Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (297 commits)
ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
ALSA: asihpi - HPI version 4.08
ALSA: asihpi - Add volume mute controls
ALSA: asihpi - Control name updates
ALSA: asihpi - Use size_t for sizeof result
ALSA: asihpi - Explicitly include mutex.h
ALSA: asihpi - Add new node and message defines
ALSA: asihpi - Make local function static
ALSA: asihpi - Fix minor typos and spelling
ALSA: asihpi - Remove unused structures, macros and functions
ALSA: asihpi - Remove spurious adapter index check
ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
ALSA: asihpi - DSP code loader API now independent of OS
ALSA: asihpi - Remove controlex structs and associated special data transfer code
ALSA: asihpi - Increase request and response buffer sizes
ALSA: asihpi - Give more meaningful name to hpi request message type
ALSA: usb-audio - Add quirk for Roland / BOSS BR-800
ALSA: hda - Remove a superfluous argument of via_auto_init_output()
ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
ALSA: hda - Add documentation for codec-specific mixer controls
...
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 58ced23..598c22f 100644
--- a/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -1164,7 +1164,7 @@
}
chip->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_mychip_interrupt,
- IRQF_SHARED, "My Chip", chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
snd_mychip_free(chip);
return -EBUSY;
@@ -1197,7 +1197,7 @@
/* pci_driver definition */
static struct pci_driver driver = {
- .name = "My Own Chip",
+ .name = KBUILD_MODNAME,
.id_table = snd_mychip_ids,
.probe = snd_mychip_probe,
.remove = __devexit_p(snd_mychip_remove),
@@ -1340,7 +1340,7 @@
<programlisting>
<![CDATA[
if (request_irq(pci->irq, snd_mychip_interrupt,
- IRQF_SHARED, "My Chip", chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
snd_mychip_free(chip);
return -EBUSY;
@@ -1616,7 +1616,7 @@
<programlisting>
<![CDATA[
static struct pci_driver driver = {
- .name = "My Own Chip",
+ .name = KBUILD_MODNAME,
.id_table = snd_mychip_ids,
.probe = snd_mychip_probe,
.remove = __devexit_p(snd_mychip_remove),
@@ -5816,7 +5816,7 @@
<programlisting>
<![CDATA[
static struct pci_driver driver = {
- .name = "My Chip",
+ .name = KBUILD_MODNAME,
.id_table = snd_my_ids,
.probe = snd_my_probe,
.remove = __devexit_p(snd_my_remove),
diff --git a/Documentation/sound/alsa/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt
new file mode 100644
index 0000000..1482035
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio-Controls.txt
@@ -0,0 +1,100 @@
+This file explains the codec-specific mixer controls.
+
+Realtek codecs
+--------------
+
+* Channel Mode
+ This is an enum control to change the surround-channel setup,
+ appears only when the surround channels are available.
+ It gives the number of channels to be used, "2ch", "4ch", "6ch",
+ and "8ch". According to the configuration, this also controls the
+ jack-retasking of multi-I/O jacks.
+
+* Auto-Mute Mode
+ This is an enum control to change the auto-mute behavior of the
+ headphone and line-out jacks. If built-in speakers and headphone
+ and/or line-out jacks are available on a machine, this controls
+ appears.
+ When there are only either headphones or line-out jacks, it gives
+ "Disabled" and "Enabled" state. When enabled, the speaker is muted
+ automatically when a jack is plugged.
+
+ When both headphone and line-out jacks are present, it gives
+ "Disabled", "Speaker Only" and "Line-Out+Speaker". When
+ speaker-only is chosen, plugging into a headphone or a line-out jack
+ mutes the speakers, but not line-outs. When line-out+speaker is
+ selected, plugging to a headphone jack mutes both speakers and
+ line-outs.
+
+
+IDT/Sigmatel codecs
+-------------------
+
+* Analog Loopback
+ This control enables/disables the analog-loopback circuit. This
+ appears only when "loopback" is set to true in a codec hint
+ (see HD-Audio.txt). Note that on some codecs the analog-loopback
+ and the normal PCM playback are exclusive, i.e. when this is on, you
+ won't hear any PCM stream.
+
+* Swap Center/LFE
+ Swaps the center and LFE channel order. Normally, the left
+ corresponds to the center and the right to the LFE. When this is
+ ON, the left to the LFE and the right to the center.
+
+* Headphone as Line Out
+ When this control is ON, treat the headphone jacks as line-out
+ jacks. That is, the headphone won't auto-mute the other line-outs,
+ and no HP-amp is set to the pins.
+
+* Mic Jack Mode, Line Jack Mode, etc
+ These enum controls the direction and the bias of the input jack
+ pins. Depending on the jack type, it can set as "Mic In" and "Line
+ In", for determining the input bias, or it can be set to "Line Out"
+ when the pin is a multi-I/O jack for surround channels.
+
+
+VIA codecs
+----------
+
+* Smart 5.1
+ An enum control to re-task the multi-I/O jacks for surround outputs.
+ When it's ON, the corresponding input jacks (usually a line-in and a
+ mic-in) are switched as the surround and the CLFE output jacks.
+
+* Independent HP
+ When this enum control is enabled, the headphone output is routed
+ from an individual stream (the third PCM such as hw:0,2) instead of
+ the primary stream. In the case the headphone DAC is shared with a
+ side or a CLFE-channel DAC, the DAC is switched to the headphone
+ automatically.
+
+* Loopback Mixing
+ An enum control to determine whether the analog-loopback route is
+ enabled or not. When it's enabled, the analog-loopback is mixed to
+ the front-channel. Also, the same route is used for the headphone
+ and speaker outputs. As a side-effect, when this mode is set, the
+ individual volume controls will be no longer available for
+ headphones and speakers because there is only one DAC connected to a
+ mixer widget.
+
+* Dynamic Power-Control
+ This control determines whether the dynamic power-control per jack
+ detection is enabled or not. When enabled, the widgets power state
+ (D0/D3) are changed dynamically depending on the jack plugging
+ state for saving power consumptions. However, if your system
+ doesn't provide a proper jack-detection, this won't work; in such a
+ case, turn this control OFF.
+
+* Jack Detect
+ This control is provided only for VT1708 codec which gives no proper
+ unsolicited event per jack plug. When this is on, the driver polls
+ the jack detection so that the headphone auto-mute can work, while
+ turning this off would reduce the power consumption.
+
+
+Conexant codecs
+---------------
+
+* Auto-Mute Mode
+ See Reatek codecs.
diff --git a/MAINTAINERS b/MAINTAINERS
index 41ec646..7521440 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -534,6 +534,8 @@
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
W: http://wiki.analog.com/
S: Supported
+F: sound/soc/codecs/adau*
+F: sound/soc/codecs/adav*
F: sound/soc/codecs/ad1*
F: sound/soc/codecs/ssm*
diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h
index 0570930..74173c5 100644
--- a/include/linux/pci_ids.h
+++ b/include/linux/pci_ids.h
@@ -1308,6 +1308,7 @@
#define PCI_SUBDEVICE_ID_CREATIVE_SB08801 0x0041
#define PCI_SUBDEVICE_ID_CREATIVE_SB08802 0x0042
#define PCI_SUBDEVICE_ID_CREATIVE_SB08803 0x0043
+#define PCI_SUBDEVICE_ID_CREATIVE_SB1270 0x0062
#define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX 0x6000
#define PCI_VENDOR_ID_ECTIVA 0x1102 /* duplicate: CREATIVE */
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index 2480e7d..6b14359 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -27,6 +27,7 @@
#include <linux/spinlock.h>
#include <linux/wait.h>
#include <linux/mutex.h>
+#include <linux/workqueue.h>
#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
#include "seq_device.h"
@@ -63,6 +64,7 @@
};
struct snd_rawmidi_runtime {
+ struct snd_rawmidi_substream *substream;
unsigned int drain: 1, /* drain stage */
oss: 1; /* OSS compatible mode */
/* midi stream buffer */
@@ -79,7 +81,7 @@
/* event handler (new bytes, input only) */
void (*event)(struct snd_rawmidi_substream *substream);
/* defers calls to event [input] or ops->trigger [output] */
- struct tasklet_struct tasklet;
+ struct work_struct event_work;
/* private data */
void *private_data;
void (*private_free)(struct snd_rawmidi_substream *substream);
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 1bafe95..5ad5f3a 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -209,6 +209,10 @@
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
unsigned int symmetric_rates:1;
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
};
/*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c46e7d8..e09505c 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -348,6 +348,8 @@
void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num);
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
@@ -429,6 +431,7 @@
/* status */
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
+ u32 weak:1; /* path ignored for power management */
int (*connected)(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink);
@@ -444,6 +447,7 @@
char *name; /* widget name */
char *sname; /* stream name */
struct snd_soc_codec *codec;
+ struct snd_soc_platform *platform;
struct list_head list;
struct snd_soc_dapm_context *dapm;
@@ -507,10 +511,11 @@
struct device *dev; /* from parent - for debug */
struct snd_soc_codec *codec; /* parent codec */
+ struct snd_soc_platform *platform; /* parent platform */
struct snd_soc_card *card; /* parent card */
/* used during DAPM updates */
- int dev_power;
+ enum snd_soc_bias_level target_bias_level;
struct list_head list;
#ifdef CONFIG_DEBUG_FS
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3a4bd3a..aa19f5a 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -203,6 +203,16 @@
SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
/*
+ * Component probe and remove ordering levels for components with runtime
+ * dependencies.
+ */
+#define SND_SOC_COMP_ORDER_FIRST -2
+#define SND_SOC_COMP_ORDER_EARLY -1
+#define SND_SOC_COMP_ORDER_NORMAL 0
+#define SND_SOC_COMP_ORDER_LATE 1
+#define SND_SOC_COMP_ORDER_LAST 2
+
+/*
* Bias levels
*
* @ON: Bias is fully on for audio playback and capture operations.
@@ -214,10 +224,10 @@
* @OFF: Power Off. No restrictions on transition times.
*/
enum snd_soc_bias_level {
- SND_SOC_BIAS_OFF,
- SND_SOC_BIAS_STANDBY,
- SND_SOC_BIAS_PREPARE,
- SND_SOC_BIAS_ON,
+ SND_SOC_BIAS_OFF = 0,
+ SND_SOC_BIAS_STANDBY = 1,
+ SND_SOC_BIAS_PREPARE = 2,
+ SND_SOC_BIAS_ON = 3,
};
struct snd_jack;
@@ -258,6 +268,11 @@
SND_SOC_RBTREE_COMPRESSION
};
+enum snd_soc_pcm_subclass {
+ SND_SOC_PCM_CLASS_PCM = 0,
+ SND_SOC_PCM_CLASS_BE = 1,
+};
+
int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
unsigned int freq, int dir);
int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -297,6 +312,10 @@
unsigned int reg);
int snd_soc_default_writable_register(struct snd_soc_codec *codec,
unsigned int reg);
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+ unsigned int reg);
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+ unsigned int reg, unsigned int val);
/* Utility functions to get clock rates from various things */
int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
@@ -349,6 +368,8 @@
const char *prefix);
int snd_soc_add_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
+ const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@@ -612,6 +633,10 @@
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
};
/* SoC platform interface */
@@ -623,10 +648,17 @@
int (*resume)(struct snd_soc_dai *dai);
/* pcm creation and destruction */
- int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
- struct snd_pcm *);
+ int (*pcm_new)(struct snd_soc_pcm_runtime *);
void (*pcm_free)(struct snd_pcm *);
+ /* Default control and setup, added after probe() is run */
+ const struct snd_kcontrol_new *controls;
+ int num_controls;
+ const struct snd_soc_dapm_widget *dapm_widgets;
+ int num_dapm_widgets;
+ const struct snd_soc_dapm_route *dapm_routes;
+ int num_dapm_routes;
+
/*
* For platform caused delay reporting.
* Optional.
@@ -636,6 +668,14 @@
/* platform stream ops */
struct snd_pcm_ops *ops;
+
+ /* probe ordering - for components with runtime dependencies */
+ int probe_order;
+ int remove_order;
+
+ /* platform IO - used for platform DAPM */
+ unsigned int (*read)(struct snd_soc_platform *, unsigned int);
+ int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
};
struct snd_soc_platform {
@@ -650,6 +690,8 @@
struct snd_soc_card *card;
struct list_head list;
struct list_head card_list;
+
+ struct snd_soc_dapm_context dapm;
};
struct snd_soc_dai_link {
@@ -725,8 +767,10 @@
/* callbacks */
int (*set_bias_level)(struct snd_soc_card *,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
int (*set_bias_level_post)(struct snd_soc_card *,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
long pmdown_time;
@@ -789,6 +833,9 @@
struct device dev;
struct snd_soc_card *card;
struct snd_soc_dai_link *dai_link;
+ struct mutex pcm_mutex;
+ enum snd_soc_pcm_subclass pcm_subclass;
+ struct snd_pcm_ops ops;
unsigned int complete:1;
unsigned int dev_registered:1;
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index ae973d2..603f5a0 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -9,6 +9,7 @@
struct snd_soc_jack;
struct snd_soc_codec;
+struct snd_soc_platform;
struct snd_soc_card;
struct snd_soc_dapm_widget;
@@ -59,6 +60,50 @@
);
+DECLARE_EVENT_CLASS(snd_soc_preg,
+
+ TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+ unsigned int val),
+
+ TP_ARGS(platform, reg, val),
+
+ TP_STRUCT__entry(
+ __string( name, platform->name )
+ __field( int, id )
+ __field( unsigned int, reg )
+ __field( unsigned int, val )
+ ),
+
+ TP_fast_assign(
+ __assign_str(name, platform->name);
+ __entry->id = platform->id;
+ __entry->reg = reg;
+ __entry->val = val;
+ ),
+
+ TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name),
+ (int)__entry->id, (unsigned int)__entry->reg,
+ (unsigned int)__entry->val)
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write,
+
+ TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+ unsigned int val),
+
+ TP_ARGS(platform, reg, val)
+
+);
+
+DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read,
+
+ TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
+ unsigned int val),
+
+ TP_ARGS(platform, reg, val)
+
+);
+
DECLARE_EVENT_CLASS(snd_soc_card,
TP_PROTO(struct snd_soc_card *card, int val),
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index cbbed0d..849a0ed 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -92,16 +92,12 @@
(!substream->append || runtime->avail >= count);
}
-static void snd_rawmidi_input_event_tasklet(unsigned long data)
+static void snd_rawmidi_input_event_work(struct work_struct *work)
{
- struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
- substream->runtime->event(substream);
-}
-
-static void snd_rawmidi_output_trigger_tasklet(unsigned long data)
-{
- struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
- substream->ops->trigger(substream, 1);
+ struct snd_rawmidi_runtime *runtime =
+ container_of(work, struct snd_rawmidi_runtime, event_work);
+ if (runtime->event)
+ runtime->event(runtime->substream);
}
static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
@@ -110,16 +106,10 @@
if ((runtime = kzalloc(sizeof(*runtime), GFP_KERNEL)) == NULL)
return -ENOMEM;
+ runtime->substream = substream;
spin_lock_init(&runtime->lock);
init_waitqueue_head(&runtime->sleep);
- if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT)
- tasklet_init(&runtime->tasklet,
- snd_rawmidi_input_event_tasklet,
- (unsigned long)substream);
- else
- tasklet_init(&runtime->tasklet,
- snd_rawmidi_output_trigger_tasklet,
- (unsigned long)substream);
+ INIT_WORK(&runtime->event_work, snd_rawmidi_input_event_work);
runtime->event = NULL;
runtime->buffer_size = PAGE_SIZE;
runtime->avail_min = 1;
@@ -150,12 +140,7 @@
{
if (!substream->opened)
return;
- if (up) {
- tasklet_schedule(&substream->runtime->tasklet);
- } else {
- tasklet_kill(&substream->runtime->tasklet);
- substream->ops->trigger(substream, 0);
- }
+ substream->ops->trigger(substream, up);
}
static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
@@ -163,8 +148,8 @@
if (!substream->opened)
return;
substream->ops->trigger(substream, up);
- if (!up && substream->runtime->event)
- tasklet_kill(&substream->runtime->tasklet);
+ if (!up)
+ cancel_work_sync(&substream->runtime->event_work);
}
int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream)
@@ -641,10 +626,10 @@
return -EINVAL;
}
if (params->buffer_size != runtime->buffer_size) {
- newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
+ newbuf = krealloc(runtime->buffer, params->buffer_size,
+ GFP_KERNEL);
if (!newbuf)
return -ENOMEM;
- kfree(runtime->buffer);
runtime->buffer = newbuf;
runtime->buffer_size = params->buffer_size;
runtime->avail = runtime->buffer_size;
@@ -668,10 +653,10 @@
return -EINVAL;
}
if (params->buffer_size != runtime->buffer_size) {
- newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
+ newbuf = krealloc(runtime->buffer, params->buffer_size,
+ GFP_KERNEL);
if (!newbuf)
return -ENOMEM;
- kfree(runtime->buffer);
runtime->buffer = newbuf;
runtime->buffer_size = params->buffer_size;
}
@@ -926,7 +911,7 @@
}
if (result > 0) {
if (runtime->event)
- tasklet_schedule(&runtime->tasklet);
+ schedule_work(&runtime->event_work);
else if (snd_rawmidi_ready(substream))
wake_up(&runtime->sleep);
}
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 5466de8..3fc257d 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -171,7 +171,7 @@
err = snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_PERIOD_TIME,
- 5000, 8192000);
+ 5000, UINT_MAX);
if (err < 0)
return err;
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index d8f6fd6..2015036 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -944,7 +944,7 @@
spin_lock_init(&chip->lock); /* only now can we call ad1889_free */
if (request_irq(pci->irq, snd_ad1889_interrupt,
- IRQF_SHARED, card->driver, chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR PFX "cannot obtain IRQ %d\n", pci->irq);
snd_ad1889_free(chip);
return -EBUSY;
@@ -1055,7 +1055,7 @@
MODULE_DEVICE_TABLE(pci, snd_ad1889_ids);
static struct pci_driver ad1889_pci_driver = {
- .name = "AD1889 Audio",
+ .name = KBUILD_MODNAME,
.id_table = snd_ad1889_ids,
.probe = snd_ad1889_probe,
.remove = __devexit_p(snd_ad1889_remove),
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 5c6e322..b444b74 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -2090,7 +2090,7 @@
codec->port = pci_resource_start(codec->pci, 0);
if (request_irq(codec->pci->irq, snd_ali_card_interrupt,
- IRQF_SHARED, "ALI 5451", codec)) {
+ IRQF_SHARED, KBUILD_MODNAME, codec)) {
snd_printk(KERN_ERR "Unable to request irq.\n");
return -EBUSY;
}
@@ -2295,7 +2295,7 @@
}
static struct pci_driver driver = {
- .name = "ALI 5451",
+ .name = KBUILD_MODNAME,
.id_table = snd_ali_ids,
.probe = snd_ali_probe,
.remove = __devexit_p(snd_ali_remove),
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index d7653cb..736c8e9 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -722,7 +722,7 @@
irq_handler = snd_als300_interrupt;
if (request_irq(pci->irq, irq_handler, IRQF_SHARED,
- card->shortname, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_als300_free(chip);
return -EBUSY;
@@ -846,7 +846,7 @@
}
static struct pci_driver driver = {
- .name = "ALS300",
+ .name = KBUILD_MODNAME,
.id_table = snd_als300_ids,
.probe = snd_als300_probe,
.remove = __devexit_p(snd_als300_remove),
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 0e247cb..a9c1af3 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -1036,7 +1036,7 @@
static struct pci_driver driver = {
- .name = "ALS4000",
+ .name = KBUILD_MODNAME,
.id_table = snd_als4000_ids,
.probe = snd_card_als4000_probe,
.remove = __devexit_p(snd_card_als4000_remove),
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index e3569bd..b941d25 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -49,19 +49,21 @@
#if defined CONFIG_SND_DEBUG
/* copied from pcm_lib.c, hope later patch will make that version public
and this copy can be removed */
-static void pcm_debug_name(struct snd_pcm_substream *substream,
- char *name, size_t len)
+static inline void
+snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
{
- snprintf(name, len, "pcmC%dD%d%c:%d",
+ snprintf(buf, size, "pcmC%dD%d%c:%d",
substream->pcm->card->number,
substream->pcm->device,
substream->stream ? 'c' : 'p',
substream->number);
}
-#define DEBUG_NAME(substream, name) char name[16]; pcm_debug_name(substream, name, sizeof(name))
#else
-#define pcm_debug_name(s, n, l) do { } while (0)
-#define DEBUG_NAME(name, substream) do { } while (0)
+static inline void
+snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
+{
+ *buf = 0;
+}
#endif
#if defined CONFIG_SND_DEBUG_VERBOSE
@@ -304,7 +306,8 @@
static void print_hwparams(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *p)
{
- DEBUG_NAME(substream, name);
+ char name[16];
+ snd_pcm_debug_name(substream, name, sizeof(name));
snd_printd("%s HWPARAMS\n", name);
snd_printd(" samplerate %d Hz\n", params_rate(p));
snd_printd(" channels %d\n", params_channels(p));
@@ -576,8 +579,9 @@
struct snd_card_asihpi *card = snd_pcm_substream_chip(substream);
struct snd_pcm_substream *s;
u16 e;
- DEBUG_NAME(substream, name);
+ char name[16];
+ snd_pcm_debug_name(substream, name, sizeof(name));
snd_printdd("%s trigger\n", name);
switch (cmd) {
@@ -741,7 +745,9 @@
int loops = 0;
u16 state;
u32 buffer_size, bytes_avail, samples_played, on_card_bytes;
- DEBUG_NAME(substream, name);
+ char name[16];
+
+ snd_pcm_debug_name(substream, name, sizeof(name));
snd_printdd("%s snd_card_asihpi_timer_function\n", name);
@@ -1323,10 +1329,12 @@
"RF",
"Clock",
"Bitstream",
- "Microphone",
- "Cobranet",
+ "Mic",
+ "Net",
"Analog",
"Adapter",
+ "RTP",
+ "GPI",
};
compile_time_assert(
@@ -1341,8 +1349,10 @@
"Digital",
"RF",
"Speaker",
- "Cobranet Out",
- "Analog"
+ "Net",
+ "Analog",
+ "RTP",
+ "GPO",
};
compile_time_assert(
@@ -1476,11 +1486,40 @@
static const DECLARE_TLV_DB_SCALE(db_scale_100, -10000, VOL_STEP_mB, 0);
+#define snd_asihpi_volume_mute_info snd_ctl_boolean_mono_info
+
+static int snd_asihpi_volume_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 h_control = kcontrol->private_value;
+ u32 mute;
+
+ hpi_handle_error(hpi_volume_get_mute(h_control, &mute));
+ ucontrol->value.integer.value[0] = mute ? 0 : 1;
+
+ return 0;
+}
+
+static int snd_asihpi_volume_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u32 h_control = kcontrol->private_value;
+ int change = 1;
+ /* HPI currently only supports all or none muting of multichannel volume
+ ALSA Switch element has opposite sense to HPI mute: on==unmuted, off=muted
+ */
+ int mute = ucontrol->value.integer.value[0] ? 0 : HPI_BITMASK_ALL_CHANNELS;
+ hpi_handle_error(hpi_volume_set_mute(h_control, mute));
+ return change;
+}
+
static int __devinit snd_asihpi_volume_add(struct snd_card_asihpi *asihpi,
struct hpi_control *hpi_ctl)
{
struct snd_card *card = asihpi->card;
struct snd_kcontrol_new snd_control;
+ int err;
+ u32 mute;
asihpi_ctl_init(&snd_control, hpi_ctl, "Volume");
snd_control.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
@@ -1490,7 +1529,19 @@
snd_control.put = snd_asihpi_volume_put;
snd_control.tlv.p = db_scale_100;
- return ctl_add(card, &snd_control, asihpi);
+ err = ctl_add(card, &snd_control, asihpi);
+ if (err)
+ return err;
+
+ if (hpi_volume_get_mute(hpi_ctl->h_control, &mute) == 0) {
+ asihpi_ctl_init(&snd_control, hpi_ctl, "Switch");
+ snd_control.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ snd_control.info = snd_asihpi_volume_mute_info;
+ snd_control.get = snd_asihpi_volume_mute_get;
+ snd_control.put = snd_asihpi_volume_mute_put;
+ err = ctl_add(card, &snd_control, asihpi);
+ }
+ return err;
}
/*------------------------------------------------------------
@@ -2923,7 +2974,7 @@
MODULE_DEVICE_TABLE(pci, asihpi_pci_tbl);
static struct pci_driver driver = {
- .name = "asihpi",
+ .name = KBUILD_MODNAME,
.id_table = asihpi_pci_tbl,
.probe = snd_asihpi_probe,
.remove = __devexit_p(snd_asihpi_remove),
diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index 255429c..f207272 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -1,7 +1,7 @@
/******************************************************************************
AudioScience HPI driver
- Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
+ Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@@ -42,12 +42,11 @@
#define HPI_VER_MINOR(v) ((int)((v >> 8) & 0xFF))
#define HPI_VER_RELEASE(v) ((int)(v & 0xFF))
-/* Use single digits for versions less that 10 to avoid octal. */
-#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 6, 0)
-#define HPI_VER_STRING "4.06.00"
+#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 8, 0)
+#define HPI_VER_STRING "4.08.00"
/* Library version as documented in hpi-api-versions.txt */
-#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(9, 0, 0)
+#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(10, 0, 0)
#include <linux/types.h>
#define HPI_BUILD_EXCLUDE_DEPRECATED
@@ -211,8 +210,12 @@
HPI_SOURCENODE_COBRANET = 109,
HPI_SOURCENODE_ANALOG = 110, /**< analog input node. */
HPI_SOURCENODE_ADAPTER = 111, /**< adapter node. */
+ /** RTP stream input node - This node is a destination for
+ packets of RTP audio samples from other devices. */
+ HPI_SOURCENODE_RTP_DESTINATION = 112,
+ HPI_SOURCENODE_GP_IN = 113, /**< general purpose input. */
/* !!!Update this AND hpidebug.h if you add a new sourcenode type!!! */
- HPI_SOURCENODE_LAST_INDEX = 111 /**< largest ID */
+ HPI_SOURCENODE_LAST_INDEX = 113 /**< largest ID */
/* AX6 max sourcenode types = 15 */
};
@@ -228,7 +231,7 @@
HPI_DESTNODE_NONE = 200,
/** In Stream (Record) node. */
HPI_DESTNODE_ISTREAM = 201,
- HPI_DESTNODE_LINEOUT = 202, /**< line out node. */
+ HPI_DESTNODE_LINEOUT = 202, /**< line out node. */
HPI_DESTNODE_AESEBU_OUT = 203, /**< AES/EBU output node. */
HPI_DESTNODE_RF = 204, /**< RF output node. */
HPI_DESTNODE_SPEAKER = 205, /**< speaker output node. */
@@ -236,9 +239,12 @@
Audio samples from the device are sent out on the Cobranet network.*/
HPI_DESTNODE_COBRANET = 206,
HPI_DESTNODE_ANALOG = 207, /**< analog output node. */
-
+ /** RTP stream output node - This node is a source for
+ packets of RTP audio samples that are sent to other devices. */
+ HPI_DESTNODE_RTP_SOURCE = 208,
+ HPI_DESTNODE_GP_OUT = 209, /**< general purpose output node. */
/* !!!Update this AND hpidebug.h if you add a new destnode type!!! */
- HPI_DESTNODE_LAST_INDEX = 207 /**< largest ID */
+ HPI_DESTNODE_LAST_INDEX = 209 /**< largest ID */
/* AX6 max destnode types = 15 */
};
diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c
index df4aed5..3cc6f11 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -359,7 +359,7 @@
HPI_ERROR_PROCESSING_MESSAGE);
switch (phm->type) {
- case HPI_TYPE_MESSAGE:
+ case HPI_TYPE_REQUEST:
switch (phm->object) {
case HPI_OBJ_SUBSYSTEM:
subsys_message(phm, phr);
@@ -538,7 +538,7 @@
HPI_DEBUG_LOG(VERBOSE, "send ADAPTER_GET_INFO\n");
memset(&hm, 0, sizeof(hm));
- hm.type = HPI_TYPE_MESSAGE;
+ hm.type = HPI_TYPE_REQUEST;
hm.size = sizeof(struct hpi_message);
hm.object = HPI_OBJ_ADAPTER;
hm.function = HPI_ADAPTER_GET_INFO;
@@ -946,11 +946,8 @@
}
/* write the DSP code down into the DSPs memory */
- /*HpiDspCode_Open(nBootLoadFamily,&DspCode,pdwOsErrorCode); */
- dsp_code.ps_dev = pao->pci.pci_dev;
-
- error = hpi_dsp_code_open(boot_load_family, &dsp_code,
- pos_error_code);
+ error = hpi_dsp_code_open(boot_load_family, pao->pci.pci_dev,
+ &dsp_code, pos_error_code);
if (error)
return error;
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 9d5df54..e041a6a 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -373,6 +373,7 @@
/** Entry point to this HPI backend
* All calls to the HPI start here
*/
+static
void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
struct hpi_response *phr)
{
@@ -392,7 +393,7 @@
HPI_DEBUG_LOG(VERBOSE, "start of switch\n");
switch (phm->type) {
- case HPI_TYPE_MESSAGE:
+ case HPI_TYPE_REQUEST:
switch (phm->object) {
case HPI_OBJ_SUBSYSTEM:
subsys_message(pao, phm, phr);
@@ -402,7 +403,6 @@
adapter_message(pao, phm, phr);
break;
- case HPI_OBJ_CONTROLEX:
case HPI_OBJ_CONTROL:
control_message(pao, phm, phr);
break;
@@ -634,11 +634,12 @@
HPI_DEBUG_LOG(VERBOSE, "init ADAPTER_GET_INFO\n");
memset(&hm, 0, sizeof(hm));
- hm.type = HPI_TYPE_MESSAGE;
+ /* wAdapterIndex == version == 0 */
+ hm.type = HPI_TYPE_REQUEST;
hm.size = sizeof(hm);
hm.object = HPI_OBJ_ADAPTER;
hm.function = HPI_ADAPTER_GET_INFO;
- hm.adapter_index = 0;
+
memset(&hr, 0, sizeof(hr));
hr.size = sizeof(hr);
@@ -658,9 +659,6 @@
hr.u.ax.info.num_outstreams +
hr.u.ax.info.num_instreams;
- hpios_locked_mem_prepare((max_streams * 6) / 10, max_streams,
- 65536, pao->pci.pci_dev);
-
HPI_DEBUG_LOG(VERBOSE,
"got adapter info type %x index %d serial %d\n",
hr.u.ax.info.adapter_type, hr.u.ax.info.adapter_index,
@@ -709,9 +707,6 @@
[i]);
phw->outstream_host_buffer_size[i] = 0;
}
-
- hpios_locked_mem_unprepare(pao->pci.pci_dev);
-
kfree(phw);
}
@@ -1371,9 +1366,8 @@
return err;
/* write the DSP code down into the DSPs memory */
- dsp_code.ps_dev = pao->pci.pci_dev;
- err = hpi_dsp_code_open(boot_code_id[dsp], &dsp_code,
- pos_error_code);
+ err = hpi_dsp_code_open(boot_code_id[dsp], pao->pci.pci_dev,
+ &dsp_code, pos_error_code);
if (err)
return err;
@@ -2084,13 +2078,13 @@
u16 err = 0;
message_count++;
- if (phm->size > sizeof(interface->u)) {
+ if (phm->size > sizeof(interface->u.message_buffer)) {
phr->error = HPI_ERROR_MESSAGE_BUFFER_TOO_SMALL;
- phr->specific_error = sizeof(interface->u);
+ phr->specific_error = sizeof(interface->u.message_buffer);
phr->size = sizeof(struct hpi_response_header);
HPI_DEBUG_LOG(ERROR,
"message len %d too big for buffer %zd \n", phm->size,
- sizeof(interface->u));
+ sizeof(interface->u.message_buffer));
return 0;
}
@@ -2122,18 +2116,19 @@
/* read the result */
if (time_out) {
- if (interface->u.response_buffer.size <= phr->size)
+ if (interface->u.response_buffer.response.size <= phr->size)
memcpy(phr, &interface->u.response_buffer,
- interface->u.response_buffer.size);
+ interface->u.response_buffer.response.size);
else {
HPI_DEBUG_LOG(ERROR,
"response len %d too big for buffer %d\n",
- interface->u.response_buffer.size, phr->size);
+ interface->u.response_buffer.response.size,
+ phr->size);
memcpy(phr, &interface->u.response_buffer,
sizeof(struct hpi_response_header));
phr->error = HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
phr->specific_error =
- interface->u.response_buffer.size;
+ interface->u.response_buffer.response.size;
phr->size = sizeof(struct hpi_response_header);
}
}
@@ -2202,23 +2197,6 @@
phm->u.d.u.data.data_size, H620_HIF_GET_DATA);
break;
- case HPI_CONTROL_SET_STATE:
- if (phm->object == HPI_OBJ_CONTROLEX
- && phm->u.cx.attribute == HPI_COBRANET_SET_DATA)
- err = hpi6205_transfer_data(pao,
- phm->u.cx.u.cobranet_bigdata.pb_data,
- phm->u.cx.u.cobranet_bigdata.byte_count,
- H620_HIF_SEND_DATA);
- break;
-
- case HPI_CONTROL_GET_STATE:
- if (phm->object == HPI_OBJ_CONTROLEX
- && phm->u.cx.attribute == HPI_COBRANET_GET_DATA)
- err = hpi6205_transfer_data(pao,
- phm->u.cx.u.cobranet_bigdata.pb_data,
- phr->u.cx.u.cobranet_data.byte_count,
- H620_HIF_GET_DATA);
- break;
}
phr->error = err;
diff --git a/sound/pci/asihpi/hpi6205.h b/sound/pci/asihpi/hpi6205.h
index df2f02c..ec0827b 100644
--- a/sound/pci/asihpi/hpi6205.h
+++ b/sound/pci/asihpi/hpi6205.h
@@ -1,7 +1,7 @@
/*****************************************************************************
AudioScience HPI driver
- Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
+ Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@@ -70,15 +70,28 @@
in and out of.
************************************************************/
#define HPI6205_SIZEOF_DATA (16*1024)
+
+struct message_buffer_6205 {
+ struct hpi_message message;
+ char data[256];
+};
+
+struct response_buffer_6205 {
+ struct hpi_response response;
+ char data[256];
+};
+
+union buffer_6205 {
+ struct message_buffer_6205 message_buffer;
+ struct response_buffer_6205 response_buffer;
+ u8 b_data[HPI6205_SIZEOF_DATA];
+};
+
struct bus_master_interface {
u32 host_cmd;
u32 dsp_ack;
u32 transfer_size_in_bytes;
- union {
- struct hpi_message_header message_buffer;
- struct hpi_response_header response_buffer;
- u8 b_data[HPI6205_SIZEOF_DATA];
- } u;
+ union buffer_6205 u;
struct controlcache_6205 control_cache;
struct async_event_buffer_6205 async_buffer;
struct hpi_hostbuffer_status
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index bf5eced..d497030 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -1,7 +1,7 @@
/******************************************************************************
AudioScience HPI driver
- Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
+ Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@@ -32,12 +32,6 @@
#include "hpios.h"
/* physical memory allocation */
-void hpios_locked_mem_init(void
- );
-void hpios_locked_mem_free_all(void
- );
-#define hpios_locked_mem_prepare(a, b, c, d);
-#define hpios_locked_mem_unprepare(a)
/** Allocate and map an area of locked memory for bus master DMA operations.
@@ -226,8 +220,8 @@
HPI_COBRANET_SET = HPI_CTL_ATTR(COBRANET, 1),
HPI_COBRANET_GET = HPI_CTL_ATTR(COBRANET, 2),
- HPI_COBRANET_SET_DATA = HPI_CTL_ATTR(COBRANET, 3),
- HPI_COBRANET_GET_DATA = HPI_CTL_ATTR(COBRANET, 4),
+ /*HPI_COBRANET_SET_DATA = HPI_CTL_ATTR(COBRANET, 3), */
+ /*HPI_COBRANET_GET_DATA = HPI_CTL_ATTR(COBRANET, 4), */
HPI_COBRANET_GET_STATUS = HPI_CTL_ATTR(COBRANET, 5),
HPI_COBRANET_SEND_PACKET = HPI_CTL_ATTR(COBRANET, 6),
HPI_COBRANET_GET_PACKET = HPI_CTL_ATTR(COBRANET, 7),
@@ -364,10 +358,12 @@
#define HPI_ADAPTER_ASI(f) (f)
enum HPI_MESSAGE_TYPES {
- HPI_TYPE_MESSAGE = 1,
+ HPI_TYPE_REQUEST = 1,
HPI_TYPE_RESPONSE = 2,
HPI_TYPE_DATA = 3,
- HPI_TYPE_SSX2BYPASS_MESSAGE = 4
+ HPI_TYPE_SSX2BYPASS_MESSAGE = 4,
+ HPI_TYPE_COMMAND = 5,
+ HPI_TYPE_NOTIFICATION = 6
};
enum HPI_OBJECT_TYPES {
@@ -383,7 +379,7 @@
HPI_OBJ_WATCHDOG = 10,
HPI_OBJ_CLOCK = 11,
HPI_OBJ_PROFILE = 12,
- HPI_OBJ_CONTROLEX = 13,
+ /* HPI_ OBJ_ CONTROLEX = 13, */
HPI_OBJ_ASYNCEVENT = 14
#define HPI_OBJ_MAXINDEX 14
};
@@ -608,7 +604,7 @@
#endif
struct hpi_buffer {
- /** placehoder for backward compatibility (see dwBufferSize) */
+ /** placeholder for backward compatibility (see dwBufferSize) */
struct hpi_msg_format reserved;
u32 command; /**< HPI_BUFFER_CMD_xxx*/
u32 pci_address; /**< PCI physical address of buffer for DSP DMA */
@@ -912,95 +908,13 @@
u32 remaining_chars;
} chars8;
char c_data12[12];
-};
-
-/* HPI_CONTROLX_STRUCTURES */
-
-/* Message */
-
-/** Used for all HMI variables where max length <= 8 bytes
-*/
-struct hpi_controlx_msg_cobranet_data {
- u32 hmi_address;
- u32 byte_count;
- u32 data[2];
-};
-
-/** Used for string data, and for packet bridge
-*/
-struct hpi_controlx_msg_cobranet_bigdata {
- u32 hmi_address;
- u32 byte_count;
- u8 *pb_data;
-#ifndef HPI64BIT
- u32 padding;
-#endif
-};
-
-/** Used for PADS control reading of string fields.
-*/
-struct hpi_controlx_msg_pad_data {
- u32 field;
- u32 byte_count;
- u8 *pb_data;
-#ifndef HPI64BIT
- u32 padding;
-#endif
-};
-
-/** Used for generic data
-*/
-
-struct hpi_controlx_msg_generic {
- u32 param1;
- u32 param2;
-};
-
-struct hpi_controlx_msg {
- u16 attribute; /* control attribute or property */
- u16 saved_index;
union {
- struct hpi_controlx_msg_cobranet_data cobranet_data;
- struct hpi_controlx_msg_cobranet_bigdata cobranet_bigdata;
- struct hpi_controlx_msg_generic generic;
- struct hpi_controlx_msg_pad_data pad_data;
- /*struct param_value universal_value; */
- /* nothing extra to send for status read */
- } u;
-};
-
-/* Response */
-/**
-*/
-struct hpi_controlx_res_cobranet_data {
- u32 byte_count;
- u32 data[2];
-};
-
-struct hpi_controlx_res_cobranet_bigdata {
- u32 byte_count;
-};
-
-struct hpi_controlx_res_cobranet_status {
- u32 status;
- u32 readable_size;
- u32 writeable_size;
-};
-
-struct hpi_controlx_res_generic {
- u32 param1;
- u32 param2;
-};
-
-struct hpi_controlx_res {
- union {
- struct hpi_controlx_res_cobranet_bigdata cobranet_bigdata;
- struct hpi_controlx_res_cobranet_data cobranet_data;
- struct hpi_controlx_res_cobranet_status cobranet_status;
- struct hpi_controlx_res_generic generic;
- /*struct param_info universal_info; */
- /*struct param_value universal_value; */
- } u;
+ struct {
+ u32 status;
+ u32 readable_size;
+ u32 writeable_size;
+ } status;
+ } cobranet;
};
struct hpi_nvmemory_msg {
@@ -1126,7 +1040,6 @@
/* identical to struct hpi_control_msg,
but field naming is improved */
struct hpi_control_union_msg cu;
- struct hpi_controlx_msg cx; /* extended mixer control; */
struct hpi_nvmemory_msg n;
struct hpi_gpio_msg l; /* digital i/o */
struct hpi_watchdog_msg w;
@@ -1151,7 +1064,7 @@
sizeof(struct hpi_message_header) + sizeof(struct hpi_watchdog_msg),\
sizeof(struct hpi_message_header) + sizeof(struct hpi_clock_msg),\
sizeof(struct hpi_message_header) + sizeof(struct hpi_profile_msg),\
- sizeof(struct hpi_message_header) + sizeof(struct hpi_controlx_msg),\
+ sizeof(struct hpi_message_header), /* controlx obj removed */ \
sizeof(struct hpi_message_header) + sizeof(struct hpi_async_msg) \
}
@@ -1188,7 +1101,6 @@
struct hpi_control_res c; /* mixer control; */
/* identical to hpi_control_res, but field naming is improved */
union hpi_control_union_res cu;
- struct hpi_controlx_res cx; /* extended mixer control; */
struct hpi_nvmemory_res n;
struct hpi_gpio_res l; /* digital i/o */
struct hpi_watchdog_res w;
@@ -1213,7 +1125,7 @@
sizeof(struct hpi_response_header) + sizeof(struct hpi_watchdog_res),\
sizeof(struct hpi_response_header) + sizeof(struct hpi_clock_res),\
sizeof(struct hpi_response_header) + sizeof(struct hpi_profile_res),\
- sizeof(struct hpi_response_header) + sizeof(struct hpi_controlx_res),\
+ sizeof(struct hpi_response_header), /* controlx obj removed */ \
sizeof(struct hpi_response_header) + sizeof(struct hpi_async_res) \
}
@@ -1308,6 +1220,30 @@
u8 bytes[256];
};
+struct hpi_msg_cobranet_hmi {
+ u16 attribute;
+ u16 padding;
+ u32 hmi_address;
+ u32 byte_count;
+};
+
+struct hpi_msg_cobranet_hmiwrite {
+ struct hpi_message_header h;
+ struct hpi_msg_cobranet_hmi p;
+ u8 bytes[256];
+};
+
+struct hpi_msg_cobranet_hmiread {
+ struct hpi_message_header h;
+ struct hpi_msg_cobranet_hmi p;
+};
+
+struct hpi_res_cobranet_hmiread {
+ struct hpi_response_header h;
+ u32 byte_count;
+ u8 bytes[256];
+};
+
#if 1
#define hpi_message_header_v1 hpi_message_header
#define hpi_response_header_v1 hpi_response_header
@@ -1338,7 +1274,6 @@
union hpi_mixerx_msg mx;
struct hpi_control_msg c;
struct hpi_control_union_msg cu;
- struct hpi_controlx_msg cx;
struct hpi_nvmemory_msg n;
struct hpi_gpio_msg l;
struct hpi_watchdog_msg w;
@@ -1358,7 +1293,6 @@
union hpi_mixerx_res mx;
struct hpi_control_res c;
union hpi_control_union_res cu;
- struct hpi_controlx_res cx;
struct hpi_nvmemory_res n;
struct hpi_gpio_res l;
struct hpi_watchdog_res w;
@@ -1493,12 +1427,6 @@
char temp_padding[6];
};
-struct hpi_control_cache_generic {
- struct hpi_control_cache_info i;
- u32 dw1;
- u32 dw2;
-};
-
struct hpi_control_cache_single {
union {
struct hpi_control_cache_info i;
@@ -1514,7 +1442,6 @@
struct hpi_control_cache_silencedetector silence;
struct hpi_control_cache_sampleclock clk;
struct hpi_control_cache_microphone microphone;
- struct hpi_control_cache_generic generic;
} u;
};
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index b15a02e..65b7ca1 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -57,7 +57,7 @@
}
if (phr->function != phm->function) {
- HPI_DEBUG_LOG(ERROR, "header type %d invalid\n",
+ HPI_DEBUG_LOG(ERROR, "header function %d invalid\n",
phr->function);
return HPI_ERROR_INVALID_RESPONSE;
}
@@ -315,8 +315,7 @@
short found = 1;
struct hpi_control_cache_info *pI;
struct hpi_control_cache_single *pC;
- struct hpi_control_cache_pad *p_pad;
-
+ size_t response_size;
if (!find_control(phm->obj_index, p_cache, &pI)) {
HPI_DEBUG_LOG(VERBOSE,
"HPICMN find_control() failed for adap %d\n",
@@ -326,11 +325,15 @@
phr->error = 0;
+ /* set the default response size */
+ response_size =
+ sizeof(struct hpi_response_header) +
+ sizeof(struct hpi_control_res);
+
/* pC is the default cached control strucure. May be cast to
something else in the following switch statement.
*/
pC = (struct hpi_control_cache_single *)pI;
- p_pad = (struct hpi_control_cache_pad *)pI;
switch (pI->control_type) {
@@ -529,9 +532,7 @@
pI->control_index, pI->control_type, phm->u.c.attribute);
if (found)
- phr->size =
- sizeof(struct hpi_response_header) +
- sizeof(struct hpi_control_res);
+ phr->size = (u16)response_size;
return found;
}
@@ -682,7 +683,7 @@
void HPI_COMMON(struct hpi_message *phm, struct hpi_response *phr)
{
switch (phm->type) {
- case HPI_TYPE_MESSAGE:
+ case HPI_TYPE_REQUEST:
switch (phm->object) {
case HPI_OBJ_SUBSYSTEM:
subsys_message(phm, phr);
diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c
index 5c6ea11..3a7afa3 100644
--- a/sound/pci/asihpi/hpidspcd.c
+++ b/sound/pci/asihpi/hpidspcd.c
@@ -1,8 +1,8 @@
/***********************************************************************/
-/*!
+/**
AudioScience HPI driver
- Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
+ Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@@ -18,90 +18,59 @@
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
\file
-Functions for reading DSP code to load into DSP
-
-(Linux only:) If DSPCODE_FIRMWARE_LOADER is defined, code is read using
+Functions for reading DSP code using
hotplug firmware loader from individual dsp code files
-
-If neither of the above is defined, code is read from linked arrays.
-DSPCODE_ARRAY is defined.
-
-HPI_INCLUDE_**** must be defined
-and the appropriate hzz?????.c or hex?????.c linked in
-
- */
+*/
/***********************************************************************/
#define SOURCEFILE_NAME "hpidspcd.c"
#include "hpidspcd.h"
#include "hpidebug.h"
-/**
- Header structure for binary dsp code file (see asidsp.doc)
- This structure must match that used in s2bin.c for generation of asidsp.bin
- */
-
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(push, 1)
-#endif
-
-struct code_header {
- u32 size;
- char type[4];
- u32 adapter;
- u32 version;
- u32 crc;
+struct dsp_code_private {
+ /** Firmware descriptor */
+ const struct firmware *firmware;
+ struct pci_dev *dev;
};
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(pop)
-#endif
-
#define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \
HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER)))
-/***********************************************************************/
-#include <linux/pci.h>
/*-------------------------------------------------------------------*/
-short hpi_dsp_code_open(u32 adapter, struct dsp_code *ps_dsp_code,
- u32 *pos_error_code)
+short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code,
+ u32 *os_error_code)
{
- const struct firmware *ps_firmware = ps_dsp_code->ps_firmware;
+ const struct firmware *firmware;
+ struct pci_dev *dev = os_data;
struct code_header header;
char fw_name[20];
int err;
sprintf(fw_name, "asihpi/dsp%04x.bin", adapter);
- err = request_firmware(&ps_firmware, fw_name,
- &ps_dsp_code->ps_dev->dev);
+ err = request_firmware(&firmware, fw_name, &dev->dev);
- if (err != 0) {
- dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
+ if (err || !firmware) {
+ dev_printk(KERN_ERR, &dev->dev,
"%d, request_firmware failed for %s\n", err,
fw_name);
goto error1;
}
- if (ps_firmware->size < sizeof(header)) {
- dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
- "Header size too small %s\n", fw_name);
+ if (firmware->size < sizeof(header)) {
+ dev_printk(KERN_ERR, &dev->dev, "Header size too small %s\n",
+ fw_name);
goto error2;
}
- memcpy(&header, ps_firmware->data, sizeof(header));
- if (header.adapter != adapter) {
- dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
- "Adapter type incorrect %4x != %4x\n", header.adapter,
- adapter);
- goto error2;
- }
- if (header.size != ps_firmware->size) {
- dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
- "Code size wrong %d != %ld\n", header.size,
- (unsigned long)ps_firmware->size);
+ memcpy(&header, firmware->data, sizeof(header));
+
+ if ((header.type != 0x45444F43) || /* "CODE" */
+ (header.adapter != adapter)
+ || (header.size != firmware->size)) {
+ dev_printk(KERN_ERR, &dev->dev, "Invalid firmware file\n");
goto error2;
}
- if (header.version / 100 != HPI_VER_DECIMAL / 100) {
- dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
+ if ((header.version / 100 & ~1) != (HPI_VER_DECIMAL / 100 & ~1)) {
+ dev_printk(KERN_ERR, &dev->dev,
"Incompatible firmware version "
"DSP image %d != Driver %d\n", header.version,
HPI_VER_DECIMAL);
@@ -109,67 +78,70 @@
}
if (header.version != HPI_VER_DECIMAL) {
- dev_printk(KERN_WARNING, &ps_dsp_code->ps_dev->dev,
+ dev_printk(KERN_WARNING, &dev->dev,
"Firmware: release version mismatch DSP image %d != Driver %d\n",
header.version, HPI_VER_DECIMAL);
}
HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name);
- ps_dsp_code->ps_firmware = ps_firmware;
- ps_dsp_code->block_length = header.size / sizeof(u32);
- ps_dsp_code->word_count = sizeof(header) / sizeof(u32);
- ps_dsp_code->version = header.version;
- ps_dsp_code->crc = header.crc;
+ dsp_code->pvt = kmalloc(sizeof(*dsp_code->pvt), GFP_KERNEL);
+ if (!dsp_code->pvt)
+ return HPI_ERROR_MEMORY_ALLOC;
+
+ dsp_code->pvt->dev = dev;
+ dsp_code->pvt->firmware = firmware;
+ dsp_code->header = header;
+ dsp_code->block_length = header.size / sizeof(u32);
+ dsp_code->word_count = sizeof(header) / sizeof(u32);
return 0;
error2:
- release_firmware(ps_firmware);
+ release_firmware(firmware);
error1:
- ps_dsp_code->ps_firmware = NULL;
- ps_dsp_code->block_length = 0;
+ dsp_code->block_length = 0;
return HPI_ERROR_DSP_FILE_NOT_FOUND;
}
/*-------------------------------------------------------------------*/
-void hpi_dsp_code_close(struct dsp_code *ps_dsp_code)
+void hpi_dsp_code_close(struct dsp_code *dsp_code)
{
- if (ps_dsp_code->ps_firmware != NULL) {
+ if (dsp_code->pvt->firmware) {
HPI_DEBUG_LOG(DEBUG, "dsp code closed\n");
- release_firmware(ps_dsp_code->ps_firmware);
- ps_dsp_code->ps_firmware = NULL;
+ release_firmware(dsp_code->pvt->firmware);
+ dsp_code->pvt->firmware = NULL;
}
+ kfree(dsp_code->pvt);
}
/*-------------------------------------------------------------------*/
-void hpi_dsp_code_rewind(struct dsp_code *ps_dsp_code)
+void hpi_dsp_code_rewind(struct dsp_code *dsp_code)
{
/* Go back to start of data, after header */
- ps_dsp_code->word_count = sizeof(struct code_header) / sizeof(u32);
+ dsp_code->word_count = sizeof(struct code_header) / sizeof(u32);
}
/*-------------------------------------------------------------------*/
-short hpi_dsp_code_read_word(struct dsp_code *ps_dsp_code, u32 *pword)
+short hpi_dsp_code_read_word(struct dsp_code *dsp_code, u32 *pword)
{
- if (ps_dsp_code->word_count + 1 > ps_dsp_code->block_length)
+ if (dsp_code->word_count + 1 > dsp_code->block_length)
return HPI_ERROR_DSP_FILE_FORMAT;
- *pword = ((u32 *)(ps_dsp_code->ps_firmware->data))[ps_dsp_code->
+ *pword = ((u32 *)(dsp_code->pvt->firmware->data))[dsp_code->
word_count];
- ps_dsp_code->word_count++;
+ dsp_code->word_count++;
return 0;
}
/*-------------------------------------------------------------------*/
short hpi_dsp_code_read_block(size_t words_requested,
- struct dsp_code *ps_dsp_code, u32 **ppblock)
+ struct dsp_code *dsp_code, u32 **ppblock)
{
- if (ps_dsp_code->word_count + words_requested >
- ps_dsp_code->block_length)
+ if (dsp_code->word_count + words_requested > dsp_code->block_length)
return HPI_ERROR_DSP_FILE_FORMAT;
*ppblock =
- ((u32 *)(ps_dsp_code->ps_firmware->data)) +
- ps_dsp_code->word_count;
- ps_dsp_code->word_count += words_requested;
+ ((u32 *)(dsp_code->pvt->firmware->data)) +
+ dsp_code->word_count;
+ dsp_code->word_count += words_requested;
return 0;
}
diff --git a/sound/pci/asihpi/hpidspcd.h b/sound/pci/asihpi/hpidspcd.h
index 65f0ca7..b228811 100644
--- a/sound/pci/asihpi/hpidspcd.h
+++ b/sound/pci/asihpi/hpidspcd.h
@@ -2,7 +2,7 @@
/**
AudioScience HPI driver
- Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
+ Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@@ -20,19 +20,6 @@
\file
Functions for reading DSP code to load into DSP
- hpi_dspcode_defines HPI DSP code loading method
-Define exactly one of these to select how the DSP code is supplied to
-the adapter.
-
-End users writing applications that use the HPI interface do not have to
-use any of the below defines; they are only necessary for building drivers
-
-HPI_DSPCODE_FILE:
-DSP code is supplied as a file that is opened and read from by the driver.
-
-HPI_DSPCODE_FIRMWARE:
-DSP code is read using the hotplug firmware loader module.
- Only valid when compiling the HPI kernel driver under Linux.
*/
/***********************************************************************/
#ifndef _HPIDSPCD_H_
@@ -40,37 +27,56 @@
#include "hpi_internal.h"
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(push, 1)
-#endif
+/** Code header version is decimal encoded e.g. 4.06.10 is 40601 */
+#define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \
+HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER)))
+
+/** Header structure for dsp firmware file
+ This structure must match that used in s2bin.c for generation of asidsp.bin
+ */
+/*#ifndef DISABLE_PRAGMA_PACK1 */
+/*#pragma pack(push, 1) */
+/*#endif */
+struct code_header {
+ /** Size in bytes including header */
+ u32 size;
+ /** File type tag "CODE" == 0x45444F43 */
+ u32 type;
+ /** Adapter model number */
+ u32 adapter;
+ /** Firmware version*/
+ u32 version;
+ /** Data checksum */
+ u32 checksum;
+};
+/*#ifndef DISABLE_PRAGMA_PACK1 */
+/*#pragma pack(pop) */
+/*#endif */
+
+/*? Don't need the pragmas? */
+compile_time_assert((sizeof(struct code_header) == 20), code_header_size);
/** Descriptor for dspcode from firmware loader */
struct dsp_code {
- /** Firmware descriptor */
- const struct firmware *ps_firmware;
- struct pci_dev *ps_dev;
+ /** copy of file header */
+ struct code_header header;
/** Expected number of words in the whole dsp code,INCL header */
- long int block_length;
+ u32 block_length;
/** Number of words read so far */
- long int word_count;
- /** Version read from dsp code file */
- u32 version;
- /** CRC read from dsp code file */
- u32 crc;
+ u32 word_count;
+
+ /** internal state of DSP code reader */
+ struct dsp_code_private *pvt;
};
-#ifndef DISABLE_PRAGMA_PACK1
-#pragma pack(pop)
-#endif
-
-/** Prepare *psDspCode to refer to the requuested adapter.
- Searches the file, or selects the appropriate linked array
+/** Prepare *psDspCode to refer to the requested adapter's firmware.
+Code file name is obtained from HpiOs_GetDspCodePath
\return 0 for success, or error code if requested code is not available
*/
short hpi_dsp_code_open(
/** Code identifier, usually adapter family */
- u32 adapter,
+ u32 adapter, void *pci_dev,
/** Pointer to DSP code control structure */
struct dsp_code *ps_dsp_code,
/** Pointer to dword to receive OS specific error code */
diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c
index 7397b16..ebb568d 100644
--- a/sound/pci/asihpi/hpifunc.c
+++ b/sound/pci/asihpi/hpifunc.c
@@ -1663,68 +1663,64 @@
u16 hpi_cobranet_hmi_write(u32 h_control, u32 hmi_address, u32 byte_count,
u8 *pb_data)
{
- struct hpi_message hm;
- struct hpi_response hr;
+ struct hpi_msg_cobranet_hmiwrite hm;
+ struct hpi_response_header hr;
- hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
- HPI_CONTROL_SET_STATE);
- if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
+ hpi_init_message_responseV1(&hm.h, sizeof(hm), &hr, sizeof(hr),
+ HPI_OBJ_CONTROL, HPI_CONTROL_SET_STATE);
+
+ if (hpi_handle_indexes(h_control, &hm.h.adapter_index,
+ &hm.h.obj_index))
return HPI_ERROR_INVALID_HANDLE;
- hm.u.cx.u.cobranet_data.byte_count = byte_count;
- hm.u.cx.u.cobranet_data.hmi_address = hmi_address;
+ if (byte_count > sizeof(hm.bytes))
+ return HPI_ERROR_MESSAGE_BUFFER_TOO_SMALL;
- if (byte_count <= 8) {
- memcpy(hm.u.cx.u.cobranet_data.data, pb_data, byte_count);
- hm.u.cx.attribute = HPI_COBRANET_SET;
- } else {
- hm.u.cx.u.cobranet_bigdata.pb_data = pb_data;
- hm.u.cx.attribute = HPI_COBRANET_SET_DATA;
- }
+ hm.p.attribute = HPI_COBRANET_SET;
+ hm.p.byte_count = byte_count;
+ hm.p.hmi_address = hmi_address;
+ memcpy(hm.bytes, pb_data, byte_count);
+ hm.h.size = (u16)(sizeof(hm.h) + sizeof(hm.p) + byte_count);
- hpi_send_recv(&hm, &hr);
-
+ hpi_send_recvV1(&hm.h, &hr);
return hr.error;
}
u16 hpi_cobranet_hmi_read(u32 h_control, u32 hmi_address, u32 max_byte_count,
u32 *pbyte_count, u8 *pb_data)
{
- struct hpi_message hm;
- struct hpi_response hr;
+ struct hpi_msg_cobranet_hmiread hm;
+ struct hpi_res_cobranet_hmiread hr;
- hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
- HPI_CONTROL_GET_STATE);
- if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
+ hpi_init_message_responseV1(&hm.h, sizeof(hm), &hr.h, sizeof(hr),
+ HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE);
+
+ if (hpi_handle_indexes(h_control, &hm.h.adapter_index,
+ &hm.h.obj_index))
return HPI_ERROR_INVALID_HANDLE;
- hm.u.cx.u.cobranet_data.byte_count = max_byte_count;
- hm.u.cx.u.cobranet_data.hmi_address = hmi_address;
+ if (max_byte_count > sizeof(hr.bytes))
+ return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
- if (max_byte_count <= 8) {
- hm.u.cx.attribute = HPI_COBRANET_GET;
- } else {
- hm.u.cx.u.cobranet_bigdata.pb_data = pb_data;
- hm.u.cx.attribute = HPI_COBRANET_GET_DATA;
- }
+ hm.p.attribute = HPI_COBRANET_GET;
+ hm.p.byte_count = max_byte_count;
+ hm.p.hmi_address = hmi_address;
- hpi_send_recv(&hm, &hr);
- if (!hr.error && pb_data) {
+ hpi_send_recvV1(&hm.h, &hr.h);
- *pbyte_count = hr.u.cx.u.cobranet_data.byte_count;
+ if (!hr.h.error && pb_data) {
+ if (hr.byte_count > sizeof(hr.bytes))
- if (*pbyte_count < max_byte_count)
+ return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
+
+ *pbyte_count = hr.byte_count;
+
+ if (hr.byte_count < max_byte_count)
max_byte_count = *pbyte_count;
- if (hm.u.cx.attribute == HPI_COBRANET_GET) {
- memcpy(pb_data, hr.u.cx.u.cobranet_data.data,
- max_byte_count);
- } else {
-
- }
-
+ memcpy(pb_data, hr.bytes, max_byte_count);
}
- return hr.error;
+ return hr.h.error;
}
u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus,
@@ -1733,23 +1729,23 @@
struct hpi_message hm;
struct hpi_response hr;
- hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
+ hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
HPI_CONTROL_GET_STATE);
if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
return HPI_ERROR_INVALID_HANDLE;
- hm.u.cx.attribute = HPI_COBRANET_GET_STATUS;
+ hm.u.c.attribute = HPI_COBRANET_GET_STATUS;
hpi_send_recv(&hm, &hr);
if (!hr.error) {
if (pstatus)
- *pstatus = hr.u.cx.u.cobranet_status.status;
+ *pstatus = hr.u.cu.cobranet.status.status;
if (preadable_size)
*preadable_size =
- hr.u.cx.u.cobranet_status.readable_size;
+ hr.u.cu.cobranet.status.readable_size;
if (pwriteable_size)
*pwriteable_size =
- hr.u.cx.u.cobranet_status.writeable_size;
+ hr.u.cu.cobranet.status.writeable_size;
}
return hr.error;
}
diff --git a/sound/pci/asihpi/hpimsginit.c b/sound/pci/asihpi/hpimsginit.c
index 628376c..52400a6 100644
--- a/sound/pci/asihpi/hpimsginit.c
+++ b/sound/pci/asihpi/hpimsginit.c
@@ -46,7 +46,7 @@
if (gwSSX2_bypass)
phm->type = HPI_TYPE_SSX2BYPASS_MESSAGE;
else
- phm->type = HPI_TYPE_MESSAGE;
+ phm->type = HPI_TYPE_REQUEST;
phm->object = object;
phm->function = function;
phm->version = 0;
@@ -89,7 +89,7 @@
memset(phm, 0, sizeof(*phm));
if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) {
phm->size = size;
- phm->type = HPI_TYPE_MESSAGE;
+ phm->type = HPI_TYPE_REQUEST;
phm->object = object;
phm->function = function;
phm->version = 1;
diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c
index 7352a5f..2e77942 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -16,7 +16,7 @@
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
-Extended Message Function With Response Cacheing
+Extended Message Function With Response Caching
(C) Copyright AudioScience Inc. 2002
*****************************************************************************/
@@ -186,7 +186,6 @@
/* Initialize this module's internal state */
hpios_msgxlock_init(&msgx_lock);
memset(&hpi_entry_points, 0, sizeof(hpi_entry_points));
- hpios_locked_mem_init();
/* Init subsys_findadapters response to no-adapters */
HPIMSGX__reset(HPIMSGX_ALLADAPTERS);
hpi_init_response(phr, HPI_OBJ_SUBSYSTEM,
@@ -197,7 +196,6 @@
case HPI_SUBSYS_DRIVER_UNLOAD:
HPI_COMMON(phm, phr);
HPIMSGX__cleanup(HPIMSGX_ALLADAPTERS, h_owner);
- hpios_locked_mem_free_all();
hpi_init_response(phr, HPI_OBJ_SUBSYSTEM,
HPI_SUBSYS_DRIVER_UNLOAD, 0);
return;
@@ -315,7 +313,7 @@
{
HPI_DEBUG_MESSAGE(DEBUG, phm);
- if (phm->type != HPI_TYPE_MESSAGE) {
+ if (phm->type != HPI_TYPE_REQUEST) {
hpi_init_response(phr, phm->object, phm->function,
HPI_ERROR_INVALID_TYPE);
return;
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index d8e7047..65fcf47 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -1,7 +1,7 @@
/*******************************************************************************
AudioScience HPI driver
- Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
+ Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@@ -157,11 +157,6 @@
goto out;
}
- if (hm->h.adapter_index >= HPI_MAX_ADAPTERS) {
- err = -EINVAL;
- goto out;
- }
-
switch (hm->h.function) {
case HPI_SUBSYS_CREATE_ADAPTER:
case HPI_ADAPTER_DELETE:
@@ -187,7 +182,6 @@
/* -1=no data 0=read from user mem, 1=write to user mem */
int wrflag = -1;
u32 adapter = hm->h.adapter_index;
- pa = &adapters[adapter];
if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) {
hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER,
@@ -203,6 +197,8 @@
goto out;
}
+ pa = &adapters[adapter];
+
if (mutex_lock_interruptible(&adapters[adapter].mutex)) {
err = -EINTR;
goto out;
diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index 742ee12..ff2a19b 100644
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -39,10 +39,6 @@
}
-void hpios_locked_mem_init(void)
-{
-}
-
/** Allocated an area of locked memory for bus master DMA operations.
On error, return -ENOMEM, and *pMemArea.size = 0
@@ -85,7 +81,3 @@
return 1;
}
}
-
-void hpios_locked_mem_free_all(void)
-{
-}
diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h
index 03273e7..2f605e3 100644
--- a/sound/pci/asihpi/hpios.h
+++ b/sound/pci/asihpi/hpios.h
@@ -38,6 +38,7 @@
#include <linux/firmware.h>
#include <linux/interrupt.h>
#include <linux/pci.h>
+#include <linux/mutex.h>
#define HPI_NO_OS_FILE_OPS
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 3119cd9..537e0a2 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1624,7 +1624,7 @@
}
if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
- card->shortname, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_atiixp_free(chip);
return -EBUSY;
@@ -1701,7 +1701,7 @@
}
static struct pci_driver driver = {
- .name = "ATI IXP AC97 controller",
+ .name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
.remove = __devexit_p(snd_atiixp_remove),
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 2f74c2f..45df275 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1260,7 +1260,7 @@
}
if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
- card->shortname, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_atiixp_free(chip);
return -EBUSY;
@@ -1332,7 +1332,7 @@
}
static struct pci_driver driver = {
- .name = "ATI IXP MC97 controller",
+ .name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
.remove = __devexit_p(snd_atiixp_remove),
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 7b72c88..a384699 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -196,7 +196,7 @@
}
if ((err = request_irq(pci->irq, vortex_interrupt,
- IRQF_SHARED, CARD_NAME_SHORT,
+ IRQF_SHARED, KBUILD_MODNAME,
chip)) != 0) {
printk(KERN_ERR "cannot grab irq\n");
goto irq_out;
@@ -375,7 +375,7 @@
// pci_driver definition
static struct pci_driver driver = {
- .name = CARD_NAME_SHORT,
+ .name = KBUILD_MODNAME,
.id_table = snd_vortex_ids,
.probe = snd_vortex_probe,
.remove = __devexit_p(snd_vortex_remove),
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index c150022..f8569b1 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -171,7 +171,7 @@
/* pci_driver definition */
static struct pci_driver driver = {
- .name = "Emagic Audiowerk 2",
+ .name = KBUILD_MODNAME,
.id_table = snd_aw2_ids,
.probe = snd_aw2_probe,
.remove = __devexit_p(snd_aw2_remove),
@@ -317,7 +317,7 @@
snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
- IRQF_SHARED, "Audiowerk2", chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "aw2: Cannot grab irq %d\n", pci->irq);
iounmap(chip->iobase_virt);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 9b7a634..e4d76a2 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2559,7 +2559,7 @@
codec_setup->name = "I2S_OUT";
if (request_irq(pci->irq, snd_azf3328_interrupt,
- IRQF_SHARED, card->shortname, chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
err = -EBUSY;
goto out_err;
@@ -2860,7 +2860,7 @@
static struct pci_driver driver = {
- .name = "AZF3328",
+ .name = KBUILD_MODNAME,
.id_table = snd_azf3328_ids,
.probe = snd_azf3328_probe,
.remove = __devexit_p(snd_azf3328_remove),
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 2958a05..3918033 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -760,7 +760,7 @@
snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS);
err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
- "Bt87x audio", chip);
+ KBUILD_MODNAME, chip);
if (err < 0) {
snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
goto fail;
@@ -965,7 +965,7 @@
};
static struct pci_driver driver = {
- .name = "Bt87x",
+ .name = KBUILD_MODNAME,
.id_table = snd_bt87x_ids,
.probe = snd_bt87x_probe,
.remove = __devexit_p(snd_bt87x_remove),
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 4377592..061b7e6 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1666,7 +1666,7 @@
}
if (request_irq(pci->irq, snd_ca0106_interrupt,
- IRQF_SHARED, "snd_ca0106", chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
@@ -1933,7 +1933,7 @@
// pci_driver definition
static struct pci_driver driver = {
- .name = "CA0106",
+ .name = KBUILD_MODNAME,
.id_table = snd_ca0106_ids,
.probe = snd_ca0106_probe,
.remove = __devexit_p(snd_ca0106_remove),
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index f4e5735..9cf99fb 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3053,7 +3053,7 @@
cm->iobase = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_cmipci_interrupt,
- IRQF_SHARED, card->driver, cm)) {
+ IRQF_SHARED, KBUILD_MODNAME, cm)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_cmipci_free(cm);
return -EBUSY;
@@ -3398,7 +3398,7 @@
#endif /* CONFIG_PM */
static struct pci_driver driver = {
- .name = "C-Media PCI",
+ .name = KBUILD_MODNAME,
.id_table = snd_cmipci_ids,
.probe = snd_cmipci_probe,
.remove = __devexit_p(snd_cmipci_remove),
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 6772070..07f04e3 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1382,7 +1382,7 @@
}
if (request_irq(pci->irq, snd_cs4281_interrupt, IRQF_SHARED,
- "CS4281", chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_cs4281_free(chip);
return -ENOMEM;
@@ -2085,7 +2085,7 @@
#endif /* CONFIG_PM */
static struct pci_driver driver = {
- .name = "CS4281",
+ .name = KBUILD_MODNAME,
.id_table = snd_cs4281_ids,
.probe = snd_cs4281_probe,
.remove = __devexit_p(snd_cs4281_remove),
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 767fa7f..1af9555 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -162,7 +162,7 @@
}
static struct pci_driver driver = {
- .name = "Sound Fusion CS46xx",
+ .name = KBUILD_MODNAME,
.id_table = snd_cs46xx_ids,
.probe = snd_card_cs46xx_probe,
.remove = __devexit_p(snd_card_cs46xx_remove),
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index aad3708..9546bf0 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -3835,7 +3835,7 @@
}
if (request_irq(pci->irq, snd_cs46xx_interrupt, IRQF_SHARED,
- "CS46XX", chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_cs46xx_free(chip);
return -EBUSY;
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index bc07e27..a466934 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -285,7 +285,7 @@
}
static struct pci_driver driver = {
- .name = "CS5530_Audio",
+ .name = KBUILD_MODNAME,
.id_table = snd_cs5530_ids,
.probe = snd_cs5530_probe,
.remove = __devexit_p(snd_cs5530_remove),
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index afb8037..10d22ed 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -311,7 +311,7 @@
cs5535au->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_cs5535audio_interrupt,
- IRQF_SHARED, "CS5535 Audio", cs5535au)) {
+ IRQF_SHARED, KBUILD_MODNAME, cs5535au)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
err = -EBUSY;
goto sndfail;
@@ -395,7 +395,7 @@
}
static struct pci_driver driver = {
- .name = DRIVER_NAME,
+ .name = KBUILD_MODNAME,
.id_table = snd_cs5535audio_ids,
.probe = snd_cs5535audio_probe,
.remove = __devexit_p(snd_cs5535audio_remove),
diff --git a/sound/pci/ctxfi/ct20k2reg.h b/sound/pci/ctxfi/ct20k2reg.h
index e0394e3..ca501ba 100644
--- a/sound/pci/ctxfi/ct20k2reg.h
+++ b/sound/pci/ctxfi/ct20k2reg.h
@@ -55,6 +55,7 @@
/* GPIO Registers */
#define GPIO_DATA 0x1B7020
#define GPIO_CTRL 0x1B7024
+#define GPIO_EXT_DATA 0x1B70A0
/* Virtual memory registers */
#define VMEM_PTPAL 0x1C6300 /* 0x1C6300 + (16 * Chn) */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 13f33c0..d8a4423 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -18,7 +18,6 @@
#include "ctatc.h"
#include "ctpcm.h"
#include "ctmixer.h"
-#include "cthardware.h"
#include "ctsrc.h"
#include "ctamixer.h"
#include "ctdaio.h"
@@ -30,7 +29,6 @@
#include <sound/asoundef.h>
#define MONO_SUM_SCALE 0x19a8 /* 2^(-0.5) in 14-bit floating format */
-#define DAIONUM 7
#define MAX_MULTI_CHN 8
#define IEC958_DEFAULT_CON ((IEC958_AES0_NONAUDIO \
@@ -53,6 +51,8 @@
static struct snd_pci_quirk __devinitdata subsys_20k2_list[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB0760,
"SB0760", CTSB0760),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB1270,
+ "SB1270", CTSB1270),
SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08801,
"SB0880", CTSB0880),
SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08802,
@@ -75,6 +75,7 @@
[CTSB0760] = "SB076x",
[CTHENDRIX] = "Hendrix",
[CTSB0880] = "SB0880",
+ [CTSB1270] = "SB1270",
[CT20K2_UNKNOWN] = "Unknown",
};
@@ -459,12 +460,12 @@
apcm->substream->runtime->rate);
*n_srcc = 0;
- if (1 == atc->msr) {
+ if (1 == atc->msr) { /* FIXME: do we really need SRC here if pitch==1 */
*n_srcc = apcm->substream->runtime->channels;
conf[0].pitch = pitch;
conf[0].mix_msr = conf[0].imp_msr = conf[0].msr = 1;
conf[0].vo = 1;
- } else if (2 == atc->msr) {
+ } else if (2 <= atc->msr) {
if (0x8000000 < pitch) {
/* Need two-stage SRCs, SRCIMPs and
* AMIXERs for converting format */
@@ -970,11 +971,39 @@
return 0;
}
-static int atc_have_digit_io_switch(struct ct_atc *atc)
+static struct capabilities atc_capabilities(struct ct_atc *atc)
{
struct hw *hw = atc->hw;
- return hw->have_digit_io_switch(hw);
+ return hw->capabilities(hw);
+}
+
+static int atc_output_switch_get(struct ct_atc *atc)
+{
+ struct hw *hw = atc->hw;
+
+ return hw->output_switch_get(hw);
+}
+
+static int atc_output_switch_put(struct ct_atc *atc, int position)
+{
+ struct hw *hw = atc->hw;
+
+ return hw->output_switch_put(hw, position);
+}
+
+static int atc_mic_source_switch_get(struct ct_atc *atc)
+{
+ struct hw *hw = atc->hw;
+
+ return hw->mic_source_switch_get(hw);
+}
+
+static int atc_mic_source_switch_put(struct ct_atc *atc, int position)
+{
+ struct hw *hw = atc->hw;
+
+ return hw->mic_source_switch_put(hw, position);
}
static int atc_select_digit_io(struct ct_atc *atc)
@@ -1045,6 +1074,11 @@
return atc_daio_unmute(atc, state, LINEIM);
}
+static int atc_mic_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, MIC);
+}
+
static int atc_spdif_out_unmute(struct ct_atc *atc, unsigned char state)
{
return atc_daio_unmute(atc, state, SPDIFOO);
@@ -1331,17 +1365,20 @@
struct srcimp_mgr *srcimp_mgr;
struct sum_desc sum_dsc = {0};
struct sum_mgr *sum_mgr;
- int err, i;
+ int err, i, num_srcs, num_daios;
- atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL);
+ num_daios = ((atc->model == CTSB1270) ? 8 : 7);
+ num_srcs = ((atc->model == CTSB1270) ? 6 : 4);
+
+ atc->daios = kzalloc(sizeof(void *)*num_daios, GFP_KERNEL);
if (!atc->daios)
return -ENOMEM;
- atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
+ atc->srcs = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL);
if (!atc->srcs)
return -ENOMEM;
- atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
+ atc->srcimps = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL);
if (!atc->srcimps)
return -ENOMEM;
@@ -1351,8 +1388,9 @@
daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO];
da_desc.msr = atc->msr;
- for (i = 0, atc->n_daio = 0; i < DAIONUM-1; i++) {
- da_desc.type = i;
+ for (i = 0, atc->n_daio = 0; i < num_daios; i++) {
+ da_desc.type = (atc->model != CTSB073X) ? i :
+ ((i == SPDIFIO) ? SPDIFI1 : i);
err = daio_mgr->get_daio(daio_mgr, &da_desc,
(struct daio **)&atc->daios[i]);
if (err) {
@@ -1362,23 +1400,12 @@
}
atc->n_daio++;
}
- if (atc->model == CTSB073X)
- da_desc.type = SPDIFI1;
- else
- da_desc.type = SPDIFIO;
- err = daio_mgr->get_daio(daio_mgr, &da_desc,
- (struct daio **)&atc->daios[i]);
- if (err) {
- printk(KERN_ERR "ctxfi: Failed to get S/PDIF-in resource!!!\n");
- return err;
- }
- atc->n_daio++;
src_mgr = atc->rsc_mgrs[SRC];
src_dsc.multi = 1;
src_dsc.msr = atc->msr;
src_dsc.mode = ARCRW;
- for (i = 0, atc->n_src = 0; i < (2*2); i++) {
+ for (i = 0, atc->n_src = 0; i < num_srcs; i++) {
err = src_mgr->get_src(src_mgr, &src_dsc,
(struct src **)&atc->srcs[i]);
if (err)
@@ -1388,8 +1415,8 @@
}
srcimp_mgr = atc->rsc_mgrs[SRCIMP];
- srcimp_dsc.msr = 8; /* SRCIMPs for S/PDIFIn SRT */
- for (i = 0, atc->n_srcimp = 0; i < (2*1); i++) {
+ srcimp_dsc.msr = 8;
+ for (i = 0, atc->n_srcimp = 0; i < num_srcs; i++) {
err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
(struct srcimp **)&atc->srcimps[i]);
if (err)
@@ -1397,15 +1424,6 @@
atc->n_srcimp++;
}
- srcimp_dsc.msr = 8; /* SRCIMPs for LINE/MICIn SRT */
- for (i = 0; i < (2*1); i++) {
- err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
- (struct srcimp **)&atc->srcimps[2*1+i]);
- if (err)
- return err;
-
- atc->n_srcimp++;
- }
sum_mgr = atc->rsc_mgrs[SUM];
sum_dsc.msr = atc->msr;
@@ -1488,6 +1506,18 @@
src = atc->srcs[3];
mixer->set_input_right(mixer, MIX_LINE_IN, &src->rsc);
+ if (atc->model == CTSB1270) {
+ /* Titanium HD has a dedicated ADC for the Mic. */
+ dai = container_of(atc->daios[MIC], struct dai, daio);
+ atc_connect_dai(atc->rsc_mgrs[SRC], dai,
+ (struct src **)&atc->srcs[4],
+ (struct srcimp **)&atc->srcimps[4]);
+ src = atc->srcs[4];
+ mixer->set_input_left(mixer, MIX_MIC_IN, &src->rsc);
+ src = atc->srcs[5];
+ mixer->set_input_right(mixer, MIX_MIC_IN, &src->rsc);
+ }
+
dai = container_of(atc->daios[SPDIFIO], struct dai, daio);
atc_connect_dai(atc->rsc_mgrs[SRC], dai,
(struct src **)&atc->srcs[0],
@@ -1606,12 +1636,17 @@
.line_clfe_unmute = atc_line_clfe_unmute,
.line_rear_unmute = atc_line_rear_unmute,
.line_in_unmute = atc_line_in_unmute,
+ .mic_unmute = atc_mic_unmute,
.spdif_out_unmute = atc_spdif_out_unmute,
.spdif_in_unmute = atc_spdif_in_unmute,
.spdif_out_get_status = atc_spdif_out_get_status,
.spdif_out_set_status = atc_spdif_out_set_status,
.spdif_out_passthru = atc_spdif_out_passthru,
- .have_digit_io_switch = atc_have_digit_io_switch,
+ .capabilities = atc_capabilities,
+ .output_switch_get = atc_output_switch_get,
+ .output_switch_put = atc_output_switch_put,
+ .mic_source_switch_get = atc_mic_source_switch_get,
+ .mic_source_switch_put = atc_mic_source_switch_put,
#ifdef CONFIG_PM
.suspend = atc_suspend,
.resume = atc_resume,
diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h
index 7167c01..3a0def6 100644
--- a/sound/pci/ctxfi/ctatc.h
+++ b/sound/pci/ctxfi/ctatc.h
@@ -25,6 +25,7 @@
#include <sound/core.h>
#include "ctvmem.h"
+#include "cthardware.h"
#include "ctresource.h"
enum CTALSADEVS { /* Types of alsa devices */
@@ -115,12 +116,17 @@
int (*line_clfe_unmute)(struct ct_atc *atc, unsigned char state);
int (*line_rear_unmute)(struct ct_atc *atc, unsigned char state);
int (*line_in_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*mic_unmute)(struct ct_atc *atc, unsigned char state);
int (*spdif_out_unmute)(struct ct_atc *atc, unsigned char state);
int (*spdif_in_unmute)(struct ct_atc *atc, unsigned char state);
int (*spdif_out_get_status)(struct ct_atc *atc, unsigned int *status);
int (*spdif_out_set_status)(struct ct_atc *atc, unsigned int status);
int (*spdif_out_passthru)(struct ct_atc *atc, unsigned char state);
- int (*have_digit_io_switch)(struct ct_atc *atc);
+ struct capabilities (*capabilities)(struct ct_atc *atc);
+ int (*output_switch_get)(struct ct_atc *atc);
+ int (*output_switch_put)(struct ct_atc *atc, int position);
+ int (*mic_source_switch_get)(struct ct_atc *atc);
+ int (*mic_source_switch_put)(struct ct_atc *atc, int position);
/* Don't touch! Used for internal object. */
void *rsc_mgrs[NUM_RSCTYP]; /* chip resource managers */
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 47d9ea9..0c00eb4 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -22,20 +22,9 @@
#include <linux/slab.h>
#include <linux/kernel.h>
-#define DAIO_RESOURCE_NUM NUM_DAIOTYP
#define DAIO_OUT_MAX SPDIFOO
-union daio_usage {
- struct {
- unsigned short lineo1:1;
- unsigned short lineo2:1;
- unsigned short lineo3:1;
- unsigned short lineo4:1;
- unsigned short spdifoo:1;
- unsigned short lineim:1;
- unsigned short spdifio:1;
- unsigned short spdifi1:1;
- } bf;
+struct daio_usage {
unsigned short data;
};
@@ -61,6 +50,7 @@
[LINEO3] = {.left = 0x50, .right = 0x51},
[LINEO4] = {.left = 0x70, .right = 0x71},
[LINEIM] = {.left = 0x45, .right = 0xc5},
+ [MIC] = {.left = 0x55, .right = 0xd5},
[SPDIFOO] = {.left = 0x00, .right = 0x01},
[SPDIFIO] = {.left = 0x05, .right = 0x85},
};
@@ -138,6 +128,7 @@
case LINEO3: return 5;
case LINEO4: return 6;
case LINEIM: return 4;
+ case MIC: return 5;
default: return -EINVAL;
}
default:
@@ -519,17 +510,17 @@
static int daio_mgr_get_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
{
- if (((union daio_usage *)mgr->rscs)->data & (0x1 << type))
+ if (((struct daio_usage *)mgr->rscs)->data & (0x1 << type))
return -ENOENT;
- ((union daio_usage *)mgr->rscs)->data |= (0x1 << type);
+ ((struct daio_usage *)mgr->rscs)->data |= (0x1 << type);
return 0;
}
static int daio_mgr_put_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
{
- ((union daio_usage *)mgr->rscs)->data &= ~(0x1 << type);
+ ((struct daio_usage *)mgr->rscs)->data &= ~(0x1 << type);
return 0;
}
@@ -712,7 +703,7 @@
if (!daio_mgr)
return -ENOMEM;
- err = rsc_mgr_init(&daio_mgr->mgr, DAIO, DAIO_RESOURCE_NUM, hw);
+ err = rsc_mgr_init(&daio_mgr->mgr, DAIO, NUM_DAIOTYP, hw);
if (err)
goto error1;
diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h
index 0f52ce5..85ccb6e 100644
--- a/sound/pci/ctxfi/ctdaio.h
+++ b/sound/pci/ctxfi/ctdaio.h
@@ -33,6 +33,7 @@
SPDIFOO, /* S/PDIF Out (Flexijack/Optical) */
LINEIM,
SPDIFIO, /* S/PDIF In (Flexijack/Optical) on the card */
+ MIC, /* Dedicated mic on Titanium HD */
SPDIFI1, /* S/PDIF In on internal Drive Bay */
NUM_DAIOTYP
};
diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h
index af55405..908315b 100644
--- a/sound/pci/ctxfi/cthardware.h
+++ b/sound/pci/ctxfi/cthardware.h
@@ -39,6 +39,7 @@
CT20K2_MODEL_FIRST = CTSB0760,
CTHENDRIX,
CTSB0880,
+ CTSB1270,
CT20K2_UNKNOWN,
NUM_CTCARDS /* This should always be the last */
};
@@ -60,6 +61,13 @@
unsigned int msr; /* master sample rate in rsrs */
};
+struct capabilities {
+ unsigned int digit_io_switch:1;
+ unsigned int dedicated_mic:1;
+ unsigned int output_switch:1;
+ unsigned int mic_source_switch:1;
+};
+
struct hw {
int (*card_init)(struct hw *hw, struct card_conf *info);
int (*card_stop)(struct hw *hw);
@@ -70,7 +78,11 @@
#endif
int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source);
int (*select_adc_source)(struct hw *hw, enum ADCSRC source);
- int (*have_digit_io_switch)(struct hw *hw);
+ struct capabilities (*capabilities)(struct hw *hw);
+ int (*output_switch_get)(struct hw *hw);
+ int (*output_switch_put)(struct hw *hw, int position);
+ int (*mic_source_switch_get)(struct hw *hw);
+ int (*mic_source_switch_put)(struct hw *hw, int position);
/* SRC operations */
int (*src_rsc_get_ctrl_blk)(void **rblk);
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index a5c957d..a7df197 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1777,10 +1777,17 @@
return adc_init_SBx(hw, info->input, info->mic20db);
}
-static int hw_have_digit_io_switch(struct hw *hw)
+static struct capabilities hw_capabilities(struct hw *hw)
{
+ struct capabilities cap;
+
/* SB073x and Vista compatible cards have no digit IO switch */
- return !(hw->model == CTSB073X || hw->model == CTUAA);
+ cap.digit_io_switch = !(hw->model == CTSB073X || hw->model == CTUAA);
+ cap.dedicated_mic = 0;
+ cap.output_switch = 0;
+ cap.mic_source_switch = 0;
+
+ return cap;
}
#define CTLBITS(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
@@ -1933,7 +1940,7 @@
if (hw->irq < 0) {
err = request_irq(pci->irq, ct_20k1_interrupt, IRQF_SHARED,
- "ctxfi", hw);
+ KBUILD_MODNAME, hw);
if (err < 0) {
printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
goto error2;
@@ -2172,7 +2179,7 @@
.pll_init = hw_pll_init,
.is_adc_source_selected = hw_is_adc_input_selected,
.select_adc_source = hw_adc_input_select,
- .have_digit_io_switch = hw_have_digit_io_switch,
+ .capabilities = hw_capabilities,
#ifdef CONFIG_PM
.suspend = hw_suspend,
.resume = hw_resume,
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index 5364164..d6c54b5 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -8,7 +8,7 @@
* @File cthw20k2.c
*
* @Brief
- * This file contains the implementation of hardware access methord for 20k2.
+ * This file contains the implementation of hardware access method for 20k2.
*
* @Author Liu Chun
* @Date May 14 2008
@@ -38,6 +38,8 @@
unsigned char dev_id;
unsigned char addr_size;
unsigned char data_size;
+
+ int mic_source;
};
static u32 hw_read_20kx(struct hw *hw, u32 reg);
@@ -1163,7 +1165,12 @@
hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x01010101);
hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
} else if (2 == info->msr) {
- hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111);
+ if (hw->model != CTSB1270) {
+ hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111);
+ } else {
+ /* PCM4220 on Titanium HD is different. */
+ hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11011111);
+ }
/* Specify all playing 96khz
* EA [0] - Enabled
* RTA [4:5] - 96kHz
@@ -1175,6 +1182,10 @@
* RTD [28:29] - 96kHz */
hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x11111111);
hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
+ } else if ((4 == info->msr) && (hw->model == CTSB1270)) {
+ hw_write_20kx(hw, AUDIO_IO_MCLK, 0x21011111);
+ hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x21212121);
+ hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
} else {
printk(KERN_ALERT "ctxfi: ERROR!!! Invalid sampling rate!!!\n");
return -EINVAL;
@@ -1182,6 +1193,8 @@
for (i = 0; i < 8; i++) {
if (i <= 3) {
+ /* This comment looks wrong since loop is over 4 */
+ /* channels and emu20k2 supports 4 spdif IOs. */
/* 1st 3 channels are SPDIFs (SB0960) */
if (i == 3)
data = 0x1001001;
@@ -1206,12 +1219,16 @@
hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_H+(0x40*i), 0x0B);
} else {
+ /* Again, loop is over 4 channels not 5. */
/* Next 5 channels are I2S (SB0960) */
data = 0x11;
hw_write_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i), data);
if (2 == info->msr) {
/* Four channels per sample period */
data |= 0x1000;
+ } else if (4 == info->msr) {
+ /* FIXME: check this against the chip spec */
+ data |= 0x2000;
}
hw_write_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i), data);
}
@@ -1299,21 +1316,18 @@
pllenb = 0xB;
hw_write_20kx(hw, PLL_ENB, pllenb);
- pllctl = 0x20D00000;
- set_field(&pllctl, PLLCTL_FD, 16 - 4);
- hw_write_20kx(hw, PLL_CTL, pllctl);
- mdelay(40);
- pllctl = hw_read_20kx(hw, PLL_CTL);
+ pllctl = 0x20C00000;
set_field(&pllctl, PLLCTL_B, 0);
- if (48000 == rsr) {
- set_field(&pllctl, PLLCTL_FD, 16 - 2);
- set_field(&pllctl, PLLCTL_RD, 1 - 1); /* 3000*16/1 = 48000 */
- } else { /* 44100 */
- set_field(&pllctl, PLLCTL_FD, 147 - 2);
- set_field(&pllctl, PLLCTL_RD, 10 - 1); /* 3000*147/10 = 44100 */
- }
+ set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4);
+ set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1);
hw_write_20kx(hw, PLL_CTL, pllctl);
mdelay(40);
+
+ pllctl = hw_read_20kx(hw, PLL_CTL);
+ set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2);
+ hw_write_20kx(hw, PLL_CTL, pllctl);
+ mdelay(40);
+
for (i = 0; i < 1000; i++) {
pllstat = hw_read_20kx(hw, PLL_STAT);
if (get_field(pllstat, PLLSTAT_PD))
@@ -1557,7 +1571,7 @@
hw_write_20kx(hw, I2C_IF_STATUS, i2c_status);
hw20k2_i2c_wait_data_ready(hw);
- /* Dummy write to trigger the write oprtation */
+ /* Dummy write to trigger the write operation */
hw_write_20kx(hw, I2C_IF_WDATA, 0);
hw20k2_i2c_wait_data_ready(hw);
@@ -1568,6 +1582,30 @@
return 0;
}
+static void hw_dac_stop(struct hw *hw)
+{
+ u32 data;
+ data = hw_read_20kx(hw, GPIO_DATA);
+ data &= 0xFFFFFFFD;
+ hw_write_20kx(hw, GPIO_DATA, data);
+ mdelay(10);
+}
+
+static void hw_dac_start(struct hw *hw)
+{
+ u32 data;
+ data = hw_read_20kx(hw, GPIO_DATA);
+ data |= 0x2;
+ hw_write_20kx(hw, GPIO_DATA, data);
+ mdelay(50);
+}
+
+static void hw_dac_reset(struct hw *hw)
+{
+ hw_dac_stop(hw);
+ hw_dac_start(hw);
+}
+
static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
{
int err;
@@ -1594,6 +1632,21 @@
0x00000000 /* Vol Control B4 */
};
+ if (hw->model == CTSB1270) {
+ hw_dac_stop(hw);
+ data = hw_read_20kx(hw, GPIO_DATA);
+ data &= ~0x0600;
+ if (1 == info->msr)
+ data |= 0x0000; /* Single Speed Mode 0-50kHz */
+ else if (2 == info->msr)
+ data |= 0x0200; /* Double Speed Mode 50-100kHz */
+ else
+ data |= 0x0600; /* Quad Speed Mode 100-200kHz */
+ hw_write_20kx(hw, GPIO_DATA, data);
+ hw_dac_start(hw);
+ return 0;
+ }
+
/* Set DAC reset bit as output */
data = hw_read_20kx(hw, GPIO_CTRL);
data |= 0x02;
@@ -1606,22 +1659,8 @@
for (i = 0; i < 2; i++) {
/* Reset DAC twice just in-case the chip
* didn't initialized properly */
- data = hw_read_20kx(hw, GPIO_DATA);
- /* GPIO data bit 1 */
- data &= 0xFFFFFFFD;
- hw_write_20kx(hw, GPIO_DATA, data);
- mdelay(10);
- data |= 0x2;
- hw_write_20kx(hw, GPIO_DATA, data);
- mdelay(50);
-
- /* Reset the 2nd time */
- data &= 0xFFFFFFFD;
- hw_write_20kx(hw, GPIO_DATA, data);
- mdelay(10);
- data |= 0x2;
- hw_write_20kx(hw, GPIO_DATA, data);
- mdelay(50);
+ hw_dac_reset(hw);
+ hw_dac_reset(hw);
if (hw20k2_i2c_read(hw, CS4382_MC1, &cs_read.mode_control_1))
continue;
@@ -1725,7 +1764,11 @@
static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type)
{
u32 data;
-
+ if (hw->model == CTSB1270) {
+ /* Titanium HD has two ADC chips, one for line in and one */
+ /* for MIC. We don't need to switch the ADC input. */
+ return 1;
+ }
data = hw_read_20kx(hw, GPIO_DATA);
switch (type) {
case ADC_MICIN:
@@ -1742,35 +1785,47 @@
#define MIC_BOOST_0DB 0xCF
#define MIC_BOOST_STEPS_PER_DB 2
-#define MIC_BOOST_20DB (MIC_BOOST_0DB + 20 * MIC_BOOST_STEPS_PER_DB)
+
+static void hw_wm8775_input_select(struct hw *hw, u8 input, s8 gain_in_db)
+{
+ u32 adcmc, gain;
+
+ if (input > 3)
+ input = 3;
+
+ adcmc = ((u32)1 << input) | 0x100; /* Link L+R gain... */
+
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, adcmc),
+ MAKE_WM8775_DATA(adcmc));
+
+ if (gain_in_db < -103)
+ gain_in_db = -103;
+ if (gain_in_db > 24)
+ gain_in_db = 24;
+
+ gain = gain_in_db * MIC_BOOST_STEPS_PER_DB + MIC_BOOST_0DB;
+
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, gain),
+ MAKE_WM8775_DATA(gain));
+ /* ...so there should be no need for the following. */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, gain),
+ MAKE_WM8775_DATA(gain));
+}
static int hw_adc_input_select(struct hw *hw, enum ADCSRC type)
{
u32 data;
-
data = hw_read_20kx(hw, GPIO_DATA);
switch (type) {
case ADC_MICIN:
data |= (0x1 << 14);
hw_write_20kx(hw, GPIO_DATA, data);
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
- MAKE_WM8775_DATA(0x101)); /* Mic-in */
- hw20k2_i2c_write(hw,
- MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB),
- MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
- hw20k2_i2c_write(hw,
- MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB),
- MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
+ hw_wm8775_input_select(hw, 0, 20); /* Mic, 20dB */
break;
case ADC_LINEIN:
data &= ~(0x1 << 14);
hw_write_20kx(hw, GPIO_DATA, data);
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
- MAKE_WM8775_DATA(0x102)); /* Line-in */
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
- MAKE_WM8775_DATA(0xCF)); /* No boost */
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
- MAKE_WM8775_DATA(0xCF)); /* No boost */
+ hw_wm8775_input_select(hw, 1, 0); /* Line-in, 0dB */
break;
default:
break;
@@ -1782,7 +1837,7 @@
static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
{
int err;
- u32 mux = 2, data, ctl;
+ u32 data, ctl;
/* Set ADC reset bit as output */
data = hw_read_20kx(hw, GPIO_CTRL);
@@ -1796,19 +1851,42 @@
goto error;
}
- /* Make ADC in normal operation */
+ /* Reset the ADC (reset is active low). */
data = hw_read_20kx(hw, GPIO_DATA);
data &= ~(0x1 << 15);
+ hw_write_20kx(hw, GPIO_DATA, data);
+
+ if (hw->model == CTSB1270) {
+ /* Set up the PCM4220 ADC on Titanium HD */
+ data &= ~0x0C;
+ if (1 == info->msr)
+ data |= 0x00; /* Single Speed Mode 32-50kHz */
+ else if (2 == info->msr)
+ data |= 0x08; /* Double Speed Mode 50-108kHz */
+ else
+ data |= 0x04; /* Quad Speed Mode 108kHz-216kHz */
+ hw_write_20kx(hw, GPIO_DATA, data);
+ }
+
mdelay(10);
+ /* Return the ADC to normal operation. */
data |= (0x1 << 15);
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(50);
+ /* I2C write to register offset 0x0B to set ADC LRCLK polarity */
+ /* invert bit, interface format to I2S, word length to 24-bit, */
+ /* enable ADC high pass filter. Fixes bug 5323? */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_IC, 0x26),
+ MAKE_WM8775_DATA(0x26));
+
/* Set the master mode (256fs) */
if (1 == info->msr) {
+ /* slave mode, 128x oversampling 256fs */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x02),
MAKE_WM8775_DATA(0x02));
- } else if (2 == info->msr) {
+ } else if ((2 == info->msr) || (4 == info->msr)) {
+ /* slave mode, 64x oversampling, 256fs */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A),
MAKE_WM8775_DATA(0x0A));
} else {
@@ -1818,55 +1896,113 @@
goto error;
}
- /* Configure GPIO bit 14 change to line-in/mic-in */
- ctl = hw_read_20kx(hw, GPIO_CTRL);
- ctl |= 0x1 << 14;
- hw_write_20kx(hw, GPIO_CTRL, ctl);
-
- /* Check using Mic-in or Line-in */
- data = hw_read_20kx(hw, GPIO_DATA);
-
- if (mux == 1) {
- /* Configures GPIO data to select Mic-in */
- data |= 0x1 << 14;
- hw_write_20kx(hw, GPIO_DATA, data);
-
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
- MAKE_WM8775_DATA(0x101)); /* Mic-in */
- hw20k2_i2c_write(hw,
- MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB),
- MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
- hw20k2_i2c_write(hw,
- MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB),
- MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
- } else if (mux == 2) {
- /* Configures GPIO data to select Line-in */
- data &= ~(0x1 << 14);
- hw_write_20kx(hw, GPIO_DATA, data);
-
- /* Setup ADC */
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
- MAKE_WM8775_DATA(0x102)); /* Line-in */
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
- MAKE_WM8775_DATA(0xCF)); /* No boost */
- hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
- MAKE_WM8775_DATA(0xCF)); /* No boost */
+ if (hw->model != CTSB1270) {
+ /* Configure GPIO bit 14 change to line-in/mic-in */
+ ctl = hw_read_20kx(hw, GPIO_CTRL);
+ ctl |= 0x1 << 14;
+ hw_write_20kx(hw, GPIO_CTRL, ctl);
+ hw_adc_input_select(hw, ADC_LINEIN);
} else {
- printk(KERN_ALERT "ctxfi: ERROR!!! Invalid input mux!!!\n");
- err = -EINVAL;
- goto error;
+ hw_wm8775_input_select(hw, 0, 0);
}
return 0;
-
error:
hw20k2_i2c_uninit(hw);
return err;
}
-static int hw_have_digit_io_switch(struct hw *hw)
+static struct capabilities hw_capabilities(struct hw *hw)
{
- return 0;
+ struct capabilities cap;
+
+ cap.digit_io_switch = 0;
+ cap.dedicated_mic = hw->model == CTSB1270;
+ cap.output_switch = hw->model == CTSB1270;
+ cap.mic_source_switch = hw->model == CTSB1270;
+
+ return cap;
+}
+
+static int hw_output_switch_get(struct hw *hw)
+{
+ u32 data = hw_read_20kx(hw, GPIO_EXT_DATA);
+
+ switch (data & 0x30) {
+ case 0x00:
+ return 0;
+ case 0x10:
+ return 1;
+ case 0x20:
+ return 2;
+ default:
+ return 3;
+ }
+}
+
+static int hw_output_switch_put(struct hw *hw, int position)
+{
+ u32 data;
+
+ if (position == hw_output_switch_get(hw))
+ return 0;
+
+ /* Mute line and headphones (intended for anti-pop). */
+ data = hw_read_20kx(hw, GPIO_DATA);
+ data |= (0x03 << 11);
+ hw_write_20kx(hw, GPIO_DATA, data);
+
+ data = hw_read_20kx(hw, GPIO_EXT_DATA) & ~0x30;
+ switch (position) {
+ case 0:
+ break;
+ case 1:
+ data |= 0x10;
+ break;
+ default:
+ data |= 0x20;
+ }
+ hw_write_20kx(hw, GPIO_EXT_DATA, data);
+
+ /* Unmute line and headphones. */
+ data = hw_read_20kx(hw, GPIO_DATA);
+ data &= ~(0x03 << 11);
+ hw_write_20kx(hw, GPIO_DATA, data);
+
+ return 1;
+}
+
+static int hw_mic_source_switch_get(struct hw *hw)
+{
+ struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
+
+ return hw20k2->mic_source;
+}
+
+static int hw_mic_source_switch_put(struct hw *hw, int position)
+{
+ struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
+
+ if (position == hw20k2->mic_source)
+ return 0;
+
+ switch (position) {
+ case 0:
+ hw_wm8775_input_select(hw, 0, 0); /* Mic, 0dB */
+ break;
+ case 1:
+ hw_wm8775_input_select(hw, 1, 0); /* FP Mic, 0dB */
+ break;
+ case 2:
+ hw_wm8775_input_select(hw, 3, 0); /* Aux Ext, 0dB */
+ break;
+ default:
+ return 0;
+ }
+
+ hw20k2->mic_source = position;
+
+ return 1;
}
static irqreturn_t ct_20k2_interrupt(int irq, void *dev_id)
@@ -1925,7 +2061,7 @@
if (hw->irq < 0) {
err = request_irq(pci->irq, ct_20k2_interrupt, IRQF_SHARED,
- "ctxfi", hw);
+ KBUILD_MODNAME, hw);
if (err < 0) {
printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
goto error2;
@@ -2023,13 +2159,16 @@
/* Reset all SRC pending interrupts */
hw_write_20kx(hw, SRC_IP, 0);
- /* TODO: detect the card ID and configure GPIO accordingly. */
- /* Configures GPIO (0xD802 0x98028) */
- /*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/
- /* Configures GPIO (SB0880) */
- /*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/
- hw_write_20kx(hw, GPIO_CTRL, 0xD802);
-
+ if (hw->model != CTSB1270) {
+ /* TODO: detect the card ID and configure GPIO accordingly. */
+ /* Configures GPIO (0xD802 0x98028) */
+ /*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/
+ /* Configures GPIO (SB0880) */
+ /*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/
+ hw_write_20kx(hw, GPIO_CTRL, 0xD802);
+ } else {
+ hw_write_20kx(hw, GPIO_CTRL, 0x9E5F);
+ }
/* Enable audio ring */
hw_write_20kx(hw, MIXER_AR_ENABLE, 0x01);
@@ -2106,7 +2245,11 @@
.pll_init = hw_pll_init,
.is_adc_source_selected = hw_is_adc_input_selected,
.select_adc_source = hw_adc_input_select,
- .have_digit_io_switch = hw_have_digit_io_switch,
+ .capabilities = hw_capabilities,
+ .output_switch_get = hw_output_switch_get,
+ .output_switch_put = hw_output_switch_put,
+ .mic_source_switch_get = hw_mic_source_switch_get,
+ .mic_source_switch_put = hw_mic_source_switch_put,
#ifdef CONFIG_PM
.suspend = hw_suspend,
.resume = hw_resume,
diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c
index c3519ff..0cc13ee 100644
--- a/sound/pci/ctxfi/ctmixer.c
+++ b/sound/pci/ctxfi/ctmixer.c
@@ -86,9 +86,7 @@
MIXER_LINEIN_C_S,
MIXER_MIC_C_S,
MIXER_SPDIFI_C_S,
- MIXER_LINEIN_P_S,
MIXER_SPDIFO_P_S,
- MIXER_SPDIFI_P_S,
MIXER_WAVEF_P_S,
MIXER_WAVER_P_S,
MIXER_WAVEC_P_S,
@@ -137,11 +135,11 @@
},
[MIXER_LINEIN_P] = {
.ctl = 1,
- .name = "Line-in Playback Volume",
+ .name = "Line Playback Volume",
},
[MIXER_LINEIN_C] = {
.ctl = 1,
- .name = "Line-in Capture Volume",
+ .name = "Line Capture Volume",
},
[MIXER_MIC_P] = {
.ctl = 1,
@@ -153,15 +151,15 @@
},
[MIXER_SPDIFI_P] = {
.ctl = 1,
- .name = "S/PDIF-in Playback Volume",
+ .name = "IEC958 Playback Volume",
},
[MIXER_SPDIFI_C] = {
.ctl = 1,
- .name = "S/PDIF-in Capture Volume",
+ .name = "IEC958 Capture Volume",
},
[MIXER_SPDIFO_P] = {
.ctl = 1,
- .name = "S/PDIF-out Playback Volume",
+ .name = "Digital Playback Volume",
},
[MIXER_WAVEF_P] = {
.ctl = 1,
@@ -179,14 +177,13 @@
.ctl = 1,
.name = "Surround Playback Volume",
},
-
[MIXER_PCM_C_S] = {
.ctl = 1,
.name = "PCM Capture Switch",
},
[MIXER_LINEIN_C_S] = {
.ctl = 1,
- .name = "Line-in Capture Switch",
+ .name = "Line Capture Switch",
},
[MIXER_MIC_C_S] = {
.ctl = 1,
@@ -194,19 +191,11 @@
},
[MIXER_SPDIFI_C_S] = {
.ctl = 1,
- .name = "S/PDIF-in Capture Switch",
- },
- [MIXER_LINEIN_P_S] = {
- .ctl = 1,
- .name = "Line-in Playback Switch",
+ .name = "IEC958 Capture Switch",
},
[MIXER_SPDIFO_P_S] = {
.ctl = 1,
- .name = "S/PDIF-out Playback Switch",
- },
- [MIXER_SPDIFI_P_S] = {
- .ctl = 1,
- .name = "S/PDIF-in Playback Switch",
+ .name = "Digital Playback Switch",
},
[MIXER_WAVEF_P_S] = {
.ctl = 1,
@@ -236,6 +225,8 @@
static void
ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type);
+/* FIXME: this static looks like it would fail if more than one card was */
+/* installed. */
static struct snd_kcontrol *kctls[2] = {NULL};
static enum CT_AMIXER_CTL get_amixer_index(enum CTALSA_MIXER_CTL alsa_index)
@@ -420,6 +411,77 @@
.tlv = { .p = ct_vol_db_scale },
};
+static int output_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *info)
+{
+ static const char *const names[3] = {
+ "FP Headphones", "Headphones", "Speakers"
+ };
+
+ return snd_ctl_enum_info(info, 1, 3, names);
+}
+
+static int output_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = atc->output_switch_get(atc);
+ return 0;
+}
+
+static int output_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ return atc->output_switch_put(atc, ucontrol->value.enumerated.item[0]);
+}
+
+static struct snd_kcontrol_new output_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Output Playback Enum",
+ .info = output_switch_info,
+ .get = output_switch_get,
+ .put = output_switch_put,
+};
+
+static int mic_source_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *info)
+{
+ static const char *const names[3] = {
+ "Mic", "FP Mic", "Aux"
+ };
+
+ return snd_ctl_enum_info(info, 1, 3, names);
+}
+
+static int mic_source_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = atc->mic_source_switch_get(atc);
+ return 0;
+}
+
+static int mic_source_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ return atc->mic_source_switch_put(atc,
+ ucontrol->value.enumerated.item[0]);
+}
+
+static struct snd_kcontrol_new mic_source_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Source Capture Enum",
+ .info = mic_source_switch_info,
+ .get = mic_source_switch_get,
+ .put = mic_source_switch_put,
+};
+
static void
do_line_mic_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type)
{
@@ -465,6 +527,7 @@
static void do_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type, int state)
{
struct ct_mixer *mixer = atc->mixer;
+ struct capabilities cap = atc->capabilities(atc);
/* Do changes in mixer. */
if ((SWH_CAPTURE_START <= type) && (SWH_CAPTURE_END >= type)) {
@@ -477,8 +540,17 @@
}
}
/* Do changes out of mixer. */
- if (state && (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type))
- do_line_mic_switch(atc, type);
+ if (!cap.dedicated_mic &&
+ (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type)) {
+ if (state)
+ do_line_mic_switch(atc, type);
+ atc->line_in_unmute(atc, state);
+ } else if (cap.dedicated_mic && (MIXER_LINEIN_C_S == type))
+ atc->line_in_unmute(atc, state);
+ else if (cap.dedicated_mic && (MIXER_MIC_C_S == type))
+ atc->mic_unmute(atc, state);
+ else if (MIXER_SPDIFI_C_S == type)
+ atc->spdif_in_unmute(atc, state);
else if (MIXER_WAVEF_P_S == type)
atc->line_front_unmute(atc, state);
else if (MIXER_WAVES_P_S == type)
@@ -487,12 +559,8 @@
atc->line_clfe_unmute(atc, state);
else if (MIXER_WAVER_P_S == type)
atc->line_rear_unmute(atc, state);
- else if (MIXER_LINEIN_P_S == type)
- atc->line_in_unmute(atc, state);
else if (MIXER_SPDIFO_P_S == type)
atc->spdif_out_unmute(atc, state);
- else if (MIXER_SPDIFI_P_S == type)
- atc->spdif_in_unmute(atc, state);
else if (MIXER_DIGITAL_IO_S == type)
do_digit_io_switch(atc, state);
@@ -671,6 +739,7 @@
{
enum CTALSA_MIXER_CTL type;
struct ct_atc *atc = mixer->atc;
+ struct capabilities cap = atc->capabilities(atc);
int err;
/* Create snd kcontrol instances on demand */
@@ -684,8 +753,8 @@
}
}
- ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl =
- atc->have_digit_io_switch(atc);
+ ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl = cap.digit_io_switch;
+
for (type = SWH_MIXER_START; type <= SWH_MIXER_END; type++) {
if (ct_kcontrol_init_table[type].ctl) {
swh_ctl.name = ct_kcontrol_init_table[type].name;
@@ -708,6 +777,17 @@
if (err)
return err;
+ if (cap.output_switch) {
+ err = ct_mixer_kcontrol_new(mixer, &output_ctl);
+ if (err)
+ return err;
+ }
+
+ if (cap.mic_source_switch) {
+ err = ct_mixer_kcontrol_new(mixer, &mic_source_ctl);
+ if (err)
+ return err;
+ }
atc->line_front_unmute(atc, 1);
set_switch_state(mixer, MIXER_WAVEF_P_S, 1);
atc->line_surround_unmute(atc, 0);
@@ -719,13 +799,12 @@
atc->spdif_out_unmute(atc, 0);
set_switch_state(mixer, MIXER_SPDIFO_P_S, 0);
atc->line_in_unmute(atc, 0);
- set_switch_state(mixer, MIXER_LINEIN_P_S, 0);
+ if (cap.dedicated_mic)
+ atc->mic_unmute(atc, 0);
atc->spdif_in_unmute(atc, 0);
- set_switch_state(mixer, MIXER_SPDIFI_P_S, 0);
-
- set_switch_state(mixer, MIXER_PCM_C_S, 1);
- set_switch_state(mixer, MIXER_LINEIN_C_S, 1);
- set_switch_state(mixer, MIXER_SPDIFI_C_S, 1);
+ set_switch_state(mixer, MIXER_PCM_C_S, 0);
+ set_switch_state(mixer, MIXER_LINEIN_C_S, 0);
+ set_switch_state(mixer, MIXER_SPDIFI_C_S, 0);
return 0;
}
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
index f42e7e1..b259aa0 100644
--- a/sound/pci/ctxfi/xfi.c
+++ b/sound/pci/ctxfi/xfi.c
@@ -80,11 +80,11 @@
"are 48000 and 44100, Value 48000 is assumed.\n");
reference_rate = 48000;
}
- if ((multiple != 1) && (multiple != 2)) {
+ if ((multiple != 1) && (multiple != 2) && (multiple != 4)) {
printk(KERN_ERR "ctxfi: Invalid multiple value %u!!!\n",
multiple);
printk(KERN_ERR "ctxfi: The valid values for multiple are "
- "1 and 2, Value 2 is assumed.\n");
+ "1, 2 and 4, Value 2 is assumed.\n");
multiple = 2;
}
err = ct_atc_create(card, pci, reference_rate, multiple,
@@ -143,7 +143,7 @@
#endif
static struct pci_driver ct_driver = {
- .name = "SB-XFi",
+ .name = KBUILD_MODNAME,
.id_table = ct_pci_dev_ids,
.probe = ct_card_probe,
.remove = __devexit_p(ct_card_remove),
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 20763dd..d730698 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1995,7 +1995,7 @@
ioremap_nocache(chip->dsp_registers_phys, sz);
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
- ECHOCARD_NAME, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_echo_free(chip);
snd_printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
@@ -2286,7 +2286,7 @@
kfree(commpage_bak);
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
- ECHOCARD_NAME, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_echo_free(chip);
snd_printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
@@ -2327,7 +2327,7 @@
/* pci_driver definition */
static struct pci_driver driver = {
- .name = "Echoaudio " ECHOCARD_NAME,
+ .name = KBUILD_MODNAME,
.id_table = snd_echo_ids,
.probe = snd_echo_probe,
.remove = __devexit_p(snd_echo_remove),
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index aff8387..a9c45d2 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -264,7 +264,7 @@
#endif
static struct pci_driver driver = {
- .name = "EMU10K1_Audigy",
+ .name = KBUILD_MODNAME,
.id_table = snd_emu10k1_ids,
.probe = snd_card_emu10k1_probe,
.remove = __devexit_p(snd_card_emu10k1_remove),
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 15f0161..fcd4935 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1912,7 +1912,7 @@
/* irq handler must be registered after I/O ports are activated */
if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
- "EMU10K1", emu)) {
+ KBUILD_MODNAME, emu)) {
err = -EBUSY;
goto error;
}
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 0c701e4..d4fde1b 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -925,7 +925,7 @@
}
if (request_irq(pci->irq, snd_emu10k1x_interrupt,
- IRQF_SHARED, "EMU10K1X", chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "emu10k1x: cannot grab irq %d\n", pci->irq);
snd_emu10k1x_free(chip);
return -EBUSY;
@@ -1613,7 +1613,7 @@
// pci_driver definition
static struct pci_driver driver = {
- .name = "EMU10K1X",
+ .name = KBUILD_MODNAME,
.id_table = snd_emu10k1x_ids,
.probe = snd_emu10k1x_probe,
.remove = __devexit_p(snd_emu10k1x_remove),
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 863eafe..f02e2f8 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2120,7 +2120,7 @@
}
ensoniq->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_audiopci_interrupt, IRQF_SHARED,
- "Ensoniq AudioPCI", ensoniq)) {
+ KBUILD_MODNAME, ensoniq)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_ensoniq_free(ensoniq);
return -EBUSY;
@@ -2489,7 +2489,7 @@
}
static struct pci_driver driver = {
- .name = DRIVER_NAME,
+ .name = KBUILD_MODNAME,
.id_table = snd_audiopci_ids,
.probe = snd_audiopci_probe,
.remove = __devexit_p(snd_audiopci_remove),
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 553b752..26a5a2f 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1514,7 +1514,7 @@
}
if (request_irq(pci->irq, snd_es1938_interrupt,
- IRQF_SHARED, "ES1938", chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "es1938: unable to grab IRQ %d, "
"disabling device\n", pci->irq);
snd_card_disconnect(card);
@@ -1636,7 +1636,7 @@
chip->mpu_port = pci_resource_start(pci, 3);
chip->game_port = pci_resource_start(pci, 4);
if (request_irq(pci->irq, snd_es1938_interrupt, IRQF_SHARED,
- "ES1938", chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_es1938_free(chip);
return -EBUSY;
@@ -1882,7 +1882,7 @@
}
static struct pci_driver driver = {
- .name = "ESS ES1938 (Solo-1)",
+ .name = KBUILD_MODNAME,
.id_table = snd_es1938_ids,
.probe = snd_es1938_probe,
.remove = __devexit_p(snd_es1938_remove),
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index ab0a615..99ea932 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -554,9 +554,8 @@
#else
struct snd_kcontrol *master_switch; /* for h/w volume control */
struct snd_kcontrol *master_volume;
- spinlock_t ac97_lock;
- struct tasklet_struct hwvol_tq;
#endif
+ struct work_struct hwvol_work;
#ifdef CONFIG_SND_ES1968_RADIO
struct snd_tea575x tea;
@@ -646,38 +645,23 @@
static void snd_es1968_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
{
struct es1968 *chip = ac97->private_data;
-#ifndef CONFIG_SND_ES1968_INPUT
- unsigned long flags;
-#endif
snd_es1968_ac97_wait(chip);
/* Write the bus */
-#ifndef CONFIG_SND_ES1968_INPUT
- spin_lock_irqsave(&chip->ac97_lock, flags);
-#endif
outw(val, chip->io_port + ESM_AC97_DATA);
/*msleep(1);*/
outb(reg, chip->io_port + ESM_AC97_INDEX);
/*msleep(1);*/
-#ifndef CONFIG_SND_ES1968_INPUT
- spin_unlock_irqrestore(&chip->ac97_lock, flags);
-#endif
}
static unsigned short snd_es1968_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
u16 data = 0;
struct es1968 *chip = ac97->private_data;
-#ifndef CONFIG_SND_ES1968_INPUT
- unsigned long flags;
-#endif
snd_es1968_ac97_wait(chip);
-#ifndef CONFIG_SND_ES1968_INPUT
- spin_lock_irqsave(&chip->ac97_lock, flags);
-#endif
outb(reg | 0x80, chip->io_port + ESM_AC97_INDEX);
/*msleep(1);*/
@@ -685,9 +669,6 @@
data = inw(chip->io_port + ESM_AC97_DATA);
/*msleep(1);*/
}
-#ifndef CONFIG_SND_ES1968_INPUT
- spin_unlock_irqrestore(&chip->ac97_lock, flags);
-#endif
return data;
}
@@ -1904,13 +1885,10 @@
(without wrap around) in response to volume button presses and then
generating an interrupt. The pair of counters is stored in bits 1-3 and 5-7
of a byte wide register. The meaning of bits 0 and 4 is unknown. */
-static void es1968_update_hw_volume(unsigned long private_data)
+static void es1968_update_hw_volume(struct work_struct *work)
{
- struct es1968 *chip = (struct es1968 *) private_data;
+ struct es1968 *chip = container_of(work, struct es1968, hwvol_work);
int x, val;
-#ifndef CONFIG_SND_ES1968_INPUT
- unsigned long flags;
-#endif
/* Figure out which volume control button was pushed,
based on differences from the default register
@@ -1929,18 +1907,11 @@
if (! chip->master_switch || ! chip->master_volume)
return;
- /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
- spin_lock_irqsave(&chip->ac97_lock, flags);
- val = chip->ac97->regs[AC97_MASTER];
+ val = snd_ac97_read(chip->ac97, AC97_MASTER);
switch (x) {
case 0x88:
/* mute */
val ^= 0x8000;
- chip->ac97->regs[AC97_MASTER] = val;
- outw(val, chip->io_port + ESM_AC97_DATA);
- outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &chip->master_switch->id);
break;
case 0xaa:
/* volume up */
@@ -1948,11 +1919,6 @@
val--;
if ((val & 0x7f00) > 0)
val -= 0x0100;
- chip->ac97->regs[AC97_MASTER] = val;
- outw(val, chip->io_port + ESM_AC97_DATA);
- outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &chip->master_volume->id);
break;
case 0x66:
/* volume down */
@@ -1960,14 +1926,11 @@
val++;
if ((val & 0x7f00) < 0x1f00)
val += 0x0100;
- chip->ac97->regs[AC97_MASTER] = val;
- outw(val, chip->io_port + ESM_AC97_DATA);
- outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &chip->master_volume->id);
break;
}
- spin_unlock_irqrestore(&chip->ac97_lock, flags);
+ if (snd_ac97_update(chip->ac97, AC97_MASTER, val))
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
#else
if (!chip->input_dev)
return;
@@ -2013,11 +1976,7 @@
outw(inw(chip->io_port + 4) & 1, chip->io_port + 4);
if (event & ESM_HWVOL_IRQ)
-#ifdef CONFIG_SND_ES1968_INPUT
- es1968_update_hw_volume((unsigned long)chip);
-#else
- tasklet_schedule(&chip->hwvol_tq); /* we'll do this later */
-#endif
+ schedule_work(&chip->hwvol_work);
/* else ack 'em all, i imagine */
outb(0xFF, chip->io_port + 0x1A);
@@ -2426,6 +2385,7 @@
return 0;
chip->in_suspend = 1;
+ cancel_work_sync(&chip->hwvol_work);
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
snd_pcm_suspend_all(chip->pcm);
snd_ac97_suspend(chip->ac97);
@@ -2638,6 +2598,7 @@
static int snd_es1968_free(struct es1968 *chip)
{
+ cancel_work_sync(&chip->hwvol_work);
#ifdef CONFIG_SND_ES1968_INPUT
if (chip->input_dev)
input_unregister_device(chip->input_dev);
@@ -2728,10 +2689,7 @@
INIT_LIST_HEAD(&chip->buf_list);
INIT_LIST_HEAD(&chip->substream_list);
mutex_init(&chip->memory_mutex);
-#ifndef CONFIG_SND_ES1968_INPUT
- spin_lock_init(&chip->ac97_lock);
- tasklet_init(&chip->hwvol_tq, es1968_update_hw_volume, (unsigned long)chip);
-#endif
+ INIT_WORK(&chip->hwvol_work, es1968_update_hw_volume);
chip->card = card;
chip->pci = pci;
chip->irq = -1;
@@ -2746,7 +2704,7 @@
}
chip->io_port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_es1968_interrupt, IRQF_SHARED,
- "ESS Maestro", chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_es1968_free(chip);
return -EBUSY;
@@ -2925,7 +2883,7 @@
}
static struct pci_driver driver = {
- .name = "ES1968 (ESS Maestro)",
+ .name = KBUILD_MODNAME,
.id_table = snd_es1968_ids,
.probe = snd_es1968_probe,
.remove = __devexit_p(snd_es1968_remove),
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index a7ec703..f9123f0 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1199,7 +1199,7 @@
chip->port = pci_resource_start(pci, 0);
if ((tea575x_tuner & TUNER_ONLY) == 0) {
if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED,
- "FM801", chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
snd_fm801_free(chip);
return -EBUSY;
@@ -1394,7 +1394,7 @@
#endif
static struct pci_driver driver = {
- .name = "FM801",
+ .name = KBUILD_MODNAME,
.id_table = snd_fm801_ids,
.probe = snd_card_fm801_probe,
.remove = __devexit_p(snd_card_fm801_remove),
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 0ea5cc6..7489b46 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -14,6 +14,19 @@
if SND_HDA_INTEL
+config SND_HDA_PREALLOC_SIZE
+ int "Pre-allocated buffer size for HD-audio driver"
+ range 0 32768
+ default 64
+ help
+ Specifies the default pre-allocated buffer-size in kB for the
+ HD-audio driver. A larger buffer (e.g. 2048) is preferred
+ for systems using PulseAudio. The default 64 is chosen just
+ for compatibility reasons.
+
+ Note that the pre-allocation size can be changed dynamically
+ via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too.
+
config SND_HDA_HWDEP
bool "Build hwdep interface for HD-audio driver"
select SND_HWDEP
@@ -83,6 +96,19 @@
snd-hda-codec-realtek.
This module is automatically loaded at probing.
+config SND_HDA_ENABLE_REALTEK_QUIRKS
+ bool "Build static quirks for Realtek codecs"
+ depends on SND_HDA_CODEC_REALTEK
+ default y
+ help
+ Say Y here to build the static quirks codes for Realtek codecs.
+ If you need the "model" preset that the default BIOS auto-parser
+ can't handle, turn this option on.
+
+ If your device works with model=auto option, basically you don't
+ need the quirk code. By turning this off, you can reduce the
+ module size quite a lot.
+
config SND_HDA_CODEC_ANALOG
bool "Build Analog Device HD-audio codec support"
default y
@@ -171,6 +197,19 @@
snd-hda-codec-ca0110.
This module is automatically loaded at probing.
+config SND_HDA_CODEC_CA0132
+ bool "Build Creative CA0132 codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Creative CA0132 codec support in
+ snd-hda-intel driver.
+
+ When the HD-audio driver is built as a module, the codec
+ support code is also built as another module,
+ snd-hda-codec-ca0132.
+ This module is automatically loaded at probing.
+
config SND_HDA_CODEC_CMEDIA
bool "Build C-Media HD-audio codec support"
default y
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 17ef365..87365d5 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -13,6 +13,7 @@
snd-hda-codec-si3054-objs := patch_si3054.o
snd-hda-codec-cirrus-objs := patch_cirrus.o
snd-hda-codec-ca0110-objs := patch_ca0110.o
+snd-hda-codec-ca0132-objs := patch_ca0132.o
snd-hda-codec-conexant-objs := patch_conexant.o
snd-hda-codec-via-objs := patch_via.o
snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o
@@ -42,6 +43,9 @@
ifdef CONFIG_SND_HDA_CODEC_CA0110
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
endif
+ifdef CONFIG_SND_HDA_CODEC_CA0132
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o
+endif
ifdef CONFIG_SND_HDA_CODEC_CONEXANT
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
endif
diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c
new file mode 100644
index 0000000..21ec2cb
--- /dev/null
+++ b/sound/pci/hda/alc260_quirks.c
@@ -0,0 +1,1272 @@
+/*
+ * ALC260 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC260 models */
+enum {
+ ALC260_AUTO,
+ ALC260_BASIC,
+ ALC260_HP,
+ ALC260_HP_DC7600,
+ ALC260_HP_3013,
+ ALC260_FUJITSU_S702X,
+ ALC260_ACER,
+ ALC260_WILL,
+ ALC260_REPLACER_672V,
+ ALC260_FAVORIT100,
+#ifdef CONFIG_SND_DEBUG
+ ALC260_TEST,
+#endif
+ ALC260_MODEL_LAST /* last tag */
+};
+
+static const hda_nid_t alc260_dac_nids[1] = {
+ /* front */
+ 0x02,
+};
+
+static const hda_nid_t alc260_adc_nids[1] = {
+ /* ADC0 */
+ 0x04,
+};
+
+static const hda_nid_t alc260_adc_nids_alt[1] = {
+ /* ADC1 */
+ 0x05,
+};
+
+/* NIDs used when simultaneous access to both ADCs makes sense. Note that
+ * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
+ */
+static const hda_nid_t alc260_dual_adc_nids[2] = {
+ /* ADC0, ADC1 */
+ 0x04, 0x05
+};
+
+#define ALC260_DIGOUT_NID 0x03
+#define ALC260_DIGIN_NID 0x06
+
+static const struct hda_input_mux alc260_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
+ * headphone jack and the internal CD lines since these are the only pins at
+ * which audio can appear. For flexibility, also allow the option of
+ * recording the mixer output on the second ADC (ADC0 doesn't have a
+ * connection to the mixer output).
+ */
+static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
+ {
+ .num_items = 3,
+ .items = {
+ { "Mic/Line", 0x0 },
+ { "CD", 0x4 },
+ { "Headphone", 0x2 },
+ },
+ },
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic/Line", 0x0 },
+ { "CD", 0x4 },
+ { "Headphone", 0x2 },
+ { "Mixer", 0x5 },
+ },
+ },
+
+};
+
+/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
+ * the Fujitsu S702x, but jacks are marked differently.
+ */
+static const struct hda_input_mux alc260_acer_capture_sources[2] = {
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Headphone", 0x5 },
+ },
+ },
+ {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Headphone", 0x6 },
+ { "Mixer", 0x5 },
+ },
+ },
+};
+
+/* Maxdata Favorit 100XS */
+static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
+ {
+ .num_items = 2,
+ .items = {
+ { "Line/Mic", 0x0 },
+ { "CD", 0x4 },
+ },
+ },
+ {
+ .num_items = 3,
+ .items = {
+ { "Line/Mic", 0x0 },
+ { "CD", 0x4 },
+ { "Mixer", 0x5 },
+ },
+ },
+};
+
+/*
+ * This is just place-holder, so there's something for alc_build_pcms to look
+ * at when it calculates the maximum number of channels. ALC260 has no mixer
+ * element which allows changing the channel mode, so the verb list is
+ * never used.
+ */
+static const struct hda_channel_mode alc260_modes[1] = {
+ { 2, NULL },
+};
+
+
+/* Mixer combinations
+ *
+ * basic: base_output + input + pc_beep + capture
+ * HP: base_output + input + capture_alt
+ * HP_3013: hp_3013 + input + capture
+ * fujitsu: fujitsu + capture
+ * acer: acer + capture
+ */
+
+static const struct snd_kcontrol_new alc260_base_output_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc260_input_mixer[] = {
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+/* update HP, line and mono out pins according to the master switch */
+static void alc260_hp_master_update(struct hda_codec *codec)
+{
+ update_speakers(codec);
+}
+
+static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ *ucontrol->value.integer.value = !spec->master_mute;
+ return 0;
+}
+
+static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int val = !*ucontrol->value.integer.value;
+
+ if (val == spec->master_mute)
+ return 0;
+ spec->master_mute = val;
+ alc260_hp_master_update(codec);
+ return 1;
+}
+
+static const struct snd_kcontrol_new alc260_hp_output_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc260_hp_unsol_verbs[] = {
+ {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x0f;
+ spec->autocfg.speaker_pins[0] = 0x10;
+ spec->autocfg.speaker_pins[1] = 0x11;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static const struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static void alc260_hp_3013_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x10;
+ spec->autocfg.speaker_pins[1] = 0x11;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static const struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc260_dc7600_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol),
+ HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc260_hp_3013_unsol_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_3012_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x10;
+ spec->autocfg.speaker_pins[0] = 0x0f;
+ spec->autocfg.speaker_pins[1] = 0x11;
+ spec->autocfg.speaker_pins[2] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
+ * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
+ */
+static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
+ ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+ { } /* end */
+};
+
+/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
+ * versions of the ALC260 don't act on requests to enable mic bias from NID
+ * 0x0f (used to drive the headphone jack in these laptops). The ALC260
+ * datasheet doesn't mention this restriction. At this stage it's not clear
+ * whether this behaviour is intentional or is a hardware bug in chip
+ * revisions available in early 2006. Therefore for now allow the
+ * "Headphone Jack Mode" control to span all choices, but if it turns out
+ * that the lack of mic bias for this NID is intentional we could change the
+ * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ *
+ * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
+ * don't appear to make the mic bias available from the "line" jack, even
+ * though the NID used for this jack (0x14) can supply it. The theory is
+ * that perhaps Acer have included blocking capacitors between the ALC260
+ * and the output jack. If this turns out to be the case for all such
+ * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
+ * to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ *
+ * The C20x Tablet series have a mono internal speaker which is controlled
+ * via the chip's Mono sum widget and pin complex, so include the necessary
+ * controls for such models. On models without a "mono speaker" the control
+ * won't do anything.
+ */
+static const struct snd_kcontrol_new alc260_acer_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
+ ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
+ HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ { } /* end */
+};
+
+/* Maxdata Favorit 100XS: one output and one input (0x12) jack
+ */
+static const struct snd_kcontrol_new alc260_favorit100_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
+ ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ { } /* end */
+};
+
+/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
+ * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
+ */
+static const struct snd_kcontrol_new alc260_will_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ { } /* end */
+};
+
+/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
+ * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
+ */
+static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ { } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+static const struct hda_verb alc260_init_verbs[] = {
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /* LINE-2 is used for line-out in rear */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* select line-out */
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* LINE-OUT pin */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* enable HP */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* enable Mono */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* mute capture amp left and right */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* set connection select to line in (default select for this ADC) */
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* set vol=0 Line-Out mixer amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* set vol=0 HP mixer amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* set vol=0 Mono mixer amp left and right */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* unmute LINE-2 out pin */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* mute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* mute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* mute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { }
+};
+
+#if 0 /* should be identical with alc260_init_verbs? */
+static const struct hda_verb alc260_hp_init_verbs[] = {
+ /* Headphone and output */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ /* mono output */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* Line-2 pin widget for output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* unmute amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* unmute Line-Out mixer amp left and right (volume = 0) */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* unmute HP mixer amp left and right (volume = 0) */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* Unmute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ { }
+};
+#endif
+
+static const struct hda_verb alc260_hp_3013_init_verbs[] = {
+ /* Line out and output */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* mono output */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* Headphone pin widget for output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* unmute amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* unmute Line-Out mixer amp left and right (volume = 0) */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* unmute HP mixer amp left and right (volume = 0) */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* Unmute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ { }
+};
+
+/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
+ * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
+ * audio = 0x16, internal speaker = 0x10.
+ */
+static const struct hda_verb alc260_fujitsu_init_verbs[] = {
+ /* Disable all GPIOs */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0},
+ /* Internal speaker is connected to headphone pin */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Headphone/Line-out jack connects to Line1 pin; make it an output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Mic/Line-in jack is connected to mic1 pin, so make it an input */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Ensure all other unused pins are disabled and muted. */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+ /* Disable digital (SPDIF) pins */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure Line1 pin widget takes its input from the OUT1 sum bus
+ * when acting as an output.
+ */
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Line1 pin widget output buffer since it starts as an output.
+ * If the pin mode is changed by the user the pin mode control will
+ * take care of enabling the pin's input/output buffers as needed.
+ * Therefore there's no need to enable the input buffer at this
+ * stage.
+ */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute input buffer of pin widget used for Line-in (no equiv
+ * mixer ctrl)
+ */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to match default mixer setting - line
+ * in (on mic1 pin)
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Do the same for the second ADC: mute capture input amp and
+ * set ADC connection to line in (on mic1 pin)
+ */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+
+/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
+ * similar laptops (adapted from Fujitsu init verbs).
+ */
+static const struct hda_verb alc260_acer_init_verbs[] = {
+ /* On TravelMate laptops, GPIO 0 enables the internal speaker and
+ * the headphone jack. Turn this on and rely on the standard mute
+ * methods whenever the user wants to turn these outputs off.
+ */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+ /* Internal speaker/Headphone jack is connected to Line-out pin */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Internal microphone/Mic jack is connected to Mic1 pin */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ /* Line In jack is connected to Line1 pin */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Ensure all other unused pins are disabled and muted. */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Disable digital (SPDIF) pins */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
+ * bus when acting as outputs.
+ */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Unmute Line-out pin widget amp left and right
+ * (no equiv mixer ctrl)
+ */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute mono pin widget amp output (no equiv mixer ctrl) */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Mic1 and Line1 pin widget input buffers since they start as
+ * inputs. If the pin mode is changed by the user the pin mode control
+ * will take care of enabling the pin's input/output buffers as needed.
+ * Therefore there's no need to enable the input buffer at this
+ * stage.
+ */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to match default mixer setting - mic
+ * (on mic1 pin)
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Do similar with the second ADC: mute capture input amp and
+ * set ADC connection to mic to match ALSA's default state.
+ */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+
+/* Initialisation sequence for Maxdata Favorit 100XS
+ * (adapted from Acer init verbs).
+ */
+static const struct hda_verb alc260_favorit100_init_verbs[] = {
+ /* GPIO 0 enables the output jack.
+ * Turn this on and rely on the standard mute
+ * methods whenever the user wants to turn these outputs off.
+ */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+ /* Line/Mic input jack is connected to Mic1 pin */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ /* Ensure all other unused pins are disabled and muted. */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Disable digital (SPDIF) pins */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
+ * bus when acting as outputs.
+ */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Unmute Line-out pin widget amp left and right
+ * (no equiv mixer ctrl)
+ */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Mic1 and Line1 pin widget input buffers since they start as
+ * inputs. If the pin mode is changed by the user the pin mode control
+ * will take care of enabling the pin's input/output buffers as needed.
+ * Therefore there's no need to enable the input buffer at this
+ * stage.
+ */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to match default mixer setting - mic
+ * (on mic1 pin)
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Do similar with the second ADC: mute capture input amp and
+ * set ADC connection to mic to match ALSA's default state.
+ */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+
+static const struct hda_verb alc260_will_verbs[] = {
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
+ {}
+};
+
+static const struct hda_verb alc260_replacer_672v_verbs[] = {
+ {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
+
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
+
+ {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc260_replacer_672v_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
+ present = snd_hda_jack_detect(codec, 0x0f);
+ if (present) {
+ snd_hda_codec_write_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 1);
+ snd_hda_codec_write_cache(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_HP);
+ } else {
+ snd_hda_codec_write_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 0);
+ snd_hda_codec_write_cache(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_OUT);
+ }
+}
+
+static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC_HP_EVENT)
+ alc260_replacer_672v_automute(codec);
+}
+
+static const struct hda_verb alc260_hp_dc7600_verbs[] = {
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+/* Test configuration for debugging, modelled after the ALC880 test
+ * configuration.
+ */
+#ifdef CONFIG_SND_DEBUG
+static const hda_nid_t alc260_test_dac_nids[1] = {
+ 0x02,
+};
+static const hda_nid_t alc260_test_adc_nids[2] = {
+ 0x04, 0x05,
+};
+/* For testing the ALC260, each input MUX needs its own definition since
+ * the signal assignments are different. This assumes that the first ADC
+ * is NID 0x04.
+ */
+static const struct hda_input_mux alc260_test_capture_sources[2] = {
+ {
+ .num_items = 7,
+ .items = {
+ { "MIC1 pin", 0x0 },
+ { "MIC2 pin", 0x1 },
+ { "LINE1 pin", 0x2 },
+ { "LINE2 pin", 0x3 },
+ { "CD pin", 0x4 },
+ { "LINE-OUT pin", 0x5 },
+ { "HP-OUT pin", 0x6 },
+ },
+ },
+ {
+ .num_items = 8,
+ .items = {
+ { "MIC1 pin", 0x0 },
+ { "MIC2 pin", 0x1 },
+ { "LINE1 pin", 0x2 },
+ { "LINE2 pin", 0x3 },
+ { "CD pin", 0x4 },
+ { "Mixer", 0x5 },
+ { "LINE-OUT pin", 0x6 },
+ { "HP-OUT pin", 0x7 },
+ },
+ },
+};
+static const struct snd_kcontrol_new alc260_test_mixer[] = {
+ /* Output driver widgets */
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
+
+ /* Modes for retasking pin widgets
+ * Note: the ALC260 doesn't seem to act on requests to enable mic
+ * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
+ * mention this restriction. At this stage it's not clear whether
+ * this behaviour is intentional or is a hardware bug in chip
+ * revisions available at least up until early 2006. Therefore for
+ * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
+ * choices, but if it turns out that the lack of mic bias for these
+ * NIDs is intentional we could change their modes from
+ * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ */
+ ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
+
+ /* Loopback mixer controls */
+ HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
+ HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
+
+ /* Controls for GPIO pins, assuming they are configured as outputs */
+ ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
+
+ /* Switches to allow the digital IO pins to be enabled. The datasheet
+ * is ambigious as to which NID is which; testing on laptops which
+ * make this output available should provide clarification.
+ */
+ ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
+ ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
+
+ /* A switch allowing EAPD to be enabled. Some laptops seem to use
+ * this output to turn on an external amplifier.
+ */
+ ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
+ ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
+
+ { } /* end */
+};
+static const struct hda_verb alc260_test_init_verbs[] = {
+ /* Enable all GPIOs as outputs with an initial value of 0 */
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
+
+ /* Enable retasking pins as output, initially without power amp */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* Disable digital (SPDIF) pins initially, but users can enable
+ * them via a mixer switch. In the case of SPDIF-out, this initverb
+ * payload also sets the generation to 0, output to be in "consumer"
+ * PCM format, copyright asserted, no pre-emphasis and no validity
+ * control.
+ */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
+ * OUT1 sum bus when acting as an output.
+ */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Unmute retasking pin widget output buffers since the default
+ * state appears to be output. As the pin mode is changed by the
+ * user the pin mode control will take care of enabling the pin's
+ * input/output buffers as needed.
+ */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Also unmute the mono-out pin widget */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to match default mixer setting (mic1
+ * pin)
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Do the same for the second ADC: mute capture input amp and
+ * set ADC connection to mic1 pin
+ */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+#endif
+
+/*
+ * ALC260 configurations
+ */
+static const char * const alc260_models[ALC260_MODEL_LAST] = {
+ [ALC260_BASIC] = "basic",
+ [ALC260_HP] = "hp",
+ [ALC260_HP_3013] = "hp-3013",
+ [ALC260_HP_DC7600] = "hp-dc7600",
+ [ALC260_FUJITSU_S702X] = "fujitsu",
+ [ALC260_ACER] = "acer",
+ [ALC260_WILL] = "will",
+ [ALC260_REPLACER_672V] = "replacer",
+ [ALC260_FAVORIT100] = "favorit100",
+#ifdef CONFIG_SND_DEBUG
+ [ALC260_TEST] = "test",
+#endif
+ [ALC260_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc260_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
+ SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
+ SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
+ SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
+ SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
+ SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
+ SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
+ SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
+ SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
+ SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
+ SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
+ SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
+ SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
+ {}
+};
+
+static const struct alc_config_preset alc260_presets[] = {
+ [ALC260_BASIC] = {
+ .mixers = { alc260_base_output_mixer,
+ alc260_input_mixer },
+ .init_verbs = { alc260_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+ .adc_nids = alc260_dual_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ },
+ [ALC260_HP] = {
+ .mixers = { alc260_hp_output_mixer,
+ alc260_input_mixer },
+ .init_verbs = { alc260_init_verbs,
+ alc260_hp_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+ .adc_nids = alc260_adc_nids_alt,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc260_hp_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC260_HP_DC7600] = {
+ .mixers = { alc260_hp_dc7600_mixer,
+ alc260_input_mixer },
+ .init_verbs = { alc260_init_verbs,
+ alc260_hp_dc7600_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+ .adc_nids = alc260_adc_nids_alt,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc260_hp_3012_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC260_HP_3013] = {
+ .mixers = { alc260_hp_3013_mixer,
+ alc260_input_mixer },
+ .init_verbs = { alc260_hp_3013_init_verbs,
+ alc260_hp_3013_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+ .adc_nids = alc260_adc_nids_alt,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc260_hp_3013_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC260_FUJITSU_S702X] = {
+ .mixers = { alc260_fujitsu_mixer },
+ .init_verbs = { alc260_fujitsu_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+ .adc_nids = alc260_dual_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
+ .input_mux = alc260_fujitsu_capture_sources,
+ },
+ [ALC260_ACER] = {
+ .mixers = { alc260_acer_mixer },
+ .init_verbs = { alc260_acer_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+ .adc_nids = alc260_dual_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
+ .input_mux = alc260_acer_capture_sources,
+ },
+ [ALC260_FAVORIT100] = {
+ .mixers = { alc260_favorit100_mixer },
+ .init_verbs = { alc260_favorit100_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+ .adc_nids = alc260_dual_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
+ .input_mux = alc260_favorit100_capture_sources,
+ },
+ [ALC260_WILL] = {
+ .mixers = { alc260_will_mixer },
+ .init_verbs = { alc260_init_verbs, alc260_will_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+ .adc_nids = alc260_adc_nids,
+ .dig_out_nid = ALC260_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ },
+ [ALC260_REPLACER_672V] = {
+ .mixers = { alc260_replacer_672v_mixer },
+ .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+ .adc_nids = alc260_adc_nids,
+ .dig_out_nid = ALC260_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc260_replacer_672v_unsol_event,
+ .init_hook = alc260_replacer_672v_automute,
+ },
+#ifdef CONFIG_SND_DEBUG
+ [ALC260_TEST] = {
+ .mixers = { alc260_test_mixer },
+ .init_verbs = { alc260_test_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
+ .dac_nids = alc260_test_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
+ .adc_nids = alc260_test_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
+ .input_mux = alc260_test_capture_sources,
+ },
+#endif
+};
+
diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c
new file mode 100644
index 0000000..8d2097d
--- /dev/null
+++ b/sound/pci/hda/alc262_quirks.c
@@ -0,0 +1,1353 @@
+/*
+ * ALC262 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC262 models */
+enum {
+ ALC262_AUTO,
+ ALC262_BASIC,
+ ALC262_HIPPO,
+ ALC262_HIPPO_1,
+ ALC262_FUJITSU,
+ ALC262_HP_BPC,
+ ALC262_HP_BPC_D7000_WL,
+ ALC262_HP_BPC_D7000_WF,
+ ALC262_HP_TC_T5735,
+ ALC262_HP_RP5700,
+ ALC262_BENQ_ED8,
+ ALC262_SONY_ASSAMD,
+ ALC262_BENQ_T31,
+ ALC262_ULTRA,
+ ALC262_LENOVO_3000,
+ ALC262_NEC,
+ ALC262_TOSHIBA_S06,
+ ALC262_TOSHIBA_RX1,
+ ALC262_TYAN,
+ ALC262_MODEL_LAST /* last tag */
+};
+
+#define ALC262_DIGOUT_NID ALC880_DIGOUT_NID
+#define ALC262_DIGIN_NID ALC880_DIGIN_NID
+
+#define alc262_dac_nids alc260_dac_nids
+#define alc262_adc_nids alc882_adc_nids
+#define alc262_adc_nids_alt alc882_adc_nids_alt
+#define alc262_capsrc_nids alc882_capsrc_nids
+#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt
+
+#define alc262_modes alc260_modes
+#define alc262_capture_source alc882_capture_source
+
+static const hda_nid_t alc262_dmic_adc_nids[1] = {
+ /* ADC0 */
+ 0x09
+};
+
+static const hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
+
+static const struct snd_kcontrol_new alc262_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/* update HP, line and mono-out pins according to the master switch */
+#define alc262_hp_master_update alc260_hp_master_update
+
+static void alc262_hp_bpc_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x16;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static void alc262_hp_wildwest_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x16;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+#define alc262_hp_master_sw_get alc260_hp_master_sw_get
+#define alc262_hp_master_sw_put alc260_hp_master_sw_put
+
+#define ALC262_HP_MASTER_SWITCH \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Master Playback Switch", \
+ .info = snd_ctl_boolean_mono_info, \
+ .get = alc262_hp_master_sw_get, \
+ .put = alc262_hp_master_sw_put, \
+ }, \
+ { \
+ .iface = NID_MAPPING, \
+ .name = "Master Playback Switch", \
+ .private_value = 0x15 | (0x16 << 8) | (0x1b << 16), \
+ }
+
+
+static const struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
+ ALC262_HP_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
+ ALC262_HP_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x1a, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
+ HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_hp_t5735_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static const struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_hp_t5735_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static const struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_hp_rp5700_verbs[] = {
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {}
+};
+
+static const struct hda_input_mux alc262_hp_rp5700_capture_source = {
+ .num_items = 1,
+ .items = {
+ { "Line", 0x1 },
+ },
+};
+
+/* bind hp and internal speaker mute (with plug check) as master switch */
+#define alc262_hippo_master_update alc262_hp_master_update
+#define alc262_hippo_master_sw_get alc262_hp_master_sw_get
+#define alc262_hippo_master_sw_put alc262_hp_master_sw_put
+
+#define ALC262_HIPPO_MASTER_SWITCH \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Master Playback Switch", \
+ .info = snd_ctl_boolean_mono_info, \
+ .get = alc262_hippo_master_sw_get, \
+ .put = alc262_hippo_master_sw_put, \
+ }, \
+ { \
+ .iface = NID_MAPPING, \
+ .name = "Master Playback Switch", \
+ .subdevice = SUBDEV_HP(0) | (SUBDEV_LINE(0) << 8) | \
+ (SUBDEV_SPEAKER(0) << 16), \
+ }
+
+static const struct snd_kcontrol_new alc262_hippo_mixer[] = {
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc262_hippo1_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_hippo_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc262_hippo1_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+
+static const struct snd_kcontrol_new alc262_sony_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc262_tyan_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_tyan_verbs[] = {
+ /* Headphone automute */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* P11 AUX_IN, white 4-pin connector */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1},
+ {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93},
+ {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19},
+
+ {}
+};
+
+/* unsolicited event for HP jack sensing */
+static void alc262_tyan_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+
+#define alc262_capture_mixer alc882_capture_mixer
+#define alc262_capture_alt_mixer alc882_capture_alt_mixer
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc262_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+
+ { }
+};
+
+static const struct hda_verb alc262_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static const struct hda_verb alc262_hippo1_unsol_verbs[] = {
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static const struct hda_verb alc262_sony_unsol_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, // Front Mic
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static const struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_toshiba_s06_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x22, AC_VERB_SET_CONNECT_SEL, 0x09},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static void alc262_toshiba_s06_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+/*
+ * nec model
+ * 0x15 = headphone
+ * 0x16 = internal speaker
+ * 0x18 = external mic
+ */
+
+static const struct snd_kcontrol_new alc262_nec_mixer[] = {
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_nec_verbs[] = {
+ /* Unmute Speaker */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Headphone */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* External mic to headphone */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* External mic to speaker */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {}
+};
+
+/*
+ * fujitsu model
+ * 0x14 = headphone/spdif-out, 0x15 = internal speaker,
+ * 0x1b = port replicator headphone out
+ */
+
+static const struct hda_verb alc262_fujitsu_unsol_verbs[] = {
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static const struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static const struct hda_verb alc262_lenovo_3000_init_verbs[] = {
+ /* Front Mic pin: input vref at 50% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {}
+};
+
+static const struct hda_input_mux alc262_fujitsu_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc262_HP_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "AUX IN", 0x6 },
+ },
+};
+
+static const struct hda_input_mux alc262_HP_D7000_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x2 },
+ { "Line", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
+static void alc262_fujitsu_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.hp_pins[1] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/* bind volumes of both NID 0x0c and 0x0d */
+static const struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc262_hp_master_sw_get,
+ .put = alc262_hp_master_sw_put,
+ },
+ {
+ .iface = NID_MAPPING,
+ .name = "Master Playback Switch",
+ .private_value = 0x1b,
+ },
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static void alc262_lenovo_3000_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc262_hp_master_sw_get,
+ .put = alc262_hp_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* additional init verbs for Benq laptops */
+static const struct hda_verb alc262_EAPD_verbs[] = {
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
+ {}
+};
+
+static const struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+ {}
+};
+
+/* Samsung Q1 Ultra Vista model setup */
+static const struct snd_kcontrol_new alc262_ultra_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Mic Boost Volume", 0x15, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_ultra_verbs[] = {
+ /* output mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* speaker */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ /* internal mic */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* ADC, choose mic */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)},
+ {}
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_ultra_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+
+ mute = 0;
+ /* auto-mute only when HP is used as HP */
+ if (!spec->cur_mux[0]) {
+ spec->jack_present = snd_hda_jack_detect(codec, 0x15);
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE;
+ }
+ /* mute/unmute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ /* mute/unmute HP */
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE);
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_ultra_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC_HP_EVENT)
+ return;
+ alc262_ultra_automute(codec);
+}
+
+static const struct hda_input_mux alc262_ultra_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Headphone", 0x7 },
+ },
+};
+
+static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int ret;
+
+ ret = alc_mux_enum_put(kcontrol, ucontrol);
+ if (!ret)
+ return 0;
+ /* reprogram the HP pin as mic or HP according to the input source */
+ snd_hda_codec_write_cache(codec, 0x15, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->cur_mux[0] ? PIN_VREF80 : PIN_HP);
+ alc262_ultra_automute(codec); /* mute/unmute HP */
+ return ret;
+}
+
+static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc262_ultra_mux_enum_put,
+ },
+ {
+ .iface = NID_MAPPING,
+ .name = "Capture Source",
+ .private_value = 0x15,
+ },
+ { } /* end */
+};
+
+static const struct hda_verb alc262_HP_BPC_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */
+ /* Input mixer1: only unmute Mic */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { }
+};
+
+static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for front
+ * panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Mono */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* rear MIC */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* Line in */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Line out */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD in */
+
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, /*rear MIC*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/
+ /* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { }
+};
+
+static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc262_models[ALC262_MODEL_LAST] = {
+ [ALC262_BASIC] = "basic",
+ [ALC262_HIPPO] = "hippo",
+ [ALC262_HIPPO_1] = "hippo_1",
+ [ALC262_FUJITSU] = "fujitsu",
+ [ALC262_HP_BPC] = "hp-bpc",
+ [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
+ [ALC262_HP_TC_T5735] = "hp-tc-t5735",
+ [ALC262_HP_RP5700] = "hp-rp5700",
+ [ALC262_BENQ_ED8] = "benq",
+ [ALC262_BENQ_T31] = "benq-t31",
+ [ALC262_SONY_ASSAMD] = "sony-assamd",
+ [ALC262_TOSHIBA_S06] = "toshiba-s06",
+ [ALC262_TOSHIBA_RX1] = "toshiba-rx1",
+ [ALC262_ULTRA] = "ultra",
+ [ALC262_LENOVO_3000] = "lenovo-3000",
+ [ALC262_NEC] = "nec",
+ [ALC262_TYAN] = "tyan",
+ [ALC262_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc262_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
+ SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+ ALC262_AUTO),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
+ ALC262_HP_TC_T5735),
+ SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
+ SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
+ SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
+ SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+ SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
+#if 0 /* disable the quirk since model=auto works better in recent versions */
+ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
+ ALC262_SONY_ASSAMD),
+#endif
+ SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
+ ALC262_TOSHIBA_RX1),
+ SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
+ SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
+ SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
+ SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN),
+ SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1",
+ ALC262_ULTRA),
+ SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
+ SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
+ SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
+ SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
+ SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
+ {}
+};
+
+static const struct alc_config_preset alc262_presets[] = {
+ [ALC262_BASIC] = {
+ .mixers = { alc262_base_mixer },
+ .init_verbs = { alc262_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ },
+ [ALC262_HIPPO] = {
+ .mixers = { alc262_hippo_mixer },
+ .init_verbs = { alc262_init_verbs, alc_hp15_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HIPPO_1] = {
+ .mixers = { alc262_hippo1_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hippo1_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x02,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hippo1_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_FUJITSU] = {
+ .mixers = { alc262_fujitsu_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+ alc262_fujitsu_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_fujitsu_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_fujitsu_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_BPC] = {
+ .mixers = { alc262_HP_BPC_mixer },
+ .init_verbs = { alc262_HP_BPC_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_bpc_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_BPC_D7000_WF] = {
+ .mixers = { alc262_HP_BPC_WildWest_mixer },
+ .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_wildwest_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_BPC_D7000_WL] = {
+ .mixers = { alc262_HP_BPC_WildWest_mixer,
+ alc262_HP_BPC_WildWest_option_mixer },
+ .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_wildwest_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_TC_T5735] = {
+ .mixers = { alc262_hp_t5735_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_t5735_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_RP5700] = {
+ .mixers = { alc262_hp_rp5700_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_hp_rp5700_capture_source,
+ },
+ [ALC262_BENQ_ED8] = {
+ .mixers = { alc262_base_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ },
+ [ALC262_SONY_ASSAMD] = {
+ .mixers = { alc262_sony_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_BENQ_T31] = {
+ .mixers = { alc262_benq_t31_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
+ alc_hp15_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_ULTRA] = {
+ .mixers = { alc262_ultra_mixer },
+ .cap_mixer = alc262_ultra_capture_mixer,
+ .init_verbs = { alc262_ultra_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_ultra_capture_source,
+ .adc_nids = alc262_adc_nids, /* ADC0 */
+ .capsrc_nids = alc262_capsrc_nids,
+ .num_adc_nids = 1, /* single ADC */
+ .unsol_event = alc262_ultra_unsol_event,
+ .init_hook = alc262_ultra_automute,
+ },
+ [ALC262_LENOVO_3000] = {
+ .mixers = { alc262_lenovo_3000_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+ alc262_lenovo_3000_unsol_verbs,
+ alc262_lenovo_3000_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_fujitsu_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_lenovo_3000_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_NEC] = {
+ .mixers = { alc262_nec_mixer },
+ .init_verbs = { alc262_nec_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ },
+ [ALC262_TOSHIBA_S06] = {
+ .mixers = { alc262_toshiba_s06_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_toshiba_s06_verbs,
+ alc262_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .capsrc_nids = alc262_dmic_capsrc_nids,
+ .dac_nids = alc262_dac_nids,
+ .adc_nids = alc262_dmic_adc_nids, /* ADC0 */
+ .num_adc_nids = 1, /* single ADC */
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_toshiba_s06_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_TOSHIBA_RX1] = {
+ .mixers = { alc262_toshiba_rx1_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_TYAN] = {
+ .mixers = { alc262_tyan_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_tyan_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x02,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_tyan_setup,
+ .init_hook = alc_hp_automute,
+ },
+};
+
diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
new file mode 100644
index 0000000..be58bf2
--- /dev/null
+++ b/sound/pci/hda/alc268_quirks.c
@@ -0,0 +1,636 @@
+/*
+ * ALC267/ALC268 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC268 models */
+enum {
+ ALC268_AUTO,
+ ALC267_QUANTA_IL1,
+ ALC268_3ST,
+ ALC268_TOSHIBA,
+ ALC268_ACER,
+ ALC268_ACER_DMIC,
+ ALC268_ACER_ASPIRE_ONE,
+ ALC268_DELL,
+ ALC268_ZEPTO,
+#ifdef CONFIG_SND_DEBUG
+ ALC268_TEST,
+#endif
+ ALC268_MODEL_LAST /* last tag */
+};
+
+/*
+ * ALC268 channel source setting (2 channel)
+ */
+#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID
+#define alc268_modes alc260_modes
+
+static const hda_nid_t alc268_dac_nids[2] = {
+ /* front, hp */
+ 0x02, 0x03
+};
+
+static const hda_nid_t alc268_adc_nids[2] = {
+ /* ADC0-1 */
+ 0x08, 0x07
+};
+
+static const hda_nid_t alc268_adc_nids_alt[1] = {
+ /* ADC0 */
+ 0x08
+};
+
+static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
+
+static const struct snd_kcontrol_new alc268_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
+static const struct snd_kcontrol_new alc268_toshiba_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
+static const struct hda_verb alc268_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/* Toshiba specific */
+static const struct hda_verb alc268_toshiba_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+/* Acer specific */
+/* bind volumes of both NID 0x02 and 0x03 */
+static const struct hda_bind_ctls alc268_acer_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static void alc268_acer_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+#define alc268_acer_master_sw_get alc262_hp_master_sw_get
+#define alc268_acer_master_sw_put alc262_hp_master_sw_put
+
+static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x15,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc268_acer_master_sw_get,
+ .put = alc268_acer_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
+ { }
+};
+
+static const struct snd_kcontrol_new alc268_acer_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc268_acer_master_sw_get,
+ .put = alc268_acer_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
+static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc268_acer_master_sw_get,
+ .put = alc268_acer_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
+static const struct hda_verb alc268_acer_aspire_one_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
+ { }
+};
+
+static const struct hda_verb alc268_acer_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+/* unsolicited event for HP jack sensing */
+#define alc268_toshiba_setup alc262_hippo_setup
+
+static void alc268_acer_lc_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+static const struct snd_kcontrol_new alc268_dell_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static const struct hda_verb alc268_dell_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+ { }
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc268_dell_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static const struct hda_verb alc267_quanta_il1_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static void alc267_quanta_il1_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc268_base_init_verbs[] = {
+ /* Unmute DAC0-1 and set vol = 0 */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ /* set PCBEEP vol = 0, mute connections */
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* Unmute Selector 23h,24h and set the default input to mic-in */
+
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ { }
+};
+
+/* only for model=test */
+#ifdef CONFIG_SND_DEBUG
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc268_volume_init_verbs[] = {
+ /* set output DAC */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ { }
+};
+#endif /* CONFIG_SND_DEBUG */
+
+static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
+ _DEFINE_CAPSRC(1),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc268_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
+ _DEFINE_CAPSRC(2),
+ { } /* end */
+};
+
+static const struct hda_input_mux alc268_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x3 },
+ },
+};
+
+static const struct hda_input_mux alc268_acer_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static const struct hda_input_mux alc268_acer_dmic_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x6 },
+ { "Line", 0x2 },
+ },
+};
+
+#ifdef CONFIG_SND_DEBUG
+static const struct snd_kcontrol_new alc268_test_mixer[] = {
+ /* Volume widgets */
+ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
+ HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
+ HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
+ HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
+ HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
+ HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
+ /* The below appears problematic on some hardwares */
+ /*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
+ HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
+
+ /* Modes for retasking pin widgets */
+ ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
+
+ /* Controls for GPIO pins, assuming they are configured as outputs */
+ ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
+
+ /* Switches to allow the digital SPDIF output pin to be enabled.
+ * The ALC268 does not have an SPDIF input.
+ */
+ ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
+
+ /* A switch allowing EAPD to be enabled. Some laptops seem to use
+ * this output to turn on an external amplifier.
+ */
+ ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
+ ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
+
+ { } /* end */
+};
+#endif
+
+/*
+ * configuration and preset
+ */
+static const char * const alc268_models[ALC268_MODEL_LAST] = {
+ [ALC267_QUANTA_IL1] = "quanta-il1",
+ [ALC268_3ST] = "3stack",
+ [ALC268_TOSHIBA] = "toshiba",
+ [ALC268_ACER] = "acer",
+ [ALC268_ACER_DMIC] = "acer-dmic",
+ [ALC268_ACER_ASPIRE_ONE] = "acer-aspire",
+ [ALC268_DELL] = "dell",
+ [ALC268_ZEPTO] = "zepto",
+#ifdef CONFIG_SND_DEBUG
+ [ALC268_TEST] = "test",
+#endif
+ [ALC268_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc268_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
+ ALC268_ACER_ASPIRE_ONE),
+ SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
+ SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
+ SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
+ "Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
+ /* almost compatible with toshiba but with optional digital outs;
+ * auto-probing seems working fine
+ */
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
+ ALC268_AUTO),
+ SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
+ SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
+ SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
+ {}
+};
+
+/* Toshiba laptops have no unique PCI SSID but only codec SSID */
+static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO),
+ SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO),
+ SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
+ ALC268_TOSHIBA),
+ {}
+};
+
+static const struct alc_config_preset alc268_presets[] = {
+ [ALC267_QUANTA_IL1] = {
+ .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
+ alc268_capture_nosrc_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc267_quanta_il1_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc267_quanta_il1_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC268_3ST] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ },
+ [ALC268_TOSHIBA] = {
+ .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_toshiba_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc268_toshiba_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC268_ACER] = {
+ .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_acer_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_acer_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc268_acer_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC268_ACER_DMIC] = {
+ .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_acer_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_acer_dmic_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc268_acer_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC268_ACER_ASPIRE_ONE] = {
+ .mixers = { alc268_acer_aspire_one_mixer,
+ alc268_beep_mixer,
+ alc268_capture_nosrc_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_acer_aspire_one_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc268_acer_lc_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC268_DELL] = {
+ .mixers = { alc268_dell_mixer, alc268_beep_mixer,
+ alc268_capture_nosrc_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_dell_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc268_dell_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC268_ZEPTO] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_toshiba_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc268_toshiba_setup,
+ .init_hook = alc_inithook,
+ },
+#ifdef CONFIG_SND_DEBUG
+ [ALC268_TEST] = {
+ .mixers = { alc268_test_mixer, alc268_capture_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_volume_init_verbs,
+ alc268_beep_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ },
+#endif
+};
+
diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
new file mode 100644
index 0000000..14fdcf2
--- /dev/null
+++ b/sound/pci/hda/alc269_quirks.c
@@ -0,0 +1,681 @@
+/*
+ * ALC269/ALC270/ALC275/ALC276 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC269 models */
+enum {
+ ALC269_AUTO,
+ ALC269_BASIC,
+ ALC269_QUANTA_FL1,
+ ALC269_AMIC,
+ ALC269_DMIC,
+ ALC269VB_AMIC,
+ ALC269VB_DMIC,
+ ALC269_FUJITSU,
+ ALC269_LIFEBOOK,
+ ALC271_ACER,
+ ALC269_MODEL_LAST /* last tag */
+};
+
+/*
+ * ALC269 channel source setting (2 channel)
+ */
+#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID
+
+#define alc269_dac_nids alc260_dac_nids
+
+static const hda_nid_t alc269_adc_nids[1] = {
+ /* ADC1 */
+ 0x08,
+};
+
+static const hda_nid_t alc269_capsrc_nids[1] = {
+ 0x23,
+};
+
+static const hda_nid_t alc269vb_adc_nids[1] = {
+ /* ADC1 */
+ 0x09,
+};
+
+static const hda_nid_t alc269vb_capsrc_nids[1] = {
+ 0x22,
+};
+
+#define alc269_modes alc260_modes
+#define alc269_capture_source alc880_lg_lw_capture_source
+
+static const struct snd_kcontrol_new alc269_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_AMP_FLAG,
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc268_acer_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static const struct snd_kcontrol_new alc269_lifebook_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_AMP_FLAG,
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc268_acer_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT),
+ { }
+};
+
+static const struct snd_kcontrol_new alc269_laptop_mixer[] = {
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc269_asus_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+/* capture mixer elements */
+static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* FSC amilo */
+#define alc269_fujitsu_mixer alc269_laptop_mixer
+
+static const struct hda_verb alc269_quanta_fl1_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ { }
+};
+
+static const struct hda_verb alc269_lifebook_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
+{
+ alc_hp_automute(codec);
+
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 0x0c);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF, 0x680);
+
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 0x0c);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF, 0x480);
+}
+
+#define alc269_lifebook_speaker_automute \
+ alc269_quanta_fl1_speaker_automute
+
+static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
+{
+ unsigned int present_laptop;
+ unsigned int present_dock;
+
+ present_laptop = snd_hda_jack_detect(codec, 0x18);
+ present_dock = snd_hda_jack_detect(codec, 0x1b);
+
+ /* Laptop mic port overrides dock mic port, design decision */
+ if (present_dock)
+ snd_hda_codec_write(codec, 0x23, 0,
+ AC_VERB_SET_CONNECT_SEL, 0x3);
+ if (present_laptop)
+ snd_hda_codec_write(codec, 0x23, 0,
+ AC_VERB_SET_CONNECT_SEL, 0x0);
+ if (!present_dock && !present_laptop)
+ snd_hda_codec_write(codec, 0x23, 0,
+ AC_VERB_SET_CONNECT_SEL, 0x1);
+}
+
+static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC_HP_EVENT:
+ alc269_quanta_fl1_speaker_automute(codec);
+ break;
+ case ALC_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc269_lifebook_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC_HP_EVENT)
+ alc269_lifebook_speaker_automute(codec);
+ if ((res >> 26) == ALC_MIC_EVENT)
+ alc269_lifebook_mic_autoswitch(codec);
+}
+
+static void alc269_quanta_fl1_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
+{
+ alc269_quanta_fl1_speaker_automute(codec);
+ alc_mic_automute(codec);
+}
+
+static void alc269_lifebook_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.hp_pins[1] = 0x1a;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+}
+
+static void alc269_lifebook_init_hook(struct hda_codec *codec)
+{
+ alc269_lifebook_speaker_automute(codec);
+ alc269_lifebook_mic_autoswitch(codec);
+}
+
+static const struct hda_verb alc269_laptop_dmic_init_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc269_laptop_amic_init_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x22, AC_VERB_SET_CONNECT_SEL, 0x06},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x22, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc271_acer_dmic_verbs[] = {
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x4000},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x22, AC_VERB_SET_CONNECT_SEL, 6},
+ { }
+};
+
+static void alc269_laptop_amic_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+static void alc269_laptop_dmic_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc269_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /*
+ * Set up output mixers (0x02 - 0x03)
+ */
+ /* set vol=0 to output mixers */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* FIXME: use Mux-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* set EAPD */
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static const struct hda_verb alc269vb_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /*
+ * Set up output mixers (0x02 - 0x03)
+ */
+ /* set vol=0 to output mixers */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* FIXME: use Mux-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x22, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* set EAPD */
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc269_models[ALC269_MODEL_LAST] = {
+ [ALC269_BASIC] = "basic",
+ [ALC269_QUANTA_FL1] = "quanta",
+ [ALC269_AMIC] = "laptop-amic",
+ [ALC269_DMIC] = "laptop-dmic",
+ [ALC269_FUJITSU] = "fujitsu",
+ [ALC269_LIFEBOOK] = "lifebook",
+ [ALC269_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc269_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
+ SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER),
+ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
+ ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
+ ALC269_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
+ ALC269_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
+ SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
+ SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
+ SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
+ SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
+ SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC),
+ SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC),
+ SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC),
+ SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC),
+ {}
+};
+
+static const struct alc_config_preset alc269_presets[] = {
+ [ALC269_BASIC] = {
+ .mixers = { alc269_base_mixer },
+ .init_verbs = { alc269_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_capture_source,
+ },
+ [ALC269_QUANTA_FL1] = {
+ .mixers = { alc269_quanta_fl1_mixer },
+ .init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_capture_source,
+ .unsol_event = alc269_quanta_fl1_unsol_event,
+ .setup = alc269_quanta_fl1_setup,
+ .init_hook = alc269_quanta_fl1_init_hook,
+ },
+ [ALC269_AMIC] = {
+ .mixers = { alc269_laptop_mixer },
+ .cap_mixer = alc269_laptop_analog_capture_mixer,
+ .init_verbs = { alc269_init_verbs,
+ alc269_laptop_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc269_laptop_amic_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC269_DMIC] = {
+ .mixers = { alc269_laptop_mixer },
+ .cap_mixer = alc269_laptop_digital_capture_mixer,
+ .init_verbs = { alc269_init_verbs,
+ alc269_laptop_dmic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc269_laptop_dmic_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC269VB_AMIC] = {
+ .mixers = { alc269vb_laptop_mixer },
+ .cap_mixer = alc269vb_laptop_analog_capture_mixer,
+ .init_verbs = { alc269vb_init_verbs,
+ alc269vb_laptop_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc269vb_laptop_amic_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC269VB_DMIC] = {
+ .mixers = { alc269vb_laptop_mixer },
+ .cap_mixer = alc269vb_laptop_digital_capture_mixer,
+ .init_verbs = { alc269vb_init_verbs,
+ alc269vb_laptop_dmic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc269vb_laptop_dmic_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC269_FUJITSU] = {
+ .mixers = { alc269_fujitsu_mixer },
+ .cap_mixer = alc269_laptop_digital_capture_mixer,
+ .init_verbs = { alc269_init_verbs,
+ alc269_laptop_dmic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc269_laptop_dmic_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC269_LIFEBOOK] = {
+ .mixers = { alc269_lifebook_mixer },
+ .init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_capture_source,
+ .unsol_event = alc269_lifebook_unsol_event,
+ .setup = alc269_lifebook_setup,
+ .init_hook = alc269_lifebook_init_hook,
+ },
+ [ALC271_ACER] = {
+ .mixers = { alc269_asus_mixer },
+ .cap_mixer = alc269vb_laptop_digital_capture_mixer,
+ .init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .adc_nids = alc262_dmic_adc_nids,
+ .num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids),
+ .capsrc_nids = alc262_dmic_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_capture_source,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc269vb_laptop_dmic_setup,
+ .init_hook = alc_inithook,
+ },
+};
+
diff --git a/sound/pci/hda/alc662_quirks.c b/sound/pci/hda/alc662_quirks.c
new file mode 100644
index 0000000..e69a6ea
--- /dev/null
+++ b/sound/pci/hda/alc662_quirks.c
@@ -0,0 +1,1408 @@
+/*
+ * ALC662/ALC663/ALC665/ALC670 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC662 models */
+enum {
+ ALC662_AUTO,
+ ALC662_3ST_2ch_DIG,
+ ALC662_3ST_6ch_DIG,
+ ALC662_3ST_6ch,
+ ALC662_5ST_DIG,
+ ALC662_LENOVO_101E,
+ ALC662_ASUS_EEEPC_P701,
+ ALC662_ASUS_EEEPC_EP20,
+ ALC663_ASUS_M51VA,
+ ALC663_ASUS_G71V,
+ ALC663_ASUS_H13,
+ ALC663_ASUS_G50V,
+ ALC662_ECS,
+ ALC663_ASUS_MODE1,
+ ALC662_ASUS_MODE2,
+ ALC663_ASUS_MODE3,
+ ALC663_ASUS_MODE4,
+ ALC663_ASUS_MODE5,
+ ALC663_ASUS_MODE6,
+ ALC663_ASUS_MODE7,
+ ALC663_ASUS_MODE8,
+ ALC272_DELL,
+ ALC272_DELL_ZM1,
+ ALC272_SAMSUNG_NC10,
+ ALC662_MODEL_LAST,
+};
+
+#define ALC662_DIGOUT_NID 0x06
+#define ALC662_DIGIN_NID 0x0a
+
+static const hda_nid_t alc662_dac_nids[3] = {
+ /* front, rear, clfe */
+ 0x02, 0x03, 0x04
+};
+
+static const hda_nid_t alc272_dac_nids[2] = {
+ 0x02, 0x03
+};
+
+static const hda_nid_t alc662_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x09, 0x08
+};
+
+static const hda_nid_t alc272_adc_nids[1] = {
+ /* ADC1-2 */
+ 0x08,
+};
+
+static const hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
+static const hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
+
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+static const struct hda_input_mux alc662_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc662_lenovo_101e_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static const struct hda_input_mux alc663_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+#if 0 /* set to 1 for testing other input sources below */
+static const struct hda_input_mux alc272_nc10_capture_source = {
+ .num_items = 16,
+ .items = {
+ { "Autoselect Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "In-0x02", 0x2 },
+ { "In-0x03", 0x3 },
+ { "In-0x04", 0x4 },
+ { "In-0x05", 0x5 },
+ { "In-0x06", 0x6 },
+ { "In-0x07", 0x7 },
+ { "In-0x08", 0x8 },
+ { "In-0x09", 0x9 },
+ { "In-0x0a", 0x0a },
+ { "In-0x0b", 0x0b },
+ { "In-0x0c", 0x0c },
+ { "In-0x0d", 0x0d },
+ { "In-0x0e", 0x0e },
+ { "In-0x0f", 0x0f },
+ },
+};
+#endif
+
+/*
+ * 2ch mode
+ */
+static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
+ { 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc662_3ST_ch2_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc662_3ST_ch6_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
+ { 2, alc662_3ST_ch2_init },
+ { 6, alc662_3ST_ch6_init },
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc662_sixstack_ch6_init[] = {
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc662_sixstack_ch8_init[] = {
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc662_5stack_modes[2] = {
+ { 2, alc662_sixstack_ch6_init },
+ { 6, alc662_sixstack_ch8_init },
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+
+static const struct snd_kcontrol_new alc662_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ /*Input mixer control */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc663_asus_one_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_m51va_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_tree_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_four_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_1bjd_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc663_asus_two_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume",
+ &alc663_asus_two_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_g71v_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_g50v_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_mode7_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_mode8_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+static const struct snd_kcontrol_new alc662_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static const struct hda_verb alc662_init_verbs[] = {
+ /* ADC: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ { }
+};
+
+static const struct hda_verb alc662_eapd_init_verbs[] = {
+ /* always trun on EAPD */
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static const struct hda_verb alc662_sue_init_verbs[] = {
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc662_eeepc_sue_init_verbs[] = {
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+/* Set Unsolicited Event*/
+static const struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_m51va_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_21jd_amic_init_verbs[] = {
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc662_1bjd_amic_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_15jd_amic_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_g71v_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
+ /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */
+
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_g50v_init_verbs[] = {
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
+
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc662_ecs_init_verbs[] = {
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc272_dell_zm1_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc272_dell_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_mode7_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_mode8_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc272_auto_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static void alc662_lenovo_101e_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.line_out_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->detect_line = 1;
+ spec->automute_lines = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc662_eeepc_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ alc262_hippo1_setup(codec);
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc663_m51va_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode1 ******************************/
+static void alc663_mode1_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode2 ******************************/
+static void alc662_mode2_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode3 ******************************/
+static void alc663_mode3_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode4 ******************************/
+static void alc663_mode4_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute_mixer_nid[1] = 0x0e;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode5 ******************************/
+static void alc663_mode5_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute_mixer_nid[1] = 0x0e;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode6 ******************************/
+static void alc663_mode6_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode7 ******************************/
+static void alc663_mode7_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode8 ******************************/
+static void alc663_mode8_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.hp_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+static void alc663_g71v_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.line_out_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+ spec->detect_line = 1;
+ spec->automute_lines = 1;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+#define alc663_g50v_setup alc663_m51va_setup
+
+static const struct snd_kcontrol_new alc662_ecs_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+
+ HDA_CODEC_VOLUME("Mic/LineIn Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc272_nc10_mixer[] = {
+ /* Master Playback automatically created from Speaker and Headphone */
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+
+/*
+ * configuration and preset
+ */
+static const char * const alc662_models[ALC662_MODEL_LAST] = {
+ [ALC662_3ST_2ch_DIG] = "3stack-dig",
+ [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig",
+ [ALC662_3ST_6ch] = "3stack-6ch",
+ [ALC662_5ST_DIG] = "5stack-dig",
+ [ALC662_LENOVO_101E] = "lenovo-101e",
+ [ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
+ [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
+ [ALC662_ECS] = "ecs",
+ [ALC663_ASUS_M51VA] = "m51va",
+ [ALC663_ASUS_G71V] = "g71v",
+ [ALC663_ASUS_H13] = "h13",
+ [ALC663_ASUS_G50V] = "g50v",
+ [ALC663_ASUS_MODE1] = "asus-mode1",
+ [ALC662_ASUS_MODE2] = "asus-mode2",
+ [ALC663_ASUS_MODE3] = "asus-mode3",
+ [ALC663_ASUS_MODE4] = "asus-mode4",
+ [ALC663_ASUS_MODE5] = "asus-mode5",
+ [ALC663_ASUS_MODE6] = "asus-mode6",
+ [ALC663_ASUS_MODE7] = "asus-mode7",
+ [ALC663_ASUS_MODE8] = "asus-mode8",
+ [ALC272_DELL] = "dell",
+ [ALC272_DELL_ZM1] = "dell-zm1",
+ [ALC272_SAMSUNG_NC10] = "samsung-nc10",
+ [ALC662_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc662_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
+ SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL),
+ SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1),
+ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
+ SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
+ SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
+ SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
+ /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
+ SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
+ /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
+ SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x19d3, "ASUS NB", ALC663_ASUS_M51VA),
+ SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
+ SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
+ SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
+ SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
+ SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
+ SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
+ SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
+ SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
+ ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
+ {}
+};
+
+static const struct alc_config_preset alc662_presets[] = {
+ [ALC662_3ST_2ch_DIG] = {
+ .mixers = { alc662_3ST_2ch_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_3ST_6ch_DIG] = {
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_3ST_6ch] = {
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_5ST_DIG] = {
+ .mixers = { alc662_base_mixer, alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
+ .channel_mode = alc662_5stack_modes,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_LENOVO_101E] = {
+ .mixers = { alc662_lenovo_101e_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_lenovo_101e_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_lenovo_101e_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ASUS_EEEPC_P701] = {
+ .mixers = { alc662_eeepc_p701_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_eeepc_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_eeepc_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ASUS_EEEPC_EP20] = {
+ .mixers = { alc662_eeepc_ep20_mixer,
+ alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_eeepc_ep20_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .input_mux = &alc662_lenovo_101e_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_eeepc_ep20_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ECS] = {
+ .mixers = { alc662_ecs_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_ecs_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_eeepc_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_M51VA] = {
+ .mixers = { alc663_m51va_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_m51va_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_m51va_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_G71V] = {
+ .mixers = { alc663_g71v_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_g71v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_g71v_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_H13] = {
+ .mixers = { alc663_m51va_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_m51va_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .setup = alc663_m51va_setup,
+ .unsol_event = alc_sku_unsol_event,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_G50V] = {
+ .mixers = { alc663_g50v_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_g50v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .input_mux = &alc663_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_g50v_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE1] = {
+ .mixers = { alc663_m51va_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_21jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode1_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ASUS_MODE2] = {
+ .mixers = { alc662_1bjd_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_1bjd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_mode2_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE3] = {
+ .mixers = { alc663_two_hp_m1_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_two_hp_amic_m1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode3_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE4] = {
+ .mixers = { alc663_asus_21jd_clfe_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_21jd_amic_init_verbs},
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode4_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE5] = {
+ .mixers = { alc663_asus_15jd_clfe_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_15jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode5_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE6] = {
+ .mixers = { alc663_two_hp_m2_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_two_hp_amic_m2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode6_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE7] = {
+ .mixers = { alc663_mode7_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_mode7_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode7_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE8] = {
+ .mixers = { alc663_mode8_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_mode8_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode8_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC272_DELL] = {
+ .mixers = { alc663_m51va_mixer },
+ .cap_mixer = alc272_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc272_dell_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc272_dac_nids),
+ .dac_nids = alc272_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .adc_nids = alc272_adc_nids,
+ .num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
+ .capsrc_nids = alc272_capsrc_nids,
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_m51va_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC272_DELL_ZM1] = {
+ .mixers = { alc663_m51va_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc272_dell_zm1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc272_dac_nids),
+ .dac_nids = alc272_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .adc_nids = alc662_adc_nids,
+ .num_adc_nids = 1,
+ .capsrc_nids = alc662_capsrc_nids,
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_m51va_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC272_SAMSUNG_NC10] = {
+ .mixers = { alc272_nc10_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_21jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc272_dac_nids),
+ .dac_nids = alc272_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ /*.input_mux = &alc272_nc10_capture_source,*/
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode4_setup,
+ .init_hook = alc_inithook,
+ },
+};
+
+
diff --git a/sound/pci/hda/alc680_quirks.c b/sound/pci/hda/alc680_quirks.c
new file mode 100644
index 0000000..0eeb227
--- /dev/null
+++ b/sound/pci/hda/alc680_quirks.c
@@ -0,0 +1,222 @@
+/*
+ * ALC680 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC680 models */
+enum {
+ ALC680_AUTO,
+ ALC680_BASE,
+ ALC680_MODEL_LAST,
+};
+
+#define ALC680_DIGIN_NID ALC880_DIGIN_NID
+#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID
+#define alc680_modes alc260_modes
+
+static const hda_nid_t alc680_dac_nids[3] = {
+ /* Lout1, Lout2, hp */
+ 0x02, 0x03, 0x04
+};
+
+static const hda_nid_t alc680_adc_nids[3] = {
+ /* ADC0-2 */
+ /* DMIC, MIC, Line-in*/
+ 0x07, 0x08, 0x09
+};
+
+/*
+ * Analog capture ADC cgange
+ */
+static hda_nid_t alc680_get_cur_adc(struct hda_codec *codec)
+{
+ static hda_nid_t pins[] = {0x18, 0x19};
+ static hda_nid_t adcs[] = {0x08, 0x09};
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(pins); i++) {
+ if (!is_jack_detectable(codec, pins[i]))
+ continue;
+ if (snd_hda_jack_detect(codec, pins[i]))
+ return adcs[i];
+ }
+ return 0x07;
+}
+
+static void alc680_rec_autoswitch(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid = alc680_get_cur_adc(codec);
+ if (spec->cur_adc && nid != spec->cur_adc) {
+ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
+ spec->cur_adc = nid;
+ snd_hda_codec_setup_stream(codec, nid,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+ }
+}
+
+static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid = alc680_get_cur_adc(codec);
+
+ spec->cur_adc = nid;
+ spec->cur_adc_stream_tag = stream_tag;
+ spec->cur_adc_format = format;
+ snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+ return 0;
+}
+
+static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = 0;
+ return 0;
+}
+
+static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
+ .substreams = 1, /* can be overridden */
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+ .ops = {
+ .prepare = alc680_capture_pcm_prepare,
+ .cleanup = alc680_capture_pcm_cleanup
+ },
+};
+
+static const struct snd_kcontrol_new alc680_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static const struct hda_bind_ctls alc680_bind_cap_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc680_bind_cap_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc680_master_capture_mixer[] = {
+ HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
+ HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
+ { } /* end */
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc680_init_verbs[] = {
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc680_base_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x16;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.num_inputs = 2;
+ spec->autocfg.inputs[0].pin = 0x18;
+ spec->autocfg.inputs[0].type = AUTO_PIN_MIC;
+ spec->autocfg.inputs[1].pin = 0x19;
+ spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc680_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC_HP_EVENT)
+ alc_hp_automute(codec);
+ if ((res >> 26) == ALC_MIC_EVENT)
+ alc680_rec_autoswitch(codec);
+}
+
+static void alc680_inithook(struct hda_codec *codec)
+{
+ alc_hp_automute(codec);
+ alc680_rec_autoswitch(codec);
+}
+
+/*
+ * configuration and preset
+ */
+static const char * const alc680_models[ALC680_MODEL_LAST] = {
+ [ALC680_BASE] = "base",
+ [ALC680_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc680_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
+ {}
+};
+
+static const struct alc_config_preset alc680_presets[] = {
+ [ALC680_BASE] = {
+ .mixers = { alc680_base_mixer },
+ .cap_mixer = alc680_master_capture_mixer,
+ .init_verbs = { alc680_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc680_dac_nids),
+ .dac_nids = alc680_dac_nids,
+ .dig_out_nid = ALC680_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc680_modes),
+ .channel_mode = alc680_modes,
+ .unsol_event = alc680_unsol_event,
+ .setup = alc680_base_setup,
+ .init_hook = alc680_inithook,
+
+ },
+};
diff --git a/sound/pci/hda/alc861_quirks.c b/sound/pci/hda/alc861_quirks.c
new file mode 100644
index 0000000..d719ec6
--- /dev/null
+++ b/sound/pci/hda/alc861_quirks.c
@@ -0,0 +1,725 @@
+/*
+ * ALC660/ALC861 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC861 models */
+enum {
+ ALC861_AUTO,
+ ALC861_3ST,
+ ALC660_3ST,
+ ALC861_3ST_DIG,
+ ALC861_6ST_DIG,
+ ALC861_UNIWILL_M31,
+ ALC861_TOSHIBA,
+ ALC861_ASUS,
+ ALC861_ASUS_LAPTOP,
+ ALC861_MODEL_LAST,
+};
+
+/*
+ * ALC861 channel source setting (2/6 channel selection for 3-stack)
+ */
+
+/*
+ * set the path ways for 2 channel output
+ * need to set the codec line out and mic 1 pin widgets to inputs
+ */
+static const struct hda_verb alc861_threestack_ch2_init[] = {
+ /* set pin widget 1Ah (line in) for input */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* set pin widget 18h (mic1/2) for input, for mic also enable
+ * the vref
+ */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+#endif
+ { } /* end */
+};
+/*
+ * 6ch mode
+ * need to set the codec line out and mic 1 pin widgets to outputs
+ */
+static const struct hda_verb alc861_threestack_ch6_init[] = {
+ /* set pin widget 1Ah (line in) for output (Back Surround)*/
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* set pin widget 18h (mic1) for output (CLFE)*/
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+
+ { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
+#endif
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc861_threestack_modes[2] = {
+ { 2, alc861_threestack_ch2_init },
+ { 6, alc861_threestack_ch6_init },
+};
+/* Set mic1 as input and unmute the mixer */
+static const struct hda_verb alc861_uniwill_m31_ch2_init[] = {
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+ { } /* end */
+};
+/* Set mic1 as output and mute mixer */
+static const struct hda_verb alc861_uniwill_m31_ch4_init[] = {
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
+ { 2, alc861_uniwill_m31_ch2_init },
+ { 4, alc861_uniwill_m31_ch4_init },
+};
+
+/* Set mic1 and line-in as input and unmute the mixer */
+static const struct hda_verb alc861_asus_ch2_init[] = {
+ /* set pin widget 1Ah (line in) for input */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* set pin widget 18h (mic1/2) for input, for mic also enable
+ * the vref
+ */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+#endif
+ { } /* end */
+};
+/* Set mic1 nad line-in as output and mute mixer */
+static const struct hda_verb alc861_asus_ch6_init[] = {
+ /* set pin widget 1Ah (line in) for output (Back Surround)*/
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
+ /* set pin widget 18h (mic1) for output (CLFE)*/
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
+ { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
+#endif
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc861_asus_modes[2] = {
+ { 2, alc861_asus_ch2_init },
+ { 6, alc861_asus_ch6_init },
+};
+
+/* patch-ALC861 */
+
+static const struct snd_kcontrol_new alc861_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+
+ /*Input mixer control */
+ /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc861_3ST_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
+
+ /* Input mixer control */
+ /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ .private_value = ARRAY_SIZE(alc861_threestack_modes),
+ },
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc861_toshiba_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
+
+ /* Input mixer control */
+ /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
+ },
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc861_asus_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+
+ /* Input mixer control */
+ HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ .private_value = ARRAY_SIZE(alc861_asus_modes),
+ },
+ { }
+};
+
+/* additional mixer */
+static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ { }
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc861_base_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+
+ { }
+};
+
+static const struct hda_verb alc861_threestack_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ { }
+};
+
+static const struct hda_verb alc861_uniwill_m31_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ /* this has to be set to VREF80 */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ { }
+};
+
+static const struct hda_verb alc861_asus_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel)
+ * according to codec#0 this is the HP jack
+ */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
+ /* route front PCM to HP */
+ { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ /* this has to be set to VREF80 */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ { }
+};
+
+/* additional init verbs for ASUS laptops */
+static const struct hda_verb alc861_asus_laptop_init_verbs[] = {
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
+ { }
+};
+
+static const struct hda_verb alc861_toshiba_init_verbs[] = {
+ {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc861_toshiba_automute(struct hda_codec *codec)
+{
+ unsigned int present = snd_hda_jack_detect(codec, 0x0f);
+
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
+ HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
+}
+
+static void alc861_toshiba_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC_HP_EVENT)
+ alc861_toshiba_automute(codec);
+}
+
+#define ALC861_DIGOUT_NID 0x07
+
+static const struct hda_channel_mode alc861_8ch_modes[1] = {
+ { 8, NULL }
+};
+
+static const hda_nid_t alc861_dac_nids[4] = {
+ /* front, surround, clfe, side */
+ 0x03, 0x06, 0x05, 0x04
+};
+
+static const hda_nid_t alc660_dac_nids[3] = {
+ /* front, clfe, surround */
+ 0x03, 0x05, 0x06
+};
+
+static const hda_nid_t alc861_adc_nids[1] = {
+ /* ADC0-2 */
+ 0x08,
+};
+
+static const struct hda_input_mux alc861_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x3 },
+ { "Line", 0x1 },
+ { "CD", 0x4 },
+ { "Mixer", 0x5 },
+ },
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc861_models[ALC861_MODEL_LAST] = {
+ [ALC861_3ST] = "3stack",
+ [ALC660_3ST] = "3stack-660",
+ [ALC861_3ST_DIG] = "3stack-dig",
+ [ALC861_6ST_DIG] = "6stack-dig",
+ [ALC861_UNIWILL_M31] = "uniwill-m31",
+ [ALC861_TOSHIBA] = "toshiba",
+ [ALC861_ASUS] = "asus",
+ [ALC861_ASUS_LAPTOP] = "asus-laptop",
+ [ALC861_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc861_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
+ SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
+ SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
+ /* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
+ * Any other models that need this preset?
+ */
+ /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
+ SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
+ SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
+ SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
+ SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
+ SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
+ /* FIXME: the below seems conflict */
+ /* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
+ SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
+ SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
+ {}
+};
+
+static const struct alc_config_preset alc861_presets[] = {
+ [ALC861_3ST] = {
+ .mixers = { alc861_3ST_mixer },
+ .init_verbs = { alc861_threestack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+ .channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_3ST_DIG] = {
+ .mixers = { alc861_base_mixer },
+ .init_verbs = { alc861_threestack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+ .channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_6ST_DIG] = {
+ .mixers = { alc861_base_mixer },
+ .init_verbs = { alc861_base_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
+ .channel_mode = alc861_8ch_modes,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC660_3ST] = {
+ .mixers = { alc861_3ST_mixer },
+ .init_verbs = { alc861_threestack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc660_dac_nids),
+ .dac_nids = alc660_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+ .channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_UNIWILL_M31] = {
+ .mixers = { alc861_uniwill_m31_mixer },
+ .init_verbs = { alc861_uniwill_m31_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
+ .channel_mode = alc861_uniwill_m31_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_TOSHIBA] = {
+ .mixers = { alc861_toshiba_mixer },
+ .init_verbs = { alc861_base_init_verbs,
+ alc861_toshiba_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ .unsol_event = alc861_toshiba_unsol_event,
+ .init_hook = alc861_toshiba_automute,
+ },
+ [ALC861_ASUS] = {
+ .mixers = { alc861_asus_mixer },
+ .init_verbs = { alc861_asus_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
+ .channel_mode = alc861_asus_modes,
+ .need_dac_fix = 1,
+ .hp_nid = 0x06,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_ASUS_LAPTOP] = {
+ .mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
+ .init_verbs = { alc861_asus_init_verbs,
+ alc861_asus_laptop_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+};
+
diff --git a/sound/pci/hda/alc861vd_quirks.c b/sound/pci/hda/alc861vd_quirks.c
new file mode 100644
index 0000000..8f28450
--- /dev/null
+++ b/sound/pci/hda/alc861vd_quirks.c
@@ -0,0 +1,605 @@
+/*
+ * ALC660-VD/ALC861-VD quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC861-VD models */
+enum {
+ ALC861VD_AUTO,
+ ALC660VD_3ST,
+ ALC660VD_3ST_DIG,
+ ALC660VD_ASUS_V1S,
+ ALC861VD_3ST,
+ ALC861VD_3ST_DIG,
+ ALC861VD_6ST_DIG,
+ ALC861VD_LENOVO,
+ ALC861VD_DALLAS,
+ ALC861VD_HP,
+ ALC861VD_MODEL_LAST,
+};
+
+#define ALC861VD_DIGOUT_NID 0x06
+
+static const hda_nid_t alc861vd_dac_nids[4] = {
+ /* front, surr, clfe, side surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+
+/* dac_nids for ALC660vd are in a different order - according to
+ * Realtek's driver.
+ * This should probably result in a different mixer for 6stack models
+ * of ALC660vd codecs, but for now there is only 3stack mixer
+ * - and it is the same as in 861vd.
+ * adc_nids in ALC660vd are (is) the same as in 861vd
+ */
+static const hda_nid_t alc660vd_dac_nids[3] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x04, 0x03
+};
+
+static const hda_nid_t alc861vd_adc_nids[1] = {
+ /* ADC0 */
+ 0x09,
+};
+
+static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+static const struct hda_input_mux alc861vd_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc861vd_dallas_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ },
+};
+
+static const struct hda_input_mux alc861vd_hp_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "ATAPI Mic", 0x1 },
+ },
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
+ { 2, NULL }
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc861vd_6stack_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static const struct hda_verb alc861vd_6stack_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc861vd_6stack_modes[2] = {
+ { 6, alc861vd_6stack_ch6_init },
+ { 8, alc861vd_6stack_ch8_init },
+};
+
+static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+static const struct snd_kcontrol_new alc861vd_6st_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc861vd_3st_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ /*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+ { } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, HP = 0x15,
+ * Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
+ */
+static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, Line-out = 0x15,
+ * Front Mic=0x18, ATAPI Mic = 0x19,
+ */
+static const struct snd_kcontrol_new alc861vd_hp_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ { } /* end */
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc861vd_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of
+ * the analog-loopback mixer widget
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /*
+ * Set up output mixers (0x02 - 0x05)
+ */
+ /* set vol=0 to output mixers */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ { }
+};
+
+/*
+ * 3-stack pin configuration:
+ * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
+ */
+static const struct hda_verb alc861vd_3stack_init_verbs[] = {
+ /*
+ * Set pin mode and muting
+ */
+ /* set front pin widgets 0x14 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * 6-stack pin configuration:
+ */
+static const struct hda_verb alc861vd_6stack_init_verbs[] = {
+ /*
+ * Set pin mode and muting
+ */
+ /* set front pin widgets 0x14 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* Side Pin: output 3 (0x0f) */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+static const struct hda_verb alc861vd_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {}
+};
+
+static void alc861vd_lenovo_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
+{
+ alc_hp_automute(codec);
+ alc88x_simple_mic_automute(codec);
+}
+
+static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC_MIC_EVENT:
+ alc88x_simple_mic_automute(codec);
+ break;
+ default:
+ alc_sku_unsol_event(codec, res);
+ break;
+ }
+}
+
+static const struct hda_verb alc861vd_dallas_verbs[] = {
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc861vd_dallas_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/*
+ * configuration and preset
+ */
+static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
+ [ALC660VD_3ST] = "3stack-660",
+ [ALC660VD_3ST_DIG] = "3stack-660-digout",
+ [ALC660VD_ASUS_V1S] = "asus-v1s",
+ [ALC861VD_3ST] = "3stack",
+ [ALC861VD_3ST_DIG] = "3stack-digout",
+ [ALC861VD_6ST_DIG] = "6stack-digout",
+ [ALC861VD_LENOVO] = "lenovo",
+ [ALC861VD_DALLAS] = "dallas",
+ [ALC861VD_HP] = "hp",
+ [ALC861VD_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc861vd_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
+ SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
+ SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
+ /*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
+ SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
+ SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
+ SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
+ SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
+ /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
+ SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
+ SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
+ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
+ {}
+};
+
+static const struct alc_config_preset alc861vd_presets[] = {
+ [ALC660VD_3ST] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+ .dac_nids = alc660vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC660VD_3ST_DIG] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+ .dac_nids = alc660vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_3ST] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_3ST_DIG] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_6ST_DIG] = {
+ .mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_6stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
+ .channel_mode = alc861vd_6stack_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_LENOVO] = {
+ .mixers = { alc861vd_lenovo_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs,
+ alc861vd_eapd_verbs,
+ alc861vd_lenovo_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+ .dac_nids = alc660vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ .unsol_event = alc861vd_lenovo_unsol_event,
+ .setup = alc861vd_lenovo_setup,
+ .init_hook = alc861vd_lenovo_init_hook,
+ },
+ [ALC861VD_DALLAS] = {
+ .mixers = { alc861vd_dallas_mixer },
+ .init_verbs = { alc861vd_dallas_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_dallas_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc861vd_dallas_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC861VD_HP] = {
+ .mixers = { alc861vd_hp_mixer },
+ .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_hp_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc861vd_dallas_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC660VD_ASUS_V1S] = {
+ .mixers = { alc861vd_lenovo_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs,
+ alc861vd_eapd_verbs,
+ alc861vd_lenovo_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+ .dac_nids = alc660vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ .unsol_event = alc861vd_lenovo_unsol_event,
+ .setup = alc861vd_lenovo_setup,
+ .init_hook = alc861vd_lenovo_init_hook,
+ },
+};
+
diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
new file mode 100644
index 0000000..c844d2b
--- /dev/null
+++ b/sound/pci/hda/alc880_quirks.c
@@ -0,0 +1,1898 @@
+/*
+ * ALC880 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC880 board config type */
+enum {
+ ALC880_AUTO,
+ ALC880_3ST,
+ ALC880_3ST_DIG,
+ ALC880_5ST,
+ ALC880_5ST_DIG,
+ ALC880_W810,
+ ALC880_Z71V,
+ ALC880_6ST,
+ ALC880_6ST_DIG,
+ ALC880_F1734,
+ ALC880_ASUS,
+ ALC880_ASUS_DIG,
+ ALC880_ASUS_W1V,
+ ALC880_ASUS_DIG2,
+ ALC880_FUJITSU,
+ ALC880_UNIWILL_DIG,
+ ALC880_UNIWILL,
+ ALC880_UNIWILL_P53,
+ ALC880_CLEVO,
+ ALC880_TCL_S700,
+ ALC880_LG,
+ ALC880_LG_LW,
+ ALC880_MEDION_RIM,
+#ifdef CONFIG_SND_DEBUG
+ ALC880_TEST,
+#endif
+ ALC880_MODEL_LAST /* last tag */
+};
+
+/*
+ * ALC880 3-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
+ * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
+ * F-Mic = 0x1b, HP = 0x19
+ */
+
+static const hda_nid_t alc880_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x05, 0x04, 0x03
+};
+
+static const hda_nid_t alc880_adc_nids[3] = {
+ /* ADC0-2 */
+ 0x07, 0x08, 0x09,
+};
+
+/* The datasheet says the node 0x07 is connected from inputs,
+ * but it shows zero connection in the real implementation on some devices.
+ * Note: this is a 915GAV bug, fixed on 915GLV
+ */
+static const hda_nid_t alc880_adc_nids_alt[2] = {
+ /* ADC1-2 */
+ 0x08, 0x09,
+};
+
+#define ALC880_DIGOUT_NID 0x06
+#define ALC880_DIGIN_NID 0x0a
+#define ALC880_PIN_CD_NID 0x1c
+
+static const struct hda_input_mux alc880_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x3 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* channel source setting (2/6 channel selection for 3-stack) */
+/* 2ch mode */
+static const struct hda_verb alc880_threestack_ch2_init[] = {
+ /* set line-in to input, mute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ /* set mic-in to input vref 80%, mute it */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/* 6ch mode */
+static const struct hda_verb alc880_threestack_ch6_init[] = {
+ /* set line-in to output, unmute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ /* set mic-in to output, unmute it */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc880_threestack_modes[2] = {
+ { 2, alc880_threestack_ch2_init },
+ { 6, alc880_threestack_ch6_init },
+};
+
+static const struct snd_kcontrol_new alc880_three_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/*
+ * ALC880 5-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
+ * Side = 0x02 (0xd)
+ * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
+ * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
+ */
+
+/* additional mixers to alc880_three_stack_mixer */
+static const struct snd_kcontrol_new alc880_five_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
+ { } /* end */
+};
+
+/* channel source setting (6/8 channel selection for 5-stack) */
+/* 6ch mode */
+static const struct hda_verb alc880_fivestack_ch6_init[] = {
+ /* set line-in to input, mute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/* 8ch mode */
+static const struct hda_verb alc880_fivestack_ch8_init[] = {
+ /* set line-in to output, unmute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc880_fivestack_modes[2] = {
+ { 6, alc880_fivestack_ch6_init },
+ { 8, alc880_fivestack_ch8_init },
+};
+
+
+/*
+ * ALC880 6-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
+ * Side = 0x05 (0x0f)
+ * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
+ * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
+ */
+
+static const hda_nid_t alc880_6st_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+
+static const struct hda_input_mux alc880_6stack_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* fixed 8-channels */
+static const struct hda_channel_mode alc880_sixstack_modes[1] = {
+ { 8, NULL },
+};
+
+static const struct snd_kcontrol_new alc880_six_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+
+/*
+ * ALC880 W810 model
+ *
+ * W810 has rear IO for:
+ * Front (DAC 02)
+ * Surround (DAC 03)
+ * Center/LFE (DAC 04)
+ * Digital out (06)
+ *
+ * The system also has a pair of internal speakers, and a headphone jack.
+ * These are both connected to Line2 on the codec, hence to DAC 02.
+ *
+ * There is a variable resistor to control the speaker or headphone
+ * volume. This is a hardware-only device without a software API.
+ *
+ * Plugging headphones in will disable the internal speakers. This is
+ * implemented in hardware, not via the driver using jack sense. In
+ * a similar fashion, plugging into the rear socket marked "front" will
+ * disable both the speakers and headphones.
+ *
+ * For input, there's a microphone jack, and an "audio in" jack.
+ * These may not do anything useful with this driver yet, because I
+ * haven't setup any initialization verbs for these yet...
+ */
+
+static const hda_nid_t alc880_w810_dac_nids[3] = {
+ /* front, rear/surround, clfe */
+ 0x02, 0x03, 0x04
+};
+
+/* fixed 6 channels */
+static const struct hda_channel_mode alc880_w810_modes[1] = {
+ { 6, NULL }
+};
+
+/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
+static const struct snd_kcontrol_new alc880_w810_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+
+/*
+ * Z710V model
+ *
+ * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
+ * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
+ * Line = 0x1a
+ */
+
+static const hda_nid_t alc880_z71v_dac_nids[1] = {
+ 0x02
+};
+#define ALC880_Z71V_HP_DAC 0x03
+
+/* fixed 2 channels */
+static const struct hda_channel_mode alc880_2_jack_modes[1] = {
+ { 2, NULL }
+};
+
+static const struct snd_kcontrol_new alc880_z71v_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+/*
+ * ALC880 F1734 model
+ *
+ * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
+ * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
+ */
+
+static const hda_nid_t alc880_f1734_dac_nids[1] = {
+ 0x03
+};
+#define ALC880_F1734_HP_DAC 0x02
+
+static const struct snd_kcontrol_new alc880_f1734_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_input_mux alc880_f1734_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
+
+/*
+ * ALC880 ASUS model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ * Mic = 0x18, Line = 0x1a
+ */
+
+#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
+#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
+
+static const struct snd_kcontrol_new alc880_asus_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/*
+ * ALC880 ASUS W1V model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ * Mic = 0x18, Line = 0x1a, Line2 = 0x1b
+ */
+
+/* additional mixers to alc880_asus_mixer */
+static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
+ HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
+ { } /* end */
+};
+
+/* TCL S700 */
+static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+/* Uniwill */
+static const struct snd_kcontrol_new alc880_uniwill_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+/*
+ * initialize the codec volumes, etc
+ */
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static const struct hda_verb alc880_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for front
+ * panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0f)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ { }
+};
+
+/*
+ * 3-stack pin configuration:
+ * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
+ */
+static const struct hda_verb alc880_pin_3stack_init_verbs[] = {
+ /*
+ * preset connection lists of input pins
+ * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
+ */
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
+
+ /*
+ * Set pin mode and muting
+ */
+ /* set front pin widgets 0x14 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line2 (as front mic) pin widget for input and vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * 5-stack pin configuration:
+ * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
+ * line-in/side = 0x1a, f-mic = 0x1b
+ */
+static const struct hda_verb alc880_pin_5stack_init_verbs[] = {
+ /*
+ * preset connection lists of input pins
+ * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
+ */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
+
+ /*
+ * Set pin mode and muting
+ */
+ /* set pin widgets 0x14-0x17 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* unmute pins for output (no gain on this amp) */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line2 (as front mic) pin widget for input and vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * W810 pin configuration:
+ * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
+ */
+static const struct hda_verb alc880_pin_w810_init_verbs[] = {
+ /* hphone/speaker input selector: front DAC */
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ { }
+};
+
+/*
+ * Z71V pin configuration:
+ * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
+ */
+static const struct hda_verb alc880_pin_z71v_init_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * 6-stack pin configuration:
+ * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
+ * f-mic = 0x19, line = 0x1a, HP = 0x1b
+ */
+static const struct hda_verb alc880_pin_6stack_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * Uniwill pin configuration:
+ * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
+ * line = 0x1a
+ */
+static const struct hda_verb alc880_uniwill_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */
+ /* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+
+ { }
+};
+
+/*
+* Uniwill P53
+* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
+ */
+static const struct hda_verb alc880_uniwill_p53_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_DCVOL_EVENT},
+
+ { }
+};
+
+static const struct hda_verb alc880_beep_init_verbs[] = {
+ { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
+ { }
+};
+
+static void alc880_uniwill_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x16;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc880_uniwill_init_hook(struct hda_codec *codec)
+{
+ alc_hp_automute(codec);
+ alc88x_simple_mic_automute(codec);
+}
+
+static void alc880_uniwill_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ switch (res >> 28) {
+ case ALC_MIC_EVENT:
+ alc88x_simple_mic_automute(codec);
+ break;
+ default:
+ alc_sku_unsol_event(codec, res);
+ break;
+ }
+}
+
+static void alc880_uniwill_p53_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
+ present &= HDA_AMP_VOLMASK;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
+ HDA_AMP_VOLMASK, present);
+ snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
+ HDA_AMP_VOLMASK, present);
+}
+
+static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ if ((res >> 28) == ALC_DCVOL_EVENT)
+ alc880_uniwill_p53_dcvol_automute(codec);
+ else
+ alc_sku_unsol_event(codec, res);
+}
+
+/*
+ * F1734 pin configuration:
+ * HP = 0x14, speaker-out = 0x15, mic = 0x18
+ */
+static const struct hda_verb alc880_pin_f1734_init_verbs[] = {
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_DCVOL_EVENT},
+
+ { }
+};
+
+/*
+ * ASUS pin configuration:
+ * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
+ */
+static const struct hda_verb alc880_pin_asus_init_verbs[] = {
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/* Enable GPIO mask and set output */
+#define alc880_gpio1_init_verbs alc_gpio1_init_verbs
+#define alc880_gpio2_init_verbs alc_gpio2_init_verbs
+#define alc880_gpio3_init_verbs alc_gpio3_init_verbs
+
+/* Clevo m520g init */
+static const struct hda_verb alc880_pin_clevo_init_verbs[] = {
+ /* headphone output */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* line-out */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line-in */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* CD */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Mic1 (rear panel) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Mic2 (front panel) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* headphone */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
+ { }
+};
+
+static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
+ /* Headphone output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Front output*/
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
+
+ { }
+};
+
+/*
+ * LG m1 express dual
+ *
+ * Pin assignment:
+ * Rear Line-In/Out (blue): 0x14
+ * Build-in Mic-In: 0x15
+ * Speaker-out: 0x17
+ * HP-Out (green): 0x1b
+ * Mic-In/Out (red): 0x19
+ * SPDIF-Out: 0x1e
+ */
+
+/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
+static const hda_nid_t alc880_lg_dac_nids[3] = {
+ 0x05, 0x02, 0x03
+};
+
+/* seems analog CD is not working */
+static const struct hda_input_mux alc880_lg_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x5 },
+ { "Internal Mic", 0x6 },
+ },
+};
+
+/* 2,4,6 channel modes */
+static const struct hda_verb alc880_lg_ch2_init[] = {
+ /* set line-in and mic-in to input */
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { }
+};
+
+static const struct hda_verb alc880_lg_ch4_init[] = {
+ /* set line-in to out and mic-in to input */
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { }
+};
+
+static const struct hda_verb alc880_lg_ch6_init[] = {
+ /* set line-in and mic-in to output */
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { }
+};
+
+static const struct hda_channel_mode alc880_lg_ch_modes[3] = {
+ { 2, alc880_lg_ch2_init },
+ { 4, alc880_lg_ch4_init },
+ { 6, alc880_lg_ch6_init },
+};
+
+static const struct snd_kcontrol_new alc880_lg_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static const struct hda_verb alc880_lg_init_verbs[] = {
+ /* set capture source to mic-in */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* mute all amp mixer inputs */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* line-in to input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* built-in mic */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* speaker-out */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* mic-in to input */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* HP-out */
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* jack sense */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/*
+ * LG LW20
+ *
+ * Pin assignment:
+ * Speaker-out: 0x14
+ * Mic-In: 0x18
+ * Built-in Mic-In: 0x19
+ * Line-In: 0x1b
+ * HP-Out: 0x1a
+ * SPDIF-Out: 0x1e
+ */
+
+static const struct hda_input_mux alc880_lg_lw_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "Line In", 0x2 },
+ },
+};
+
+#define alc880_lg_lw_modes alc880_threestack_modes
+
+static const struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static const struct hda_verb alc880_lg_lw_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
+
+ /* set capture source to mic-in */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* speaker-out */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* HP-out */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* mic-in to input */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* built-in mic */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* jack sense */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_lw_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_input_mux alc880_medion_rim_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ },
+};
+
+static const struct hda_verb alc880_medion_rim_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Internal Speaker */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_medion_rim_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc_hp_automute(codec);
+ /* toggle EAPD */
+ if (spec->jack_present)
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
+ else
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
+}
+
+static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ if ((res >> 28) == ALC_HP_EVENT)
+ alc880_medion_rim_automute(codec);
+}
+
+static void alc880_medion_rim_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static const struct hda_amp_list alc880_lg_loopbacks[] = {
+ { 0x0b, HDA_INPUT, 1 },
+ { 0x0b, HDA_INPUT, 6 },
+ { 0x0b, HDA_INPUT, 7 },
+ { } /* end */
+};
+#endif
+
+/*
+ * Test configuration for debugging
+ *
+ * Almost all inputs/outputs are enabled. I/O pins can be configured via
+ * enum controls.
+ */
+#ifdef CONFIG_SND_DEBUG
+static const hda_nid_t alc880_test_dac_nids[4] = {
+ 0x02, 0x03, 0x04, 0x05
+};
+
+static const struct hda_input_mux alc880_test_capture_source = {
+ .num_items = 7,
+ .items = {
+ { "In-1", 0x0 },
+ { "In-2", 0x1 },
+ { "In-3", 0x2 },
+ { "In-4", 0x3 },
+ { "CD", 0x4 },
+ { "Front", 0x5 },
+ { "Surround", 0x6 },
+ },
+};
+
+static const struct hda_channel_mode alc880_test_modes[4] = {
+ { 2, NULL },
+ { 4, NULL },
+ { 6, NULL },
+ { 8, NULL },
+};
+
+static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "N/A", "Line Out", "HP Out",
+ "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item >= 8)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int pin_ctl, item = 0;
+
+ pin_ctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (pin_ctl & AC_PINCTL_OUT_EN) {
+ if (pin_ctl & AC_PINCTL_HP_EN)
+ item = 2;
+ else
+ item = 1;
+ } else if (pin_ctl & AC_PINCTL_IN_EN) {
+ switch (pin_ctl & AC_PINCTL_VREFEN) {
+ case AC_PINCTL_VREF_HIZ: item = 3; break;
+ case AC_PINCTL_VREF_50: item = 4; break;
+ case AC_PINCTL_VREF_GRD: item = 5; break;
+ case AC_PINCTL_VREF_80: item = 6; break;
+ case AC_PINCTL_VREF_100: item = 7; break;
+ }
+ }
+ ucontrol->value.enumerated.item[0] = item;
+ return 0;
+}
+
+static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ static const unsigned int ctls[] = {
+ 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
+ };
+ unsigned int old_ctl, new_ctl;
+
+ old_ctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ new_ctl = ctls[ucontrol->value.enumerated.item[0]];
+ if (old_ctl != new_ctl) {
+ int val;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ new_ctl);
+ val = ucontrol->value.enumerated.item[0] >= 3 ?
+ HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, val);
+ return 1;
+ }
+ return 0;
+}
+
+static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "Front", "Surround", "CLFE", "Side"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= 4)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int sel;
+
+ sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
+ ucontrol->value.enumerated.item[0] = sel & 3;
+ return 0;
+}
+
+static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int sel;
+
+ sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
+ if (ucontrol->value.enumerated.item[0] != sel) {
+ sel = ucontrol->value.enumerated.item[0] & 3;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, sel);
+ return 1;
+ }
+ return 0;
+}
+
+#define PIN_CTL_TEST(xname,nid) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+ .info = alc_test_pin_ctl_info, \
+ .get = alc_test_pin_ctl_get, \
+ .put = alc_test_pin_ctl_put, \
+ .private_value = nid \
+ }
+
+#define PIN_SRC_TEST(xname,nid) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+ .info = alc_test_pin_src_info, \
+ .get = alc_test_pin_src_get, \
+ .put = alc_test_pin_src_put, \
+ .private_value = nid \
+ }
+
+static const struct snd_kcontrol_new alc880_test_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ PIN_CTL_TEST("Front Pin Mode", 0x14),
+ PIN_CTL_TEST("Surround Pin Mode", 0x15),
+ PIN_CTL_TEST("CLFE Pin Mode", 0x16),
+ PIN_CTL_TEST("Side Pin Mode", 0x17),
+ PIN_CTL_TEST("In-1 Pin Mode", 0x18),
+ PIN_CTL_TEST("In-2 Pin Mode", 0x19),
+ PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
+ PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
+ PIN_SRC_TEST("In-1 Pin Source", 0x18),
+ PIN_SRC_TEST("In-2 Pin Source", 0x19),
+ PIN_SRC_TEST("In-3 Pin Source", 0x1a),
+ PIN_SRC_TEST("In-4 Pin Source", 0x1b),
+ HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static const struct hda_verb alc880_test_init_verbs[] = {
+ /* Unmute inputs of 0x0c - 0x0f */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Vol output for 0x0c-0x0f */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Set output pins 0x14-0x17 */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Unmute output pins 0x14-0x17 */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Set input pins 0x18-0x1c */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Mute input pins 0x18-0x1b */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* ADC set up */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Analog input/passthru */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ { }
+};
+#endif
+
+/*
+ */
+
+static const char * const alc880_models[ALC880_MODEL_LAST] = {
+ [ALC880_3ST] = "3stack",
+ [ALC880_TCL_S700] = "tcl",
+ [ALC880_3ST_DIG] = "3stack-digout",
+ [ALC880_CLEVO] = "clevo",
+ [ALC880_5ST] = "5stack",
+ [ALC880_5ST_DIG] = "5stack-digout",
+ [ALC880_W810] = "w810",
+ [ALC880_Z71V] = "z71v",
+ [ALC880_6ST] = "6stack",
+ [ALC880_6ST_DIG] = "6stack-digout",
+ [ALC880_ASUS] = "asus",
+ [ALC880_ASUS_W1V] = "asus-w1v",
+ [ALC880_ASUS_DIG] = "asus-dig",
+ [ALC880_ASUS_DIG2] = "asus-dig2",
+ [ALC880_UNIWILL_DIG] = "uniwill",
+ [ALC880_UNIWILL_P53] = "uniwill-p53",
+ [ALC880_FUJITSU] = "fujitsu",
+ [ALC880_F1734] = "F1734",
+ [ALC880_LG] = "lg",
+ [ALC880_LG_LW] = "lg-lw",
+ [ALC880_MEDION_RIM] = "medion",
+#ifdef CONFIG_SND_DEBUG
+ [ALC880_TEST] = "test",
+#endif
+ [ALC880_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc880_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810),
+ SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST),
+ SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
+ SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
+ SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V),
+ /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
+ SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
+ SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
+ SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */
+ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
+ SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
+ SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
+ SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST),
+ SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST),
+ SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO),
+ SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO),
+ SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2),
+ SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG),
+ SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734),
+ SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
+ SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
+ SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
+ SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
+ SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
+ SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
+ SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734),
+ SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
+ SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
+ SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
+ SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG),
+ SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
+ SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW),
+ SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
+ SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */
+ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
+ /* default Intel */
+ SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST),
+ SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
+ {}
+};
+
+/*
+ * ALC880 codec presets
+ */
+static const struct alc_config_preset alc880_presets[] = {
+ [ALC880_3ST] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_3ST_DIG] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_TCL_S700] = {
+ .mixers = { alc880_tcl_s700_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_tcl_S700_init_verbs,
+ alc880_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */
+ .num_adc_nids = 1, /* single ADC */
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_5ST] = {
+ .mixers = { alc880_three_stack_mixer,
+ alc880_five_stack_mixer},
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_5stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+ .channel_mode = alc880_fivestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_5ST_DIG] = {
+ .mixers = { alc880_three_stack_mixer,
+ alc880_five_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_5stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+ .channel_mode = alc880_fivestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_6ST] = {
+ .mixers = { alc880_six_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_6stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+ .dac_nids = alc880_6st_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+ .channel_mode = alc880_sixstack_modes,
+ .input_mux = &alc880_6stack_capture_source,
+ },
+ [ALC880_6ST_DIG] = {
+ .mixers = { alc880_six_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_6stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+ .dac_nids = alc880_6st_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+ .channel_mode = alc880_sixstack_modes,
+ .input_mux = &alc880_6stack_capture_source,
+ },
+ [ALC880_W810] = {
+ .mixers = { alc880_w810_base_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_w810_init_verbs,
+ alc880_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
+ .dac_nids = alc880_w810_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
+ .channel_mode = alc880_w810_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_Z71V] = {
+ .mixers = { alc880_z71v_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_z71v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
+ .dac_nids = alc880_z71v_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_F1734] = {
+ .mixers = { alc880_f1734_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_f1734_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
+ .dac_nids = alc880_f1734_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_f1734_capture_source,
+ .unsol_event = alc880_uniwill_p53_unsol_event,
+ .setup = alc880_uniwill_p53_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC880_ASUS] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_DIG] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_DIG2] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio2_init_verbs }, /* use GPIO2 */
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_W1V] = {
+ .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_UNIWILL_DIG] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_UNIWILL] = {
+ .mixers = { alc880_uniwill_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_uniwill_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ .unsol_event = alc880_uniwill_unsol_event,
+ .setup = alc880_uniwill_setup,
+ .init_hook = alc880_uniwill_init_hook,
+ },
+ [ALC880_UNIWILL_P53] = {
+ .mixers = { alc880_uniwill_p53_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_uniwill_p53_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
+ .channel_mode = alc880_threestack_modes,
+ .input_mux = &alc880_capture_source,
+ .unsol_event = alc880_uniwill_p53_unsol_event,
+ .setup = alc880_uniwill_p53_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC880_FUJITSU] = {
+ .mixers = { alc880_fujitsu_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_uniwill_p53_init_verbs,
+ alc880_beep_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ .unsol_event = alc880_uniwill_p53_unsol_event,
+ .setup = alc880_uniwill_p53_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC880_CLEVO] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_clevo_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_LG] = {
+ .mixers = { alc880_lg_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_lg_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
+ .dac_nids = alc880_lg_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
+ .channel_mode = alc880_lg_ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_lg_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc880_lg_setup,
+ .init_hook = alc_hp_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .loopbacks = alc880_lg_loopbacks,
+#endif
+ },
+ [ALC880_LG_LW] = {
+ .mixers = { alc880_lg_lw_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_lg_lw_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes),
+ .channel_mode = alc880_lg_lw_modes,
+ .input_mux = &alc880_lg_lw_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc880_lg_lw_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC880_MEDION_RIM] = {
+ .mixers = { alc880_medion_rim_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_medion_rim_init_verbs,
+ alc_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_medion_rim_capture_source,
+ .unsol_event = alc880_medion_rim_unsol_event,
+ .setup = alc880_medion_rim_setup,
+ .init_hook = alc880_medion_rim_automute,
+ },
+#ifdef CONFIG_SND_DEBUG
+ [ALC880_TEST] = {
+ .mixers = { alc880_test_mixer },
+ .init_verbs = { alc880_test_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
+ .dac_nids = alc880_test_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_test_modes),
+ .channel_mode = alc880_test_modes,
+ .input_mux = &alc880_test_capture_source,
+ },
+#endif
+};
+
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
new file mode 100644
index 0000000..617d047
--- /dev/null
+++ b/sound/pci/hda/alc882_quirks.c
@@ -0,0 +1,3755 @@
+/*
+ * ALC882/ALC883/ALC888/ALC889 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC882 models */
+enum {
+ ALC882_AUTO,
+ ALC882_3ST_DIG,
+ ALC882_6ST_DIG,
+ ALC882_ARIMA,
+ ALC882_W2JC,
+ ALC882_TARGA,
+ ALC882_ASUS_A7J,
+ ALC882_ASUS_A7M,
+ ALC885_MACPRO,
+ ALC885_MBA21,
+ ALC885_MBP3,
+ ALC885_MB5,
+ ALC885_MACMINI3,
+ ALC885_IMAC24,
+ ALC885_IMAC91,
+ ALC883_3ST_2ch_DIG,
+ ALC883_3ST_6ch_DIG,
+ ALC883_3ST_6ch,
+ ALC883_6ST_DIG,
+ ALC883_TARGA_DIG,
+ ALC883_TARGA_2ch_DIG,
+ ALC883_TARGA_8ch_DIG,
+ ALC883_ACER,
+ ALC883_ACER_ASPIRE,
+ ALC888_ACER_ASPIRE_4930G,
+ ALC888_ACER_ASPIRE_6530G,
+ ALC888_ACER_ASPIRE_8930G,
+ ALC888_ACER_ASPIRE_7730G,
+ ALC883_MEDION,
+ ALC883_MEDION_WIM2160,
+ ALC883_LAPTOP_EAPD,
+ ALC883_LENOVO_101E_2ch,
+ ALC883_LENOVO_NB0763,
+ ALC888_LENOVO_MS7195_DIG,
+ ALC888_LENOVO_SKY,
+ ALC883_HAIER_W66,
+ ALC888_3ST_HP,
+ ALC888_6ST_DELL,
+ ALC883_MITAC,
+ ALC883_CLEVO_M540R,
+ ALC883_CLEVO_M720,
+ ALC883_FUJITSU_PI2515,
+ ALC888_FUJITSU_XA3530,
+ ALC883_3ST_6ch_INTEL,
+ ALC889A_INTEL,
+ ALC889_INTEL,
+ ALC888_ASUS_M90V,
+ ALC888_ASUS_EEE1601,
+ ALC889A_MB31,
+ ALC1200_ASUS_P5Q,
+ ALC883_SONY_VAIO_TT,
+ ALC882_MODEL_LAST,
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc888_4ST_ch2_intel_init[] = {
+/* Mic-in jack as mic in */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+/* Line-in jack as Line in */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+/* Line-Out as Front */
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static const struct hda_verb alc888_4ST_ch4_intel_init[] = {
+/* Mic-in jack as mic in */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+/* Line-in jack as Surround */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as Front */
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc888_4ST_ch6_intel_init[] = {
+/* Mic-in jack as CLFE */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-in jack as Surround */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static const struct hda_verb alc888_4ST_ch8_intel_init[] = {
+/* Mic-in jack as CLFE */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-in jack as Surround */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+/* Line-Out as Side */
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
+ { 2, alc888_4ST_ch2_intel_init },
+ { 4, alc888_4ST_ch4_intel_init },
+ { 6, alc888_4ST_ch6_intel_init },
+ { 8, alc888_4ST_ch8_intel_init },
+};
+
+/*
+ * ALC888 Fujitsu Siemens Amillo xa3530
+ */
+
+static const struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
+/* Front Mic: set to PIN_IN (empty by default) */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Connect Internal HP to Front */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Bass HP to Front */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Line-Out side jack (SPDIF) to Side */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+/* Connect Mic jack to CLFE */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
+/* Connect Line-in jack to Surround */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+/* Connect HP out jack to Front */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Enable unsolicited event for HP jack and Line-out jack */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {}
+};
+
+static void alc889_automute_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x17;
+ spec->autocfg.speaker_pins[3] = 0x19;
+ spec->autocfg.speaker_pins[4] = 0x1a;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc889_intel_init_hook(struct hda_codec *codec)
+{
+ alc889_coef_init(codec);
+ alc_hp_automute(codec);
+}
+
+static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x17; /* line-out */
+ spec->autocfg.hp_pins[1] = 0x1b; /* hp */
+ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
+ spec->autocfg.speaker_pins[1] = 0x15; /* bass */
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/*
+ * ALC888 Acer Aspire 4930G model
+ */
+
+static const struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
+/* Front Mic: set to PIN_IN (empty by default) */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Connect Internal HP to front */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect HP out to front */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/*
+ * ALC888 Acer Aspire 6530G model
+ */
+
+static const struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+/* Route to built-in subwoofer as well as speakers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+/* Bias voltage on for external mic port */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Front Mic: set to PIN_IN (empty by default) */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/*
+ *ALC888 Acer Aspire 7730G model
+ */
+
+static const struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+/* Bias voltage on for external mic port */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Front Mic: set to PIN_IN (empty by default) */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/*Enable internal subwoofer */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x17, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/*
+ * ALC889 Acer Aspire 8930G model
+ */
+
+static const struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
+/* Front Mic: set to PIN_IN (empty by default) */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+/* Connect Internal Front to Front */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Internal Rear to Rear */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+/* Connect Internal CLFE to CLFE */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+/* Connect HP out to Front */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Enable all DACs */
+/* DAC DISABLE/MUTE 1? */
+/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x03},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0000},
+/* DAC DISABLE/MUTE 2? */
+/* some bit here disables the other DACs. Init=0x4900 */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x08},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0000},
+/* DMIC fix
+ * This laptop has a stereo digital microphone. The mics are only 1cm apart
+ * which makes the stereo useless. However, either the mic or the ALC889
+ * makes the signal become a difference/sum signal instead of standard
+ * stereo, which is annoying. So instead we flip this bit which makes the
+ * codec replicate the sum signal to both channels, turning it into a
+ * normal mono mic.
+ */
+/* DMIC_CONTROL? Init value = 0x0001 */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0003},
+ { }
+};
+
+static const struct hda_input_mux alc888_2_capture_sources[2] = {
+ /* Front mic only available on one ADC */
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Front Mic", 0xb },
+ },
+ },
+ {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+ }
+};
+
+static const struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
+ /* Interal mic only available on one ADC */
+ {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line In", 0x2 },
+ { "CD", 0x4 },
+ { "Input Mix", 0xa },
+ { "Internal Mic", 0xb },
+ },
+ },
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line In", 0x2 },
+ { "CD", 0x4 },
+ { "Input Mix", 0xa },
+ },
+ }
+};
+
+static const struct hda_input_mux alc889_capture_sources[3] = {
+ /* Digital mic only available on first "ADC" */
+ {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Front Mic", 0xb },
+ { "Input Mix", 0xa },
+ },
+ },
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Input Mix", 0xa },
+ },
+ },
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Input Mix", 0xa },
+ },
+ }
+};
+
+static const struct snd_kcontrol_new alc888_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Internal LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Internal LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x1b;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+#define ALC882_DIGOUT_NID 0x06
+#define ALC882_DIGIN_NID 0x0a
+#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID
+#define ALC883_DIGIN_NID ALC882_DIGIN_NID
+#define ALC1200_DIGOUT_NID 0x10
+
+
+static const struct hda_channel_mode alc882_ch_modes[1] = {
+ { 8, NULL }
+};
+
+/* DACs */
+static const hda_nid_t alc882_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+#define alc883_dac_nids alc882_dac_nids
+
+/* ADCs */
+#define alc882_adc_nids alc880_adc_nids
+#define alc882_adc_nids_alt alc880_adc_nids_alt
+#define alc883_adc_nids alc882_adc_nids_alt
+static const hda_nid_t alc883_adc_nids_alt[1] = { 0x08 };
+static const hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 };
+#define alc889_adc_nids alc880_adc_nids
+
+static const hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
+static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
+#define alc883_capsrc_nids alc882_capsrc_nids_alt
+static const hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
+#define alc889_capsrc_nids alc882_capsrc_nids
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+
+static const struct hda_input_mux alc882_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+#define alc883_capture_source alc882_capture_source
+
+static const struct hda_input_mux alc889_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Mic", 0x3 },
+ { "Line", 0x2 },
+ },
+};
+
+static const struct hda_input_mux mb5_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x7 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux macmini3_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc883_3stack_6ch_intel = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x1 },
+ { "Front Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc883_lenovo_101e_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static const struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ },
+};
+
+static const struct hda_input_mux alc883_lenovo_sky_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc883_asus_eee1601_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ },
+};
+
+static const struct hda_input_mux alc889A_mb31_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ /* Front Mic (0x01) unused */
+ { "Line", 0x2 },
+ /* Line 2 (0x03) unused */
+ /* CD (0x04) unused? */
+ },
+};
+
+static const struct hda_input_mux alc889A_imac91_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x01 },
+ { "Line", 0x2 }, /* Not sure! */
+ },
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
+ { 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc882_3ST_ch2_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static const struct hda_verb alc882_3ST_ch4_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc882_3ST_ch6_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
+ { 2, alc882_3ST_ch2_init },
+ { 4, alc882_3ST_ch4_init },
+ { 6, alc882_3ST_ch6_init },
+};
+
+#define alc883_3ST_6ch_modes alc882_3ST_6ch_modes
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc883_3ST_ch2_clevo_init[] = {
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static const struct hda_verb alc883_3ST_ch4_clevo_init[] = {
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc883_3ST_ch6_clevo_init[] = {
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = {
+ { 2, alc883_3ST_ch2_clevo_init },
+ { 4, alc883_3ST_ch4_clevo_init },
+ { 6, alc883_3ST_ch6_clevo_init },
+};
+
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc882_sixstack_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static const struct hda_verb alc882_sixstack_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc882_sixstack_modes[2] = {
+ { 6, alc882_sixstack_ch6_init },
+ { 8, alc882_sixstack_ch8_init },
+};
+
+
+/* Macbook Air 2,1 */
+
+static const struct hda_channel_mode alc885_mba21_ch_modes[1] = {
+ { 2, NULL },
+};
+
+/*
+ * macbook pro ALC885 can switch LineIn to LineOut without losing Mic
+ */
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc885_mbp_ch2_init[] = {
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static const struct hda_verb alc885_mbp_ch4_init[] = {
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
+ { 2, alc885_mbp_ch2_init },
+ { 4, alc885_mbp_ch4_init },
+};
+
+/*
+ * 2ch
+ * Speakers/Woofer/HP = Front
+ * LineIn = Input
+ */
+static const struct hda_verb alc885_mb5_ch2_init[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ * Speakers/HP = Front
+ * Woofer = LFE
+ * LineIn = Surround
+ */
+static const struct hda_verb alc885_mb5_ch6_init[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
+ { 2, alc885_mb5_ch2_init },
+ { 6, alc885_mb5_ch6_init },
+};
+
+#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc883_4ST_ch2_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static const struct hda_verb alc883_4ST_ch4_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc883_4ST_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static const struct hda_verb alc883_4ST_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
+ { 2, alc883_4ST_ch2_init },
+ { 4, alc883_4ST_ch4_init },
+ { 6, alc883_4ST_ch6_init },
+ { 8, alc883_4ST_ch8_init },
+};
+
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc883_3ST_ch2_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static const struct hda_verb alc883_3ST_ch4_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc883_3ST_ch6_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
+ { 2, alc883_3ST_ch2_intel_init },
+ { 4, alc883_3ST_ch4_intel_init },
+ { 6, alc883_3ST_ch6_intel_init },
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc889_ch2_intel_init[] = {
+ { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc889_ch6_intel_init[] = {
+ { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static const struct hda_verb alc889_ch8_intel_init[] = {
+ { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc889_8ch_intel_modes[3] = {
+ { 2, alc889_ch2_intel_init },
+ { 6, alc889_ch6_intel_init },
+ { 8, alc889_ch8_intel_init },
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc883_sixstack_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static const struct hda_verb alc883_sixstack_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc883_sixstack_modes[2] = {
+ { 6, alc883_sixstack_ch6_init },
+ { 8, alc883_sixstack_ch8_init },
+};
+
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+static const struct snd_kcontrol_new alc882_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+/* Macbook Air 2,1 same control for HP and internal Speaker */
+
+static const struct snd_kcontrol_new alc885_mba21_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT),
+ { }
+};
+
+
+static const struct snd_kcontrol_new alc885_mbp3_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc885_mb5_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc885_macmini3_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc885_imac91_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ { } /* end */
+};
+
+
+static const struct snd_kcontrol_new alc882_w2jc_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc882_targa_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ???
+ * Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c
+ */
+static const struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mobile Front Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mobile Line Playback Volume", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc882_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static const struct hda_verb alc882_base_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* CLFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Side mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* Side Pin: output 3 (0x0f) */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ { }
+};
+
+static const struct hda_verb alc882_adc1_init_verbs[] = {
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { }
+};
+
+static const struct hda_verb alc882_eapd_verbs[] = {
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+ { }
+};
+
+static const struct hda_verb alc889_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static const struct hda_verb alc_hp15_unsol_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static const struct hda_verb alc885_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* CLFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Side mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Front HP Pin: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Rear Pin: output 1 (0x0d) */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x19, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* Side Pin: output 3 (0x0f) */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ /* Mixer elements: 0x18, , 0x1a, 0x1b */
+ /* Input mixer1 */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+ { }
+};
+
+static const struct hda_verb alc885_init_input_verbs[] = {
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ { }
+};
+
+
+/* Unmute Selector 24h and set the default input to front mic */
+static const struct hda_verb alc889_init_input_verbs[] = {
+ {0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ { }
+};
+
+
+#define alc883_init_verbs alc882_base_init_verbs
+
+/* Mac Pro test */
+static const struct snd_kcontrol_new alc882_macpro_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ /* FIXME: this looks suspicious...
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ */
+ { } /* end */
+};
+
+static const struct hda_verb alc882_macpro_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Speaker: output */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x04},
+ /* Headphone output (output 0 - 0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ { }
+};
+
+/* Macbook 5,1 */
+static const struct hda_verb alc885_mb5_init_verbs[] = {
+ /* DACs */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Front mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Surround mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* LFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* HP mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin (0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* LFE Pin (0x0e) */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* HP Pin (0x0f) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
+ { }
+};
+
+/* Macmini 3,1 */
+static const struct hda_verb alc885_macmini3_init_verbs[] = {
+ /* DACs */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Front mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Surround mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* LFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* HP mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin (0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* LFE Pin (0x0e) */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* HP Pin (0x0f) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ /* Line In pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ { }
+};
+
+
+static const struct hda_verb alc885_mba21_init_verbs[] = {
+ /*Internal and HP Speaker Mixer*/
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /*Internal Speaker Pin (0x0c)*/
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP Pin: output 0 (0x0e) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
+ /* Line in (is hp when jack connected)*/
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ { }
+ };
+
+
+/* Macbook Pro rev3 */
+static const struct hda_verb alc885_mbp3_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* HP mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP Pin: output 0 (0x0e) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: use output 1 when in LineOut mode */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ { }
+};
+
+/* iMac 9,1 */
+static const struct hda_verb alc885_imac91_init_verbs[] = {
+ /* Internal Speaker Pin (0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP Pin: Rear */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
+ /* Line in Rear */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { }
+};
+
+/* iMac 24 mixer. */
+static const struct snd_kcontrol_new alc885_imac24_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT),
+ { } /* end */
+};
+
+/* iMac 24 init verbs. */
+static const struct hda_verb alc885_imac24_init_verbs[] = {
+ /* Internal speakers: output 0 (0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Internal speakers: output 0 (0x0c) */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Headphone: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ /* Front Mic: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ { }
+};
+
+/* Toggle speaker-output according to the hp-jack state */
+static void alc885_imac24_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x18;
+ spec->autocfg.speaker_pins[1] = 0x1a;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+#define alc885_mb5_setup alc885_imac24_setup
+#define alc885_macmini3_setup alc885_imac24_setup
+
+/* Macbook Air 2,1 */
+static void alc885_mba21_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x18;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+
+
+static void alc885_mbp3_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc885_imac91_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x18;
+ spec->autocfg.speaker_pins[1] = 0x1a;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct hda_verb alc882_targa_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc882_targa_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc_hp_automute(codec);
+ snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+ spec->jack_present ? 1 : 3);
+}
+
+static void alc882_targa_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) == ALC_HP_EVENT)
+ alc882_targa_automute(codec);
+}
+
+static const struct hda_verb alc882_asus_a7j_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ { } /* end */
+};
+
+static const struct hda_verb alc882_asus_a7m_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ { } /* end */
+};
+
+static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
+{
+ unsigned int gpiostate, gpiomask, gpiodir;
+
+ gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DATA, 0);
+
+ if (!muted)
+ gpiostate |= (1 << pin);
+ else
+ gpiostate &= ~(1 << pin);
+
+ gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_MASK, 0);
+ gpiomask |= (1 << pin);
+
+ gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DIRECTION, 0);
+ gpiodir |= (1 << pin);
+
+
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_MASK, gpiomask);
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DIRECTION, gpiodir);
+
+ msleep(1);
+
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DATA, gpiostate);
+}
+
+/* set up GPIO at initialization */
+static void alc885_macpro_init_hook(struct hda_codec *codec)
+{
+ alc882_gpio_mute(codec, 0, 0);
+ alc882_gpio_mute(codec, 1, 0);
+}
+
+/* set up GPIO and update auto-muting at initialization */
+static void alc885_imac24_init_hook(struct hda_codec *codec)
+{
+ alc885_macpro_init_hook(codec);
+ alc_hp_automute(codec);
+}
+
+/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
+static const struct hda_verb alc889A_mb31_ch2_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
+ { } /* end */
+};
+
+/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
+static const struct hda_verb alc889A_mb31_ch4_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
+ { } /* end */
+};
+
+/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
+static const struct hda_verb alc889A_mb31_ch5_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
+ { } /* end */
+};
+
+/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
+static const struct hda_verb alc889A_mb31_ch6_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
+ { 2, alc889A_mb31_ch2_init },
+ { 4, alc889A_mb31_ch4_init },
+ { 5, alc889A_mb31_ch5_init },
+ { 6, alc889A_mb31_ch6_init },
+};
+
+static const struct hda_verb alc883_medion_eapd_verbs[] = {
+ /* eanable EAPD on medion laptop */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
+ { }
+};
+
+#define alc883_base_mixer alc882_base_mixer
+
+static const struct snd_kcontrol_new alc883_mitac_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc885_8ch_intel_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x1b, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_fivestack_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_targa_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc883_medion_wim2160_verbs[] = {
+ /* Unmute front mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Set speaker pin to front mixer */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Init headphone pin */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_medion_wim2160_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1a;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume",
+ 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc889A_mb31_mixer[] = {
+ /* Output mixers */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
+ /* Output switches */
+ HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
+ /* Boost mixers */
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
+ /* Input mixers */
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_vaiott_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc883_bind_cap_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc883_bind_cap_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
+ HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol),
+ HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc883_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_mitac_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct hda_verb alc883_mitac_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Subwoofer */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN}, */
+
+ { } /* end */
+};
+
+static const struct hda_verb alc883_clevo_m540r_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Int speaker */
+ /*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/
+
+ /* enable unsolicited event */
+ /*
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+ */
+
+ { } /* end */
+};
+
+static const struct hda_verb alc883_clevo_m720_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Int speaker */
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+static const struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
+ /* HP */
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Subwoofer */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+static const struct hda_verb alc883_targa_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+/* Connect Line-Out side jack (SPDIF) to Side */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+/* Connect Mic jack to CLFE */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
+/* Connect Line-in jack to Surround */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+/* Connect HP out jack to Front */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+static const struct hda_verb alc883_lenovo_101e_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_FRONT_EVENT|AC_USRSP_EN},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT|AC_USRSP_EN},
+ { } /* end */
+};
+
+static const struct hda_verb alc883_lenovo_nb0763_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ { } /* end */
+};
+
+static const struct hda_verb alc888_lenovo_ms7195_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_FRONT_EVENT | AC_USRSP_EN},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static const struct hda_verb alc883_haier_w66_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ { } /* end */
+};
+
+static const struct hda_verb alc888_lenovo_sky_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static const struct hda_verb alc888_6st_dell_verbs[] = {
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static const struct hda_verb alc883_vaiott_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+
+ /* enable unsolicited event */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+static void alc888_3st_hp_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x18;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct hda_verb alc888_3st_hp_verbs[] = {
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc888_3st_hp_2ch_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static const struct hda_verb alc888_3st_hp_4ch_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc888_3st_hp_6ch_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc888_3st_hp_modes[3] = {
+ { 2, alc888_3st_hp_2ch_init },
+ { 4, alc888_3st_hp_4ch_init },
+ { 6, alc888_3st_hp_6ch_init },
+};
+
+static void alc888_lenovo_ms7195_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.line_out_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_lenovo_nb0763_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/* toggle speaker-output according to the hp-jack state */
+#define alc883_targa_init_hook alc882_targa_init_hook
+#define alc883_targa_unsol_event alc882_targa_unsol_event
+
+static void alc883_clevo_m720_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
+{
+ alc_hp_automute(codec);
+ alc88x_simple_mic_automute(codec);
+}
+
+static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC_MIC_EVENT:
+ alc88x_simple_mic_automute(codec);
+ break;
+ default:
+ alc_sku_unsol_event(codec, res);
+ break;
+ }
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc883_haier_w66_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc883_lenovo_101e_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.line_out_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->detect_line = 1;
+ spec->automute_lines = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_acer_aspire_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct hda_verb alc883_acer_eapd_verbs[] = {
+ /* HP Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front Pin: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* eanable EAPD on medion laptop */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static void alc888_6st_dell_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[2] = 0x16;
+ spec->autocfg.speaker_pins[3] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc888_lenovo_sky_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[2] = 0x16;
+ spec->autocfg.speaker_pins[3] = 0x17;
+ spec->autocfg.speaker_pins[4] = 0x1a;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc883_vaiott_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct hda_verb alc888_asus_m90v_verbs[] = {
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* enable unsolicited event */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static void alc883_mode2_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[2] = 0x16;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static const struct hda_verb alc888_asus_eee1601_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0838},
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static void alc883_eee1601_inithook(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ alc_hp_automute(codec);
+}
+
+static const struct hda_verb alc889A_mb31_verbs[] = {
+ /* Init rear pin (used as headphone output) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ /* Init line pin (used as output in 4ch and 6ch mode) */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */
+ /* Init line 2 pin (used as headphone out by default) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
+ { } /* end */
+};
+
+/* Mute speakers according to the headphone jack state */
+static void alc889A_mb31_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ /* Mute only in 2ch or 4ch mode */
+ if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
+ == 0x00) {
+ present = snd_hda_jack_detect(codec, 0x15);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ }
+}
+
+static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) == ALC_HP_EVENT)
+ alc889A_mb31_automute(codec);
+}
+
+static const hda_nid_t alc883_slave_dig_outs[] = {
+ ALC1200_DIGOUT_NID, 0,
+};
+
+static const hda_nid_t alc1200_slave_dig_outs[] = {
+ ALC883_DIGOUT_NID, 0,
+};
+
+/*
+ * configuration and preset
+ */
+static const char * const alc882_models[ALC882_MODEL_LAST] = {
+ [ALC882_3ST_DIG] = "3stack-dig",
+ [ALC882_6ST_DIG] = "6stack-dig",
+ [ALC882_ARIMA] = "arima",
+ [ALC882_W2JC] = "w2jc",
+ [ALC882_TARGA] = "targa",
+ [ALC882_ASUS_A7J] = "asus-a7j",
+ [ALC882_ASUS_A7M] = "asus-a7m",
+ [ALC885_MACPRO] = "macpro",
+ [ALC885_MB5] = "mb5",
+ [ALC885_MACMINI3] = "macmini3",
+ [ALC885_MBA21] = "mba21",
+ [ALC885_MBP3] = "mbp3",
+ [ALC885_IMAC24] = "imac24",
+ [ALC885_IMAC91] = "imac91",
+ [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig",
+ [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig",
+ [ALC883_3ST_6ch] = "3stack-6ch",
+ [ALC883_6ST_DIG] = "alc883-6stack-dig",
+ [ALC883_TARGA_DIG] = "targa-dig",
+ [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
+ [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig",
+ [ALC883_ACER] = "acer",
+ [ALC883_ACER_ASPIRE] = "acer-aspire",
+ [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g",
+ [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g",
+ [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g",
+ [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g",
+ [ALC883_MEDION] = "medion",
+ [ALC883_MEDION_WIM2160] = "medion-wim2160",
+ [ALC883_LAPTOP_EAPD] = "laptop-eapd",
+ [ALC883_LENOVO_101E_2ch] = "lenovo-101e",
+ [ALC883_LENOVO_NB0763] = "lenovo-nb0763",
+ [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
+ [ALC888_LENOVO_SKY] = "lenovo-sky",
+ [ALC883_HAIER_W66] = "haier-w66",
+ [ALC888_3ST_HP] = "3stack-hp",
+ [ALC888_6ST_DELL] = "6stack-dell",
+ [ALC883_MITAC] = "mitac",
+ [ALC883_CLEVO_M540R] = "clevo-m540r",
+ [ALC883_CLEVO_M720] = "clevo-m720",
+ [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
+ [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
+ [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel",
+ [ALC889A_INTEL] = "intel-alc889a",
+ [ALC889_INTEL] = "intel-x58",
+ [ALC1200_ASUS_P5Q] = "asus-p5q",
+ [ALC889A_MB31] = "mb31",
+ [ALC883_SONY_VAIO_TT] = "sony-vaio-tt",
+ [ALC882_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc882_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
+
+ SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G",
+ ALC888_ACER_ASPIRE_4930G),
+ SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
+ ALC888_ACER_ASPIRE_4930G),
+ SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G",
+ ALC888_ACER_ASPIRE_8930G),
+ SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
+ ALC888_ACER_ASPIRE_8930G),
+ SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO),
+ SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO),
+ SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
+ ALC888_ACER_ASPIRE_6530G),
+ SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
+ ALC888_ACER_ASPIRE_6530G),
+ SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
+ ALC888_ACER_ASPIRE_7730G),
+ /* default Acer -- disabled as it causes more problems.
+ * model=auto should work fine now
+ */
+ /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */
+
+ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
+
+ SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
+ SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
+ SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
+ SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
+
+ SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
+ SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
+ SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
+ SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
+ SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
+
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
+ SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
+ SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
+ SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
+ SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
+
+ SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
+ SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO),
+ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
+
+ SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
+ SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
+ SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
+ SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
+ /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */
+ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
+ SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx",
+ ALC883_FUJITSU_PI2515),
+ SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1130, "Fujitsu AMILO Xa35xx",
+ ALC888_FUJITSU_XA3530),
+ SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
+ SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
+ SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
+ SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
+
+ SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
+ SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL),
+ SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG),
+
+ {}
+};
+
+/* codec SSID table for Intel Mac */
+static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO),
+ SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
+ SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
+ SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
+ SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M),
+ SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
+ SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
+ SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
+ SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
+ SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
+ SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
+ /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
+ * so apparently no perfect solution yet
+ */
+ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+ SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
+ SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3),
+ {} /* terminator */
+};
+
+static const struct alc_config_preset alc882_presets[] = {
+ [ALC882_3ST_DIG] = {
+ .mixers = { alc882_base_mixer },
+ .init_verbs = { alc882_base_init_verbs,
+ alc882_adc1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_6ST_DIG] = {
+ .mixers = { alc882_base_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs,
+ alc882_adc1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+ .channel_mode = alc882_sixstack_modes,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_ARIMA] = {
+ .mixers = { alc882_base_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+ .channel_mode = alc882_sixstack_modes,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_W2JC] = {
+ .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_eapd_verbs, alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ },
+ [ALC885_MBA21] = {
+ .mixers = { alc885_mba21_mixer },
+ .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs },
+ .num_dacs = 2,
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_mba21_ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
+ .input_mux = &alc882_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc885_mba21_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC885_MBP3] = {
+ .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_mbp3_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = 2,
+ .dac_nids = alc882_dac_nids,
+ .hp_nid = 0x04,
+ .channel_mode = alc885_mbp_4ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
+ .input_mux = &alc882_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc885_mbp3_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC885_MB5] = {
+ .mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_mb5_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_mb5_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
+ .input_mux = &mb5_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc885_mb5_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC885_MACMINI3] = {
+ .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_macmini3_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_macmini3_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes),
+ .input_mux = &macmini3_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc885_macmini3_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC885_MACPRO] = {
+ .mixers = { alc882_macpro_mixer },
+ .init_verbs = { alc882_macpro_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .input_mux = &alc882_capture_source,
+ .init_hook = alc885_macpro_init_hook,
+ },
+ [ALC885_IMAC24] = {
+ .mixers = { alc885_imac24_mixer },
+ .init_verbs = { alc885_imac24_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .input_mux = &alc882_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc885_imac24_setup,
+ .init_hook = alc885_imac24_init_hook,
+ },
+ [ALC885_IMAC91] = {
+ .mixers = {alc885_imac91_mixer},
+ .init_verbs = { alc885_imac91_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_mba21_ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
+ .input_mux = &alc889A_imac91_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc885_imac91_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC882_TARGA] = {
+ .mixers = { alc882_targa_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc880_gpio3_init_verbs, alc882_targa_verbs},
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+ .adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+ .channel_mode = alc882_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
+ },
+ [ALC882_ASUS_A7J] = {
+ .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_asus_a7j_verbs},
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+ .adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+ .channel_mode = alc882_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_ASUS_A7M] = {
+ .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_eapd_verbs, alc880_gpio1_init_verbs,
+ alc882_asus_a7m_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC883_3ST_2ch_DIG] = {
+ .mixers = { alc883_3ST_2ch_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_3ST_6ch_DIG] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_3ST_6ch] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_3ST_6ch_INTEL] = {
+ .mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .slave_dig_outs = alc883_slave_dig_outs,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
+ .channel_mode = alc883_3ST_6ch_intel_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_3stack_6ch_intel,
+ },
+ [ALC889A_INTEL] = {
+ .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc885_init_verbs, alc885_init_input_verbs,
+ alc_hp15_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+ .adc_nids = alc889_adc_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .slave_dig_outs = alc883_slave_dig_outs,
+ .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+ .channel_mode = alc889_8ch_intel_modes,
+ .capsrc_nids = alc889_capsrc_nids,
+ .input_mux = &alc889_capture_source,
+ .setup = alc889_automute_setup,
+ .init_hook = alc_hp_automute,
+ .unsol_event = alc_sku_unsol_event,
+ .need_dac_fix = 1,
+ },
+ [ALC889_INTEL] = {
+ .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc885_init_verbs, alc889_init_input_verbs,
+ alc889_eapd_verbs, alc_hp15_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+ .adc_nids = alc889_adc_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .slave_dig_outs = alc883_slave_dig_outs,
+ .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+ .channel_mode = alc889_8ch_intel_modes,
+ .capsrc_nids = alc889_capsrc_nids,
+ .input_mux = &alc889_capture_source,
+ .setup = alc889_automute_setup,
+ .init_hook = alc889_intel_init_hook,
+ .unsol_event = alc_sku_unsol_event,
+ .need_dac_fix = 1,
+ },
+ [ALC883_6ST_DIG] = {
+ .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_TARGA_DIG] = {
+ .mixers = { alc883_targa_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_targa_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_targa_unsol_event,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
+ },
+ [ALC883_TARGA_2ch_DIG] = {
+ .mixers = { alc883_targa_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_targa_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .adc_nids = alc883_adc_nids_alt,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_targa_unsol_event,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
+ },
+ [ALC883_TARGA_8ch_DIG] = {
+ .mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_targa_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes),
+ .channel_mode = alc883_4ST_8ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_targa_unsol_event,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
+ },
+ [ALC883_ACER] = {
+ .mixers = { alc883_base_mixer },
+ /* On TravelMate laptops, GPIO 0 enables the internal speaker
+ * and the headphone jack. Turn this on and rely on the
+ * standard mute methods whenever the user wants to turn
+ * these outputs off.
+ */
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_ACER_ASPIRE] = {
+ .mixers = { alc883_acer_aspire_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_acer_aspire_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_ACER_ASPIRE_4930G] = {
+ .mixers = { alc888_acer_aspire_4930g_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+ alc888_acer_aspire_4930g_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .const_channel_count = 6,
+ .num_mux_defs =
+ ARRAY_SIZE(alc888_2_capture_sources),
+ .input_mux = alc888_2_capture_sources,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_acer_aspire_4930g_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_ACER_ASPIRE_6530G] = {
+ .mixers = { alc888_acer_aspire_6530_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+ alc888_acer_aspire_6530g_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .num_mux_defs =
+ ARRAY_SIZE(alc888_2_capture_sources),
+ .input_mux = alc888_acer_aspire_6530_sources,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_acer_aspire_6530g_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_ACER_ASPIRE_8930G] = {
+ .mixers = { alc889_acer_aspire_8930g_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+ alc889_acer_aspire_8930g_verbs,
+ alc889_eapd_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+ .adc_nids = alc889_adc_nids,
+ .capsrc_nids = alc889_capsrc_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .const_channel_count = 6,
+ .num_mux_defs =
+ ARRAY_SIZE(alc889_capture_sources),
+ .input_mux = alc889_capture_sources,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc889_acer_aspire_8930g_setup,
+ .init_hook = alc_hp_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .power_hook = alc_power_eapd,
+#endif
+ },
+ [ALC888_ACER_ASPIRE_7730G] = {
+ .mixers = { alc883_3ST_6ch_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+ alc888_acer_aspire_7730G_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .const_channel_count = 6,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_acer_aspire_7730g_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC883_MEDION] = {
+ .mixers = { alc883_fivestack_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs,
+ alc883_medion_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .adc_nids = alc883_adc_nids_alt,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_MEDION_WIM2160] = {
+ .mixers = { alc883_medion_wim2160_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .adc_nids = alc883_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_medion_wim2160_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC883_LAPTOP_EAPD] = {
+ .mixers = { alc883_base_mixer },
+ .init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_CLEVO_M540R] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_clevo_modes),
+ .channel_mode = alc883_3ST_6ch_clevo_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ /* This machine has the hardware HP auto-muting, thus
+ * we need no software mute via unsol event
+ */
+ },
+ [ALC883_CLEVO_M720] = {
+ .mixers = { alc883_clevo_m720_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_clevo_m720_unsol_event,
+ .setup = alc883_clevo_m720_setup,
+ .init_hook = alc883_clevo_m720_init_hook,
+ },
+ [ALC883_LENOVO_101E_2ch] = {
+ .mixers = { alc883_lenovo_101e_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .adc_nids = alc883_adc_nids_alt,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_lenovo_101e_capture_source,
+ .setup = alc883_lenovo_101e_setup,
+ .unsol_event = alc_sku_unsol_event,
+ .init_hook = alc_inithook,
+ },
+ [ALC883_LENOVO_NB0763] = {
+ .mixers = { alc883_lenovo_nb0763_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_lenovo_nb0763_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_lenovo_nb0763_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_LENOVO_MS7195_DIG] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_lenovo_ms7195_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_lenovo_ms7195_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC883_HAIER_W66] = {
+ .mixers = { alc883_targa_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_haier_w66_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_3ST_HP] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
+ .channel_mode = alc888_3st_hp_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_3st_hp_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_6ST_DELL] = {
+ .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_6st_dell_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC883_MITAC] = {
+ .mixers = { alc883_mitac_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_mitac_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_mitac_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC883_FUJITSU_PI2515] = {
+ .mixers = { alc883_2ch_fujitsu_pi2515_mixer },
+ .init_verbs = { alc883_init_verbs,
+ alc883_2ch_fujitsu_pi2515_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_fujitsu_pi2515_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_2ch_fujitsu_pi2515_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_FUJITSU_XA3530] = {
+ .mixers = { alc888_base_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs,
+ alc888_fujitsu_xa3530_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc888_4ST_8ch_intel_modes),
+ .channel_mode = alc888_4ST_8ch_intel_modes,
+ .num_mux_defs =
+ ARRAY_SIZE(alc888_2_capture_sources),
+ .input_mux = alc888_2_capture_sources,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_fujitsu_xa3530_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_LENOVO_SKY] = {
+ .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_lenovo_sky_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc888_lenovo_sky_setup,
+ .init_hook = alc_hp_automute,
+ },
+ [ALC888_ASUS_M90V] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_asus_m90v_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_fujitsu_pi2515_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_mode2_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC888_ASUS_EEE1601] = {
+ .mixers = { alc883_asus_eee1601_mixer },
+ .cap_mixer = alc883_asus_eee1601_cap_mixer,
+ .init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_asus_eee1601_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .init_hook = alc883_eee1601_inithook,
+ },
+ [ALC1200_ASUS_P5Q] = {
+ .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC1200_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .slave_dig_outs = alc1200_slave_dig_outs,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC889A_MB31] = {
+ .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
+ .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
+ alc880_gpio1_init_verbs },
+ .adc_nids = alc883_adc_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .capsrc_nids = alc883_capsrc_nids,
+ .dac_nids = alc883_dac_nids,
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .channel_mode = alc889A_mb31_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
+ .input_mux = &alc889A_mb31_capture_source,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .unsol_event = alc889A_mb31_unsol_event,
+ .init_hook = alc889A_mb31_automute,
+ },
+ [ALC883_SONY_VAIO_TT] = {
+ .mixers = { alc883_vaiott_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_vaiott_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_vaiott_setup,
+ .init_hook = alc_hp_automute,
+ },
+};
+
+
diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c
new file mode 100644
index 0000000..2be1129
--- /dev/null
+++ b/sound/pci/hda/alc_quirks.c
@@ -0,0 +1,467 @@
+/*
+ * Common codes for Realtek codec quirks
+ * included by patch_realtek.c
+ */
+
+/*
+ * configuration template - to be copied to the spec instance
+ */
+struct alc_config_preset {
+ const struct snd_kcontrol_new *mixers[5]; /* should be identical size
+ * with spec
+ */
+ const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
+ const struct hda_verb *init_verbs[5];
+ unsigned int num_dacs;
+ const hda_nid_t *dac_nids;
+ hda_nid_t dig_out_nid; /* optional */
+ hda_nid_t hp_nid; /* optional */
+ const hda_nid_t *slave_dig_outs;
+ unsigned int num_adc_nids;
+ const hda_nid_t *adc_nids;
+ const hda_nid_t *capsrc_nids;
+ hda_nid_t dig_in_nid;
+ unsigned int num_channel_mode;
+ const struct hda_channel_mode *channel_mode;
+ int need_dac_fix;
+ int const_channel_count;
+ unsigned int num_mux_defs;
+ const struct hda_input_mux *input_mux;
+ void (*unsol_event)(struct hda_codec *, unsigned int);
+ void (*setup)(struct hda_codec *);
+ void (*init_hook)(struct hda_codec *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ const struct hda_amp_list *loopbacks;
+ void (*power_hook)(struct hda_codec *codec);
+#endif
+};
+
+/*
+ * channel mode setting
+ */
+static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
+ spec->num_channel_mode);
+}
+
+static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ spec->ext_channel_count);
+}
+
+static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ &spec->ext_channel_count);
+ if (err >= 0 && !spec->const_channel_count) {
+ spec->multiout.max_channels = spec->ext_channel_count;
+ if (spec->need_dac_fix)
+ spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+ }
+ return err;
+}
+
+/*
+ * Control the mode of pin widget settings via the mixer. "pc" is used
+ * instead of "%" to avoid consequences of accidentally treating the % as
+ * being part of a format specifier. Maximum allowed length of a value is
+ * 63 characters plus NULL terminator.
+ *
+ * Note: some retasking pin complexes seem to ignore requests for input
+ * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
+ * are requested. Therefore order this list so that this behaviour will not
+ * cause problems when mixer clients move through the enum sequentially.
+ * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
+ * March 2006.
+ */
+static const char * const alc_pin_mode_names[] = {
+ "Mic 50pc bias", "Mic 80pc bias",
+ "Line in", "Line out", "Headphone out",
+};
+static const unsigned char alc_pin_mode_values[] = {
+ PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
+};
+/* The control can present all 5 options, or it can limit the options based
+ * in the pin being assumed to be exclusively an input or an output pin. In
+ * addition, "input" pins may or may not process the mic bias option
+ * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
+ * accept requests for bias as of chip versions up to March 2006) and/or
+ * wiring in the computer.
+ */
+#define ALC_PIN_DIR_IN 0x00
+#define ALC_PIN_DIR_OUT 0x01
+#define ALC_PIN_DIR_INOUT 0x02
+#define ALC_PIN_DIR_IN_NOMICBIAS 0x03
+#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
+
+/* Info about the pin modes supported by the different pin direction modes.
+ * For each direction the minimum and maximum values are given.
+ */
+static const signed char alc_pin_mode_dir_info[5][2] = {
+ { 0, 2 }, /* ALC_PIN_DIR_IN */
+ { 3, 4 }, /* ALC_PIN_DIR_OUT */
+ { 0, 4 }, /* ALC_PIN_DIR_INOUT */
+ { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
+ { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
+};
+#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
+#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
+#define alc_pin_mode_n_items(_dir) \
+ (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
+
+static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int item_num = uinfo->value.enumerated.item;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
+
+ if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
+ item_num = alc_pin_mode_min(dir);
+ strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
+ return 0;
+}
+
+static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int i;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL,
+ 0x00);
+
+ /* Find enumerated value for current pinctl setting */
+ i = alc_pin_mode_min(dir);
+ while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
+ i++;
+ *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
+ return 0;
+}
+
+static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ signed int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL,
+ 0x00);
+
+ if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
+ val = alc_pin_mode_min(dir);
+
+ change = pinctl != alc_pin_mode_values[val];
+ if (change) {
+ /* Set pin mode to that requested */
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ alc_pin_mode_values[val]);
+
+ /* Also enable the retasking pin's input/output as required
+ * for the requested pin mode. Enum values of 2 or less are
+ * input modes.
+ *
+ * Dynamically switching the input/output buffers probably
+ * reduces noise slightly (particularly on input) so we'll
+ * do it. However, having both input and output buffers
+ * enabled simultaneously doesn't seem to be problematic if
+ * this turns out to be necessary in the future.
+ */
+ if (val <= 2) {
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+ HDA_AMP_MUTE, 0);
+ } else {
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, 0);
+ }
+ }
+ return change;
+}
+
+#define ALC_PIN_MODE(xname, nid, dir) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+ .info = alc_pin_mode_info, \
+ .get = alc_pin_mode_get, \
+ .put = alc_pin_mode_put, \
+ .private_value = nid | (dir<<16) }
+
+/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
+ * together using a mask with more than one bit set. This control is
+ * currently used only by the ALC260 test model. At this stage they are not
+ * needed for any "production" models.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_gpio_data_info snd_ctl_boolean_mono_info
+
+static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_DATA, 0x00);
+
+ *valp = (val & mask) != 0;
+ return 0;
+}
+static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ signed int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_DATA,
+ 0x00);
+
+ /* Set/unset the masked GPIO bit(s) as needed */
+ change = (val == 0 ? 0 : mask) != (gpio_data & mask);
+ if (val == 0)
+ gpio_data &= ~mask;
+ else
+ gpio_data |= mask;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_GPIO_DATA, gpio_data);
+
+ return change;
+}
+#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+ .info = alc_gpio_data_info, \
+ .get = alc_gpio_data_get, \
+ .put = alc_gpio_data_put, \
+ .private_value = nid | (mask<<16) }
+#endif /* CONFIG_SND_DEBUG */
+
+/* A switch control to allow the enabling of the digital IO pins on the
+ * ALC260. This is incredibly simplistic; the intention of this control is
+ * to provide something in the test model allowing digital outputs to be
+ * identified if present. If models are found which can utilise these
+ * outputs a more complete mixer control can be devised for those models if
+ * necessary.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
+
+static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1, 0x00);
+
+ *valp = (val & mask) != 0;
+ return 0;
+}
+static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ signed int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1,
+ 0x00);
+
+ /* Set/unset the masked control bit(s) as needed */
+ change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
+ if (val==0)
+ ctrl_data &= ~mask;
+ else
+ ctrl_data |= mask;
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+ ctrl_data);
+
+ return change;
+}
+#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+ .info = alc_spdif_ctrl_info, \
+ .get = alc_spdif_ctrl_get, \
+ .put = alc_spdif_ctrl_put, \
+ .private_value = nid | (mask<<16) }
+#endif /* CONFIG_SND_DEBUG */
+
+/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
+ * Again, this is only used in the ALC26x test models to help identify when
+ * the EAPD line must be asserted for features to work.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
+
+static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_EAPD_BTLENABLE, 0x00);
+
+ *valp = (val & mask) != 0;
+ return 0;
+}
+
+static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_EAPD_BTLENABLE,
+ 0x00);
+
+ /* Set/unset the masked control bit(s) as needed */
+ change = (!val ? 0 : mask) != (ctrl_data & mask);
+ if (!val)
+ ctrl_data &= ~mask;
+ else
+ ctrl_data |= mask;
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ ctrl_data);
+
+ return change;
+}
+
+#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
+ .info = alc_eapd_ctrl_info, \
+ .get = alc_eapd_ctrl_get, \
+ .put = alc_eapd_ctrl_put, \
+ .private_value = nid | (mask<<16) }
+#endif /* CONFIG_SND_DEBUG */
+
+static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+
+ if (!cfg->line_outs) {
+ while (cfg->line_outs < AUTO_CFG_MAX_OUTS &&
+ cfg->line_out_pins[cfg->line_outs])
+ cfg->line_outs++;
+ }
+ if (!cfg->speaker_outs) {
+ while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS &&
+ cfg->speaker_pins[cfg->speaker_outs])
+ cfg->speaker_outs++;
+ }
+ if (!cfg->hp_outs) {
+ while (cfg->hp_outs < AUTO_CFG_MAX_OUTS &&
+ cfg->hp_pins[cfg->hp_outs])
+ cfg->hp_outs++;
+ }
+}
+
+/*
+ * set up from the preset table
+ */
+static void setup_preset(struct hda_codec *codec,
+ const struct alc_config_preset *preset)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
+ add_mixer(spec, preset->mixers[i]);
+ spec->cap_mixer = preset->cap_mixer;
+ for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
+ i++)
+ add_verb(spec, preset->init_verbs[i]);
+
+ spec->channel_mode = preset->channel_mode;
+ spec->num_channel_mode = preset->num_channel_mode;
+ spec->need_dac_fix = preset->need_dac_fix;
+ spec->const_channel_count = preset->const_channel_count;
+
+ if (preset->const_channel_count)
+ spec->multiout.max_channels = preset->const_channel_count;
+ else
+ spec->multiout.max_channels = spec->channel_mode[0].channels;
+ spec->ext_channel_count = spec->channel_mode[0].channels;
+
+ spec->multiout.num_dacs = preset->num_dacs;
+ spec->multiout.dac_nids = preset->dac_nids;
+ spec->multiout.dig_out_nid = preset->dig_out_nid;
+ spec->multiout.slave_dig_outs = preset->slave_dig_outs;
+ spec->multiout.hp_nid = preset->hp_nid;
+
+ spec->num_mux_defs = preset->num_mux_defs;
+ if (!spec->num_mux_defs)
+ spec->num_mux_defs = 1;
+ spec->input_mux = preset->input_mux;
+
+ spec->num_adc_nids = preset->num_adc_nids;
+ spec->adc_nids = preset->adc_nids;
+ spec->capsrc_nids = preset->capsrc_nids;
+ spec->dig_in_nid = preset->dig_in_nid;
+
+ spec->unsol_event = preset->unsol_event;
+ spec->init_hook = preset->init_hook;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->power_hook = preset->power_hook;
+ spec->loopback.amplist = preset->loopbacks;
+#endif
+
+ if (preset->setup)
+ preset->setup(codec);
+
+ alc_fixup_autocfg_pin_nums(codec);
+}
+
+
+/* auto-toggle front mic */
+static void alc88x_simple_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_jack_detect(codec, 0x18);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
+}
+
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 45b4a8d..9c27a3a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -243,7 +243,8 @@
{
unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm);
unsigned int res;
- codec_exec_verb(codec, cmd, &res);
+ if (codec_exec_verb(codec, cmd, &res))
+ return -1;
return res;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_read);
@@ -307,14 +308,65 @@
}
EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes);
-static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
- hda_nid_t *conn_list, int max_conns);
-static bool add_conn_list(struct snd_array *array, hda_nid_t nid);
-static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
- hda_nid_t *src, int len);
+/* look up the cached results */
+static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid)
+{
+ int i, len;
+ for (i = 0; i < array->used; ) {
+ hda_nid_t *p = snd_array_elem(array, i);
+ if (nid == *p)
+ return p;
+ len = p[1];
+ i += len + 2;
+ }
+ return NULL;
+}
/**
- * snd_hda_get_connections - get connection list
+ * snd_hda_get_conn_list - get connection list
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @listp: the pointer to store NID list
+ *
+ * Parses the connection list of the given widget and stores the list
+ * of NIDs.
+ *
+ * Returns the number of connections, or a negative error code.
+ */
+int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
+ const hda_nid_t **listp)
+{
+ struct snd_array *array = &codec->conn_lists;
+ int len, err;
+ hda_nid_t list[HDA_MAX_CONNECTIONS];
+ hda_nid_t *p;
+ bool added = false;
+
+ again:
+ /* if the connection-list is already cached, read it */
+ p = lookup_conn_list(array, nid);
+ if (p) {
+ if (listp)
+ *listp = p + 2;
+ return p[1];
+ }
+ if (snd_BUG_ON(added))
+ return -EINVAL;
+
+ /* read the connection and add to the cache */
+ len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
+ if (len < 0)
+ return len;
+ err = snd_hda_override_conn_list(codec, nid, len, list);
+ if (err < 0)
+ return err;
+ added = true;
+ goto again;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
+
+/**
+ * snd_hda_get_connections - copy connection list
* @codec: the HDA codec
* @nid: NID to parse
* @conn_list: connection list array
@@ -328,42 +380,35 @@
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns)
{
- struct snd_array *array = &codec->conn_lists;
- int i, len, old_used;
- hda_nid_t list[HDA_MAX_CONNECTIONS];
+ const hda_nid_t *list;
+ int len = snd_hda_get_conn_list(codec, nid, &list);
- /* look up the cached results */
- for (i = 0; i < array->used; ) {
- hda_nid_t *p = snd_array_elem(array, i);
- len = p[1];
- if (nid == *p)
- return copy_conn_list(nid, conn_list, max_conns,
- p + 2, len);
- i += len + 2;
- }
-
- len = _hda_get_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
- if (len < 0)
+ if (len <= 0)
return len;
-
- /* add to the cache */
- old_used = array->used;
- if (!add_conn_list(array, nid) || !add_conn_list(array, len))
- goto error_add;
- for (i = 0; i < len; i++)
- if (!add_conn_list(array, list[i]))
- goto error_add;
-
- return copy_conn_list(nid, conn_list, max_conns, list, len);
-
- error_add:
- array->used = old_used;
- return -ENOMEM;
+ if (len > max_conns) {
+ snd_printk(KERN_ERR "hda_codec: "
+ "Too many connections %d for NID 0x%x\n",
+ len, nid);
+ return -EINVAL;
+ }
+ memcpy(conn_list, list, len * sizeof(hda_nid_t));
+ return len;
}
EXPORT_SYMBOL_HDA(snd_hda_get_connections);
-static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
- hda_nid_t *conn_list, int max_conns)
+/**
+ * snd_hda_get_raw_connections - copy connection list without cache
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @conn_list: connection list array
+ * @max_conns: max. number of connections to store
+ *
+ * Like snd_hda_get_connections(), copy the connection list but without
+ * checking through the connection-list cache.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *conn_list, int max_conns)
{
unsigned int parm;
int i, conn_len, conns;
@@ -376,11 +421,8 @@
wcaps = get_wcaps(codec, nid);
if (!(wcaps & AC_WCAP_CONN_LIST) &&
- get_wcaps_type(wcaps) != AC_WID_VOL_KNB) {
- snd_printk(KERN_WARNING "hda_codec: "
- "connection list not available for 0x%x\n", nid);
- return -EINVAL;
- }
+ get_wcaps_type(wcaps) != AC_WID_VOL_KNB)
+ return 0;
parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
if (parm & AC_CLIST_LONG) {
@@ -470,18 +512,77 @@
return true;
}
-static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
- hda_nid_t *src, int len)
+/**
+ * snd_hda_override_conn_list - add/modify the connection-list to cache
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @len: number of connection list entries
+ * @list: the list of connection entries
+ *
+ * Add or modify the given connection-list to the cache. If the corresponding
+ * cache already exists, invalidate it and append a new one.
+ *
+ * Returns zero or a negative error code.
+ */
+int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
+ const hda_nid_t *list)
{
- if (len > max_dst) {
- snd_printk(KERN_ERR "hda_codec: "
- "Too many connections %d for NID 0x%x\n",
- len, nid);
- return -EINVAL;
- }
- memcpy(dst, src, len * sizeof(hda_nid_t));
- return len;
+ struct snd_array *array = &codec->conn_lists;
+ hda_nid_t *p;
+ int i, old_used;
+
+ p = lookup_conn_list(array, nid);
+ if (p)
+ *p = -1; /* invalidate the old entry */
+
+ old_used = array->used;
+ if (!add_conn_list(array, nid) || !add_conn_list(array, len))
+ goto error_add;
+ for (i = 0; i < len; i++)
+ if (!add_conn_list(array, list[i]))
+ goto error_add;
+ return 0;
+
+ error_add:
+ array->used = old_used;
+ return -ENOMEM;
}
+EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
+
+/**
+ * snd_hda_get_conn_index - get the connection index of the given NID
+ * @codec: the HDA codec
+ * @mux: NID containing the list
+ * @nid: NID to select
+ * @recursive: 1 when searching NID recursively, otherwise 0
+ *
+ * Parses the connection list of the widget @mux and checks whether the
+ * widget @nid is present. If it is, return the connection index.
+ * Otherwise it returns -1.
+ */
+int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
+ hda_nid_t nid, int recursive)
+{
+ hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+ int i, nums;
+
+ nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
+ for (i = 0; i < nums; i++)
+ if (conn[i] == nid)
+ return i;
+ if (!recursive)
+ return -1;
+ if (recursive > 5) {
+ snd_printd("hda_codec: too deep connection for 0x%x\n", nid);
+ return -1;
+ }
+ recursive++;
+ for (i = 0; i < nums; i++)
+ if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
+ return i;
+ return -1;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
/**
* snd_hda_queue_unsol_event - add an unsolicited event to queue
@@ -1083,6 +1184,7 @@
snd_array_free(&codec->mixers);
snd_array_free(&codec->nids);
snd_array_free(&codec->conn_lists);
+ snd_array_free(&codec->spdif_out);
codec->bus->caddr_tbl[codec->addr] = NULL;
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
@@ -1144,6 +1246,7 @@
snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16);
snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8);
snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64);
+ snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16);
if (codec->bus->modelname) {
codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
if (!codec->modelname) {
@@ -2555,11 +2658,13 @@
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int idx = kcontrol->private_value;
+ struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
- ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff;
- ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff;
- ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff;
- ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff;
+ ucontrol->value.iec958.status[0] = spdif->status & 0xff;
+ ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff;
return 0;
}
@@ -2644,23 +2749,23 @@
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value;
+ int idx = kcontrol->private_value;
+ struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ hda_nid_t nid = spdif->nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
- codec->spdif_status = ucontrol->value.iec958.status[0] |
+ spdif->status = ucontrol->value.iec958.status[0] |
((unsigned int)ucontrol->value.iec958.status[1] << 8) |
((unsigned int)ucontrol->value.iec958.status[2] << 16) |
((unsigned int)ucontrol->value.iec958.status[3] << 24);
- val = convert_from_spdif_status(codec->spdif_status);
- val |= codec->spdif_ctls & 1;
- change = codec->spdif_ctls != val;
- codec->spdif_ctls = val;
-
- if (change)
+ val = convert_from_spdif_status(spdif->status);
+ val |= spdif->ctls & 1;
+ change = spdif->ctls != val;
+ spdif->ctls = val;
+ if (change && nid != (u16)-1)
set_dig_out_convert(codec, nid, val & 0xff, (val >> 8) & 0xff);
-
mutex_unlock(&codec->spdif_mutex);
return change;
}
@@ -2671,33 +2776,42 @@
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int idx = kcontrol->private_value;
+ struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
- ucontrol->value.integer.value[0] = codec->spdif_ctls & AC_DIG1_ENABLE;
+ ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
return 0;
}
+static inline void set_spdif_ctls(struct hda_codec *codec, hda_nid_t nid,
+ int dig1, int dig2)
+{
+ set_dig_out_convert(codec, nid, dig1, dig2);
+ /* unmute amp switch (if any) */
+ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
+ (dig1 & AC_DIG1_ENABLE))
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, 0);
+}
+
static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value;
+ int idx = kcontrol->private_value;
+ struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ hda_nid_t nid = spdif->nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
- val = codec->spdif_ctls & ~AC_DIG1_ENABLE;
+ val = spdif->ctls & ~AC_DIG1_ENABLE;
if (ucontrol->value.integer.value[0])
val |= AC_DIG1_ENABLE;
- change = codec->spdif_ctls != val;
- if (change) {
- codec->spdif_ctls = val;
- set_dig_out_convert(codec, nid, val & 0xff, -1);
- /* unmute amp switch (if any) */
- if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
- (val & AC_DIG1_ENABLE))
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, 0);
- }
+ change = spdif->ctls != val;
+ spdif->ctls = val;
+ if (change && nid != (u16)-1)
+ set_spdif_ctls(codec, nid, val & 0xff, -1);
mutex_unlock(&codec->spdif_mutex);
return change;
}
@@ -2744,36 +2858,79 @@
*
* Returns 0 if successful, or a negative error code.
*/
-int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
+ hda_nid_t associated_nid,
+ hda_nid_t cvt_nid)
{
int err;
struct snd_kcontrol *kctl;
struct snd_kcontrol_new *dig_mix;
int idx;
+ struct hda_spdif_out *spdif;
idx = find_empty_mixer_ctl_idx(codec, "IEC958 Playback Switch");
if (idx < 0) {
printk(KERN_ERR "hda_codec: too many IEC958 outputs\n");
return -EBUSY;
}
+ spdif = snd_array_new(&codec->spdif_out);
for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
kctl = snd_ctl_new1(dig_mix, codec);
if (!kctl)
return -ENOMEM;
kctl->id.index = idx;
- kctl->private_value = nid;
- err = snd_hda_ctl_add(codec, nid, kctl);
+ kctl->private_value = codec->spdif_out.used - 1;
+ err = snd_hda_ctl_add(codec, associated_nid, kctl);
if (err < 0)
return err;
}
- codec->spdif_ctls =
- snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_DIGI_CONVERT_1, 0);
- codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls);
+ spdif->nid = cvt_nid;
+ spdif->ctls = snd_hda_codec_read(codec, cvt_nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1, 0);
+ spdif->status = convert_to_spdif_status(spdif->ctls);
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
+struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
+ hda_nid_t nid)
+{
+ int i;
+ for (i = 0; i < codec->spdif_out.used; i++) {
+ struct hda_spdif_out *spdif =
+ snd_array_elem(&codec->spdif_out, i);
+ if (spdif->nid == nid)
+ return spdif;
+ }
+ return NULL;
+}
+EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid);
+
+void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
+{
+ struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+
+ mutex_lock(&codec->spdif_mutex);
+ spdif->nid = (u16)-1;
+ mutex_unlock(&codec->spdif_mutex);
+}
+EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign);
+
+void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
+{
+ struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ unsigned short val;
+
+ mutex_lock(&codec->spdif_mutex);
+ if (spdif->nid != nid) {
+ spdif->nid = nid;
+ val = spdif->ctls;
+ set_spdif_ctls(codec, nid, val & 0xff, (val >> 8) & 0xff);
+ }
+ mutex_unlock(&codec->spdif_mutex);
+}
+EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_assign);
+
/*
* SPDIF sharing with analog output
*/
@@ -3356,7 +3513,7 @@
*
* Returns 0 if successful, otherwise a negative error code.
*/
-static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
{
unsigned int i, val, wcaps;
@@ -3448,6 +3605,7 @@
return 0;
}
+EXPORT_SYMBOL_HDA(snd_hda_query_supported_pcm);
/**
* snd_hda_is_supported_format - Check the validity of the format
@@ -4177,10 +4335,12 @@
static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
unsigned int stream_tag, unsigned int format)
{
+ struct hda_spdif_out *spdif = snd_hda_spdif_out_of_nid(codec, nid);
+
/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
- if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+ if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
set_dig_out_convert(codec, nid,
- codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff,
+ spdif->ctls & ~AC_DIG1_ENABLE & 0xff,
-1);
snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
if (codec->slave_dig_outs) {
@@ -4190,9 +4350,9 @@
format);
}
/* turn on again (if needed) */
- if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+ if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
set_dig_out_convert(codec, nid,
- codec->spdif_ctls & 0xff, -1);
+ spdif->ctls & 0xff, -1);
}
static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
@@ -4348,6 +4508,8 @@
{
const hda_nid_t *nids = mout->dac_nids;
int chs = substream->runtime->channels;
+ struct hda_spdif_out *spdif =
+ snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
int i;
mutex_lock(&codec->spdif_mutex);
@@ -4356,7 +4518,7 @@
if (chs == 2 &&
snd_hda_is_supported_format(codec, mout->dig_out_nid,
format) &&
- !(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
+ !(spdif->status & IEC958_AES0_NONAUDIO)) {
mout->dig_out_used = HDA_DIG_ANALOG_DUP;
setup_dig_out_stream(codec, mout->dig_out_nid,
stream_tag, format);
@@ -4528,7 +4690,7 @@
unsigned int wid_caps = get_wcaps(codec, nid);
unsigned int wid_type = get_wcaps_type(wid_caps);
unsigned int def_conf;
- short assoc, loc;
+ short assoc, loc, conn, dev;
/* read all default configuration for pin complex */
if (wid_type != AC_WID_PIN)
@@ -4538,10 +4700,19 @@
continue;
def_conf = snd_hda_codec_get_pincfg(codec, nid);
- if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+ conn = get_defcfg_connect(def_conf);
+ if (conn == AC_JACK_PORT_NONE)
continue;
loc = get_defcfg_location(def_conf);
- switch (get_defcfg_device(def_conf)) {
+ dev = get_defcfg_device(def_conf);
+
+ /* workaround for buggy BIOS setups */
+ if (dev == AC_JACK_LINE_OUT) {
+ if (conn == AC_JACK_PORT_FIXED)
+ dev = AC_JACK_SPEAKER;
+ }
+
+ switch (dev) {
case AC_JACK_LINE_OUT:
seq = get_defcfg_sequence(def_conf);
assoc = get_defcfg_association(def_conf);
@@ -4957,17 +5128,15 @@
{
if (array->used >= array->alloced) {
int num = array->alloced + array->alloc_align;
+ int size = (num + 1) * array->elem_size;
+ int oldsize = array->alloced * array->elem_size;
void *nlist;
if (snd_BUG_ON(num >= 4096))
return NULL;
- nlist = kcalloc(num + 1, array->elem_size, GFP_KERNEL);
+ nlist = krealloc(array->list, size, GFP_KERNEL);
if (!nlist)
return NULL;
- if (array->list) {
- memcpy(nlist, array->list,
- array->elem_size * array->alloced);
- kfree(array->list);
- }
+ memset(nlist + oldsize, 0, size - oldsize);
array->list = nlist;
array->alloced = num;
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 59c9730..f465e07 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -829,8 +829,7 @@
struct mutex spdif_mutex;
struct mutex control_mutex;
- unsigned int spdif_status; /* IEC958 status bits */
- unsigned short spdif_ctls; /* SPDIF control bits */
+ struct snd_array spdif_out;
unsigned int spdif_in_enable; /* SPDIF input enable? */
const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
struct snd_array init_pins; /* initial (BIOS) pin configurations */
@@ -904,6 +903,16 @@
hda_nid_t *start_id);
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
+int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *conn_list, int max_conns);
+int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
+ const hda_nid_t **listp);
+int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
+ const hda_nid_t *list);
+int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
+ hda_nid_t nid, int recursive);
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+ u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
struct hda_verb {
hda_nid_t nid;
@@ -947,6 +956,17 @@
hda_nid_t nid, unsigned int cfg); /* for hwdep */
void snd_hda_shutup_pins(struct hda_codec *codec);
+/* SPDIF controls */
+struct hda_spdif_out {
+ hda_nid_t nid; /* Converter nid values relate to */
+ unsigned int status; /* IEC958 status bits */
+ unsigned short ctls; /* SPDIF control bits */
+};
+struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
+ hda_nid_t nid);
+void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx);
+void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid);
+
/*
* Mixer
*/
@@ -997,17 +1017,15 @@
int snd_hda_resume(struct hda_bus *bus);
#endif
-#ifdef CONFIG_SND_HDA_POWER_SAVE
static inline
int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
+#ifdef CONFIG_SND_HDA_POWER_SAVE
if (codec->patch_ops.check_power_status)
return codec->patch_ops.check_power_status(codec, nid);
+#endif
return 0;
}
-#else
-#define hda_call_check_power_status(codec, nid) 0
-#endif
/*
* get widget information
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index e3e8531..28ce17d 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -580,43 +580,45 @@
#endif /* CONFIG_PROC_FS */
/* update PCM info based on ELD */
-void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
- struct hda_pcm_stream *codec_pars)
+void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
+ struct hda_pcm_stream *hinfo)
{
+ u32 rates;
+ u64 formats;
+ unsigned int maxbps;
+ unsigned int channels_max;
int i;
/* assume basic audio support (the basic audio flag is not in ELD;
* however, all audio capable sinks are required to support basic
* audio) */
- pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
- pcm->formats = SNDRV_PCM_FMTBIT_S16_LE;
- pcm->maxbps = 16;
- pcm->channels_max = 2;
+ rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000;
+ formats = SNDRV_PCM_FMTBIT_S16_LE;
+ maxbps = 16;
+ channels_max = 2;
for (i = 0; i < eld->sad_count; i++) {
struct cea_sad *a = &eld->sad[i];
- pcm->rates |= a->rates;
- if (a->channels > pcm->channels_max)
- pcm->channels_max = a->channels;
+ rates |= a->rates;
+ if (a->channels > channels_max)
+ channels_max = a->channels;
if (a->format == AUDIO_CODING_TYPE_LPCM) {
if (a->sample_bits & AC_SUPPCM_BITS_20) {
- pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
- if (pcm->maxbps < 20)
- pcm->maxbps = 20;
+ formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ if (maxbps < 20)
+ maxbps = 20;
}
if (a->sample_bits & AC_SUPPCM_BITS_24) {
- pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
- if (pcm->maxbps < 24)
- pcm->maxbps = 24;
+ formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ if (maxbps < 24)
+ maxbps = 24;
}
}
}
- if (!codec_pars)
- return;
-
/* restrict the parameters by the values the codec provides */
- pcm->rates &= codec_pars->rates;
- pcm->formats &= codec_pars->formats;
- pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max);
- pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
+ hinfo->rates &= rates;
+ hinfo->formats &= formats;
+ hinfo->maxbps = min(hinfo->maxbps, maxbps);
+ hinfo->channels_max = min(hinfo->channels_max, channels_max);
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 486f6de..be69822 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -177,7 +177,8 @@
#define ICH6_REG_INTCTL 0x20
#define ICH6_REG_INTSTS 0x24
#define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */
-#define ICH6_REG_SYNC 0x34
+#define ICH6_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */
+#define ICH6_REG_SSYNC 0x38
#define ICH6_REG_CORBLBASE 0x40
#define ICH6_REG_CORBUBASE 0x44
#define ICH6_REG_CORBWP 0x48
@@ -479,6 +480,7 @@
#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */
#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */
#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */
+#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@@ -1706,13 +1708,16 @@
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned int bufsize, period_bytes, format_val, stream_tag;
int err;
+ struct hda_spdif_out *spdif =
+ snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid);
+ unsigned short ctls = spdif ? spdif->ctls : 0;
azx_stream_reset(chip, azx_dev);
format_val = snd_hda_calc_stream_format(runtime->rate,
runtime->channels,
runtime->format,
hinfo->maxbps,
- apcm->codec->spdif_ctls);
+ ctls);
if (!format_val) {
snd_printk(KERN_ERR SFX
"invalid format_val, rate=%d, ch=%d, format=%d\n",
@@ -1792,7 +1797,11 @@
spin_lock(&chip->reg_lock);
if (nsync > 1) {
/* first, set SYNC bits of corresponding streams */
- azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits);
+ if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
+ azx_writel(chip, OLD_SSYNC,
+ azx_readl(chip, OLD_SSYNC) | sbits);
+ else
+ azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits);
}
snd_pcm_group_for_each_entry(s, substream) {
if (s->pcm->card != substream->pcm->card)
@@ -1848,7 +1857,11 @@
if (nsync > 1) {
spin_lock(&chip->reg_lock);
/* reset SYNC bits */
- azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits);
+ if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
+ azx_writel(chip, OLD_SSYNC,
+ azx_readl(chip, OLD_SSYNC) & ~sbits);
+ else
+ azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits);
spin_unlock(&chip->reg_lock);
}
return 0;
@@ -1863,7 +1876,7 @@
unsigned int fifo_size;
link_pos = azx_sd_readl(azx_dev, SD_LPIB);
- if (azx_dev->index >= 4) {
+ if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* Playback, no problem using link position */
return link_pos;
}
@@ -1927,6 +1940,17 @@
default:
/* use the position buffer */
pos = le32_to_cpu(*azx_dev->posbuf);
+ if (chip->position_fix[stream] == POS_FIX_AUTO) {
+ if (!pos || pos == (u32)-1) {
+ printk(KERN_WARNING
+ "hda-intel: Invalid position buffer, "
+ "using LPIB read method instead.\n");
+ chip->position_fix[stream] = POS_FIX_LPIB;
+ pos = azx_sd_readl(azx_dev, SD_LPIB);
+ } else
+ chip->position_fix[stream] = POS_FIX_POSBUF;
+ }
+ break;
}
if (pos >= azx_dev->bufsize)
@@ -1964,16 +1988,6 @@
stream = azx_dev->substream->stream;
pos = azx_get_position(chip, azx_dev);
- if (chip->position_fix[stream] == POS_FIX_AUTO) {
- if (!pos) {
- printk(KERN_WARNING
- "hda-intel: Invalid position buffer, "
- "using LPIB read method instead.\n");
- chip->position_fix[stream] = POS_FIX_LPIB;
- pos = azx_get_position(chip, azx_dev);
- } else
- chip->position_fix[stream] = POS_FIX_POSBUF;
- }
if (WARN_ONCE(!azx_dev->period_bytes,
"hda-intel: zero azx_dev->period_bytes"))
@@ -2061,6 +2075,8 @@
}
}
+#define MAX_PREALLOC_SIZE (32 * 1024 * 1024)
+
static int
azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
struct hda_pcm *cpcm)
@@ -2069,6 +2085,7 @@
struct snd_pcm *pcm;
struct azx_pcm *apcm;
int pcm_dev = cpcm->device;
+ unsigned int size;
int s, err;
if (pcm_dev >= HDA_MAX_PCMS) {
@@ -2104,9 +2121,12 @@
snd_pcm_set_ops(pcm, s, &azx_pcm_ops);
}
/* buffer pre-allocation */
+ size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024;
+ if (size > MAX_PREALLOC_SIZE)
+ size = MAX_PREALLOC_SIZE;
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
snd_dma_pci_data(chip->pci),
- 1024 * 64, 32 * 1024 * 1024);
+ size, MAX_PREALLOC_SIZE);
return 0;
}
@@ -2149,7 +2169,7 @@
{
if (request_irq(chip->pci->irq, azx_interrupt,
chip->msi ? 0 : IRQF_SHARED,
- "hda_intel", chip)) {
+ KBUILD_MODNAME, chip)) {
printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
"disabling device\n", chip->pci->irq);
if (do_disconnect)
@@ -2347,28 +2367,20 @@
* white/black-listing for position_fix
*/
static struct snd_pci_quirk position_fix_list[] __devinitdata = {
- SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB),
SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
- SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
{}
};
@@ -2815,6 +2827,22 @@
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
+ { PCI_DEVICE(0x8086, 0x2668),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH6 */
+ { PCI_DEVICE(0x8086, 0x27d8),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH7 */
+ { PCI_DEVICE(0x8086, 0x269a),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ESB2 */
+ { PCI_DEVICE(0x8086, 0x284b),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH8 */
+ { PCI_DEVICE(0x8086, 0x293e),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
+ { PCI_DEVICE(0x8086, 0x293f),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
+ { PCI_DEVICE(0x8086, 0x3a3e),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
+ { PCI_DEVICE(0x8086, 0x3a6e),
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
/* Generic Intel */
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
.class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
@@ -2908,7 +2936,7 @@
/* pci_driver definition */
static struct pci_driver driver = {
- .name = "HDA Intel",
+ .name = KBUILD_MODNAME,
.id_table = azx_ids,
.probe = azx_probe,
.remove = __devexit_p(azx_remove),
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 08ec073..88b277e 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -212,7 +212,9 @@
/*
* SPDIF I/O
*/
-int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
+ hda_nid_t associated_nid,
+ hda_nid_t cvt_nid);
int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
/*
@@ -563,7 +565,6 @@
* power-management
*/
-#ifdef CONFIG_SND_HDA_POWER_SAVE
void snd_hda_schedule_power_save(struct hda_codec *codec);
struct hda_amp_list {
@@ -580,7 +581,6 @@
int snd_hda_check_amp_list_power(struct hda_codec *codec,
struct hda_loopback_check *check,
hda_nid_t nid);
-#endif /* CONFIG_SND_HDA_POWER_SAVE */
/*
* AMP control callbacks
@@ -639,8 +639,8 @@
int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
void snd_hdmi_show_eld(struct hdmi_eld *eld);
-void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
- struct hda_pcm_stream *codec_pars);
+void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
+ struct hda_pcm_stream *hinfo);
#ifdef CONFIG_PROC_FS
int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index bfe74c2..2be57b0 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -636,7 +636,7 @@
wid_caps |= AC_WCAP_CONN_LIST;
if (wid_caps & AC_WCAP_CONN_LIST)
- conn_len = snd_hda_get_connections(codec, nid, conn,
+ conn_len = snd_hda_get_raw_connections(codec, nid, conn,
HDA_MAX_CONNECTIONS);
if (wid_caps & AC_WCAP_IN_AMP) {
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d694e9d..1362c8b 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -213,7 +213,9 @@
return err;
}
if (spec->multiout.dig_out_nid) {
- err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid,
+ spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@@ -1920,7 +1922,8 @@
spec->mixers[0] = ad1981_hp_mixers;
spec->num_init_verbs = 2;
spec->init_verbs[1] = ad1981_hp_init_verbs;
- spec->multiout.dig_out_nid = 0;
+ if (!is_jack_available(codec, 0x0a))
+ spec->multiout.dig_out_nid = 0;
spec->input_mux = &ad1981_hp_capture_source;
codec->patch_ops.init = ad1981_hp_init;
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 61b9263..6b40684 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -240,7 +240,8 @@
}
if (spec->dig_out) {
- err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out);
+ err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+ spec->dig_out);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
new file mode 100644
index 0000000..d9a2254
--- /dev/null
+++ b/sound/pci/hda/patch_ca0132.c
@@ -0,0 +1,1097 @@
+/*
+ * HD audio interface patch for Creative CA0132 chip
+ *
+ * Copyright (c) 2011, Creative Technology Ltd.
+ *
+ * Based on patch_ca0110.c
+ * Copyright (c) 2008 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+#define WIDGET_CHIP_CTRL 0x15
+#define WIDGET_DSP_CTRL 0x16
+
+#define WUH_MEM_CONNID 10
+#define DSP_MEM_CONNID 16
+
+enum hda_cmd_vendor_io {
+ /* for DspIO node */
+ VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000,
+ VENDOR_DSPIO_SCP_WRITE_DATA_HIGH = 0x100,
+
+ VENDOR_DSPIO_STATUS = 0xF01,
+ VENDOR_DSPIO_SCP_POST_READ_DATA = 0x702,
+ VENDOR_DSPIO_SCP_READ_DATA = 0xF02,
+ VENDOR_DSPIO_DSP_INIT = 0x703,
+ VENDOR_DSPIO_SCP_POST_COUNT_QUERY = 0x704,
+ VENDOR_DSPIO_SCP_READ_COUNT = 0xF04,
+
+ /* for ChipIO node */
+ VENDOR_CHIPIO_ADDRESS_LOW = 0x000,
+ VENDOR_CHIPIO_ADDRESS_HIGH = 0x100,
+ VENDOR_CHIPIO_STREAM_FORMAT = 0x200,
+ VENDOR_CHIPIO_DATA_LOW = 0x300,
+ VENDOR_CHIPIO_DATA_HIGH = 0x400,
+
+ VENDOR_CHIPIO_GET_PARAMETER = 0xF00,
+ VENDOR_CHIPIO_STATUS = 0xF01,
+ VENDOR_CHIPIO_HIC_POST_READ = 0x702,
+ VENDOR_CHIPIO_HIC_READ_DATA = 0xF03,
+
+ VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE = 0x70A,
+
+ VENDOR_CHIPIO_PLL_PMU_WRITE = 0x70C,
+ VENDOR_CHIPIO_PLL_PMU_READ = 0xF0C,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW = 0x70D,
+ VENDOR_CHIPIO_8051_ADDRESS_HIGH = 0x70E,
+ VENDOR_CHIPIO_FLAG_SET = 0x70F,
+ VENDOR_CHIPIO_FLAGS_GET = 0xF0F,
+ VENDOR_CHIPIO_PARAMETER_SET = 0x710,
+ VENDOR_CHIPIO_PARAMETER_GET = 0xF10,
+
+ VENDOR_CHIPIO_PORT_ALLOC_CONFIG_SET = 0x711,
+ VENDOR_CHIPIO_PORT_ALLOC_SET = 0x712,
+ VENDOR_CHIPIO_PORT_ALLOC_GET = 0xF12,
+ VENDOR_CHIPIO_PORT_FREE_SET = 0x713,
+
+ VENDOR_CHIPIO_PARAMETER_EX_ID_GET = 0xF17,
+ VENDOR_CHIPIO_PARAMETER_EX_ID_SET = 0x717,
+ VENDOR_CHIPIO_PARAMETER_EX_VALUE_GET = 0xF18,
+ VENDOR_CHIPIO_PARAMETER_EX_VALUE_SET = 0x718
+};
+
+/*
+ * Control flag IDs
+ */
+enum control_flag_id {
+ /* Connection manager stream setup is bypassed/enabled */
+ CONTROL_FLAG_C_MGR = 0,
+ /* DSP DMA is bypassed/enabled */
+ CONTROL_FLAG_DMA = 1,
+ /* 8051 'idle' mode is disabled/enabled */
+ CONTROL_FLAG_IDLE_ENABLE = 2,
+ /* Tracker for the SPDIF-in path is bypassed/enabled */
+ CONTROL_FLAG_TRACKER = 3,
+ /* DigitalOut to Spdif2Out connection is disabled/enabled */
+ CONTROL_FLAG_SPDIF2OUT = 4,
+ /* Digital Microphone is disabled/enabled */
+ CONTROL_FLAG_DMIC = 5,
+ /* ADC_B rate is 48 kHz/96 kHz */
+ CONTROL_FLAG_ADC_B_96KHZ = 6,
+ /* ADC_C rate is 48 kHz/96 kHz */
+ CONTROL_FLAG_ADC_C_96KHZ = 7,
+ /* DAC rate is 48 kHz/96 kHz (affects all DACs) */
+ CONTROL_FLAG_DAC_96KHZ = 8,
+ /* DSP rate is 48 kHz/96 kHz */
+ CONTROL_FLAG_DSP_96KHZ = 9,
+ /* SRC clock is 98 MHz/196 MHz (196 MHz forces rate to 96 KHz) */
+ CONTROL_FLAG_SRC_CLOCK_196MHZ = 10,
+ /* SRC rate is 48 kHz/96 kHz (48 kHz disabled when clock is 196 MHz) */
+ CONTROL_FLAG_SRC_RATE_96KHZ = 11,
+ /* Decode Loop (DSP->SRC->DSP) is disabled/enabled */
+ CONTROL_FLAG_DECODE_LOOP = 12,
+ /* De-emphasis filter on DAC-1 disabled/enabled */
+ CONTROL_FLAG_DAC1_DEEMPHASIS = 13,
+ /* De-emphasis filter on DAC-2 disabled/enabled */
+ CONTROL_FLAG_DAC2_DEEMPHASIS = 14,
+ /* De-emphasis filter on DAC-3 disabled/enabled */
+ CONTROL_FLAG_DAC3_DEEMPHASIS = 15,
+ /* High-pass filter on ADC_B disabled/enabled */
+ CONTROL_FLAG_ADC_B_HIGH_PASS = 16,
+ /* High-pass filter on ADC_C disabled/enabled */
+ CONTROL_FLAG_ADC_C_HIGH_PASS = 17,
+ /* Common mode on Port_A disabled/enabled */
+ CONTROL_FLAG_PORT_A_COMMON_MODE = 18,
+ /* Common mode on Port_D disabled/enabled */
+ CONTROL_FLAG_PORT_D_COMMON_MODE = 19,
+ /* Impedance for ramp generator on Port_A 16 Ohm/10K Ohm */
+ CONTROL_FLAG_PORT_A_10KOHM_LOAD = 20,
+ /* Impedance for ramp generator on Port_D, 16 Ohm/10K Ohm */
+ CONTROL_FLAG_PORT_D_10K0HM_LOAD = 21,
+ /* ASI rate is 48kHz/96kHz */
+ CONTROL_FLAG_ASI_96KHZ = 22,
+ /* DAC power settings able to control attached ports no/yes */
+ CONTROL_FLAG_DACS_CONTROL_PORTS = 23,
+ /* Clock Stop OK reporting is disabled/enabled */
+ CONTROL_FLAG_CONTROL_STOP_OK_ENABLE = 24,
+ /* Number of control flags */
+ CONTROL_FLAGS_MAX = (CONTROL_FLAG_CONTROL_STOP_OK_ENABLE+1)
+};
+
+/*
+ * Control parameter IDs
+ */
+enum control_parameter_id {
+ /* 0: force HDA, 1: allow DSP if HDA Spdif1Out stream is idle */
+ CONTROL_PARAM_SPDIF1_SOURCE = 2,
+
+ /* Stream Control */
+
+ /* Select stream with the given ID */
+ CONTROL_PARAM_STREAM_ID = 24,
+ /* Source connection point for the selected stream */
+ CONTROL_PARAM_STREAM_SOURCE_CONN_POINT = 25,
+ /* Destination connection point for the selected stream */
+ CONTROL_PARAM_STREAM_DEST_CONN_POINT = 26,
+ /* Number of audio channels in the selected stream */
+ CONTROL_PARAM_STREAMS_CHANNELS = 27,
+ /*Enable control for the selected stream */
+ CONTROL_PARAM_STREAM_CONTROL = 28,
+
+ /* Connection Point Control */
+
+ /* Select connection point with the given ID */
+ CONTROL_PARAM_CONN_POINT_ID = 29,
+ /* Connection point sample rate */
+ CONTROL_PARAM_CONN_POINT_SAMPLE_RATE = 30,
+
+ /* Node Control */
+
+ /* Select HDA node with the given ID */
+ CONTROL_PARAM_NODE_ID = 31
+};
+
+/*
+ * Dsp Io Status codes
+ */
+enum hda_vendor_status_dspio {
+ /* Success */
+ VENDOR_STATUS_DSPIO_OK = 0x00,
+ /* Busy, unable to accept new command, the host must retry */
+ VENDOR_STATUS_DSPIO_BUSY = 0x01,
+ /* SCP command queue is full */
+ VENDOR_STATUS_DSPIO_SCP_COMMAND_QUEUE_FULL = 0x02,
+ /* SCP response queue is empty */
+ VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY = 0x03
+};
+
+/*
+ * Chip Io Status codes
+ */
+enum hda_vendor_status_chipio {
+ /* Success */
+ VENDOR_STATUS_CHIPIO_OK = 0x00,
+ /* Busy, unable to accept new command, the host must retry */
+ VENDOR_STATUS_CHIPIO_BUSY = 0x01
+};
+
+/*
+ * CA0132 sample rate
+ */
+enum ca0132_sample_rate {
+ SR_6_000 = 0x00,
+ SR_8_000 = 0x01,
+ SR_9_600 = 0x02,
+ SR_11_025 = 0x03,
+ SR_16_000 = 0x04,
+ SR_22_050 = 0x05,
+ SR_24_000 = 0x06,
+ SR_32_000 = 0x07,
+ SR_44_100 = 0x08,
+ SR_48_000 = 0x09,
+ SR_88_200 = 0x0A,
+ SR_96_000 = 0x0B,
+ SR_144_000 = 0x0C,
+ SR_176_400 = 0x0D,
+ SR_192_000 = 0x0E,
+ SR_384_000 = 0x0F,
+
+ SR_COUNT = 0x10,
+
+ SR_RATE_UNKNOWN = 0x1F
+};
+
+/*
+ * Scp Helper function
+ */
+enum get_set {
+ IS_SET = 0,
+ IS_GET = 1,
+};
+
+/*
+ * Duplicated from ca0110 codec
+ */
+
+static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
+{
+ if (pin) {
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
+ }
+ if (dac)
+ snd_hda_codec_write(codec, dac, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
+}
+
+static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
+{
+ if (pin) {
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_VREF80);
+ if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+ }
+ if (adc)
+ snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+}
+
+static char *dirstr[2] = { "Playback", "Capture" };
+
+static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+ int chan, int dir)
+{
+ char namestr[44];
+ int type = dir ? HDA_INPUT : HDA_OUTPUT;
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+ sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+ int chan, int dir)
+{
+ char namestr[44];
+ int type = dir ? HDA_INPUT : HDA_OUTPUT;
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+ sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0)
+#define add_out_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 0)
+#define add_in_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 1)
+#define add_in_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 1)
+#define add_mono_switch(codec, nid, pfx, chan) \
+ _add_switch(codec, nid, pfx, chan, 0)
+#define add_mono_volume(codec, nid, pfx, chan) \
+ _add_volume(codec, nid, pfx, chan, 0)
+#define add_in_mono_switch(codec, nid, pfx, chan) \
+ _add_switch(codec, nid, pfx, chan, 1)
+#define add_in_mono_volume(codec, nid, pfx, chan) \
+ _add_volume(codec, nid, pfx, chan, 1)
+
+
+/*
+ * CA0132 specific
+ */
+
+struct ca0132_spec {
+ struct auto_pin_cfg autocfg;
+ struct hda_multi_out multiout;
+ hda_nid_t out_pins[AUTO_CFG_MAX_OUTS];
+ hda_nid_t dacs[AUTO_CFG_MAX_OUTS];
+ hda_nid_t hp_dac;
+ hda_nid_t input_pins[AUTO_PIN_LAST];
+ hda_nid_t adcs[AUTO_PIN_LAST];
+ hda_nid_t dig_out;
+ hda_nid_t dig_in;
+ unsigned int num_inputs;
+ long curr_hp_switch;
+ long curr_hp_volume[2];
+ long curr_speaker_switch;
+ struct mutex chipio_mutex;
+ const char *input_labels[AUTO_PIN_LAST];
+ struct hda_pcm pcm_rec[2]; /* PCM information */
+};
+
+/* Chip access helper function */
+static int chipio_send(struct hda_codec *codec,
+ unsigned int reg,
+ unsigned int data)
+{
+ unsigned int res;
+ int retry = 50;
+
+ /* send bits of data specified by reg */
+ do {
+ res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
+ reg, data);
+ if (res == VENDOR_STATUS_CHIPIO_OK)
+ return 0;
+ } while (--retry);
+ return -EIO;
+}
+
+/*
+ * Write chip address through the vendor widget -- NOT protected by the Mutex!
+ */
+static int chipio_write_address(struct hda_codec *codec,
+ unsigned int chip_addx)
+{
+ int res;
+
+ /* send low 16 bits of the address */
+ res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_LOW,
+ chip_addx & 0xffff);
+
+ if (res != -EIO) {
+ /* send high 16 bits of the address */
+ res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_HIGH,
+ chip_addx >> 16);
+ }
+
+ return res;
+}
+
+/*
+ * Write data through the vendor widget -- NOT protected by the Mutex!
+ */
+
+static int chipio_write_data(struct hda_codec *codec, unsigned int data)
+{
+ int res;
+
+ /* send low 16 bits of the data */
+ res = chipio_send(codec, VENDOR_CHIPIO_DATA_LOW, data & 0xffff);
+
+ if (res != -EIO) {
+ /* send high 16 bits of the data */
+ res = chipio_send(codec, VENDOR_CHIPIO_DATA_HIGH,
+ data >> 16);
+ }
+
+ return res;
+}
+
+/*
+ * Read data through the vendor widget -- NOT protected by the Mutex!
+ */
+static int chipio_read_data(struct hda_codec *codec, unsigned int *data)
+{
+ int res;
+
+ /* post read */
+ res = chipio_send(codec, VENDOR_CHIPIO_HIC_POST_READ, 0);
+
+ if (res != -EIO) {
+ /* read status */
+ res = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0);
+ }
+
+ if (res != -EIO) {
+ /* read data */
+ *data = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_HIC_READ_DATA,
+ 0);
+ }
+
+ return res;
+}
+
+/*
+ * Write given value to the given address through the chip I/O widget.
+ * protected by the Mutex
+ */
+static int chipio_write(struct hda_codec *codec,
+ unsigned int chip_addx, const unsigned int data)
+{
+ struct ca0132_spec *spec = codec->spec;
+ int err;
+
+ mutex_lock(&spec->chipio_mutex);
+
+ /* write the address, and if successful proceed to write data */
+ err = chipio_write_address(codec, chip_addx);
+ if (err < 0)
+ goto exit;
+
+ err = chipio_write_data(codec, data);
+ if (err < 0)
+ goto exit;
+
+exit:
+ mutex_unlock(&spec->chipio_mutex);
+ return err;
+}
+
+/*
+ * Read the given address through the chip I/O widget
+ * protected by the Mutex
+ */
+static int chipio_read(struct hda_codec *codec,
+ unsigned int chip_addx, unsigned int *data)
+{
+ struct ca0132_spec *spec = codec->spec;
+ int err;
+
+ mutex_lock(&spec->chipio_mutex);
+
+ /* write the address, and if successful proceed to write data */
+ err = chipio_write_address(codec, chip_addx);
+ if (err < 0)
+ goto exit;
+
+ err = chipio_read_data(codec, data);
+ if (err < 0)
+ goto exit;
+
+exit:
+ mutex_unlock(&spec->chipio_mutex);
+ return err;
+}
+
+/*
+ * PCM stuffs
+ */
+static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid,
+ u32 stream_tag,
+ int channel_id, int format)
+{
+ unsigned int oldval, newval;
+
+ if (!nid)
+ return;
+
+ snd_printdd("ca0132_setup_stream: "
+ "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
+ nid, stream_tag, channel_id, format);
+
+ /* update the format-id if changed */
+ oldval = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_STREAM_FORMAT,
+ 0);
+ if (oldval != format) {
+ msleep(20);
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_STREAM_FORMAT,
+ format);
+ }
+
+ oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
+ newval = (stream_tag << 4) | channel_id;
+ if (oldval != newval) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CHANNEL_STREAMID,
+ newval);
+ }
+}
+
+static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
+{
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+}
+
+/*
+ * PCM callbacks
+ */
+static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
+
+ return 0;
+}
+
+static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_cleanup_stream(codec, spec->dacs[0]);
+
+ return 0;
+}
+
+/*
+ * Digital out
+ */
+static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format);
+
+ return 0;
+}
+
+static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_cleanup_stream(codec, spec->dig_out);
+
+ return 0;
+}
+
+/*
+ * Analog capture
+ */
+static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_setup_stream(codec, spec->adcs[substream->number],
+ stream_tag, 0, format);
+
+ return 0;
+}
+
+static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_cleanup_stream(codec, spec->adcs[substream->number]);
+
+ return 0;
+}
+
+/*
+ * Digital capture
+ */
+static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format);
+
+ return 0;
+}
+
+static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_cleanup_stream(codec, spec->dig_in);
+
+ return 0;
+}
+
+/*
+ */
+static struct hda_pcm_stream ca0132_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .prepare = ca0132_playback_pcm_prepare,
+ .cleanup = ca0132_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ca0132_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .prepare = ca0132_capture_pcm_prepare,
+ .cleanup = ca0132_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ca0132_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .prepare = ca0132_dig_playback_pcm_prepare,
+ .cleanup = ca0132_dig_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ca0132_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .prepare = ca0132_dig_capture_pcm_prepare,
+ .cleanup = ca0132_dig_capture_pcm_cleanup
+ },
+};
+
+static int ca0132_build_pcms(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->pcm_info = info;
+ codec->num_pcms = 0;
+
+ info->name = "CA0132 Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+ spec->multiout.max_channels;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
+ codec->num_pcms++;
+
+ if (!spec->dig_out && !spec->dig_in)
+ return 0;
+
+ info++;
+ info->name = "CA0132 Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->dig_out) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ ca0132_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
+ }
+ if (spec->dig_in) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ ca0132_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+ }
+ codec->num_pcms++;
+
+ return 0;
+}
+
+#define REG_CODEC_MUTE 0x18b014
+#define REG_CODEC_HP_VOL_L 0x18b070
+#define REG_CODEC_HP_VOL_R 0x18b074
+
+static int ca0132_hp_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ long *valp = ucontrol->value.integer.value;
+
+ *valp = spec->curr_hp_switch;
+ return 0;
+}
+
+static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int data;
+ int err;
+
+ /* any change? */
+ if (spec->curr_hp_switch == *valp)
+ return 0;
+
+ snd_hda_power_up(codec);
+
+ err = chipio_read(codec, REG_CODEC_MUTE, &data);
+ if (err < 0)
+ return err;
+
+ /* *valp 0 is mute, 1 is unmute */
+ data = (data & 0x7f) | (*valp ? 0 : 0x80);
+ chipio_write(codec, REG_CODEC_MUTE, data);
+ if (err < 0)
+ return err;
+
+ spec->curr_hp_switch = *valp;
+
+ snd_hda_power_down(codec);
+ return 1;
+}
+
+static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ long *valp = ucontrol->value.integer.value;
+
+ *valp = spec->curr_speaker_switch;
+ return 0;
+}
+
+static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int data;
+ int err;
+
+ /* any change? */
+ if (spec->curr_speaker_switch == *valp)
+ return 0;
+
+ snd_hda_power_up(codec);
+
+ err = chipio_read(codec, REG_CODEC_MUTE, &data);
+ if (err < 0)
+ return err;
+
+ /* *valp 0 is mute, 1 is unmute */
+ data = (data & 0xef) | (*valp ? 0 : 0x10);
+ chipio_write(codec, REG_CODEC_MUTE, data);
+ if (err < 0)
+ return err;
+
+ spec->curr_speaker_switch = *valp;
+
+ snd_hda_power_down(codec);
+ return 1;
+}
+
+static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ long *valp = ucontrol->value.integer.value;
+
+ *valp++ = spec->curr_hp_volume[0];
+ *valp = spec->curr_hp_volume[1];
+ return 0;
+}
+
+static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ca0132_spec *spec = codec->spec;
+ long *valp = ucontrol->value.integer.value;
+ long left_vol, right_vol;
+ unsigned int data;
+ int val;
+ int err;
+
+ left_vol = *valp++;
+ right_vol = *valp;
+
+ /* any change? */
+ if ((spec->curr_hp_volume[0] == left_vol) &&
+ (spec->curr_hp_volume[1] == right_vol))
+ return 0;
+
+ snd_hda_power_up(codec);
+
+ err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data);
+ if (err < 0)
+ return err;
+
+ val = 31 - left_vol;
+ data = (data & 0xe0) | val;
+ chipio_write(codec, REG_CODEC_HP_VOL_L, data);
+ if (err < 0)
+ return err;
+
+ val = 31 - right_vol;
+ data = (data & 0xe0) | val;
+ chipio_write(codec, REG_CODEC_HP_VOL_R, data);
+ if (err < 0)
+ return err;
+
+ spec->curr_hp_volume[0] = left_vol;
+ spec->curr_hp_volume[1] = right_vol;
+
+ snd_hda_power_down(codec);
+ return 1;
+}
+
+static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("Headphone Playback Switch",
+ nid, 1, 0, HDA_OUTPUT);
+ knew.get = ca0132_hp_switch_get;
+ knew.put = ca0132_hp_switch_put;
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static int add_hp_volume(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_VOLUME_MONO("Headphone Playback Volume",
+ nid, 3, 0, HDA_OUTPUT);
+ knew.get = ca0132_hp_volume_get;
+ knew.put = ca0132_hp_volume_put;
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static int add_speaker_switch(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch",
+ nid, 1, 0, HDA_OUTPUT);
+ knew.get = ca0132_speaker_switch_get;
+ knew.put = ca0132_speaker_switch_put;
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+static void ca0132_fix_hp_caps(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int caps;
+
+ /* set mute-capable, 1db step, 32 steps, ofs 6 */
+ caps = 0x80031f06;
+ snd_hda_override_amp_caps(codec, cfg->hp_pins[0], HDA_OUTPUT, caps);
+}
+
+static int ca0132_build_controls(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, err;
+
+ if (spec->multiout.num_dacs) {
+ err = add_speaker_switch(codec, spec->out_pins[0]);
+ if (err < 0)
+ return err;
+ }
+
+ if (cfg->hp_outs) {
+ ca0132_fix_hp_caps(codec);
+ err = add_hp_switch(codec, cfg->hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = add_hp_volume(codec, cfg->hp_pins[0]);
+ if (err < 0)
+ return err;
+ }
+
+ for (i = 0; i < spec->num_inputs; i++) {
+ const char *label = spec->input_labels[i];
+
+ err = add_in_switch(codec, spec->adcs[i], label);
+ if (err < 0)
+ return err;
+ err = add_in_volume(codec, spec->adcs[i], label);
+ if (err < 0)
+ return err;
+ if (cfg->inputs[i].type == AUTO_PIN_MIC) {
+ /* add Mic-Boost */
+ err = add_in_mono_volume(codec, spec->input_pins[i],
+ "Mic Boost", 1);
+ if (err < 0)
+ return err;
+ }
+ }
+
+ if (spec->dig_out) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
+ spec->dig_out);
+ if (err < 0)
+ return err;
+ err = add_out_volume(codec, spec->dig_out, "IEC958");
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->dig_in) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+ if (err < 0)
+ return err;
+ err = add_in_volume(codec, spec->dig_in, "IEC958");
+ }
+ return 0;
+}
+
+
+static void ca0132_set_ct_ext(struct hda_codec *codec, int enable)
+{
+ /* Set Creative extension */
+ snd_printdd("SET CREATIVE EXTENSION\n");
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE,
+ enable);
+ msleep(20);
+}
+
+
+static void ca0132_config(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+
+ /* line-outs */
+ cfg->line_outs = 1;
+ cfg->line_out_pins[0] = 0x0b; /* front */
+ cfg->line_out_type = AUTO_PIN_LINE_OUT;
+
+ spec->dacs[0] = 0x02;
+ spec->out_pins[0] = 0x0b;
+ spec->multiout.dac_nids = spec->dacs;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.max_channels = 2;
+
+ /* headphone */
+ cfg->hp_outs = 1;
+ cfg->hp_pins[0] = 0x0f;
+
+ spec->hp_dac = 0;
+ spec->multiout.hp_nid = 0;
+
+ /* inputs */
+ cfg->num_inputs = 2; /* Mic-in and line-in */
+ cfg->inputs[0].pin = 0x12;
+ cfg->inputs[0].type = AUTO_PIN_MIC;
+ cfg->inputs[1].pin = 0x11;
+ cfg->inputs[1].type = AUTO_PIN_LINE_IN;
+
+ /* Mic-in */
+ spec->input_pins[0] = 0x12;
+ spec->input_labels[0] = "Mic-In";
+ spec->adcs[0] = 0x07;
+
+ /* Line-In */
+ spec->input_pins[1] = 0x11;
+ spec->input_labels[1] = "Line-In";
+ spec->adcs[1] = 0x08;
+ spec->num_inputs = 2;
+}
+
+static void ca0132_init_chip(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ mutex_init(&spec->chipio_mutex);
+}
+
+static void ca0132_exit_chip(struct hda_codec *codec)
+{
+ /* put any chip cleanup stuffs here. */
+}
+
+static int ca0132_init(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ for (i = 0; i < spec->multiout.num_dacs; i++) {
+ init_output(codec, spec->out_pins[i],
+ spec->multiout.dac_nids[i]);
+ }
+ init_output(codec, cfg->hp_pins[0], spec->hp_dac);
+ init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
+
+ for (i = 0; i < spec->num_inputs; i++)
+ init_input(codec, spec->input_pins[i], spec->adcs[i]);
+
+ init_input(codec, cfg->dig_in_pin, spec->dig_in);
+
+ ca0132_set_ct_ext(codec, 1);
+
+ return 0;
+}
+
+
+static void ca0132_free(struct hda_codec *codec)
+{
+ ca0132_set_ct_ext(codec, 0);
+ ca0132_exit_chip(codec);
+ kfree(codec->spec);
+}
+
+static struct hda_codec_ops ca0132_patch_ops = {
+ .build_controls = ca0132_build_controls,
+ .build_pcms = ca0132_build_pcms,
+ .init = ca0132_init,
+ .free = ca0132_free,
+};
+
+
+
+static int patch_ca0132(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec;
+
+ snd_printdd("patch_ca0132\n");
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ ca0132_init_chip(codec);
+
+ ca0132_config(codec);
+
+ codec->patch_ops = ca0132_patch_ops;
+
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+static struct hda_codec_preset snd_hda_preset_ca0132[] = {
+ { .id = 0x11020011, .name = "CA0132", .patch = patch_ca0132 },
+ {} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:11020011");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Creative CA0132, CA0132 HD-audio codec");
+
+static struct hda_codec_preset_list ca0132_list = {
+ .preset = snd_hda_preset_ca0132,
+ .owner = THIS_MODULE,
+};
+
+static int __init patch_ca0132_init(void)
+{
+ return snd_hda_add_codec_preset(&ca0132_list);
+}
+
+static void __exit patch_ca0132_exit(void)
+{
+ snd_hda_delete_codec_preset(&ca0132_list);
+}
+
+module_init(patch_ca0132_init)
+module_exit(patch_ca0132_exit)
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 26a15210..7f93739 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -346,21 +346,15 @@
nid = codec->start_nid;
for (i = 0; i < codec->num_nodes; i++, nid++) {
- hda_nid_t pins[2];
unsigned int type;
- int j, nums;
+ int idx;
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
- nums = snd_hda_get_connections(codec, nid, pins,
- ARRAY_SIZE(pins));
- if (nums <= 0)
- continue;
- for (j = 0; j < nums; j++) {
- if (pins[j] == pin) {
- *idxp = j;
- return nid;
- }
+ idx = snd_hda_get_conn_index(codec, nid, pin, 0);
+ if (idx >= 0) {
+ *idxp = idx;
+ return nid;
}
}
return 0;
@@ -821,7 +815,8 @@
if (!spec->multiout.dig_out_nid)
return 0;
- err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid,
+ spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index ab3308d..cd2cf5e 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -327,7 +327,9 @@
return err;
}
if (spec->multiout.dig_out_nid) {
- err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid,
+ spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@@ -396,12 +398,11 @@
{
struct cmi_spec *spec = codec->spec;
hda_nid_t nid;
- int i, j, k, len;
+ int i, j, k;
/* clear the table, only one c-media dac assumed here */
memset(spec->multi_init, 0, sizeof(spec->multi_init));
for (j = 0, i = 0; i < cfg->line_outs; i++) {
- hda_nid_t conn[4];
nid = cfg->line_out_pins[i];
/* set as output */
spec->multi_init[j].nid = nid;
@@ -414,12 +415,10 @@
spec->multi_init[j].verb = AC_VERB_SET_CONNECT_SEL;
spec->multi_init[j].param = 0;
/* find the index in connect list */
- len = snd_hda_get_connections(codec, nid, conn, 4);
- for (k = 0; k < len; k++)
- if (conn[k] == spec->dac_nids[i]) {
- spec->multi_init[j].param = k;
- break;
- }
+ k = snd_hda_get_conn_index(codec, nid,
+ spec->dac_nids[i], 0);
+ if (k >= 0)
+ spec->multi_init[j].param = k;
j++;
}
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7bbc5f2..884f67b 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -155,6 +155,10 @@
unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
unsigned int beep_amp;
+
+ /* extra EAPD pins */
+ unsigned int num_eapds;
+ hda_nid_t eapds[4];
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -510,6 +514,7 @@
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
@@ -1123,10 +1128,8 @@
board_config = snd_hda_check_board_config(codec, CXT5045_MODELS,
cxt5045_models,
cxt5045_cfg_tbl);
-#if 0 /* use the old method just for safety */
if (board_config < 0)
- board_config = CXT5045_AUTO;
-#endif
+ board_config = CXT5045_AUTO; /* model=auto as default */
if (board_config == CXT5045_AUTO)
return patch_conexant_auto(codec);
@@ -1564,10 +1567,8 @@
board_config = snd_hda_check_board_config(codec, CXT5047_MODELS,
cxt5047_models,
cxt5047_cfg_tbl);
-#if 0 /* not enabled as default, as BIOS often broken for this codec */
if (board_config < 0)
- board_config = CXT5047_AUTO;
-#endif
+ board_config = CXT5047_AUTO; /* model=auto as default */
if (board_config == CXT5047_AUTO)
return patch_conexant_auto(codec);
@@ -1993,10 +1994,8 @@
board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
cxt5051_models,
cxt5051_cfg_tbl);
-#if 0 /* use the old method just for safety */
if (board_config < 0)
- board_config = CXT5051_AUTO;
-#endif
+ board_config = CXT5051_AUTO; /* model=auto as default */
if (board_config == CXT5051_AUTO)
return patch_conexant_auto(codec);
@@ -3114,10 +3113,8 @@
board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
cxt5066_models, cxt5066_cfg_tbl);
-#if 0 /* use the old method just for safety */
if (board_config < 0)
- board_config = CXT5066_AUTO;
-#endif
+ board_config = CXT5066_AUTO; /* model=auto as default */
if (board_config == CXT5066_AUTO)
return patch_conexant_auto(codec);
@@ -3308,19 +3305,8 @@
static const hda_nid_t cx_auto_adc_nids[] = { 0x14 };
-/* get the connection index of @nid in the widget @mux */
-static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
- hda_nid_t nid)
-{
- hda_nid_t conn[HDA_MAX_NUM_INPUTS];
- int i, nums;
-
- nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
- for (i = 0; i < nums; i++)
- if (conn[i] == nid)
- return i;
- return -1;
-}
+#define get_connection_index(codec, mux, nid)\
+ snd_hda_get_conn_index(codec, mux, nid, 0)
/* get an unassigned DAC from the given list.
* Return the nid if found and reduce the DAC list, or return zero if
@@ -3919,6 +3905,38 @@
#define cx_auto_parse_beep(codec)
#endif
+static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+ int i;
+ for (i = 0; i < nums; i++)
+ if (list[i] == nid)
+ return true;
+ return false;
+}
+
+/* parse extra-EAPD that aren't assigned to any pins */
+static void cx_auto_parse_eapd(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t nid, end_nid;
+
+ end_nid = codec->start_nid + codec->num_nodes;
+ for (nid = codec->start_nid; nid < end_nid; nid++) {
+ if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+ continue;
+ if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD))
+ continue;
+ if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) ||
+ found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) ||
+ found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs))
+ continue;
+ spec->eapds[spec->num_eapds++] = nid;
+ if (spec->num_eapds >= ARRAY_SIZE(spec->eapds))
+ break;
+ }
+}
+
static int cx_auto_parse_auto_config(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@@ -3932,6 +3950,7 @@
cx_auto_parse_input(codec);
cx_auto_parse_digital(codec);
cx_auto_parse_beep(codec);
+ cx_auto_parse_eapd(codec);
return 0;
}
@@ -4019,6 +4038,8 @@
}
}
cx_auto_update_speakers(codec);
+ /* turn on/off extra EAPDs, too */
+ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
}
static void cx_auto_init_input(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index bd0ae69..19cb72d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -43,7 +43,7 @@
/*
* The HDMI/DisplayPort configuration can be highly dynamic. A graphics device
- * could support two independent pipes, each of them can be connected to one or
+ * could support N independent pipes, each of them can be connected to one or
* more ports (DVI, HDMI or DisplayPort).
*
* The HDA correspondence of pipes/ports are converter/pin nodes.
@@ -51,30 +51,33 @@
#define MAX_HDMI_CVTS 4
#define MAX_HDMI_PINS 4
+struct hdmi_spec_per_cvt {
+ hda_nid_t cvt_nid;
+ int assigned;
+ unsigned int channels_min;
+ unsigned int channels_max;
+ u32 rates;
+ u64 formats;
+ unsigned int maxbps;
+};
+
+struct hdmi_spec_per_pin {
+ hda_nid_t pin_nid;
+ int num_mux_nids;
+ hda_nid_t mux_nids[HDA_MAX_CONNECTIONS];
+ struct hdmi_eld sink_eld;
+};
+
struct hdmi_spec {
int num_cvts;
+ struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS];
+
int num_pins;
- hda_nid_t cvt[MAX_HDMI_CVTS+1]; /* audio sources */
- hda_nid_t pin[MAX_HDMI_PINS+1]; /* audio sinks */
+ struct hdmi_spec_per_pin pins[MAX_HDMI_PINS];
+ struct hda_pcm pcm_rec[MAX_HDMI_PINS];
/*
- * source connection for each pin
- */
- hda_nid_t pin_cvt[MAX_HDMI_PINS+1];
-
- /*
- * HDMI sink attached to each pin
- */
- struct hdmi_eld sink_eld[MAX_HDMI_PINS];
-
- /*
- * export one pcm per pipe
- */
- struct hda_pcm pcm_rec[MAX_HDMI_CVTS];
- struct hda_pcm_stream codec_pcm_pars[MAX_HDMI_CVTS];
-
- /*
- * ati/nvhdmi specific
+ * Non-generic ATI/NVIDIA specific
*/
struct hda_multi_out multiout;
const struct hda_pcm_stream *pcm_playback;
@@ -284,15 +287,40 @@
* HDMI routines
*/
-static int hda_node_index(hda_nid_t *nids, hda_nid_t nid)
+static int pin_nid_to_pin_index(struct hdmi_spec *spec, hda_nid_t pin_nid)
{
- int i;
+ int pin_idx;
- for (i = 0; nids[i]; i++)
- if (nids[i] == nid)
- return i;
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++)
+ if (spec->pins[pin_idx].pin_nid == pin_nid)
+ return pin_idx;
- snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid);
+ snd_printk(KERN_WARNING "HDMI: pin nid %d not registered\n", pin_nid);
+ return -EINVAL;
+}
+
+static int hinfo_to_pin_index(struct hdmi_spec *spec,
+ struct hda_pcm_stream *hinfo)
+{
+ int pin_idx;
+
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++)
+ if (&spec->pcm_rec[pin_idx].stream[0] == hinfo)
+ return pin_idx;
+
+ snd_printk(KERN_WARNING "HDMI: hinfo %p not registered\n", hinfo);
+ return -EINVAL;
+}
+
+static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid)
+{
+ int cvt_idx;
+
+ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++)
+ if (spec->cvts[cvt_idx].cvt_nid == cvt_nid)
+ return cvt_idx;
+
+ snd_printk(KERN_WARNING "HDMI: cvt nid %d not registered\n", cvt_nid);
return -EINVAL;
}
@@ -326,28 +354,28 @@
snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val);
}
-static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid)
+static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid)
{
/* Unmute */
if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, pin_nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
- /* Enable pin out */
+ /* Disable pin out until stream is active*/
snd_hda_codec_write(codec, pin_nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
}
-static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid)
+static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid)
{
- return 1 + snd_hda_codec_read(codec, nid, 0,
+ return 1 + snd_hda_codec_read(codec, cvt_nid, 0,
AC_VERB_GET_CVT_CHAN_COUNT, 0);
}
static void hdmi_set_channel_count(struct hda_codec *codec,
- hda_nid_t nid, int chs)
+ hda_nid_t cvt_nid, int chs)
{
- if (chs != hdmi_get_channel_count(codec, nid))
- snd_hda_codec_write(codec, nid, 0,
+ if (chs != hdmi_get_channel_count(codec, cvt_nid))
+ snd_hda_codec_write(codec, cvt_nid, 0,
AC_VERB_SET_CVT_CHAN_COUNT, chs - 1);
}
@@ -384,11 +412,8 @@
*
* TODO: it could select the wrong CA from multiple candidates.
*/
-static int hdmi_channel_allocation(struct hda_codec *codec, hda_nid_t nid,
- int channels)
+static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels)
{
- struct hdmi_spec *spec = codec->spec;
- struct hdmi_eld *eld;
int i;
int ca = 0;
int spk_mask = 0;
@@ -400,19 +425,6 @@
if (channels <= 2)
return 0;
- i = hda_node_index(spec->pin_cvt, nid);
- if (i < 0)
- return 0;
- eld = &spec->sink_eld[i];
-
- /*
- * HDMI sink's ELD info cannot always be retrieved for now, e.g.
- * in console or for audio devices. Assume the highest speakers
- * configuration, to _not_ prohibit multi-channel audio playback.
- */
- if (!eld->spk_alloc)
- eld->spk_alloc = 0xffff;
-
/*
* expand ELD's speaker allocation mask
*
@@ -608,67 +620,63 @@
return true;
}
-static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
+static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx,
struct snd_pcm_substream *substream)
{
struct hdmi_spec *spec = codec->spec;
- hda_nid_t pin_nid;
+ struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ hda_nid_t pin_nid = per_pin->pin_nid;
int channels = substream->runtime->channels;
+ struct hdmi_eld *eld;
int ca;
- int i;
union audio_infoframe ai;
- ca = hdmi_channel_allocation(codec, nid, channels);
+ eld = &spec->pins[pin_idx].sink_eld;
+ if (!eld->monitor_present)
+ return;
- for (i = 0; i < spec->num_pins; i++) {
- if (spec->pin_cvt[i] != nid)
- continue;
- if (!spec->sink_eld[i].monitor_present)
- continue;
+ ca = hdmi_channel_allocation(eld, channels);
- pin_nid = spec->pin[i];
+ memset(&ai, 0, sizeof(ai));
+ if (eld->conn_type == 0) { /* HDMI */
+ struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi;
- memset(&ai, 0, sizeof(ai));
- if (spec->sink_eld[i].conn_type == 0) { /* HDMI */
- struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi;
+ hdmi_ai->type = 0x84;
+ hdmi_ai->ver = 0x01;
+ hdmi_ai->len = 0x0a;
+ hdmi_ai->CC02_CT47 = channels - 1;
+ hdmi_ai->CA = ca;
+ hdmi_checksum_audio_infoframe(hdmi_ai);
+ } else if (eld->conn_type == 1) { /* DisplayPort */
+ struct dp_audio_infoframe *dp_ai = &ai.dp;
- hdmi_ai->type = 0x84;
- hdmi_ai->ver = 0x01;
- hdmi_ai->len = 0x0a;
- hdmi_ai->CC02_CT47 = channels - 1;
- hdmi_ai->CA = ca;
- hdmi_checksum_audio_infoframe(hdmi_ai);
- } else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */
- struct dp_audio_infoframe *dp_ai = &ai.dp;
+ dp_ai->type = 0x84;
+ dp_ai->len = 0x1b;
+ dp_ai->ver = 0x11 << 2;
+ dp_ai->CC02_CT47 = channels - 1;
+ dp_ai->CA = ca;
+ } else {
+ snd_printd("HDMI: unknown connection type at pin %d\n",
+ pin_nid);
+ return;
+ }
- dp_ai->type = 0x84;
- dp_ai->len = 0x1b;
- dp_ai->ver = 0x11 << 2;
- dp_ai->CC02_CT47 = channels - 1;
- dp_ai->CA = ca;
- } else {
- snd_printd("HDMI: unknown connection type at pin %d\n",
- pin_nid);
- continue;
- }
-
- /*
- * sizeof(ai) is used instead of sizeof(*hdmi_ai) or
- * sizeof(*dp_ai) to avoid partial match/update problems when
- * the user switches between HDMI/DP monitors.
- */
- if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes,
- sizeof(ai))) {
- snd_printdd("hdmi_setup_audio_infoframe: "
- "cvt=%d pin=%d channels=%d\n",
- nid, pin_nid,
- channels);
- hdmi_setup_channel_mapping(codec, pin_nid, ca);
- hdmi_stop_infoframe_trans(codec, pin_nid);
- hdmi_fill_audio_infoframe(codec, pin_nid,
- ai.bytes, sizeof(ai));
- hdmi_start_infoframe_trans(codec, pin_nid);
- }
+ /*
+ * sizeof(ai) is used instead of sizeof(*hdmi_ai) or
+ * sizeof(*dp_ai) to avoid partial match/update problems when
+ * the user switches between HDMI/DP monitors.
+ */
+ if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes,
+ sizeof(ai))) {
+ snd_printdd("hdmi_setup_audio_infoframe: "
+ "pin=%d channels=%d\n",
+ pin_nid,
+ channels);
+ hdmi_setup_channel_mapping(codec, pin_nid, ca);
+ hdmi_stop_infoframe_trans(codec, pin_nid);
+ hdmi_fill_audio_infoframe(codec, pin_nid,
+ ai.bytes, sizeof(ai));
+ hdmi_start_infoframe_trans(codec, pin_nid);
}
}
@@ -686,17 +694,27 @@
int pin_nid = res >> AC_UNSOL_RES_TAG_SHIFT;
int pd = !!(res & AC_UNSOL_RES_PD);
int eldv = !!(res & AC_UNSOL_RES_ELDV);
- int index;
+ int pin_idx;
+ struct hdmi_eld *eld;
printk(KERN_INFO
- "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
- pin_nid, pd, eldv);
+ "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
+ codec->addr, pin_nid, pd, eldv);
- index = hda_node_index(spec->pin, pin_nid);
- if (index < 0)
+ pin_idx = pin_nid_to_pin_index(spec, pin_nid);
+ if (pin_idx < 0)
return;
+ eld = &spec->pins[pin_idx].sink_eld;
- hdmi_present_sense(codec, pin_nid, &spec->sink_eld[index]);
+ hdmi_present_sense(codec, pin_nid, eld);
+
+ /*
+ * HDMI sink's ELD info cannot always be retrieved for now, e.g.
+ * in console or for audio devices. Assume the highest speakers
+ * configuration, to _not_ prohibit multi-channel audio playback.
+ */
+ if (!eld->spk_alloc)
+ eld->spk_alloc = 0xffff;
}
static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
@@ -707,7 +725,8 @@
int cp_ready = !!(res & AC_UNSOL_RES_CP_READY);
printk(KERN_INFO
- "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
+ "HDMI CP event: CODEC=%d PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
+ codec->addr,
tag,
subtag,
cp_state,
@@ -727,7 +746,7 @@
int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
- if (hda_node_index(spec->pin, tag) < 0) {
+ if (pin_nid_to_pin_index(spec, tag) < 0) {
snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag);
return;
}
@@ -746,21 +765,14 @@
#define is_hbr_format(format) \
((format & AC_FMT_TYPE_NON_PCM) && (format & AC_FMT_CHAN_MASK) == 7)
-static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
- u32 stream_tag, int format)
+static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
+ hda_nid_t pin_nid, u32 stream_tag, int format)
{
- struct hdmi_spec *spec = codec->spec;
int pinctl;
int new_pinctl = 0;
- int i;
- for (i = 0; i < spec->num_pins; i++) {
- if (spec->pin_cvt[i] != nid)
- continue;
- if (!(snd_hda_query_pin_caps(codec, spec->pin[i]) & AC_PINCAP_HBR))
- continue;
-
- pinctl = snd_hda_codec_read(codec, spec->pin[i], 0,
+ if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) {
+ pinctl = snd_hda_codec_read(codec, pin_nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
new_pinctl = pinctl & ~AC_PINCTL_EPT;
@@ -771,22 +783,22 @@
snd_printdd("hdmi_setup_stream: "
"NID=0x%x, %spinctl=0x%x\n",
- spec->pin[i],
+ pin_nid,
pinctl == new_pinctl ? "" : "new-",
new_pinctl);
if (pinctl != new_pinctl)
- snd_hda_codec_write(codec, spec->pin[i], 0,
+ snd_hda_codec_write(codec, pin_nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
new_pinctl);
- }
+ }
if (is_hbr_format(format) && !new_pinctl) {
snd_printdd("hdmi_setup_stream: HBR is not supported\n");
return -EINVAL;
}
- snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+ snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format);
return 0;
}
@@ -798,37 +810,70 @@
struct snd_pcm_substream *substream)
{
struct hdmi_spec *spec = codec->spec;
- struct hdmi_eld *eld;
- struct hda_pcm_stream *codec_pars;
struct snd_pcm_runtime *runtime = substream->runtime;
- unsigned int idx;
+ int pin_idx, cvt_idx, mux_idx = 0;
+ struct hdmi_spec_per_pin *per_pin;
+ struct hdmi_eld *eld;
+ struct hdmi_spec_per_cvt *per_cvt = NULL;
+ int pinctl;
- for (idx = 0; idx < spec->num_cvts; idx++)
- if (hinfo->nid == spec->cvt[idx])
- break;
- if (snd_BUG_ON(idx >= spec->num_cvts) ||
- snd_BUG_ON(idx >= spec->num_pins))
+ /* Validate hinfo */
+ pin_idx = hinfo_to_pin_index(spec, hinfo);
+ if (snd_BUG_ON(pin_idx < 0))
return -EINVAL;
+ per_pin = &spec->pins[pin_idx];
+ eld = &per_pin->sink_eld;
- /* save the PCM info the codec provides */
- codec_pars = &spec->codec_pcm_pars[idx];
- if (!codec_pars->rates)
- *codec_pars = *hinfo;
+ /* Dynamically assign converter to stream */
+ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) {
+ per_cvt = &spec->cvts[cvt_idx];
- eld = &spec->sink_eld[idx];
- if (!static_hdmi_pcm && eld->eld_valid && eld->sad_count > 0) {
- hdmi_eld_update_pcm_info(eld, hinfo, codec_pars);
+ /* Must not already be assigned */
+ if (per_cvt->assigned)
+ continue;
+ /* Must be in pin's mux's list of converters */
+ for (mux_idx = 0; mux_idx < per_pin->num_mux_nids; mux_idx++)
+ if (per_pin->mux_nids[mux_idx] == per_cvt->cvt_nid)
+ break;
+ /* Not in mux list */
+ if (mux_idx == per_pin->num_mux_nids)
+ continue;
+ break;
+ }
+ /* No free converters */
+ if (cvt_idx == spec->num_cvts)
+ return -ENODEV;
+
+ /* Claim converter */
+ per_cvt->assigned = 1;
+ hinfo->nid = per_cvt->cvt_nid;
+
+ snd_hda_codec_write(codec, per_pin->pin_nid, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ mux_idx);
+ pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_codec_write(codec, per_pin->pin_nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinctl | PIN_OUT);
+ snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid);
+
+ /* Initially set the converter's capabilities */
+ hinfo->channels_min = per_cvt->channels_min;
+ hinfo->channels_max = per_cvt->channels_max;
+ hinfo->rates = per_cvt->rates;
+ hinfo->formats = per_cvt->formats;
+ hinfo->maxbps = per_cvt->maxbps;
+
+ /* Restrict capabilities by ELD if this isn't disabled */
+ if (!static_hdmi_pcm && eld->eld_valid) {
+ snd_hdmi_eld_update_pcm_info(eld, hinfo);
if (hinfo->channels_min > hinfo->channels_max ||
!hinfo->rates || !hinfo->formats)
return -ENODEV;
- } else {
- /* fallback to the codec default */
- hinfo->channels_max = codec_pars->channels_max;
- hinfo->rates = codec_pars->rates;
- hinfo->formats = codec_pars->formats;
- hinfo->maxbps = codec_pars->maxbps;
}
- /* store the updated parameters */
+
+ /* Store the updated parameters */
runtime->hw.channels_min = hinfo->channels_min;
runtime->hw.channels_max = hinfo->channels_max;
runtime->hw.formats = hinfo->formats;
@@ -842,12 +887,11 @@
/*
* HDA/HDMI auto parsing
*/
-static int hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid)
+static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx)
{
struct hdmi_spec *spec = codec->spec;
- hda_nid_t conn_list[HDA_MAX_CONNECTIONS];
- int conn_len, curr;
- int index;
+ struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ hda_nid_t pin_nid = per_pin->pin_nid;
if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) {
snd_printk(KERN_WARNING
@@ -857,19 +901,9 @@
return -EINVAL;
}
- conn_len = snd_hda_get_connections(codec, pin_nid, conn_list,
- HDA_MAX_CONNECTIONS);
- if (conn_len > 1)
- curr = snd_hda_codec_read(codec, pin_nid, 0,
- AC_VERB_GET_CONNECT_SEL, 0);
- else
- curr = 0;
-
- index = hda_node_index(spec->pin, pin_nid);
- if (index < 0)
- return -EINVAL;
-
- spec->pin_cvt[index] = conn_list[curr];
+ per_pin->num_mux_nids = snd_hda_get_connections(codec, pin_nid,
+ per_pin->mux_nids,
+ HDA_MAX_CONNECTIONS);
return 0;
}
@@ -896,8 +930,8 @@
eld->eld_valid = 0;
printk(KERN_INFO
- "HDMI status: Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
- pin_nid, eld->monitor_present, eld->eld_valid);
+ "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
+ codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
if (eld->eld_valid)
if (!snd_hdmi_get_eld(eld, codec, pin_nid))
@@ -909,47 +943,75 @@
static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
{
struct hdmi_spec *spec = codec->spec;
+ unsigned int caps, config;
+ int pin_idx;
+ struct hdmi_spec_per_pin *per_pin;
+ struct hdmi_eld *eld;
int err;
- if (spec->num_pins >= MAX_HDMI_PINS) {
- snd_printk(KERN_WARNING
- "HDMI: no space for pin %d\n", pin_nid);
+ caps = snd_hda_param_read(codec, pin_nid, AC_PAR_PIN_CAP);
+ if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP)))
+ return 0;
+
+ config = snd_hda_codec_read(codec, pin_nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
+ return 0;
+
+ if (snd_BUG_ON(spec->num_pins >= MAX_HDMI_PINS))
return -E2BIG;
- }
+
+ pin_idx = spec->num_pins;
+ per_pin = &spec->pins[pin_idx];
+ eld = &per_pin->sink_eld;
+
+ per_pin->pin_nid = pin_nid;
err = snd_hda_input_jack_add(codec, pin_nid,
SND_JACK_VIDEOOUT, NULL);
if (err < 0)
return err;
- hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]);
+ err = hdmi_read_pin_conn(codec, pin_idx);
+ if (err < 0)
+ return err;
- spec->pin[spec->num_pins] = pin_nid;
spec->num_pins++;
- return hdmi_read_pin_conn(codec, pin_nid);
+ hdmi_present_sense(codec, pin_nid, eld);
+
+ return 0;
}
-static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid)
+static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
{
- int i, found_pin = 0;
struct hdmi_spec *spec = codec->spec;
-
- for (i = 0; i < spec->num_pins; i++)
- if (nid == spec->pin_cvt[i]) {
- found_pin = 1;
- break;
- }
-
- if (!found_pin) {
- snd_printdd("HDMI: Skipping node %d (no connection)\n", nid);
- return -EINVAL;
- }
+ int cvt_idx;
+ struct hdmi_spec_per_cvt *per_cvt;
+ unsigned int chans;
+ int err;
if (snd_BUG_ON(spec->num_cvts >= MAX_HDMI_CVTS))
return -E2BIG;
- spec->cvt[spec->num_cvts] = nid;
+ chans = get_wcaps(codec, cvt_nid);
+ chans = get_wcaps_channels(chans);
+
+ cvt_idx = spec->num_cvts;
+ per_cvt = &spec->cvts[cvt_idx];
+
+ per_cvt->cvt_nid = cvt_nid;
+ per_cvt->channels_min = 2;
+ if (chans <= 16)
+ per_cvt->channels_max = chans;
+
+ err = snd_hda_query_supported_pcm(codec, cvt_nid,
+ &per_cvt->rates,
+ &per_cvt->formats,
+ &per_cvt->maxbps);
+ if (err < 0)
+ return err;
+
spec->num_cvts++;
return 0;
@@ -959,8 +1021,6 @@
{
hda_nid_t nid;
int i, nodes;
- int num_tmp_cvts = 0;
- hda_nid_t tmp_cvt[MAX_HDMI_CVTS];
nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
if (!nid || nodes < 0) {
@@ -971,7 +1031,6 @@
for (i = 0; i < nodes; i++, nid++) {
unsigned int caps;
unsigned int type;
- unsigned int config;
caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP);
type = get_wcaps_type(caps);
@@ -981,32 +1040,14 @@
switch (type) {
case AC_WID_AUD_OUT:
- if (num_tmp_cvts >= MAX_HDMI_CVTS) {
- snd_printk(KERN_WARNING
- "HDMI: no space for converter %d\n", nid);
- continue;
- }
- tmp_cvt[num_tmp_cvts] = nid;
- num_tmp_cvts++;
+ hdmi_add_cvt(codec, nid);
break;
case AC_WID_PIN:
- caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
- if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP)))
- continue;
-
- config = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_CONFIG_DEFAULT, 0);
- if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
- continue;
-
hdmi_add_pin(codec, nid);
break;
}
}
- for (i = 0; i < num_tmp_cvts; i++)
- hdmi_add_cvt(codec, tmp_cvt[i]);
-
/*
* G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event
* can be lost and presence sense verb will become inaccurate if the
@@ -1023,7 +1064,7 @@
/*
*/
-static char *generic_hdmi_pcm_names[MAX_HDMI_CVTS] = {
+static char *generic_hdmi_pcm_names[MAX_HDMI_PINS] = {
"HDMI 0",
"HDMI 1",
"HDMI 2",
@@ -1040,51 +1081,84 @@
unsigned int format,
struct snd_pcm_substream *substream)
{
- hdmi_set_channel_count(codec, hinfo->nid,
- substream->runtime->channels);
+ hda_nid_t cvt_nid = hinfo->nid;
+ struct hdmi_spec *spec = codec->spec;
+ int pin_idx = hinfo_to_pin_index(spec, hinfo);
+ hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid;
- hdmi_setup_audio_infoframe(codec, hinfo->nid, substream);
+ hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels);
- return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
+ hdmi_setup_audio_infoframe(codec, pin_idx, substream);
+
+ return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
}
-static const struct hda_pcm_stream generic_hdmi_pcm_playback = {
- .substreams = 1,
- .channels_min = 2,
- .ops = {
- .open = hdmi_pcm_open,
- .prepare = generic_hdmi_playback_pcm_prepare,
- },
+static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct hdmi_spec *spec = codec->spec;
+ int cvt_idx, pin_idx;
+ struct hdmi_spec_per_cvt *per_cvt;
+ struct hdmi_spec_per_pin *per_pin;
+ int pinctl;
+
+ snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+
+ if (hinfo->nid) {
+ cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid);
+ if (snd_BUG_ON(cvt_idx < 0))
+ return -EINVAL;
+ per_cvt = &spec->cvts[cvt_idx];
+
+ snd_BUG_ON(!per_cvt->assigned);
+ per_cvt->assigned = 0;
+ hinfo->nid = 0;
+
+ pin_idx = hinfo_to_pin_index(spec, hinfo);
+ if (snd_BUG_ON(pin_idx < 0))
+ return -EINVAL;
+ per_pin = &spec->pins[pin_idx];
+
+ pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_codec_write(codec, per_pin->pin_nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinctl & ~PIN_OUT);
+ snd_hda_spdif_ctls_unassign(codec, pin_idx);
+ }
+
+ return 0;
+}
+
+static const struct hda_pcm_ops generic_ops = {
+ .open = hdmi_pcm_open,
+ .prepare = generic_hdmi_playback_pcm_prepare,
+ .cleanup = generic_hdmi_playback_pcm_cleanup,
};
static int generic_hdmi_build_pcms(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
- struct hda_pcm *info = spec->pcm_rec;
- int i;
+ int pin_idx;
- codec->num_pcms = spec->num_cvts;
- codec->pcm_info = info;
-
- for (i = 0; i < codec->num_pcms; i++, info++) {
- unsigned int chans;
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+ struct hda_pcm *info;
struct hda_pcm_stream *pstr;
- chans = get_wcaps(codec, spec->cvt[i]);
- chans = get_wcaps_channels(chans);
-
- info->name = generic_hdmi_pcm_names[i];
+ info = &spec->pcm_rec[pin_idx];
+ info->name = generic_hdmi_pcm_names[pin_idx];
info->pcm_type = HDA_PCM_TYPE_HDMI;
+
pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
- if (spec->pcm_playback)
- *pstr = *spec->pcm_playback;
- else
- *pstr = generic_hdmi_pcm_playback;
- pstr->nid = spec->cvt[i];
- if (pstr->channels_max <= 2 && chans && chans <= 16)
- pstr->channels_max = chans;
+ pstr->substreams = 1;
+ pstr->ops = generic_ops;
+ /* other pstr fields are set in open */
}
+ codec->num_pcms = spec->num_pins;
+ codec->pcm_info = spec->pcm_rec;
+
return 0;
}
@@ -1092,12 +1166,16 @@
{
struct hdmi_spec *spec = codec->spec;
int err;
- int i;
+ int pin_idx;
- for (i = 0; i < codec->num_pcms; i++) {
- err = snd_hda_create_spdif_out_ctls(codec, spec->cvt[i]);
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+ struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ err = snd_hda_create_spdif_out_ctls(codec,
+ per_pin->pin_nid,
+ per_pin->mux_nids[0]);
if (err < 0)
return err;
+ snd_hda_spdif_ctls_unassign(codec, pin_idx);
}
return 0;
@@ -1106,13 +1184,19 @@
static int generic_hdmi_init(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
- int i;
+ int pin_idx;
- for (i = 0; spec->pin[i]; i++) {
- hdmi_enable_output(codec, spec->pin[i]);
- snd_hda_codec_write(codec, spec->pin[i], 0,
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+ struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ hda_nid_t pin_nid = per_pin->pin_nid;
+ struct hdmi_eld *eld = &per_pin->sink_eld;
+
+ hdmi_init_pin(codec, pin_nid);
+ snd_hda_codec_write(codec, pin_nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | spec->pin[i]);
+ AC_USRSP_EN | pin_nid);
+
+ snd_hda_eld_proc_new(codec, eld, pin_idx);
}
return 0;
}
@@ -1120,10 +1204,14 @@
static void generic_hdmi_free(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
- int i;
+ int pin_idx;
- for (i = 0; i < spec->num_pins; i++)
- snd_hda_eld_proc_free(codec, &spec->sink_eld[i]);
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+ struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+ struct hdmi_eld *eld = &per_pin->sink_eld;
+
+ snd_hda_eld_proc_free(codec, eld);
+ }
snd_hda_input_jack_free(codec);
kfree(spec);
@@ -1140,7 +1228,6 @@
static int patch_generic_hdmi(struct hda_codec *codec)
{
struct hdmi_spec *spec;
- int i;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -1154,15 +1241,69 @@
}
codec->patch_ops = generic_hdmi_patch_ops;
- for (i = 0; i < spec->num_pins; i++)
- snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i);
-
init_channel_allocations();
return 0;
}
/*
+ * Shared non-generic implementations
+ */
+
+static int simple_playback_build_pcms(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+ int i;
+
+ codec->num_pcms = spec->num_cvts;
+ codec->pcm_info = info;
+
+ for (i = 0; i < codec->num_pcms; i++, info++) {
+ unsigned int chans;
+ struct hda_pcm_stream *pstr;
+
+ chans = get_wcaps(codec, spec->cvts[i].cvt_nid);
+ chans = get_wcaps_channels(chans);
+
+ info->name = generic_hdmi_pcm_names[i];
+ info->pcm_type = HDA_PCM_TYPE_HDMI;
+ pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
+ snd_BUG_ON(!spec->pcm_playback);
+ *pstr = *spec->pcm_playback;
+ pstr->nid = spec->cvts[i].cvt_nid;
+ if (pstr->channels_max <= 2 && chans && chans <= 16)
+ pstr->channels_max = chans;
+ }
+
+ return 0;
+}
+
+static int simple_playback_build_controls(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+ int err;
+ int i;
+
+ for (i = 0; i < codec->num_pcms; i++) {
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->cvts[i].cvt_nid,
+ spec->cvts[i].cvt_nid);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static void simple_playback_free(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+
+ kfree(spec);
+}
+
+/*
* Nvidia specific implementations
*/
@@ -1352,6 +1493,9 @@
int chs;
unsigned int dataDCC1, dataDCC2, channel_id;
int i;
+ struct hdmi_spec *spec = codec->spec;
+ struct hda_spdif_out *spdif =
+ snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
mutex_lock(&codec->spdif_mutex);
@@ -1361,12 +1505,12 @@
dataDCC2 = 0x2;
/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
- if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+ if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
snd_hda_codec_write(codec,
nvhdmi_master_con_nid_7x,
0,
AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
+ spdif->ctls & ~AC_DIG1_ENABLE & 0xff);
/* set the stream id */
snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0,
@@ -1378,12 +1522,12 @@
/* turn on again (if needed) */
/* enable and set the channel status audio/data flag */
- if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) {
+ if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE)) {
snd_hda_codec_write(codec,
nvhdmi_master_con_nid_7x,
0,
AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls & 0xff);
+ spdif->ctls & 0xff);
snd_hda_codec_write(codec,
nvhdmi_master_con_nid_7x,
0,
@@ -1400,12 +1544,12 @@
*otherwise the IEC958 bits won't be updated
*/
if (codec->spdif_status_reset &&
- (codec->spdif_ctls & AC_DIG1_ENABLE))
+ (spdif->ctls & AC_DIG1_ENABLE))
snd_hda_codec_write(codec,
nvhdmi_con_nids_7x[i],
0,
AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
+ spdif->ctls & ~AC_DIG1_ENABLE & 0xff);
/* set the stream id */
snd_hda_codec_write(codec,
nvhdmi_con_nids_7x[i],
@@ -1421,12 +1565,12 @@
/* turn on again (if needed) */
/* enable and set the channel status audio/data flag */
if (codec->spdif_status_reset &&
- (codec->spdif_ctls & AC_DIG1_ENABLE)) {
+ (spdif->ctls & AC_DIG1_ENABLE)) {
snd_hda_codec_write(codec,
nvhdmi_con_nids_7x[i],
0,
AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls & 0xff);
+ spdif->ctls & 0xff);
snd_hda_codec_write(codec,
nvhdmi_con_nids_7x[i],
0,
@@ -1471,17 +1615,17 @@
};
static const struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = {
- .build_controls = generic_hdmi_build_controls,
- .build_pcms = generic_hdmi_build_pcms,
+ .build_controls = simple_playback_build_controls,
+ .build_pcms = simple_playback_build_pcms,
.init = nvhdmi_7x_init,
- .free = generic_hdmi_free,
+ .free = simple_playback_free,
};
static const struct hda_codec_ops nvhdmi_patch_ops_2ch = {
- .build_controls = generic_hdmi_build_controls,
- .build_pcms = generic_hdmi_build_pcms,
+ .build_controls = simple_playback_build_controls,
+ .build_pcms = simple_playback_build_pcms,
.init = nvhdmi_7x_init,
- .free = generic_hdmi_free,
+ .free = simple_playback_free,
};
static int patch_nvhdmi_2ch(struct hda_codec *codec)
@@ -1498,7 +1642,7 @@
spec->multiout.max_channels = 2;
spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x;
spec->num_cvts = 1;
- spec->cvt[0] = nvhdmi_master_con_nid_7x;
+ spec->cvts[0].cvt_nid = nvhdmi_master_con_nid_7x;
spec->pcm_playback = &nvhdmi_pcm_playback_2ch;
codec->patch_ops = nvhdmi_patch_ops_2ch;
@@ -1549,11 +1693,11 @@
substream);
if (err < 0)
return err;
- snd_hda_codec_write(codec, spec->cvt[0], 0, AC_VERB_SET_CVT_CHAN_COUNT,
- chans - 1);
+ snd_hda_codec_write(codec, spec->cvts[0].cvt_nid, 0,
+ AC_VERB_SET_CVT_CHAN_COUNT, chans - 1);
/* FIXME: XXX */
for (i = 0; i < chans; i++) {
- snd_hda_codec_write(codec, spec->cvt[0], 0,
+ snd_hda_codec_write(codec, spec->cvts[0].cvt_nid, 0,
AC_VERB_SET_HDMI_CHAN_SLOT,
(i << 4) | i);
}
@@ -1584,18 +1728,18 @@
snd_hda_sequence_write(codec, atihdmi_basic_init);
/* SI codec requires to unmute the pin */
- if (get_wcaps(codec, spec->pin[0]) & AC_WCAP_OUT_AMP)
- snd_hda_codec_write(codec, spec->pin[0], 0,
+ if (get_wcaps(codec, spec->pins[0].pin_nid) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, spec->pins[0].pin_nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
return 0;
}
static const struct hda_codec_ops atihdmi_patch_ops = {
- .build_controls = generic_hdmi_build_controls,
- .build_pcms = generic_hdmi_build_pcms,
+ .build_controls = simple_playback_build_controls,
+ .build_pcms = simple_playback_build_pcms,
.init = atihdmi_init,
- .free = generic_hdmi_free,
+ .free = simple_playback_free,
};
@@ -1613,8 +1757,8 @@
spec->multiout.max_channels = 2;
spec->multiout.dig_out_nid = ATIHDMI_CVT_NID;
spec->num_cvts = 1;
- spec->cvt[0] = ATIHDMI_CVT_NID;
- spec->pin[0] = ATIHDMI_PIN_NID;
+ spec->cvts[0].cvt_nid = ATIHDMI_CVT_NID;
+ spec->pins[0].pin_nid = ATIHDMI_PIN_NID;
spec->pcm_playback = &atihdmi_pcm_digital_playback;
codec->patch_ops = atihdmi_patch_ops;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b48fb43..52ce075 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1,7 +1,7 @@
/*
* Universal Interface for Intel High Definition Audio Codec
*
- * HD audio interface patch for ALC 260/880/882 codecs
+ * HD audio interface patch for Realtek ALC codecs
*
* Copyright (c) 2004 Kailang Yang <kailang@realtek.com.tw>
* PeiSen Hou <pshou@realtek.com.tw>
@@ -33,236 +33,11 @@
#include "hda_local.h"
#include "hda_beep.h"
-#define ALC880_FRONT_EVENT 0x01
-#define ALC880_DCVOL_EVENT 0x02
-#define ALC880_HP_EVENT 0x04
-#define ALC880_MIC_EVENT 0x08
-
-/* ALC880 board config type */
-enum {
- ALC880_3ST,
- ALC880_3ST_DIG,
- ALC880_5ST,
- ALC880_5ST_DIG,
- ALC880_W810,
- ALC880_Z71V,
- ALC880_6ST,
- ALC880_6ST_DIG,
- ALC880_F1734,
- ALC880_ASUS,
- ALC880_ASUS_DIG,
- ALC880_ASUS_W1V,
- ALC880_ASUS_DIG2,
- ALC880_FUJITSU,
- ALC880_UNIWILL_DIG,
- ALC880_UNIWILL,
- ALC880_UNIWILL_P53,
- ALC880_CLEVO,
- ALC880_TCL_S700,
- ALC880_LG,
- ALC880_LG_LW,
- ALC880_MEDION_RIM,
-#ifdef CONFIG_SND_DEBUG
- ALC880_TEST,
-#endif
- ALC880_AUTO,
- ALC880_MODEL_LAST /* last tag */
-};
-
-/* ALC260 models */
-enum {
- ALC260_BASIC,
- ALC260_HP,
- ALC260_HP_DC7600,
- ALC260_HP_3013,
- ALC260_FUJITSU_S702X,
- ALC260_ACER,
- ALC260_WILL,
- ALC260_REPLACER_672V,
- ALC260_FAVORIT100,
-#ifdef CONFIG_SND_DEBUG
- ALC260_TEST,
-#endif
- ALC260_AUTO,
- ALC260_MODEL_LAST /* last tag */
-};
-
-/* ALC262 models */
-enum {
- ALC262_BASIC,
- ALC262_HIPPO,
- ALC262_HIPPO_1,
- ALC262_FUJITSU,
- ALC262_HP_BPC,
- ALC262_HP_BPC_D7000_WL,
- ALC262_HP_BPC_D7000_WF,
- ALC262_HP_TC_T5735,
- ALC262_HP_RP5700,
- ALC262_BENQ_ED8,
- ALC262_SONY_ASSAMD,
- ALC262_BENQ_T31,
- ALC262_ULTRA,
- ALC262_LENOVO_3000,
- ALC262_NEC,
- ALC262_TOSHIBA_S06,
- ALC262_TOSHIBA_RX1,
- ALC262_TYAN,
- ALC262_AUTO,
- ALC262_MODEL_LAST /* last tag */
-};
-
-/* ALC268 models */
-enum {
- ALC267_QUANTA_IL1,
- ALC268_3ST,
- ALC268_TOSHIBA,
- ALC268_ACER,
- ALC268_ACER_DMIC,
- ALC268_ACER_ASPIRE_ONE,
- ALC268_DELL,
- ALC268_ZEPTO,
-#ifdef CONFIG_SND_DEBUG
- ALC268_TEST,
-#endif
- ALC268_AUTO,
- ALC268_MODEL_LAST /* last tag */
-};
-
-/* ALC269 models */
-enum {
- ALC269_BASIC,
- ALC269_QUANTA_FL1,
- ALC269_AMIC,
- ALC269_DMIC,
- ALC269VB_AMIC,
- ALC269VB_DMIC,
- ALC269_FUJITSU,
- ALC269_LIFEBOOK,
- ALC271_ACER,
- ALC269_AUTO,
- ALC269_MODEL_LAST /* last tag */
-};
-
-/* ALC861 models */
-enum {
- ALC861_3ST,
- ALC660_3ST,
- ALC861_3ST_DIG,
- ALC861_6ST_DIG,
- ALC861_UNIWILL_M31,
- ALC861_TOSHIBA,
- ALC861_ASUS,
- ALC861_ASUS_LAPTOP,
- ALC861_AUTO,
- ALC861_MODEL_LAST,
-};
-
-/* ALC861-VD models */
-enum {
- ALC660VD_3ST,
- ALC660VD_3ST_DIG,
- ALC660VD_ASUS_V1S,
- ALC861VD_3ST,
- ALC861VD_3ST_DIG,
- ALC861VD_6ST_DIG,
- ALC861VD_LENOVO,
- ALC861VD_DALLAS,
- ALC861VD_HP,
- ALC861VD_AUTO,
- ALC861VD_MODEL_LAST,
-};
-
-/* ALC662 models */
-enum {
- ALC662_3ST_2ch_DIG,
- ALC662_3ST_6ch_DIG,
- ALC662_3ST_6ch,
- ALC662_5ST_DIG,
- ALC662_LENOVO_101E,
- ALC662_ASUS_EEEPC_P701,
- ALC662_ASUS_EEEPC_EP20,
- ALC663_ASUS_M51VA,
- ALC663_ASUS_G71V,
- ALC663_ASUS_H13,
- ALC663_ASUS_G50V,
- ALC662_ECS,
- ALC663_ASUS_MODE1,
- ALC662_ASUS_MODE2,
- ALC663_ASUS_MODE3,
- ALC663_ASUS_MODE4,
- ALC663_ASUS_MODE5,
- ALC663_ASUS_MODE6,
- ALC663_ASUS_MODE7,
- ALC663_ASUS_MODE8,
- ALC272_DELL,
- ALC272_DELL_ZM1,
- ALC272_SAMSUNG_NC10,
- ALC662_AUTO,
- ALC662_MODEL_LAST,
-};
-
-/* ALC882 models */
-enum {
- ALC882_3ST_DIG,
- ALC882_6ST_DIG,
- ALC882_ARIMA,
- ALC882_W2JC,
- ALC882_TARGA,
- ALC882_ASUS_A7J,
- ALC882_ASUS_A7M,
- ALC885_MACPRO,
- ALC885_MBA21,
- ALC885_MBP3,
- ALC885_MB5,
- ALC885_MACMINI3,
- ALC885_IMAC24,
- ALC885_IMAC91,
- ALC883_3ST_2ch_DIG,
- ALC883_3ST_6ch_DIG,
- ALC883_3ST_6ch,
- ALC883_6ST_DIG,
- ALC883_TARGA_DIG,
- ALC883_TARGA_2ch_DIG,
- ALC883_TARGA_8ch_DIG,
- ALC883_ACER,
- ALC883_ACER_ASPIRE,
- ALC888_ACER_ASPIRE_4930G,
- ALC888_ACER_ASPIRE_6530G,
- ALC888_ACER_ASPIRE_8930G,
- ALC888_ACER_ASPIRE_7730G,
- ALC883_MEDION,
- ALC883_MEDION_WIM2160,
- ALC883_LAPTOP_EAPD,
- ALC883_LENOVO_101E_2ch,
- ALC883_LENOVO_NB0763,
- ALC888_LENOVO_MS7195_DIG,
- ALC888_LENOVO_SKY,
- ALC883_HAIER_W66,
- ALC888_3ST_HP,
- ALC888_6ST_DELL,
- ALC883_MITAC,
- ALC883_CLEVO_M540R,
- ALC883_CLEVO_M720,
- ALC883_FUJITSU_PI2515,
- ALC888_FUJITSU_XA3530,
- ALC883_3ST_6ch_INTEL,
- ALC889A_INTEL,
- ALC889_INTEL,
- ALC888_ASUS_M90V,
- ALC888_ASUS_EEE1601,
- ALC889A_MB31,
- ALC1200_ASUS_P5Q,
- ALC883_SONY_VAIO_TT,
- ALC882_AUTO,
- ALC882_MODEL_LAST,
-};
-
-/* ALC680 models */
-enum {
- ALC680_BASE,
- ALC680_AUTO,
- ALC680_MODEL_LAST,
-};
+/* unsol event tags */
+#define ALC_FRONT_EVENT 0x01
+#define ALC_DCVOL_EVENT 0x02
+#define ALC_HP_EVENT 0x04
+#define ALC_MIC_EVENT 0x08
/* for GPIO Poll */
#define GPIO_MASK 0x03
@@ -276,14 +51,6 @@
ALC_INIT_GPIO3,
};
-struct alc_mic_route {
- hda_nid_t pin;
- unsigned char mux_idx;
- unsigned char amix_idx;
-};
-
-#define MUX_IDX_UNDEF ((unsigned char)-1)
-
struct alc_customize_define {
unsigned int sku_cfg;
unsigned char port_connectivity;
@@ -348,9 +115,9 @@
const hda_nid_t *adc_nids;
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid; /* digital-in NID; optional */
+ hda_nid_t mixer_nid; /* analog-mixer NID */
/* capture setup for dynamic dual-adc switch */
- unsigned int cur_adc_idx;
hda_nid_t cur_adc;
unsigned int cur_adc_stream_tag;
unsigned int cur_adc_format;
@@ -359,9 +126,9 @@
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
unsigned int cur_mux[3];
- struct alc_mic_route ext_mic;
- struct alc_mic_route dock_mic;
- struct alc_mic_route int_mic;
+ hda_nid_t ext_mic_pin;
+ hda_nid_t dock_mic_pin;
+ hda_nid_t int_mic_pin;
/* channel model */
const struct hda_channel_mode *channel_mode;
@@ -381,6 +148,9 @@
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
hda_nid_t private_adc_nids[AUTO_CFG_MAX_OUTS];
hda_nid_t private_capsrc_nids[AUTO_CFG_MAX_OUTS];
+ hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS];
+ unsigned int dyn_adc_idx[HDA_MAX_NUM_INPUTS];
+ int int_mic_idx, ext_mic_idx, dock_mic_idx; /* for auto-mic */
/* hooks */
void (*init_hook)(struct hda_codec *codec);
@@ -395,6 +165,7 @@
unsigned int line_jack_present:1;
unsigned int master_mute:1;
unsigned int auto_mic:1;
+ unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */
unsigned int automute:1; /* HP automute enabled */
unsigned int detect_line:1; /* Line-out detection enabled */
unsigned int automute_lines:1; /* automute line-out as well */
@@ -402,8 +173,9 @@
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
- unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */
+ unsigned int dyn_adc_switch:1; /* switch ADCs (for ALC275) */
unsigned int single_input_src:1;
+ unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
/* auto-mute control */
int automute_mode;
@@ -432,39 +204,23 @@
struct alc_multi_io multi_io[4];
};
-/*
- * configuration template - to be copied to the spec instance
- */
-struct alc_config_preset {
- const struct snd_kcontrol_new *mixers[5]; /* should be identical size
- * with spec
- */
- const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
- const struct hda_verb *init_verbs[5];
- unsigned int num_dacs;
- const hda_nid_t *dac_nids;
- hda_nid_t dig_out_nid; /* optional */
- hda_nid_t hp_nid; /* optional */
- const hda_nid_t *slave_dig_outs;
- unsigned int num_adc_nids;
- const hda_nid_t *adc_nids;
- const hda_nid_t *capsrc_nids;
- hda_nid_t dig_in_nid;
- unsigned int num_channel_mode;
- const struct hda_channel_mode *channel_mode;
- int need_dac_fix;
- int const_channel_count;
- unsigned int num_mux_defs;
- const struct hda_input_mux *input_mux;
- void (*unsol_event)(struct hda_codec *, unsigned int);
- void (*setup)(struct hda_codec *);
- void (*init_hook)(struct hda_codec *);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- const struct hda_amp_list *loopbacks;
- void (*power_hook)(struct hda_codec *codec);
-#endif
-};
+#define ALC_MODEL_AUTO 0 /* common for all chips */
+static bool check_amp_caps(struct hda_codec *codec, hda_nid_t nid,
+ int dir, unsigned int bits)
+{
+ if (!nid)
+ return false;
+ if (get_wcaps(codec, nid) & (1 << (dir + 1)))
+ if (query_amp_caps(codec, nid, dir) & bits)
+ return true;
+ return false;
+}
+
+#define nid_has_mute(codec, nid, dir) \
+ check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE)
+#define nid_has_volume(codec, nid, dir) \
+ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS)
/*
* input MUX handling
@@ -493,387 +249,82 @@
return 0;
}
-static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static bool alc_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t new_adc = spec->adc_nids[spec->dyn_adc_idx[cur]];
+
+ if (spec->cur_adc && spec->cur_adc != new_adc) {
+ /* stream is running, let's swap the current ADC */
+ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
+ spec->cur_adc = new_adc;
+ snd_hda_codec_setup_stream(codec, new_adc,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+ return true;
+ }
+ return false;
+}
+
+/* select the given imux item; either unmute exclusively or select the route */
+static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
+ unsigned int idx, bool force)
+{
struct alc_spec *spec = codec->spec;
const struct hda_input_mux *imux;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
unsigned int mux_idx;
- hda_nid_t nid = spec->capsrc_nids ?
- spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
- unsigned int type;
+ int i, type;
+ hda_nid_t nid;
mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
imux = &spec->input_mux[mux_idx];
if (!imux->num_items && mux_idx > 0)
imux = &spec->input_mux[0];
+ if (idx >= imux->num_items)
+ idx = imux->num_items - 1;
+ if (spec->cur_mux[adc_idx] == idx && !force)
+ return 0;
+ spec->cur_mux[adc_idx] = idx;
+
+ if (spec->dyn_adc_switch) {
+ alc_dyn_adc_pcm_resetup(codec, idx);
+ adc_idx = spec->dyn_adc_idx[idx];
+ }
+
+ nid = spec->capsrc_nids ?
+ spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
+
+ /* no selection? */
+ if (snd_hda_get_conn_list(codec, nid, NULL) <= 1)
+ return 1;
+
type = get_wcaps_type(get_wcaps(codec, nid));
if (type == AC_WID_AUD_MIX) {
/* Matrix-mixer style (e.g. ALC882) */
- unsigned int *cur_val = &spec->cur_mux[adc_idx];
- unsigned int i, idx;
-
- idx = ucontrol->value.enumerated.item[0];
- if (idx >= imux->num_items)
- idx = imux->num_items - 1;
- if (*cur_val == idx)
- return 0;
for (i = 0; i < imux->num_items; i++) {
unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
imux->items[i].index,
HDA_AMP_MUTE, v);
}
- *cur_val = idx;
- return 1;
} else {
/* MUX style (e.g. ALC880) */
- return snd_hda_input_mux_put(codec, imux, ucontrol, nid,
- &spec->cur_mux[adc_idx]);
- }
-}
-
-/*
- * channel mode setting
- */
-static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
- spec->num_channel_mode);
-}
-
-static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode,
- spec->ext_channel_count);
-}
-
-static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode,
- &spec->ext_channel_count);
- if (err >= 0 && !spec->const_channel_count) {
- spec->multiout.max_channels = spec->ext_channel_count;
- if (spec->need_dac_fix)
- spec->multiout.num_dacs = spec->multiout.max_channels / 2;
- }
- return err;
-}
-
-/*
- * Control the mode of pin widget settings via the mixer. "pc" is used
- * instead of "%" to avoid consequences of accidentally treating the % as
- * being part of a format specifier. Maximum allowed length of a value is
- * 63 characters plus NULL terminator.
- *
- * Note: some retasking pin complexes seem to ignore requests for input
- * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
- * are requested. Therefore order this list so that this behaviour will not
- * cause problems when mixer clients move through the enum sequentially.
- * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
- * March 2006.
- */
-static const char * const alc_pin_mode_names[] = {
- "Mic 50pc bias", "Mic 80pc bias",
- "Line in", "Line out", "Headphone out",
-};
-static const unsigned char alc_pin_mode_values[] = {
- PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
-};
-/* The control can present all 5 options, or it can limit the options based
- * in the pin being assumed to be exclusively an input or an output pin. In
- * addition, "input" pins may or may not process the mic bias option
- * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
- * accept requests for bias as of chip versions up to March 2006) and/or
- * wiring in the computer.
- */
-#define ALC_PIN_DIR_IN 0x00
-#define ALC_PIN_DIR_OUT 0x01
-#define ALC_PIN_DIR_INOUT 0x02
-#define ALC_PIN_DIR_IN_NOMICBIAS 0x03
-#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
-
-/* Info about the pin modes supported by the different pin direction modes.
- * For each direction the minimum and maximum values are given.
- */
-static const signed char alc_pin_mode_dir_info[5][2] = {
- { 0, 2 }, /* ALC_PIN_DIR_IN */
- { 3, 4 }, /* ALC_PIN_DIR_OUT */
- { 0, 4 }, /* ALC_PIN_DIR_INOUT */
- { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
- { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
-};
-#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
-#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
-#define alc_pin_mode_n_items(_dir) \
- (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
-
-static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- unsigned int item_num = uinfo->value.enumerated.item;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
-
- if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
- item_num = alc_pin_mode_min(dir);
- strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
- return 0;
-}
-
-static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- unsigned int i;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL,
- 0x00);
-
- /* Find enumerated value for current pinctl setting */
- i = alc_pin_mode_min(dir);
- while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
- i++;
- *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
- return 0;
-}
-
-static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL,
- 0x00);
-
- if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
- val = alc_pin_mode_min(dir);
-
- change = pinctl != alc_pin_mode_values[val];
- if (change) {
- /* Set pin mode to that requested */
snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- alc_pin_mode_values[val]);
-
- /* Also enable the retasking pin's input/output as required
- * for the requested pin mode. Enum values of 2 or less are
- * input modes.
- *
- * Dynamically switching the input/output buffers probably
- * reduces noise slightly (particularly on input) so we'll
- * do it. However, having both input and output buffers
- * enabled simultaneously doesn't seem to be problematic if
- * this turns out to be necessary in the future.
- */
- if (val <= 2) {
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
- HDA_AMP_MUTE, 0);
- } else {
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, 0);
- }
+ AC_VERB_SET_CONNECT_SEL,
+ imux->items[idx].index);
}
- return change;
+ return 1;
}
-#define ALC_PIN_MODE(xname, nid, dir) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_pin_mode_info, \
- .get = alc_pin_mode_get, \
- .put = alc_pin_mode_put, \
- .private_value = nid | (dir<<16) }
-
-/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
- * together using a mask with more than one bit set. This control is
- * currently used only by the ALC260 test model. At this stage they are not
- * needed for any "production" models.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_gpio_data_info snd_ctl_boolean_mono_info
-
-static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_GPIO_DATA, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ return alc_mux_select(codec, adc_idx,
+ ucontrol->value.enumerated.item[0], false);
}
-static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_GPIO_DATA,
- 0x00);
-
- /* Set/unset the masked GPIO bit(s) as needed */
- change = (val == 0 ? 0 : mask) != (gpio_data & mask);
- if (val == 0)
- gpio_data &= ~mask;
- else
- gpio_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_GPIO_DATA, gpio_data);
-
- return change;
-}
-#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_gpio_data_info, \
- .get = alc_gpio_data_get, \
- .put = alc_gpio_data_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
-/* A switch control to allow the enabling of the digital IO pins on the
- * ALC260. This is incredibly simplistic; the intention of this control is
- * to provide something in the test model allowing digital outputs to be
- * identified if present. If models are found which can utilise these
- * outputs a more complete mixer control can be devised for those models if
- * necessary.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
-
-static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_DIGI_CONVERT_1, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_DIGI_CONVERT_1,
- 0x00);
-
- /* Set/unset the masked control bit(s) as needed */
- change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
- if (val==0)
- ctrl_data &= ~mask;
- else
- ctrl_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- ctrl_data);
-
- return change;
-}
-#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_spdif_ctrl_info, \
- .get = alc_spdif_ctrl_get, \
- .put = alc_spdif_ctrl_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
-/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
- * Again, this is only used in the ALC26x test models to help identify when
- * the EAPD line must be asserted for features to work.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
-
-static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_EAPD_BTLENABLE, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-
-static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_EAPD_BTLENABLE,
- 0x00);
-
- /* Set/unset the masked control bit(s) as needed */
- change = (!val ? 0 : mask) != (ctrl_data & mask);
- if (!val)
- ctrl_data &= ~mask;
- else
- ctrl_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
- ctrl_data);
-
- return change;
-}
-
-#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_eapd_ctrl_info, \
- .get = alc_eapd_ctrl_get, \
- .put = alc_eapd_ctrl_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
/*
* set up the input pin config (depending on the given auto-pin type)
@@ -903,29 +354,10 @@
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
}
-static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
-
- if (!cfg->line_outs) {
- while (cfg->line_outs < AUTO_CFG_MAX_OUTS &&
- cfg->line_out_pins[cfg->line_outs])
- cfg->line_outs++;
- }
- if (!cfg->speaker_outs) {
- while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS &&
- cfg->speaker_pins[cfg->speaker_outs])
- cfg->speaker_outs++;
- }
- if (!cfg->hp_outs) {
- while (cfg->hp_outs < AUTO_CFG_MAX_OUTS &&
- cfg->hp_pins[cfg->hp_outs])
- cfg->hp_outs++;
- }
-}
-
/*
+ * Append the given mixer and verb elements for the later use
+ * The mixer array is referred in build_controls(), and init_verbs are
+ * called in init().
*/
static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix)
{
@@ -942,61 +374,8 @@
}
/*
- * set up from the preset table
+ * GPIO setup tables, used in initialization
*/
-static void setup_preset(struct hda_codec *codec,
- const struct alc_config_preset *preset)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
- add_mixer(spec, preset->mixers[i]);
- spec->cap_mixer = preset->cap_mixer;
- for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
- i++)
- add_verb(spec, preset->init_verbs[i]);
-
- spec->channel_mode = preset->channel_mode;
- spec->num_channel_mode = preset->num_channel_mode;
- spec->need_dac_fix = preset->need_dac_fix;
- spec->const_channel_count = preset->const_channel_count;
-
- if (preset->const_channel_count)
- spec->multiout.max_channels = preset->const_channel_count;
- else
- spec->multiout.max_channels = spec->channel_mode[0].channels;
- spec->ext_channel_count = spec->channel_mode[0].channels;
-
- spec->multiout.num_dacs = preset->num_dacs;
- spec->multiout.dac_nids = preset->dac_nids;
- spec->multiout.dig_out_nid = preset->dig_out_nid;
- spec->multiout.slave_dig_outs = preset->slave_dig_outs;
- spec->multiout.hp_nid = preset->hp_nid;
-
- spec->num_mux_defs = preset->num_mux_defs;
- if (!spec->num_mux_defs)
- spec->num_mux_defs = 1;
- spec->input_mux = preset->input_mux;
-
- spec->num_adc_nids = preset->num_adc_nids;
- spec->adc_nids = preset->adc_nids;
- spec->capsrc_nids = preset->capsrc_nids;
- spec->dig_in_nid = preset->dig_in_nid;
-
- spec->unsol_event = preset->unsol_event;
- spec->init_hook = preset->init_hook;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->power_hook = preset->power_hook;
- spec->loopback.amplist = preset->loopbacks;
-#endif
-
- if (preset->setup)
- preset->setup(codec);
-
- alc_fixup_autocfg_pin_nums(codec);
-}
-
/* Enable GPIO mask and set output */
static const struct hda_verb alc_gpio1_init_verbs[] = {
{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
@@ -1051,14 +430,19 @@
alc_fix_pll(codec);
}
+/*
+ * Jack-reporting via input-jack layer
+ */
+
+/* initialization of jacks; currently checks only a few known pins */
static int alc_init_jacks(struct hda_codec *codec)
{
#ifdef CONFIG_SND_HDA_INPUT_JACK
struct alc_spec *spec = codec->spec;
int err;
unsigned int hp_nid = spec->autocfg.hp_pins[0];
- unsigned int mic_nid = spec->ext_mic.pin;
- unsigned int dock_nid = spec->dock_mic.pin;
+ unsigned int mic_nid = spec->ext_mic_pin;
+ unsigned int dock_nid = spec->dock_mic_pin;
if (hp_nid) {
err = snd_hda_input_jack_add(codec, hp_nid,
@@ -1086,7 +470,12 @@
return 0;
}
-static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
+/*
+ * Jack detections for HP auto-mute and mic-switch
+ */
+
+/* check each pin in the given array; returns true if any of them is plugged */
+static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
{
int i, present = 0;
@@ -1100,6 +489,7 @@
return present;
}
+/* standard HP/line-out auto-mute helper */
static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
bool mute, bool hp_out)
{
@@ -1170,6 +560,7 @@
spec->autocfg.line_out_pins, on, false);
}
+/* standard HP-automute helper */
static void alc_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1182,6 +573,7 @@
update_speakers(codec);
}
+/* standard line-out-automute helper */
static void alc_line_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1194,106 +586,33 @@
update_speakers(codec);
}
-static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
- hda_nid_t nid)
-{
- hda_nid_t conn[HDA_MAX_NUM_INPUTS];
- int i, nums;
+#define get_connection_index(codec, mux, nid) \
+ snd_hda_get_conn_index(codec, mux, nid, 0)
- nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
- for (i = 0; i < nums; i++)
- if (conn[i] == nid)
- return i;
- return -1;
-}
-
-/* switch the current ADC according to the jack state */
-static void alc_dual_mic_adc_auto_switch(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- unsigned int present;
- hda_nid_t new_adc;
-
- present = snd_hda_jack_detect(codec, spec->ext_mic.pin);
- if (present)
- spec->cur_adc_idx = 1;
- else
- spec->cur_adc_idx = 0;
- new_adc = spec->adc_nids[spec->cur_adc_idx];
- if (spec->cur_adc && spec->cur_adc != new_adc) {
- /* stream is running, let's swap the current ADC */
- __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
- spec->cur_adc = new_adc;
- snd_hda_codec_setup_stream(codec, new_adc,
- spec->cur_adc_stream_tag, 0,
- spec->cur_adc_format);
- }
-}
-
+/* standard mic auto-switch helper */
static void alc_mic_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- struct alc_mic_route *dead1, *dead2, *alive;
- unsigned int present, type;
- hda_nid_t cap_nid;
+ hda_nid_t *pins = spec->imux_pins;
- if (!spec->auto_mic)
- return;
- if (!spec->int_mic.pin || !spec->ext_mic.pin)
+ if (!spec->auto_mic || !spec->auto_mic_valid_imux)
return;
if (snd_BUG_ON(!spec->adc_nids))
return;
-
- if (spec->dual_adc_switch) {
- alc_dual_mic_adc_auto_switch(codec);
+ if (snd_BUG_ON(spec->int_mic_idx < 0 || spec->ext_mic_idx < 0))
return;
- }
- cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
+ if (snd_hda_jack_detect(codec, pins[spec->ext_mic_idx]))
+ alc_mux_select(codec, 0, spec->ext_mic_idx, false);
+ else if (spec->dock_mic_idx >= 0 &&
+ snd_hda_jack_detect(codec, pins[spec->dock_mic_idx]))
+ alc_mux_select(codec, 0, spec->dock_mic_idx, false);
+ else
+ alc_mux_select(codec, 0, spec->int_mic_idx, false);
- alive = &spec->int_mic;
- dead1 = &spec->ext_mic;
- dead2 = &spec->dock_mic;
-
- present = snd_hda_jack_detect(codec, spec->ext_mic.pin);
- if (present) {
- alive = &spec->ext_mic;
- dead1 = &spec->int_mic;
- dead2 = &spec->dock_mic;
- }
- if (!present && spec->dock_mic.pin > 0) {
- present = snd_hda_jack_detect(codec, spec->dock_mic.pin);
- if (present) {
- alive = &spec->dock_mic;
- dead1 = &spec->int_mic;
- dead2 = &spec->ext_mic;
- }
- snd_hda_input_jack_report(codec, spec->dock_mic.pin);
- }
-
- type = get_wcaps_type(get_wcaps(codec, cap_nid));
- if (type == AC_WID_AUD_MIX) {
- /* Matrix-mixer style (e.g. ALC882) */
- snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
- alive->mux_idx,
- HDA_AMP_MUTE, 0);
- if (dead1->pin > 0)
- snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
- dead1->mux_idx,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- if (dead2->pin > 0)
- snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
- dead2->mux_idx,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* MUX style (e.g. ALC880) */
- snd_hda_codec_write_cache(codec, cap_nid, 0,
- AC_VERB_SET_CONNECT_SEL,
- alive->mux_idx);
- }
- snd_hda_input_jack_report(codec, spec->ext_mic.pin);
-
- /* FIXME: analog mixer */
+ snd_hda_input_jack_report(codec, pins[spec->ext_mic_idx]);
+ if (spec->dock_mic_idx >= 0)
+ snd_hda_input_jack_report(codec, pins[spec->dock_mic_idx]);
}
/* unsolicited event for HP jack sensing */
@@ -1304,18 +623,19 @@
else
res >>= 26;
switch (res) {
- case ALC880_HP_EVENT:
+ case ALC_HP_EVENT:
alc_hp_automute(codec);
break;
- case ALC880_FRONT_EVENT:
+ case ALC_FRONT_EVENT:
alc_line_automute(codec);
break;
- case ALC880_MIC_EVENT:
+ case ALC_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
+/* call init functions of standard auto-mute helpers */
static void alc_inithook(struct hda_codec *codec)
{
alc_hp_automute(codec);
@@ -1341,6 +661,7 @@
AC_VERB_SET_PROC_COEF, 0x3030);
}
+/* additional initialization for ALC889 variants */
static void alc889_coef_init(struct hda_codec *codec)
{
unsigned int tmp;
@@ -1365,28 +686,12 @@
static void alc_auto_setup_eapd(struct hda_codec *codec, bool on)
{
/* We currently only handle front, HP */
- switch (codec->vendor_id) {
- case 0x10ec0260:
- set_eapd(codec, 0x0f, on);
- set_eapd(codec, 0x10, on);
- break;
- case 0x10ec0262:
- case 0x10ec0267:
- case 0x10ec0268:
- case 0x10ec0269:
- case 0x10ec0270:
- case 0x10ec0272:
- case 0x10ec0660:
- case 0x10ec0662:
- case 0x10ec0663:
- case 0x10ec0665:
- case 0x10ec0862:
- case 0x10ec0889:
- case 0x10ec0892:
- set_eapd(codec, 0x14, on);
- set_eapd(codec, 0x15, on);
- break;
- }
+ static hda_nid_t pins[] = {
+ 0x0f, 0x10, 0x14, 0x15, 0
+ };
+ hda_nid_t *p;
+ for (p = pins; *p; p++)
+ set_eapd(codec, *p, on);
}
/* generic shutup callback;
@@ -1398,10 +703,12 @@
msleep(200);
}
+/* generic EAPD initialization */
static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
unsigned int tmp;
+ alc_auto_setup_eapd(codec, true);
switch (type) {
case ALC_INIT_GPIO1:
snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
@@ -1413,7 +720,6 @@
snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
break;
case ALC_INIT_DEFAULT:
- alc_auto_setup_eapd(codec, true);
switch (codec->vendor_id) {
case 0x10ec0260:
snd_hda_codec_write(codec, 0x1a, 0,
@@ -1457,6 +763,9 @@
}
}
+/*
+ * Auto-Mute mode mixer enum support
+ */
static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -1543,7 +852,11 @@
.put = alc_automute_mode_put,
};
-static struct snd_kcontrol_new *alc_kcontrol_new(struct alc_spec *spec);
+static struct snd_kcontrol_new *alc_kcontrol_new(struct alc_spec *spec)
+{
+ snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32);
+ return snd_array_new(&spec->kctls);
+}
static int alc_add_automute_mode_enum(struct hda_codec *codec)
{
@@ -1560,6 +873,10 @@
return 0;
}
+/*
+ * Check the availability of HP/line-out auto-mute;
+ * Set up appropriately if really supported
+ */
static void alc_init_auto_hp(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1598,7 +915,7 @@
nid);
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | ALC880_HP_EVENT);
+ AC_USRSP_EN | ALC_HP_EVENT);
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_PIN;
}
@@ -1613,7 +930,7 @@
"on NID 0x%x\n", nid);
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | ALC880_FRONT_EVENT);
+ AC_USRSP_EN | ALC_FRONT_EVENT);
spec->detect_line = 1;
}
spec->automute_lines = spec->detect_line;
@@ -1626,6 +943,132 @@
}
}
+/* return the position of NID in the list, or -1 if not found */
+static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+ int i;
+ for (i = 0; i < nums; i++)
+ if (list[i] == nid)
+ return i;
+ return -1;
+}
+
+/* check whether dynamic ADC-switching is available */
+static bool alc_check_dyn_adc_switch(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, n, idx;
+ hda_nid_t cap, pin;
+
+ if (imux != spec->input_mux) /* no dynamic imux? */
+ return false;
+
+ for (n = 0; n < spec->num_adc_nids; n++) {
+ cap = spec->private_capsrc_nids[n];
+ for (i = 0; i < imux->num_items; i++) {
+ pin = spec->imux_pins[i];
+ if (!pin)
+ return false;
+ if (get_connection_index(codec, cap, pin) < 0)
+ break;
+ }
+ if (i >= imux->num_items)
+ return true; /* no ADC-switch is needed */
+ }
+
+ for (i = 0; i < imux->num_items; i++) {
+ pin = spec->imux_pins[i];
+ for (n = 0; n < spec->num_adc_nids; n++) {
+ cap = spec->private_capsrc_nids[n];
+ idx = get_connection_index(codec, cap, pin);
+ if (idx >= 0) {
+ imux->items[i].index = idx;
+ spec->dyn_adc_idx[i] = n;
+ break;
+ }
+ }
+ }
+
+ snd_printdd("realtek: enabling ADC switching\n");
+ spec->dyn_adc_switch = 1;
+ return true;
+}
+
+/* rebuild imux for matching with the given auto-mic pins (if not yet) */
+static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct hda_input_mux *imux;
+ static char * const texts[3] = {
+ "Mic", "Internal Mic", "Dock Mic"
+ };
+ int i;
+
+ if (!spec->auto_mic)
+ return false;
+ imux = &spec->private_imux[0];
+ if (spec->input_mux == imux)
+ return true;
+ spec->imux_pins[0] = spec->ext_mic_pin;
+ spec->imux_pins[1] = spec->int_mic_pin;
+ spec->imux_pins[2] = spec->dock_mic_pin;
+ for (i = 0; i < 3; i++) {
+ strcpy(imux->items[i].label, texts[i]);
+ if (spec->imux_pins[i])
+ imux->num_items = i + 1;
+ }
+ spec->num_mux_defs = 1;
+ spec->input_mux = imux;
+ return true;
+}
+
+/* check whether all auto-mic pins are valid; setup indices if OK */
+static bool alc_auto_mic_check_imux(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ const struct hda_input_mux *imux;
+
+ if (!spec->auto_mic)
+ return false;
+ if (spec->auto_mic_valid_imux)
+ return true; /* already checked */
+
+ /* fill up imux indices */
+ if (!alc_check_dyn_adc_switch(codec)) {
+ spec->auto_mic = 0;
+ return false;
+ }
+
+ imux = spec->input_mux;
+ spec->ext_mic_idx = find_idx_in_nid_list(spec->ext_mic_pin,
+ spec->imux_pins, imux->num_items);
+ spec->int_mic_idx = find_idx_in_nid_list(spec->int_mic_pin,
+ spec->imux_pins, imux->num_items);
+ spec->dock_mic_idx = find_idx_in_nid_list(spec->dock_mic_pin,
+ spec->imux_pins, imux->num_items);
+ if (spec->ext_mic_idx < 0 || spec->int_mic_idx < 0) {
+ spec->auto_mic = 0;
+ return false; /* no corresponding imux */
+ }
+
+ snd_hda_codec_write_cache(codec, spec->ext_mic_pin, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC_MIC_EVENT);
+ if (spec->dock_mic_pin)
+ snd_hda_codec_write_cache(codec, spec->dock_mic_pin, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC_MIC_EVENT);
+
+ spec->auto_mic_valid_imux = 1;
+ spec->auto_mic = 1;
+ return true;
+}
+
+/*
+ * Check the availability of auto-mic switch;
+ * Set up if really supported
+ */
static void alc_init_auto_mic(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1633,6 +1076,8 @@
hda_nid_t fixed, ext, dock;
int i;
+ spec->ext_mic_idx = spec->int_mic_idx = spec->dock_mic_idx = -1;
+
fixed = ext = dock = 0;
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
@@ -1674,21 +1119,32 @@
return; /* no unsol support */
if (dock && !is_jack_detectable(codec, dock))
return; /* no unsol support */
+
+ /* check imux indices */
+ spec->ext_mic_pin = ext;
+ spec->int_mic_pin = fixed;
+ spec->dock_mic_pin = dock;
+
+ spec->auto_mic = 1;
+ if (!alc_auto_mic_check_imux(codec))
+ return;
+
snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n",
ext, fixed, dock);
- spec->ext_mic.pin = ext;
- spec->dock_mic.pin = dock;
- spec->int_mic.pin = fixed;
- spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
- spec->dock_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
- spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
- spec->auto_mic = 1;
- snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | ALC880_MIC_EVENT);
spec->unsol_event = alc_sku_unsol_event;
}
+/* check the availabilities of auto-mute and auto-mic switches */
+static void alc_auto_check_switches(struct hda_codec *codec)
+{
+ alc_init_auto_hp(codec);
+ alc_init_auto_mic(codec);
+}
+
+/*
+ * Realtek SSID verification
+ */
+
/* Could be any non-zero and even value. When used as fixup, tells
* the driver to ignore any present sku defines.
*/
@@ -1759,6 +1215,12 @@
return 0;
}
+/* return true if the given NID is found in the list */
+static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+ return find_idx_in_nid_list(nid, list, nums) >= 0;
+}
+
/* check subsystem ID and set up device-specific initialization;
* return 1 if initialized, 0 if invalid SSID
*/
@@ -1868,27 +1330,24 @@
nid = porti;
else
return 1;
- for (i = 0; i < spec->autocfg.line_outs; i++)
- if (spec->autocfg.line_out_pins[i] == nid)
- return 1;
+ if (found_in_nid_list(nid, spec->autocfg.line_out_pins,
+ spec->autocfg.line_outs))
+ return 1;
spec->autocfg.hp_pins[0] = nid;
}
return 1;
}
-static void alc_ssid_check(struct hda_codec *codec,
- hda_nid_t porta, hda_nid_t porte,
- hda_nid_t portd, hda_nid_t porti)
+/* Check the validity of ALC subsystem-id
+ * ports contains an array of 4 pin NIDs for port-A, E, D and I */
+static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports)
{
- if (!alc_subsystem_id(codec, porta, porte, portd, porti)) {
+ if (!alc_subsystem_id(codec, ports[0], ports[1], ports[2], ports[3])) {
struct alc_spec *spec = codec->spec;
snd_printd("realtek: "
"Enable default setup for auto mode as fallback\n");
spec->init_amp = ALC_INIT_DEFAULT;
}
-
- alc_init_auto_hp(codec);
- alc_init_auto_mic(codec);
}
/*
@@ -2036,6 +1495,9 @@
}
}
+/*
+ * COEF access helper functions
+ */
static int alc_read_coef_idx(struct hda_codec *codec,
unsigned int coef_idx)
{
@@ -2056,20 +1518,32 @@
coef_val);
}
+/*
+ * Digital I/O handling
+ */
+
/* set right pin controls for digital I/O */
static void alc_auto_init_digital(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
- hda_nid_t pin;
+ hda_nid_t pin, dac;
for (i = 0; i < spec->autocfg.dig_outs; i++) {
pin = spec->autocfg.dig_out_pins[i];
- if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
- }
+ if (!pin)
+ continue;
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ if (!i)
+ dac = spec->multiout.dig_out_nid;
+ else
+ dac = spec->slave_dig_outs[i - 1];
+ if (!dac || !(get_wcaps(codec, dac) & AC_WCAP_OUT_AMP))
+ continue;
+ snd_hda_codec_write(codec, dac, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
}
pin = spec->autocfg.dig_in_pin;
if (pin)
@@ -2087,11 +1561,13 @@
/* support multiple SPDIFs; the secondary is set up as a slave */
for (i = 0; i < spec->autocfg.dig_outs; i++) {
+ hda_nid_t conn[4];
err = snd_hda_get_connections(codec,
spec->autocfg.dig_out_pins[i],
- &dig_nid, 1);
+ conn, ARRAY_SIZE(conn));
if (err < 0)
continue;
+ dig_nid = conn[0]; /* assume the first element is audio-out */
if (!i) {
spec->multiout.dig_out_nid = dig_nid;
spec->dig_out_type = spec->autocfg.dig_out_type[0];
@@ -2124,572 +1600,22 @@
}
/*
- * ALC888
+ * capture mixer elements
*/
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc888_4ST_ch2_intel_init[] = {
-/* Mic-in jack as mic in */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-/* Line-in jack as Line in */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-/* Line-Out as Front */
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc888_4ST_ch4_intel_init[] = {
-/* Mic-in jack as mic in */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-/* Line-in jack as Surround */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-/* Line-Out as Front */
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc888_4ST_ch6_intel_init[] = {
-/* Mic-in jack as CLFE */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-/* Line-in jack as Surround */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-/* Line-Out as CLFE (workaround because Mic-in is not loud enough) */
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static const struct hda_verb alc888_4ST_ch8_intel_init[] = {
-/* Mic-in jack as CLFE */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-/* Line-in jack as Surround */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-/* Line-Out as Side */
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
- { } /* end */
-};
-
-static const struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = {
- { 2, alc888_4ST_ch2_intel_init },
- { 4, alc888_4ST_ch4_intel_init },
- { 6, alc888_4ST_ch6_intel_init },
- { 8, alc888_4ST_ch8_intel_init },
-};
-
-/*
- * ALC888 Fujitsu Siemens Amillo xa3530
- */
-
-static const struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
-/* Front Mic: set to PIN_IN (empty by default) */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-/* Connect Internal HP to Front */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-/* Connect Bass HP to Front */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-/* Connect Line-Out side jack (SPDIF) to Side */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
-/* Connect Mic jack to CLFE */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
-/* Connect Line-in jack to Surround */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-/* Connect HP out jack to Front */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-/* Enable unsolicited event for HP jack and Line-out jack */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {}
-};
-
-static void alc889_automute_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->autocfg.speaker_pins[2] = 0x17;
- spec->autocfg.speaker_pins[3] = 0x19;
- spec->autocfg.speaker_pins[4] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc889_intel_init_hook(struct hda_codec *codec)
-{
- alc889_coef_init(codec);
- alc_hp_automute(codec);
-}
-
-static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x17; /* line-out */
- spec->autocfg.hp_pins[1] = 0x1b; /* hp */
- spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
- spec->autocfg.speaker_pins[1] = 0x15; /* bass */
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/*
- * ALC888 Acer Aspire 4930G model
- */
-
-static const struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
-/* Front Mic: set to PIN_IN (empty by default) */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-/* Unselect Front Mic by default in input mixer 3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
-/* Enable unsolicited event for HP jack */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-/* Connect Internal HP to front */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-/* Connect HP out to front */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-/*
- * ALC888 Acer Aspire 6530G model
- */
-
-static const struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
-/* Route to built-in subwoofer as well as speakers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-/* Bias voltage on for external mic port */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
-/* Front Mic: set to PIN_IN (empty by default) */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-/* Unselect Front Mic by default in input mixer 3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
-/* Enable unsolicited event for HP jack */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-/* Enable speaker output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-/* Enable headphone output */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-/*
- *ALC888 Acer Aspire 7730G model
- */
-
-static const struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
-/* Bias voltage on for external mic port */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
-/* Front Mic: set to PIN_IN (empty by default) */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-/* Unselect Front Mic by default in input mixer 3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
-/* Enable unsolicited event for HP jack */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-/* Enable speaker output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-/* Enable headphone output */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
-/*Enable internal subwoofer */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x17, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-/*
- * ALC889 Acer Aspire 8930G model
- */
-
-static const struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
-/* Front Mic: set to PIN_IN (empty by default) */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-/* Unselect Front Mic by default in input mixer 3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
-/* Enable unsolicited event for HP jack */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-/* Connect Internal Front to Front */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-/* Connect Internal Rear to Rear */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
-/* Connect Internal CLFE to CLFE */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
-/* Connect HP out to Front */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-/* Enable all DACs */
-/* DAC DISABLE/MUTE 1? */
-/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x03},
- {0x20, AC_VERB_SET_PROC_COEF, 0x0000},
-/* DAC DISABLE/MUTE 2? */
-/* some bit here disables the other DACs. Init=0x4900 */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x08},
- {0x20, AC_VERB_SET_PROC_COEF, 0x0000},
-/* DMIC fix
- * This laptop has a stereo digital microphone. The mics are only 1cm apart
- * which makes the stereo useless. However, either the mic or the ALC889
- * makes the signal become a difference/sum signal instead of standard
- * stereo, which is annoying. So instead we flip this bit which makes the
- * codec replicate the sum signal to both channels, turning it into a
- * normal mono mic.
- */
-/* DMIC_CONTROL? Init value = 0x0001 */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
- {0x20, AC_VERB_SET_PROC_COEF, 0x0003},
- { }
-};
-
-static const struct hda_input_mux alc888_2_capture_sources[2] = {
- /* Front mic only available on one ADC */
- {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Front Mic", 0xb },
- },
- },
- {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
- }
-};
-
-static const struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
- /* Interal mic only available on one ADC */
- {
- .num_items = 5,
- .items = {
- { "Mic", 0x0 },
- { "Line In", 0x2 },
- { "CD", 0x4 },
- { "Input Mix", 0xa },
- { "Internal Mic", 0xb },
- },
- },
- {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line In", 0x2 },
- { "CD", 0x4 },
- { "Input Mix", 0xa },
- },
- }
-};
-
-static const struct hda_input_mux alc889_capture_sources[3] = {
- /* Digital mic only available on first "ADC" */
- {
- .num_items = 5,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Front Mic", 0xb },
- { "Input Mix", 0xa },
- },
- },
- {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Input Mix", 0xa },
- },
- },
- {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Input Mix", 0xa },
- },
- }
-};
-
-static const struct snd_kcontrol_new alc888_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Internal LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Internal LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-
-static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->autocfg.speaker_pins[2] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->autocfg.speaker_pins[2] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->autocfg.speaker_pins[2] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->autocfg.speaker_pins[2] = 0x1b;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/*
- * ALC880 3-stack model
- *
- * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
- * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
- * F-Mic = 0x1b, HP = 0x19
- */
-
-static const hda_nid_t alc880_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x05, 0x04, 0x03
-};
-
-static const hda_nid_t alc880_adc_nids[3] = {
- /* ADC0-2 */
- 0x07, 0x08, 0x09,
-};
-
-/* The datasheet says the node 0x07 is connected from inputs,
- * but it shows zero connection in the real implementation on some devices.
- * Note: this is a 915GAV bug, fixed on 915GLV
- */
-static const hda_nid_t alc880_adc_nids_alt[2] = {
- /* ADC1-2 */
- 0x08, 0x09,
-};
-
-#define ALC880_DIGOUT_NID 0x06
-#define ALC880_DIGIN_NID 0x0a
-
-static const struct hda_input_mux alc880_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x3 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* channel source setting (2/6 channel selection for 3-stack) */
-/* 2ch mode */
-static const struct hda_verb alc880_threestack_ch2_init[] = {
- /* set line-in to input, mute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- /* set mic-in to input vref 80%, mute it */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/* 6ch mode */
-static const struct hda_verb alc880_threestack_ch6_init[] = {
- /* set line-in to output, unmute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- /* set mic-in to output, unmute it */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc880_threestack_modes[2] = {
- { 2, alc880_threestack_ch2_init },
- { 6, alc880_threestack_ch6_init },
-};
-
-static const struct snd_kcontrol_new alc880_three_stack_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-/* capture mixer elements */
static int alc_cap_vol_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
+ unsigned long val;
int err;
mutex_lock(&codec->control_mutex);
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
- HDA_INPUT);
+ if (spec->vol_in_capsrc)
+ val = HDA_COMPOSE_AMP_VAL(spec->capsrc_nids[0], 3, 0, HDA_OUTPUT);
+ else
+ val = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT);
+ kcontrol->private_value = val;
err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo);
mutex_unlock(&codec->control_mutex);
return err;
@@ -2700,11 +1626,15 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
+ unsigned long val;
int err;
mutex_lock(&codec->control_mutex);
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0,
- HDA_INPUT);
+ if (spec->vol_in_capsrc)
+ val = HDA_COMPOSE_AMP_VAL(spec->capsrc_nids[0], 3, 0, HDA_OUTPUT);
+ else
+ val = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT);
+ kcontrol->private_value = val;
err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv);
mutex_unlock(&codec->control_mutex);
return err;
@@ -2722,7 +1652,7 @@
int i, err = 0;
mutex_lock(&codec->control_mutex);
- if (check_adc_switch && spec->dual_adc_switch) {
+ if (check_adc_switch && spec->dyn_adc_switch) {
for (i = 0; i < spec->num_adc_nids; i++) {
kcontrol->private_value =
HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
@@ -2733,9 +1663,14 @@
}
} else {
i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- kcontrol->private_value =
- HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
- 3, 0, HDA_INPUT);
+ if (spec->vol_in_capsrc)
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->capsrc_nids[i],
+ 3, 0, HDA_OUTPUT);
+ else
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+ 3, 0, HDA_INPUT);
err = func(kcontrol, ucontrol);
}
error:
@@ -2830,335 +1765,6 @@
DEFINE_CAPMIX_NOSRC(3);
/*
- * ALC880 5-stack model
- *
- * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
- * Side = 0x02 (0xd)
- * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
- * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
- */
-
-/* additional mixers to alc880_three_stack_mixer */
-static const struct snd_kcontrol_new alc880_five_stack_mixer[] = {
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
- { } /* end */
-};
-
-/* channel source setting (6/8 channel selection for 5-stack) */
-/* 6ch mode */
-static const struct hda_verb alc880_fivestack_ch6_init[] = {
- /* set line-in to input, mute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/* 8ch mode */
-static const struct hda_verb alc880_fivestack_ch8_init[] = {
- /* set line-in to output, unmute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc880_fivestack_modes[2] = {
- { 6, alc880_fivestack_ch6_init },
- { 8, alc880_fivestack_ch8_init },
-};
-
-
-/*
- * ALC880 6-stack model
- *
- * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
- * Side = 0x05 (0x0f)
- * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
- * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
- */
-
-static const hda_nid_t alc880_6st_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x03, 0x04, 0x05
-};
-
-static const struct hda_input_mux alc880_6stack_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* fixed 8-channels */
-static const struct hda_channel_mode alc880_sixstack_modes[1] = {
- { 8, NULL },
-};
-
-static const struct snd_kcontrol_new alc880_six_stack_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-
-/*
- * ALC880 W810 model
- *
- * W810 has rear IO for:
- * Front (DAC 02)
- * Surround (DAC 03)
- * Center/LFE (DAC 04)
- * Digital out (06)
- *
- * The system also has a pair of internal speakers, and a headphone jack.
- * These are both connected to Line2 on the codec, hence to DAC 02.
- *
- * There is a variable resistor to control the speaker or headphone
- * volume. This is a hardware-only device without a software API.
- *
- * Plugging headphones in will disable the internal speakers. This is
- * implemented in hardware, not via the driver using jack sense. In
- * a similar fashion, plugging into the rear socket marked "front" will
- * disable both the speakers and headphones.
- *
- * For input, there's a microphone jack, and an "audio in" jack.
- * These may not do anything useful with this driver yet, because I
- * haven't setup any initialization verbs for these yet...
- */
-
-static const hda_nid_t alc880_w810_dac_nids[3] = {
- /* front, rear/surround, clfe */
- 0x02, 0x03, 0x04
-};
-
-/* fixed 6 channels */
-static const struct hda_channel_mode alc880_w810_modes[1] = {
- { 6, NULL }
-};
-
-/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
-static const struct snd_kcontrol_new alc880_w810_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-
-/*
- * Z710V model
- *
- * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
- * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
- * Line = 0x1a
- */
-
-static const hda_nid_t alc880_z71v_dac_nids[1] = {
- 0x02
-};
-#define ALC880_Z71V_HP_DAC 0x03
-
-/* fixed 2 channels */
-static const struct hda_channel_mode alc880_2_jack_modes[1] = {
- { 2, NULL }
-};
-
-static const struct snd_kcontrol_new alc880_z71v_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-
-/*
- * ALC880 F1734 model
- *
- * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
- * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
- */
-
-static const hda_nid_t alc880_f1734_dac_nids[1] = {
- 0x03
-};
-#define ALC880_F1734_HP_DAC 0x02
-
-static const struct snd_kcontrol_new alc880_f1734_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_input_mux alc880_f1734_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 },
- { "CD", 0x4 },
- },
-};
-
-
-/*
- * ALC880 ASUS model
- *
- * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
- * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
- * Mic = 0x18, Line = 0x1a
- */
-
-#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
-#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
-
-static const struct snd_kcontrol_new alc880_asus_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-/*
- * ALC880 ASUS W1V model
- *
- * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
- * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
- * Mic = 0x18, Line = 0x1a, Line2 = 0x1b
- */
-
-/* additional mixers to alc880_asus_mixer */
-static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
- HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
- { } /* end */
-};
-
-/* TCL S700 */
-static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-/* Uniwill */
-static const struct snd_kcontrol_new alc880_uniwill_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-/*
* virtual master controls
*/
@@ -3237,6 +1843,7 @@
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
@@ -3368,789 +1975,6 @@
/*
- * initialize the codec volumes, etc
- */
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc880_volume_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for front
- * panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-
- /*
- * Set up output mixers (0x0c - 0x0f)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- { }
-};
-
-/*
- * 3-stack pin configuration:
- * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
- */
-static const struct hda_verb alc880_pin_3stack_init_verbs[] = {
- /*
- * preset connection lists of input pins
- * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
- */
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
-
- /*
- * Set pin mode and muting
- */
- /* set front pin widgets 0x14 for output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mic2 (as headphone out) for HP output */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line2 (as front mic) pin widget for input and vref at 80% */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * 5-stack pin configuration:
- * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
- * line-in/side = 0x1a, f-mic = 0x1b
- */
-static const struct hda_verb alc880_pin_5stack_init_verbs[] = {
- /*
- * preset connection lists of input pins
- * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
- */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
-
- /*
- * Set pin mode and muting
- */
- /* set pin widgets 0x14-0x17 for output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* unmute pins for output (no gain on this amp) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mic2 (as headphone out) for HP output */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line2 (as front mic) pin widget for input and vref at 80% */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * W810 pin configuration:
- * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
- */
-static const struct hda_verb alc880_pin_w810_init_verbs[] = {
- /* hphone/speaker input selector: front DAC */
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- { }
-};
-
-/*
- * Z71V pin configuration:
- * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
- */
-static const struct hda_verb alc880_pin_z71v_init_verbs[] = {
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * 6-stack pin configuration:
- * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
- * f-mic = 0x19, line = 0x1a, HP = 0x1b
- */
-static const struct hda_verb alc880_pin_6stack_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * Uniwill pin configuration:
- * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
- * line = 0x1a
- */
-static const struct hda_verb alc880_uniwill_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */
- /* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
-
- { }
-};
-
-/*
-* Uniwill P53
-* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
- */
-static const struct hda_verb alc880_uniwill_p53_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_DCVOL_EVENT},
-
- { }
-};
-
-static const struct hda_verb alc880_beep_init_verbs[] = {
- { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
- { }
-};
-
-/* auto-toggle front mic */
-static void alc88x_simple_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_jack_detect(codec, 0x18);
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
-}
-
-static void alc880_uniwill_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc880_uniwill_init_hook(struct hda_codec *codec)
-{
- alc_hp_automute(codec);
- alc88x_simple_mic_automute(codec);
-}
-
-static void alc880_uniwill_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 28 bit!
- */
- switch (res >> 28) {
- case ALC880_MIC_EVENT:
- alc88x_simple_mic_automute(codec);
- break;
- default:
- alc_sku_unsol_event(codec, res);
- break;
- }
-}
-
-static void alc880_uniwill_p53_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x21, 0,
- AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
- present &= HDA_AMP_VOLMASK;
- snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
- HDA_AMP_VOLMASK, present);
- snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
- HDA_AMP_VOLMASK, present);
-}
-
-static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 28 bit!
- */
- if ((res >> 28) == ALC880_DCVOL_EVENT)
- alc880_uniwill_p53_dcvol_automute(codec);
- else
- alc_sku_unsol_event(codec, res);
-}
-
-/*
- * F1734 pin configuration:
- * HP = 0x14, speaker-out = 0x15, mic = 0x18
- */
-static const struct hda_verb alc880_pin_f1734_init_verbs[] = {
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_DCVOL_EVENT},
-
- { }
-};
-
-/*
- * ASUS pin configuration:
- * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
- */
-static const struct hda_verb alc880_pin_asus_init_verbs[] = {
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/* Enable GPIO mask and set output */
-#define alc880_gpio1_init_verbs alc_gpio1_init_verbs
-#define alc880_gpio2_init_verbs alc_gpio2_init_verbs
-#define alc880_gpio3_init_verbs alc_gpio3_init_verbs
-
-/* Clevo m520g init */
-static const struct hda_verb alc880_pin_clevo_init_verbs[] = {
- /* headphone output */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* line-out */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Line-in */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* CD */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Mic1 (rear panel) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Mic2 (front panel) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* headphone */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* change to EAPD mode */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
-
- { }
-};
-
-static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
- /* change to EAPD mode */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
-
- /* Headphone output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Front output*/
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-
- /* change to EAPD mode */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
-
- { }
-};
-
-/*
- * LG m1 express dual
- *
- * Pin assignment:
- * Rear Line-In/Out (blue): 0x14
- * Build-in Mic-In: 0x15
- * Speaker-out: 0x17
- * HP-Out (green): 0x1b
- * Mic-In/Out (red): 0x19
- * SPDIF-Out: 0x1e
- */
-
-/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
-static const hda_nid_t alc880_lg_dac_nids[3] = {
- 0x05, 0x02, 0x03
-};
-
-/* seems analog CD is not working */
-static const struct hda_input_mux alc880_lg_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x1 },
- { "Line", 0x5 },
- { "Internal Mic", 0x6 },
- },
-};
-
-/* 2,4,6 channel modes */
-static const struct hda_verb alc880_lg_ch2_init[] = {
- /* set line-in and mic-in to input */
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { }
-};
-
-static const struct hda_verb alc880_lg_ch4_init[] = {
- /* set line-in to out and mic-in to input */
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { }
-};
-
-static const struct hda_verb alc880_lg_ch6_init[] = {
- /* set line-in and mic-in to output */
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { }
-};
-
-static const struct hda_channel_mode alc880_lg_ch_modes[3] = {
- { 2, alc880_lg_ch2_init },
- { 4, alc880_lg_ch4_init },
- { 6, alc880_lg_ch6_init },
-};
-
-static const struct snd_kcontrol_new alc880_lg_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc880_lg_init_verbs[] = {
- /* set capture source to mic-in */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* mute all amp mixer inputs */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* line-in to input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* built-in mic */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* speaker-out */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* mic-in to input */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* HP-out */
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* jack sense */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- { }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/*
- * LG LW20
- *
- * Pin assignment:
- * Speaker-out: 0x14
- * Mic-In: 0x18
- * Built-in Mic-In: 0x19
- * Line-In: 0x1b
- * HP-Out: 0x1a
- * SPDIF-Out: 0x1e
- */
-
-static const struct hda_input_mux alc880_lg_lw_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- { "Line In", 0x2 },
- },
-};
-
-#define alc880_lg_lw_modes alc880_threestack_modes
-
-static const struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc880_lg_lw_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
-
- /* set capture source to mic-in */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* speaker-out */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* HP-out */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* mic-in to input */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* built-in mic */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* jack sense */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- { }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_lw_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_input_mux alc880_medion_rim_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- },
-};
-
-static const struct hda_verb alc880_medion_rim_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mic2 (as headphone out) for HP output */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Internal Speaker */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- { }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc880_medion_rim_automute(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc_hp_automute(codec);
- /* toggle EAPD */
- if (spec->jack_present)
- snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
- else
- snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
-}
-
-static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 28 bit!
- */
- if ((res >> 28) == ALC880_HP_EVENT)
- alc880_medion_rim_automute(codec);
-}
-
-static void alc880_medion_rim_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x1b;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list alc880_loopbacks[] = {
- { 0x0b, HDA_INPUT, 0 },
- { 0x0b, HDA_INPUT, 1 },
- { 0x0b, HDA_INPUT, 2 },
- { 0x0b, HDA_INPUT, 3 },
- { 0x0b, HDA_INPUT, 4 },
- { } /* end */
-};
-
-static const struct hda_amp_list alc880_lg_loopbacks[] = {
- { 0x0b, HDA_INPUT, 1 },
- { 0x0b, HDA_INPUT, 6 },
- { 0x0b, HDA_INPUT, 7 },
- { } /* end */
-};
-#endif
-
-/*
* Common callbacks
*/
@@ -4196,7 +2020,7 @@
/*
* Analog playback callbacks
*/
-static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
+static int alc_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
@@ -4205,7 +2029,7 @@
hinfo);
}
-static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int alc_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
@@ -4216,7 +2040,7 @@
stream_tag, format, substream);
}
-static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int alc_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
@@ -4227,7 +2051,7 @@
/*
* Digital out
*/
-static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+static int alc_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
@@ -4235,7 +2059,7 @@
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
-static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int alc_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
@@ -4246,7 +2070,7 @@
stream_tag, format, substream);
}
-static int alc880_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int alc_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
@@ -4254,7 +2078,7 @@
return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
}
-static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+static int alc_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
@@ -4265,7 +2089,7 @@
/*
* Analog capture
*/
-static int alc880_alt_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int alc_alt_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
@@ -4278,7 +2102,7 @@
return 0;
}
-static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int alc_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
@@ -4290,21 +2114,21 @@
}
/* analog capture with dynamic dual-adc changes */
-static int dualmic_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
- spec->cur_adc = spec->adc_nids[spec->cur_adc_idx];
+ spec->cur_adc = spec->adc_nids[spec->dyn_adc_idx[spec->cur_mux[0]]];
spec->cur_adc_stream_tag = stream_tag;
spec->cur_adc_format = format;
snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
return 0;
}
-static int dualmic_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
@@ -4314,70 +2138,70 @@
return 0;
}
-static const struct hda_pcm_stream dualmic_pcm_analog_capture = {
+static const struct hda_pcm_stream dyn_adc_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
.nid = 0, /* fill later */
.ops = {
- .prepare = dualmic_capture_pcm_prepare,
- .cleanup = dualmic_capture_pcm_cleanup
+ .prepare = dyn_adc_capture_pcm_prepare,
+ .cleanup = dyn_adc_capture_pcm_cleanup
},
};
/*
*/
-static const struct hda_pcm_stream alc880_pcm_analog_playback = {
+static const struct hda_pcm_stream alc_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
/* NID is set in alc_build_pcms */
.ops = {
- .open = alc880_playback_pcm_open,
- .prepare = alc880_playback_pcm_prepare,
- .cleanup = alc880_playback_pcm_cleanup
+ .open = alc_playback_pcm_open,
+ .prepare = alc_playback_pcm_prepare,
+ .cleanup = alc_playback_pcm_cleanup
},
};
-static const struct hda_pcm_stream alc880_pcm_analog_capture = {
+static const struct hda_pcm_stream alc_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
};
-static const struct hda_pcm_stream alc880_pcm_analog_alt_playback = {
+static const struct hda_pcm_stream alc_pcm_analog_alt_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
};
-static const struct hda_pcm_stream alc880_pcm_analog_alt_capture = {
+static const struct hda_pcm_stream alc_pcm_analog_alt_capture = {
.substreams = 2, /* can be overridden */
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
- .prepare = alc880_alt_capture_pcm_prepare,
- .cleanup = alc880_alt_capture_pcm_cleanup
+ .prepare = alc_alt_capture_pcm_prepare,
+ .cleanup = alc_alt_capture_pcm_cleanup
},
};
-static const struct hda_pcm_stream alc880_pcm_digital_playback = {
+static const struct hda_pcm_stream alc_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
- .open = alc880_dig_playback_pcm_open,
- .close = alc880_dig_playback_pcm_close,
- .prepare = alc880_dig_playback_pcm_prepare,
- .cleanup = alc880_dig_playback_pcm_cleanup
+ .open = alc_dig_playback_pcm_open,
+ .close = alc_dig_playback_pcm_close,
+ .prepare = alc_dig_playback_pcm_prepare,
+ .cleanup = alc_dig_playback_pcm_cleanup
},
};
-static const struct hda_pcm_stream alc880_pcm_digital_capture = {
+static const struct hda_pcm_stream alc_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
@@ -4395,6 +2219,7 @@
{
struct alc_spec *spec = codec->spec;
struct hda_pcm *info = spec->pcm_rec;
+ const struct hda_pcm_stream *p;
int i;
codec->num_pcms = 1;
@@ -4407,16 +2232,22 @@
"%s Analog", codec->chip_name);
info->name = spec->stream_name_analog;
- if (spec->stream_analog_playback) {
- if (snd_BUG_ON(!spec->multiout.dac_nids))
- return -EINVAL;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
+ if (spec->multiout.dac_nids > 0) {
+ p = spec->stream_analog_playback;
+ if (!p)
+ p = &alc_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
}
- if (spec->stream_analog_capture) {
- if (snd_BUG_ON(!spec->adc_nids))
- return -EINVAL;
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
+ if (spec->adc_nids) {
+ p = spec->stream_analog_capture;
+ if (!p) {
+ if (spec->dyn_adc_switch)
+ p = &dyn_adc_pcm_analog_capture;
+ else
+ p = &alc_pcm_analog_capture;
+ }
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
}
@@ -4443,14 +2274,18 @@
info->pcm_type = spec->dig_out_type;
else
info->pcm_type = HDA_PCM_TYPE_SPDIF;
- if (spec->multiout.dig_out_nid &&
- spec->stream_digital_playback) {
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
+ if (spec->multiout.dig_out_nid) {
+ p = spec->stream_digital_playback;
+ if (!p)
+ p = &alc_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
}
- if (spec->dig_in_nid &&
- spec->stream_digital_capture) {
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture);
+ if (spec->dig_in_nid) {
+ p = spec->stream_digital_capture;
+ if (!p)
+ p = &alc_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
}
/* FIXME: do we need this for all Realtek codec models? */
@@ -4464,14 +2299,15 @@
* model, configure a second analog capture-only PCM.
*/
/* Additional Analaog capture for index #2 */
- if ((spec->alt_dac_nid && spec->stream_analog_alt_playback) ||
- (spec->num_adc_nids > 1 && spec->stream_analog_alt_capture)) {
+ if (spec->alt_dac_nid || spec->num_adc_nids > 1) {
codec->num_pcms = 3;
info = spec->pcm_rec + 2;
info->name = spec->stream_name_analog;
if (spec->alt_dac_nid) {
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
- *spec->stream_analog_alt_playback;
+ p = spec->stream_analog_alt_playback;
+ if (!p)
+ p = &alc_pcm_analog_alt_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
spec->alt_dac_nid;
} else {
@@ -4479,9 +2315,11 @@
alc_pcm_null_stream;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
}
- if (spec->num_adc_nids > 1 && spec->stream_analog_alt_capture) {
- info->stream[SNDRV_PCM_STREAM_CAPTURE] =
- *spec->stream_analog_alt_capture;
+ if (spec->num_adc_nids > 1) {
+ p = spec->stream_analog_alt_capture;
+ if (!p)
+ p = &alc_pcm_analog_alt_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
spec->adc_nids[1];
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
@@ -4591,679 +2429,6 @@
}
/*
- * Test configuration for debugging
- *
- * Almost all inputs/outputs are enabled. I/O pins can be configured via
- * enum controls.
- */
-#ifdef CONFIG_SND_DEBUG
-static const hda_nid_t alc880_test_dac_nids[4] = {
- 0x02, 0x03, 0x04, 0x05
-};
-
-static const struct hda_input_mux alc880_test_capture_source = {
- .num_items = 7,
- .items = {
- { "In-1", 0x0 },
- { "In-2", 0x1 },
- { "In-3", 0x2 },
- { "In-4", 0x3 },
- { "CD", 0x4 },
- { "Front", 0x5 },
- { "Surround", 0x6 },
- },
-};
-
-static const struct hda_channel_mode alc880_test_modes[4] = {
- { 2, NULL },
- { 4, NULL },
- { 6, NULL },
- { 8, NULL },
-};
-
-static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = {
- "N/A", "Line Out", "HP Out",
- "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
- };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 8;
- if (uinfo->value.enumerated.item >= 8)
- uinfo->value.enumerated.item = 7;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- unsigned int pin_ctl, item = 0;
-
- pin_ctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- if (pin_ctl & AC_PINCTL_OUT_EN) {
- if (pin_ctl & AC_PINCTL_HP_EN)
- item = 2;
- else
- item = 1;
- } else if (pin_ctl & AC_PINCTL_IN_EN) {
- switch (pin_ctl & AC_PINCTL_VREFEN) {
- case AC_PINCTL_VREF_HIZ: item = 3; break;
- case AC_PINCTL_VREF_50: item = 4; break;
- case AC_PINCTL_VREF_GRD: item = 5; break;
- case AC_PINCTL_VREF_80: item = 6; break;
- case AC_PINCTL_VREF_100: item = 7; break;
- }
- }
- ucontrol->value.enumerated.item[0] = item;
- return 0;
-}
-
-static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- static const unsigned int ctls[] = {
- 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
- };
- unsigned int old_ctl, new_ctl;
-
- old_ctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- new_ctl = ctls[ucontrol->value.enumerated.item[0]];
- if (old_ctl != new_ctl) {
- int val;
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- new_ctl);
- val = ucontrol->value.enumerated.item[0] >= 3 ?
- HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, val);
- return 1;
- }
- return 0;
-}
-
-static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = {
- "Front", "Surround", "CLFE", "Side"
- };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 4;
- if (uinfo->value.enumerated.item >= 4)
- uinfo->value.enumerated.item = 3;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- unsigned int sel;
-
- sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
- ucontrol->value.enumerated.item[0] = sel & 3;
- return 0;
-}
-
-static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- unsigned int sel;
-
- sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
- if (ucontrol->value.enumerated.item[0] != sel) {
- sel = ucontrol->value.enumerated.item[0] & 3;
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL, sel);
- return 1;
- }
- return 0;
-}
-
-#define PIN_CTL_TEST(xname,nid) { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = xname, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_test_pin_ctl_info, \
- .get = alc_test_pin_ctl_get, \
- .put = alc_test_pin_ctl_put, \
- .private_value = nid \
- }
-
-#define PIN_SRC_TEST(xname,nid) { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = xname, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_test_pin_src_info, \
- .get = alc_test_pin_src_get, \
- .put = alc_test_pin_src_put, \
- .private_value = nid \
- }
-
-static const struct snd_kcontrol_new alc880_test_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- PIN_CTL_TEST("Front Pin Mode", 0x14),
- PIN_CTL_TEST("Surround Pin Mode", 0x15),
- PIN_CTL_TEST("CLFE Pin Mode", 0x16),
- PIN_CTL_TEST("Side Pin Mode", 0x17),
- PIN_CTL_TEST("In-1 Pin Mode", 0x18),
- PIN_CTL_TEST("In-2 Pin Mode", 0x19),
- PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
- PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
- PIN_SRC_TEST("In-1 Pin Source", 0x18),
- PIN_SRC_TEST("In-2 Pin Source", 0x19),
- PIN_SRC_TEST("In-3 Pin Source", 0x1a),
- PIN_SRC_TEST("In-4 Pin Source", 0x1b),
- HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc880_test_init_verbs[] = {
- /* Unmute inputs of 0x0c - 0x0f */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Vol output for 0x0c-0x0f */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Set output pins 0x14-0x17 */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Unmute output pins 0x14-0x17 */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Set input pins 0x18-0x1c */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Mute input pins 0x18-0x1b */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* ADC set up */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Analog input/passthru */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- { }
-};
-#endif
-
-/*
- */
-
-static const char * const alc880_models[ALC880_MODEL_LAST] = {
- [ALC880_3ST] = "3stack",
- [ALC880_TCL_S700] = "tcl",
- [ALC880_3ST_DIG] = "3stack-digout",
- [ALC880_CLEVO] = "clevo",
- [ALC880_5ST] = "5stack",
- [ALC880_5ST_DIG] = "5stack-digout",
- [ALC880_W810] = "w810",
- [ALC880_Z71V] = "z71v",
- [ALC880_6ST] = "6stack",
- [ALC880_6ST_DIG] = "6stack-digout",
- [ALC880_ASUS] = "asus",
- [ALC880_ASUS_W1V] = "asus-w1v",
- [ALC880_ASUS_DIG] = "asus-dig",
- [ALC880_ASUS_DIG2] = "asus-dig2",
- [ALC880_UNIWILL_DIG] = "uniwill",
- [ALC880_UNIWILL_P53] = "uniwill-p53",
- [ALC880_FUJITSU] = "fujitsu",
- [ALC880_F1734] = "F1734",
- [ALC880_LG] = "lg",
- [ALC880_LG_LW] = "lg-lw",
- [ALC880_MEDION_RIM] = "medion",
-#ifdef CONFIG_SND_DEBUG
- [ALC880_TEST] = "test",
-#endif
- [ALC880_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc880_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810),
- SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST),
- SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
- SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
- SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V),
- /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
- SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
- SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
- SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */
- SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
- SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
- SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
- SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST),
- SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST),
- SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO),
- SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO),
- SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2),
- SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG),
- SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734),
- SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
- SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
- SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
- SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
- SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
- SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
- SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734),
- SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
- SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
- SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
- SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG),
- SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
- SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW),
- SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
- SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */
- SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
- /* default Intel */
- SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST),
- SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
- {}
-};
-
-/*
- * ALC880 codec presets
- */
-static const struct alc_config_preset alc880_presets[] = {
- [ALC880_3ST] = {
- .mixers = { alc880_three_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_3ST_DIG] = {
- .mixers = { alc880_three_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_TCL_S700] = {
- .mixers = { alc880_tcl_s700_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_tcl_S700_init_verbs,
- alc880_gpio2_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */
- .num_adc_nids = 1, /* single ADC */
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_5ST] = {
- .mixers = { alc880_three_stack_mixer,
- alc880_five_stack_mixer},
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_5stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
- .channel_mode = alc880_fivestack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_5ST_DIG] = {
- .mixers = { alc880_three_stack_mixer,
- alc880_five_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_5stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
- .channel_mode = alc880_fivestack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_6ST] = {
- .mixers = { alc880_six_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_6stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
- .dac_nids = alc880_6st_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
- .channel_mode = alc880_sixstack_modes,
- .input_mux = &alc880_6stack_capture_source,
- },
- [ALC880_6ST_DIG] = {
- .mixers = { alc880_six_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_6stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
- .dac_nids = alc880_6st_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
- .channel_mode = alc880_sixstack_modes,
- .input_mux = &alc880_6stack_capture_source,
- },
- [ALC880_W810] = {
- .mixers = { alc880_w810_base_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_w810_init_verbs,
- alc880_gpio2_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
- .dac_nids = alc880_w810_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
- .channel_mode = alc880_w810_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_Z71V] = {
- .mixers = { alc880_z71v_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_z71v_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
- .dac_nids = alc880_z71v_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_F1734] = {
- .mixers = { alc880_f1734_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_f1734_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
- .dac_nids = alc880_f1734_dac_nids,
- .hp_nid = 0x02,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_f1734_capture_source,
- .unsol_event = alc880_uniwill_p53_unsol_event,
- .setup = alc880_uniwill_p53_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC880_ASUS] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_ASUS_DIG] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_ASUS_DIG2] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio2_init_verbs }, /* use GPIO2 */
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_ASUS_W1V] = {
- .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_UNIWILL_DIG] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_UNIWILL] = {
- .mixers = { alc880_uniwill_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_uniwill_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- .unsol_event = alc880_uniwill_unsol_event,
- .setup = alc880_uniwill_setup,
- .init_hook = alc880_uniwill_init_hook,
- },
- [ALC880_UNIWILL_P53] = {
- .mixers = { alc880_uniwill_p53_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_uniwill_p53_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
- .channel_mode = alc880_threestack_modes,
- .input_mux = &alc880_capture_source,
- .unsol_event = alc880_uniwill_p53_unsol_event,
- .setup = alc880_uniwill_p53_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC880_FUJITSU] = {
- .mixers = { alc880_fujitsu_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_uniwill_p53_init_verbs,
- alc880_beep_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_capture_source,
- .unsol_event = alc880_uniwill_p53_unsol_event,
- .setup = alc880_uniwill_p53_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC880_CLEVO] = {
- .mixers = { alc880_three_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_clevo_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_LG] = {
- .mixers = { alc880_lg_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_lg_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
- .dac_nids = alc880_lg_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
- .channel_mode = alc880_lg_ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_lg_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc880_lg_setup,
- .init_hook = alc_hp_automute,
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- .loopbacks = alc880_lg_loopbacks,
-#endif
- },
- [ALC880_LG_LW] = {
- .mixers = { alc880_lg_lw_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_lg_lw_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes),
- .channel_mode = alc880_lg_lw_modes,
- .input_mux = &alc880_lg_lw_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc880_lg_lw_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC880_MEDION_RIM] = {
- .mixers = { alc880_medion_rim_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_medion_rim_init_verbs,
- alc_gpio2_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_medion_rim_capture_source,
- .unsol_event = alc880_medion_rim_unsol_event,
- .setup = alc880_medion_rim_setup,
- .init_hook = alc880_medion_rim_automute,
- },
-#ifdef CONFIG_SND_DEBUG
- [ALC880_TEST] = {
- .mixers = { alc880_test_mixer },
- .init_verbs = { alc880_test_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
- .dac_nids = alc880_test_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_test_modes),
- .channel_mode = alc880_test_modes,
- .input_mux = &alc880_test_capture_source,
- },
-#endif
-};
-
-/*
* Automatic parse of I/O pins from the BIOS configuration
*/
@@ -5272,18 +2437,12 @@
ALC_CTL_WIDGET_MUTE,
ALC_CTL_BIND_MUTE,
};
-static const struct snd_kcontrol_new alc880_control_templates[] = {
+static const struct snd_kcontrol_new alc_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
HDA_BIND_MUTE(NULL, 0, 0, 0),
};
-static struct snd_kcontrol_new *alc_kcontrol_new(struct alc_spec *spec)
-{
- snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32);
- return snd_array_new(&spec->kctls);
-}
-
/* add dynamic controls */
static int add_control(struct alc_spec *spec, int type, const char *name,
int cidx, unsigned long val)
@@ -5293,7 +2452,7 @@
knew = alc_kcontrol_new(spec);
if (!knew)
return -ENOMEM;
- *knew = alc880_control_templates[type];
+ *knew = alc_control_templates[type];
knew->name = kstrdup(name, GFP_KERNEL);
if (!knew->name)
return -ENOMEM;
@@ -5322,60 +2481,15 @@
#define __add_pb_sw_ctrl(spec, type, pfx, cidx, val) \
add_control_with_pfx(spec, type, pfx, "Playback", "Switch", cidx, val)
-#define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17)
-#define alc880_fixed_pin_idx(nid) ((nid) - 0x14)
-#define alc880_is_multi_pin(nid) ((nid) >= 0x18)
-#define alc880_multi_pin_idx(nid) ((nid) - 0x18)
-#define alc880_idx_to_dac(nid) ((nid) + 0x02)
-#define alc880_dac_to_idx(nid) ((nid) - 0x02)
-#define alc880_idx_to_mixer(nid) ((nid) + 0x0c)
-#define alc880_idx_to_selector(nid) ((nid) + 0x10)
-#define ALC880_PIN_CD_NID 0x1c
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int alc880_auto_fill_dac_nids(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- hda_nid_t nid;
- int assigned[4];
- int i, j;
-
- memset(assigned, 0, sizeof(assigned));
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- /* check the pins hardwired to audio widget */
- for (i = 0; i < cfg->line_outs; i++) {
- nid = cfg->line_out_pins[i];
- if (alc880_is_fixed_pin(nid)) {
- int idx = alc880_fixed_pin_idx(nid);
- spec->private_dac_nids[i] = alc880_idx_to_dac(idx);
- assigned[idx] = 1;
- }
- }
- /* left pins can be connect to any audio widget */
- for (i = 0; i < cfg->line_outs; i++) {
- nid = cfg->line_out_pins[i];
- if (alc880_is_fixed_pin(nid))
- continue;
- /* search for an empty channel */
- for (j = 0; j < cfg->line_outs; j++) {
- if (!assigned[j]) {
- spec->private_dac_nids[i] =
- alc880_idx_to_dac(j);
- assigned[j] = 1;
- break;
- }
- }
- }
- spec->multiout.num_dacs = cfg->line_outs;
- return 0;
-}
-
-static const char *alc_get_line_out_pfx(struct alc_spec *spec,
- bool can_be_master)
+static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
+ bool can_be_master, int *index)
{
struct auto_pin_cfg *cfg = &spec->autocfg;
+ static const char * const chname[4] = {
+ "Front", "Surround", NULL /*CLFE*/, "Side"
+ };
+ *index = 0;
if (cfg->line_outs == 1 && !spec->multi_ios &&
!cfg->hp_outs && !cfg->speaker_outs && can_be_master)
return "Master";
@@ -5386,120 +2500,17 @@
return "Speaker";
break;
case AUTO_PIN_HP_OUT:
+ /* for multi-io case, only the primary out */
+ if (ch && spec->multi_ios)
+ break;
+ *index = ch;
return "Headphone";
default:
if (cfg->line_outs == 1 && !spec->multi_ios)
return "PCM";
break;
}
- return NULL;
-}
-
-/* add playback controls from the parsed DAC table */
-static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- static const char * const chname[4] = {
- "Front", "Surround", NULL /*CLFE*/, "Side"
- };
- const char *pfx = alc_get_line_out_pfx(spec, false);
- hda_nid_t nid;
- int i, err, noutputs;
-
- noutputs = cfg->line_outs;
- if (spec->multi_ios > 0)
- noutputs += spec->multi_ios;
-
- for (i = 0; i < noutputs; i++) {
- if (!spec->multiout.dac_nids[i])
- continue;
- nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i]));
- if (!pfx && i == 2) {
- /* Center/LFE */
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
- "Center",
- HDA_COMPOSE_AMP_VAL(nid, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
- "LFE",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
- "Center",
- HDA_COMPOSE_AMP_VAL(nid, 1, 2,
- HDA_INPUT));
- if (err < 0)
- return err;
- err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
- "LFE",
- HDA_COMPOSE_AMP_VAL(nid, 2, 2,
- HDA_INPUT));
- if (err < 0)
- return err;
- } else {
- const char *name = pfx;
- int index = i;
- if (!name) {
- name = chname[i];
- index = 0;
- }
- err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
- name, index,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
- name, index,
- HDA_COMPOSE_AMP_VAL(nid, 3, 2,
- HDA_INPUT));
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-/* add playback controls for speaker and HP outputs */
-static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
- const char *pfx)
-{
- hda_nid_t nid;
- int err;
-
- if (!pin)
- return 0;
-
- if (alc880_is_fixed_pin(pin)) {
- nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
- /* specify the DAC as the extra output */
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = nid;
- else
- spec->multiout.extra_out_nid[0] = nid;
- /* control HP volume/switch on the output mixer amp */
- nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin));
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx,
- HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
- if (err < 0)
- return err;
- } else if (alc880_is_multi_pin(pin)) {
- /* set manual connection */
- /* we have only a switch on HP-out PIN */
- err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- return 0;
+ return chname[ch];
}
/* create input playback/capture controls for the given pin */
@@ -5526,17 +2537,72 @@
return (pincap & AC_PINCAP_IN) != 0;
}
-/* create playback/capture controls for input pins */
-static int alc_auto_create_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- hda_nid_t mixer,
- hda_nid_t cap1, hda_nid_t cap2)
+/* Parse the codec tree and retrieve ADCs and corresponding capsrc MUXs */
+static int alc_auto_fill_adc_caps(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
+ hda_nid_t nid;
+ hda_nid_t *adc_nids = spec->private_adc_nids;
+ hda_nid_t *cap_nids = spec->private_capsrc_nids;
+ int max_nums = ARRAY_SIZE(spec->private_adc_nids);
+ bool indep_capsrc = false;
+ int i, nums = 0;
+
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ hda_nid_t src;
+ const hda_nid_t *list;
+ unsigned int caps = get_wcaps(codec, nid);
+ int type = get_wcaps_type(caps);
+
+ if (type != AC_WID_AUD_IN || (caps & AC_WCAP_DIGITAL))
+ continue;
+ adc_nids[nums] = nid;
+ cap_nids[nums] = nid;
+ src = nid;
+ for (;;) {
+ int n;
+ type = get_wcaps_type(get_wcaps(codec, src));
+ if (type == AC_WID_PIN)
+ break;
+ if (type == AC_WID_AUD_SEL) {
+ cap_nids[nums] = src;
+ indep_capsrc = true;
+ break;
+ }
+ n = snd_hda_get_conn_list(codec, src, &list);
+ if (n > 1) {
+ cap_nids[nums] = src;
+ indep_capsrc = true;
+ break;
+ } else if (n != 1)
+ break;
+ src = *list;
+ }
+ if (++nums >= max_nums)
+ break;
+ }
+ spec->adc_nids = spec->private_adc_nids;
+ spec->capsrc_nids = spec->private_capsrc_nids;
+ spec->num_adc_nids = nums;
+ return nums;
+}
+
+/* create playback/capture controls for input pins */
+static int alc_auto_create_input_ctls(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t mixer = spec->mixer_nid;
struct hda_input_mux *imux = &spec->private_imux[0];
- int i, err, idx, type_idx = 0;
+ int num_adcs;
+ int i, c, err, idx, type_idx = 0;
const char *prev_label = NULL;
+ num_adcs = alc_auto_fill_adc_caps(codec);
+ if (num_adcs < 0)
+ return 0;
+
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t pin;
const char *label;
@@ -5563,21 +2629,22 @@
}
}
- if (!cap1)
- continue;
- idx = get_connection_index(codec, cap1, pin);
- if (idx < 0 && cap2)
- idx = get_connection_index(codec, cap2, pin);
- if (idx >= 0)
- snd_hda_add_imux_item(imux, label, idx, NULL);
+ for (c = 0; c < num_adcs; c++) {
+ hda_nid_t cap = spec->capsrc_nids ?
+ spec->capsrc_nids[c] : spec->adc_nids[c];
+ idx = get_connection_index(codec, cap, pin);
+ if (idx >= 0) {
+ spec->imux_pins[imux->num_items] = pin;
+ snd_hda_add_imux_item(imux, label, idx, NULL);
+ break;
+ }
+ }
}
- return 0;
-}
-static int alc880_auto_create_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x08, 0x09);
+ spec->num_mux_defs = 1;
+ spec->input_mux = imux;
+
+ return 0;
}
static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
@@ -5586,25 +2653,11 @@
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_type);
/* unmute pin */
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ if (nid_has_mute(codec, nid, HDA_OUTPUT))
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
}
-static void alc880_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type,
- int dac_idx)
-{
- alc_set_pin_output(codec, nid, pin_type);
- /* need the manual connection? */
- if (alc880_is_multi_pin(nid)) {
- struct alc_spec *spec = codec->spec;
- int idx = alc880_multi_pin_idx(nid);
- snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
- AC_VERB_SET_CONNECT_SEL,
- alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
- }
-}
-
static int get_pin_type(int line_out_type)
{
if (line_out_type == AUTO_PIN_HP_OUT)
@@ -5613,32 +2666,7 @@
return PIN_OUT;
}
-static void alc880_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < spec->autocfg.line_outs; i++) {
- hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- alc880_auto_set_output_and_unmute(codec, nid, pin_type, i);
- }
-}
-
-static void alc880_auto_init_extra_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
-
- pin = spec->autocfg.speaker_pins[0];
- if (pin) /* connect to front */
- alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
- pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front */
- alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
-}
-
-static void alc880_auto_init_analog_input(struct hda_codec *codec)
+static void alc_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
@@ -5648,13274 +2676,31 @@
hda_nid_t nid = cfg->inputs[i].pin;
if (alc_is_input_pin(codec, nid)) {
alc_set_input_pin(codec, nid, cfg->inputs[i].type);
- if (nid != ALC880_PIN_CD_NID &&
- (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
+ if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
}
}
-}
-static void alc880_auto_init_input_src(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int c;
-
- for (c = 0; c < spec->num_adc_nids; c++) {
- unsigned int mux_idx;
- const struct hda_input_mux *imux;
- mux_idx = c >= spec->num_mux_defs ? 0 : c;
- imux = &spec->input_mux[mux_idx];
- if (!imux->num_items && mux_idx > 0)
- imux = &spec->input_mux[0];
- if (imux)
- snd_hda_codec_write(codec, spec->adc_nids[c], 0,
- AC_VERB_SET_CONNECT_SEL,
- imux->items[0].index);
- }
-}
-
-static int alc_auto_add_multi_channel_mode(struct hda_codec *codec);
-
-/* parse the BIOS configuration and set up the alc_spec */
-/* return 1 if successful, 0 if the proper config is not found,
- * or a negative error code
- */
-static int alc880_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc880_ignore);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs)
- return 0; /* can't find valid BIOS pin config */
-
- err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc_auto_add_multi_channel_mode(codec);
- if (err < 0)
- return err;
- err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc880_auto_create_extra_out(spec,
- spec->autocfg.speaker_pins[0],
- "Speaker");
- if (err < 0)
- return err;
- err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
- "Headphone");
- if (err < 0)
- return err;
- err = alc880_auto_create_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- alc_auto_parse_digital(codec);
-
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- add_verb(spec, alc880_volume_init_verbs);
-
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
-
- return 1;
-}
-
-/* additional initialization for auto-configuration model */
-static void alc880_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc880_auto_init_multi_out(codec);
- alc880_auto_init_extra_out(codec);
- alc880_auto_init_analog_input(codec);
- alc880_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-/* check the ADC/MUX contains all input pins; some ADC/MUX contains only
- * one of two digital mic pins, e.g. on ALC272
- */
-static void fixup_automic_adc(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < spec->num_adc_nids; i++) {
- hda_nid_t cap = spec->capsrc_nids ?
- spec->capsrc_nids[i] : spec->adc_nids[i];
- int iidx, eidx;
-
- iidx = get_connection_index(codec, cap, spec->int_mic.pin);
- if (iidx < 0)
- continue;
- eidx = get_connection_index(codec, cap, spec->ext_mic.pin);
- if (eidx < 0)
- continue;
- spec->int_mic.mux_idx = iidx;
- spec->ext_mic.mux_idx = eidx;
- if (spec->capsrc_nids)
- spec->capsrc_nids += i;
- spec->adc_nids += i;
- spec->num_adc_nids = 1;
- /* optional dock-mic */
- eidx = get_connection_index(codec, cap, spec->dock_mic.pin);
- if (eidx < 0)
- spec->dock_mic.pin = 0;
- else
- spec->dock_mic.mux_idx = eidx;
- return;
- }
- snd_printd(KERN_INFO "hda_codec: %s: "
- "No ADC/MUX containing both 0x%x and 0x%x pins\n",
- codec->chip_name, spec->int_mic.pin, spec->ext_mic.pin);
- spec->auto_mic = 0; /* disable auto-mic to be sure */
-}
-
-/* select or unmute the given capsrc route */
-static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap,
- int idx)
-{
- if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
- snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
- HDA_AMP_MUTE, 0);
- } else {
- snd_hda_codec_write_cache(codec, cap, 0,
- AC_VERB_SET_CONNECT_SEL, idx);
- }
-}
-
-/* set the default connection to that pin */
-static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- if (!pin)
- return 0;
- for (i = 0; i < spec->num_adc_nids; i++) {
- hda_nid_t cap = spec->capsrc_nids ?
- spec->capsrc_nids[i] : spec->adc_nids[i];
- int idx;
-
- idx = get_connection_index(codec, cap, pin);
- if (idx < 0)
- continue;
- select_or_unmute_capsrc(codec, cap, idx);
- return i; /* return the found index */
- }
- return -1; /* not found */
-}
-
-/* choose the ADC/MUX containing the input pin and initialize the setup */
-static void fixup_single_adc(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i;
-
- /* search for the input pin; there must be only one */
- if (cfg->num_inputs != 1)
- return;
- i = init_capsrc_for_pin(codec, cfg->inputs[0].pin);
- if (i >= 0) {
- /* use only this ADC */
- if (spec->capsrc_nids)
- spec->capsrc_nids += i;
- spec->adc_nids += i;
- spec->num_adc_nids = 1;
- spec->single_input_src = 1;
- }
-}
-
-/* initialize dual adcs */
-static void fixup_dual_adc_switch(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- init_capsrc_for_pin(codec, spec->ext_mic.pin);
- init_capsrc_for_pin(codec, spec->dock_mic.pin);
- init_capsrc_for_pin(codec, spec->int_mic.pin);
-}
-
-/* initialize some special cases for input sources */
-static void alc_init_special_input_src(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- if (spec->dual_adc_switch)
- fixup_dual_adc_switch(codec);
- else if (spec->single_input_src)
- init_capsrc_for_pin(codec, spec->autocfg.inputs[0].pin);
-}
-
-static void set_capture_mixer(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- static const struct snd_kcontrol_new *caps[2][3] = {
- { alc_capture_mixer_nosrc1,
- alc_capture_mixer_nosrc2,
- alc_capture_mixer_nosrc3 },
- { alc_capture_mixer1,
- alc_capture_mixer2,
- alc_capture_mixer3 },
- };
- if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
- int mux = 0;
- int num_adcs = spec->num_adc_nids;
- if (spec->dual_adc_switch)
- num_adcs = 1;
- else if (spec->auto_mic)
- fixup_automic_adc(codec);
- else if (spec->input_mux) {
- if (spec->input_mux->num_items > 1)
- mux = 1;
- else if (spec->input_mux->num_items == 1)
- fixup_single_adc(codec);
- }
- spec->cap_mixer = caps[mux][num_adcs - 1];
- }
-}
-
-/* fill adc_nids (and capsrc_nids) containing all active input pins */
-static void fillup_priv_adc_nids(struct hda_codec *codec, const hda_nid_t *nids,
- int num_nids)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int n;
- hda_nid_t fallback_adc = 0, fallback_cap = 0;
-
- for (n = 0; n < num_nids; n++) {
- hda_nid_t adc, cap;
- hda_nid_t conn[HDA_MAX_NUM_INPUTS];
- int nconns, i, j;
-
- adc = nids[n];
- if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN)
- continue;
- cap = adc;
- nconns = snd_hda_get_connections(codec, cap, conn,
- ARRAY_SIZE(conn));
- if (nconns == 1) {
- cap = conn[0];
- nconns = snd_hda_get_connections(codec, cap, conn,
- ARRAY_SIZE(conn));
- }
- if (nconns <= 0)
- continue;
- if (!fallback_adc) {
- fallback_adc = adc;
- fallback_cap = cap;
- }
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
- for (j = 0; j < nconns; j++) {
- if (conn[j] == nid)
- break;
- }
- if (j >= nconns)
- break;
- }
- if (i >= cfg->num_inputs) {
- int num_adcs = spec->num_adc_nids;
- spec->private_adc_nids[num_adcs] = adc;
- spec->private_capsrc_nids[num_adcs] = cap;
- spec->num_adc_nids++;
- spec->adc_nids = spec->private_adc_nids;
- if (adc != cap)
- spec->capsrc_nids = spec->private_capsrc_nids;
- }
- }
- if (!spec->num_adc_nids) {
- printk(KERN_WARNING "hda_codec: %s: no valid ADC found;"
- " using fallback 0x%x\n",
- codec->chip_name, fallback_adc);
- spec->private_adc_nids[0] = fallback_adc;
- spec->adc_nids = spec->private_adc_nids;
- if (fallback_adc != fallback_cap) {
- spec->private_capsrc_nids[0] = fallback_cap;
- spec->capsrc_nids = spec->private_adc_nids;
- }
- }
-}
-
-#ifdef CONFIG_SND_HDA_INPUT_BEEP
-#define set_beep_amp(spec, nid, idx, dir) \
- ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
-
-static const struct snd_pci_quirk beep_white_list[] = {
- SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
- SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
- SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
- SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
- SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
- {}
-};
-
-static inline int has_cdefine_beep(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- const struct snd_pci_quirk *q;
- q = snd_pci_quirk_lookup(codec->bus->pci, beep_white_list);
- if (q)
- return q->value;
- return spec->cdefine.enable_pcbeep;
-}
-#else
-#define set_beep_amp(spec, nid, idx, dir) /* NOP */
-#define has_cdefine_beep(codec) 0
-#endif
-
-/*
- * OK, here we have finally the patch for ALC880
- */
-
-static int patch_alc880(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int board_config;
- int err;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- board_config = snd_hda_check_board_config(codec, ALC880_MODEL_LAST,
- alc880_models,
- alc880_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC880_AUTO;
- }
-
- if (board_config == ALC880_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc880_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using 3-stack mode...\n");
- board_config = ALC880_3ST;
- }
- }
-
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
-
- if (board_config != ALC880_AUTO)
- setup_preset(codec, &alc880_presets[board_config]);
-
- spec->stream_analog_playback = &alc880_pcm_analog_playback;
- spec->stream_analog_capture = &alc880_pcm_analog_capture;
- spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
-
- spec->stream_digital_playback = &alc880_pcm_digital_playback;
- spec->stream_digital_capture = &alc880_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- /* check whether NID 0x07 is valid */
- unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]);
- /* get type */
- wcap = get_wcaps_type(wcap);
- if (wcap != AC_WID_AUD_IN) {
- spec->adc_nids = alc880_adc_nids_alt;
- spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
- } else {
- spec->adc_nids = alc880_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
- }
- }
- set_capture_mixer(codec);
- set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
-
- spec->vmaster_nid = 0x0c;
-
- codec->patch_ops = alc_patch_ops;
- if (board_config == ALC880_AUTO)
- spec->init_hook = alc880_auto_init;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc880_loopbacks;
-#endif
-
- return 0;
-}
-
-
-/*
- * ALC260 support
- */
-
-static const hda_nid_t alc260_dac_nids[1] = {
- /* front */
- 0x02,
-};
-
-static const hda_nid_t alc260_adc_nids[1] = {
- /* ADC0 */
- 0x04,
-};
-
-static const hda_nid_t alc260_adc_nids_alt[1] = {
- /* ADC1 */
- 0x05,
-};
-
-/* NIDs used when simultaneous access to both ADCs makes sense. Note that
- * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
- */
-static const hda_nid_t alc260_dual_adc_nids[2] = {
- /* ADC0, ADC1 */
- 0x04, 0x05
-};
-
-#define ALC260_DIGOUT_NID 0x03
-#define ALC260_DIGIN_NID 0x06
-
-static const struct hda_input_mux alc260_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
- * headphone jack and the internal CD lines since these are the only pins at
- * which audio can appear. For flexibility, also allow the option of
- * recording the mixer output on the second ADC (ADC0 doesn't have a
- * connection to the mixer output).
- */
-static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
- {
- .num_items = 3,
- .items = {
- { "Mic/Line", 0x0 },
- { "CD", 0x4 },
- { "Headphone", 0x2 },
- },
- },
- {
- .num_items = 4,
- .items = {
- { "Mic/Line", 0x0 },
- { "CD", 0x4 },
- { "Headphone", 0x2 },
- { "Mixer", 0x5 },
- },
- },
-
-};
-
-/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
- * the Fujitsu S702x, but jacks are marked differently.
- */
-static const struct hda_input_mux alc260_acer_capture_sources[2] = {
- {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Headphone", 0x5 },
- },
- },
- {
- .num_items = 5,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Headphone", 0x6 },
- { "Mixer", 0x5 },
- },
- },
-};
-
-/* Maxdata Favorit 100XS */
-static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
- {
- .num_items = 2,
- .items = {
- { "Line/Mic", 0x0 },
- { "CD", 0x4 },
- },
- },
- {
- .num_items = 3,
- .items = {
- { "Line/Mic", 0x0 },
- { "CD", 0x4 },
- { "Mixer", 0x5 },
- },
- },
-};
-
-/*
- * This is just place-holder, so there's something for alc_build_pcms to look
- * at when it calculates the maximum number of channels. ALC260 has no mixer
- * element which allows changing the channel mode, so the verb list is
- * never used.
- */
-static const struct hda_channel_mode alc260_modes[1] = {
- { 2, NULL },
-};
-
-
-/* Mixer combinations
- *
- * basic: base_output + input + pc_beep + capture
- * HP: base_output + input + capture_alt
- * HP_3013: hp_3013 + input + capture
- * fujitsu: fujitsu + capture
- * acer: acer + capture
- */
-
-static const struct snd_kcontrol_new alc260_base_output_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc260_input_mixer[] = {
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
- { } /* end */
-};
-
-/* update HP, line and mono out pins according to the master switch */
-static void alc260_hp_master_update(struct hda_codec *codec)
-{
- update_speakers(codec);
-}
-
-static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- *ucontrol->value.integer.value = !spec->master_mute;
- return 0;
-}
-
-static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- int val = !*ucontrol->value.integer.value;
-
- if (val == spec->master_mute)
- return 0;
- spec->master_mute = val;
- alc260_hp_master_update(codec);
- return 1;
-}
-
-static const struct snd_kcontrol_new alc260_hp_output_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
- .info = snd_ctl_boolean_mono_info,
- .get = alc260_hp_master_sw_get,
- .put = alc260_hp_master_sw_put,
- },
- HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc260_hp_unsol_verbs[] = {
- {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {},
-};
-
-static void alc260_hp_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x0f;
- spec->autocfg.speaker_pins[0] = 0x10;
- spec->autocfg.speaker_pins[1] = 0x11;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
- .info = snd_ctl_boolean_mono_info,
- .get = alc260_hp_master_sw_get,
- .put = alc260_hp_master_sw_put,
- },
- HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static void alc260_hp_3013_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x10;
- spec->autocfg.speaker_pins[1] = 0x11;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls alc260_dc7600_bind_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol),
- HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc260_hp_3013_unsol_verbs[] = {
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {},
-};
-
-static void alc260_hp_3012_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x10;
- spec->autocfg.speaker_pins[0] = 0x0f;
- spec->autocfg.speaker_pins[1] = 0x11;
- spec->autocfg.speaker_pins[2] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
- * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
- */
-static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
- { } /* end */
-};
-
-/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
- * versions of the ALC260 don't act on requests to enable mic bias from NID
- * 0x0f (used to drive the headphone jack in these laptops). The ALC260
- * datasheet doesn't mention this restriction. At this stage it's not clear
- * whether this behaviour is intentional or is a hardware bug in chip
- * revisions available in early 2006. Therefore for now allow the
- * "Headphone Jack Mode" control to span all choices, but if it turns out
- * that the lack of mic bias for this NID is intentional we could change the
- * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
- *
- * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
- * don't appear to make the mic bias available from the "line" jack, even
- * though the NID used for this jack (0x14) can supply it. The theory is
- * that perhaps Acer have included blocking capacitors between the ALC260
- * and the output jack. If this turns out to be the case for all such
- * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
- * to ALC_PIN_DIR_INOUT_NOMICBIAS.
- *
- * The C20x Tablet series have a mono internal speaker which is controlled
- * via the chip's Mono sum widget and pin complex, so include the necessary
- * controls for such models. On models without a "mono speaker" the control
- * won't do anything.
- */
-static const struct snd_kcontrol_new alc260_acer_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
- HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- { } /* end */
-};
-
-/* Maxdata Favorit 100XS: one output and one input (0x12) jack
- */
-static const struct snd_kcontrol_new alc260_favorit100_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- { } /* end */
-};
-
-/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
- * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
- */
-static const struct snd_kcontrol_new alc260_will_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- { } /* end */
-};
-
-/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
- * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
- */
-static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb alc260_init_verbs[] = {
- /* Line In pin widget for input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* CD pin widget for input */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* LINE-2 is used for line-out in rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* select line-out */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LINE-OUT pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* enable HP */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* enable Mono */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* set connection select to line in (default select for this ADC) */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* mute capture amp left and right */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* set connection select to line in (default select for this ADC) */
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* set vol=0 Line-Out mixer amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* set vol=0 HP mixer amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* set vol=0 Mono mixer amp left and right */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* unmute LINE-2 out pin */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
- * Line In 2 = 0x03
- */
- /* mute analog inputs */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* mute Front out path */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* mute Headphone out path */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* mute Mono out path */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { }
-};
-
-#if 0 /* should be identical with alc260_init_verbs? */
-static const struct hda_verb alc260_hp_init_verbs[] = {
- /* Headphone and output */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- /* mono output */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Line In pin widget for input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- /* Line-2 pin widget for output */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* CD pin widget for input */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- /* unmute amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
- /* set connection select to line in (default select for this ADC) */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* unmute Line-Out mixer amp left and right (volume = 0) */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* unmute HP mixer amp left and right (volume = 0) */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
- * Line In 2 = 0x03
- */
- /* mute analog inputs */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* Unmute Front out path */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Headphone out path */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Mono out path */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- { }
-};
-#endif
-
-static const struct hda_verb alc260_hp_3013_init_verbs[] = {
- /* Line out and output */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* mono output */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Line In pin widget for input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- /* Headphone pin widget for output */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- /* CD pin widget for input */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- /* unmute amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
- /* set connection select to line in (default select for this ADC) */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* unmute Line-Out mixer amp left and right (volume = 0) */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* unmute HP mixer amp left and right (volume = 0) */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
- * Line In 2 = 0x03
- */
- /* mute analog inputs */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* Unmute Front out path */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Headphone out path */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Mono out path */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- { }
-};
-
-/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
- * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
- * audio = 0x16, internal speaker = 0x10.
- */
-static const struct hda_verb alc260_fujitsu_init_verbs[] = {
- /* Disable all GPIOs */
- {0x01, AC_VERB_SET_GPIO_MASK, 0},
- /* Internal speaker is connected to headphone pin */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Headphone/Line-out jack connects to Line1 pin; make it an output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Mic/Line-in jack is connected to mic1 pin, so make it an input */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Ensure all other unused pins are disabled and muted. */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-
- /* Disable digital (SPDIF) pins */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure Line1 pin widget takes its input from the OUT1 sum bus
- * when acting as an output.
- */
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Line1 pin widget output buffer since it starts as an output.
- * If the pin mode is changed by the user the pin mode control will
- * take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute input buffer of pin widget used for Line-in (no equiv
- * mixer ctrl)
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting - line
- * in (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do the same for the second ADC: mute capture input amp and
- * set ADC connection to line in (on mic1 pin)
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-
-/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
- * similar laptops (adapted from Fujitsu init verbs).
- */
-static const struct hda_verb alc260_acer_init_verbs[] = {
- /* On TravelMate laptops, GPIO 0 enables the internal speaker and
- * the headphone jack. Turn this on and rely on the standard mute
- * methods whenever the user wants to turn these outputs off.
- */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
- /* Internal speaker/Headphone jack is connected to Line-out pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Internal microphone/Mic jack is connected to Mic1 pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- /* Line In jack is connected to Line1 pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Ensure all other unused pins are disabled and muted. */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Disable digital (SPDIF) pins */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
- * bus when acting as outputs.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute Line-out pin widget amp left and right
- * (no equiv mixer ctrl)
- */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute mono pin widget amp output (no equiv mixer ctrl) */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Mic1 and Line1 pin widget input buffers since they start as
- * inputs. If the pin mode is changed by the user the pin mode control
- * will take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting - mic
- * (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do similar with the second ADC: mute capture input amp and
- * set ADC connection to mic to match ALSA's default state.
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-
-/* Initialisation sequence for Maxdata Favorit 100XS
- * (adapted from Acer init verbs).
- */
-static const struct hda_verb alc260_favorit100_init_verbs[] = {
- /* GPIO 0 enables the output jack.
- * Turn this on and rely on the standard mute
- * methods whenever the user wants to turn these outputs off.
- */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
- /* Line/Mic input jack is connected to Mic1 pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- /* Ensure all other unused pins are disabled and muted. */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Disable digital (SPDIF) pins */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
- * bus when acting as outputs.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute Line-out pin widget amp left and right
- * (no equiv mixer ctrl)
- */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Mic1 and Line1 pin widget input buffers since they start as
- * inputs. If the pin mode is changed by the user the pin mode control
- * will take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting - mic
- * (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do similar with the second ADC: mute capture input amp and
- * set ADC connection to mic to match ALSA's default state.
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-
-static const struct hda_verb alc260_will_verbs[] = {
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
- {}
-};
-
-static const struct hda_verb alc260_replacer_672v_verbs[] = {
- {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
-
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
-
- {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc260_replacer_672v_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
- present = snd_hda_jack_detect(codec, 0x0f);
- if (present) {
- snd_hda_codec_write_cache(codec, 0x01, 0,
- AC_VERB_SET_GPIO_DATA, 1);
- snd_hda_codec_write_cache(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_HP);
- } else {
- snd_hda_codec_write_cache(codec, 0x01, 0,
- AC_VERB_SET_GPIO_DATA, 0);
- snd_hda_codec_write_cache(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
- }
-}
-
-static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc260_replacer_672v_automute(codec);
-}
-
-static const struct hda_verb alc260_hp_dc7600_verbs[] = {
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-/* Test configuration for debugging, modelled after the ALC880 test
- * configuration.
- */
-#ifdef CONFIG_SND_DEBUG
-static const hda_nid_t alc260_test_dac_nids[1] = {
- 0x02,
-};
-static const hda_nid_t alc260_test_adc_nids[2] = {
- 0x04, 0x05,
-};
-/* For testing the ALC260, each input MUX needs its own definition since
- * the signal assignments are different. This assumes that the first ADC
- * is NID 0x04.
- */
-static const struct hda_input_mux alc260_test_capture_sources[2] = {
- {
- .num_items = 7,
- .items = {
- { "MIC1 pin", 0x0 },
- { "MIC2 pin", 0x1 },
- { "LINE1 pin", 0x2 },
- { "LINE2 pin", 0x3 },
- { "CD pin", 0x4 },
- { "LINE-OUT pin", 0x5 },
- { "HP-OUT pin", 0x6 },
- },
- },
- {
- .num_items = 8,
- .items = {
- { "MIC1 pin", 0x0 },
- { "MIC2 pin", 0x1 },
- { "LINE1 pin", 0x2 },
- { "LINE2 pin", 0x3 },
- { "CD pin", 0x4 },
- { "Mixer", 0x5 },
- { "LINE-OUT pin", 0x6 },
- { "HP-OUT pin", 0x7 },
- },
- },
-};
-static const struct snd_kcontrol_new alc260_test_mixer[] = {
- /* Output driver widgets */
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
-
- /* Modes for retasking pin widgets
- * Note: the ALC260 doesn't seem to act on requests to enable mic
- * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
- * mention this restriction. At this stage it's not clear whether
- * this behaviour is intentional or is a hardware bug in chip
- * revisions available at least up until early 2006. Therefore for
- * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
- * choices, but if it turns out that the lack of mic bias for these
- * NIDs is intentional we could change their modes from
- * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
- */
- ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
-
- /* Loopback mixer controls */
- HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
- HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
-
- /* Controls for GPIO pins, assuming they are configured as outputs */
- ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
- ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
- ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
- ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
-
- /* Switches to allow the digital IO pins to be enabled. The datasheet
- * is ambigious as to which NID is which; testing on laptops which
- * make this output available should provide clarification.
- */
- ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
- ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
-
- /* A switch allowing EAPD to be enabled. Some laptops seem to use
- * this output to turn on an external amplifier.
- */
- ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
- ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
-
- { } /* end */
-};
-static const struct hda_verb alc260_test_init_verbs[] = {
- /* Enable all GPIOs as outputs with an initial value of 0 */
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
- {0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
-
- /* Enable retasking pins as output, initially without power amp */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* Disable digital (SPDIF) pins initially, but users can enable
- * them via a mixer switch. In the case of SPDIF-out, this initverb
- * payload also sets the generation to 0, output to be in "consumer"
- * PCM format, copyright asserted, no pre-emphasis and no validity
- * control.
- */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
- * OUT1 sum bus when acting as an output.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute retasking pin widget output buffers since the default
- * state appears to be output. As the pin mode is changed by the
- * user the pin mode control will take care of enabling the pin's
- * input/output buffers as needed.
- */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Also unmute the mono-out pin widget */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting (mic1
- * pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do the same for the second ADC: mute capture input amp and
- * set ADC connection to mic1 pin
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mute all inputs to mixer widget (even unconnected ones) */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
-
- { }
-};
-#endif
-
-#define alc260_pcm_analog_playback alc880_pcm_analog_alt_playback
-#define alc260_pcm_analog_capture alc880_pcm_analog_capture
-
-#define alc260_pcm_digital_playback alc880_pcm_digital_playback
-#define alc260_pcm_digital_capture alc880_pcm_digital_capture
-
-/*
- * for BIOS auto-configuration
- */
-
-static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
- const char *pfx, int *vol_bits)
-{
- hda_nid_t nid_vol;
- unsigned long vol_val, sw_val;
- int err;
-
- if (nid >= 0x0f && nid < 0x11) {
- nid_vol = nid - 0x7;
- vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
- sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
- } else if (nid == 0x11) {
- nid_vol = nid - 0x7;
- vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT);
- sw_val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
- } else if (nid >= 0x12 && nid <= 0x15) {
- nid_vol = 0x08;
- vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
- sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
- } else
- return 0; /* N/A */
-
- if (!(*vol_bits & (1 << nid_vol))) {
- /* first control for the volume widget */
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, vol_val);
- if (err < 0)
- return err;
- *vol_bits |= (1 << nid_vol);
- }
- err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, sw_val);
- if (err < 0)
- return err;
- return 1;
-}
-
-/* add playback controls from the parsed DAC table */
-static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- hda_nid_t nid;
- int err;
- int vols = 0;
-
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = spec->private_dac_nids;
- spec->private_dac_nids[0] = 0x02;
-
- nid = cfg->line_out_pins[0];
- if (nid) {
- const char *pfx;
- if (!cfg->speaker_pins[0] && !cfg->hp_pins[0])
- pfx = "Master";
- else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
- pfx = "Speaker";
- else
- pfx = "Front";
- err = alc260_add_playback_controls(spec, nid, pfx, &vols);
- if (err < 0)
- return err;
- }
-
- nid = cfg->speaker_pins[0];
- if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
- if (err < 0)
- return err;
- }
-
- nid = cfg->hp_pins[0];
- if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Headphone",
- &vols);
- if (err < 0)
- return err;
- }
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int alc260_auto_create_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- return alc_auto_create_input_ctls(codec, cfg, 0x07, 0x04, 0x05);
-}
-
-static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type,
- int sel_idx)
-{
- alc_set_pin_output(codec, nid, pin_type);
- /* need the manual connection? */
- if (nid >= 0x12) {
- int idx = nid - 0x12;
- snd_hda_codec_write(codec, idx + 0x0b, 0,
- AC_VERB_SET_CONNECT_SEL, sel_idx);
- }
-}
-
-static void alc260_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t nid;
-
- nid = spec->autocfg.line_out_pins[0];
- if (nid) {
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- alc260_auto_set_output_and_unmute(codec, nid, pin_type, 0);
- }
-
- nid = spec->autocfg.speaker_pins[0];
- if (nid)
- alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
-
- nid = spec->autocfg.hp_pins[0];
- if (nid)
- alc260_auto_set_output_and_unmute(codec, nid, PIN_HP, 0);
-}
-
-#define ALC260_PIN_CD_NID 0x16
-static void alc260_auto_init_analog_input(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
- if (nid >= 0x12) {
- alc_set_input_pin(codec, nid, cfg->inputs[i].type);
- if (nid != ALC260_PIN_CD_NID &&
- (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
- }
- }
-}
-
-#define alc260_auto_init_input_src alc880_auto_init_input_src
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc260_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for
- * front panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- /* mute analog inputs */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /*
- * Set up output mixers (0x08 - 0x0a)
- */
- /* set vol=0 to output mixers */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- { }
-};
-
-static int alc260_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc260_ignore[] = { 0x17, 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc260_ignore);
- if (err < 0)
- return err;
- err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (!spec->kctls.list)
- return 0; /* can't find valid BIOS pin config */
- err = alc260_auto_create_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = 2;
-
- if (spec->autocfg.dig_outs)
- spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- add_verb(spec, alc260_volume_init_verbs);
-
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- alc_ssid_check(codec, 0x10, 0x15, 0x0f, 0);
-
- return 1;
-}
-
-/* additional initialization for auto-configuration model */
-static void alc260_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc260_auto_init_multi_out(codec);
- alc260_auto_init_analog_input(codec);
- alc260_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list alc260_loopbacks[] = {
- { 0x07, HDA_INPUT, 0 },
- { 0x07, HDA_INPUT, 1 },
- { 0x07, HDA_INPUT, 2 },
- { 0x07, HDA_INPUT, 3 },
- { 0x07, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
-/*
- * Pin config fixes
- */
-enum {
- PINFIX_HP_DC5750,
-};
-
-static const struct alc_fixup alc260_fixups[] = {
- [PINFIX_HP_DC5750] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x11, 0x90130110 }, /* speaker */
- { }
- }
- },
-};
-
-static const struct snd_pci_quirk alc260_fixup_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750),
- {}
-};
-
-/*
- * ALC260 configurations
- */
-static const char * const alc260_models[ALC260_MODEL_LAST] = {
- [ALC260_BASIC] = "basic",
- [ALC260_HP] = "hp",
- [ALC260_HP_3013] = "hp-3013",
- [ALC260_HP_DC7600] = "hp-dc7600",
- [ALC260_FUJITSU_S702X] = "fujitsu",
- [ALC260_ACER] = "acer",
- [ALC260_WILL] = "will",
- [ALC260_REPLACER_672V] = "replacer",
- [ALC260_FAVORIT100] = "favorit100",
-#ifdef CONFIG_SND_DEBUG
- [ALC260_TEST] = "test",
-#endif
- [ALC260_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc260_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
- SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
- SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
- SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
- SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
- SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
- SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
- SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
- SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
- SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
- SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
- SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
- SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
- SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
- SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
- SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
- SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
- {}
-};
-
-static const struct alc_config_preset alc260_presets[] = {
- [ALC260_BASIC] = {
- .mixers = { alc260_base_output_mixer,
- alc260_input_mixer },
- .init_verbs = { alc260_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- },
- [ALC260_HP] = {
- .mixers = { alc260_hp_output_mixer,
- alc260_input_mixer },
- .init_verbs = { alc260_init_verbs,
- alc260_hp_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
- .adc_nids = alc260_adc_nids_alt,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc260_hp_setup,
- .init_hook = alc_inithook,
- },
- [ALC260_HP_DC7600] = {
- .mixers = { alc260_hp_dc7600_mixer,
- alc260_input_mixer },
- .init_verbs = { alc260_init_verbs,
- alc260_hp_dc7600_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
- .adc_nids = alc260_adc_nids_alt,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc260_hp_3012_setup,
- .init_hook = alc_inithook,
- },
- [ALC260_HP_3013] = {
- .mixers = { alc260_hp_3013_mixer,
- alc260_input_mixer },
- .init_verbs = { alc260_hp_3013_init_verbs,
- alc260_hp_3013_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
- .adc_nids = alc260_adc_nids_alt,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc260_hp_3013_setup,
- .init_hook = alc_inithook,
- },
- [ALC260_FUJITSU_S702X] = {
- .mixers = { alc260_fujitsu_mixer },
- .init_verbs = { alc260_fujitsu_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
- .input_mux = alc260_fujitsu_capture_sources,
- },
- [ALC260_ACER] = {
- .mixers = { alc260_acer_mixer },
- .init_verbs = { alc260_acer_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
- .input_mux = alc260_acer_capture_sources,
- },
- [ALC260_FAVORIT100] = {
- .mixers = { alc260_favorit100_mixer },
- .init_verbs = { alc260_favorit100_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
- .input_mux = alc260_favorit100_capture_sources,
- },
- [ALC260_WILL] = {
- .mixers = { alc260_will_mixer },
- .init_verbs = { alc260_init_verbs, alc260_will_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
- .adc_nids = alc260_adc_nids,
- .dig_out_nid = ALC260_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- },
- [ALC260_REPLACER_672V] = {
- .mixers = { alc260_replacer_672v_mixer },
- .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
- .adc_nids = alc260_adc_nids,
- .dig_out_nid = ALC260_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- .unsol_event = alc260_replacer_672v_unsol_event,
- .init_hook = alc260_replacer_672v_automute,
- },
-#ifdef CONFIG_SND_DEBUG
- [ALC260_TEST] = {
- .mixers = { alc260_test_mixer },
- .init_verbs = { alc260_test_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
- .dac_nids = alc260_test_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
- .adc_nids = alc260_test_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
- .input_mux = alc260_test_capture_sources,
- },
-#endif
-};
-
-static int patch_alc260(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int err, board_config;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- board_config = snd_hda_check_board_config(codec, ALC260_MODEL_LAST,
- alc260_models,
- alc260_cfg_tbl);
- if (board_config < 0) {
- snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC260_AUTO;
- }
-
- if (board_config == ALC260_AUTO) {
- alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC260_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc260_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC260_BASIC;
- }
- }
-
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
-
- if (board_config != ALC260_AUTO)
- setup_preset(codec, &alc260_presets[board_config]);
-
- spec->stream_analog_playback = &alc260_pcm_analog_playback;
- spec->stream_analog_capture = &alc260_pcm_analog_capture;
- spec->stream_analog_alt_capture = &alc260_pcm_analog_capture;
-
- spec->stream_digital_playback = &alc260_pcm_digital_playback;
- spec->stream_digital_capture = &alc260_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- /* check whether NID 0x04 is valid */
- unsigned int wcap = get_wcaps(codec, 0x04);
- wcap = get_wcaps_type(wcap);
- /* get type */
- if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
- spec->adc_nids = alc260_adc_nids_alt;
- spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
- } else {
- spec->adc_nids = alc260_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
- }
- }
- set_capture_mixer(codec);
- set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
-
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
- spec->vmaster_nid = 0x08;
-
- codec->patch_ops = alc_patch_ops;
- if (board_config == ALC260_AUTO)
- spec->init_hook = alc260_auto_init;
- spec->shutup = alc_eapd_shutup;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc260_loopbacks;
-#endif
-
- return 0;
-}
-
-
-/*
- * ALC882/883/885/888/889 support
- *
- * ALC882 is almost identical with ALC880 but has cleaner and more flexible
- * configuration. Each pin widget can choose any input DACs and a mixer.
- * Each ADC is connected from a mixer of all inputs. This makes possible
- * 6-channel independent captures.
- *
- * In addition, an independent DAC for the multi-playback (not used in this
- * driver yet).
- */
-#define ALC882_DIGOUT_NID 0x06
-#define ALC882_DIGIN_NID 0x0a
-#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID
-#define ALC883_DIGIN_NID ALC882_DIGIN_NID
-#define ALC1200_DIGOUT_NID 0x10
-
-
-static const struct hda_channel_mode alc882_ch_modes[1] = {
- { 8, NULL }
-};
-
-/* DACs */
-static const hda_nid_t alc882_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x03, 0x04, 0x05
-};
-#define alc883_dac_nids alc882_dac_nids
-
-/* ADCs */
-#define alc882_adc_nids alc880_adc_nids
-#define alc882_adc_nids_alt alc880_adc_nids_alt
-#define alc883_adc_nids alc882_adc_nids_alt
-static const hda_nid_t alc883_adc_nids_alt[1] = { 0x08 };
-static const hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 };
-#define alc889_adc_nids alc880_adc_nids
-
-static const hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
-static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
-#define alc883_capsrc_nids alc882_capsrc_nids_alt
-static const hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
-#define alc889_capsrc_nids alc882_capsrc_nids
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-
-static const struct hda_input_mux alc882_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-#define alc883_capture_source alc882_capture_source
-
-static const struct hda_input_mux alc889_capture_source = {
- .num_items = 3,
- .items = {
- { "Front Mic", 0x0 },
- { "Mic", 0x3 },
- { "Line", 0x2 },
- },
-};
-
-static const struct hda_input_mux mb5_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x1 },
- { "Line", 0x7 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux macmini3_capture_source = {
- .num_items = 2,
- .items = {
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc883_3stack_6ch_intel = {
- .num_items = 4,
- .items = {
- { "Mic", 0x1 },
- { "Front Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc883_lenovo_101e_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 },
- { "Line", 0x2 },
- },
-};
-
-static const struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- },
-};
-
-static const struct hda_input_mux alc883_lenovo_sky_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc883_asus_eee1601_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- },
-};
-
-static const struct hda_input_mux alc889A_mb31_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- /* Front Mic (0x01) unused */
- { "Line", 0x2 },
- /* Line 2 (0x03) unused */
- /* CD (0x04) unused? */
- },
-};
-
-static const struct hda_input_mux alc889A_imac91_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x01 },
- { "Line", 0x2 }, /* Not sure! */
- },
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
- { 2, NULL }
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc882_3ST_ch2_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc882_3ST_ch4_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc882_3ST_ch6_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
- { 2, alc882_3ST_ch2_init },
- { 4, alc882_3ST_ch4_init },
- { 6, alc882_3ST_ch6_init },
-};
-
-#define alc883_3ST_6ch_modes alc882_3ST_6ch_modes
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc883_3ST_ch2_clevo_init[] = {
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc883_3ST_ch4_clevo_init[] = {
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc883_3ST_ch6_clevo_init[] = {
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = {
- { 2, alc883_3ST_ch2_clevo_init },
- { 4, alc883_3ST_ch4_clevo_init },
- { 6, alc883_3ST_ch6_clevo_init },
-};
-
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc882_sixstack_ch6_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static const struct hda_verb alc882_sixstack_ch8_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc882_sixstack_modes[2] = {
- { 6, alc882_sixstack_ch6_init },
- { 8, alc882_sixstack_ch8_init },
-};
-
-
-/* Macbook Air 2,1 */
-
-static const struct hda_channel_mode alc885_mba21_ch_modes[1] = {
- { 2, NULL },
-};
-
-/*
- * macbook pro ALC885 can switch LineIn to LineOut without losing Mic
- */
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc885_mbp_ch2_init[] = {
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc885_mbp_ch4_init[] = {
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- { } /* end */
-};
-
-static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
- { 2, alc885_mbp_ch2_init },
- { 4, alc885_mbp_ch4_init },
-};
-
-/*
- * 2ch
- * Speakers/Woofer/HP = Front
- * LineIn = Input
- */
-static const struct hda_verb alc885_mb5_ch2_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- { } /* end */
-};
-
-/*
- * 6ch mode
- * Speakers/HP = Front
- * Woofer = LFE
- * LineIn = Surround
- */
-static const struct hda_verb alc885_mb5_ch6_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- { } /* end */
-};
-
-static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
- { 2, alc885_mb5_ch2_init },
- { 6, alc885_mb5_ch6_init },
-};
-
-#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc883_4ST_ch2_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc883_4ST_ch4_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc883_4ST_ch6_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static const struct hda_verb alc883_4ST_ch8_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
- { 2, alc883_4ST_ch2_init },
- { 4, alc883_4ST_ch4_init },
- { 6, alc883_4ST_ch6_init },
- { 8, alc883_4ST_ch8_init },
-};
-
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc883_3ST_ch2_intel_init[] = {
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc883_3ST_ch4_intel_init[] = {
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc883_3ST_ch6_intel_init[] = {
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
- { 2, alc883_3ST_ch2_intel_init },
- { 4, alc883_3ST_ch4_intel_init },
- { 6, alc883_3ST_ch6_intel_init },
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc889_ch2_intel_init[] = {
- { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc889_ch6_intel_init[] = {
- { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static const struct hda_verb alc889_ch8_intel_init[] = {
- { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc889_8ch_intel_modes[3] = {
- { 2, alc889_ch2_intel_init },
- { 6, alc889_ch6_intel_init },
- { 8, alc889_ch8_intel_init },
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc883_sixstack_ch6_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static const struct hda_verb alc883_sixstack_ch8_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc883_sixstack_modes[2] = {
- { 6, alc883_sixstack_ch6_init },
- { 8, alc883_sixstack_ch8_init },
-};
-
-
-/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
- * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
- */
-static const struct snd_kcontrol_new alc882_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-/* Macbook Air 2,1 same control for HP and internal Speaker */
-
-static const struct snd_kcontrol_new alc885_mba21_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT),
- { }
-};
-
-
-static const struct snd_kcontrol_new alc885_mbp3_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc885_mb5_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc885_macmini3_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc885_imac91_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
- { } /* end */
-};
-
-
-static const struct snd_kcontrol_new alc882_w2jc_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc882_targa_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { } /* end */
-};
-
-/* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ???
- * Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c
- */
-static const struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Mobile Front Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mobile Line Playback Volume", 0x0b, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc882_chmode_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc882_base_init_verbs[] = {
- /* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* CLFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Side mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- /* Front Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Rear Pin: output 1 (0x0d) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* CLFE Pin: output 2 (0x0e) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* Side Pin: output 3 (0x0f) */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line-2 In: Headphone output (output 0 - 0x0c) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* ADC2: mute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC3: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- { }
-};
-
-static const struct hda_verb alc882_adc1_init_verbs[] = {
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* ADC1: mute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- { }
-};
-
-static const struct hda_verb alc882_eapd_verbs[] = {
- /* change to EAPD mode */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
- { }
-};
-
-static const struct hda_verb alc889_eapd_verbs[] = {
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-static const struct hda_verb alc_hp15_unsol_verbs[] = {
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {}
-};
-
-static const struct hda_verb alc885_init_verbs[] = {
- /* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* CLFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Side mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Front HP Pin: output 0 (0x0c) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Front Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Rear Pin: output 1 (0x0d) */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x19, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* CLFE Pin: output 2 (0x0e) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* Side Pin: output 3 (0x0f) */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic (rear) pin: input vref at 80% */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- /* Mixer elements: 0x18, , 0x1a, 0x1b */
- /* Input mixer1 */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* ADC2: mute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* ADC3: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-
- { }
-};
-
-static const struct hda_verb alc885_init_input_verbs[] = {
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- { }
-};
-
-
-/* Unmute Selector 24h and set the default input to front mic */
-static const struct hda_verb alc889_init_input_verbs[] = {
- {0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- { }
-};
-
-
-#define alc883_init_verbs alc882_base_init_verbs
-
-/* Mac Pro test */
-static const struct snd_kcontrol_new alc882_macpro_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
- /* FIXME: this looks suspicious...
- HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT),
- */
- { } /* end */
-};
-
-static const struct hda_verb alc882_macpro_init_verbs[] = {
- /* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Front Pin: output 0 (0x0c) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Speaker: output */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x04},
- /* Headphone output (output 0 - 0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* ADC1: mute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC2: mute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC3: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- { }
-};
-
-/* Macbook 5,1 */
-static const struct hda_verb alc885_mb5_init_verbs[] = {
- /* DACs */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Front mixer */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Surround mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* LFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* HP mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Front Pin (0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LFE Pin (0x0e) */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* HP Pin (0x0f) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
- { }
-};
-
-/* Macmini 3,1 */
-static const struct hda_verb alc885_macmini3_init_verbs[] = {
- /* DACs */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Front mixer */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Surround mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* LFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* HP mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Front Pin (0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LFE Pin (0x0e) */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* HP Pin (0x0f) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- /* Line In pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- { }
-};
-
-
-static const struct hda_verb alc885_mba21_init_verbs[] = {
- /*Internal and HP Speaker Mixer*/
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /*Internal Speaker Pin (0x0c)*/
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP Pin: output 0 (0x0e) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)},
- /* Line in (is hp when jack connected)*/
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- { }
- };
-
-
-/* Macbook Pro rev3 */
-static const struct hda_verb alc885_mbp3_init_verbs[] = {
- /* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* HP mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Front Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP Pin: output 0 (0x0e) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: use output 1 when in LineOut mode */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* ADC1: mute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC2: mute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC3: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- { }
-};
-
-/* iMac 9,1 */
-static const struct hda_verb alc885_imac91_init_verbs[] = {
- /* Internal Speaker Pin (0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP Pin: Rear */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)},
- /* Line in Rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- { }
-};
-
-/* iMac 24 mixer. */
-static const struct snd_kcontrol_new alc885_imac24_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT),
- { } /* end */
-};
-
-/* iMac 24 init verbs. */
-static const struct hda_verb alc885_imac24_init_verbs[] = {
- /* Internal speakers: output 0 (0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Internal speakers: output 0 (0x0c) */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Headphone: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- /* Front Mic: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- { }
-};
-
-/* Toggle speaker-output according to the hp-jack state */
-static void alc885_imac24_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x18;
- spec->autocfg.speaker_pins[1] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-#define alc885_mb5_setup alc885_imac24_setup
-#define alc885_macmini3_setup alc885_imac24_setup
-
-/* Macbook Air 2,1 */
-static void alc885_mba21_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x18;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-
-
-static void alc885_mbp3_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc885_imac91_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x18;
- spec->autocfg.speaker_pins[1] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct hda_verb alc882_targa_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc882_targa_automute(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc_hp_automute(codec);
- snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
- spec->jack_present ? 1 : 3);
-}
-
-static void alc882_targa_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x1b;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc882_targa_automute(codec);
-}
-
-static const struct hda_verb alc882_asus_a7j_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
-
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- { } /* end */
-};
-
-static const struct hda_verb alc882_asus_a7m_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
-
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- { } /* end */
-};
-
-static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
-{
- unsigned int gpiostate, gpiomask, gpiodir;
-
- gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
- AC_VERB_GET_GPIO_DATA, 0);
-
- if (!muted)
- gpiostate |= (1 << pin);
- else
- gpiostate &= ~(1 << pin);
-
- gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
- AC_VERB_GET_GPIO_MASK, 0);
- gpiomask |= (1 << pin);
-
- gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
- AC_VERB_GET_GPIO_DIRECTION, 0);
- gpiodir |= (1 << pin);
-
-
- snd_hda_codec_write(codec, codec->afg, 0,
- AC_VERB_SET_GPIO_MASK, gpiomask);
- snd_hda_codec_write(codec, codec->afg, 0,
- AC_VERB_SET_GPIO_DIRECTION, gpiodir);
-
- msleep(1);
-
- snd_hda_codec_write(codec, codec->afg, 0,
- AC_VERB_SET_GPIO_DATA, gpiostate);
-}
-
-/* set up GPIO at initialization */
-static void alc885_macpro_init_hook(struct hda_codec *codec)
-{
- alc882_gpio_mute(codec, 0, 0);
- alc882_gpio_mute(codec, 1, 0);
-}
-
-/* set up GPIO and update auto-muting at initialization */
-static void alc885_imac24_init_hook(struct hda_codec *codec)
-{
- alc885_macpro_init_hook(codec);
- alc_hp_automute(codec);
-}
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc883_auto_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /*
- * Set up output mixers (0x0c - 0x0f)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- { }
-};
-
-/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
-static const struct hda_verb alc889A_mb31_ch2_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
- { } /* end */
-};
-
-/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
-static const struct hda_verb alc889A_mb31_ch4_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
- { } /* end */
-};
-
-/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
-static const struct hda_verb alc889A_mb31_ch5_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
- { } /* end */
-};
-
-/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
-static const struct hda_verb alc889A_mb31_ch6_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
- { } /* end */
-};
-
-static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
- { 2, alc889A_mb31_ch2_init },
- { 4, alc889A_mb31_ch4_init },
- { 5, alc889A_mb31_ch5_init },
- { 6, alc889A_mb31_ch6_init },
-};
-
-static const struct hda_verb alc883_medion_eapd_verbs[] = {
- /* eanable EAPD on medion laptop */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
- { }
-};
-
-#define alc883_base_mixer alc882_base_mixer
-
-static const struct snd_kcontrol_new alc883_mitac_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc885_8ch_intel_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x1b, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_fivestack_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_targa_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc883_medion_wim2160_verbs[] = {
- /* Unmute front mixer */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Set speaker pin to front mixer */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Init headphone pin */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-
- { } /* end */
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc883_medion_wim2160_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1a;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume",
- 0x0d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc889A_mb31_mixer[] = {
- /* Output mixers */
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
- /* Output switches */
- HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
- HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
- /* Boost mixers */
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
- /* Input mixers */
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_vaiott_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_bind_ctls alc883_bind_cap_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls alc883_bind_cap_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
- HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol),
- HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = alc_mux_enum_info,
- .get = alc_mux_enum_get,
- .put = alc_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_chmode_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc883_mitac_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct hda_verb alc883_mitac_verbs[] = {
- /* HP */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Subwoofer */
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* enable unsolicited event */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, */
-
- { } /* end */
-};
-
-static const struct hda_verb alc883_clevo_m540r_verbs[] = {
- /* HP */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Int speaker */
- /*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/
-
- /* enable unsolicited event */
- /*
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
- */
-
- { } /* end */
-};
-
-static const struct hda_verb alc883_clevo_m720_verbs[] = {
- /* HP */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Int speaker */
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* enable unsolicited event */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
-
- { } /* end */
-};
-
-static const struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
- /* HP */
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Subwoofer */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* enable unsolicited event */
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-
- { } /* end */
-};
-
-static const struct hda_verb alc883_targa_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
-/* Connect Line-Out side jack (SPDIF) to Side */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
-/* Connect Mic jack to CLFE */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
-/* Connect Line-in jack to Surround */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-/* Connect HP out jack to Front */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-
- { } /* end */
-};
-
-static const struct hda_verb alc883_lenovo_101e_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT|AC_USRSP_EN},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT|AC_USRSP_EN},
- { } /* end */
-};
-
-static const struct hda_verb alc883_lenovo_nb0763_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- { } /* end */
-};
-
-static const struct hda_verb alc888_lenovo_ms7195_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT | AC_USRSP_EN},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
-static const struct hda_verb alc883_haier_w66_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- { } /* end */
-};
-
-static const struct hda_verb alc888_lenovo_sky_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
-static const struct hda_verb alc888_6st_dell_verbs[] = {
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { }
-};
-
-static const struct hda_verb alc883_vaiott_verbs[] = {
- /* HP */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
- /* enable unsolicited event */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-
- { } /* end */
-};
-
-static void alc888_3st_hp_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->autocfg.speaker_pins[2] = 0x18;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct hda_verb alc888_3st_hp_verbs[] = {
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc888_3st_hp_2ch_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc888_3st_hp_4ch_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc888_3st_hp_6ch_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc888_3st_hp_modes[3] = {
- { 2, alc888_3st_hp_2ch_init },
- { 4, alc888_3st_hp_4ch_init },
- { 6, alc888_3st_hp_6ch_init },
-};
-
-static void alc888_lenovo_ms7195_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.line_out_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc883_lenovo_nb0763_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/* toggle speaker-output according to the hp-jack state */
-#define alc883_targa_init_hook alc882_targa_init_hook
-#define alc883_targa_unsol_event alc882_targa_unsol_event
-
-static void alc883_clevo_m720_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
-{
- alc_hp_automute(codec);
- alc88x_simple_mic_automute(codec);
-}
-
-static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case ALC880_MIC_EVENT:
- alc88x_simple_mic_automute(codec);
- break;
- default:
- alc_sku_unsol_event(codec, res);
- break;
- }
-}
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc883_haier_w66_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc883_lenovo_101e_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.line_out_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->detect_line = 1;
- spec->automute_lines = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc883_acer_aspire_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct hda_verb alc883_acer_eapd_verbs[] = {
- /* HP Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Front Pin: output 0 (0x0c) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* eanable EAPD on medion laptop */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
- /* enable unsolicited event */
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { }
-};
-
-static void alc888_6st_dell_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x15;
- spec->autocfg.speaker_pins[2] = 0x16;
- spec->autocfg.speaker_pins[3] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc888_lenovo_sky_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x15;
- spec->autocfg.speaker_pins[2] = 0x16;
- spec->autocfg.speaker_pins[3] = 0x17;
- spec->autocfg.speaker_pins[4] = 0x1a;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc883_vaiott_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct hda_verb alc888_asus_m90v_verbs[] = {
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* enable unsolicited event */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
-static void alc883_mode2_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x15;
- spec->autocfg.speaker_pins[2] = 0x16;
- spec->ext_mic.pin = 0x18;
- spec->int_mic.pin = 0x19;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct hda_verb alc888_asus_eee1601_verbs[] = {
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
- {0x20, AC_VERB_SET_PROC_COEF, 0x0838},
- /* enable unsolicited event */
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
-static void alc883_eee1601_inithook(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x1b;
- alc_hp_automute(codec);
-}
-
-static const struct hda_verb alc889A_mb31_verbs[] = {
- /* Init rear pin (used as headphone output) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- /* Init line pin (used as output in 4ch and 6ch mode) */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */
- /* Init line 2 pin (used as headphone out by default) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
- { } /* end */
-};
-
-/* Mute speakers according to the headphone jack state */
-static void alc889A_mb31_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- /* Mute only in 2ch or 4ch mode */
- if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
- == 0x00) {
- present = snd_hda_jack_detect(codec, 0x15);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- }
-}
-
-static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc889A_mb31_automute(codec);
-}
-
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc882_loopbacks alc880_loopbacks
-#endif
-
-/* pcm configuration: identical with ALC880 */
-#define alc882_pcm_analog_playback alc880_pcm_analog_playback
-#define alc882_pcm_analog_capture alc880_pcm_analog_capture
-#define alc882_pcm_digital_playback alc880_pcm_digital_playback
-#define alc882_pcm_digital_capture alc880_pcm_digital_capture
-
-static const hda_nid_t alc883_slave_dig_outs[] = {
- ALC1200_DIGOUT_NID, 0,
-};
-
-static const hda_nid_t alc1200_slave_dig_outs[] = {
- ALC883_DIGOUT_NID, 0,
-};
-
-/*
- * configuration and preset
- */
-static const char * const alc882_models[ALC882_MODEL_LAST] = {
- [ALC882_3ST_DIG] = "3stack-dig",
- [ALC882_6ST_DIG] = "6stack-dig",
- [ALC882_ARIMA] = "arima",
- [ALC882_W2JC] = "w2jc",
- [ALC882_TARGA] = "targa",
- [ALC882_ASUS_A7J] = "asus-a7j",
- [ALC882_ASUS_A7M] = "asus-a7m",
- [ALC885_MACPRO] = "macpro",
- [ALC885_MB5] = "mb5",
- [ALC885_MACMINI3] = "macmini3",
- [ALC885_MBA21] = "mba21",
- [ALC885_MBP3] = "mbp3",
- [ALC885_IMAC24] = "imac24",
- [ALC885_IMAC91] = "imac91",
- [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig",
- [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig",
- [ALC883_3ST_6ch] = "3stack-6ch",
- [ALC883_6ST_DIG] = "alc883-6stack-dig",
- [ALC883_TARGA_DIG] = "targa-dig",
- [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
- [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig",
- [ALC883_ACER] = "acer",
- [ALC883_ACER_ASPIRE] = "acer-aspire",
- [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g",
- [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g",
- [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g",
- [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g",
- [ALC883_MEDION] = "medion",
- [ALC883_MEDION_WIM2160] = "medion-wim2160",
- [ALC883_LAPTOP_EAPD] = "laptop-eapd",
- [ALC883_LENOVO_101E_2ch] = "lenovo-101e",
- [ALC883_LENOVO_NB0763] = "lenovo-nb0763",
- [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
- [ALC888_LENOVO_SKY] = "lenovo-sky",
- [ALC883_HAIER_W66] = "haier-w66",
- [ALC888_3ST_HP] = "3stack-hp",
- [ALC888_6ST_DELL] = "6stack-dell",
- [ALC883_MITAC] = "mitac",
- [ALC883_CLEVO_M540R] = "clevo-m540r",
- [ALC883_CLEVO_M720] = "clevo-m720",
- [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
- [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
- [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel",
- [ALC889A_INTEL] = "intel-alc889a",
- [ALC889_INTEL] = "intel-x58",
- [ALC1200_ASUS_P5Q] = "asus-p5q",
- [ALC889A_MB31] = "mb31",
- [ALC883_SONY_VAIO_TT] = "sony-vaio-tt",
- [ALC882_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc882_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
-
- SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
- SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
- SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
- SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
- SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
- SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
- SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G",
- ALC888_ACER_ASPIRE_4930G),
- SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
- ALC888_ACER_ASPIRE_4930G),
- SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G",
- ALC888_ACER_ASPIRE_8930G),
- SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
- ALC888_ACER_ASPIRE_8930G),
- SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO),
- SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO),
- SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
- ALC888_ACER_ASPIRE_6530G),
- SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
- ALC888_ACER_ASPIRE_6530G),
- SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
- ALC888_ACER_ASPIRE_7730G),
- /* default Acer -- disabled as it causes more problems.
- * model=auto should work fine now
- */
- /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */
-
- SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
-
- SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
- SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
- SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
- SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
-
- SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
- SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
- SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
- SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
- SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
- SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
- SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
-
- SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
- SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
- SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
- SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
- SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
- SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
-
- SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
- SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
- SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
- SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
- SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO),
- SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
- SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG),
- SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG),
- SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
- SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
-
- SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
- SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
- SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
- SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
- SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
- /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */
- SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
- SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx",
- ALC883_FUJITSU_PI2515),
- SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1130, "Fujitsu AMILO Xa35xx",
- ALC888_FUJITSU_XA3530),
- SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
- SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
- SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
- SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
- SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
- SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
- SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
-
- SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
- SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
- SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
- SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL),
- SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL),
- SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL),
- SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG),
-
- {}
-};
-
-/* codec SSID table for Intel Mac */
-static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
- SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO),
- SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
- SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
- SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
- SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M),
- SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
- SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
- SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
- SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
- SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
- SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
- /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
- * so apparently no perfect solution yet
- */
- SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
- SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
- SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3),
- {} /* terminator */
-};
-
-static const struct alc_config_preset alc882_presets[] = {
- [ALC882_3ST_DIG] = {
- .mixers = { alc882_base_mixer },
- .init_verbs = { alc882_base_init_verbs,
- alc882_adc1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
- .channel_mode = alc882_ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_6ST_DIG] = {
- .mixers = { alc882_base_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_base_init_verbs,
- alc882_adc1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
- .channel_mode = alc882_sixstack_modes,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_ARIMA] = {
- .mixers = { alc882_base_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
- alc882_eapd_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
- .channel_mode = alc882_sixstack_modes,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_W2JC] = {
- .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
- alc882_eapd_verbs, alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- },
- [ALC885_MBA21] = {
- .mixers = { alc885_mba21_mixer },
- .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs },
- .num_dacs = 2,
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_mba21_ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
- .input_mux = &alc882_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_mba21_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_MBP3] = {
- .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_mbp3_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = 2,
- .dac_nids = alc882_dac_nids,
- .hp_nid = 0x04,
- .channel_mode = alc885_mbp_4ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
- .input_mux = &alc882_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_mbp3_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_MB5] = {
- .mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_mb5_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_mb5_6ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
- .input_mux = &mb5_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_mb5_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_MACMINI3] = {
- .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_macmini3_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_macmini3_6ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes),
- .input_mux = &macmini3_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_macmini3_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_MACPRO] = {
- .mixers = { alc882_macpro_mixer },
- .init_verbs = { alc882_macpro_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
- .channel_mode = alc882_ch_modes,
- .input_mux = &alc882_capture_source,
- .init_hook = alc885_macpro_init_hook,
- },
- [ALC885_IMAC24] = {
- .mixers = { alc885_imac24_mixer },
- .init_verbs = { alc885_imac24_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
- .channel_mode = alc882_ch_modes,
- .input_mux = &alc882_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_imac24_setup,
- .init_hook = alc885_imac24_init_hook,
- },
- [ALC885_IMAC91] = {
- .mixers = {alc885_imac91_mixer},
- .init_verbs = { alc885_imac91_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_mba21_ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
- .input_mux = &alc889A_imac91_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_imac91_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC882_TARGA] = {
- .mixers = { alc882_targa_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
- alc880_gpio3_init_verbs, alc882_targa_verbs},
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
- .adc_nids = alc882_adc_nids,
- .capsrc_nids = alc882_capsrc_nids,
- .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
- .channel_mode = alc882_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc882_targa_setup,
- .init_hook = alc882_targa_automute,
- },
- [ALC882_ASUS_A7J] = {
- .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
- alc882_asus_a7j_verbs},
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
- .adc_nids = alc882_adc_nids,
- .capsrc_nids = alc882_capsrc_nids,
- .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
- .channel_mode = alc882_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_ASUS_A7M] = {
- .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
- alc882_eapd_verbs, alc880_gpio1_init_verbs,
- alc882_asus_a7m_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- },
- [ALC883_3ST_2ch_DIG] = {
- .mixers = { alc883_3ST_2ch_mixer },
- .init_verbs = { alc883_init_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- },
- [ALC883_3ST_6ch_DIG] = {
- .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_capture_source,
- },
- [ALC883_3ST_6ch] = {
- .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_capture_source,
- },
- [ALC883_3ST_6ch_INTEL] = {
- .mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .slave_dig_outs = alc883_slave_dig_outs,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
- .channel_mode = alc883_3ST_6ch_intel_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_3stack_6ch_intel,
- },
- [ALC889A_INTEL] = {
- .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
- .init_verbs = { alc885_init_verbs, alc885_init_input_verbs,
- alc_hp15_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
- .adc_nids = alc889_adc_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .slave_dig_outs = alc883_slave_dig_outs,
- .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
- .channel_mode = alc889_8ch_intel_modes,
- .capsrc_nids = alc889_capsrc_nids,
- .input_mux = &alc889_capture_source,
- .setup = alc889_automute_setup,
- .init_hook = alc_hp_automute,
- .unsol_event = alc_sku_unsol_event,
- .need_dac_fix = 1,
- },
- [ALC889_INTEL] = {
- .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
- .init_verbs = { alc885_init_verbs, alc889_init_input_verbs,
- alc889_eapd_verbs, alc_hp15_unsol_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
- .adc_nids = alc889_adc_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .slave_dig_outs = alc883_slave_dig_outs,
- .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
- .channel_mode = alc889_8ch_intel_modes,
- .capsrc_nids = alc889_capsrc_nids,
- .input_mux = &alc889_capture_source,
- .setup = alc889_automute_setup,
- .init_hook = alc889_intel_init_hook,
- .unsol_event = alc_sku_unsol_event,
- .need_dac_fix = 1,
- },
- [ALC883_6ST_DIG] = {
- .mixers = { alc883_base_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
- .channel_mode = alc883_sixstack_modes,
- .input_mux = &alc883_capture_source,
- },
- [ALC883_TARGA_DIG] = {
- .mixers = { alc883_targa_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
- alc883_targa_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc883_targa_unsol_event,
- .setup = alc882_targa_setup,
- .init_hook = alc882_targa_automute,
- },
- [ALC883_TARGA_2ch_DIG] = {
- .mixers = { alc883_targa_2ch_mixer},
- .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
- alc883_targa_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .adc_nids = alc883_adc_nids_alt,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
- .capsrc_nids = alc883_capsrc_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc883_targa_unsol_event,
- .setup = alc882_targa_setup,
- .init_hook = alc882_targa_automute,
- },
- [ALC883_TARGA_8ch_DIG] = {
- .mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer,
- alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
- alc883_targa_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
- .adc_nids = alc883_adc_nids_rev,
- .capsrc_nids = alc883_capsrc_nids_rev,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes),
- .channel_mode = alc883_4ST_8ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc883_targa_unsol_event,
- .setup = alc882_targa_setup,
- .init_hook = alc882_targa_automute,
- },
- [ALC883_ACER] = {
- .mixers = { alc883_base_mixer },
- /* On TravelMate laptops, GPIO 0 enables the internal speaker
- * and the headphone jack. Turn this on and rely on the
- * standard mute methods whenever the user wants to turn
- * these outputs off.
- */
- .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- },
- [ALC883_ACER_ASPIRE] = {
- .mixers = { alc883_acer_aspire_mixer },
- .init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_acer_aspire_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_ACER_ASPIRE_4930G] = {
- .mixers = { alc888_acer_aspire_4930g_mixer,
- alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
- alc888_acer_aspire_4930g_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
- .adc_nids = alc883_adc_nids_rev,
- .capsrc_nids = alc883_capsrc_nids_rev,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .const_channel_count = 6,
- .num_mux_defs =
- ARRAY_SIZE(alc888_2_capture_sources),
- .input_mux = alc888_2_capture_sources,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_acer_aspire_4930g_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_ACER_ASPIRE_6530G] = {
- .mixers = { alc888_acer_aspire_6530_mixer },
- .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
- alc888_acer_aspire_6530g_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
- .adc_nids = alc883_adc_nids_rev,
- .capsrc_nids = alc883_capsrc_nids_rev,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .num_mux_defs =
- ARRAY_SIZE(alc888_2_capture_sources),
- .input_mux = alc888_acer_aspire_6530_sources,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_acer_aspire_6530g_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_ACER_ASPIRE_8930G] = {
- .mixers = { alc889_acer_aspire_8930g_mixer,
- alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
- alc889_acer_aspire_8930g_verbs,
- alc889_eapd_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
- .adc_nids = alc889_adc_nids,
- .capsrc_nids = alc889_capsrc_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .const_channel_count = 6,
- .num_mux_defs =
- ARRAY_SIZE(alc889_capture_sources),
- .input_mux = alc889_capture_sources,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc889_acer_aspire_8930g_setup,
- .init_hook = alc_hp_automute,
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- .power_hook = alc_power_eapd,
-#endif
- },
- [ALC888_ACER_ASPIRE_7730G] = {
- .mixers = { alc883_3ST_6ch_mixer,
- alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
- alc888_acer_aspire_7730G_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
- .adc_nids = alc883_adc_nids_rev,
- .capsrc_nids = alc883_capsrc_nids_rev,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .const_channel_count = 6,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_acer_aspire_7730g_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC883_MEDION] = {
- .mixers = { alc883_fivestack_mixer,
- alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs,
- alc883_medion_eapd_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .adc_nids = alc883_adc_nids_alt,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
- .capsrc_nids = alc883_capsrc_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
- .channel_mode = alc883_sixstack_modes,
- .input_mux = &alc883_capture_source,
- },
- [ALC883_MEDION_WIM2160] = {
- .mixers = { alc883_medion_wim2160_mixer },
- .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .adc_nids = alc883_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_medion_wim2160_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC883_LAPTOP_EAPD] = {
- .mixers = { alc883_base_mixer },
- .init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- },
- [ALC883_CLEVO_M540R] = {
- .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_clevo_modes),
- .channel_mode = alc883_3ST_6ch_clevo_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_capture_source,
- /* This machine has the hardware HP auto-muting, thus
- * we need no software mute via unsol event
- */
- },
- [ALC883_CLEVO_M720] = {
- .mixers = { alc883_clevo_m720_mixer },
- .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc883_clevo_m720_unsol_event,
- .setup = alc883_clevo_m720_setup,
- .init_hook = alc883_clevo_m720_init_hook,
- },
- [ALC883_LENOVO_101E_2ch] = {
- .mixers = { alc883_lenovo_101e_2ch_mixer},
- .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .adc_nids = alc883_adc_nids_alt,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
- .capsrc_nids = alc883_capsrc_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_lenovo_101e_capture_source,
- .setup = alc883_lenovo_101e_setup,
- .unsol_event = alc_sku_unsol_event,
- .init_hook = alc_inithook,
- },
- [ALC883_LENOVO_NB0763] = {
- .mixers = { alc883_lenovo_nb0763_mixer },
- .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_lenovo_nb0763_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_lenovo_nb0763_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_LENOVO_MS7195_DIG] = {
- .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc888_lenovo_ms7195_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_lenovo_ms7195_setup,
- .init_hook = alc_inithook,
- },
- [ALC883_HAIER_W66] = {
- .mixers = { alc883_targa_2ch_mixer},
- .init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_haier_w66_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_3ST_HP] = {
- .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
- .channel_mode = alc888_3st_hp_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_3st_hp_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_6ST_DELL] = {
- .mixers = { alc883_base_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
- .channel_mode = alc883_sixstack_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_6st_dell_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC883_MITAC] = {
- .mixers = { alc883_mitac_mixer },
- .init_verbs = { alc883_init_verbs, alc883_mitac_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_mitac_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC883_FUJITSU_PI2515] = {
- .mixers = { alc883_2ch_fujitsu_pi2515_mixer },
- .init_verbs = { alc883_init_verbs,
- alc883_2ch_fujitsu_pi2515_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_fujitsu_pi2515_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_2ch_fujitsu_pi2515_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_FUJITSU_XA3530] = {
- .mixers = { alc888_base_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs,
- alc888_fujitsu_xa3530_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
- .adc_nids = alc883_adc_nids_rev,
- .capsrc_nids = alc883_capsrc_nids_rev,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc888_4ST_8ch_intel_modes),
- .channel_mode = alc888_4ST_8ch_intel_modes,
- .num_mux_defs =
- ARRAY_SIZE(alc888_2_capture_sources),
- .input_mux = alc888_2_capture_sources,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_fujitsu_xa3530_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_LENOVO_SKY] = {
- .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs},
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
- .channel_mode = alc883_sixstack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_lenovo_sky_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc888_lenovo_sky_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC888_ASUS_M90V] = {
- .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc888_asus_m90v_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
- .channel_mode = alc883_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_fujitsu_pi2515_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_mode2_setup,
- .init_hook = alc_inithook,
- },
- [ALC888_ASUS_EEE1601] = {
- .mixers = { alc883_asus_eee1601_mixer },
- .cap_mixer = alc883_asus_eee1601_cap_mixer,
- .init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc883_asus_eee1601_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .init_hook = alc883_eee1601_inithook,
- },
- [ALC1200_ASUS_P5Q] = {
- .mixers = { alc883_base_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .dig_out_nid = ALC1200_DIGOUT_NID,
- .dig_in_nid = ALC883_DIGIN_NID,
- .slave_dig_outs = alc1200_slave_dig_outs,
- .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
- .channel_mode = alc883_sixstack_modes,
- .input_mux = &alc883_capture_source,
- },
- [ALC889A_MB31] = {
- .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
- .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
- alc880_gpio1_init_verbs },
- .adc_nids = alc883_adc_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .capsrc_nids = alc883_capsrc_nids,
- .dac_nids = alc883_dac_nids,
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .channel_mode = alc889A_mb31_6ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
- .input_mux = &alc889A_mb31_capture_source,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .unsol_event = alc889A_mb31_unsol_event,
- .init_hook = alc889A_mb31_automute,
- },
- [ALC883_SONY_VAIO_TT] = {
- .mixers = { alc883_vaiott_mixer },
- .init_verbs = { alc883_init_verbs, alc883_vaiott_verbs },
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .dac_nids = alc883_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .input_mux = &alc883_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc883_vaiott_setup,
- .init_hook = alc_hp_automute,
- },
-};
-
-
-/*
- * Pin config fixes
- */
-enum {
- PINFIX_ABIT_AW9D_MAX,
- PINFIX_LENOVO_Y530,
- PINFIX_PB_M5210,
- PINFIX_ACER_ASPIRE_7736,
-};
-
-static const struct alc_fixup alc882_fixups[] = {
- [PINFIX_ABIT_AW9D_MAX] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x15, 0x01080104 }, /* side */
- { 0x16, 0x01011012 }, /* rear */
- { 0x17, 0x01016011 }, /* clfe */
- { }
- }
- },
- [PINFIX_LENOVO_Y530] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x15, 0x99130112 }, /* rear int speakers */
- { 0x16, 0x99130111 }, /* subwoofer */
- { }
- }
- },
- [PINFIX_PB_M5210] = {
- .type = ALC_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
- {}
- }
- },
- [PINFIX_ACER_ASPIRE_7736] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
- },
-};
-
-static const struct snd_pci_quirk alc882_fixup_tbl[] = {
- SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210),
- SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
- SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
- SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
- {}
-};
-
-/*
- * BIOS auto configuration
- */
-static int alc882_auto_create_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x23, 0x22);
-}
-
-static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type,
- hda_nid_t dac)
-{
- int idx;
-
- /* set as output */
- alc_set_pin_output(codec, nid, pin_type);
-
- if (dac == 0x25)
- idx = 4;
- else if (dac >= 0x02 && dac <= 0x05)
- idx = dac - 2;
- else
- return;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
-}
-
-static void alc882_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i <= HDA_SIDE; i++) {
- hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- if (nid)
- alc882_auto_set_output_and_unmute(codec, nid, pin_type,
- spec->multiout.dac_nids[i]);
- }
-}
-
-static void alc882_auto_init_hp_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t pin, dac;
- int i;
-
- if (spec->autocfg.line_out_type != AUTO_PIN_HP_OUT) {
- for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) {
- pin = spec->autocfg.hp_pins[i];
- if (!pin)
- break;
- dac = spec->multiout.hp_nid;
- if (!dac)
- dac = spec->multiout.dac_nids[0]; /* to front */
- alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
- }
- }
-
- if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT) {
- for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
- pin = spec->autocfg.speaker_pins[i];
- if (!pin)
- break;
- dac = spec->multiout.extra_out_nid[0];
- if (!dac)
- dac = spec->multiout.dac_nids[0]; /* to front */
- alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
- }
- }
-}
-
-static void alc882_auto_init_analog_input(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
- alc_set_input_pin(codec, nid, cfg->inputs[i].type);
- if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
- snd_hda_codec_write(codec, nid, 0,
+ /* mute all loopback inputs */
+ if (spec->mixer_nid) {
+ int nums = snd_hda_get_conn_list(codec, spec->mixer_nid, NULL);
+ for (i = 0; i < nums; i++)
+ snd_hda_codec_write(codec, spec->mixer_nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
- }
-}
-
-static void alc882_auto_init_input_src(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int c;
-
- for (c = 0; c < spec->num_adc_nids; c++) {
- hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
- hda_nid_t nid = spec->capsrc_nids[c];
- unsigned int mux_idx;
- const struct hda_input_mux *imux;
- int conns, mute, idx, item;
-
- /* mute ADC */
- snd_hda_codec_write(codec, spec->adc_nids[c], 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_MUTE(0));
-
- conns = snd_hda_get_connections(codec, nid, conn_list,
- ARRAY_SIZE(conn_list));
- if (conns < 0)
- continue;
- mux_idx = c >= spec->num_mux_defs ? 0 : c;
- imux = &spec->input_mux[mux_idx];
- if (!imux->num_items && mux_idx > 0)
- imux = &spec->input_mux[0];
- for (idx = 0; idx < conns; idx++) {
- /* if the current connection is the selected one,
- * unmute it as default - otherwise mute it
- */
- mute = AMP_IN_MUTE(idx);
- for (item = 0; item < imux->num_items; item++) {
- if (imux->items[item].index == idx) {
- if (spec->cur_mux[c] == item)
- mute = AMP_IN_UNMUTE(idx);
- break;
- }
- }
- /* check if we have a selector or mixer
- * we could check for the widget type instead, but
- * just check for Amp-In presence (in case of mixer
- * without amp-in there is something wrong, this
- * function shouldn't be used or capsrc nid is wrong)
- */
- if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- mute);
- else if (mute != AMP_IN_MUTE(idx))
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL,
- idx);
- }
- }
-}
-
-/* add mic boosts if needed */
-static int alc_auto_add_mic_boost(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i, err;
- int type_idx = 0;
- hda_nid_t nid;
- const char *prev_label = NULL;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- if (cfg->inputs[i].type > AUTO_PIN_MIC)
- break;
- nid = cfg->inputs[i].pin;
- if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) {
- const char *label;
- char boost_label[32];
-
- label = hda_get_autocfg_input_label(codec, cfg, i);
- if (prev_label && !strcmp(label, prev_label))
- type_idx++;
- else
- type_idx = 0;
- prev_label = label;
-
- snprintf(boost_label, sizeof(boost_label),
- "%s Boost Volume", label);
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- boost_label, type_idx,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-/* almost identical with ALC880 parser... */
-static int alc882_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- static const hda_nid_t alc882_ignore[] = { 0x1d, 0 };
- int err;
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc882_ignore);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs)
- return 0; /* can't find valid BIOS pin config */
-
- err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc_auto_add_multi_channel_mode(codec);
- if (err < 0)
- return err;
- err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
- "Headphone");
- if (err < 0)
- return err;
- err = alc880_auto_create_extra_out(spec,
- spec->autocfg.speaker_pins[0],
- "Speaker");
- if (err < 0)
- return err;
- err = alc882_auto_create_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- alc_auto_parse_digital(codec);
-
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- add_verb(spec, alc883_auto_init_verbs);
- /* if ADC 0x07 is available, initialize it, too */
- if (get_wcaps_type(get_wcaps(codec, 0x07)) == AC_WID_AUD_IN)
- add_verb(spec, alc882_adc1_init_verbs);
-
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- return 1; /* config found */
-}
-
-/* additional initialization for auto-configuration model */
-static void alc882_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc882_auto_init_multi_out(codec);
- alc882_auto_init_hp_out(codec);
- alc882_auto_init_analog_input(codec);
- alc882_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-static int patch_alc882(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int err, board_config;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- switch (codec->vendor_id) {
- case 0x10ec0882:
- case 0x10ec0885:
- break;
- default:
- /* ALC883 and variants */
- alc_fix_pll_init(codec, 0x20, 0x0a, 10);
- break;
- }
-
- board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
- alc882_models,
- alc882_cfg_tbl);
-
- if (board_config < 0 || board_config >= ALC882_MODEL_LAST)
- board_config = snd_hda_check_board_codec_sid_config(codec,
- ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl);
-
- if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC882_AUTO;
- }
-
- if (board_config == ALC882_AUTO) {
- alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- alc_auto_parse_customize_define(codec);
-
- if (board_config == ALC882_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc882_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC882_3ST_DIG;
- }
- }
-
- if (has_cdefine_beep(codec)) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
- }
-
- if (board_config != ALC882_AUTO)
- setup_preset(codec, &alc882_presets[board_config]);
-
- spec->stream_analog_playback = &alc882_pcm_analog_playback;
- spec->stream_analog_capture = &alc882_pcm_analog_capture;
- /* FIXME: setup DAC5 */
- /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
- spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
-
- spec->stream_digital_playback = &alc882_pcm_digital_playback;
- spec->stream_digital_capture = &alc882_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- int i, j;
- spec->num_adc_nids = 0;
- for (i = 0; i < ARRAY_SIZE(alc882_adc_nids); i++) {
- const struct hda_input_mux *imux = spec->input_mux;
- hda_nid_t cap;
- hda_nid_t items[16];
- hda_nid_t nid = alc882_adc_nids[i];
- unsigned int wcap = get_wcaps(codec, nid);
- /* get type */
- wcap = get_wcaps_type(wcap);
- if (wcap != AC_WID_AUD_IN)
- continue;
- spec->private_adc_nids[spec->num_adc_nids] = nid;
- err = snd_hda_get_connections(codec, nid, &cap, 1);
- if (err < 0)
- continue;
- err = snd_hda_get_connections(codec, cap, items,
- ARRAY_SIZE(items));
- if (err < 0)
- continue;
- for (j = 0; j < imux->num_items; j++)
- if (imux->items[j].index >= err)
- break;
- if (j < imux->num_items)
- continue;
- spec->private_capsrc_nids[spec->num_adc_nids] = cap;
- spec->num_adc_nids++;
- }
- spec->adc_nids = spec->private_adc_nids;
- spec->capsrc_nids = spec->private_capsrc_nids;
- }
-
- set_capture_mixer(codec);
-
- if (has_cdefine_beep(codec))
- set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
-
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
- spec->vmaster_nid = 0x0c;
-
- codec->patch_ops = alc_patch_ops;
- if (board_config == ALC882_AUTO)
- spec->init_hook = alc882_auto_init;
-
- alc_init_jacks(codec);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc882_loopbacks;
-#endif
-
- return 0;
-}
-
-
-/*
- * ALC262 support
- */
-
-#define ALC262_DIGOUT_NID ALC880_DIGOUT_NID
-#define ALC262_DIGIN_NID ALC880_DIGIN_NID
-
-#define alc262_dac_nids alc260_dac_nids
-#define alc262_adc_nids alc882_adc_nids
-#define alc262_adc_nids_alt alc882_adc_nids_alt
-#define alc262_capsrc_nids alc882_capsrc_nids
-#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt
-
-#define alc262_modes alc260_modes
-#define alc262_capture_source alc882_capture_source
-
-static const hda_nid_t alc262_dmic_adc_nids[1] = {
- /* ADC0 */
- 0x09
-};
-
-static const hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
-
-static const struct snd_kcontrol_new alc262_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-/* update HP, line and mono-out pins according to the master switch */
-#define alc262_hp_master_update alc260_hp_master_update
-
-static void alc262_hp_bpc_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static void alc262_hp_wildwest_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-#define alc262_hp_master_sw_get alc260_hp_master_sw_get
-#define alc262_hp_master_sw_put alc260_hp_master_sw_put
-
-#define ALC262_HP_MASTER_SWITCH \
- { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = "Master Playback Switch", \
- .info = snd_ctl_boolean_mono_info, \
- .get = alc262_hp_master_sw_get, \
- .put = alc262_hp_master_sw_put, \
- }, \
- { \
- .iface = NID_MAPPING, \
- .name = "Master Playback Switch", \
- .private_value = 0x15 | (0x16 << 8) | (0x1b << 16), \
- }
-
-
-static const struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
- ALC262_HP_MASTER_SWITCH,
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
- ALC262_HP_MASTER_SWITCH,
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x1a, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
- HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT),
- { } /* end */
-};
-
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hp_t5735_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc262_hp_t5735_verbs[] = {
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { }
-};
-
-static const struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc262_hp_rp5700_verbs[] = {
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
- {}
-};
-
-static const struct hda_input_mux alc262_hp_rp5700_capture_source = {
- .num_items = 1,
- .items = {
- { "Line", 0x1 },
- },
-};
-
-/* bind hp and internal speaker mute (with plug check) as master switch */
-#define alc262_hippo_master_update alc262_hp_master_update
-#define alc262_hippo_master_sw_get alc262_hp_master_sw_get
-#define alc262_hippo_master_sw_put alc262_hp_master_sw_put
-
-#define ALC262_HIPPO_MASTER_SWITCH \
- { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = "Master Playback Switch", \
- .info = snd_ctl_boolean_mono_info, \
- .get = alc262_hippo_master_sw_get, \
- .put = alc262_hippo_master_sw_put, \
- }, \
- { \
- .iface = NID_MAPPING, \
- .name = "Master Playback Switch", \
- .subdevice = SUBDEV_HP(0) | (SUBDEV_LINE(0) << 8) | \
- (SUBDEV_SPEAKER(0) << 16), \
- }
-
-static const struct snd_kcontrol_new alc262_hippo_mixer[] = {
- ALC262_HIPPO_MASTER_SWITCH,
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_hippo1_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC262_HIPPO_MASTER_SWITCH,
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { } /* end */
-};
-
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hippo_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc262_hippo1_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-
-static const struct snd_kcontrol_new alc262_sony_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC262_HIPPO_MASTER_SWITCH,
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC262_HIPPO_MASTER_SWITCH,
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_tyan_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc262_tyan_verbs[] = {
- /* Headphone automute */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* P11 AUX_IN, white 4-pin connector */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1},
- {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93},
- {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19},
-
- {}
-};
-
-/* unsolicited event for HP jack sensing */
-static void alc262_tyan_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-
-#define alc262_capture_mixer alc882_capture_mixer
-#define alc262_capture_alt_mixer alc882_capture_alt_mixer
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc262_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for
- * front panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /*
- * Set up output mixers (0x0c - 0x0e)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
-
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-
- { }
-};
-
-static const struct hda_verb alc262_eapd_verbs[] = {
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-static const struct hda_verb alc262_hippo1_unsol_verbs[] = {
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
-
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {}
-};
-
-static const struct hda_verb alc262_sony_unsol_verbs[] = {
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, // Front Mic
-
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {}
-};
-
-static const struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc262_toshiba_s06_verbs[] = {
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x09},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static void alc262_toshiba_s06_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 9;
- spec->auto_mic = 1;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-/*
- * nec model
- * 0x15 = headphone
- * 0x16 = internal speaker
- * 0x18 = external mic
- */
-
-static const struct snd_kcontrol_new alc262_nec_mixer[] = {
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc262_nec_verbs[] = {
- /* Unmute Speaker */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Headphone */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* External mic to headphone */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* External mic to speaker */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {}
-};
-
-/*
- * fujitsu model
- * 0x14 = headphone/spdif-out, 0x15 = internal speaker,
- * 0x1b = port replicator headphone out
- */
-
-#define ALC_HP_EVENT ALC880_HP_EVENT
-
-static const struct hda_verb alc262_fujitsu_unsol_verbs[] = {
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {}
-};
-
-static const struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {}
-};
-
-static const struct hda_verb alc262_lenovo_3000_init_verbs[] = {
- /* Front Mic pin: input vref at 50% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {}
-};
-
-static const struct hda_input_mux alc262_fujitsu_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc262_HP_capture_source = {
- .num_items = 5,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "AUX IN", 0x6 },
- },
-};
-
-static const struct hda_input_mux alc262_HP_D7000_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x2 },
- { "Line", 0x1 },
- { "CD", 0x4 },
- },
-};
-
-static void alc262_fujitsu_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.hp_pins[1] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-/* bind volumes of both NID 0x0c and 0x0d */
-static const struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
- .info = snd_ctl_boolean_mono_info,
- .get = alc262_hp_master_sw_get,
- .put = alc262_hp_master_sw_put,
- },
- {
- .iface = NID_MAPPING,
- .name = "Master Playback Switch",
- .private_value = 0x1b,
- },
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static void alc262_lenovo_3000_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
- .info = snd_ctl_boolean_mono_info,
- .get = alc262_hp_master_sw_get,
- .put = alc262_hp_master_sw_put,
- },
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
- ALC262_HIPPO_MASTER_SWITCH,
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { } /* end */
-};
-
-/* additional init verbs for Benq laptops */
-static const struct hda_verb alc262_EAPD_verbs[] = {
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
- {}
-};
-
-static const struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
-
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
- {}
-};
-
-/* Samsung Q1 Ultra Vista model setup */
-static const struct snd_kcontrol_new alc262_ultra_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Mic Boost Volume", 0x15, 0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_verb alc262_ultra_verbs[] = {
- /* output mixer */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* speaker */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- /* internal mic */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* ADC, choose mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)},
- {}
-};
-
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_ultra_automute(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- unsigned int mute;
-
- mute = 0;
- /* auto-mute only when HP is used as HP */
- if (!spec->cur_mux[0]) {
- spec->jack_present = snd_hda_jack_detect(codec, 0x15);
- if (spec->jack_present)
- mute = HDA_AMP_MUTE;
- }
- /* mute/unmute internal speaker */
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- /* mute/unmute HP */
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE);
-}
-
-/* unsolicited event for HP jack sensing */
-static void alc262_ultra_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != ALC880_HP_EVENT)
- return;
- alc262_ultra_automute(codec);
-}
-
-static const struct hda_input_mux alc262_ultra_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 },
- { "Headphone", 0x7 },
- },
-};
-
-static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- int ret;
-
- ret = alc_mux_enum_put(kcontrol, ucontrol);
- if (!ret)
- return 0;
- /* reprogram the HP pin as mic or HP according to the input source */
- snd_hda_codec_write_cache(codec, 0x15, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->cur_mux[0] ? PIN_VREF80 : PIN_HP);
- alc262_ultra_automute(codec); /* mute/unmute HP */
- return ret;
-}
-
-static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = alc_mux_enum_info,
- .get = alc_mux_enum_get,
- .put = alc262_ultra_mux_enum_put,
- },
- {
- .iface = NID_MAPPING,
- .name = "Capture Source",
- .private_value = 0x15,
- },
- { } /* end */
-};
-
-/* We use two mixers depending on the output pin; 0x16 is a mono output
- * and thus it's bound with a different mixer.
- * This function returns which mixer amp should be used.
- */
-static int alc262_check_volbit(hda_nid_t nid)
-{
- if (!nid)
- return 0;
- else if (nid == 0x16)
- return 2;
- else
- return 1;
-}
-
-static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid,
- const char *pfx, int *vbits, int idx)
-{
- unsigned long val;
- int vbit;
-
- vbit = alc262_check_volbit(nid);
- if (!vbit)
- return 0;
- if (*vbits & vbit) /* a volume control for this mixer already there */
- return 0;
- *vbits |= vbit;
- if (vbit == 2)
- val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT);
- else
- val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT);
- return __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, idx, val);
-}
-
-static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid,
- const char *pfx, int idx)
-{
- unsigned long val;
-
- if (!nid)
- return 0;
- if (nid == 0x16)
- val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
- else
- val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
- return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, idx, val);
-}
-
-/* add playback controls from the parsed DAC table */
-static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- const char *pfx;
- int vbits;
- int i, err;
-
- spec->multiout.num_dacs = 1; /* only use one dac */
- spec->multiout.dac_nids = spec->private_dac_nids;
- spec->private_dac_nids[0] = 2;
-
- pfx = alc_get_line_out_pfx(spec, true);
- if (!pfx)
- pfx = "Front";
- for (i = 0; i < 2; i++) {
- err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[i], pfx, i);
- if (err < 0)
- return err;
- if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
- err = alc262_add_out_sw_ctl(spec, cfg->speaker_pins[i],
- "Speaker", i);
- if (err < 0)
- return err;
- }
- if (cfg->line_out_type != AUTO_PIN_HP_OUT) {
- err = alc262_add_out_sw_ctl(spec, cfg->hp_pins[i],
- "Headphone", i);
- if (err < 0)
- return err;
- }
- }
-
- vbits = alc262_check_volbit(cfg->line_out_pins[0]) |
- alc262_check_volbit(cfg->speaker_pins[0]) |
- alc262_check_volbit(cfg->hp_pins[0]);
- if (vbits == 1 || vbits == 2)
- pfx = "Master"; /* only one mixer is used */
- vbits = 0;
- for (i = 0; i < 2; i++) {
- err = alc262_add_out_vol_ctl(spec, cfg->line_out_pins[i], pfx,
- &vbits, i);
- if (err < 0)
- return err;
- if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
- err = alc262_add_out_vol_ctl(spec, cfg->speaker_pins[i],
- "Speaker", &vbits, i);
- if (err < 0)
- return err;
- }
- if (cfg->line_out_type != AUTO_PIN_HP_OUT) {
- err = alc262_add_out_vol_ctl(spec, cfg->hp_pins[i],
- "Headphone", &vbits, i);
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-#define alc262_auto_create_input_ctls \
- alc882_auto_create_input_ctls
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc262_volume_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for
- * front panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /*
- * Set up output mixers (0x0c - 0x0f)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-
- { }
-};
-
-static const struct hda_verb alc262_HP_BPC_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for
- * front panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-
- /*
- * Set up output mixers (0x0c - 0x0e)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */
- /* Input mixer1: only unmute Mic */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
-
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-
- { }
-};
-
-static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for front
- * panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /*
- * Set up output mixers (0x0c - 0x0e)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Mono */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* rear MIC */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* Line in */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Line out */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD in */
-
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-
- /* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, /*rear MIC*/
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/
- /* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
-
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-
- { }
-};
-
-static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
-
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-/*
- * Pin config fixes
- */
-enum {
- PINFIX_FSC_H270,
- PINFIX_HP_Z200,
-};
-
-static const struct alc_fixup alc262_fixups[] = {
- [PINFIX_FSC_H270] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x15, 0x0221142f }, /* front HP */
- { 0x1b, 0x0121141f }, /* rear HP */
- { }
- }
- },
- [PINFIX_HP_Z200] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x16, 0x99130120 }, /* internal speaker */
- { }
- }
- },
-};
-
-static const struct snd_pci_quirk alc262_fixup_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
- SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
- {}
-};
-
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc262_loopbacks alc880_loopbacks
-#endif
-
-/* pcm configuration: identical with ALC880 */
-#define alc262_pcm_analog_playback alc880_pcm_analog_playback
-#define alc262_pcm_analog_capture alc880_pcm_analog_capture
-#define alc262_pcm_digital_playback alc880_pcm_digital_playback
-#define alc262_pcm_digital_capture alc880_pcm_digital_capture
-
-/*
- * BIOS auto configuration
- */
-static int alc262_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc262_ignore[] = { 0x1d, 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc262_ignore);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs) {
- if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
- spec->multiout.max_channels = 2;
- spec->no_analog = 1;
- goto dig_only;
- }
- return 0; /* can't find valid BIOS pin config */
- }
- err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc262_auto_create_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- dig_only:
- alc_auto_parse_digital(codec);
-
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- add_verb(spec, alc262_volume_init_verbs);
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
-
- return 1;
-}
-
-#define alc262_auto_init_multi_out alc882_auto_init_multi_out
-#define alc262_auto_init_hp_out alc882_auto_init_hp_out
-#define alc262_auto_init_analog_input alc882_auto_init_analog_input
-#define alc262_auto_init_input_src alc882_auto_init_input_src
-
-
-/* init callback for auto-configuration model -- overriding the default init */
-static void alc262_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc262_auto_init_multi_out(codec);
- alc262_auto_init_hp_out(codec);
- alc262_auto_init_analog_input(codec);
- alc262_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-/*
- * configuration and preset
- */
-static const char * const alc262_models[ALC262_MODEL_LAST] = {
- [ALC262_BASIC] = "basic",
- [ALC262_HIPPO] = "hippo",
- [ALC262_HIPPO_1] = "hippo_1",
- [ALC262_FUJITSU] = "fujitsu",
- [ALC262_HP_BPC] = "hp-bpc",
- [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
- [ALC262_HP_TC_T5735] = "hp-tc-t5735",
- [ALC262_HP_RP5700] = "hp-rp5700",
- [ALC262_BENQ_ED8] = "benq",
- [ALC262_BENQ_T31] = "benq-t31",
- [ALC262_SONY_ASSAMD] = "sony-assamd",
- [ALC262_TOSHIBA_S06] = "toshiba-s06",
- [ALC262_TOSHIBA_RX1] = "toshiba-rx1",
- [ALC262_ULTRA] = "ultra",
- [ALC262_LENOVO_3000] = "lenovo-3000",
- [ALC262_NEC] = "nec",
- [ALC262_TYAN] = "tyan",
- [ALC262_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc262_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
- SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
- ALC262_HP_BPC),
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
- ALC262_HP_BPC),
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
- ALC262_HP_BPC),
- SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
- ALC262_AUTO),
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
- ALC262_HP_BPC),
- SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
- SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
- SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
- SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF),
- SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
- SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF),
- SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL),
- SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF),
- SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
- SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
- SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
- SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
- ALC262_HP_TC_T5735),
- SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
- SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
- SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
- SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
- SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
- SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
- SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
- SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
-#if 0 /* disable the quirk since model=auto works better in recent versions */
- SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
- ALC262_SONY_ASSAMD),
-#endif
- SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
- ALC262_TOSHIBA_RX1),
- SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
- SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
- SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
- SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN),
- SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1",
- ALC262_ULTRA),
- SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
- SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
- SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
- SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
- SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
- {}
-};
-
-static const struct alc_config_preset alc262_presets[] = {
- [ALC262_BASIC] = {
- .mixers = { alc262_base_mixer },
- .init_verbs = { alc262_init_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- },
- [ALC262_HIPPO] = {
- .mixers = { alc262_hippo_mixer },
- .init_verbs = { alc262_init_verbs, alc_hp15_unsol_verbs},
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .dig_out_nid = ALC262_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hippo_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_HIPPO_1] = {
- .mixers = { alc262_hippo1_mixer },
- .init_verbs = { alc262_init_verbs, alc262_hippo1_unsol_verbs},
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x02,
- .dig_out_nid = ALC262_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hippo1_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_FUJITSU] = {
- .mixers = { alc262_fujitsu_mixer },
- .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
- alc262_fujitsu_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .dig_out_nid = ALC262_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_fujitsu_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_fujitsu_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_HP_BPC] = {
- .mixers = { alc262_HP_BPC_mixer },
- .init_verbs = { alc262_HP_BPC_init_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_HP_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hp_bpc_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_HP_BPC_D7000_WF] = {
- .mixers = { alc262_HP_BPC_WildWest_mixer },
- .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_HP_D7000_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hp_wildwest_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_HP_BPC_D7000_WL] = {
- .mixers = { alc262_HP_BPC_WildWest_mixer,
- alc262_HP_BPC_WildWest_option_mixer },
- .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_HP_D7000_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hp_wildwest_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_HP_TC_T5735] = {
- .mixers = { alc262_hp_t5735_mixer },
- .init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hp_t5735_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_HP_RP5700] = {
- .mixers = { alc262_hp_rp5700_mixer },
- .init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_hp_rp5700_capture_source,
- },
- [ALC262_BENQ_ED8] = {
- .mixers = { alc262_base_mixer },
- .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- },
- [ALC262_SONY_ASSAMD] = {
- .mixers = { alc262_sony_mixer },
- .init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs},
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x02,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hippo_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_BENQ_T31] = {
- .mixers = { alc262_benq_t31_mixer },
- .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
- alc_hp15_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hippo_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_ULTRA] = {
- .mixers = { alc262_ultra_mixer },
- .cap_mixer = alc262_ultra_capture_mixer,
- .init_verbs = { alc262_ultra_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_ultra_capture_source,
- .adc_nids = alc262_adc_nids, /* ADC0 */
- .capsrc_nids = alc262_capsrc_nids,
- .num_adc_nids = 1, /* single ADC */
- .unsol_event = alc262_ultra_unsol_event,
- .init_hook = alc262_ultra_automute,
- },
- [ALC262_LENOVO_3000] = {
- .mixers = { alc262_lenovo_3000_mixer },
- .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
- alc262_lenovo_3000_unsol_verbs,
- alc262_lenovo_3000_init_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .dig_out_nid = ALC262_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_fujitsu_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_lenovo_3000_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_NEC] = {
- .mixers = { alc262_nec_mixer },
- .init_verbs = { alc262_nec_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- },
- [ALC262_TOSHIBA_S06] = {
- .mixers = { alc262_toshiba_s06_mixer },
- .init_verbs = { alc262_init_verbs, alc262_toshiba_s06_verbs,
- alc262_eapd_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .capsrc_nids = alc262_dmic_capsrc_nids,
- .dac_nids = alc262_dac_nids,
- .adc_nids = alc262_dmic_adc_nids, /* ADC0 */
- .num_adc_nids = 1, /* single ADC */
- .dig_out_nid = ALC262_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_toshiba_s06_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_TOSHIBA_RX1] = {
- .mixers = { alc262_toshiba_rx1_mixer },
- .init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_hippo_setup,
- .init_hook = alc_inithook,
- },
- [ALC262_TYAN] = {
- .mixers = { alc262_tyan_mixer },
- .init_verbs = { alc262_init_verbs, alc262_tyan_verbs},
- .num_dacs = ARRAY_SIZE(alc262_dac_nids),
- .dac_nids = alc262_dac_nids,
- .hp_nid = 0x02,
- .dig_out_nid = ALC262_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc262_modes),
- .channel_mode = alc262_modes,
- .input_mux = &alc262_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc262_tyan_setup,
- .init_hook = alc_hp_automute,
- },
-};
-
-static int patch_alc262(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int board_config;
- int err;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-#if 0
- /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is
- * under-run
- */
- {
- int tmp;
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
- tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
- }
-#endif
- alc_auto_parse_customize_define(codec);
-
- alc_fix_pll_init(codec, 0x20, 0x0a, 10);
-
- board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST,
- alc262_models,
- alc262_cfg_tbl);
-
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC262_AUTO;
- }
-
- if (board_config == ALC262_AUTO) {
- alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC262_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc262_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC262_BASIC;
- }
- }
-
- if (!spec->no_analog && has_cdefine_beep(codec)) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
- }
-
- if (board_config != ALC262_AUTO)
- setup_preset(codec, &alc262_presets[board_config]);
-
- spec->stream_analog_playback = &alc262_pcm_analog_playback;
- spec->stream_analog_capture = &alc262_pcm_analog_capture;
-
- spec->stream_digital_playback = &alc262_pcm_digital_playback;
- spec->stream_digital_capture = &alc262_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- int i;
- /* check whether the digital-mic has to be supported */
- for (i = 0; i < spec->input_mux->num_items; i++) {
- if (spec->input_mux->items[i].index >= 9)
- break;
- }
- if (i < spec->input_mux->num_items) {
- /* use only ADC0 */
- spec->adc_nids = alc262_dmic_adc_nids;
- spec->num_adc_nids = 1;
- spec->capsrc_nids = alc262_dmic_capsrc_nids;
- } else {
- /* all analog inputs */
- /* check whether NID 0x07 is valid */
- unsigned int wcap = get_wcaps(codec, 0x07);
-
- /* get type */
- wcap = get_wcaps_type(wcap);
- if (wcap != AC_WID_AUD_IN) {
- spec->adc_nids = alc262_adc_nids_alt;
- spec->num_adc_nids =
- ARRAY_SIZE(alc262_adc_nids_alt);
- spec->capsrc_nids = alc262_capsrc_nids_alt;
- } else {
- spec->adc_nids = alc262_adc_nids;
- spec->num_adc_nids =
- ARRAY_SIZE(alc262_adc_nids);
- spec->capsrc_nids = alc262_capsrc_nids;
- }
- }
- }
- if (!spec->cap_mixer && !spec->no_analog)
- set_capture_mixer(codec);
- if (!spec->no_analog && has_cdefine_beep(codec))
- set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
-
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
- spec->vmaster_nid = 0x0c;
-
- codec->patch_ops = alc_patch_ops;
- if (board_config == ALC262_AUTO)
- spec->init_hook = alc262_auto_init;
- spec->shutup = alc_eapd_shutup;
-
- alc_init_jacks(codec);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc262_loopbacks;
-#endif
-
- return 0;
-}
-
-/*
- * ALC268 channel source setting (2 channel)
- */
-#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID
-#define alc268_modes alc260_modes
-
-static const hda_nid_t alc268_dac_nids[2] = {
- /* front, hp */
- 0x02, 0x03
-};
-
-static const hda_nid_t alc268_adc_nids[2] = {
- /* ADC0-1 */
- 0x08, 0x07
-};
-
-static const hda_nid_t alc268_adc_nids_alt[1] = {
- /* ADC0 */
- 0x08
-};
-
-static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
-
-static const struct snd_kcontrol_new alc268_base_mixer[] = {
- /* output mixer control */
- HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
- { }
-};
-
-static const struct snd_kcontrol_new alc268_toshiba_mixer[] = {
- /* output mixer control */
- HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
- ALC262_HIPPO_MASTER_SWITCH,
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
- { }
-};
-
-/* bind Beep switches of both NID 0x0f and 0x10 */
-static const struct hda_bind_ctls alc268_bind_beep_sw = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x10, 3, 1, HDA_INPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc268_beep_mixer[] = {
- HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT),
- HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw),
- { }
-};
-
-static const struct hda_verb alc268_eapd_verbs[] = {
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-/* Toshiba specific */
-static const struct hda_verb alc268_toshiba_verbs[] = {
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { } /* end */
-};
-
-/* Acer specific */
-/* bind volumes of both NID 0x02 and 0x03 */
-static const struct hda_bind_ctls alc268_acer_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static void alc268_acer_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-#define alc268_acer_master_sw_get alc262_hp_master_sw_get
-#define alc268_acer_master_sw_put alc262_hp_master_sw_put
-
-static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
- /* output mixer control */
- HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x15,
- .info = snd_ctl_boolean_mono_info,
- .get = alc268_acer_master_sw_get,
- .put = alc268_acer_master_sw_put,
- },
- HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
- { }
-};
-
-static const struct snd_kcontrol_new alc268_acer_mixer[] = {
- /* output mixer control */
- HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
- .info = snd_ctl_boolean_mono_info,
- .get = alc268_acer_master_sw_get,
- .put = alc268_acer_master_sw_put,
- },
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
- { }
-};
-
-static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
- /* output mixer control */
- HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
- .info = snd_ctl_boolean_mono_info,
- .get = alc268_acer_master_sw_get,
- .put = alc268_acer_master_sw_put,
- },
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
- { }
-};
-
-static const struct hda_verb alc268_acer_aspire_one_verbs[] = {
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
- { }
-};
-
-static const struct hda_verb alc268_acer_verbs[] = {
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- { }
-};
-
-/* unsolicited event for HP jack sensing */
-#define alc268_toshiba_setup alc262_hippo_setup
-
-static void alc268_acer_lc_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 6;
- spec->auto_mic = 1;
-}
-
-static const struct snd_kcontrol_new alc268_dell_mixer[] = {
- /* output mixer control */
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { }
-};
-
-static const struct hda_verb alc268_dell_verbs[] = {
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
- { }
-};
-
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc268_dell_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { }
-};
-
-static const struct hda_verb alc267_quanta_il1_verbs[] = {
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
- { }
-};
-
-static void alc267_quanta_il1_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-}
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc268_base_init_verbs[] = {
- /* Unmute DAC0-1 and set vol = 0 */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /*
- * Set up output mixers (0x0c - 0x0e)
- */
- /* set vol=0 to output mixers */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- /* set PCBEEP vol = 0, mute connections */
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- /* Unmute Selector 23h,24h and set the default input to mic-in */
-
- {0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- { }
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc268_volume_init_verbs[] = {
- /* set output DAC */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
-
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- /* set PCBEEP vol = 0, mute connections */
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- { }
-};
-
-static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
- _DEFINE_CAPSRC(1),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc268_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
- _DEFINE_CAPSRC(2),
- { } /* end */
-};
-
-static const struct hda_input_mux alc268_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x3 },
- },
-};
-
-static const struct hda_input_mux alc268_acer_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- { "Line", 0x2 },
- },
-};
-
-static const struct hda_input_mux alc268_acer_dmic_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x6 },
- { "Line", 0x2 },
- },
-};
-
-#ifdef CONFIG_SND_DEBUG
-static const struct snd_kcontrol_new alc268_test_mixer[] = {
- /* Volume widgets */
- HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
- HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
- HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
- HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
- HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
- HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
- HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
- HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
- /* The below appears problematic on some hardwares */
- /*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
- HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
- HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
-
- /* Modes for retasking pin widgets */
- ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
-
- /* Controls for GPIO pins, assuming they are configured as outputs */
- ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
- ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
- ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
- ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
-
- /* Switches to allow the digital SPDIF output pin to be enabled.
- * The ALC268 does not have an SPDIF input.
- */
- ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
-
- /* A switch allowing EAPD to be enabled. Some laptops seem to use
- * this output to turn on an external amplifier.
- */
- ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
- ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
-
- { } /* end */
-};
-#endif
-
-/* create input playback/capture controls for the given pin */
-static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
- const char *ctlname, int idx)
-{
- hda_nid_t dac;
- int err;
-
- switch (nid) {
- case 0x14:
- case 0x16:
- dac = 0x02;
- break;
- case 0x15:
- case 0x1a: /* ALC259/269 only */
- case 0x1b: /* ALC259/269 only */
- case 0x21: /* ALC269vb has this pin, too */
- dac = 0x03;
- break;
- default:
- snd_printd(KERN_WARNING "hda_codec: "
- "ignoring pin 0x%x as unknown\n", nid);
- return 0;
- }
- if (spec->multiout.dac_nids[0] != dac &&
- spec->multiout.dac_nids[1] != dac) {
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname,
- HDA_COMPOSE_AMP_VAL(dac, 3, idx,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
- }
-
- if (nid != 0x16)
- err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname,
- HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
- else /* mono */
- err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname,
- HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT));
- if (err < 0)
- return err;
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- hda_nid_t nid;
- int err;
-
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- nid = cfg->line_out_pins[0];
- if (nid) {
- const char *name;
- if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
- name = "Speaker";
- else
- name = "Front";
- err = alc268_new_analog_output(spec, nid, name, 0);
- if (err < 0)
- return err;
- }
-
- nid = cfg->speaker_pins[0];
- if (nid == 0x1d) {
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, "Speaker",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
- } else if (nid) {
- err = alc268_new_analog_output(spec, nid, "Speaker", 0);
- if (err < 0)
- return err;
- }
- nid = cfg->hp_pins[0];
- if (nid) {
- err = alc268_new_analog_output(spec, nid, "Headphone", 0);
- if (err < 0)
- return err;
- }
-
- nid = cfg->line_out_pins[1] | cfg->line_out_pins[2];
- if (nid == 0x16) {
- err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, "Mono",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int alc268_auto_create_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- return alc_auto_create_input_ctls(codec, cfg, 0, 0x23, 0x24);
-}
-
-static void alc268_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type)
-{
- int idx;
-
- alc_set_pin_output(codec, nid, pin_type);
- if (nid == 0x14 || nid == 0x16)
- idx = 0;
- else
- idx = 1;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
-}
-
-static void alc268_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < spec->autocfg.line_outs; i++) {
- hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- alc268_auto_set_output_and_unmute(codec, nid, pin_type);
- }
-}
-
-static void alc268_auto_init_hp_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
- int i;
-
- for (i = 0; i < spec->autocfg.hp_outs; i++) {
- pin = spec->autocfg.hp_pins[i];
- alc268_auto_set_output_and_unmute(codec, pin, PIN_HP);
- }
- for (i = 0; i < spec->autocfg.speaker_outs; i++) {
- pin = spec->autocfg.speaker_pins[i];
- alc268_auto_set_output_and_unmute(codec, pin, PIN_OUT);
- }
- if (spec->autocfg.mono_out_pin)
- snd_hda_codec_write(codec, spec->autocfg.mono_out_pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-}
-
-static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0];
- hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
- hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
- unsigned int dac_vol1, dac_vol2;
-
- if (line_nid == 0x1d || speaker_nid == 0x1d) {
- snd_hda_codec_write(codec, speaker_nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
- /* mute mixer inputs from 0x1d */
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(1));
- snd_hda_codec_write(codec, 0x10, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(1));
- } else {
- /* unmute mixer inputs from 0x1d */
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
- snd_hda_codec_write(codec, 0x10, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
- }
-
- dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */
- if (line_nid == 0x14)
- dac_vol2 = AMP_OUT_ZERO;
- else if (line_nid == 0x15)
- dac_vol1 = AMP_OUT_ZERO;
- if (hp_nid == 0x14)
- dac_vol2 = AMP_OUT_ZERO;
- else if (hp_nid == 0x15)
- dac_vol1 = AMP_OUT_ZERO;
- if (line_nid != 0x16 || hp_nid != 0x16 ||
- spec->autocfg.line_out_pins[1] != 0x16 ||
- spec->autocfg.line_out_pins[2] != 0x16)
- dac_vol1 = dac_vol2 = AMP_OUT_ZERO;
-
- snd_hda_codec_write(codec, 0x02, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, dac_vol1);
- snd_hda_codec_write(codec, 0x03, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2);
-}
-
-/* pcm configuration: identical with ALC880 */
-#define alc268_pcm_analog_playback alc880_pcm_analog_playback
-#define alc268_pcm_analog_capture alc880_pcm_analog_capture
-#define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
-#define alc268_pcm_digital_playback alc880_pcm_digital_playback
-
-/*
- * BIOS auto configuration
- */
-static int alc268_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc268_ignore[] = { 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc268_ignore);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs) {
- if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
- spec->multiout.max_channels = 2;
- spec->no_analog = 1;
- goto dig_only;
- }
- return 0; /* can't find valid BIOS pin config */
- }
- err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc268_auto_create_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = 2;
-
- dig_only:
- /* digital only support output */
- alc_auto_parse_digital(codec);
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d)
- add_mixer(spec, alc268_beep_mixer);
-
- add_verb(spec, alc268_volume_init_verbs);
- spec->num_mux_defs = 2;
- spec->input_mux = &spec->private_imux[0];
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
-
- return 1;
-}
-
-#define alc268_auto_init_analog_input alc882_auto_init_analog_input
-#define alc268_auto_init_input_src alc882_auto_init_input_src
-
-/* init callback for auto-configuration model -- overriding the default init */
-static void alc268_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc268_auto_init_multi_out(codec);
- alc268_auto_init_hp_out(codec);
- alc268_auto_init_mono_speaker_out(codec);
- alc268_auto_init_analog_input(codec);
- alc268_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-/*
- * configuration and preset
- */
-static const char * const alc268_models[ALC268_MODEL_LAST] = {
- [ALC267_QUANTA_IL1] = "quanta-il1",
- [ALC268_3ST] = "3stack",
- [ALC268_TOSHIBA] = "toshiba",
- [ALC268_ACER] = "acer",
- [ALC268_ACER_DMIC] = "acer-dmic",
- [ALC268_ACER_ASPIRE_ONE] = "acer-aspire",
- [ALC268_DELL] = "dell",
- [ALC268_ZEPTO] = "zepto",
-#ifdef CONFIG_SND_DEBUG
- [ALC268_TEST] = "test",
-#endif
- [ALC268_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc268_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
- SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
- SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
- SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
- SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
- SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
- ALC268_ACER_ASPIRE_ONE),
- SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
- SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
- SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
- "Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
- /* almost compatible with toshiba but with optional digital outs;
- * auto-probing seems working fine
- */
- SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
- ALC268_AUTO),
- SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
- SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
- SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
- SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
- {}
-};
-
-/* Toshiba laptops have no unique PCI SSID but only codec SSID */
-static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO),
- SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO),
- SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
- ALC268_TOSHIBA),
- {}
-};
-
-static const struct alc_config_preset alc268_presets[] = {
- [ALC267_QUANTA_IL1] = {
- .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc267_quanta_il1_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc267_quanta_il1_setup,
- .init_hook = alc_inithook,
- },
- [ALC268_3ST] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
- .init_verbs = { alc268_base_init_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x03,
- .dig_out_nid = ALC268_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .input_mux = &alc268_capture_source,
- },
- [ALC268_TOSHIBA] = {
- .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc268_toshiba_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .input_mux = &alc268_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc268_toshiba_setup,
- .init_hook = alc_inithook,
- },
- [ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc268_acer_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x02,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .input_mux = &alc268_acer_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc268_acer_setup,
- .init_hook = alc_inithook,
- },
- [ALC268_ACER_DMIC] = {
- .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc268_acer_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x02,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .input_mux = &alc268_acer_dmic_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc268_acer_setup,
- .init_hook = alc_inithook,
- },
- [ALC268_ACER_ASPIRE_ONE] = {
- .mixers = { alc268_acer_aspire_one_mixer,
- alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc268_acer_aspire_one_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc268_acer_lc_setup,
- .init_hook = alc_inithook,
- },
- [ALC268_DELL] = {
- .mixers = { alc268_dell_mixer, alc268_beep_mixer,
- alc268_capture_nosrc_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc268_dell_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x02,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc268_dell_setup,
- .init_hook = alc_inithook,
- },
- [ALC268_ZEPTO] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
- alc268_beep_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc268_toshiba_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x03,
- .dig_out_nid = ALC268_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .input_mux = &alc268_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc268_toshiba_setup,
- .init_hook = alc_inithook,
- },
-#ifdef CONFIG_SND_DEBUG
- [ALC268_TEST] = {
- .mixers = { alc268_test_mixer, alc268_capture_mixer },
- .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
- alc268_volume_init_verbs },
- .num_dacs = ARRAY_SIZE(alc268_dac_nids),
- .dac_nids = alc268_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
- .adc_nids = alc268_adc_nids_alt,
- .capsrc_nids = alc268_capsrc_nids,
- .hp_nid = 0x03,
- .dig_out_nid = ALC268_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc268_modes),
- .channel_mode = alc268_modes,
- .input_mux = &alc268_capture_source,
- },
-#endif
-};
-
-static int patch_alc268(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int board_config;
- int i, has_beep, err;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- board_config = snd_hda_check_board_config(codec, ALC268_MODEL_LAST,
- alc268_models,
- alc268_cfg_tbl);
-
- if (board_config < 0 || board_config >= ALC268_MODEL_LAST)
- board_config = snd_hda_check_board_codec_sid_config(codec,
- ALC268_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
-
- if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC268_AUTO;
- }
-
- if (board_config == ALC268_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc268_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC268_3ST;
- }
- }
-
- if (board_config != ALC268_AUTO)
- setup_preset(codec, &alc268_presets[board_config]);
-
- spec->stream_analog_playback = &alc268_pcm_analog_playback;
- spec->stream_analog_capture = &alc268_pcm_analog_capture;
- spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture;
-
- spec->stream_digital_playback = &alc268_pcm_digital_playback;
-
- has_beep = 0;
- for (i = 0; i < spec->num_mixers; i++) {
- if (spec->mixers[i] == alc268_beep_mixer) {
- has_beep = 1;
- break;
- }
- }
-
- if (has_beep) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
- if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
- /* override the amp caps for beep generator */
- snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
- (0x0c << AC_AMPCAP_OFFSET_SHIFT) |
- (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (0 << AC_AMPCAP_MUTE_SHIFT));
- }
-
- if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
- /* check whether NID 0x07 is valid */
- unsigned int wcap = get_wcaps(codec, 0x07);
-
- spec->capsrc_nids = alc268_capsrc_nids;
- /* get type */
- wcap = get_wcaps_type(wcap);
- if (spec->auto_mic ||
- wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
- spec->adc_nids = alc268_adc_nids_alt;
- spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
- if (spec->auto_mic)
- fixup_automic_adc(codec);
- if (spec->auto_mic || spec->input_mux->num_items == 1)
- add_mixer(spec, alc268_capture_nosrc_mixer);
- else
- add_mixer(spec, alc268_capture_alt_mixer);
- } else {
- spec->adc_nids = alc268_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
- add_mixer(spec, alc268_capture_mixer);
- }
- }
-
- spec->vmaster_nid = 0x02;
-
- codec->patch_ops = alc_patch_ops;
- if (board_config == ALC268_AUTO)
- spec->init_hook = alc268_auto_init;
- spec->shutup = alc_eapd_shutup;
-
- alc_init_jacks(codec);
-
- return 0;
-}
-
-/*
- * ALC269 channel source setting (2 channel)
- */
-#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID
-
-#define alc269_dac_nids alc260_dac_nids
-
-static const hda_nid_t alc269_adc_nids[1] = {
- /* ADC1 */
- 0x08,
-};
-
-static const hda_nid_t alc269_capsrc_nids[1] = {
- 0x23,
-};
-
-static const hda_nid_t alc269vb_adc_nids[1] = {
- /* ADC1 */
- 0x09,
-};
-
-static const hda_nid_t alc269vb_capsrc_nids[1] = {
- 0x22,
-};
-
-static const hda_nid_t alc269_adc_candidates[] = {
- 0x08, 0x09, 0x07, 0x11,
-};
-
-#define alc269_modes alc260_modes
-#define alc269_capture_source alc880_lg_lw_capture_source
-
-static const struct snd_kcontrol_new alc269_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
- /* output mixer control */
- HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = alc268_acer_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { }
-};
-
-static const struct snd_kcontrol_new alc269_lifebook_mixer[] = {
- /* output mixer control */
- HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = alc268_acer_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- },
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
- HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT),
- { }
-};
-
-static const struct snd_kcontrol_new alc269_laptop_mixer[] = {
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc269_asus_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-/* capture mixer elements */
-static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- { } /* end */
-};
-
-/* FSC amilo */
-#define alc269_fujitsu_mixer alc269_laptop_mixer
-
-static const struct hda_verb alc269_quanta_fl1_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- { }
-};
-
-static const struct hda_verb alc269_lifebook_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- { }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
-{
- alc_hp_automute(codec);
-
- snd_hda_codec_write(codec, 0x20, 0,
- AC_VERB_SET_COEF_INDEX, 0x0c);
- snd_hda_codec_write(codec, 0x20, 0,
- AC_VERB_SET_PROC_COEF, 0x680);
-
- snd_hda_codec_write(codec, 0x20, 0,
- AC_VERB_SET_COEF_INDEX, 0x0c);
- snd_hda_codec_write(codec, 0x20, 0,
- AC_VERB_SET_PROC_COEF, 0x480);
-}
-
-#define alc269_lifebook_speaker_automute \
- alc269_quanta_fl1_speaker_automute
-
-static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
-{
- unsigned int present_laptop;
- unsigned int present_dock;
-
- present_laptop = snd_hda_jack_detect(codec, 0x18);
- present_dock = snd_hda_jack_detect(codec, 0x1b);
-
- /* Laptop mic port overrides dock mic port, design decision */
- if (present_dock)
- snd_hda_codec_write(codec, 0x23, 0,
- AC_VERB_SET_CONNECT_SEL, 0x3);
- if (present_laptop)
- snd_hda_codec_write(codec, 0x23, 0,
- AC_VERB_SET_CONNECT_SEL, 0x0);
- if (!present_dock && !present_laptop)
- snd_hda_codec_write(codec, 0x23, 0,
- AC_VERB_SET_CONNECT_SEL, 0x1);
-}
-
-static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc269_quanta_fl1_speaker_automute(codec);
- break;
- case ALC880_MIC_EVENT:
- alc_mic_automute(codec);
- break;
- }
-}
-
-static void alc269_lifebook_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc269_lifebook_speaker_automute(codec);
- if ((res >> 26) == ALC880_MIC_EVENT)
- alc269_lifebook_mic_autoswitch(codec);
-}
-
-static void alc269_quanta_fl1_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
-{
- alc269_quanta_fl1_speaker_automute(codec);
- alc_mic_automute(codec);
-}
-
-static void alc269_lifebook_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.hp_pins[1] = 0x1a;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
-}
-
-static void alc269_lifebook_init_hook(struct hda_codec *codec)
-{
- alc269_lifebook_speaker_automute(codec);
- alc269_lifebook_mic_autoswitch(codec);
-}
-
-static const struct hda_verb alc269_laptop_dmic_init_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc269_laptop_amic_init_verbs[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x06},
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc271_acer_dmic_verbs[] = {
- {0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
- {0x20, AC_VERB_SET_PROC_COEF, 0x4000},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x22, AC_VERB_SET_CONNECT_SEL, 6},
- { }
-};
-
-static void alc269_laptop_amic_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-static void alc269_laptop_dmic_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 5;
- spec->auto_mic = 1;
-}
-
-static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 6;
- spec->auto_mic = 1;
-}
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc269_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /*
- * Set up output mixers (0x02 - 0x03)
- */
- /* set vol=0 to output mixers */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* FIXME: use Mux-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* set EAPD */
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-static const struct hda_verb alc269vb_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /*
- * Set up output mixers (0x02 - 0x03)
- */
- /* set vol=0 to output mixers */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* FIXME: use Mux-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x22, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* set EAPD */
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-#define alc269_auto_create_multi_out_ctls \
- alc268_auto_create_multi_out_ctls
-#define alc269_auto_create_input_ctls \
- alc268_auto_create_input_ctls
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc269_loopbacks alc880_loopbacks
-#endif
-
-/* pcm configuration: identical with ALC880 */
-#define alc269_pcm_analog_playback alc880_pcm_analog_playback
-#define alc269_pcm_analog_capture alc880_pcm_analog_capture
-#define alc269_pcm_digital_playback alc880_pcm_digital_playback
-#define alc269_pcm_digital_capture alc880_pcm_digital_capture
-
-static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 8,
- .rates = SNDRV_PCM_RATE_44100, /* fixed rate */
- /* NID is set in alc_build_pcms */
- .ops = {
- .open = alc880_playback_pcm_open,
- .prepare = alc880_playback_pcm_prepare,
- .cleanup = alc880_playback_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .rates = SNDRV_PCM_RATE_44100, /* fixed rate */
- /* NID is set in alc_build_pcms */
-};
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static int alc269_mic2_for_mute_led(struct hda_codec *codec)
-{
- switch (codec->subsystem_id) {
- case 0x103c1586:
- return 1;
- }
- return 0;
-}
-
-static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid)
-{
- /* update mute-LED according to the speaker mute state */
- if (nid == 0x01 || nid == 0x14) {
- int pinval;
- if (snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0) &
- HDA_AMP_MUTE)
- pinval = 0x24;
- else
- pinval = 0x20;
- /* mic2 vref pin is used for mute LED control */
- snd_hda_codec_update_cache(codec, 0x19, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinval);
- }
- return alc_check_power_status(codec, nid);
-}
-#endif /* CONFIG_SND_HDA_POWER_SAVE */
-
-static int alc275_setup_dual_adc(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- if (codec->vendor_id != 0x10ec0275 || !spec->auto_mic)
- return 0;
- if ((spec->ext_mic.pin >= 0x18 && spec->int_mic.pin <= 0x13) ||
- (spec->ext_mic.pin <= 0x12 && spec->int_mic.pin >= 0x18)) {
- if (spec->ext_mic.pin <= 0x12) {
- spec->private_adc_nids[0] = 0x08;
- spec->private_adc_nids[1] = 0x11;
- spec->private_capsrc_nids[0] = 0x23;
- spec->private_capsrc_nids[1] = 0x22;
- } else {
- spec->private_adc_nids[0] = 0x11;
- spec->private_adc_nids[1] = 0x08;
- spec->private_capsrc_nids[0] = 0x22;
- spec->private_capsrc_nids[1] = 0x23;
- }
- spec->adc_nids = spec->private_adc_nids;
- spec->capsrc_nids = spec->private_capsrc_nids;
- spec->num_adc_nids = 2;
- spec->dual_adc_switch = 1;
- snd_printdd("realtek: enabling dual ADC switchg (%02x:%02x)\n",
- spec->adc_nids[0], spec->adc_nids[1]);
- return 1;
- }
- return 0;
-}
-
-/* different alc269-variants */
-enum {
- ALC269_TYPE_NORMAL,
- ALC269_TYPE_ALC258,
- ALC269_TYPE_ALC259,
- ALC269_TYPE_ALC269VB,
- ALC269_TYPE_ALC270,
- ALC269_TYPE_ALC271X,
-};
-
-/*
- * BIOS auto configuration
- */
-static int alc269_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc269_ignore[] = { 0x1d, 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc269_ignore);
- if (err < 0)
- return err;
-
- err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (spec->codec_variant == ALC269_TYPE_NORMAL)
- err = alc269_auto_create_input_ctls(codec, &spec->autocfg);
- else
- err = alc_auto_create_input_ctls(codec, &spec->autocfg, 0,
- 0x22, 0);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- alc_auto_parse_digital(codec);
-
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- if (spec->codec_variant != ALC269_TYPE_NORMAL) {
- add_verb(spec, alc269vb_init_verbs);
- alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21);
- } else {
- add_verb(spec, alc269_init_verbs);
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
- }
-
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- if (!alc275_setup_dual_adc(codec))
- fillup_priv_adc_nids(codec, alc269_adc_candidates,
- sizeof(alc269_adc_candidates));
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- if (!spec->cap_mixer && !spec->no_analog)
- set_capture_mixer(codec);
-
- return 1;
-}
-
-#define alc269_auto_init_multi_out alc268_auto_init_multi_out
-#define alc269_auto_init_hp_out alc268_auto_init_hp_out
-#define alc269_auto_init_analog_input alc882_auto_init_analog_input
-#define alc269_auto_init_input_src alc882_auto_init_input_src
-
-
-/* init callback for auto-configuration model -- overriding the default init */
-static void alc269_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc269_auto_init_multi_out(codec);
- alc269_auto_init_hp_out(codec);
- alc269_auto_init_analog_input(codec);
- if (!spec->dual_adc_switch)
- alc269_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-static void alc269_toggle_power_output(struct hda_codec *codec, int power_up)
-{
- int val = alc_read_coef_idx(codec, 0x04);
- if (power_up)
- val |= 1 << 11;
- else
- val &= ~(1 << 11);
- alc_write_coef_idx(codec, 0x04, val);
-}
-
-static void alc269_shutup(struct hda_codec *codec)
-{
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017)
- alc269_toggle_power_output(codec, 0);
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
- alc269_toggle_power_output(codec, 0);
- msleep(150);
- }
-}
-
-#ifdef SND_HDA_NEEDS_RESUME
-static int alc269_resume(struct hda_codec *codec)
-{
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
- alc269_toggle_power_output(codec, 0);
- msleep(150);
- }
-
- codec->patch_ops.init(codec);
-
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
- alc269_toggle_power_output(codec, 1);
- msleep(200);
- }
-
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018)
- alc269_toggle_power_output(codec, 1);
-
- snd_hda_codec_resume_amp(codec);
- snd_hda_codec_resume_cache(codec);
- hda_call_check_power_status(codec, 0x01);
- return 0;
-}
-#endif /* SND_HDA_NEEDS_RESUME */
-
-static void alc269_fixup_hweq(struct hda_codec *codec,
- const struct alc_fixup *fix, int action)
-{
- int coef;
-
- if (action != ALC_FIXUP_ACT_INIT)
- return;
- coef = alc_read_coef_idx(codec, 0x1e);
- alc_write_coef_idx(codec, 0x1e, coef | 0x80);
-}
-
-static void alc271_fixup_dmic(struct hda_codec *codec,
- const struct alc_fixup *fix, int action)
-{
- static const struct hda_verb verbs[] = {
- {0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
- {0x20, AC_VERB_SET_PROC_COEF, 0x4000},
- {}
- };
- unsigned int cfg;
-
- if (strcmp(codec->chip_name, "ALC271X"))
- return;
- cfg = snd_hda_codec_get_pincfg(codec, 0x12);
- if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
- snd_hda_sequence_write(codec, verbs);
-}
-
-enum {
- ALC269_FIXUP_SONY_VAIO,
- ALC275_FIXUP_SONY_VAIO_GPIO2,
- ALC269_FIXUP_DELL_M101Z,
- ALC269_FIXUP_SKU_IGNORE,
- ALC269_FIXUP_ASUS_G73JW,
- ALC269_FIXUP_LENOVO_EAPD,
- ALC275_FIXUP_SONY_HWEQ,
- ALC271_FIXUP_DMIC,
-};
-
-static const struct alc_fixup alc269_fixups[] = {
- [ALC269_FIXUP_SONY_VAIO] = {
- .type = ALC_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
- {}
- }
- },
- [ALC275_FIXUP_SONY_VAIO_GPIO2] = {
- .type = ALC_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- {0x01, AC_VERB_SET_GPIO_MASK, 0x04},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
- { }
- },
- .chained = true,
- .chain_id = ALC269_FIXUP_SONY_VAIO
- },
- [ALC269_FIXUP_DELL_M101Z] = {
- .type = ALC_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- /* Enables internal speaker */
- {0x20, AC_VERB_SET_COEF_INDEX, 13},
- {0x20, AC_VERB_SET_PROC_COEF, 0x4040},
- {}
- }
- },
- [ALC269_FIXUP_SKU_IGNORE] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
- },
- [ALC269_FIXUP_ASUS_G73JW] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x17, 0x99130111 }, /* subwoofer */
- { }
- }
- },
- [ALC269_FIXUP_LENOVO_EAPD] = {
- .type = ALC_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0},
- {}
- }
- },
- [ALC275_FIXUP_SONY_HWEQ] = {
- .type = ALC_FIXUP_FUNC,
- .v.func = alc269_fixup_hweq,
- .chained = true,
- .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
- },
- [ALC271_FIXUP_DMIC] = {
- .type = ALC_FIXUP_FUNC,
- .v.func = alc271_fixup_dmic,
- },
-};
-
-static const struct snd_pci_quirk alc269_fixup_tbl[] = {
- SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
- SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
- SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
- SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
- SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
- SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
- SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
- SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
- {}
-};
-
-
-/*
- * configuration and preset
- */
-static const char * const alc269_models[ALC269_MODEL_LAST] = {
- [ALC269_BASIC] = "basic",
- [ALC269_QUANTA_FL1] = "quanta",
- [ALC269_AMIC] = "laptop-amic",
- [ALC269_DMIC] = "laptop-dmic",
- [ALC269_FUJITSU] = "fujitsu",
- [ALC269_LIFEBOOK] = "lifebook",
- [ALC269_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc269_cfg_tbl[] = {
- SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
- SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER),
- SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
- ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
- SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
- ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
- SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
- SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
- SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
- SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
- SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
- SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC),
- SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC),
- SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC),
- SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC),
- {}
-};
-
-static const struct alc_config_preset alc269_presets[] = {
- [ALC269_BASIC] = {
- .mixers = { alc269_base_mixer },
- .init_verbs = { alc269_init_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .input_mux = &alc269_capture_source,
- },
- [ALC269_QUANTA_FL1] = {
- .mixers = { alc269_quanta_fl1_mixer },
- .init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .input_mux = &alc269_capture_source,
- .unsol_event = alc269_quanta_fl1_unsol_event,
- .setup = alc269_quanta_fl1_setup,
- .init_hook = alc269_quanta_fl1_init_hook,
- },
- [ALC269_AMIC] = {
- .mixers = { alc269_laptop_mixer },
- .cap_mixer = alc269_laptop_analog_capture_mixer,
- .init_verbs = { alc269_init_verbs,
- alc269_laptop_amic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc269_laptop_amic_setup,
- .init_hook = alc_inithook,
- },
- [ALC269_DMIC] = {
- .mixers = { alc269_laptop_mixer },
- .cap_mixer = alc269_laptop_digital_capture_mixer,
- .init_verbs = { alc269_init_verbs,
- alc269_laptop_dmic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc269_laptop_dmic_setup,
- .init_hook = alc_inithook,
- },
- [ALC269VB_AMIC] = {
- .mixers = { alc269vb_laptop_mixer },
- .cap_mixer = alc269vb_laptop_analog_capture_mixer,
- .init_verbs = { alc269vb_init_verbs,
- alc269vb_laptop_amic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc269vb_laptop_amic_setup,
- .init_hook = alc_inithook,
- },
- [ALC269VB_DMIC] = {
- .mixers = { alc269vb_laptop_mixer },
- .cap_mixer = alc269vb_laptop_digital_capture_mixer,
- .init_verbs = { alc269vb_init_verbs,
- alc269vb_laptop_dmic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc269vb_laptop_dmic_setup,
- .init_hook = alc_inithook,
- },
- [ALC269_FUJITSU] = {
- .mixers = { alc269_fujitsu_mixer },
- .cap_mixer = alc269_laptop_digital_capture_mixer,
- .init_verbs = { alc269_init_verbs,
- alc269_laptop_dmic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc269_laptop_dmic_setup,
- .init_hook = alc_inithook,
- },
- [ALC269_LIFEBOOK] = {
- .mixers = { alc269_lifebook_mixer },
- .init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .input_mux = &alc269_capture_source,
- .unsol_event = alc269_lifebook_unsol_event,
- .setup = alc269_lifebook_setup,
- .init_hook = alc269_lifebook_init_hook,
- },
- [ALC271_ACER] = {
- .mixers = { alc269_asus_mixer },
- .cap_mixer = alc269vb_laptop_digital_capture_mixer,
- .init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs },
- .num_dacs = ARRAY_SIZE(alc269_dac_nids),
- .dac_nids = alc269_dac_nids,
- .adc_nids = alc262_dmic_adc_nids,
- .num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids),
- .capsrc_nids = alc262_dmic_capsrc_nids,
- .num_channel_mode = ARRAY_SIZE(alc269_modes),
- .channel_mode = alc269_modes,
- .input_mux = &alc269_capture_source,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc269vb_laptop_dmic_setup,
- .init_hook = alc_inithook,
- },
-};
-
-static int alc269_fill_coef(struct hda_codec *codec)
-{
- int val;
-
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) < 0x015) {
- alc_write_coef_idx(codec, 0xf, 0x960b);
- alc_write_coef_idx(codec, 0xe, 0x8817);
- }
-
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x016) {
- alc_write_coef_idx(codec, 0xf, 0x960b);
- alc_write_coef_idx(codec, 0xe, 0x8814);
- }
-
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
- val = alc_read_coef_idx(codec, 0x04);
- /* Power up output pin */
- alc_write_coef_idx(codec, 0x04, val | (1<<11));
- }
-
- if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
- val = alc_read_coef_idx(codec, 0xd);
- if ((val & 0x0c00) >> 10 != 0x1) {
- /* Capless ramp up clock control */
- alc_write_coef_idx(codec, 0xd, val | (1<<10));
- }
- val = alc_read_coef_idx(codec, 0x17);
- if ((val & 0x01c0) >> 6 != 0x4) {
- /* Class D power on reset */
- alc_write_coef_idx(codec, 0x17, val | (1<<7));
- }
- }
-
- val = alc_read_coef_idx(codec, 0xd); /* Class D */
- alc_write_coef_idx(codec, 0xd, val | (1<<14));
-
- val = alc_read_coef_idx(codec, 0x4); /* HP */
- alc_write_coef_idx(codec, 0x4, val | (1<<11));
-
- return 0;
-}
-
-static int patch_alc269(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int board_config, coef;
- int err;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- alc_auto_parse_customize_define(codec);
-
- if (codec->vendor_id == 0x10ec0269) {
- coef = alc_read_coef_idx(codec, 0);
- if ((coef & 0x00f0) == 0x0010) {
- if (codec->bus->pci->subsystem_vendor == 0x1025 &&
- spec->cdefine.platform_type == 1) {
- alc_codec_rename(codec, "ALC271X");
- spec->codec_variant = ALC269_TYPE_ALC271X;
- } else if ((coef & 0xf000) == 0x1000) {
- spec->codec_variant = ALC269_TYPE_ALC270;
- } else if ((coef & 0xf000) == 0x2000) {
- alc_codec_rename(codec, "ALC259");
- spec->codec_variant = ALC269_TYPE_ALC259;
- } else if ((coef & 0xf000) == 0x3000) {
- alc_codec_rename(codec, "ALC258");
- spec->codec_variant = ALC269_TYPE_ALC258;
- } else {
- alc_codec_rename(codec, "ALC269VB");
- spec->codec_variant = ALC269_TYPE_ALC269VB;
- }
- } else
- alc_fix_pll_init(codec, 0x20, 0x04, 15);
- alc269_fill_coef(codec);
- }
-
- board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST,
- alc269_models,
- alc269_cfg_tbl);
-
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC269_AUTO;
- }
-
- if (board_config == ALC269_AUTO) {
- alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC269_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc269_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC269_BASIC;
- }
- }
-
- if (has_cdefine_beep(codec)) {
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
- }
-
- if (board_config != ALC269_AUTO)
- setup_preset(codec, &alc269_presets[board_config]);
-
- if (board_config == ALC269_QUANTA_FL1) {
- /* Due to a hardware problem on Lenovo Ideadpad, we need to
- * fix the sample rate of analog I/O to 44.1kHz
- */
- spec->stream_analog_playback = &alc269_44k_pcm_analog_playback;
- spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
- } else if (spec->dual_adc_switch) {
- spec->stream_analog_playback = &alc269_pcm_analog_playback;
- /* switch ADC dynamically */
- spec->stream_analog_capture = &dualmic_pcm_analog_capture;
- } else {
- spec->stream_analog_playback = &alc269_pcm_analog_playback;
- spec->stream_analog_capture = &alc269_pcm_analog_capture;
- }
- spec->stream_digital_playback = &alc269_pcm_digital_playback;
- spec->stream_digital_capture = &alc269_pcm_digital_capture;
-
- if (!spec->adc_nids) { /* wasn't filled automatically? use default */
- if (spec->codec_variant == ALC269_TYPE_NORMAL) {
- spec->adc_nids = alc269_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
- spec->capsrc_nids = alc269_capsrc_nids;
- } else {
- spec->adc_nids = alc269vb_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
- spec->capsrc_nids = alc269vb_capsrc_nids;
- }
- }
-
- if (!spec->cap_mixer)
- set_capture_mixer(codec);
- if (has_cdefine_beep(codec))
- set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
-
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
- spec->vmaster_nid = 0x02;
-
- codec->patch_ops = alc_patch_ops;
-#ifdef SND_HDA_NEEDS_RESUME
- codec->patch_ops.resume = alc269_resume;
-#endif
- if (board_config == ALC269_AUTO)
- spec->init_hook = alc269_auto_init;
- spec->shutup = alc269_shutup;
-
- alc_init_jacks(codec);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc269_loopbacks;
- if (alc269_mic2_for_mute_led(codec))
- codec->patch_ops.check_power_status = alc269_mic2_mute_check_ps;
-#endif
-
- return 0;
-}
-
-/*
- * ALC861 channel source setting (2/6 channel selection for 3-stack)
- */
-
-/*
- * set the path ways for 2 channel output
- * need to set the codec line out and mic 1 pin widgets to inputs
- */
-static const struct hda_verb alc861_threestack_ch2_init[] = {
- /* set pin widget 1Ah (line in) for input */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* set pin widget 18h (mic1/2) for input, for mic also enable
- * the vref
- */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
-#if 0
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
-#endif
- { } /* end */
-};
-/*
- * 6ch mode
- * need to set the codec line out and mic 1 pin widgets to outputs
- */
-static const struct hda_verb alc861_threestack_ch6_init[] = {
- /* set pin widget 1Ah (line in) for output (Back Surround)*/
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* set pin widget 18h (mic1) for output (CLFE)*/
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-
- { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
-
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-#if 0
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
-#endif
- { } /* end */
-};
-
-static const struct hda_channel_mode alc861_threestack_modes[2] = {
- { 2, alc861_threestack_ch2_init },
- { 6, alc861_threestack_ch6_init },
-};
-/* Set mic1 as input and unmute the mixer */
-static const struct hda_verb alc861_uniwill_m31_ch2_init[] = {
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
- { } /* end */
-};
-/* Set mic1 as output and mute mixer */
-static const struct hda_verb alc861_uniwill_m31_ch4_init[] = {
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
- { } /* end */
-};
-
-static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
- { 2, alc861_uniwill_m31_ch2_init },
- { 4, alc861_uniwill_m31_ch4_init },
-};
-
-/* Set mic1 and line-in as input and unmute the mixer */
-static const struct hda_verb alc861_asus_ch2_init[] = {
- /* set pin widget 1Ah (line in) for input */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* set pin widget 18h (mic1/2) for input, for mic also enable
- * the vref
- */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
-#if 0
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
-#endif
- { } /* end */
-};
-/* Set mic1 nad line-in as output and mute mixer */
-static const struct hda_verb alc861_asus_ch6_init[] = {
- /* set pin widget 1Ah (line in) for output (Back Surround)*/
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
- /* set pin widget 18h (mic1) for output (CLFE)*/
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
- { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
- { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
-
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-#if 0
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
-#endif
- { } /* end */
-};
-
-static const struct hda_channel_mode alc861_asus_modes[2] = {
- { 2, alc861_asus_ch2_init },
- { 6, alc861_asus_ch6_init },
-};
-
-/* patch-ALC861 */
-
-static const struct snd_kcontrol_new alc861_base_mixer[] = {
- /* output mixer control */
- HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
-
- /*Input mixer control */
- /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
-
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_3ST_mixer[] = {
- /* output mixer control */
- HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
- /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
-
- /* Input mixer control */
- /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
-
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- .private_value = ARRAY_SIZE(alc861_threestack_modes),
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_toshiba_mixer[] = {
- /* output mixer control */
- HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
-
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
- /* output mixer control */
- HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
- /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
-
- /* Input mixer control */
- /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
-
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc861_asus_mixer[] = {
- /* output mixer control */
- HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
-
- /* Input mixer control */
- HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
-
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- .private_value = ARRAY_SIZE(alc861_asus_modes),
- },
- { }
-};
-
-/* additional mixer */
-static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
- HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
- { }
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc861_base_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- /* port-A for surround (rear panel) */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-C for line-in (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* port-D for Front */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-E for HP out (front panel) */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* route front PCM to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-H for side (rear panel) */
- { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* route front mic to ADC1*/
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Unmute DAC0~3 & spdif out*/
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Unmute Mixer 14 (mic) 1c (Line in)*/
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Unmute Stereo Mixer 15 */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* hp used DAC 3 (Front) */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-
- { }
-};
-
-static const struct hda_verb alc861_threestack_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- /* port-A for surround (rear panel) */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- /* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-C for line-in (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* port-D for Front */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-E for HP out (front panel) */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
- /* route front PCM to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- /* port-H for side (rear panel) */
- { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- /* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* route front mic to ADC1*/
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute DAC0~3 & spdif out*/
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Unmute Mixer 14 (mic) 1c (Line in)*/
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Unmute Stereo Mixer 15 */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* hp used DAC 3 (Front) */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- { }
-};
-
-static const struct hda_verb alc861_uniwill_m31_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- /* port-A for surround (rear panel) */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- /* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-C for line-in (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* port-D for Front */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-E for HP out (front panel) */
- /* this has to be set to VREF80 */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* route front PCM to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- /* port-H for side (rear panel) */
- { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- /* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* route front mic to ADC1*/
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute DAC0~3 & spdif out*/
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Unmute Mixer 14 (mic) 1c (Line in)*/
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Unmute Stereo Mixer 15 */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* hp used DAC 3 (Front) */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- { }
-};
-
-static const struct hda_verb alc861_asus_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- /* port-A for surround (rear panel)
- * according to codec#0 this is the HP jack
- */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
- /* route front PCM to HP */
- { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
- /* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-C for line-in (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* port-D for Front */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-E for HP out (front panel) */
- /* this has to be set to VREF80 */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* route front PCM to HP */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
- /* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* port-H for side (rear panel) */
- { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* route front mic to ADC1*/
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute DAC0~3 & spdif out*/
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Mixer 14 (mic) 1c (Line in)*/
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Unmute Stereo Mixer 15 */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
-
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* hp used DAC 3 (Front) */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- { }
-};
-
-/* additional init verbs for ASUS laptops */
-static const struct hda_verb alc861_asus_laptop_init_verbs[] = {
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
- { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
- { }
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc861_auto_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- /* {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Unmute DAC0~3 & spdif out*/
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Unmute Mixer 14 (mic) 1c (Line in)*/
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Unmute Stereo Mixer 15 */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c},
-
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, /* set Mic 1 */
-
- { }
-};
-
-static const struct hda_verb alc861_toshiba_init_verbs[] = {
- {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
-
- { }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc861_toshiba_automute(struct hda_codec *codec)
-{
- unsigned int present = snd_hda_jack_detect(codec, 0x0f);
-
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
- HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
-}
-
-static void alc861_toshiba_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc861_toshiba_automute(codec);
-}
-
-/* pcm configuration: identical with ALC880 */
-#define alc861_pcm_analog_playback alc880_pcm_analog_playback
-#define alc861_pcm_analog_capture alc880_pcm_analog_capture
-#define alc861_pcm_digital_playback alc880_pcm_digital_playback
-#define alc861_pcm_digital_capture alc880_pcm_digital_capture
-
-
-#define ALC861_DIGOUT_NID 0x07
-
-static const struct hda_channel_mode alc861_8ch_modes[1] = {
- { 8, NULL }
-};
-
-static const hda_nid_t alc861_dac_nids[4] = {
- /* front, surround, clfe, side */
- 0x03, 0x06, 0x05, 0x04
-};
-
-static const hda_nid_t alc660_dac_nids[3] = {
- /* front, clfe, surround */
- 0x03, 0x05, 0x06
-};
-
-static const hda_nid_t alc861_adc_nids[1] = {
- /* ADC0-2 */
- 0x08,
-};
-
-static const struct hda_input_mux alc861_capture_source = {
- .num_items = 5,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x3 },
- { "Line", 0x1 },
- { "CD", 0x4 },
- { "Mixer", 0x5 },
- },
-};
-
-static hda_nid_t alc861_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t mix, srcs[5];
- int i, j, num;
-
- if (snd_hda_get_connections(codec, pin, &mix, 1) != 1)
- return 0;
- num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
- if (num < 0)
- return 0;
- for (i = 0; i < num; i++) {
- unsigned int type;
- type = get_wcaps_type(get_wcaps(codec, srcs[i]));
- if (type != AC_WID_AUD_OUT)
- continue;
- for (j = 0; j < spec->multiout.num_dacs; j++)
- if (spec->multiout.dac_nids[j] == srcs[i])
- break;
- if (j >= spec->multiout.num_dacs)
- return srcs[i];
- }
- return 0;
-}
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int alc861_auto_fill_dac_nids(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- struct alc_spec *spec = codec->spec;
- int i;
- hda_nid_t nid, dac;
-
- spec->multiout.dac_nids = spec->private_dac_nids;
- for (i = 0; i < cfg->line_outs; i++) {
- nid = cfg->line_out_pins[i];
- dac = alc861_look_for_dac(codec, nid);
- if (!dac)
- continue;
- spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
- }
- return 0;
-}
-
-static int __alc861_create_out_sw(struct hda_codec *codec, const char *pfx,
- hda_nid_t nid, int idx, unsigned int chs)
-{
- return __add_pb_sw_ctrl(codec->spec, ALC_CTL_WIDGET_MUTE, pfx, idx,
- HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
-}
-
-#define alc861_create_out_sw(codec, pfx, nid, chs) \
- __alc861_create_out_sw(codec, pfx, nid, 0, chs)
-
-/* add playback controls from the parsed DAC table */
-static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- struct alc_spec *spec = codec->spec;
- static const char * const chname[4] = {
- "Front", "Surround", NULL /*CLFE*/, "Side"
- };
- const char *pfx = alc_get_line_out_pfx(spec, true);
- hda_nid_t nid;
- int i, err, noutputs;
-
- noutputs = cfg->line_outs;
- if (spec->multi_ios > 0)
- noutputs += spec->multi_ios;
-
- for (i = 0; i < noutputs; i++) {
- nid = spec->multiout.dac_nids[i];
- if (!nid)
- continue;
- if (!pfx && i == 2) {
- /* Center/LFE */
- err = alc861_create_out_sw(codec, "Center", nid, 1);
- if (err < 0)
- return err;
- err = alc861_create_out_sw(codec, "LFE", nid, 2);
- if (err < 0)
- return err;
- } else {
- const char *name = pfx;
- int index = i;
- if (!name) {
- name = chname[i];
- index = 0;
- }
- err = __alc861_create_out_sw(codec, name, nid, index, 3);
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-static int alc861_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- hda_nid_t nid;
-
- if (!pin)
- return 0;
-
- if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) {
- nid = alc861_look_for_dac(codec, pin);
- if (nid) {
- err = alc861_create_out_sw(codec, "Headphone", nid, 3);
- if (err < 0)
- return err;
- spec->multiout.hp_nid = nid;
- }
- }
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int alc861_auto_create_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x08, 0);
-}
-
-static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid,
- int pin_type, hda_nid_t dac)
-{
- hda_nid_t mix, srcs[5];
- int i, num;
-
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
- snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_UNMUTE);
- if (snd_hda_get_connections(codec, nid, &mix, 1) != 1)
- return;
- num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
- if (num < 0)
- return;
- for (i = 0; i < num; i++) {
- unsigned int mute;
- if (srcs[i] == dac || srcs[i] == 0x15)
- mute = AMP_IN_UNMUTE(i);
- else
- mute = AMP_IN_MUTE(i);
- snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- mute);
- }
-}
-
-static void alc861_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < spec->autocfg.line_outs; i++) {
- hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- if (nid)
- alc861_auto_set_output_and_unmute(codec, nid, pin_type,
- spec->multiout.dac_nids[i]);
- }
-}
-
-static void alc861_auto_init_hp_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- if (spec->autocfg.hp_outs)
- alc861_auto_set_output_and_unmute(codec,
- spec->autocfg.hp_pins[0],
- PIN_HP,
- spec->multiout.hp_nid);
- if (spec->autocfg.speaker_outs)
- alc861_auto_set_output_and_unmute(codec,
- spec->autocfg.speaker_pins[0],
- PIN_OUT,
- spec->multiout.dac_nids[0]);
-}
-
-static void alc861_auto_init_analog_input(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
- if (nid >= 0x0c && nid <= 0x11)
- alc_set_input_pin(codec, nid, cfg->inputs[i].type);
- }
-}
-
-/* parse the BIOS configuration and set up the alc_spec */
-/* return 1 if successful, 0 if the proper config is not found,
- * or a negative error code
- */
-static int alc861_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc861_ignore[] = { 0x1d, 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc861_ignore);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs)
- return 0; /* can't find valid BIOS pin config */
-
- err = alc861_auto_fill_dac_nids(codec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc_auto_add_multi_channel_mode(codec);
- if (err < 0)
- return err;
- err = alc861_auto_create_multi_out_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc861_auto_create_hp_ctls(codec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = alc861_auto_create_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- alc_auto_parse_digital(codec);
-
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- add_verb(spec, alc861_auto_init_verbs);
-
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- spec->adc_nids = alc861_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
- set_capture_mixer(codec);
-
- alc_ssid_check(codec, 0x0e, 0x0f, 0x0b, 0);
-
- return 1;
-}
-
-/* additional initialization for auto-configuration model */
-static void alc861_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc861_auto_init_multi_out(codec);
- alc861_auto_init_hp_out(codec);
- alc861_auto_init_analog_input(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list alc861_loopbacks[] = {
- { 0x15, HDA_INPUT, 0 },
- { 0x15, HDA_INPUT, 1 },
- { 0x15, HDA_INPUT, 2 },
- { 0x15, HDA_INPUT, 3 },
- { } /* end */
-};
-#endif
-
-
-/*
- * configuration and preset
- */
-static const char * const alc861_models[ALC861_MODEL_LAST] = {
- [ALC861_3ST] = "3stack",
- [ALC660_3ST] = "3stack-660",
- [ALC861_3ST_DIG] = "3stack-dig",
- [ALC861_6ST_DIG] = "6stack-dig",
- [ALC861_UNIWILL_M31] = "uniwill-m31",
- [ALC861_TOSHIBA] = "toshiba",
- [ALC861_ASUS] = "asus",
- [ALC861_ASUS_LAPTOP] = "asus-laptop",
- [ALC861_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc861_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
- SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
- SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
- SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
- SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
- /* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
- * Any other models that need this preset?
- */
- /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
- SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
- SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
- SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
- SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
- SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
- /* FIXME: the below seems conflict */
- /* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
- SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
- SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
- {}
-};
-
-static const struct alc_config_preset alc861_presets[] = {
- [ALC861_3ST] = {
- .mixers = { alc861_3ST_mixer },
- .init_verbs = { alc861_threestack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861_dac_nids),
- .dac_nids = alc861_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
- .channel_mode = alc861_threestack_modes,
- .need_dac_fix = 1,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- },
- [ALC861_3ST_DIG] = {
- .mixers = { alc861_base_mixer },
- .init_verbs = { alc861_threestack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861_dac_nids),
- .dac_nids = alc861_dac_nids,
- .dig_out_nid = ALC861_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
- .channel_mode = alc861_threestack_modes,
- .need_dac_fix = 1,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- },
- [ALC861_6ST_DIG] = {
- .mixers = { alc861_base_mixer },
- .init_verbs = { alc861_base_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861_dac_nids),
- .dac_nids = alc861_dac_nids,
- .dig_out_nid = ALC861_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
- .channel_mode = alc861_8ch_modes,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- },
- [ALC660_3ST] = {
- .mixers = { alc861_3ST_mixer },
- .init_verbs = { alc861_threestack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc660_dac_nids),
- .dac_nids = alc660_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
- .channel_mode = alc861_threestack_modes,
- .need_dac_fix = 1,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- },
- [ALC861_UNIWILL_M31] = {
- .mixers = { alc861_uniwill_m31_mixer },
- .init_verbs = { alc861_uniwill_m31_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861_dac_nids),
- .dac_nids = alc861_dac_nids,
- .dig_out_nid = ALC861_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
- .channel_mode = alc861_uniwill_m31_modes,
- .need_dac_fix = 1,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- },
- [ALC861_TOSHIBA] = {
- .mixers = { alc861_toshiba_mixer },
- .init_verbs = { alc861_base_init_verbs,
- alc861_toshiba_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861_dac_nids),
- .dac_nids = alc861_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- .unsol_event = alc861_toshiba_unsol_event,
- .init_hook = alc861_toshiba_automute,
- },
- [ALC861_ASUS] = {
- .mixers = { alc861_asus_mixer },
- .init_verbs = { alc861_asus_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861_dac_nids),
- .dac_nids = alc861_dac_nids,
- .dig_out_nid = ALC861_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
- .channel_mode = alc861_asus_modes,
- .need_dac_fix = 1,
- .hp_nid = 0x06,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- },
- [ALC861_ASUS_LAPTOP] = {
- .mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
- .init_verbs = { alc861_asus_init_verbs,
- alc861_asus_laptop_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861_dac_nids),
- .dac_nids = alc861_dac_nids,
- .dig_out_nid = ALC861_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
- .channel_mode = alc883_3ST_2ch_modes,
- .need_dac_fix = 1,
- .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
- .adc_nids = alc861_adc_nids,
- .input_mux = &alc861_capture_source,
- },
-};
-
-/* Pin config fixes */
-enum {
- PINFIX_FSC_AMILO_PI1505,
-};
-
-static const struct alc_fixup alc861_fixups[] = {
- [PINFIX_FSC_AMILO_PI1505] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x0b, 0x0221101f }, /* HP */
- { 0x0f, 0x90170310 }, /* speaker */
- { }
- }
- },
-};
-
-static const struct snd_pci_quirk alc861_fixup_tbl[] = {
- SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
- {}
-};
-
-static int patch_alc861(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int board_config;
- int err;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- board_config = snd_hda_check_board_config(codec, ALC861_MODEL_LAST,
- alc861_models,
- alc861_cfg_tbl);
-
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC861_AUTO;
- }
-
- if (board_config == ALC861_AUTO) {
- alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC861_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc861_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC861_3ST_DIG;
- }
- }
-
- err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
-
- if (board_config != ALC861_AUTO)
- setup_preset(codec, &alc861_presets[board_config]);
-
- spec->stream_analog_playback = &alc861_pcm_analog_playback;
- spec->stream_analog_capture = &alc861_pcm_analog_capture;
-
- spec->stream_digital_playback = &alc861_pcm_digital_playback;
- spec->stream_digital_capture = &alc861_pcm_digital_capture;
-
- if (!spec->cap_mixer)
- set_capture_mixer(codec);
- set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
-
- spec->vmaster_nid = 0x03;
-
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
- codec->patch_ops = alc_patch_ops;
- if (board_config == ALC861_AUTO) {
- spec->init_hook = alc861_auto_init;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->power_hook = alc_power_eapd;
-#endif
- }
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc861_loopbacks;
-#endif
-
- return 0;
-}
-
-/*
- * ALC861-VD support
- *
- * Based on ALC882
- *
- * In addition, an independent DAC
- */
-#define ALC861VD_DIGOUT_NID 0x06
-
-static const hda_nid_t alc861vd_dac_nids[4] = {
- /* front, surr, clfe, side surr */
- 0x02, 0x03, 0x04, 0x05
-};
-
-/* dac_nids for ALC660vd are in a different order - according to
- * Realtek's driver.
- * This should probably result in a different mixer for 6stack models
- * of ALC660vd codecs, but for now there is only 3stack mixer
- * - and it is the same as in 861vd.
- * adc_nids in ALC660vd are (is) the same as in 861vd
- */
-static const hda_nid_t alc660vd_dac_nids[3] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x04, 0x03
-};
-
-static const hda_nid_t alc861vd_adc_nids[1] = {
- /* ADC0 */
- 0x09,
-};
-
-static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-static const struct hda_input_mux alc861vd_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc861vd_dallas_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x1 },
- },
-};
-
-static const struct hda_input_mux alc861vd_hp_capture_source = {
- .num_items = 2,
- .items = {
- { "Front Mic", 0x0 },
- { "ATAPI Mic", 0x1 },
- },
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
- { 2, NULL }
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc861vd_6stack_ch6_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static const struct hda_verb alc861vd_6stack_ch8_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc861vd_6stack_modes[2] = {
- { 6, alc861vd_6stack_ch6_init },
- { 8, alc861vd_6stack_ch8_init },
-};
-
-static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
- * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
- */
-static const struct snd_kcontrol_new alc861vd_6st_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
- HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
-
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc861vd_3st_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
-
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- /*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-
- { } /* end */
-};
-
-/* Pin assignment: Speaker=0x14, HP = 0x15,
- * Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
- */
-static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-/* Pin assignment: Speaker=0x14, Line-out = 0x15,
- * Front Mic=0x18, ATAPI Mic = 0x19,
- */
-static const struct snd_kcontrol_new alc861vd_hp_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
- { } /* end */
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc861vd_volume_init_verbs[] = {
- /*
- * Unmute ADC0 and set the default input to mic-in
- */
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of
- * the analog-loopback mixer widget
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- /*
- * Set up output mixers (0x02 - 0x05)
- */
- /* set vol=0 to output mixers */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- { }
-};
-
-/*
- * 3-stack pin configuration:
- * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
- */
-static const struct hda_verb alc861vd_3stack_init_verbs[] = {
- /*
- * Set pin mode and muting
- */
- /* set front pin widgets 0x14 for output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line-2 In: Headphone output (output 0 - 0x0c) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * 6-stack pin configuration:
- */
-static const struct hda_verb alc861vd_6stack_init_verbs[] = {
- /*
- * Set pin mode and muting
- */
- /* set front pin widgets 0x14 for output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Rear Pin: output 1 (0x0d) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* CLFE Pin: output 2 (0x0e) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* Side Pin: output 3 (0x0f) */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
-
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line-2 In: Headphone output (output 0 - 0x0c) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-static const struct hda_verb alc861vd_eapd_verbs[] = {
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-static const struct hda_verb alc660vd_eapd_verbs[] = {
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {}
-};
-
-static void alc861vd_lenovo_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
-{
- alc_hp_automute(codec);
- alc88x_simple_mic_automute(codec);
-}
-
-static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case ALC880_MIC_EVENT:
- alc88x_simple_mic_automute(codec);
- break;
- default:
- alc_sku_unsol_event(codec, res);
- break;
- }
-}
-
-static const struct hda_verb alc861vd_dallas_verbs[] = {
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
-
- { } /* end */
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc861vd_dallas_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc861vd_loopbacks alc880_loopbacks
-#endif
-
-/* pcm configuration: identical with ALC880 */
-#define alc861vd_pcm_analog_playback alc880_pcm_analog_playback
-#define alc861vd_pcm_analog_capture alc880_pcm_analog_capture
-#define alc861vd_pcm_digital_playback alc880_pcm_digital_playback
-#define alc861vd_pcm_digital_capture alc880_pcm_digital_capture
-
-/*
- * configuration and preset
- */
-static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
- [ALC660VD_3ST] = "3stack-660",
- [ALC660VD_3ST_DIG] = "3stack-660-digout",
- [ALC660VD_ASUS_V1S] = "asus-v1s",
- [ALC861VD_3ST] = "3stack",
- [ALC861VD_3ST_DIG] = "3stack-digout",
- [ALC861VD_6ST_DIG] = "6stack-digout",
- [ALC861VD_LENOVO] = "lenovo",
- [ALC861VD_DALLAS] = "dallas",
- [ALC861VD_HP] = "hp",
- [ALC861VD_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc861vd_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
- SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
- SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
- /*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
- SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
- SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
- SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
- SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
- /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
- SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
- SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
- SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
- SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
- SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
- SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
- {}
-};
-
-static const struct alc_config_preset alc861vd_presets[] = {
- [ALC660VD_3ST] = {
- .mixers = { alc861vd_3st_mixer },
- .init_verbs = { alc861vd_volume_init_verbs,
- alc861vd_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
- .dac_nids = alc660vd_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_capture_source,
- },
- [ALC660VD_3ST_DIG] = {
- .mixers = { alc861vd_3st_mixer },
- .init_verbs = { alc861vd_volume_init_verbs,
- alc861vd_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
- .dac_nids = alc660vd_dac_nids,
- .dig_out_nid = ALC861VD_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_capture_source,
- },
- [ALC861VD_3ST] = {
- .mixers = { alc861vd_3st_mixer },
- .init_verbs = { alc861vd_volume_init_verbs,
- alc861vd_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
- .dac_nids = alc861vd_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_capture_source,
- },
- [ALC861VD_3ST_DIG] = {
- .mixers = { alc861vd_3st_mixer },
- .init_verbs = { alc861vd_volume_init_verbs,
- alc861vd_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
- .dac_nids = alc861vd_dac_nids,
- .dig_out_nid = ALC861VD_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_capture_source,
- },
- [ALC861VD_6ST_DIG] = {
- .mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
- .init_verbs = { alc861vd_volume_init_verbs,
- alc861vd_6stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
- .dac_nids = alc861vd_dac_nids,
- .dig_out_nid = ALC861VD_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
- .channel_mode = alc861vd_6stack_modes,
- .input_mux = &alc861vd_capture_source,
- },
- [ALC861VD_LENOVO] = {
- .mixers = { alc861vd_lenovo_mixer },
- .init_verbs = { alc861vd_volume_init_verbs,
- alc861vd_3stack_init_verbs,
- alc861vd_eapd_verbs,
- alc861vd_lenovo_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
- .dac_nids = alc660vd_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_capture_source,
- .unsol_event = alc861vd_lenovo_unsol_event,
- .setup = alc861vd_lenovo_setup,
- .init_hook = alc861vd_lenovo_init_hook,
- },
- [ALC861VD_DALLAS] = {
- .mixers = { alc861vd_dallas_mixer },
- .init_verbs = { alc861vd_dallas_verbs },
- .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
- .dac_nids = alc861vd_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_dallas_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc861vd_dallas_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC861VD_HP] = {
- .mixers = { alc861vd_hp_mixer },
- .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
- .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
- .dac_nids = alc861vd_dac_nids,
- .dig_out_nid = ALC861VD_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_hp_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc861vd_dallas_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC660VD_ASUS_V1S] = {
- .mixers = { alc861vd_lenovo_mixer },
- .init_verbs = { alc861vd_volume_init_verbs,
- alc861vd_3stack_init_verbs,
- alc861vd_eapd_verbs,
- alc861vd_lenovo_unsol_verbs },
- .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
- .dac_nids = alc660vd_dac_nids,
- .dig_out_nid = ALC861VD_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
- .channel_mode = alc861vd_3stack_2ch_modes,
- .input_mux = &alc861vd_capture_source,
- .unsol_event = alc861vd_lenovo_unsol_event,
- .setup = alc861vd_lenovo_setup,
- .init_hook = alc861vd_lenovo_init_hook,
- },
-};
-
-/*
- * BIOS auto configuration
- */
-static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x22, 0);
-}
-
-
-static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type, int dac_idx)
-{
- alc_set_pin_output(codec, nid, pin_type);
-}
-
-static void alc861vd_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i <= HDA_SIDE; i++) {
- hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- if (nid)
- alc861vd_auto_set_output_and_unmute(codec, nid,
- pin_type, i);
- }
-}
-
-
-static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
-
- pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front and use dac 0 */
- alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
- pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
-}
-
-#define ALC861VD_PIN_CD_NID ALC880_PIN_CD_NID
-
-static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
- if (alc_is_input_pin(codec, nid)) {
- alc_set_input_pin(codec, nid, cfg->inputs[i].type);
- if (nid != ALC861VD_PIN_CD_NID &&
- (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
- }
- }
-}
-
-#define alc861vd_auto_init_input_src alc882_auto_init_input_src
-
-#define alc861vd_idx_to_mixer_vol(nid) ((nid) + 0x02)
-#define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c)
-
-/* add playback controls from the parsed DAC table */
-/* Based on ALC880 version. But ALC861VD has separate,
- * different NIDs for mute/unmute switch and volume control */
-static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- static const char * const chname[4] = {
- "Front", "Surround", "CLFE", "Side"
- };
- const char *pfx = alc_get_line_out_pfx(spec, true);
- hda_nid_t nid_v, nid_s;
- int i, err, noutputs;
-
- noutputs = cfg->line_outs;
- if (spec->multi_ios > 0)
- noutputs += spec->multi_ios;
-
- for (i = 0; i < noutputs; i++) {
- if (!spec->multiout.dac_nids[i])
- continue;
- nid_v = alc861vd_idx_to_mixer_vol(
- alc880_dac_to_idx(
- spec->multiout.dac_nids[i]));
- nid_s = alc861vd_idx_to_mixer_switch(
- alc880_dac_to_idx(
- spec->multiout.dac_nids[i]));
-
- if (!pfx && i == 2) {
- /* Center/LFE */
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
- "Center",
- HDA_COMPOSE_AMP_VAL(nid_v, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
- "LFE",
- HDA_COMPOSE_AMP_VAL(nid_v, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
- "Center",
- HDA_COMPOSE_AMP_VAL(nid_s, 1, 2,
- HDA_INPUT));
- if (err < 0)
- return err;
- err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
- "LFE",
- HDA_COMPOSE_AMP_VAL(nid_s, 2, 2,
- HDA_INPUT));
- if (err < 0)
- return err;
- } else {
- const char *name = pfx;
- int index = i;
- if (!name) {
- name = chname[i];
- index = 0;
- }
- err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
- name, index,
- HDA_COMPOSE_AMP_VAL(nid_v, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
- name, index,
- HDA_COMPOSE_AMP_VAL(nid_s, 3, 2,
- HDA_INPUT));
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-/* add playback controls for speaker and HP outputs */
-/* Based on ALC880 version. But ALC861VD has separate,
- * different NIDs for mute/unmute switch and volume control */
-static int alc861vd_auto_create_extra_out(struct alc_spec *spec,
- hda_nid_t pin, const char *pfx)
-{
- hda_nid_t nid_v, nid_s;
- int err;
-
- if (!pin)
- return 0;
-
- if (alc880_is_fixed_pin(pin)) {
- nid_v = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
- /* specify the DAC as the extra output */
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = nid_v;
- else
- spec->multiout.extra_out_nid[0] = nid_v;
- /* control HP volume/switch on the output mixer amp */
- nid_v = alc861vd_idx_to_mixer_vol(
- alc880_fixed_pin_idx(pin));
- nid_s = alc861vd_idx_to_mixer_switch(
- alc880_fixed_pin_idx(pin));
-
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx,
- HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx,
- HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT));
- if (err < 0)
- return err;
- } else if (alc880_is_multi_pin(pin)) {
- /* set manual connection */
- /* we have only a switch on HP-out PIN */
- err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
-/* parse the BIOS configuration and set up the alc_spec
- * return 1 if successful, 0 if the proper config is not found,
- * or a negative error code
- * Based on ALC880 version - had to change it to override
- * alc880_auto_create_extra_out and alc880_auto_create_multi_out_ctls */
-static int alc861vd_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc861vd_ignore[] = { 0x1d, 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc861vd_ignore);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs)
- return 0; /* can't find valid BIOS pin config */
-
- err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc_auto_add_multi_channel_mode(codec);
- if (err < 0)
- return err;
- err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = alc861vd_auto_create_extra_out(spec,
- spec->autocfg.speaker_pins[0],
- "Speaker");
- if (err < 0)
- return err;
- err = alc861vd_auto_create_extra_out(spec,
- spec->autocfg.hp_pins[0],
- "Headphone");
- if (err < 0)
- return err;
- err = alc861vd_auto_create_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- alc_auto_parse_digital(codec);
-
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- add_verb(spec, alc861vd_volume_init_verbs);
-
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
-
- return 1;
-}
-
-/* additional initialization for auto-configuration model */
-static void alc861vd_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc861vd_auto_init_multi_out(codec);
- alc861vd_auto_init_hp_out(codec);
- alc861vd_auto_init_analog_input(codec);
- alc861vd_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-enum {
- ALC660VD_FIX_ASUS_GPIO1
-};
-
-/* reset GPIO1 */
-static const struct alc_fixup alc861vd_fixups[] = {
- [ALC660VD_FIX_ASUS_GPIO1] = {
- .type = ALC_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
- { }
- }
- },
-};
-
-static const struct snd_pci_quirk alc861vd_fixup_tbl[] = {
- SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
- {}
-};
-
-static int patch_alc861vd(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int err, board_config;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- board_config = snd_hda_check_board_config(codec, ALC861VD_MODEL_LAST,
- alc861vd_models,
- alc861vd_cfg_tbl);
-
- if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC861VD_AUTO;
- }
-
- if (board_config == ALC861VD_AUTO) {
- alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC861VD_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc861vd_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC861VD_3ST;
- }
- }
-
- err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
-
- if (board_config != ALC861VD_AUTO)
- setup_preset(codec, &alc861vd_presets[board_config]);
-
- if (codec->vendor_id == 0x10ec0660) {
- /* always turn on EAPD */
- add_verb(spec, alc660vd_eapd_verbs);
- }
-
- spec->stream_analog_playback = &alc861vd_pcm_analog_playback;
- spec->stream_analog_capture = &alc861vd_pcm_analog_capture;
-
- spec->stream_digital_playback = &alc861vd_pcm_digital_playback;
- spec->stream_digital_capture = &alc861vd_pcm_digital_capture;
-
- if (!spec->adc_nids) {
- spec->adc_nids = alc861vd_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
+ AMP_IN_MUTE(i));
}
- if (!spec->capsrc_nids)
- spec->capsrc_nids = alc861vd_capsrc_nids;
-
- set_capture_mixer(codec);
- set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
-
- spec->vmaster_nid = 0x02;
-
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
- codec->patch_ops = alc_patch_ops;
-
- if (board_config == ALC861VD_AUTO)
- spec->init_hook = alc861vd_auto_init;
- spec->shutup = alc_eapd_shutup;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc861vd_loopbacks;
-#endif
-
- return 0;
-}
-
-/*
- * ALC662 support
- *
- * ALC662 is almost identical with ALC880 but has cleaner and more flexible
- * configuration. Each pin widget can choose any input DACs and a mixer.
- * Each ADC is connected from a mixer of all inputs. This makes possible
- * 6-channel independent captures.
- *
- * In addition, an independent DAC for the multi-playback (not used in this
- * driver yet).
- */
-#define ALC662_DIGOUT_NID 0x06
-#define ALC662_DIGIN_NID 0x0a
-
-static const hda_nid_t alc662_dac_nids[3] = {
- /* front, rear, clfe */
- 0x02, 0x03, 0x04
-};
-
-static const hda_nid_t alc272_dac_nids[2] = {
- 0x02, 0x03
-};
-
-static const hda_nid_t alc662_adc_nids[2] = {
- /* ADC1-2 */
- 0x09, 0x08
-};
-
-static const hda_nid_t alc272_adc_nids[1] = {
- /* ADC1-2 */
- 0x08,
-};
-
-static const hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
-static const hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
-
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-static const struct hda_input_mux alc662_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc662_lenovo_101e_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 },
- { "Line", 0x2 },
- },
-};
-
-static const struct hda_input_mux alc663_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- },
-};
-
-#if 0 /* set to 1 for testing other input sources below */
-static const struct hda_input_mux alc272_nc10_capture_source = {
- .num_items = 16,
- .items = {
- { "Autoselect Mic", 0x0 },
- { "Internal Mic", 0x1 },
- { "In-0x02", 0x2 },
- { "In-0x03", 0x3 },
- { "In-0x04", 0x4 },
- { "In-0x05", 0x5 },
- { "In-0x06", 0x6 },
- { "In-0x07", 0x7 },
- { "In-0x08", 0x8 },
- { "In-0x09", 0x9 },
- { "In-0x0a", 0x0a },
- { "In-0x0b", 0x0b },
- { "In-0x0c", 0x0c },
- { "In-0x0d", 0x0d },
- { "In-0x0e", 0x0e },
- { "In-0x0f", 0x0f },
- },
-};
-#endif
-
-/*
- * 2ch mode
- */
-static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
- { 2, NULL }
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc662_3ST_ch2_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc662_3ST_ch6_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
- { 2, alc662_3ST_ch2_init },
- { 6, alc662_3ST_ch6_init },
-};
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc662_sixstack_ch6_init[] = {
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static const struct hda_verb alc662_sixstack_ch8_init[] = {
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc662_5stack_modes[2] = {
- { 2, alc662_sixstack_ch6_init },
- { 6, alc662_sixstack_ch8_init },
-};
-
-/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
- * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
- */
-
-static const struct snd_kcontrol_new alc662_base_mixer[] = {
- /* output mixer control */
- HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-
- /*Input mixer control */
- HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- ALC262_HIPPO_MASTER_SWITCH,
-
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
- ALC262_HIPPO_MASTER_SWITCH,
- HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls alc663_asus_one_bind_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc663_m51va_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
- HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_tree_bind_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
- HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-
- { } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_four_bind_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
- HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc662_1bjd_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls alc663_asus_two_bind_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume",
- &alc663_asus_two_bind_master_vol),
- HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
- HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
- HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_g71v_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_g50v_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
- HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc663_mode7_mixer[] = {
- HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
- HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
- HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
- HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc663_mode8_mixer[] = {
- HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
- HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
- HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
- HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-
-static const struct snd_kcontrol_new alc662_chmode_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc662_init_verbs[] = {
- /* ADC: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* Front Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Rear Pin: output 1 (0x0d) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* CLFE Pin: output 2 (0x0e) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line-2 In: Headphone output (output 0 - 0x0c) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- { }
-};
-
-static const struct hda_verb alc662_eapd_init_verbs[] = {
- /* always trun on EAPD */
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
- { }
-};
-
-static const struct hda_verb alc662_sue_init_verbs[] = {
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc662_eeepc_sue_init_verbs[] = {
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-/* Set Unsolicited Event*/
-static const struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_m51va_init_verbs[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_21jd_amic_init_verbs[] = {
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc662_1bjd_amic_init_verbs[] = {
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_15jd_amic_init_verbs[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_g71v_init_verbs[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
- /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */
-
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
-
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_g50v_init_verbs[] = {
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
-
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc662_ecs_init_verbs[] = {
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc272_dell_zm1_init_verbs[] = {
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc272_dell_init_verbs[] = {
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_mode7_init_verbs[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct hda_verb alc663_mode8_init_verbs[] = {
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {}
-};
-
-static const struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc272_auto_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-static void alc662_lenovo_101e_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.line_out_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->automute = 1;
- spec->detect_line = 1;
- spec->automute_lines = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc662_eeepc_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- alc262_hippo1_setup(codec);
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x1b;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc663_m51va_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 9;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode1 ******************************/
-static void alc663_mode1_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode2 ******************************/
-static void alc662_mode2_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode3 ******************************/
-static void alc663_mode3_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode4 ******************************/
-static void alc663_mode4_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute_mixer_nid[1] = 0x0e;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode5 ******************************/
-static void alc663_mode5_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x16;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute_mixer_nid[1] = 0x0e;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode6 ******************************/
-static void alc663_mode6_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute_mixer_nid[0] = 0x0c;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_MIXER;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode7 ******************************/
-static void alc663_mode7_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x19;
- spec->int_mic.mux_idx = 1;
- spec->auto_mic = 1;
-}
-
-/* ***************** Mode8 ******************************/
-static void alc663_mode8_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.hp_pins[1] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x17;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_PIN;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 9;
- spec->auto_mic = 1;
-}
-
-static void alc663_g71v_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x21;
- spec->autocfg.line_out_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
- spec->detect_line = 1;
- spec->automute_lines = 1;
- spec->ext_mic.pin = 0x18;
- spec->ext_mic.mux_idx = 0;
- spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 9;
- spec->auto_mic = 1;
}
-#define alc663_g50v_setup alc663_m51va_setup
-
-static const struct snd_kcontrol_new alc662_ecs_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- ALC262_HIPPO_MASTER_SWITCH,
-
- HDA_CODEC_VOLUME("Mic/LineIn Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc272_nc10_mixer[] = {
- /* Master Playback automatically created from Speaker and Headphone */
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
- { } /* end */
-};
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc662_loopbacks alc880_loopbacks
-#endif
-
-
-/* pcm configuration: identical with ALC880 */
-#define alc662_pcm_analog_playback alc880_pcm_analog_playback
-#define alc662_pcm_analog_capture alc880_pcm_analog_capture
-#define alc662_pcm_digital_playback alc880_pcm_digital_playback
-#define alc662_pcm_digital_capture alc880_pcm_digital_capture
-
-/*
- * configuration and preset
- */
-static const char * const alc662_models[ALC662_MODEL_LAST] = {
- [ALC662_3ST_2ch_DIG] = "3stack-dig",
- [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig",
- [ALC662_3ST_6ch] = "3stack-6ch",
- [ALC662_5ST_DIG] = "5stack-dig",
- [ALC662_LENOVO_101E] = "lenovo-101e",
- [ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
- [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
- [ALC662_ECS] = "ecs",
- [ALC663_ASUS_M51VA] = "m51va",
- [ALC663_ASUS_G71V] = "g71v",
- [ALC663_ASUS_H13] = "h13",
- [ALC663_ASUS_G50V] = "g50v",
- [ALC663_ASUS_MODE1] = "asus-mode1",
- [ALC662_ASUS_MODE2] = "asus-mode2",
- [ALC663_ASUS_MODE3] = "asus-mode3",
- [ALC663_ASUS_MODE4] = "asus-mode4",
- [ALC663_ASUS_MODE5] = "asus-mode5",
- [ALC663_ASUS_MODE6] = "asus-mode6",
- [ALC663_ASUS_MODE7] = "asus-mode7",
- [ALC663_ASUS_MODE8] = "asus-mode8",
- [ALC272_DELL] = "dell",
- [ALC272_DELL_ZM1] = "dell-zm1",
- [ALC272_SAMSUNG_NC10] = "samsung-nc10",
- [ALC662_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc662_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
- SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL),
- SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1),
- SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
- SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
- SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
- SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
- SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
- SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
- SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
- SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
- SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
- /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
- SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
- /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
- SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x19d3, "ASUS NB", ALC663_ASUS_M51VA),
- SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
- SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
- SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
- SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
- SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
- ALC662_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
- SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
- SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
- ALC662_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
- SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
- SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
- SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
- ALC662_3ST_6ch_DIG),
- SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
- ALC663_ASUS_H13),
- SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
- {}
-};
-
-static const struct alc_config_preset alc662_presets[] = {
- [ALC662_3ST_2ch_DIG] = {
- .mixers = { alc662_3ST_2ch_mixer },
- .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .dig_in_nid = ALC662_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_capture_source,
- },
- [ALC662_3ST_6ch_DIG] = {
- .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
- .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .dig_in_nid = ALC662_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
- .channel_mode = alc662_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc662_capture_source,
- },
- [ALC662_3ST_6ch] = {
- .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
- .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
- .channel_mode = alc662_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc662_capture_source,
- },
- [ALC662_5ST_DIG] = {
- .mixers = { alc662_base_mixer, alc662_chmode_mixer },
- .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .dig_in_nid = ALC662_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
- .channel_mode = alc662_5stack_modes,
- .input_mux = &alc662_capture_source,
- },
- [ALC662_LENOVO_101E] = {
- .mixers = { alc662_lenovo_101e_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc662_sue_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_lenovo_101e_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc662_lenovo_101e_setup,
- .init_hook = alc_inithook,
- },
- [ALC662_ASUS_EEEPC_P701] = {
- .mixers = { alc662_eeepc_p701_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc662_eeepc_sue_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc662_eeepc_setup,
- .init_hook = alc_inithook,
- },
- [ALC662_ASUS_EEEPC_EP20] = {
- .mixers = { alc662_eeepc_ep20_mixer,
- alc662_chmode_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc662_eeepc_ep20_sue_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
- .channel_mode = alc662_3ST_6ch_modes,
- .input_mux = &alc662_lenovo_101e_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc662_eeepc_ep20_setup,
- .init_hook = alc_inithook,
- },
- [ALC662_ECS] = {
- .mixers = { alc662_ecs_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc662_ecs_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc662_eeepc_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_M51VA] = {
- .mixers = { alc663_m51va_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_m51va_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_m51va_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_G71V] = {
- .mixers = { alc663_g71v_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_g71v_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_g71v_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_H13] = {
- .mixers = { alc663_m51va_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_m51va_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .setup = alc663_m51va_setup,
- .unsol_event = alc_sku_unsol_event,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_G50V] = {
- .mixers = { alc663_g50v_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_g50v_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
- .channel_mode = alc662_3ST_6ch_modes,
- .input_mux = &alc663_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_g50v_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_MODE1] = {
- .mixers = { alc663_m51va_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_21jd_amic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .hp_nid = 0x03,
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode1_setup,
- .init_hook = alc_inithook,
- },
- [ALC662_ASUS_MODE2] = {
- .mixers = { alc662_1bjd_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc662_1bjd_amic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc662_mode2_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_MODE3] = {
- .mixers = { alc663_two_hp_m1_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_two_hp_amic_m1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .hp_nid = 0x03,
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode3_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_MODE4] = {
- .mixers = { alc663_asus_21jd_clfe_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_21jd_amic_init_verbs},
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .hp_nid = 0x03,
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode4_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_MODE5] = {
- .mixers = { alc663_asus_15jd_clfe_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_15jd_amic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .hp_nid = 0x03,
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode5_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_MODE6] = {
- .mixers = { alc663_two_hp_m2_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_two_hp_amic_m2_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .hp_nid = 0x03,
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode6_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_MODE7] = {
- .mixers = { alc663_mode7_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_mode7_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .hp_nid = 0x03,
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode7_setup,
- .init_hook = alc_inithook,
- },
- [ALC663_ASUS_MODE8] = {
- .mixers = { alc663_mode8_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_mode8_init_verbs },
- .num_dacs = ARRAY_SIZE(alc662_dac_nids),
- .hp_nid = 0x03,
- .dac_nids = alc662_dac_nids,
- .dig_out_nid = ALC662_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode8_setup,
- .init_hook = alc_inithook,
- },
- [ALC272_DELL] = {
- .mixers = { alc663_m51va_mixer },
- .cap_mixer = alc272_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc272_dell_init_verbs },
- .num_dacs = ARRAY_SIZE(alc272_dac_nids),
- .dac_nids = alc272_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .adc_nids = alc272_adc_nids,
- .num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
- .capsrc_nids = alc272_capsrc_nids,
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_m51va_setup,
- .init_hook = alc_inithook,
- },
- [ALC272_DELL_ZM1] = {
- .mixers = { alc663_m51va_mixer },
- .cap_mixer = alc662_auto_capture_mixer,
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc272_dell_zm1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc272_dac_nids),
- .dac_nids = alc272_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .adc_nids = alc662_adc_nids,
- .num_adc_nids = 1,
- .capsrc_nids = alc662_capsrc_nids,
- .channel_mode = alc662_3ST_2ch_modes,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_m51va_setup,
- .init_hook = alc_inithook,
- },
- [ALC272_SAMSUNG_NC10] = {
- .mixers = { alc272_nc10_mixer },
- .init_verbs = { alc662_init_verbs,
- alc662_eapd_init_verbs,
- alc663_21jd_amic_init_verbs },
- .num_dacs = ARRAY_SIZE(alc272_dac_nids),
- .dac_nids = alc272_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
- .channel_mode = alc662_3ST_2ch_modes,
- /*.input_mux = &alc272_nc10_capture_source,*/
- .unsol_event = alc_sku_unsol_event,
- .setup = alc663_mode4_setup,
- .init_hook = alc_inithook,
- },
-};
-
-
-/*
- * BIOS auto configuration
- */
-
/* convert from MIX nid to DAC */
static hda_nid_t alc_auto_mix_to_dac(struct hda_codec *codec, hda_nid_t nid)
{
hda_nid_t list[5];
int i, num;
+ if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_AUD_OUT)
+ return nid;
num = snd_hda_get_connections(codec, nid, list, ARRAY_SIZE(list));
for (i = 0; i < num; i++) {
if (get_wcaps_type(get_wcaps(codec, list[i])) == AC_WID_AUD_OUT)
@@ -18978,7 +2763,7 @@
{
struct alc_spec *spec = codec->spec;
hda_nid_t srcs[5];
- int i, j, num;
+ int i, num;
pin = alc_go_down_to_selector(codec, pin);
num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
@@ -18986,66 +2771,161 @@
hda_nid_t nid = alc_auto_mix_to_dac(codec, srcs[i]);
if (!nid)
continue;
- for (j = 0; j < spec->multiout.num_dacs; j++)
- if (spec->multiout.dac_nids[j] == nid)
- break;
- if (j >= spec->multiout.num_dacs)
- return nid;
+ if (found_in_nid_list(nid, spec->multiout.dac_nids,
+ spec->multiout.num_dacs))
+ continue;
+ if (spec->multiout.hp_nid == nid)
+ continue;
+ if (found_in_nid_list(nid, spec->multiout.extra_out_nid,
+ ARRAY_SIZE(spec->multiout.extra_out_nid)))
+ continue;
+ return nid;
}
return 0;
}
+static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin)
+{
+ hda_nid_t sel = alc_go_down_to_selector(codec, pin);
+ if (snd_hda_get_conn_list(codec, sel, NULL) == 1)
+ return alc_auto_look_for_dac(codec, pin);
+ return 0;
+}
+
/* fill in the dac_nids table from the parsed pin configuration */
-static int alc662_auto_fill_dac_nids(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
+static int alc_auto_fill_dac_nids(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ bool redone = false;
int i;
- hda_nid_t dac;
+ again:
+ spec->multiout.num_dacs = 0;
+ spec->multiout.hp_nid = 0;
+ spec->multiout.extra_out_nid[0] = 0;
+ memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
spec->multiout.dac_nids = spec->private_dac_nids;
- for (i = 0; i < cfg->line_outs; i++) {
- dac = alc_auto_look_for_dac(codec, cfg->line_out_pins[i]);
- if (!dac)
- continue;
- spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
+
+ /* fill hard-wired DACs first */
+ if (!redone) {
+ for (i = 0; i < cfg->line_outs; i++)
+ spec->private_dac_nids[i] =
+ get_dac_if_single(codec, cfg->line_out_pins[i]);
+ if (cfg->hp_outs)
+ spec->multiout.hp_nid =
+ get_dac_if_single(codec, cfg->hp_pins[0]);
+ if (cfg->speaker_outs)
+ spec->multiout.extra_out_nid[0] =
+ get_dac_if_single(codec, cfg->speaker_pins[0]);
}
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t pin = cfg->line_out_pins[i];
+ if (spec->private_dac_nids[i])
+ continue;
+ spec->private_dac_nids[i] = alc_auto_look_for_dac(codec, pin);
+ if (!spec->private_dac_nids[i] && !redone) {
+ /* if we can't find primary DACs, re-probe without
+ * checking the hard-wired DACs
+ */
+ redone = true;
+ goto again;
+ }
+ }
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (spec->private_dac_nids[i])
+ spec->multiout.num_dacs++;
+ else
+ memmove(spec->private_dac_nids + i,
+ spec->private_dac_nids + i + 1,
+ sizeof(hda_nid_t) * (cfg->line_outs - i - 1));
+ }
+
+ if (cfg->hp_outs && !spec->multiout.hp_nid)
+ spec->multiout.hp_nid =
+ alc_auto_look_for_dac(codec, cfg->hp_pins[0]);
+ if (cfg->speaker_outs && !spec->multiout.extra_out_nid[0])
+ spec->multiout.extra_out_nid[0] =
+ alc_auto_look_for_dac(codec, cfg->speaker_pins[0]);
+
return 0;
}
-static inline int __alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx,
- hda_nid_t nid, int idx, unsigned int chs)
+static int alc_auto_add_vol_ctl(struct hda_codec *codec,
+ const char *pfx, int cidx,
+ hda_nid_t nid, unsigned int chs)
{
- return __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, idx,
- HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+ if (!nid)
+ return 0;
+ return __add_pb_vol_ctrl(codec->spec, ALC_CTL_WIDGET_VOL, pfx, cidx,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
}
-static inline int __alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx,
- hda_nid_t nid, int idx, unsigned int chs)
+#define alc_auto_add_stereo_vol(codec, pfx, cidx, nid) \
+ alc_auto_add_vol_ctl(codec, pfx, cidx, nid, 3)
+
+/* create a mute-switch for the given mixer widget;
+ * if it has multiple sources (e.g. DAC and loopback), create a bind-mute
+ */
+static int alc_auto_add_sw_ctl(struct hda_codec *codec,
+ const char *pfx, int cidx,
+ hda_nid_t nid, unsigned int chs)
{
- return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, idx,
- HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT));
+ int wid_type;
+ int type;
+ unsigned long val;
+ if (!nid)
+ return 0;
+ wid_type = get_wcaps_type(get_wcaps(codec, nid));
+ if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) {
+ type = ALC_CTL_WIDGET_MUTE;
+ val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT);
+ } else if (snd_hda_get_conn_list(codec, nid, NULL) == 1) {
+ type = ALC_CTL_WIDGET_MUTE;
+ val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT);
+ } else {
+ type = ALC_CTL_BIND_MUTE;
+ val = HDA_COMPOSE_AMP_VAL(nid, chs, 2, HDA_INPUT);
+ }
+ return __add_pb_sw_ctrl(codec->spec, type, pfx, cidx, val);
}
-#define alc662_add_vol_ctl(spec, pfx, nid, chs) \
- __alc662_add_vol_ctl(spec, pfx, nid, 0, chs)
-#define alc662_add_sw_ctl(spec, pfx, nid, chs) \
- __alc662_add_sw_ctl(spec, pfx, nid, 0, chs)
-#define alc662_add_stereo_vol(spec, pfx, nid) \
- alc662_add_vol_ctl(spec, pfx, nid, 3)
-#define alc662_add_stereo_sw(spec, pfx, nid) \
- alc662_add_sw_ctl(spec, pfx, nid, 3)
+#define alc_auto_add_stereo_sw(codec, pfx, cidx, nid) \
+ alc_auto_add_sw_ctl(codec, pfx, cidx, nid, 3)
+
+static hda_nid_t alc_look_for_out_mute_nid(struct hda_codec *codec,
+ hda_nid_t pin, hda_nid_t dac)
+{
+ hda_nid_t mix = alc_auto_dac_to_mix(codec, pin, dac);
+ if (nid_has_mute(codec, pin, HDA_OUTPUT))
+ return pin;
+ else if (mix && nid_has_mute(codec, mix, HDA_INPUT))
+ return mix;
+ else if (nid_has_mute(codec, dac, HDA_OUTPUT))
+ return dac;
+ return 0;
+}
+
+static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec,
+ hda_nid_t pin, hda_nid_t dac)
+{
+ hda_nid_t mix = alc_auto_dac_to_mix(codec, pin, dac);
+ if (nid_has_volume(codec, dac, HDA_OUTPUT))
+ return dac;
+ else if (nid_has_volume(codec, mix, HDA_OUTPUT))
+ return mix;
+ else if (nid_has_volume(codec, pin, HDA_OUTPUT))
+ return pin;
+ return 0;
+}
/* add playback controls from the parsed DAC table */
-static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
+static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
struct alc_spec *spec = codec->spec;
- static const char * const chname[4] = {
- "Front", "Surround", NULL /*CLFE*/, "Side"
- };
- const char *pfx = alc_get_line_out_pfx(spec, true);
- hda_nid_t nid, mix, pin;
int i, err, noutputs;
noutputs = cfg->line_outs;
@@ -19053,41 +2933,41 @@
noutputs += spec->multi_ios;
for (i = 0; i < noutputs; i++) {
- nid = spec->multiout.dac_nids[i];
- if (!nid)
+ const char *name;
+ int index;
+ hda_nid_t dac, pin;
+ hda_nid_t sw, vol;
+
+ dac = spec->multiout.dac_nids[i];
+ if (!dac)
continue;
if (i >= cfg->line_outs)
pin = spec->multi_io[i - 1].pin;
else
pin = cfg->line_out_pins[i];
- mix = alc_auto_dac_to_mix(codec, pin, nid);
- if (!mix)
- continue;
- if (!pfx && i == 2) {
+
+ sw = alc_look_for_out_mute_nid(codec, pin, dac);
+ vol = alc_look_for_out_vol_nid(codec, pin, dac);
+ name = alc_get_line_out_pfx(spec, i, true, &index);
+ if (!name) {
/* Center/LFE */
- err = alc662_add_vol_ctl(spec, "Center", nid, 1);
+ err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1);
if (err < 0)
return err;
- err = alc662_add_vol_ctl(spec, "LFE", nid, 2);
+ err = alc_auto_add_vol_ctl(codec, "LFE", 0, vol, 2);
if (err < 0)
return err;
- err = alc662_add_sw_ctl(spec, "Center", mix, 1);
+ err = alc_auto_add_sw_ctl(codec, "Center", 0, sw, 1);
if (err < 0)
return err;
- err = alc662_add_sw_ctl(spec, "LFE", mix, 2);
+ err = alc_auto_add_sw_ctl(codec, "LFE", 0, sw, 2);
if (err < 0)
return err;
} else {
- const char *name = pfx;
- int index = i;
- if (!name) {
- name = chname[i];
- index = 0;
- }
- err = __alc662_add_vol_ctl(spec, name, nid, index, 3);
+ err = alc_auto_add_stereo_vol(codec, name, index, vol);
if (err < 0)
return err;
- err = __alc662_add_sw_ctl(spec, name, mix, index, 3);
+ err = alc_auto_add_stereo_sw(codec, name, index, sw);
if (err < 0)
return err;
}
@@ -19096,18 +2976,16 @@
}
/* add playback controls for speaker and HP outputs */
-/* return DAC nid if any new DAC is assigned */
-static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
- const char *pfx)
+static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
+ hda_nid_t dac, const char *pfx)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t nid, mix;
+ hda_nid_t sw, vol;
int err;
if (!pin)
return 0;
- nid = alc_auto_look_for_dac(codec, pin);
- if (!nid) {
+ if (!dac) {
/* the corresponding DAC is already occupied */
if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
return 0; /* no way */
@@ -19116,50 +2994,71 @@
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
}
- mix = alc_auto_dac_to_mix(codec, pin, nid);
- if (!mix)
- return 0;
- err = alc662_add_vol_ctl(spec, pfx, nid, 3);
+ sw = alc_look_for_out_mute_nid(codec, pin, dac);
+ vol = alc_look_for_out_vol_nid(codec, pin, dac);
+ err = alc_auto_add_stereo_vol(codec, pfx, 0, vol);
if (err < 0)
return err;
- err = alc662_add_sw_ctl(spec, pfx, mix, 3);
+ err = alc_auto_add_stereo_sw(codec, pfx, 0, sw);
if (err < 0)
return err;
- return nid;
+ return 0;
}
-/* create playback/capture controls for input pins */
-#define alc662_auto_create_input_ctls \
- alc882_auto_create_input_ctls
+static int alc_auto_create_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ return alc_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
+ spec->multiout.hp_nid,
+ "Headphone");
+}
-static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type,
+static int alc_auto_create_speaker_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ return alc_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0],
+ spec->multiout.extra_out_nid[0],
+ "Speaker");
+}
+
+static void alc_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t pin, int pin_type,
hda_nid_t dac)
{
int i, num;
+ hda_nid_t nid, mix = 0;
hda_nid_t srcs[HDA_MAX_CONNECTIONS];
- alc_set_pin_output(codec, nid, pin_type);
+ alc_set_pin_output(codec, pin, pin_type);
+ nid = alc_go_down_to_selector(codec, pin);
num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs));
for (i = 0; i < num; i++) {
if (alc_auto_mix_to_dac(codec, srcs[i]) != dac)
continue;
- /* need the manual connection? */
- if (num > 1)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL, i);
- /* unmute mixer widget inputs */
- snd_hda_codec_write(codec, srcs[i], 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(0));
- snd_hda_codec_write(codec, srcs[i], 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(1));
- return;
+ mix = srcs[i];
+ break;
}
+ if (!mix)
+ return;
+
+ /* need the manual connection? */
+ if (num > 1)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i);
+ /* unmute mixer widget inputs */
+ if (nid_has_mute(codec, mix, HDA_INPUT)) {
+ snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+ snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(1));
+ }
+ /* initialize volume */
+ nid = alc_look_for_out_vol_nid(codec, pin, dac);
+ if (nid)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_ZERO);
}
-static void alc662_auto_init_multi_out(struct hda_codec *codec)
+static void alc_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -19168,49 +3067,26 @@
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
if (nid)
- alc662_auto_set_output_and_unmute(codec, nid, pin_type,
+ alc_auto_set_output_and_unmute(codec, nid, pin_type,
spec->multiout.dac_nids[i]);
}
}
-static void alc662_auto_init_hp_out(struct hda_codec *codec)
+static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
if (pin)
- alc662_auto_set_output_and_unmute(codec, pin, PIN_HP,
+ alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
spec->multiout.hp_nid);
pin = spec->autocfg.speaker_pins[0];
if (pin)
- alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+ alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
spec->multiout.extra_out_nid[0]);
}
-#define ALC662_PIN_CD_NID ALC880_PIN_CD_NID
-
-static void alc662_auto_init_analog_input(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
- if (alc_is_input_pin(codec, nid)) {
- alc_set_input_pin(codec, nid, cfg->inputs[i].type);
- if (nid != ALC662_PIN_CD_NID &&
- (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
- }
- }
-}
-
-#define alc662_auto_init_input_src alc882_auto_init_input_src
-
/*
* multi-io helper
*/
@@ -19320,6 +3196,8 @@
for (i = 0; i < spec->multi_ios; i++)
alc_set_multi_io(codec, i, i < ch);
spec->multiout.max_channels = spec->ext_channel_count;
+ if (spec->need_dac_fix && !spec->const_channel_count)
+ spec->multiout.num_dacs = spec->multiout.max_channels / 2;
return 1;
}
@@ -19331,15 +3209,29 @@
.put = alc_auto_ch_mode_put,
};
-static int alc_auto_add_multi_channel_mode(struct hda_codec *codec)
+static int alc_auto_add_multi_channel_mode(struct hda_codec *codec,
+ int (*fill_dac)(struct hda_codec *))
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
unsigned int location, defcfg;
int num_pins;
+ if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->hp_outs == 1) {
+ /* use HP as primary out */
+ cfg->speaker_outs = cfg->line_outs;
+ memcpy(cfg->speaker_pins, cfg->line_out_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->line_outs = cfg->hp_outs;
+ memcpy(cfg->line_out_pins, cfg->hp_pins, sizeof(cfg->hp_pins));
+ cfg->hp_outs = 0;
+ memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+ if (fill_dac)
+ fill_dac(codec);
+ }
if (cfg->line_outs != 1 ||
- cfg->line_out_type != AUTO_PIN_LINE_OUT)
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
return 0;
defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]);
@@ -19364,79 +3256,1798 @@
return 0;
}
-static int alc662_parse_auto_config(struct hda_codec *codec)
+/* filter out invalid adc_nids (and capsrc_nids) that don't give all
+ * active input pins
+ */
+static void alc_remove_invalid_adc_nids(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ const struct hda_input_mux *imux;
+ hda_nid_t adc_nids[ARRAY_SIZE(spec->private_adc_nids)];
+ hda_nid_t capsrc_nids[ARRAY_SIZE(spec->private_adc_nids)];
+ int i, n, nums;
+
+ imux = spec->input_mux;
+ if (!imux)
+ return;
+ if (spec->dyn_adc_switch)
+ return;
+
+ nums = 0;
+ for (n = 0; n < spec->num_adc_nids; n++) {
+ hda_nid_t cap = spec->private_capsrc_nids[n];
+ int num_conns = snd_hda_get_conn_list(codec, cap, NULL);
+ for (i = 0; i < imux->num_items; i++) {
+ hda_nid_t pin = spec->imux_pins[i];
+ if (pin) {
+ if (get_connection_index(codec, cap, pin) < 0)
+ break;
+ } else if (num_conns <= imux->items[i].index)
+ break;
+ }
+ if (i >= imux->num_items) {
+ adc_nids[nums] = spec->private_adc_nids[n];
+ capsrc_nids[nums++] = cap;
+ }
+ }
+ if (!nums) {
+ /* check whether ADC-switch is possible */
+ if (!alc_check_dyn_adc_switch(codec)) {
+ printk(KERN_WARNING "hda_codec: %s: no valid ADC found;"
+ " using fallback 0x%x\n",
+ codec->chip_name, spec->private_adc_nids[0]);
+ spec->num_adc_nids = 1;
+ spec->auto_mic = 0;
+ return;
+ }
+ } else if (nums != spec->num_adc_nids) {
+ memcpy(spec->private_adc_nids, adc_nids,
+ nums * sizeof(hda_nid_t));
+ memcpy(spec->private_capsrc_nids, capsrc_nids,
+ nums * sizeof(hda_nid_t));
+ spec->num_adc_nids = nums;
+ }
+
+ if (spec->auto_mic)
+ alc_auto_mic_check_imux(codec); /* check auto-mic setups */
+ else if (spec->input_mux->num_items == 1)
+ spec->num_adc_nids = 1; /* reduce to a single ADC */
+}
+
+/*
+ * initialize ADC paths
+ */
+static void alc_auto_init_adc(struct hda_codec *codec, int adc_idx)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid;
+
+ nid = spec->adc_nids[adc_idx];
+ /* mute ADC */
+ if (nid_has_mute(codec, nid, HDA_INPUT)) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(0));
+ return;
+ }
+ if (!spec->capsrc_nids)
+ return;
+ nid = spec->capsrc_nids[adc_idx];
+ if (nid_has_mute(codec, nid, HDA_OUTPUT))
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+}
+
+static void alc_auto_init_input_src(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int c, nums;
+
+ for (c = 0; c < spec->num_adc_nids; c++)
+ alc_auto_init_adc(codec, c);
+ if (spec->dyn_adc_switch)
+ nums = 1;
+ else
+ nums = spec->num_adc_nids;
+ for (c = 0; c < nums; c++)
+ alc_mux_select(codec, 0, spec->cur_mux[c], true);
+}
+
+/* add mic boosts if needed */
+static int alc_auto_add_mic_boost(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, err;
+ int type_idx = 0;
+ hda_nid_t nid;
+ const char *prev_label = NULL;
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (cfg->inputs[i].type > AUTO_PIN_MIC)
+ break;
+ nid = cfg->inputs[i].pin;
+ if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) {
+ const char *label;
+ char boost_label[32];
+
+ label = hda_get_autocfg_input_label(codec, cfg, i);
+ if (prev_label && !strcmp(label, prev_label))
+ type_idx++;
+ else
+ type_idx = 0;
+ prev_label = label;
+
+ snprintf(boost_label, sizeof(boost_label),
+ "%s Boost Volume", label);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ boost_label, type_idx,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+/* select or unmute the given capsrc route */
+static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap,
+ int idx)
+{
+ if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+ snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+ HDA_AMP_MUTE, 0);
+ } else if (snd_hda_get_conn_list(codec, cap, NULL) > 1) {
+ snd_hda_codec_write_cache(codec, cap, 0,
+ AC_VERB_SET_CONNECT_SEL, idx);
+ }
+}
+
+/* set the default connection to that pin */
+static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ if (!pin)
+ return 0;
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ hda_nid_t cap = spec->capsrc_nids ?
+ spec->capsrc_nids[i] : spec->adc_nids[i];
+ int idx;
+
+ idx = get_connection_index(codec, cap, pin);
+ if (idx < 0)
+ continue;
+ select_or_unmute_capsrc(codec, cap, idx);
+ return i; /* return the found index */
+ }
+ return -1; /* not found */
+}
+
+/* initialize some special cases for input sources */
+static void alc_init_special_input_src(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->autocfg.num_inputs; i++)
+ init_capsrc_for_pin(codec, spec->autocfg.inputs[i].pin);
+}
+
+/* assign appropriate capture mixers */
+static void set_capture_mixer(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ static const struct snd_kcontrol_new *caps[2][3] = {
+ { alc_capture_mixer_nosrc1,
+ alc_capture_mixer_nosrc2,
+ alc_capture_mixer_nosrc3 },
+ { alc_capture_mixer1,
+ alc_capture_mixer2,
+ alc_capture_mixer3 },
+ };
+
+ /* check whether either of ADC or MUX has a volume control */
+ if (!nid_has_volume(codec, spec->adc_nids[0], HDA_INPUT)) {
+ if (!spec->capsrc_nids)
+ return; /* no volume */
+ if (!nid_has_volume(codec, spec->capsrc_nids[0], HDA_OUTPUT))
+ return; /* no volume in capsrc, too */
+ spec->vol_in_capsrc = 1;
+ }
+
+ if (spec->num_adc_nids > 0) {
+ int mux = 0;
+ int num_adcs = 0;
+
+ if (spec->input_mux && spec->input_mux->num_items > 1)
+ mux = 1;
+ if (spec->auto_mic) {
+ num_adcs = 1;
+ mux = 0;
+ } else if (spec->dyn_adc_switch)
+ num_adcs = 1;
+ if (!num_adcs) {
+ if (spec->num_adc_nids > 3)
+ spec->num_adc_nids = 3;
+ else if (!spec->num_adc_nids)
+ return;
+ num_adcs = spec->num_adc_nids;
+ }
+ spec->cap_mixer = caps[mux][num_adcs - 1];
+ }
+}
+
+/*
+ * standard auto-parser initializations
+ */
+static void alc_auto_init_std(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc_auto_init_multi_out(codec);
+ alc_auto_init_extra_out(codec);
+ alc_auto_init_analog_input(codec);
+ alc_auto_init_input_src(codec);
+ alc_auto_init_digital(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+/*
+ * Digital-beep handlers
+ */
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+#define set_beep_amp(spec, nid, idx, dir) \
+ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
+
+static const struct snd_pci_quirk beep_white_list[] = {
+ SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
+ SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
+ SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
+ SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
+ {}
+};
+
+static inline int has_cdefine_beep(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ const struct snd_pci_quirk *q;
+ q = snd_pci_quirk_lookup(codec->bus->pci, beep_white_list);
+ if (q)
+ return q->value;
+ return spec->cdefine.enable_pcbeep;
+}
+#else
+#define set_beep_amp(spec, nid, idx, dir) /* NOP */
+#define has_cdefine_beep(codec) 0
+#endif
+
+/* parse the BIOS configuration and set up the alc_spec */
+/* return 1 if successful, 0 if the proper config is not found,
+ * or a negative error code
+ */
+static int alc_parse_auto_config(struct hda_codec *codec,
+ const hda_nid_t *ignore_nids,
+ const hda_nid_t *ssid_nids)
{
struct alc_spec *spec = codec->spec;
int err;
- static const hda_nid_t alc662_ignore[] = { 0x1d, 0 };
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc662_ignore);
+ ignore_nids);
if (err < 0)
return err;
- if (!spec->autocfg.line_outs)
+ if (!spec->autocfg.line_outs) {
+ if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
+ spec->multiout.max_channels = 2;
+ spec->no_analog = 1;
+ goto dig_only;
+ }
return 0; /* can't find valid BIOS pin config */
-
- err = alc662_auto_fill_dac_nids(codec, &spec->autocfg);
+ }
+ err = alc_auto_fill_dac_nids(codec);
if (err < 0)
return err;
- err = alc_auto_add_multi_channel_mode(codec);
+ err = alc_auto_add_multi_channel_mode(codec, alc_auto_fill_dac_nids);
if (err < 0)
return err;
- err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg);
+ err = alc_auto_create_multi_out_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
- err = alc662_auto_create_extra_out(codec,
- spec->autocfg.speaker_pins[0],
- "Speaker");
+ err = alc_auto_create_hp_out(codec);
if (err < 0)
return err;
- if (err)
- spec->multiout.extra_out_nid[0] = err;
- err = alc662_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
- "Headphone");
+ err = alc_auto_create_speaker_out(codec);
if (err < 0)
return err;
- if (err)
- spec->multiout.hp_nid = err;
- err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
+ err = alc_auto_create_input_ctls(codec);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+ dig_only:
alc_auto_parse_digital(codec);
+ if (!spec->no_analog)
+ alc_remove_invalid_adc_nids(codec);
+
+ if (ssid_nids)
+ alc_ssid_check(codec, ssid_nids);
+
+ if (!spec->no_analog) {
+ alc_auto_check_switches(codec);
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+ }
+
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
- spec->num_mux_defs = 1;
- spec->input_mux = &spec->private_imux[0];
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
- codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0x21);
- else
- alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
-
return 1;
}
-/* additional initialization for auto-configuration model */
-static void alc662_auto_init(struct hda_codec *codec)
+static int alc880_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
+ static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+ return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static const struct hda_amp_list alc880_loopbacks[] = {
+ { 0x0b, HDA_INPUT, 0 },
+ { 0x0b, HDA_INPUT, 1 },
+ { 0x0b, HDA_INPUT, 2 },
+ { 0x0b, HDA_INPUT, 3 },
+ { 0x0b, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+/*
+ * board setups
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#define alc_board_config \
+ snd_hda_check_board_config
+#define alc_board_codec_sid_config \
+ snd_hda_check_board_codec_sid_config
+#include "alc_quirks.c"
+#else
+#define alc_board_config(codec, nums, models, tbl) -1
+#define alc_board_codec_sid_config(codec, nums, models, tbl) -1
+#define setup_preset(codec, x) /* NOP */
+#endif
+
+/*
+ * OK, here we have finally the patch for ALC880
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc880_quirks.c"
+#endif
+
+static int patch_alc880(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->mixer_nid = 0x0b;
+ spec->need_dac_fix = 1;
+
+ board_config = alc_board_config(codec, ALC880_MODEL_LAST,
+ alc880_models, alc880_cfg_tbl);
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc880_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using 3-stack mode...\n");
+ board_config = ALC880_3ST;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc880_presets[board_config]);
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog) {
+ err = snd_hda_attach_beep_device(codec, 0x1);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+ }
+
+ spec->vmaster_nid = 0x0c;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc880_loopbacks;
+#endif
+
+ return 0;
+}
+
+
+/*
+ * ALC260 support
+ */
+static int alc260_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc260_ignore[] = { 0x17, 0 };
+ static const hda_nid_t alc260_ssids[] = { 0x10, 0x15, 0x0f, 0 };
+ return alc_parse_auto_config(codec, alc260_ignore, alc260_ssids);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static const struct hda_amp_list alc260_loopbacks[] = {
+ { 0x07, HDA_INPUT, 0 },
+ { 0x07, HDA_INPUT, 1 },
+ { 0x07, HDA_INPUT, 2 },
+ { 0x07, HDA_INPUT, 3 },
+ { 0x07, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+/*
+ * Pin config fixes
+ */
+enum {
+ PINFIX_HP_DC5750,
+};
+
+static const struct alc_fixup alc260_fixups[] = {
+ [PINFIX_HP_DC5750] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x11, 0x90130110 }, /* speaker */
+ { }
+ }
+ },
+};
+
+static const struct snd_pci_quirk alc260_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750),
+ {}
+};
+
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc260_quirks.c"
+#endif
+
+static int patch_alc260(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->mixer_nid = 0x07;
+
+ board_config = alc_board_config(codec, ALC260_MODEL_LAST,
+ alc260_models, alc260_cfg_tbl);
+ if (board_config < 0) {
+ snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc260_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC260_BASIC;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc260_presets[board_config]);
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog) {
+ err = snd_hda_attach_beep_device(codec, 0x1);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
+ }
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
+ spec->vmaster_nid = 0x08;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
+ spec->shutup = alc_eapd_shutup;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc260_loopbacks;
+#endif
+
+ return 0;
+}
+
+
+/*
+ * ALC882/883/885/888/889 support
+ *
+ * ALC882 is almost identical with ALC880 but has cleaner and more flexible
+ * configuration. Each pin widget can choose any input DACs and a mixer.
+ * Each ADC is connected from a mixer of all inputs. This makes possible
+ * 6-channel independent captures.
+ *
+ * In addition, an independent DAC for the multi-playback (not used in this
+ * driver yet).
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc882_loopbacks alc880_loopbacks
+#endif
+
+/*
+ * Pin config fixes
+ */
+enum {
+ PINFIX_ABIT_AW9D_MAX,
+ PINFIX_LENOVO_Y530,
+ PINFIX_PB_M5210,
+ PINFIX_ACER_ASPIRE_7736,
+};
+
+static const struct alc_fixup alc882_fixups[] = {
+ [PINFIX_ABIT_AW9D_MAX] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x15, 0x01080104 }, /* side */
+ { 0x16, 0x01011012 }, /* rear */
+ { 0x17, 0x01016011 }, /* clfe */
+ { }
+ }
+ },
+ [PINFIX_LENOVO_Y530] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x15, 0x99130112 }, /* rear int speakers */
+ { 0x16, 0x99130111 }, /* subwoofer */
+ { }
+ }
+ },
+ [PINFIX_PB_M5210] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
+ {}
+ }
+ },
+ [PINFIX_ACER_ASPIRE_7736] = {
+ .type = ALC_FIXUP_SKU,
+ .v.sku = ALC_FIXUP_SKU_IGNORE,
+ },
+};
+
+static const struct snd_pci_quirk alc882_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210),
+ SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
+ SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
+ SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
+ {}
+};
+
+/*
+ * BIOS auto configuration
+ */
+/* almost identical with ALC880 parser... */
+static int alc882_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc882_ignore[] = { 0x1d, 0 };
+ static const hda_nid_t alc882_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+ return alc_parse_auto_config(codec, alc882_ignore, alc882_ssids);
+}
+
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc882_quirks.c"
+#endif
+
+static int patch_alc882(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->mixer_nid = 0x0b;
+
+ switch (codec->vendor_id) {
+ case 0x10ec0882:
+ case 0x10ec0885:
+ break;
+ default:
+ /* ALC883 and variants */
+ alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+ break;
+ }
+
+ board_config = alc_board_config(codec, ALC882_MODEL_LAST,
+ alc882_models, alc882_cfg_tbl);
+
+ if (board_config < 0)
+ board_config = alc_board_codec_sid_config(codec,
+ ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ alc_auto_parse_customize_define(codec);
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc882_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC882_3ST_DIG;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc882_presets[board_config]);
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog && has_cdefine_beep(codec)) {
+ err = snd_hda_attach_beep_device(codec, 0x1);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+ }
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
+ spec->vmaster_nid = 0x0c;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
+
+ alc_init_jacks(codec);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc882_loopbacks;
+#endif
+
+ return 0;
+}
+
+
+/*
+ * ALC262 support
+ */
+static int alc262_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc262_ignore[] = { 0x1d, 0 };
+ static const hda_nid_t alc262_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+ return alc_parse_auto_config(codec, alc262_ignore, alc262_ssids);
+}
+
+/*
+ * Pin config fixes
+ */
+enum {
+ PINFIX_FSC_H270,
+ PINFIX_HP_Z200,
+};
+
+static const struct alc_fixup alc262_fixups[] = {
+ [PINFIX_FSC_H270] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x99130110 }, /* speaker */
+ { 0x15, 0x0221142f }, /* front HP */
+ { 0x1b, 0x0121141f }, /* rear HP */
+ { }
+ }
+ },
+ [PINFIX_HP_Z200] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x16, 0x99130120 }, /* internal speaker */
+ { }
+ }
+ },
+};
+
+static const struct snd_pci_quirk alc262_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
+ SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
+ {}
+};
+
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc262_loopbacks alc880_loopbacks
+#endif
+
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc262_quirks.c"
+#endif
+
+static int patch_alc262(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->mixer_nid = 0x0b;
+
+#if 0
+ /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is
+ * under-run
+ */
+ {
+ int tmp;
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
+ }
+#endif
+ alc_auto_parse_customize_define(codec);
+
+ alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+
+ board_config = alc_board_config(codec, ALC262_MODEL_LAST,
+ alc262_models, alc262_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc262_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC262_BASIC;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc262_presets[board_config]);
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog && has_cdefine_beep(codec)) {
+ err = snd_hda_attach_beep_device(codec, 0x1);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+ }
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
+ spec->vmaster_nid = 0x0c;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
+ spec->shutup = alc_eapd_shutup;
+
+ alc_init_jacks(codec);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc262_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC268
+ */
+/* bind Beep switches of both NID 0x0f and 0x10 */
+static const struct hda_bind_ctls alc268_bind_beep_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x10, 3, 1, HDA_INPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc268_beep_mixer[] = {
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT),
+ HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw),
+ { }
+};
+
+/* set PCBEEP vol = 0, mute connections */
+static const struct hda_verb alc268_beep_init_verbs[] = {
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { }
+};
+
+/*
+ * BIOS auto configuration
+ */
+static int alc268_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc268_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+ struct alc_spec *spec = codec->spec;
+ int err = alc_parse_auto_config(codec, NULL, alc268_ssids);
+ if (err > 0) {
+ if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) {
+ add_mixer(spec, alc268_beep_mixer);
+ add_verb(spec, alc268_beep_init_verbs);
+ }
+ }
+ return err;
+}
+
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc268_quirks.c"
+#endif
+
+static int patch_alc268(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int i, has_beep, err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* ALC268 has no aa-loopback mixer */
+
+ board_config = alc_board_config(codec, ALC268_MODEL_LAST,
+ alc268_models, alc268_cfg_tbl);
+
+ if (board_config < 0)
+ board_config = alc_board_codec_sid_config(codec,
+ ALC268_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc268_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC268_3ST;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc268_presets[board_config]);
+
+ has_beep = 0;
+ for (i = 0; i < spec->num_mixers; i++) {
+ if (spec->mixers[i] == alc268_beep_mixer) {
+ has_beep = 1;
+ break;
+ }
+ }
+
+ if (has_beep) {
+ err = snd_hda_attach_beep_device(codec, 0x1);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
+ /* override the amp caps for beep generator */
+ snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
+ (0x0c << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (0 << AC_AMPCAP_MUTE_SHIFT));
+ }
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ spec->vmaster_nid = 0x02;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
+ spec->shutup = alc_eapd_shutup;
+
+ alc_init_jacks(codec);
+
+ return 0;
+}
+
+/*
+ * ALC269
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc269_loopbacks alc880_loopbacks
+#endif
+
+static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_44100, /* fixed rate */
+ /* NID is set in alc_build_pcms */
+ .ops = {
+ .open = alc_playback_pcm_open,
+ .prepare = alc_playback_pcm_prepare,
+ .cleanup = alc_playback_pcm_cleanup
+ },
+};
+
+static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_44100, /* fixed rate */
+ /* NID is set in alc_build_pcms */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc269_mic2_for_mute_led(struct hda_codec *codec)
+{
+ switch (codec->subsystem_id) {
+ case 0x103c1586:
+ return 1;
+ }
+ return 0;
+}
+
+static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid)
+{
+ /* update mute-LED according to the speaker mute state */
+ if (nid == 0x01 || nid == 0x14) {
+ int pinval;
+ if (snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0) &
+ HDA_AMP_MUTE)
+ pinval = 0x24;
+ else
+ pinval = 0x20;
+ /* mic2 vref pin is used for mute LED control */
+ snd_hda_codec_update_cache(codec, 0x19, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinval);
+ }
+ return alc_check_power_status(codec, nid);
+}
+#endif /* CONFIG_SND_HDA_POWER_SAVE */
+
+/* different alc269-variants */
+enum {
+ ALC269_TYPE_ALC269VA,
+ ALC269_TYPE_ALC269VB,
+ ALC269_TYPE_ALC269VC,
+};
+
+/*
+ * BIOS auto configuration
+ */
+static int alc269_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc269_ignore[] = { 0x1d, 0 };
+ static const hda_nid_t alc269_ssids[] = { 0, 0x1b, 0x14, 0x21 };
+ static const hda_nid_t alc269va_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+ struct alc_spec *spec = codec->spec;
+ const hda_nid_t *ssids = spec->codec_variant == ALC269_TYPE_ALC269VA ?
+ alc269va_ssids : alc269_ssids;
+
+ return alc_parse_auto_config(codec, alc269_ignore, ssids);
+}
+
+static void alc269_toggle_power_output(struct hda_codec *codec, int power_up)
+{
+ int val = alc_read_coef_idx(codec, 0x04);
+ if (power_up)
+ val |= 1 << 11;
+ else
+ val &= ~(1 << 11);
+ alc_write_coef_idx(codec, 0x04, val);
+}
+
+static void alc269_shutup(struct hda_codec *codec)
+{
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017)
+ alc269_toggle_power_output(codec, 0);
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+ alc269_toggle_power_output(codec, 0);
+ msleep(150);
+ }
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+static int alc269_resume(struct hda_codec *codec)
+{
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+ alc269_toggle_power_output(codec, 0);
+ msleep(150);
+ }
+
+ codec->patch_ops.init(codec);
+
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
+ alc269_toggle_power_output(codec, 1);
+ msleep(200);
+ }
+
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018)
+ alc269_toggle_power_output(codec, 1);
+
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ hda_call_check_power_status(codec, 0x01);
+ return 0;
+}
+#endif /* SND_HDA_NEEDS_RESUME */
+
+static void alc269_fixup_hweq(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ int coef;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ coef = alc_read_coef_idx(codec, 0x1e);
+ alc_write_coef_idx(codec, 0x1e, coef | 0x80);
+}
+
+static void alc271_fixup_dmic(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ static const struct hda_verb verbs[] = {
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x4000},
+ {}
+ };
+ unsigned int cfg;
+
+ if (strcmp(codec->chip_name, "ALC271X"))
+ return;
+ cfg = snd_hda_codec_get_pincfg(codec, 0x12);
+ if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
+ snd_hda_sequence_write(codec, verbs);
+}
+
+static void alc269_fixup_pcm_44k(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
{
struct alc_spec *spec = codec->spec;
- alc662_auto_init_multi_out(codec);
- alc662_auto_init_hp_out(codec);
- alc662_auto_init_analog_input(codec);
- alc662_auto_init_input_src(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
+
+ if (action != ALC_FIXUP_ACT_PROBE)
+ return;
+
+ /* Due to a hardware problem on Lenovo Ideadpad, we need to
+ * fix the sample rate of analog I/O to 44.1kHz
+ */
+ spec->stream_analog_playback = &alc269_44k_pcm_analog_playback;
+ spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
+}
+
+enum {
+ ALC269_FIXUP_SONY_VAIO,
+ ALC275_FIXUP_SONY_VAIO_GPIO2,
+ ALC269_FIXUP_DELL_M101Z,
+ ALC269_FIXUP_SKU_IGNORE,
+ ALC269_FIXUP_ASUS_G73JW,
+ ALC269_FIXUP_LENOVO_EAPD,
+ ALC275_FIXUP_SONY_HWEQ,
+ ALC271_FIXUP_DMIC,
+ ALC269_FIXUP_PCM_44K,
+};
+
+static const struct alc_fixup alc269_fixups[] = {
+ [ALC269_FIXUP_SONY_VAIO] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
+ {}
+ }
+ },
+ [ALC275_FIXUP_SONY_VAIO_GPIO2] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x04},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_SONY_VAIO
+ },
+ [ALC269_FIXUP_DELL_M101Z] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Enables internal speaker */
+ {0x20, AC_VERB_SET_COEF_INDEX, 13},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x4040},
+ {}
+ }
+ },
+ [ALC269_FIXUP_SKU_IGNORE] = {
+ .type = ALC_FIXUP_SKU,
+ .v.sku = ALC_FIXUP_SKU_IGNORE,
+ },
+ [ALC269_FIXUP_ASUS_G73JW] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x17, 0x99130111 }, /* subwoofer */
+ { }
+ }
+ },
+ [ALC269_FIXUP_LENOVO_EAPD] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0},
+ {}
+ }
+ },
+ [ALC275_FIXUP_SONY_HWEQ] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_hweq,
+ .chained = true,
+ .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
+ },
+ [ALC271_FIXUP_DMIC] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc271_fixup_dmic,
+ },
+ [ALC269_FIXUP_PCM_44K] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_pcm_44k,
+ },
+};
+
+static const struct snd_pci_quirk alc269_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+ SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
+ SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
+ SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
+ SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+ SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+ SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+ SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
+ SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
+ {}
+};
+
+
+static int alc269_fill_coef(struct hda_codec *codec)
+{
+ int val;
+
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) < 0x015) {
+ alc_write_coef_idx(codec, 0xf, 0x960b);
+ alc_write_coef_idx(codec, 0xe, 0x8817);
+ }
+
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x016) {
+ alc_write_coef_idx(codec, 0xf, 0x960b);
+ alc_write_coef_idx(codec, 0xe, 0x8814);
+ }
+
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
+ val = alc_read_coef_idx(codec, 0x04);
+ /* Power up output pin */
+ alc_write_coef_idx(codec, 0x04, val | (1<<11));
+ }
+
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
+ val = alc_read_coef_idx(codec, 0xd);
+ if ((val & 0x0c00) >> 10 != 0x1) {
+ /* Capless ramp up clock control */
+ alc_write_coef_idx(codec, 0xd, val | (1<<10));
+ }
+ val = alc_read_coef_idx(codec, 0x17);
+ if ((val & 0x01c0) >> 6 != 0x4) {
+ /* Class D power on reset */
+ alc_write_coef_idx(codec, 0x17, val | (1<<7));
+ }
+ }
+
+ val = alc_read_coef_idx(codec, 0xd); /* Class D */
+ alc_write_coef_idx(codec, 0xd, val | (1<<14));
+
+ val = alc_read_coef_idx(codec, 0x4); /* HP */
+ alc_write_coef_idx(codec, 0x4, val | (1<<11));
+
+ return 0;
+}
+
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc269_quirks.c"
+#endif
+
+static int patch_alc269(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config, coef;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->mixer_nid = 0x0b;
+
+ alc_auto_parse_customize_define(codec);
+
+ if (codec->vendor_id == 0x10ec0269) {
+ spec->codec_variant = ALC269_TYPE_ALC269VA;
+ coef = alc_read_coef_idx(codec, 0);
+ if ((coef & 0x00f0) == 0x0010) {
+ if (codec->bus->pci->subsystem_vendor == 0x1025 &&
+ spec->cdefine.platform_type == 1) {
+ alc_codec_rename(codec, "ALC271X");
+ } else if ((coef & 0xf000) == 0x2000) {
+ alc_codec_rename(codec, "ALC259");
+ } else if ((coef & 0xf000) == 0x3000) {
+ alc_codec_rename(codec, "ALC258");
+ } else if ((coef & 0xfff0) == 0x3010) {
+ alc_codec_rename(codec, "ALC277");
+ } else {
+ alc_codec_rename(codec, "ALC269VB");
+ }
+ spec->codec_variant = ALC269_TYPE_ALC269VB;
+ } else if ((coef & 0x00f0) == 0x0020) {
+ if (coef == 0xa023)
+ alc_codec_rename(codec, "ALC259");
+ else if (coef == 0x6023)
+ alc_codec_rename(codec, "ALC281X");
+ else if (codec->bus->pci->subsystem_vendor == 0x17aa &&
+ codec->bus->pci->subsystem_device == 0x21f3)
+ alc_codec_rename(codec, "ALC3202");
+ else
+ alc_codec_rename(codec, "ALC269VC");
+ spec->codec_variant = ALC269_TYPE_ALC269VC;
+ } else
+ alc_fix_pll_init(codec, 0x20, 0x04, 15);
+ alc269_fill_coef(codec);
+ }
+
+ board_config = alc_board_config(codec, ALC269_MODEL_LAST,
+ alc269_models, alc269_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc269_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC269_BASIC;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc269_presets[board_config]);
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog && has_cdefine_beep(codec)) {
+ err = snd_hda_attach_beep_device(codec, 0x1);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+ }
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
+ spec->vmaster_nid = 0x02;
+
+ codec->patch_ops = alc_patch_ops;
+#ifdef SND_HDA_NEEDS_RESUME
+ codec->patch_ops.resume = alc269_resume;
+#endif
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
+ spec->shutup = alc269_shutup;
+
+ alc_init_jacks(codec);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc269_loopbacks;
+ if (alc269_mic2_for_mute_led(codec))
+ codec->patch_ops.check_power_status = alc269_mic2_mute_check_ps;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC861
+ */
+
+static int alc861_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc861_ignore[] = { 0x1d, 0 };
+ static const hda_nid_t alc861_ssids[] = { 0x0e, 0x0f, 0x0b, 0 };
+ return alc_parse_auto_config(codec, alc861_ignore, alc861_ssids);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static const struct hda_amp_list alc861_loopbacks[] = {
+ { 0x15, HDA_INPUT, 0 },
+ { 0x15, HDA_INPUT, 1 },
+ { 0x15, HDA_INPUT, 2 },
+ { 0x15, HDA_INPUT, 3 },
+ { } /* end */
+};
+#endif
+
+
+/* Pin config fixes */
+enum {
+ PINFIX_FSC_AMILO_PI1505,
+};
+
+static const struct alc_fixup alc861_fixups[] = {
+ [PINFIX_FSC_AMILO_PI1505] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x0b, 0x0221101f }, /* HP */
+ { 0x0f, 0x90170310 }, /* speaker */
+ { }
+ }
+ },
+};
+
+static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
+ {}
+};
+
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc861_quirks.c"
+#endif
+
+static int patch_alc861(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->mixer_nid = 0x15;
+
+ board_config = alc_board_config(codec, ALC861_MODEL_LAST,
+ alc861_models, alc861_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc861_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC861_3ST_DIG;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc861_presets[board_config]);
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog) {
+ err = snd_hda_attach_beep_device(codec, 0x23);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
+ }
+
+ spec->vmaster_nid = 0x03;
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC_MODEL_AUTO) {
+ spec->init_hook = alc_auto_init_std;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->power_hook = alc_power_eapd;
+#endif
+ }
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC861-VD support
+ *
+ * Based on ALC882
+ *
+ * In addition, an independent DAC
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc861vd_loopbacks alc880_loopbacks
+#endif
+
+static int alc861vd_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc861vd_ignore[] = { 0x1d, 0 };
+ static const hda_nid_t alc861vd_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+ return alc_parse_auto_config(codec, alc861vd_ignore, alc861vd_ssids);
+}
+
+enum {
+ ALC660VD_FIX_ASUS_GPIO1
+};
+
+/* reset GPIO1 */
+static const struct alc_fixup alc861vd_fixups[] = {
+ [ALC660VD_FIX_ASUS_GPIO1] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+ { }
+ }
+ },
+};
+
+static const struct snd_pci_quirk alc861vd_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
+ {}
+};
+
+static const struct hda_verb alc660vd_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc861vd_quirks.c"
+#endif
+
+static int patch_alc861vd(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->mixer_nid = 0x0b;
+
+ board_config = alc_board_config(codec, ALC861VD_MODEL_LAST,
+ alc861vd_models, alc861vd_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc861vd_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC861VD_3ST;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc861vd_presets[board_config]);
+
+ if (codec->vendor_id == 0x10ec0660) {
+ /* always turn on EAPD */
+ add_verb(spec, alc660vd_eapd_verbs);
+ }
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog) {
+ err = snd_hda_attach_beep_device(codec, 0x23);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+ set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+ }
+
+ spec->vmaster_nid = 0x02;
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
+ codec->patch_ops = alc_patch_ops;
+
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
+ spec->shutup = alc_eapd_shutup;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861vd_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC662 support
+ *
+ * ALC662 is almost identical with ALC880 but has cleaner and more flexible
+ * configuration. Each pin widget can choose any input DACs and a mixer.
+ * Each ADC is connected from a mixer of all inputs. This makes possible
+ * 6-channel independent captures.
+ *
+ * In addition, an independent DAC for the multi-playback (not used in this
+ * driver yet).
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc662_loopbacks alc880_loopbacks
+#endif
+
+/*
+ * BIOS auto configuration
+ */
+
+static int alc662_parse_auto_config(struct hda_codec *codec)
+{
+ static const hda_nid_t alc662_ignore[] = { 0x1d, 0 };
+ static const hda_nid_t alc663_ssids[] = { 0x15, 0x1b, 0x14, 0x21 };
+ static const hda_nid_t alc662_ssids[] = { 0x15, 0x1b, 0x14, 0 };
+ const hda_nid_t *ssids;
+
+ if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
+ codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
+ ssids = alc663_ssids;
+ else
+ ssids = alc662_ssids;
+ return alc_parse_auto_config(codec, alc662_ignore, ssids);
}
static void alc272_fixup_mario(struct hda_codec *codec,
@@ -19459,6 +5070,7 @@
ALC272_FIXUP_MARIO,
ALC662_FIXUP_CZC_P10T,
ALC662_FIXUP_SKU_IGNORE,
+ ALC662_FIXUP_HP_RP5800,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -19491,12 +5103,22 @@
.type = ALC_FIXUP_SKU,
.v.sku = ALC_FIXUP_SKU_IGNORE,
},
+ [ALC662_FIXUP_HP_RP5800] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x0221201f }, /* HP out */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_SKU_IGNORE
+ },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
+ SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
@@ -19510,6 +5132,12 @@
};
+/*
+ */
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc662_quirks.c"
+#endif
+
static int patch_alc662(struct hda_codec *codec)
{
struct alc_spec *spec;
@@ -19522,6 +5150,8 @@
codec->spec = spec;
+ spec->mixer_nid = 0x0b;
+
alc_auto_parse_customize_define(codec);
alc_fix_pll_init(codec, 0x20, 0x04, 15);
@@ -19536,16 +5166,15 @@
else if (coef == 0x4011)
alc_codec_rename(codec, "ALC656");
- board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST,
- alc662_models,
- alc662_cfg_tbl);
+ board_config = alc_board_config(codec, ALC662_MODEL_LAST,
+ alc662_models, alc662_cfg_tbl);
if (board_config < 0) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
- board_config = ALC662_AUTO;
+ board_config = ALC_MODEL_AUTO;
}
- if (board_config == ALC662_AUTO) {
+ if (board_config == ALC_MODEL_AUTO) {
alc_pick_fixup(codec, alc662_fixup_models,
alc662_fixup_tbl, alc662_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -19554,42 +5183,35 @@
if (err < 0) {
alc_free(codec);
return err;
- } else if (!err) {
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
board_config = ALC662_3ST_2ch_DIG;
}
+#endif
}
- if (has_cdefine_beep(codec)) {
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc662_presets[board_config]);
+
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
+ if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0) {
alc_free(codec);
return err;
}
- }
-
- if (board_config != ALC662_AUTO)
- setup_preset(codec, &alc662_presets[board_config]);
-
- spec->stream_analog_playback = &alc662_pcm_analog_playback;
- spec->stream_analog_capture = &alc662_pcm_analog_capture;
-
- spec->stream_digital_playback = &alc662_pcm_digital_playback;
- spec->stream_digital_capture = &alc662_pcm_digital_capture;
-
- if (!spec->adc_nids) {
- spec->adc_nids = alc662_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
- }
- if (!spec->capsrc_nids)
- spec->capsrc_nids = alc662_capsrc_nids;
-
- if (!spec->cap_mixer)
- set_capture_mixer(codec);
-
- if (has_cdefine_beep(codec)) {
switch (codec->vendor_id) {
case 0x10ec0662:
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -19609,8 +5231,8 @@
alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC662_AUTO)
- spec->init_hook = alc662_auto_init;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
alc_init_jacks(codec);
@@ -19652,389 +5274,17 @@
/*
* ALC680 support
*/
-#define ALC680_DIGIN_NID ALC880_DIGIN_NID
-#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID
-#define alc680_modes alc260_modes
-static const hda_nid_t alc680_dac_nids[3] = {
- /* Lout1, Lout2, hp */
- 0x02, 0x03, 0x04
-};
-
-static const hda_nid_t alc680_adc_nids[3] = {
- /* ADC0-2 */
- /* DMIC, MIC, Line-in*/
- 0x07, 0x08, 0x09
-};
-
-/*
- * Analog capture ADC cgange
- */
-static void alc680_rec_autoswitch(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int pin_found = 0;
- int type_found = AUTO_PIN_LAST;
- hda_nid_t nid;
- int i;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- nid = cfg->inputs[i].pin;
- if (!is_jack_detectable(codec, nid))
- continue;
- if (snd_hda_jack_detect(codec, nid)) {
- if (cfg->inputs[i].type < type_found) {
- type_found = cfg->inputs[i].type;
- pin_found = nid;
- }
- }
- }
-
- nid = 0x07;
- if (pin_found)
- snd_hda_get_connections(codec, pin_found, &nid, 1);
-
- if (nid != spec->cur_adc)
- __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
- spec->cur_adc = nid;
- snd_hda_codec_setup_stream(codec, nid, spec->cur_adc_stream_tag, 0,
- spec->cur_adc_format);
-}
-
-static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->cur_adc = 0x07;
- spec->cur_adc_stream_tag = stream_tag;
- spec->cur_adc_format = format;
-
- alc680_rec_autoswitch(codec);
- return 0;
-}
-
-static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- snd_hda_codec_cleanup_stream(codec, 0x07);
- snd_hda_codec_cleanup_stream(codec, 0x08);
- snd_hda_codec_cleanup_stream(codec, 0x09);
- return 0;
-}
-
-static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
- .substreams = 1, /* can be overridden */
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in alc_build_pcms */
- .ops = {
- .prepare = alc680_capture_pcm_prepare,
- .cleanup = alc680_capture_pcm_cleanup
- },
-};
-
-static const struct snd_kcontrol_new alc680_base_mixer[] = {
- /* output mixer control */
- HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT),
- { }
-};
-
-static const struct hda_bind_ctls alc680_bind_cap_vol = {
- .ops = &snd_hda_bind_vol,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
- 0
- },
-};
-
-static const struct hda_bind_ctls alc680_bind_cap_switch = {
- .ops = &snd_hda_bind_sw,
- .values = {
- HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
- HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
- 0
- },
-};
-
-static const struct snd_kcontrol_new alc680_master_capture_mixer[] = {
- HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
- HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
- { } /* end */
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc680_init_verbs[] = {
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
-
- { }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc680_base_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x16;
- spec->autocfg.speaker_pins[0] = 0x14;
- spec->autocfg.speaker_pins[1] = 0x15;
- spec->autocfg.num_inputs = 2;
- spec->autocfg.inputs[0].pin = 0x18;
- spec->autocfg.inputs[0].type = AUTO_PIN_MIC;
- spec->autocfg.inputs[1].pin = 0x19;
- spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN;
- spec->automute = 1;
- spec->automute_mode = ALC_AUTOMUTE_AMP;
-}
-
-static void alc680_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc_hp_automute(codec);
- if ((res >> 26) == ALC880_MIC_EVENT)
- alc680_rec_autoswitch(codec);
-}
-
-static void alc680_inithook(struct hda_codec *codec)
-{
- alc_hp_automute(codec);
- alc680_rec_autoswitch(codec);
-}
-
-/* create input playback/capture controls for the given pin */
-static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
- const char *ctlname, int idx)
-{
- hda_nid_t dac;
- int err;
-
- switch (nid) {
- case 0x14:
- dac = 0x02;
- break;
- case 0x15:
- dac = 0x03;
- break;
- case 0x16:
- dac = 0x04;
- break;
- default:
- return 0;
- }
- if (spec->multiout.dac_nids[0] != dac &&
- spec->multiout.dac_nids[1] != dac) {
- err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname,
- HDA_COMPOSE_AMP_VAL(dac, 3, idx,
- HDA_OUTPUT));
- if (err < 0)
- return err;
-
- err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname,
- HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
-
- if (err < 0)
- return err;
- spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
- }
-
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int alc680_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- hda_nid_t nid;
- int err;
-
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- nid = cfg->line_out_pins[0];
- if (nid) {
- const char *name;
- if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
- name = "Speaker";
- else
- name = "Front";
- err = alc680_new_analog_output(spec, nid, name, 0);
- if (err < 0)
- return err;
- }
-
- nid = cfg->speaker_pins[0];
- if (nid) {
- err = alc680_new_analog_output(spec, nid, "Speaker", 0);
- if (err < 0)
- return err;
- }
- nid = cfg->hp_pins[0];
- if (nid) {
- err = alc680_new_analog_output(spec, nid, "Headphone", 0);
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
-static void alc680_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type)
-{
- alc_set_pin_output(codec, nid, pin_type);
-}
-
-static void alc680_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t nid = spec->autocfg.line_out_pins[0];
- if (nid) {
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- alc680_auto_set_output_and_unmute(codec, nid, pin_type);
- }
-}
-
-static void alc680_auto_init_hp_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
-
- pin = spec->autocfg.hp_pins[0];
- if (pin)
- alc680_auto_set_output_and_unmute(codec, pin, PIN_HP);
- pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc680_auto_set_output_and_unmute(codec, pin, PIN_OUT);
-}
-
-/* pcm configuration: identical with ALC880 */
-#define alc680_pcm_analog_playback alc880_pcm_analog_playback
-#define alc680_pcm_analog_capture alc880_pcm_analog_capture
-#define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
-#define alc680_pcm_digital_playback alc880_pcm_digital_playback
-#define alc680_pcm_digital_capture alc880_pcm_digital_capture
-
-/*
- * BIOS auto configuration
- */
static int alc680_parse_auto_config(struct hda_codec *codec)
{
- struct alc_spec *spec = codec->spec;
- int err;
- static const hda_nid_t alc680_ignore[] = { 0 };
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- alc680_ignore);
- if (err < 0)
- return err;
-
- if (!spec->autocfg.line_outs) {
- if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
- spec->multiout.max_channels = 2;
- spec->no_analog = 1;
- goto dig_only;
- }
- return 0; /* can't find valid BIOS pin config */
- }
- err = alc680_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = 2;
-
- dig_only:
- /* digital only support output */
- alc_auto_parse_digital(codec);
- if (spec->kctls.list)
- add_mixer(spec, spec->kctls.list);
-
- add_verb(spec, alc680_init_verbs);
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- return 1;
-}
-
-#define alc680_auto_init_analog_input alc882_auto_init_analog_input
-
-/* init callback for auto-configuration model -- overriding the default init */
-static void alc680_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc680_auto_init_multi_out(codec);
- alc680_auto_init_hp_out(codec);
- alc680_auto_init_analog_input(codec);
- alc_auto_init_digital(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
+ return alc_parse_auto_config(codec, NULL, NULL);
}
/*
- * configuration and preset
*/
-static const char * const alc680_models[ALC680_MODEL_LAST] = {
- [ALC680_BASE] = "base",
- [ALC680_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc680_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
- {}
-};
-
-static const struct alc_config_preset alc680_presets[] = {
- [ALC680_BASE] = {
- .mixers = { alc680_base_mixer },
- .cap_mixer = alc680_master_capture_mixer,
- .init_verbs = { alc680_init_verbs },
- .num_dacs = ARRAY_SIZE(alc680_dac_nids),
- .dac_nids = alc680_dac_nids,
- .dig_out_nid = ALC680_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc680_modes),
- .channel_mode = alc680_modes,
- .unsol_event = alc680_unsol_event,
- .setup = alc680_base_setup,
- .init_hook = alc680_inithook,
-
- },
-};
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc680_quirks.c"
+#endif
static int patch_alc680(struct hda_codec *codec)
{
@@ -20048,51 +5298,55 @@
codec->spec = spec;
- board_config = snd_hda_check_board_config(codec, ALC680_MODEL_LAST,
- alc680_models,
- alc680_cfg_tbl);
+ /* ALC680 has no aa-loopback mixer */
- if (board_config < 0 || board_config >= ALC680_MODEL_LAST) {
+ board_config = alc_board_config(codec, ALC680_MODEL_LAST,
+ alc680_models, alc680_cfg_tbl);
+
+ if (board_config < 0) {
printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
- board_config = ALC680_AUTO;
+ board_config = ALC_MODEL_AUTO;
}
- if (board_config == ALC680_AUTO) {
+ if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc680_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
- } else if (!err) {
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
board_config = ALC680_BASE;
}
+#endif
}
- if (board_config != ALC680_AUTO)
+ if (board_config != ALC_MODEL_AUTO) {
setup_preset(codec, &alc680_presets[board_config]);
-
- spec->stream_analog_playback = &alc680_pcm_analog_playback;
- spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
- spec->stream_digital_playback = &alc680_pcm_digital_playback;
- spec->stream_digital_capture = &alc680_pcm_digital_capture;
-
- if (!spec->adc_nids) {
- spec->adc_nids = alc680_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc680_adc_nids);
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
+#endif
}
- if (!spec->cap_mixer)
+ if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
+ }
+
+ if (!spec->no_analog && !spec->cap_mixer)
set_capture_mixer(codec);
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC680_AUTO)
- spec->init_hook = alc680_auto_init;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
return 0;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7f81cc2..56425a5 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1112,7 +1112,9 @@
}
if (spec->multiout.dig_out_nid) {
- err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid,
+ spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@@ -3406,30 +3408,9 @@
return 0;
}
-static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
- hda_nid_t nid)
-{
- hda_nid_t conn[HDA_MAX_NUM_INPUTS];
- int i, nums;
-
- if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST))
- return -1;
-
- nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
- for (i = 0; i < nums; i++)
- if (conn[i] == nid)
- return i;
-
- for (i = 0; i < nums; i++) {
- unsigned int wid_caps = get_wcaps(codec, conn[i]);
- unsigned int wid_type = get_wcaps_type(wid_caps);
-
- if (wid_type != AC_WID_PIN && wid_type != AC_WID_AUD_MIX)
- if (get_connection_index(codec, conn[i], nid) >= 0)
- return i;
- }
- return -1;
-}
+/* look for NID recursively */
+#define get_connection_index(codec, mux, nid) \
+ snd_hda_get_conn_index(codec, mux, nid, 1)
/* create a volume assigned to the given pin (only if supported) */
/* return 1 if the volume control is created */
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index f43bb0e..f38160b 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -54,36 +54,10 @@
#include "hda_codec.h"
#include "hda_local.h"
-#define NID_MAPPING (-1)
-
-/* amp values */
-#define AMP_VAL_IDX_SHIFT 19
-#define AMP_VAL_IDX_MASK (0x0f<<19)
-
/* Pin Widget NID */
-#define VT1708_HP_NID 0x13
-#define VT1708_DIGOUT_NID 0x14
-#define VT1708_DIGIN_NID 0x16
-#define VT1708_DIGIN_PIN 0x26
#define VT1708_HP_PIN_NID 0x20
#define VT1708_CD_PIN_NID 0x24
-#define VT1709_HP_DAC_NID 0x28
-#define VT1709_DIGOUT_NID 0x13
-#define VT1709_DIGIN_NID 0x17
-#define VT1709_DIGIN_PIN 0x25
-
-#define VT1708B_HP_NID 0x25
-#define VT1708B_DIGOUT_NID 0x12
-#define VT1708B_DIGIN_NID 0x15
-#define VT1708B_DIGIN_PIN 0x21
-
-#define VT1708S_HP_NID 0x25
-#define VT1708S_DIGOUT_NID 0x12
-
-#define VT1702_HP_NID 0x17
-#define VT1702_DIGOUT_NID 0x11
-
enum VIA_HDA_CODEC {
UNKNOWN = -1,
VT1708,
@@ -107,6 +81,39 @@
(spec)->codec_type == VT1812 ||\
(spec)->codec_type == VT1802)
+#define MAX_NID_PATH_DEPTH 5
+
+/* output-path: DAC -> ... -> pin
+ * idx[] contains the source index number of the next widget;
+ * e.g. idx[0] is the index of the DAC selected by path[1] widget
+ * multi[] indicates whether it's a selector widget with multi-connectors
+ * (i.e. the connection selection is mandatory)
+ * vol_ctl and mute_ctl contains the NIDs for the assigned mixers
+ */
+struct nid_path {
+ int depth;
+ hda_nid_t path[MAX_NID_PATH_DEPTH];
+ unsigned char idx[MAX_NID_PATH_DEPTH];
+ unsigned char multi[MAX_NID_PATH_DEPTH];
+ unsigned int vol_ctl;
+ unsigned int mute_ctl;
+};
+
+/* input-path */
+struct via_input {
+ hda_nid_t pin; /* input-pin or aa-mix */
+ int adc_idx; /* ADC index to be used */
+ int mux_idx; /* MUX index (if any) */
+ const char *label; /* input-source label */
+};
+
+#define VIA_MAX_ADCS 3
+
+enum {
+ STREAM_MULTI_OUT = (1 << 0),
+ STREAM_INDEP_HP = (1 << 1),
+};
+
struct via_spec {
/* codec parameterization */
const struct snd_kcontrol_new *mixers[6];
@@ -115,28 +122,66 @@
const struct hda_verb *init_verbs[5];
unsigned int num_iverbs;
- char *stream_name_analog;
+ char stream_name_analog[32];
+ char stream_name_hp[32];
const struct hda_pcm_stream *stream_analog_playback;
const struct hda_pcm_stream *stream_analog_capture;
- char *stream_name_digital;
+ char stream_name_digital[32];
const struct hda_pcm_stream *stream_digital_playback;
const struct hda_pcm_stream *stream_digital_capture;
/* playback */
struct hda_multi_out multiout;
hda_nid_t slave_dig_outs[2];
+ hda_nid_t hp_dac_nid;
+ hda_nid_t speaker_dac_nid;
+ int hp_indep_shared; /* indep HP-DAC is shared with side ch */
+ int opened_streams; /* STREAM_* bits */
+ int active_streams; /* STREAM_* bits */
+ int aamix_mode; /* loopback is enabled for output-path? */
+
+ /* Output-paths:
+ * There are different output-paths depending on the setup.
+ * out_path, hp_path and speaker_path are primary paths. If both
+ * direct DAC and aa-loopback routes are available, these contain
+ * the former paths. Meanwhile *_mix_path contain the paths with
+ * loopback mixer. (Since the loopback is only for front channel,
+ * no out_mix_path for surround channels.)
+ * The HP output has another path, hp_indep_path, which is used in
+ * the independent-HP mode.
+ */
+ struct nid_path out_path[HDA_SIDE + 1];
+ struct nid_path out_mix_path;
+ struct nid_path hp_path;
+ struct nid_path hp_mix_path;
+ struct nid_path hp_indep_path;
+ struct nid_path speaker_path;
+ struct nid_path speaker_mix_path;
/* capture */
unsigned int num_adc_nids;
- const hda_nid_t *adc_nids;
- hda_nid_t mux_nids[3];
+ hda_nid_t adc_nids[VIA_MAX_ADCS];
+ hda_nid_t mux_nids[VIA_MAX_ADCS];
+ hda_nid_t aa_mix_nid;
hda_nid_t dig_in_nid;
- hda_nid_t dig_in_pin;
/* capture source */
- const struct hda_input_mux *input_mux;
- unsigned int cur_mux[3];
+ bool dyn_adc_switch;
+ int num_inputs;
+ struct via_input inputs[AUTO_CFG_MAX_INS + 1];
+ unsigned int cur_mux[VIA_MAX_ADCS];
+
+ /* dynamic DAC switching */
+ unsigned int cur_dac_stream_tag;
+ unsigned int cur_dac_format;
+ unsigned int cur_hp_stream_tag;
+ unsigned int cur_hp_format;
+
+ /* dynamic ADC switching */
+ hda_nid_t cur_adc;
+ unsigned int cur_adc_stream_tag;
+ unsigned int cur_adc_format;
/* PCM information */
struct hda_pcm pcm_rec[3];
@@ -144,28 +189,38 @@
/* dynamic controls, init_verbs and input_mux */
struct auto_pin_cfg autocfg;
struct snd_array kctls;
- struct hda_input_mux private_imux[2];
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
/* HP mode source */
- const struct hda_input_mux *hp_mux;
unsigned int hp_independent_mode;
- unsigned int hp_independent_mode_index;
- unsigned int smart51_enabled;
unsigned int dmic_enabled;
+ unsigned int no_pin_power_ctl;
enum VIA_HDA_CODEC codec_type;
+ /* smart51 setup */
+ unsigned int smart51_nums;
+ hda_nid_t smart51_pins[2];
+ int smart51_idxs[2];
+ const char *smart51_labels[2];
+ unsigned int smart51_enabled;
+
/* work to check hp jack state */
struct hda_codec *codec;
struct delayed_work vt1708_hp_work;
- int vt1708_jack_detectect;
+ int vt1708_jack_detect;
int vt1708_hp_present;
void (*set_widgets_power_state)(struct hda_codec *codec);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
-#endif
+ int num_loopbacks;
+ struct hda_amp_list loopback_list[8];
+
+ /* bind capture-volume */
+ struct hda_bind_ctls *bind_cap_vol;
+ struct hda_bind_ctls *bind_cap_sw;
+
+ struct mutex config_mutex;
};
static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec);
@@ -177,6 +232,7 @@
if (spec == NULL)
return NULL;
+ mutex_init(&spec->config_mutex);
codec->spec = spec;
spec->codec = codec;
spec->codec_type = get_codec_type(codec);
@@ -237,33 +293,23 @@
#define VIA_JACK_EVENT 0x20
#define VIA_HP_EVENT 0x01
#define VIA_GPIO_EVENT 0x02
-#define VIA_MONO_EVENT 0x03
-#define VIA_SPEAKER_EVENT 0x04
-#define VIA_BIND_HP_EVENT 0x05
+#define VIA_LINE_EVENT 0x03
enum {
VIA_CTL_WIDGET_VOL,
VIA_CTL_WIDGET_MUTE,
VIA_CTL_WIDGET_ANALOG_MUTE,
- VIA_CTL_WIDGET_BIND_PIN_MUTE,
};
-enum {
- AUTO_SEQ_FRONT = 0,
- AUTO_SEQ_SURROUND,
- AUTO_SEQ_CENLFE,
- AUTO_SEQ_SIDE
-};
-
-static void analog_low_current_mode(struct hda_codec *codec, int stream_idle);
-static int is_aa_path_mute(struct hda_codec *codec);
+static void analog_low_current_mode(struct hda_codec *codec);
+static bool is_aa_path_mute(struct hda_codec *codec);
static void vt1708_start_hp_work(struct via_spec *spec)
{
if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
return;
snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
- !spec->vt1708_jack_detectect);
+ !spec->vt1708_jack_detect);
if (!delayed_work_pending(&spec->vt1708_hp_work))
schedule_delayed_work(&spec->vt1708_hp_work,
msecs_to_jiffies(100));
@@ -277,7 +323,7 @@
&& !is_aa_path_mute(spec->codec))
return;
snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
- !spec->vt1708_jack_detectect);
+ !spec->vt1708_jack_detect);
cancel_delayed_work_sync(&spec->vt1708_hp_work);
}
@@ -295,7 +341,7 @@
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
set_widgets_power_state(codec);
- analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1);
+ analog_low_current_mode(snd_kcontrol_chip(kcontrol));
if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) {
if (is_aa_path_mute(codec))
vt1708_start_hp_work(codec->spec);
@@ -315,168 +361,44 @@
.put = analog_input_switch_put, \
.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) }
-static void via_hp_bind_automute(struct hda_codec *codec);
-
-static int bind_pin_switch_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct via_spec *spec = codec->spec;
- int i;
- int change = 0;
-
- long *valp = ucontrol->value.integer.value;
- int lmute, rmute;
- if (strstr(kcontrol->id.name, "Switch") == NULL) {
- snd_printd("Invalid control!\n");
- return change;
- }
- change = snd_hda_mixer_amp_switch_put(kcontrol,
- ucontrol);
- /* Get mute value */
- lmute = *valp ? 0 : HDA_AMP_MUTE;
- valp++;
- rmute = *valp ? 0 : HDA_AMP_MUTE;
-
- /* Set hp pins */
- if (!spec->hp_independent_mode) {
- for (i = 0; i < spec->autocfg.hp_outs; i++) {
- snd_hda_codec_amp_update(
- codec, spec->autocfg.hp_pins[i],
- 0, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- lmute);
- snd_hda_codec_amp_update(
- codec, spec->autocfg.hp_pins[i],
- 1, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- rmute);
- }
- }
-
- if (!lmute && !rmute) {
- /* Line Outs */
- for (i = 0; i < spec->autocfg.line_outs; i++)
- snd_hda_codec_amp_stereo(
- codec, spec->autocfg.line_out_pins[i],
- HDA_OUTPUT, 0, HDA_AMP_MUTE, 0);
- /* Speakers */
- for (i = 0; i < spec->autocfg.speaker_outs; i++)
- snd_hda_codec_amp_stereo(
- codec, spec->autocfg.speaker_pins[i],
- HDA_OUTPUT, 0, HDA_AMP_MUTE, 0);
- /* unmute */
- via_hp_bind_automute(codec);
-
- } else {
- if (lmute) {
- /* Mute all left channels */
- for (i = 1; i < spec->autocfg.line_outs; i++)
- snd_hda_codec_amp_update(
- codec,
- spec->autocfg.line_out_pins[i],
- 0, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- lmute);
- for (i = 0; i < spec->autocfg.speaker_outs; i++)
- snd_hda_codec_amp_update(
- codec,
- spec->autocfg.speaker_pins[i],
- 0, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- lmute);
- }
- if (rmute) {
- /* mute all right channels */
- for (i = 1; i < spec->autocfg.line_outs; i++)
- snd_hda_codec_amp_update(
- codec,
- spec->autocfg.line_out_pins[i],
- 1, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- rmute);
- for (i = 0; i < spec->autocfg.speaker_outs; i++)
- snd_hda_codec_amp_update(
- codec,
- spec->autocfg.speaker_pins[i],
- 1, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- rmute);
- }
- }
- return change;
-}
-
-#define BIND_PIN_MUTE \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = NULL, \
- .index = 0, \
- .info = snd_hda_mixer_amp_switch_info, \
- .get = snd_hda_mixer_amp_switch_get, \
- .put = bind_pin_switch_put, \
- .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) }
-
static const struct snd_kcontrol_new via_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
ANALOG_INPUT_MUTE,
- BIND_PIN_MUTE,
-};
-
-static const hda_nid_t vt1708_adc_nids[2] = {
- /* ADC1-2 */
- 0x15, 0x27
-};
-
-static const hda_nid_t vt1709_adc_nids[3] = {
- /* ADC1-2 */
- 0x14, 0x15, 0x16
-};
-
-static const hda_nid_t vt1708B_adc_nids[2] = {
- /* ADC1-2 */
- 0x13, 0x14
-};
-
-static const hda_nid_t vt1708S_adc_nids[2] = {
- /* ADC1-2 */
- 0x13, 0x14
-};
-
-static const hda_nid_t vt1702_adc_nids[3] = {
- /* ADC1-2 */
- 0x12, 0x20, 0x1F
-};
-
-static const hda_nid_t vt1718S_adc_nids[2] = {
- /* ADC1-2 */
- 0x10, 0x11
-};
-
-static const hda_nid_t vt1716S_adc_nids[2] = {
- /* ADC1-2 */
- 0x13, 0x14
-};
-
-static const hda_nid_t vt2002P_adc_nids[2] = {
- /* ADC1-2 */
- 0x10, 0x11
-};
-
-static const hda_nid_t vt1812_adc_nids[2] = {
- /* ADC1-2 */
- 0x10, 0x11
};
/* add dynamic controls */
-static int __via_add_control(struct via_spec *spec, int type, const char *name,
- int idx, unsigned long val)
+static struct snd_kcontrol_new *__via_clone_ctl(struct via_spec *spec,
+ const struct snd_kcontrol_new *tmpl,
+ const char *name)
{
struct snd_kcontrol_new *knew;
snd_array_init(&spec->kctls, sizeof(*knew), 32);
knew = snd_array_new(&spec->kctls);
if (!knew)
+ return NULL;
+ *knew = *tmpl;
+ if (!name)
+ name = tmpl->name;
+ if (name) {
+ knew->name = kstrdup(name, GFP_KERNEL);
+ if (!knew->name)
+ return NULL;
+ }
+ return knew;
+}
+
+static int __via_add_control(struct via_spec *spec, int type, const char *name,
+ int idx, unsigned long val)
+{
+ struct snd_kcontrol_new *knew;
+
+ knew = __via_clone_ctl(spec, &via_control_templates[type], name);
+ if (!knew)
return -ENOMEM;
- *knew = via_control_templates[type];
- knew->name = kstrdup(name, GFP_KERNEL);
- if (!knew->name)
- return -ENOMEM;
+ knew->index = idx;
if (get_amp_nid_(val))
knew->subdevice = HDA_SUBDEV_AMP_FLAG;
knew->private_value = val;
@@ -486,21 +408,7 @@
#define via_add_control(spec, type, name, val) \
__via_add_control(spec, type, name, 0, val)
-static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec,
- const struct snd_kcontrol_new *tmpl)
-{
- struct snd_kcontrol_new *knew;
-
- snd_array_init(&spec->kctls, sizeof(*knew), 32);
- knew = snd_array_new(&spec->kctls);
- if (!knew)
- return NULL;
- *knew = *tmpl;
- knew->name = kstrdup(tmpl->name, GFP_KERNEL);
- if (!knew->name)
- return NULL;
- return knew;
-}
+#define via_clone_control(spec, tmpl) __via_clone_ctl(spec, tmpl, NULL)
static void via_free_kctls(struct hda_codec *codec)
{
@@ -535,58 +443,208 @@
return 0;
}
-static void via_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type,
- int dac_idx)
+#define get_connection_index(codec, mux, nid) \
+ snd_hda_get_conn_index(codec, mux, nid, 0)
+
+static bool check_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
+ unsigned int mask)
{
- /* set as output */
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ unsigned int caps;
+ if (!nid)
+ return false;
+ caps = get_wcaps(codec, nid);
+ if (dir == HDA_INPUT)
+ caps &= AC_WCAP_IN_AMP;
+ else
+ caps &= AC_WCAP_OUT_AMP;
+ if (!caps)
+ return false;
+ if (query_amp_caps(codec, nid, dir) & mask)
+ return true;
+ return false;
+}
+
+#define have_mute(codec, nid, dir) \
+ check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE)
+
+/* enable/disable the output-route mixers */
+static void activate_output_mix(struct hda_codec *codec, struct nid_path *path,
+ hda_nid_t mix_nid, int idx, bool enable)
+{
+ int i, num, val;
+
+ if (!path)
+ return;
+ num = snd_hda_get_conn_list(codec, mix_nid, NULL);
+ for (i = 0; i < num; i++) {
+ if (i == idx)
+ val = AMP_IN_UNMUTE(i);
+ else
+ val = AMP_IN_MUTE(i);
+ snd_hda_codec_write(codec, mix_nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, val);
+ }
+}
+
+/* enable/disable the output-route */
+static void activate_output_path(struct hda_codec *codec, struct nid_path *path,
+ bool enable, bool force)
+{
+ struct via_spec *spec = codec->spec;
+ int i;
+ for (i = 0; i < path->depth; i++) {
+ hda_nid_t src, dst;
+ int idx = path->idx[i];
+ src = path->path[i];
+ if (i < path->depth - 1)
+ dst = path->path[i + 1];
+ else
+ dst = 0;
+ if (enable && path->multi[i])
+ snd_hda_codec_write(codec, dst, 0,
+ AC_VERB_SET_CONNECT_SEL, idx);
+ if (!force && (dst == spec->aa_mix_nid))
+ continue;
+ if (have_mute(codec, dst, HDA_INPUT))
+ activate_output_mix(codec, path, dst, idx, enable);
+ if (!force && (src == path->vol_ctl || src == path->mute_ctl))
+ continue;
+ if (have_mute(codec, src, HDA_OUTPUT)) {
+ int val = enable ? AMP_OUT_UNMUTE : AMP_OUT_MUTE;
+ snd_hda_codec_write(codec, src, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, val);
+ }
+ }
+}
+
+/* set the given pin as output */
+static void init_output_pin(struct hda_codec *codec, hda_nid_t pin,
+ int pin_type)
+{
+ if (!pin)
+ return;
+ snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_type);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_UNMUTE);
- if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
- snd_hda_codec_write(codec, nid, 0,
+ if (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD)
+ snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x02);
}
+static void via_auto_init_output(struct hda_codec *codec,
+ struct nid_path *path, int pin_type)
+{
+ unsigned int caps;
+ hda_nid_t pin;
+
+ if (!path->depth)
+ return;
+ pin = path->path[path->depth - 1];
+
+ init_output_pin(codec, pin, pin_type);
+ caps = query_amp_caps(codec, pin, HDA_OUTPUT);
+ if (caps & AC_AMPCAP_MUTE) {
+ unsigned int val;
+ val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
+ snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE | val);
+ }
+ activate_output_path(codec, path, true, true); /* force on */
+}
static void via_auto_init_multi_out(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
+ struct nid_path *path;
int i;
- for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
- hda_nid_t nid = spec->autocfg.line_out_pins[i];
- if (nid)
- via_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
+ for (i = 0; i < spec->autocfg.line_outs + spec->smart51_nums; i++) {
+ path = &spec->out_path[i];
+ if (!i && spec->aamix_mode && spec->out_mix_path.depth)
+ path = &spec->out_mix_path;
+ via_auto_init_output(codec, path, PIN_OUT);
+ }
+}
+
+/* deactivate the inactive headphone-paths */
+static void deactivate_hp_paths(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int shared = spec->hp_indep_shared;
+
+ if (spec->hp_independent_mode) {
+ activate_output_path(codec, &spec->hp_path, false, false);
+ activate_output_path(codec, &spec->hp_mix_path, false, false);
+ if (shared)
+ activate_output_path(codec, &spec->out_path[shared],
+ false, false);
+ } else if (spec->aamix_mode || !spec->hp_path.depth) {
+ activate_output_path(codec, &spec->hp_indep_path, false, false);
+ activate_output_path(codec, &spec->hp_path, false, false);
+ } else {
+ activate_output_path(codec, &spec->hp_indep_path, false, false);
+ activate_output_path(codec, &spec->hp_mix_path, false, false);
}
}
static void via_auto_init_hp_out(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
- hda_nid_t pin;
- int i;
- for (i = 0; i < spec->autocfg.hp_outs; i++) {
- pin = spec->autocfg.hp_pins[i];
- if (pin) /* connect to front */
- via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ if (!spec->hp_path.depth) {
+ via_auto_init_output(codec, &spec->hp_mix_path, PIN_HP);
+ return;
+ }
+ deactivate_hp_paths(codec);
+ if (spec->hp_independent_mode)
+ via_auto_init_output(codec, &spec->hp_indep_path, PIN_HP);
+ else if (spec->aamix_mode)
+ via_auto_init_output(codec, &spec->hp_mix_path, PIN_HP);
+ else
+ via_auto_init_output(codec, &spec->hp_path, PIN_HP);
+}
+
+static void via_auto_init_speaker_out(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+
+ if (!spec->autocfg.speaker_outs)
+ return;
+ if (!spec->speaker_path.depth) {
+ via_auto_init_output(codec, &spec->speaker_mix_path, PIN_OUT);
+ return;
+ }
+ if (!spec->aamix_mode) {
+ activate_output_path(codec, &spec->speaker_mix_path,
+ false, false);
+ via_auto_init_output(codec, &spec->speaker_path, PIN_OUT);
+ } else {
+ activate_output_path(codec, &spec->speaker_path, false, false);
+ via_auto_init_output(codec, &spec->speaker_mix_path, PIN_OUT);
}
}
-static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin);
+static bool is_smart51_pins(struct hda_codec *codec, hda_nid_t pin);
+static void via_hp_automute(struct hda_codec *codec);
static void via_auto_init_analog_input(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
const struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
unsigned int ctl;
- int i;
+ int i, num_conns;
+ /* init ADCs */
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ snd_hda_codec_write(codec, spec->adc_nids[i], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+ }
+
+ /* init pins */
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
- if (spec->smart51_enabled && is_smart51_pins(spec, nid))
+ if (spec->smart51_enabled && is_smart51_pins(codec, nid))
ctl = PIN_OUT;
else if (cfg->inputs[i].type == AUTO_PIN_MIC)
ctl = PIN_VREF50;
@@ -595,6 +653,32 @@
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
}
+
+ /* init input-src */
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx;
+ if (spec->mux_nids[adc_idx]) {
+ int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx;
+ snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ mux_idx);
+ }
+ if (spec->dyn_adc_switch)
+ break; /* only one input-src */
+ }
+
+ /* init aa-mixer */
+ if (!spec->aa_mix_nid)
+ return;
+ num_conns = snd_hda_get_connections(codec, spec->aa_mix_nid, conn,
+ ARRAY_SIZE(conn));
+ for (i = 0; i < num_conns; i++) {
+ unsigned int caps = get_wcaps(codec, conn[i]);
+ if (get_wcaps_type(caps) == AC_WID_PIN)
+ snd_hda_codec_write(codec, spec->aa_mix_nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(i));
+ }
}
static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
@@ -605,9 +689,13 @@
unsigned no_presence = (def_conf & AC_DEFCFG_MISC)
>> AC_DEFCFG_MISC_SHIFT
& AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */
- unsigned present = snd_hda_jack_detect(codec, nid);
struct via_spec *spec = codec->spec;
- if ((spec->smart51_enabled && is_smart51_pins(spec, nid))
+ unsigned present = 0;
+
+ no_presence |= spec->no_pin_power_ctl;
+ if (!no_presence)
+ present = snd_hda_jack_detect(codec, nid);
+ if ((spec->smart51_enabled && is_smart51_pins(codec, nid))
|| ((no_presence || present)
&& get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) {
*affected_parm = AC_PWRST_D0; /* if it's connected */
@@ -618,191 +706,186 @@
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
}
-/*
- * input MUX handling
- */
-static int via_mux_enum_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
+static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct via_spec *spec = codec->spec;
- return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
+ static const char * const texts[] = {
+ "Disabled", "Enabled"
+ };
-static int via_mux_enum_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct via_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
- ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
return 0;
}
-static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- int ret;
-
- if (!spec->mux_nids[adc_idx])
- return -EINVAL;
- /* switch to D0 beofre change index */
- if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0,
- AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0)
- snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
-
- ret = snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- spec->mux_nids[adc_idx],
- &spec->cur_mux[adc_idx]);
- /* update jack power state */
- set_widgets_power_state(codec);
-
- return ret;
+ ucontrol->value.enumerated.item[0] = !spec->no_pin_power_ctl;
+ return 0;
}
+static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ unsigned int val = !ucontrol->value.enumerated.item[0];
+
+ if (val == spec->no_pin_power_ctl)
+ return 0;
+ spec->no_pin_power_ctl = val;
+ set_widgets_power_state(codec);
+ return 1;
+}
+
+static const struct snd_kcontrol_new via_pin_power_ctl_enum = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Dynamic Power-Control",
+ .info = via_pin_power_ctl_info,
+ .get = via_pin_power_ctl_get,
+ .put = via_pin_power_ctl_put,
+};
+
+
static int via_independent_hp_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct via_spec *spec = codec->spec;
- return snd_hda_input_mux_info(spec->hp_mux, uinfo);
+ static const char * const texts[] = { "OFF", "ON" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= 2)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
}
static int via_independent_hp_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value;
- unsigned int pinsel;
-
- /* use !! to translate conn sel 2 for VT1718S */
- pinsel = !!snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_CONNECT_SEL,
- 0x00);
- ucontrol->value.enumerated.item[0] = pinsel;
-
- return 0;
-}
-
-static void activate_ctl(struct hda_codec *codec, const char *name, int active)
-{
- struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name);
- if (ctl) {
- ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
- ctl->vd[0].access |= active
- ? 0 : SNDRV_CTL_ELEM_ACCESS_INACTIVE;
- snd_ctl_notify(codec->bus->card,
- SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id);
- }
-}
-
-static hda_nid_t side_mute_channel(struct via_spec *spec)
-{
- switch (spec->codec_type) {
- case VT1708: return 0x1b;
- case VT1709_10CH: return 0x29;
- case VT1708B_8CH: /* fall thru */
- case VT1708S: return 0x27;
- case VT2002P: return 0x19;
- case VT1802: return 0x15;
- case VT1812: return 0x15;
- default: return 0;
- }
-}
-
-static int update_side_mute_status(struct hda_codec *codec)
-{
- /* mute side channel */
struct via_spec *spec = codec->spec;
- unsigned int parm;
- hda_nid_t sw3 = side_mute_channel(spec);
- if (sw3) {
- if (VT2002P_COMPATIBLE(spec))
- parm = spec->hp_independent_mode ?
- AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
- else
- parm = spec->hp_independent_mode ?
- AMP_OUT_MUTE : AMP_OUT_UNMUTE;
- snd_hda_codec_write(codec, sw3, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, parm);
- if (spec->codec_type == VT1812)
- snd_hda_codec_write(codec, 0x1d, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, parm);
- }
+ ucontrol->value.enumerated.item[0] = spec->hp_independent_mode;
return 0;
}
+/* adjust spec->multiout setup according to the current flags */
+static void setup_playback_multi_pcm(struct via_spec *spec)
+{
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ spec->multiout.num_dacs = cfg->line_outs + spec->smart51_nums;
+ spec->multiout.hp_nid = 0;
+ if (!spec->hp_independent_mode) {
+ if (!spec->hp_indep_shared)
+ spec->multiout.hp_nid = spec->hp_dac_nid;
+ } else {
+ if (spec->hp_indep_shared)
+ spec->multiout.num_dacs = cfg->line_outs - 1;
+ }
+}
+
+/* update DAC setups according to indep-HP switch;
+ * this function is called only when indep-HP is modified
+ */
+static void switch_indep_hp_dacs(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int shared = spec->hp_indep_shared;
+ hda_nid_t shared_dac, hp_dac;
+
+ if (!spec->opened_streams)
+ return;
+
+ shared_dac = shared ? spec->multiout.dac_nids[shared] : 0;
+ hp_dac = spec->hp_dac_nid;
+ if (spec->hp_independent_mode) {
+ /* switch to indep-HP mode */
+ if (spec->active_streams & STREAM_MULTI_OUT) {
+ __snd_hda_codec_cleanup_stream(codec, hp_dac, 1);
+ __snd_hda_codec_cleanup_stream(codec, shared_dac, 1);
+ }
+ if (spec->active_streams & STREAM_INDEP_HP)
+ snd_hda_codec_setup_stream(codec, hp_dac,
+ spec->cur_hp_stream_tag, 0,
+ spec->cur_hp_format);
+ } else {
+ /* back to HP or shared-DAC */
+ if (spec->active_streams & STREAM_INDEP_HP)
+ __snd_hda_codec_cleanup_stream(codec, hp_dac, 1);
+ if (spec->active_streams & STREAM_MULTI_OUT) {
+ hda_nid_t dac;
+ int ch;
+ if (shared_dac) { /* reset mutli-ch DAC */
+ dac = shared_dac;
+ ch = shared * 2;
+ } else { /* reset HP DAC */
+ dac = hp_dac;
+ ch = 0;
+ }
+ snd_hda_codec_setup_stream(codec, dac,
+ spec->cur_dac_stream_tag, ch,
+ spec->cur_dac_format);
+ }
+ }
+ setup_playback_multi_pcm(spec);
+}
+
static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
- hda_nid_t nid = kcontrol->private_value;
- unsigned int pinsel = ucontrol->value.enumerated.item[0];
- unsigned int parm0, parm1;
- /* Get Independent Mode index of headphone pin widget */
- spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
- ? 1 : 0;
- if (spec->codec_type == VT1718S) {
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
- /* Set correct mute switch for MW3 */
- parm0 = spec->hp_independent_mode ?
- AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0);
- parm1 = spec->hp_independent_mode ?
- AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
- snd_hda_codec_write(codec, 0x1b, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, parm0);
- snd_hda_codec_write(codec, 0x1b, 0,
- AC_VERB_SET_AMP_GAIN_MUTE, parm1);
- }
- else
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL, pinsel);
+ int cur, shared;
- if (spec->codec_type == VT1812)
- snd_hda_codec_write(codec, 0x35, 0,
- AC_VERB_SET_CONNECT_SEL, pinsel);
- if (spec->multiout.hp_nid && spec->multiout.hp_nid
- != spec->multiout.dac_nids[HDA_FRONT])
- snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid,
- 0, 0, 0);
-
- update_side_mute_status(codec);
- /* update HP volume/swtich active state */
- if (spec->codec_type == VT1708S
- || spec->codec_type == VT1702
- || spec->codec_type == VT1718S
- || spec->codec_type == VT1716S
- || VT2002P_COMPATIBLE(spec)) {
- activate_ctl(codec, "Headphone Playback Volume",
- spec->hp_independent_mode);
- activate_ctl(codec, "Headphone Playback Switch",
- spec->hp_independent_mode);
+ mutex_lock(&spec->config_mutex);
+ cur = !!ucontrol->value.enumerated.item[0];
+ if (spec->hp_independent_mode == cur) {
+ mutex_unlock(&spec->config_mutex);
+ return 0;
}
+ spec->hp_independent_mode = cur;
+ shared = spec->hp_indep_shared;
+ deactivate_hp_paths(codec);
+ if (cur)
+ activate_output_path(codec, &spec->hp_indep_path, true, false);
+ else {
+ if (shared)
+ activate_output_path(codec, &spec->out_path[shared],
+ true, false);
+ if (spec->aamix_mode || !spec->hp_path.depth)
+ activate_output_path(codec, &spec->hp_mix_path,
+ true, false);
+ else
+ activate_output_path(codec, &spec->hp_path,
+ true, false);
+ }
+
+ switch_indep_hp_dacs(codec);
+ mutex_unlock(&spec->config_mutex);
+
/* update jack power state */
set_widgets_power_state(codec);
- return 0;
+ via_hp_automute(codec);
+ return 1;
}
-static const struct snd_kcontrol_new via_hp_mixer[2] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Independent HP",
- .info = via_independent_hp_info,
- .get = via_independent_hp_get,
- .put = via_independent_hp_put,
- },
- {
- .iface = NID_MAPPING,
- .name = "Independent HP",
- },
+static const struct snd_kcontrol_new via_hp_mixer = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Independent HP",
+ .info = via_independent_hp_info,
+ .get = via_independent_hp_get,
+ .put = via_independent_hp_put,
};
static int via_hp_build(struct hda_codec *codec)
@@ -810,61 +893,28 @@
struct via_spec *spec = codec->spec;
struct snd_kcontrol_new *knew;
hda_nid_t nid;
- int nums;
- hda_nid_t conn[HDA_MAX_CONNECTIONS];
- switch (spec->codec_type) {
- case VT1718S:
- nid = 0x34;
- break;
- case VT2002P:
- case VT1802:
- nid = 0x35;
- break;
- case VT1812:
- nid = 0x3d;
- break;
- default:
- nid = spec->autocfg.hp_pins[0];
- break;
- }
-
- if (spec->codec_type != VT1708) {
- nums = snd_hda_get_connections(codec, nid,
- conn, HDA_MAX_CONNECTIONS);
- if (nums <= 1)
- return 0;
- }
-
- knew = via_clone_control(spec, &via_hp_mixer[0]);
+ nid = spec->autocfg.hp_pins[0];
+ knew = via_clone_control(spec, &via_hp_mixer);
if (knew == NULL)
return -ENOMEM;
knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
- knew->private_value = nid;
-
- nid = side_mute_channel(spec);
- if (nid) {
- knew = via_clone_control(spec, &via_hp_mixer[1]);
- if (knew == NULL)
- return -ENOMEM;
- knew->subdevice = nid;
- }
return 0;
}
static void notify_aa_path_ctls(struct hda_codec *codec)
{
+ struct via_spec *spec = codec->spec;
int i;
- struct snd_ctl_elem_id id;
- const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"};
- struct snd_kcontrol *ctl;
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- for (i = 0; i < ARRAY_SIZE(labels); i++) {
- sprintf(id.name, "%s Playback Volume", labels[i]);
+ for (i = 0; i < spec->smart51_nums; i++) {
+ struct snd_kcontrol *ctl;
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ sprintf(id.name, "%s Playback Volume", spec->smart51_labels[i]);
ctl = snd_hda_find_mixer_ctl(codec, id.name);
if (ctl)
snd_ctl_notify(codec->bus->card,
@@ -876,66 +926,28 @@
static void mute_aa_path(struct hda_codec *codec, int mute)
{
struct via_spec *spec = codec->spec;
- hda_nid_t nid_mixer;
- int start_idx;
- int end_idx;
+ int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE;
int i;
- /* get nid of MW0 and start & end index */
- switch (spec->codec_type) {
- case VT1708:
- nid_mixer = 0x17;
- start_idx = 2;
- end_idx = 4;
- break;
- case VT1709_10CH:
- case VT1709_6CH:
- nid_mixer = 0x18;
- start_idx = 2;
- end_idx = 4;
- break;
- case VT1708B_8CH:
- case VT1708B_4CH:
- case VT1708S:
- case VT1716S:
- nid_mixer = 0x16;
- start_idx = 2;
- end_idx = 4;
- break;
- case VT1718S:
- nid_mixer = 0x21;
- start_idx = 1;
- end_idx = 3;
- break;
- default:
- return;
- }
+
/* check AA path's mute status */
- for (i = start_idx; i <= end_idx; i++) {
- int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE;
- snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i,
+ for (i = 0; i < spec->smart51_nums; i++) {
+ if (spec->smart51_idxs[i] < 0)
+ continue;
+ snd_hda_codec_amp_stereo(codec, spec->aa_mix_nid,
+ HDA_INPUT, spec->smart51_idxs[i],
HDA_AMP_MUTE, val);
}
}
-static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin)
+
+static bool is_smart51_pins(struct hda_codec *codec, hda_nid_t pin)
{
- const struct auto_pin_cfg *cfg = &spec->autocfg;
+ struct via_spec *spec = codec->spec;
int i;
- for (i = 0; i < cfg->num_inputs; i++) {
- if (pin == cfg->inputs[i].pin)
- return cfg->inputs[i].type <= AUTO_PIN_LINE_IN;
- }
- return 0;
-}
-
-static int via_smart51_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
+ for (i = 0; i < spec->smart51_nums; i++)
+ if (spec->smart51_pins[i] == pin)
+ return true;
+ return false;
}
static int via_smart51_get(struct snd_kcontrol *kcontrol,
@@ -943,23 +955,8 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
- const struct auto_pin_cfg *cfg = &spec->autocfg;
- int on = 1;
- int i;
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
- int ctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- if (cfg->inputs[i].type > AUTO_PIN_LINE_IN)
- continue;
- if (cfg->inputs[i].type == AUTO_PIN_MIC &&
- spec->hp_independent_mode && spec->codec_type != VT1718S)
- continue; /* ignore FMic for independent HP */
- if ((ctl & AC_PINCTL_IN_EN) && !(ctl & AC_PINCTL_OUT_EN))
- on = 0;
- }
- *ucontrol->value.integer.value = on;
+ *ucontrol->value.integer.value = spec->smart51_enabled;
return 0;
}
@@ -968,21 +965,14 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
- const struct auto_pin_cfg *cfg = &spec->autocfg;
int out_in = *ucontrol->value.integer.value
? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN;
int i;
- for (i = 0; i < cfg->num_inputs; i++) {
- hda_nid_t nid = cfg->inputs[i].pin;
+ for (i = 0; i < spec->smart51_nums; i++) {
+ hda_nid_t nid = spec->smart51_pins[i];
unsigned int parm;
- if (cfg->inputs[i].type > AUTO_PIN_LINE_IN)
- continue;
- if (cfg->inputs[i].type == AUTO_PIN_MIC &&
- spec->hp_independent_mode && spec->codec_type != VT1718S)
- continue; /* don't retask FMic for independent HP */
-
parm = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
@@ -994,171 +984,59 @@
mute_aa_path(codec, 1);
notify_aa_path_ctls(codec);
}
- if (spec->codec_type == VT1718S) {
- snd_hda_codec_amp_stereo(
- codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- HDA_AMP_UNMUTE);
- }
- if (cfg->inputs[i].type == AUTO_PIN_MIC) {
- if (spec->codec_type == VT1708S
- || spec->codec_type == VT1716S) {
- /* input = index 1 (AOW3) */
- snd_hda_codec_write(
- codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL, 1);
- snd_hda_codec_amp_stereo(
- codec, nid, HDA_OUTPUT,
- 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE);
- }
- }
}
spec->smart51_enabled = *ucontrol->value.integer.value;
set_widgets_power_state(codec);
return 1;
}
-static const struct snd_kcontrol_new via_smart51_mixer[2] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Smart 5.1",
- .count = 1,
- .info = via_smart51_info,
- .get = via_smart51_get,
- .put = via_smart51_put,
- },
- {
- .iface = NID_MAPPING,
- .name = "Smart 5.1",
- }
+static const struct snd_kcontrol_new via_smart51_mixer = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Smart 5.1",
+ .count = 1,
+ .info = snd_ctl_boolean_mono_info,
+ .get = via_smart51_get,
+ .put = via_smart51_put,
};
-static int via_smart51_build(struct via_spec *spec)
+static int via_smart51_build(struct hda_codec *codec)
{
- struct snd_kcontrol_new *knew;
- const struct auto_pin_cfg *cfg = &spec->autocfg;
- hda_nid_t nid;
- int i;
+ struct via_spec *spec = codec->spec;
- if (!cfg)
+ if (!spec->smart51_nums)
return 0;
- if (cfg->line_outs > 2)
- return 0;
-
- knew = via_clone_control(spec, &via_smart51_mixer[0]);
- if (knew == NULL)
+ if (!via_clone_control(spec, &via_smart51_mixer))
return -ENOMEM;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- nid = cfg->inputs[i].pin;
- if (cfg->inputs[i].type <= AUTO_PIN_LINE_IN) {
- knew = via_clone_control(spec, &via_smart51_mixer[1]);
- if (knew == NULL)
- return -ENOMEM;
- knew->subdevice = nid;
- break;
- }
- }
-
return 0;
}
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1708_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x27, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x27, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
-/* check AA path's mute statue */
-static int is_aa_path_mute(struct hda_codec *codec)
+/* check AA path's mute status */
+static bool is_aa_path_mute(struct hda_codec *codec)
{
- int mute = 1;
- hda_nid_t nid_mixer;
- int start_idx;
- int end_idx;
- int i;
struct via_spec *spec = codec->spec;
- /* get nid of MW0 and start & end index */
- switch (spec->codec_type) {
- case VT1708B_8CH:
- case VT1708B_4CH:
- case VT1708S:
- case VT1716S:
- nid_mixer = 0x16;
- start_idx = 2;
- end_idx = 4;
- break;
- case VT1702:
- nid_mixer = 0x1a;
- start_idx = 1;
- end_idx = 3;
- break;
- case VT1718S:
- nid_mixer = 0x21;
- start_idx = 1;
- end_idx = 3;
- break;
- case VT2002P:
- case VT1812:
- case VT1802:
- nid_mixer = 0x21;
- start_idx = 0;
- end_idx = 2;
- break;
- default:
- return 0;
- }
- /* check AA path's mute status */
- for (i = start_idx; i <= end_idx; i++) {
- unsigned int con_list = snd_hda_codec_read(
- codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4);
- int shift = 8 * (i % 4);
- hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift;
- unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin);
- if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) {
- /* check mute status while the pin is connected */
- int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0,
- HDA_INPUT, i) >> 7;
- int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1,
- HDA_INPUT, i) >> 7;
- if (!mute_l || !mute_r) {
- mute = 0;
- break;
- }
+ const struct hda_amp_list *p;
+ int i, ch, v;
+
+ for (i = 0; i < spec->num_loopbacks; i++) {
+ p = &spec->loopback_list[i];
+ for (ch = 0; ch < 2; ch++) {
+ v = snd_hda_codec_amp_read(codec, p->nid, ch, p->dir,
+ p->idx);
+ if (!(v & HDA_AMP_MUTE) && v > 0)
+ return false;
}
}
- return mute;
+ return true;
}
/* enter/exit analog low-current mode */
-static void analog_low_current_mode(struct hda_codec *codec, int stream_idle)
+static void analog_low_current_mode(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
- static int saved_stream_idle = 1; /* saved stream idle status */
- int enable = is_aa_path_mute(codec);
- unsigned int verb = 0;
- unsigned int parm = 0;
+ bool enable;
+ unsigned int verb, parm;
- if (stream_idle == -1) /* stream status did not change */
- enable = enable && saved_stream_idle;
- else {
- enable = enable && stream_idle;
- saved_stream_idle = stream_idle;
- }
+ enable = is_aa_path_mute(codec) && (spec->opened_streams != 0);
/* decide low current mode's verb & parameter */
switch (spec->codec_type) {
@@ -1193,119 +1071,69 @@
/*
* generic initialization of ADC, input mixers and output mixers
*/
-static const struct hda_verb vt1708_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- /*
- * Set up output mixers (0x19 - 0x1b)
- */
- /* set vol=0 to output mixers */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Setup default input MW0 to PW4 */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0},
- /* PW9 Output enable */
- {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+static const struct hda_verb vt1708_init_verbs[] = {
/* power down jack detect function */
{0x1, 0xf81, 0x1},
{ }
};
-static int via_playback_pcm_open(struct hda_pcm_stream *hinfo,
+static void set_stream_open(struct hda_codec *codec, int bit, bool active)
+{
+ struct via_spec *spec = codec->spec;
+
+ if (active)
+ spec->opened_streams |= bit;
+ else
+ spec->opened_streams &= ~bit;
+ analog_low_current_mode(codec);
+}
+
+static int via_playback_multi_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
- int idle = substream->pstr->substream_opened == 1
- && substream->ref_count == 0;
- analog_low_current_mode(codec, idle);
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
- hinfo);
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ int err;
+
+ spec->multiout.num_dacs = cfg->line_outs + spec->smart51_nums;
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+ set_stream_open(codec, STREAM_MULTI_OUT, true);
+ err = snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+ if (err < 0) {
+ set_stream_open(codec, STREAM_MULTI_OUT, false);
+ return err;
+ }
+ return 0;
}
-static void playback_multi_pcm_prep_0(struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
+static int via_playback_multi_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ set_stream_open(codec, STREAM_MULTI_OUT, false);
+ return 0;
+}
+
+static int via_playback_hp_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
- struct hda_multi_out *mout = &spec->multiout;
- const hda_nid_t *nids = mout->dac_nids;
- int chs = substream->runtime->channels;
- int i;
- mutex_lock(&codec->spdif_mutex);
- if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
- if (chs == 2 &&
- snd_hda_is_supported_format(codec, mout->dig_out_nid,
- format) &&
- !(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
- mout->dig_out_used = HDA_DIG_ANALOG_DUP;
- /* turn off SPDIF once; otherwise the IEC958 bits won't
- * be updated */
- if (codec->spdif_ctls & AC_DIG1_ENABLE)
- snd_hda_codec_write(codec, mout->dig_out_nid, 0,
- AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls &
- ~AC_DIG1_ENABLE & 0xff);
- snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
- stream_tag, 0, format);
- /* turn on again (if needed) */
- if (codec->spdif_ctls & AC_DIG1_ENABLE)
- snd_hda_codec_write(codec, mout->dig_out_nid, 0,
- AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls & 0xff);
- } else {
- mout->dig_out_used = 0;
- snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
- 0, 0, 0);
- }
- }
- mutex_unlock(&codec->spdif_mutex);
+ if (snd_BUG_ON(!spec->hp_dac_nid))
+ return -EINVAL;
+ set_stream_open(codec, STREAM_INDEP_HP, true);
+ return 0;
+}
- /* front */
- snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag,
- 0, format);
-
- if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT]
- && !spec->hp_independent_mode)
- /* headphone out will just decode front left/right (stereo) */
- snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag,
- 0, format);
-
- /* extra outputs copied from front */
- for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
- if (mout->extra_out_nid[i])
- snd_hda_codec_setup_stream(codec,
- mout->extra_out_nid[i],
- stream_tag, 0, format);
-
- /* surrounds */
- for (i = 1; i < mout->num_dacs; i++) {
- if (chs >= (i + 1) * 2) /* independent out */
- snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
- i * 2, format);
- else /* copy front */
- snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
- 0, format);
- }
+static int via_playback_hp_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ set_stream_open(codec, STREAM_INDEP_HP, false);
+ return 0;
}
static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -1315,18 +1143,36 @@
struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
- struct hda_multi_out *mout = &spec->multiout;
- const hda_nid_t *nids = mout->dac_nids;
- if (substream->number == 0)
- playback_multi_pcm_prep_0(codec, stream_tag, format,
- substream);
- else {
- if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
- spec->hp_independent_mode)
- snd_hda_codec_setup_stream(codec, mout->hp_nid,
- stream_tag, 0, format);
- }
+ mutex_lock(&spec->config_mutex);
+ setup_playback_multi_pcm(spec);
+ snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+ /* remember for dynamic DAC switch with indep-HP */
+ spec->active_streams |= STREAM_MULTI_OUT;
+ spec->cur_dac_stream_tag = stream_tag;
+ spec->cur_dac_format = format;
+ mutex_unlock(&spec->config_mutex);
+ vt1708_start_hp_work(spec);
+ return 0;
+}
+
+static int via_playback_hp_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+
+ mutex_lock(&spec->config_mutex);
+ if (spec->hp_independent_mode)
+ snd_hda_codec_setup_stream(codec, spec->hp_dac_nid,
+ stream_tag, 0, format);
+ spec->active_streams |= STREAM_INDEP_HP;
+ spec->cur_hp_stream_tag = stream_tag;
+ spec->cur_hp_format = format;
+ mutex_unlock(&spec->config_mutex);
vt1708_start_hp_work(spec);
return 0;
}
@@ -1336,37 +1182,26 @@
struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
- struct hda_multi_out *mout = &spec->multiout;
- const hda_nid_t *nids = mout->dac_nids;
- int i;
- if (substream->number == 0) {
- for (i = 0; i < mout->num_dacs; i++)
- snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0);
+ mutex_lock(&spec->config_mutex);
+ snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+ spec->active_streams &= ~STREAM_MULTI_OUT;
+ mutex_unlock(&spec->config_mutex);
+ vt1708_stop_hp_work(spec);
+ return 0;
+}
- if (mout->hp_nid && !spec->hp_independent_mode)
- snd_hda_codec_setup_stream(codec, mout->hp_nid,
- 0, 0, 0);
+static int via_playback_hp_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
- for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
- if (mout->extra_out_nid[i])
- snd_hda_codec_setup_stream(codec,
- mout->extra_out_nid[i],
- 0, 0, 0);
- mutex_lock(&codec->spdif_mutex);
- if (mout->dig_out_nid &&
- mout->dig_out_used == HDA_DIG_ANALOG_DUP) {
- snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
- 0, 0, 0);
- mout->dig_out_used = 0;
- }
- mutex_unlock(&codec->spdif_mutex);
- } else {
- if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
- spec->hp_independent_mode)
- snd_hda_codec_setup_stream(codec, mout->hp_nid,
- 0, 0, 0);
- }
+ mutex_lock(&spec->config_mutex);
+ if (spec->hp_independent_mode)
+ snd_hda_codec_setup_stream(codec, spec->hp_dac_nid, 0, 0, 0);
+ spec->active_streams &= ~STREAM_INDEP_HP;
+ mutex_unlock(&spec->config_mutex);
vt1708_stop_hp_work(spec);
return 0;
}
@@ -1435,47 +1270,127 @@
return 0;
}
-static const struct hda_pcm_stream vt1708_pcm_analog_playback = {
- .substreams = 2,
+/* analog capture with dynamic ADC switching */
+static int via_dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ int adc_idx = spec->inputs[spec->cur_mux[0]].adc_idx;
+
+ mutex_lock(&spec->config_mutex);
+ spec->cur_adc = spec->adc_nids[adc_idx];
+ spec->cur_adc_stream_tag = stream_tag;
+ spec->cur_adc_format = format;
+ snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+ mutex_unlock(&spec->config_mutex);
+ return 0;
+}
+
+static int via_dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+
+ mutex_lock(&spec->config_mutex);
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = 0;
+ mutex_unlock(&spec->config_mutex);
+ return 0;
+}
+
+/* re-setup the stream if running; called from input-src put */
+static bool via_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur)
+{
+ struct via_spec *spec = codec->spec;
+ int adc_idx = spec->inputs[cur].adc_idx;
+ hda_nid_t adc = spec->adc_nids[adc_idx];
+ bool ret = false;
+
+ mutex_lock(&spec->config_mutex);
+ if (spec->cur_adc && spec->cur_adc != adc) {
+ /* stream is running, let's swap the current ADC */
+ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
+ spec->cur_adc = adc;
+ snd_hda_codec_setup_stream(codec, adc,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+ ret = true;
+ }
+ mutex_unlock(&spec->config_mutex);
+ return ret;
+}
+
+static const struct hda_pcm_stream via_pcm_analog_playback = {
+ .substreams = 1,
.channels_min = 2,
.channels_max = 8,
- .nid = 0x10, /* NID to query formats and rates */
+ /* NID is set in via_build_pcms */
.ops = {
- .open = via_playback_pcm_open,
+ .open = via_playback_multi_pcm_open,
+ .close = via_playback_multi_pcm_close,
.prepare = via_playback_multi_pcm_prepare,
.cleanup = via_playback_multi_pcm_cleanup
},
};
+static const struct hda_pcm_stream via_pcm_hp_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_playback_hp_pcm_open,
+ .close = via_playback_hp_pcm_close,
+ .prepare = via_playback_hp_pcm_prepare,
+ .cleanup = via_playback_hp_pcm_cleanup
+ },
+};
+
static const struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
- .substreams = 2,
+ .substreams = 1,
.channels_min = 2,
.channels_max = 8,
- .nid = 0x10, /* NID to query formats and rates */
+ /* NID is set in via_build_pcms */
/* We got noisy outputs on the right channel on VT1708 when
* 24bit samples are used. Until any workaround is found,
* disable the 24bit format, so far.
*/
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.ops = {
- .open = via_playback_pcm_open,
+ .open = via_playback_multi_pcm_open,
+ .close = via_playback_multi_pcm_close,
.prepare = via_playback_multi_pcm_prepare,
.cleanup = via_playback_multi_pcm_cleanup
},
};
-static const struct hda_pcm_stream vt1708_pcm_analog_capture = {
- .substreams = 2,
+static const struct hda_pcm_stream via_pcm_analog_capture = {
+ .substreams = 1, /* will be changed in via_build_pcms() */
.channels_min = 2,
.channels_max = 2,
- .nid = 0x15, /* NID to query formats and rates */
+ /* NID is set in via_build_pcms */
.ops = {
.prepare = via_capture_pcm_prepare,
.cleanup = via_capture_pcm_cleanup
},
};
-static const struct hda_pcm_stream vt1708_pcm_digital_playback = {
+static const struct hda_pcm_stream via_pcm_dyn_adc_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .prepare = via_dyn_adc_capture_pcm_prepare,
+ .cleanup = via_dyn_adc_capture_pcm_cleanup,
+ },
+};
+
+static const struct hda_pcm_stream via_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
@@ -1488,19 +1403,47 @@
},
};
-static const struct hda_pcm_stream vt1708_pcm_digital_capture = {
+static const struct hda_pcm_stream via_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
};
+/*
+ * slave controls for virtual master
+ */
+static const char * const via_slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Speaker Playback Volume",
+ NULL,
+};
+
+static const char * const via_slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Speaker Playback Switch",
+ NULL,
+};
+
static int via_build_controls(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
struct snd_kcontrol *kctl;
- const struct snd_kcontrol_new *knew;
int err, i;
+ if (spec->set_widgets_power_state)
+ if (!via_clone_control(spec, &via_pin_power_ctl_enum))
+ return -ENOMEM;
+
for (i = 0; i < spec->num_mixers; i++) {
err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
if (err < 0)
@@ -1509,6 +1452,7 @@
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
@@ -1524,6 +1468,23 @@
return err;
}
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ unsigned int vmaster_tlv[4];
+ snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
+ HDA_OUTPUT, vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ vmaster_tlv, via_slave_vols);
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, via_slave_sws);
+ if (err < 0)
+ return err;
+ }
+
/* assign Capture Source enums to NID */
kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
for (i = 0; kctl && i < kctl->count; i++) {
@@ -1532,22 +1493,9 @@
return err;
}
- /* other nid->control mapping */
- for (i = 0; i < spec->num_mixers; i++) {
- for (knew = spec->mixers[i]; knew->name; knew++) {
- if (knew->iface != NID_MAPPING)
- continue;
- kctl = snd_hda_find_mixer_ctl(codec, knew->name);
- if (kctl == NULL)
- continue;
- err = snd_hda_add_nid(codec, kctl, 0,
- knew->subdevice);
- }
- }
-
/* init power states */
set_widgets_power_state(codec);
- analog_low_current_mode(codec, 1);
+ analog_low_current_mode(codec);
via_free_kctls(codec); /* no longer needed */
return 0;
@@ -1561,36 +1509,71 @@
codec->num_pcms = 1;
codec->pcm_info = info;
+ snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog),
+ "%s Analog", codec->chip_name);
info->name = spec->stream_name_analog;
+
+ if (!spec->stream_analog_playback)
+ spec->stream_analog_playback = &via_pcm_analog_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
- *(spec->stream_analog_playback);
+ *spec->stream_analog_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
spec->multiout.dac_nids[0];
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
-
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
spec->multiout.max_channels;
+ if (!spec->stream_analog_capture) {
+ if (spec->dyn_adc_switch)
+ spec->stream_analog_capture =
+ &via_pcm_dyn_adc_analog_capture;
+ else
+ spec->stream_analog_capture = &via_pcm_analog_capture;
+ }
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ *spec->stream_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+ if (!spec->dyn_adc_switch)
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
+ spec->num_adc_nids;
+
if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
codec->num_pcms++;
info++;
+ snprintf(spec->stream_name_digital,
+ sizeof(spec->stream_name_digital),
+ "%s Digital", codec->chip_name);
info->name = spec->stream_name_digital;
info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->multiout.dig_out_nid) {
+ if (!spec->stream_digital_playback)
+ spec->stream_digital_playback =
+ &via_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
- *(spec->stream_digital_playback);
+ *spec->stream_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
spec->multiout.dig_out_nid;
}
if (spec->dig_in_nid) {
+ if (!spec->stream_digital_capture)
+ spec->stream_digital_capture =
+ &via_pcm_digital_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE] =
- *(spec->stream_digital_capture);
+ *spec->stream_digital_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
spec->dig_in_nid;
}
}
+ if (spec->hp_dac_nid) {
+ codec->num_pcms++;
+ info++;
+ snprintf(spec->stream_name_hp, sizeof(spec->stream_name_hp),
+ "%s HP", codec->chip_name);
+ info->name = spec->stream_name_hp;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = via_pcm_hp_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->hp_dac_nid;
+ }
return 0;
}
@@ -1603,57 +1586,62 @@
via_free_kctls(codec);
vt1708_stop_hp_work(spec);
- kfree(codec->spec);
+ kfree(spec->bind_cap_vol);
+ kfree(spec->bind_cap_sw);
+ kfree(spec);
+}
+
+/* mute/unmute outputs */
+static void toggle_output_mutes(struct hda_codec *codec, int num_pins,
+ hda_nid_t *pins, bool mute)
+{
+ int i;
+ for (i = 0; i < num_pins; i++) {
+ unsigned int parm = snd_hda_codec_read(codec, pins[i], 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (parm & AC_PINCTL_IN_EN)
+ continue;
+ if (mute)
+ parm &= ~AC_PINCTL_OUT_EN;
+ else
+ parm |= AC_PINCTL_OUT_EN;
+ snd_hda_codec_write(codec, pins[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, parm);
+ }
+}
+
+/* mute internal speaker if line-out is plugged */
+static void via_line_automute(struct hda_codec *codec, int present)
+{
+ struct via_spec *spec = codec->spec;
+
+ if (!spec->autocfg.speaker_outs)
+ return;
+ if (!present)
+ present = snd_hda_jack_detect(codec,
+ spec->autocfg.line_out_pins[0]);
+ toggle_output_mutes(codec, spec->autocfg.speaker_outs,
+ spec->autocfg.speaker_pins,
+ present);
}
/* mute internal speaker if HP is plugged */
static void via_hp_automute(struct hda_codec *codec)
{
- unsigned int present = 0;
+ int present = 0;
+ int nums;
struct via_spec *spec = codec->spec;
- present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
+ if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0])
+ present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
- if (!spec->hp_independent_mode) {
- struct snd_ctl_elem_id id;
- /* auto mute */
- snd_hda_codec_amp_stereo(
- codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- /* notify change */
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, "Front Playback Switch");
- snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &id);
- }
-}
+ if (spec->smart51_enabled)
+ nums = spec->autocfg.line_outs + spec->smart51_nums;
+ else
+ nums = spec->autocfg.line_outs;
+ toggle_output_mutes(codec, nums, spec->autocfg.line_out_pins, present);
-/* mute mono out if HP or Line out is plugged */
-static void via_mono_automute(struct hda_codec *codec)
-{
- unsigned int hp_present, lineout_present;
- struct via_spec *spec = codec->spec;
-
- if (spec->codec_type != VT1716S)
- return;
-
- lineout_present = snd_hda_jack_detect(codec,
- spec->autocfg.line_out_pins[0]);
-
- /* Mute Mono Out if Line Out is plugged */
- if (lineout_present) {
- snd_hda_codec_amp_stereo(
- codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE);
- return;
- }
-
- hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
-
- if (!spec->hp_independent_mode)
- snd_hda_codec_amp_stereo(
- codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- hp_present ? HDA_AMP_MUTE : 0);
+ via_line_automute(codec, present);
}
static void via_gpio_control(struct hda_codec *codec)
@@ -1678,9 +1666,9 @@
if (gpio_data == 0x02) {
/* unmute line out */
- snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0],
- HDA_OUTPUT, 0, HDA_AMP_MUTE, 0);
-
+ snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_OUT);
if (vol_counter & 0x20) {
/* decrease volume */
if (vol > master_vol)
@@ -1697,73 +1685,12 @@
}
} else if (!(gpio_data & 0x02)) {
/* mute line out */
- snd_hda_codec_amp_stereo(codec,
- spec->autocfg.line_out_pins[0],
- HDA_OUTPUT, 0, HDA_AMP_MUTE,
- HDA_AMP_MUTE);
+ snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ 0);
}
}
-/* mute Internal-Speaker if HP is plugged */
-static void via_speaker_automute(struct hda_codec *codec)
-{
- unsigned int hp_present;
- struct via_spec *spec = codec->spec;
-
- if (!VT2002P_COMPATIBLE(spec))
- return;
-
- hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
-
- if (!spec->hp_independent_mode) {
- struct snd_ctl_elem_id id;
- snd_hda_codec_amp_stereo(
- codec, spec->autocfg.speaker_pins[0], HDA_OUTPUT, 0,
- HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0);
- /* notify change */
- memset(&id, 0, sizeof(id));
- id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strcpy(id.name, "Speaker Playback Switch");
- snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &id);
- }
-}
-
-/* mute line-out and internal speaker if HP is plugged */
-static void via_hp_bind_automute(struct hda_codec *codec)
-{
- /* use long instead of int below just to avoid an internal compiler
- * error with gcc 4.0.x
- */
- unsigned long hp_present, present = 0;
- struct via_spec *spec = codec->spec;
- int i;
-
- if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0])
- return;
-
- hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
-
- present = snd_hda_jack_detect(codec, spec->autocfg.line_out_pins[0]);
-
- if (!spec->hp_independent_mode) {
- /* Mute Line-Outs */
- for (i = 0; i < spec->autocfg.line_outs; i++)
- snd_hda_codec_amp_stereo(
- codec, spec->autocfg.line_out_pins[i],
- HDA_OUTPUT, 0,
- HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0);
- if (hp_present)
- present = hp_present;
- }
- /* Speakers */
- for (i = 0; i < spec->autocfg.speaker_outs; i++)
- snd_hda_codec_amp_stereo(
- codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-
/* unsolicited event for jack sensing */
static void via_unsol_event(struct hda_codec *codec,
unsigned int res)
@@ -1775,43 +1702,10 @@
res &= ~VIA_JACK_EVENT;
- if (res == VIA_HP_EVENT)
+ if (res == VIA_HP_EVENT || res == VIA_LINE_EVENT)
via_hp_automute(codec);
else if (res == VIA_GPIO_EVENT)
via_gpio_control(codec);
- else if (res == VIA_MONO_EVENT)
- via_mono_automute(codec);
- else if (res == VIA_SPEAKER_EVENT)
- via_speaker_automute(codec);
- else if (res == VIA_BIND_HP_EVENT)
- via_hp_bind_automute(codec);
-}
-
-static int via_init(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int i;
- for (i = 0; i < spec->num_iverbs; i++)
- snd_hda_sequence_write(codec, spec->init_verbs[i]);
-
- /* Lydia Add for EAPD enable */
- if (!spec->dig_in_nid) { /* No Digital In connection */
- if (spec->dig_in_pin) {
- snd_hda_codec_write(codec, spec->dig_in_pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
- snd_hda_codec_write(codec, spec->dig_in_pin, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 0x02);
- }
- } else /* enable SPDIF-input pin */
- snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
-
- /* assign slave outs */
- if (spec->slave_dig_outs[0])
- codec->slave_dig_outs = spec->slave_dig_outs;
-
- return 0;
}
#ifdef SND_HDA_NEEDS_RESUME
@@ -1833,11 +1727,15 @@
/*
*/
+
+static int via_init(struct hda_codec *codec);
+
static const struct hda_codec_ops via_patch_ops = {
.build_controls = via_build_controls,
.build_pcms = via_build_pcms,
.init = via_init,
.free = via_free,
+ .unsol_event = via_unsol_event,
#ifdef SND_HDA_NEEDS_RESUME
.suspend = via_suspend,
#endif
@@ -1846,237 +1744,835 @@
#endif
};
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt1708_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
+static bool is_empty_dac(struct hda_codec *codec, hda_nid_t dac)
{
+ struct via_spec *spec = codec->spec;
int i;
- hda_nid_t nid;
- spec->multiout.num_dacs = cfg->line_outs;
+ for (i = 0; i < spec->multiout.num_dacs; i++) {
+ if (spec->multiout.dac_nids[i] == dac)
+ return false;
+ }
+ if (spec->hp_dac_nid == dac)
+ return false;
+ return true;
+}
- spec->multiout.dac_nids = spec->private_dac_nids;
+static bool __parse_output_path(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t target_dac, int with_aa_mix,
+ struct nid_path *path, int depth)
+{
+ struct via_spec *spec = codec->spec;
+ hda_nid_t conn[8];
+ int i, nums;
- for (i = 0; i < 4; i++) {
- nid = cfg->line_out_pins[i];
- if (nid) {
- /* config dac list */
- switch (i) {
- case AUTO_SEQ_FRONT:
- spec->private_dac_nids[i] = 0x10;
- break;
- case AUTO_SEQ_CENLFE:
- spec->private_dac_nids[i] = 0x12;
- break;
- case AUTO_SEQ_SURROUND:
- spec->private_dac_nids[i] = 0x11;
- break;
- case AUTO_SEQ_SIDE:
- spec->private_dac_nids[i] = 0x13;
- break;
- }
- }
+ if (nid == spec->aa_mix_nid) {
+ if (!with_aa_mix)
+ return false;
+ with_aa_mix = 2; /* mark aa-mix is included */
}
+ nums = snd_hda_get_connections(codec, nid, conn, ARRAY_SIZE(conn));
+ for (i = 0; i < nums; i++) {
+ if (get_wcaps_type(get_wcaps(codec, conn[i])) != AC_WID_AUD_OUT)
+ continue;
+ if (conn[i] == target_dac || is_empty_dac(codec, conn[i])) {
+ /* aa-mix is requested but not included? */
+ if (!(spec->aa_mix_nid && with_aa_mix == 1))
+ goto found;
+ }
+ }
+ if (depth >= MAX_NID_PATH_DEPTH)
+ return false;
+ for (i = 0; i < nums; i++) {
+ unsigned int type;
+ type = get_wcaps_type(get_wcaps(codec, conn[i]));
+ if (type == AC_WID_AUD_OUT)
+ continue;
+ if (__parse_output_path(codec, conn[i], target_dac,
+ with_aa_mix, path, depth + 1))
+ goto found;
+ }
+ return false;
+
+ found:
+ path->path[path->depth] = conn[i];
+ path->idx[path->depth] = i;
+ if (nums > 1 && get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_AUD_MIX)
+ path->multi[path->depth] = 1;
+ path->depth++;
+ return true;
+}
+
+static bool parse_output_path(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t target_dac, int with_aa_mix,
+ struct nid_path *path)
+{
+ if (__parse_output_path(codec, nid, target_dac, with_aa_mix, path, 1)) {
+ path->path[path->depth] = nid;
+ path->depth++;
+ snd_printdd("output-path: depth=%d, %02x/%02x/%02x/%02x/%02x\n",
+ path->depth, path->path[0], path->path[1],
+ path->path[2], path->path[3], path->path[4]);
+ return true;
+ }
+ return false;
+}
+
+static int via_auto_fill_dac_nids(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, dac_num;
+ hda_nid_t nid;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ dac_num = 0;
+ for (i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t dac = 0;
+ nid = cfg->line_out_pins[i];
+ if (!nid)
+ continue;
+ if (parse_output_path(codec, nid, 0, 0, &spec->out_path[i]))
+ dac = spec->out_path[i].path[0];
+ if (!i && parse_output_path(codec, nid, dac, 1,
+ &spec->out_mix_path))
+ dac = spec->out_mix_path.path[0];
+ if (dac) {
+ spec->private_dac_nids[i] = dac;
+ dac_num++;
+ }
+ }
+ if (!spec->out_path[0].depth && spec->out_mix_path.depth) {
+ spec->out_path[0] = spec->out_mix_path;
+ spec->out_mix_path.depth = 0;
+ }
+ spec->multiout.num_dacs = dac_num;
return 0;
}
+static int create_ch_ctls(struct hda_codec *codec, const char *pfx,
+ int chs, bool check_dac, struct nid_path *path)
+{
+ struct via_spec *spec = codec->spec;
+ char name[32];
+ hda_nid_t dac, pin, sel, nid;
+ int err;
+
+ dac = check_dac ? path->path[0] : 0;
+ pin = path->path[path->depth - 1];
+ sel = path->depth > 1 ? path->path[1] : 0;
+
+ if (dac && check_amp_caps(codec, dac, HDA_OUTPUT, AC_AMPCAP_NUM_STEPS))
+ nid = dac;
+ else if (check_amp_caps(codec, pin, HDA_OUTPUT, AC_AMPCAP_NUM_STEPS))
+ nid = pin;
+ else if (check_amp_caps(codec, sel, HDA_OUTPUT, AC_AMPCAP_NUM_STEPS))
+ nid = sel;
+ else
+ nid = 0;
+ if (nid) {
+ sprintf(name, "%s Playback Volume", pfx);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ path->vol_ctl = nid;
+ }
+
+ if (dac && check_amp_caps(codec, dac, HDA_OUTPUT, AC_AMPCAP_MUTE))
+ nid = dac;
+ else if (check_amp_caps(codec, pin, HDA_OUTPUT, AC_AMPCAP_MUTE))
+ nid = pin;
+ else if (check_amp_caps(codec, sel, HDA_OUTPUT, AC_AMPCAP_MUTE))
+ nid = sel;
+ else
+ nid = 0;
+ if (nid) {
+ sprintf(name, "%s Playback Switch", pfx);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ path->mute_ctl = nid;
+ }
+ return 0;
+}
+
+static void mangle_smart51(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ struct auto_pin_cfg_item *ins = cfg->inputs;
+ int i, j, nums, attr;
+ int pins[AUTO_CFG_MAX_INS];
+
+ for (attr = INPUT_PIN_ATTR_REAR; attr >= INPUT_PIN_ATTR_NORMAL; attr--) {
+ nums = 0;
+ for (i = 0; i < cfg->num_inputs; i++) {
+ unsigned int def;
+ if (ins[i].type > AUTO_PIN_LINE_IN)
+ continue;
+ def = snd_hda_codec_get_pincfg(codec, ins[i].pin);
+ if (snd_hda_get_input_pin_attr(def) != attr)
+ continue;
+ for (j = 0; j < nums; j++)
+ if (ins[pins[j]].type < ins[i].type) {
+ memmove(pins + j + 1, pins + j,
+ (nums - j) * sizeof(int));
+ break;
+ }
+ pins[j] = i;
+ nums++;
+ }
+ if (cfg->line_outs + nums < 3)
+ continue;
+ for (i = 0; i < nums; i++) {
+ hda_nid_t pin = ins[pins[i]].pin;
+ spec->smart51_pins[spec->smart51_nums++] = pin;
+ cfg->line_out_pins[cfg->line_outs++] = pin;
+ if (cfg->line_outs == 3)
+ break;
+ }
+ return;
+ }
+}
+
+static void copy_path_mixer_ctls(struct nid_path *dst, struct nid_path *src)
+{
+ dst->vol_ctl = src->vol_ctl;
+ dst->mute_ctl = src->mute_ctl;
+}
+
/* add playback controls from the parsed DAC table */
-static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
+static int via_auto_create_multi_out_ctls(struct hda_codec *codec)
{
- char name[32];
+ struct via_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ struct nid_path *path;
static const char * const chname[4] = {
"Front", "Surround", "C/LFE", "Side"
};
- hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b};
- int i, err;
+ int i, idx, err;
+ int old_line_outs;
- for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
- nid = cfg->line_out_pins[i];
+ /* check smart51 */
+ old_line_outs = cfg->line_outs;
+ if (cfg->line_outs == 1)
+ mangle_smart51(codec);
- if (!nid)
+ err = via_auto_fill_dac_nids(codec);
+ if (err < 0)
+ return err;
+
+ if (spec->multiout.num_dacs < 3) {
+ spec->smart51_nums = 0;
+ cfg->line_outs = old_line_outs;
+ }
+ for (i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t pin, dac;
+ pin = cfg->line_out_pins[i];
+ dac = spec->multiout.dac_nids[i];
+ if (!pin || !dac)
continue;
-
- nid_vol = nid_vols[i];
-
- if (i == AUTO_SEQ_CENLFE) {
- /* Center/LFE */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
+ path = spec->out_path + i;
+ if (i == HDA_CLFE) {
+ err = create_ch_ctls(codec, "Center", 1, true, path);
if (err < 0)
return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_FRONT) {
- /* add control to mixer index 0 */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
-
- /* add control to PW3 */
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
+ err = create_ch_ctls(codec, "LFE", 2, true, path);
if (err < 0)
return err;
} else {
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
+ const char *pfx = chname[i];
+ if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
+ cfg->line_outs == 1)
+ pfx = "Speaker";
+ err = create_ch_ctls(codec, pfx, 3, true, path);
if (err < 0)
return err;
}
+ if (path != spec->out_path + i)
+ copy_path_mixer_ctls(&spec->out_path[i], path);
+ if (path == spec->out_path && spec->out_mix_path.depth)
+ copy_path_mixer_ctls(&spec->out_mix_path, path);
}
+ idx = get_connection_index(codec, spec->aa_mix_nid,
+ spec->multiout.dac_nids[0]);
+ if (idx >= 0) {
+ /* add control to mixer */
+ const char *name;
+ name = spec->out_mix_path.depth ?
+ "PCM Loopback Playback Volume" : "PCM Playback Volume";
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(spec->aa_mix_nid, 3,
+ idx, HDA_INPUT));
+ if (err < 0)
+ return err;
+ name = spec->out_mix_path.depth ?
+ "PCM Loopback Playback Switch" : "PCM Playback Switch";
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(spec->aa_mix_nid, 3,
+ idx, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+
+ cfg->line_outs = old_line_outs;
+
return 0;
}
-static void create_hp_imux(struct via_spec *spec)
+static int via_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin)
{
- int i;
- struct hda_input_mux *imux = &spec->private_imux[1];
- static const char * const texts[] = { "OFF", "ON", NULL};
-
- /* for hp mode select */
- for (i = 0; texts[i]; i++)
- snd_hda_add_imux_item(imux, texts[i], i, NULL);
-
- spec->hp_mux = &spec->private_imux[1];
-}
-
-static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
+ struct via_spec *spec = codec->spec;
+ struct nid_path *path;
+ bool check_dac;
+ int i, err;
if (!pin)
return 0;
- spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */
- spec->hp_independent_mode_index = 1;
+ if (!parse_output_path(codec, pin, 0, 0, &spec->hp_indep_path)) {
+ for (i = HDA_SIDE; i >= HDA_CLFE; i--) {
+ if (i < spec->multiout.num_dacs &&
+ parse_output_path(codec, pin,
+ spec->multiout.dac_nids[i], 0,
+ &spec->hp_indep_path)) {
+ spec->hp_indep_shared = i;
+ break;
+ }
+ }
+ }
+ if (spec->hp_indep_path.depth) {
+ spec->hp_dac_nid = spec->hp_indep_path.path[0];
+ if (!spec->hp_indep_shared)
+ spec->hp_path = spec->hp_indep_path;
+ }
+ /* optionally check front-path w/o AA-mix */
+ if (!spec->hp_path.depth)
+ parse_output_path(codec, pin,
+ spec->multiout.dac_nids[HDA_FRONT], 0,
+ &spec->hp_path);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (!parse_output_path(codec, pin, spec->multiout.dac_nids[HDA_FRONT],
+ 1, &spec->hp_mix_path) && !spec->hp_path.depth)
+ return 0;
+
+ if (spec->hp_path.depth) {
+ path = &spec->hp_path;
+ check_dac = true;
+ } else {
+ path = &spec->hp_mix_path;
+ check_dac = false;
+ }
+ err = create_ch_ctls(codec, "Headphone", 3, check_dac, path);
if (err < 0)
return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (check_dac)
+ copy_path_mixer_ctls(&spec->hp_mix_path, path);
+ else
+ copy_path_mixer_ctls(&spec->hp_path, path);
+ if (spec->hp_indep_path.depth)
+ copy_path_mixer_ctls(&spec->hp_indep_path, path);
+ return 0;
+}
+
+static int via_auto_create_speaker_ctls(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ struct nid_path *path;
+ bool check_dac;
+ hda_nid_t pin, dac;
+ int err;
+
+ pin = spec->autocfg.speaker_pins[0];
+ if (!spec->autocfg.speaker_outs || !pin)
+ return 0;
+
+ if (parse_output_path(codec, pin, 0, 0, &spec->speaker_path))
+ dac = spec->speaker_path.path[0];
+ if (!dac)
+ parse_output_path(codec, pin,
+ spec->multiout.dac_nids[HDA_FRONT], 0,
+ &spec->speaker_path);
+ if (!parse_output_path(codec, pin, spec->multiout.dac_nids[HDA_FRONT],
+ 1, &spec->speaker_mix_path) && !dac)
+ return 0;
+
+ /* no AA-path for front? */
+ if (!spec->out_mix_path.depth && spec->speaker_mix_path.depth)
+ dac = 0;
+
+ spec->speaker_dac_nid = dac;
+ spec->multiout.extra_out_nid[0] = dac;
+ if (dac) {
+ path = &spec->speaker_path;
+ check_dac = true;
+ } else {
+ path = &spec->speaker_mix_path;
+ check_dac = false;
+ }
+ err = create_ch_ctls(codec, "Speaker", 3, check_dac, path);
+ if (err < 0)
+ return err;
+ if (check_dac)
+ copy_path_mixer_ctls(&spec->speaker_mix_path, path);
+ else
+ copy_path_mixer_ctls(&spec->speaker_path, path);
+ return 0;
+}
+
+#define via_aamix_ctl_info via_pin_power_ctl_info
+
+static int via_aamix_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ ucontrol->value.enumerated.item[0] = spec->aamix_mode;
+ return 0;
+}
+
+static void update_aamix_paths(struct hda_codec *codec, int do_mix,
+ struct nid_path *nomix, struct nid_path *mix)
+{
+ if (do_mix) {
+ activate_output_path(codec, nomix, false, false);
+ activate_output_path(codec, mix, true, false);
+ } else {
+ activate_output_path(codec, mix, false, false);
+ activate_output_path(codec, nomix, true, false);
+ }
+}
+
+static int via_aamix_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ unsigned int val = ucontrol->value.enumerated.item[0];
+
+ if (val == spec->aamix_mode)
+ return 0;
+ spec->aamix_mode = val;
+ /* update front path */
+ update_aamix_paths(codec, val, &spec->out_path[0], &spec->out_mix_path);
+ /* update HP path */
+ if (!spec->hp_independent_mode) {
+ update_aamix_paths(codec, val, &spec->hp_path,
+ &spec->hp_mix_path);
+ }
+ /* update speaker path */
+ update_aamix_paths(codec, val, &spec->speaker_path,
+ &spec->speaker_mix_path);
+ return 1;
+}
+
+static const struct snd_kcontrol_new via_aamix_ctl_enum = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Loopback Mixing",
+ .info = via_aamix_ctl_info,
+ .get = via_aamix_ctl_get,
+ .put = via_aamix_ctl_put,
+};
+
+static int via_auto_create_loopback_switch(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+
+ if (!spec->aa_mix_nid || !spec->out_mix_path.depth)
+ return 0; /* no loopback switching available */
+ if (!via_clone_control(spec, &via_aamix_ctl_enum))
+ return -ENOMEM;
+ return 0;
+}
+
+/* look for ADCs */
+static int via_fill_adcs(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ hda_nid_t nid = codec->start_nid;
+ int i;
+
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ unsigned int wcaps = get_wcaps(codec, nid);
+ if (get_wcaps_type(wcaps) != AC_WID_AUD_IN)
+ continue;
+ if (wcaps & AC_WCAP_DIGITAL)
+ continue;
+ if (!(wcaps & AC_WCAP_CONN_LIST))
+ continue;
+ if (spec->num_adc_nids >= ARRAY_SIZE(spec->adc_nids))
+ return -ENOMEM;
+ spec->adc_nids[spec->num_adc_nids++] = nid;
+ }
+ return 0;
+}
+
+/* input-src control */
+static int via_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = spec->num_inputs;
+ if (uinfo->value.enumerated.item >= spec->num_inputs)
+ uinfo->value.enumerated.item = spec->num_inputs - 1;
+ strcpy(uinfo->value.enumerated.name,
+ spec->inputs[uinfo->value.enumerated.item].label);
+ return 0;
+}
+
+static int via_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[idx];
+ return 0;
+}
+
+static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ hda_nid_t mux;
+ int cur;
+
+ cur = ucontrol->value.enumerated.item[0];
+ if (cur < 0 || cur >= spec->num_inputs)
+ return -EINVAL;
+ if (spec->cur_mux[idx] == cur)
+ return 0;
+ spec->cur_mux[idx] = cur;
+ if (spec->dyn_adc_switch) {
+ int adc_idx = spec->inputs[cur].adc_idx;
+ mux = spec->mux_nids[adc_idx];
+ via_dyn_adc_pcm_resetup(codec, cur);
+ } else {
+ mux = spec->mux_nids[idx];
+ if (snd_BUG_ON(!mux))
+ return -EINVAL;
+ }
+
+ if (mux) {
+ /* switch to D0 beofre change index */
+ if (snd_hda_codec_read(codec, mux, 0,
+ AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0)
+ snd_hda_codec_write(codec, mux, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ snd_hda_codec_write(codec, mux, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ spec->inputs[cur].mux_idx);
+ }
+
+ /* update jack power state */
+ set_widgets_power_state(codec);
+ return 0;
+}
+
+static const struct snd_kcontrol_new via_input_src_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+};
+
+static int create_input_src_ctls(struct hda_codec *codec, int count)
+{
+ struct via_spec *spec = codec->spec;
+ struct snd_kcontrol_new *knew;
+
+ if (spec->num_inputs <= 1 || !count)
+ return 0; /* no need for single src */
+
+ knew = via_clone_control(spec, &via_input_src_ctl);
+ if (!knew)
+ return -ENOMEM;
+ knew->count = count;
+ return 0;
+}
+
+/* add the powersave loopback-list entry */
+static void add_loopback_list(struct via_spec *spec, hda_nid_t mix, int idx)
+{
+ struct hda_amp_list *list;
+
+ if (spec->num_loopbacks >= ARRAY_SIZE(spec->loopback_list) - 1)
+ return;
+ list = spec->loopback_list + spec->num_loopbacks;
+ list->nid = mix;
+ list->dir = HDA_INPUT;
+ list->idx = idx;
+ spec->num_loopbacks++;
+ spec->loopback.amplist = spec->loopback_list;
+}
+
+static bool is_reachable_nid(struct hda_codec *codec, hda_nid_t src,
+ hda_nid_t dst)
+{
+ return snd_hda_get_conn_index(codec, src, dst, 1) >= 0;
+}
+
+/* add the input-route to the given pin */
+static bool add_input_route(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct via_spec *spec = codec->spec;
+ int c, idx;
+
+ spec->inputs[spec->num_inputs].adc_idx = -1;
+ spec->inputs[spec->num_inputs].pin = pin;
+ for (c = 0; c < spec->num_adc_nids; c++) {
+ if (spec->mux_nids[c]) {
+ idx = get_connection_index(codec, spec->mux_nids[c],
+ pin);
+ if (idx < 0)
+ continue;
+ spec->inputs[spec->num_inputs].mux_idx = idx;
+ } else {
+ if (!is_reachable_nid(codec, spec->adc_nids[c], pin))
+ continue;
+ }
+ spec->inputs[spec->num_inputs].adc_idx = c;
+ /* Can primary ADC satisfy all inputs? */
+ if (!spec->dyn_adc_switch &&
+ spec->num_inputs > 0 && spec->inputs[0].adc_idx != c) {
+ snd_printd(KERN_INFO
+ "via: dynamic ADC switching enabled\n");
+ spec->dyn_adc_switch = 1;
+ }
+ return true;
+ }
+ return false;
+}
+
+static int get_mux_nids(struct hda_codec *codec);
+
+/* parse input-routes; fill ADCs, MUXs and input-src entries */
+static int parse_analog_inputs(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, err;
+
+ err = via_fill_adcs(codec);
+ if (err < 0)
+ return err;
+ err = get_mux_nids(codec);
if (err < 0)
return err;
- create_hp_imux(spec);
+ /* fill all input-routes */
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (add_input_route(codec, cfg->inputs[i].pin))
+ spec->inputs[spec->num_inputs++].label =
+ hda_get_autocfg_input_label(codec, cfg, i);
+ }
+
+ /* check for internal loopback recording */
+ if (spec->aa_mix_nid &&
+ add_input_route(codec, spec->aa_mix_nid))
+ spec->inputs[spec->num_inputs++].label = "Stereo Mixer";
return 0;
}
-/* create playback/capture controls for input pins */
-static int vt_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- hda_nid_t cap_nid,
- const hda_nid_t pin_idxs[],
- int num_idxs)
+/* create analog-loopback volume/switch controls */
+static int create_loopback_ctls(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
- struct hda_input_mux *imux = &spec->private_imux[0];
- int i, err, idx, type, type_idx = 0;
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ const char *prev_label = NULL;
+ int type_idx = 0;
+ int i, j, err, idx;
- /* for internal loopback recording select */
- for (idx = 0; idx < num_idxs; idx++) {
- if (pin_idxs[idx] == 0xff) {
- snd_hda_add_imux_item(imux, "Stereo Mixer", idx, NULL);
- break;
- }
- }
+ if (!spec->aa_mix_nid)
+ return 0;
for (i = 0; i < cfg->num_inputs; i++) {
- const char *label;
- type = cfg->inputs[i].type;
- for (idx = 0; idx < num_idxs; idx++)
- if (pin_idxs[idx] == cfg->inputs[i].pin)
- break;
- if (idx >= num_idxs)
- continue;
- if (i > 0 && type == cfg->inputs[i - 1].type)
+ hda_nid_t pin = cfg->inputs[i].pin;
+ const char *label = hda_get_autocfg_input_label(codec, cfg, i);
+
+ if (prev_label && !strcmp(label, prev_label))
type_idx++;
else
type_idx = 0;
- label = hda_get_autocfg_input_label(codec, cfg, i);
- if (spec->codec_type == VT1708S ||
- spec->codec_type == VT1702 ||
- spec->codec_type == VT1716S)
+ prev_label = label;
+ idx = get_connection_index(codec, spec->aa_mix_nid, pin);
+ if (idx >= 0) {
err = via_new_analog_input(spec, label, type_idx,
- idx+1, cap_nid);
- else
- err = via_new_analog_input(spec, label, type_idx,
- idx, cap_nid);
- if (err < 0)
- return err;
- snd_hda_add_imux_item(imux, label, idx, NULL);
+ idx, spec->aa_mix_nid);
+ if (err < 0)
+ return err;
+ add_loopback_list(spec, spec->aa_mix_nid, idx);
+ }
+
+ /* remember the label for smart51 control */
+ for (j = 0; j < spec->smart51_nums; j++) {
+ if (spec->smart51_pins[j] == pin) {
+ spec->smart51_idxs[j] = idx;
+ spec->smart51_labels[j] = label;
+ break;
+ }
+ }
}
return 0;
}
-/* create playback/capture controls for input pins */
-static int vt1708_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
+/* create mic-boost controls (if present) */
+static int create_mic_boost_ctls(struct hda_codec *codec)
{
- static const hda_nid_t pin_idxs[] = { 0xff, 0x24, 0x1d, 0x1e, 0x21 };
- return vt_auto_create_analog_input_ctls(codec, cfg, 0x17, pin_idxs,
- ARRAY_SIZE(pin_idxs));
+ struct via_spec *spec = codec->spec;
+ const struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, err;
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ hda_nid_t pin = cfg->inputs[i].pin;
+ unsigned int caps;
+ const char *label;
+ char name[32];
+
+ if (cfg->inputs[i].type != AUTO_PIN_MIC)
+ continue;
+ caps = query_amp_caps(codec, pin, HDA_INPUT);
+ if (caps == -1 || !(caps & AC_AMPCAP_NUM_STEPS))
+ continue;
+ label = hda_get_autocfg_input_label(codec, cfg, i);
+ snprintf(name, sizeof(name), "%s Boost Volume", label);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1708_loopbacks[] = {
- { 0x17, HDA_INPUT, 1 },
- { 0x17, HDA_INPUT, 2 },
- { 0x17, HDA_INPUT, 3 },
- { 0x17, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
+/* create capture and input-src controls for multiple streams */
+static int create_multi_adc_ctls(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int i, err;
+
+ /* create capture mixer elements */
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ hda_nid_t adc = spec->adc_nids[i];
+ err = __via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Capture Volume", i,
+ HDA_COMPOSE_AMP_VAL(adc, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = __via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Capture Switch", i,
+ HDA_COMPOSE_AMP_VAL(adc, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+
+ /* input-source control */
+ for (i = 0; i < spec->num_adc_nids; i++)
+ if (!spec->mux_nids[i])
+ break;
+ err = create_input_src_ctls(codec, i);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/* bind capture volume/switch */
+static struct snd_kcontrol_new via_bind_cap_vol_ctl =
+ HDA_BIND_VOL("Capture Volume", 0);
+static struct snd_kcontrol_new via_bind_cap_sw_ctl =
+ HDA_BIND_SW("Capture Switch", 0);
+
+static int init_bind_ctl(struct via_spec *spec, struct hda_bind_ctls **ctl_ret,
+ struct hda_ctl_ops *ops)
+{
+ struct hda_bind_ctls *ctl;
+ int i;
+
+ ctl = kzalloc(sizeof(*ctl) + sizeof(long) * 4, GFP_KERNEL);
+ if (!ctl)
+ return -ENOMEM;
+ ctl->ops = ops;
+ for (i = 0; i < spec->num_adc_nids; i++)
+ ctl->values[i] =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], 3, 0, HDA_INPUT);
+ *ctl_ret = ctl;
+ return 0;
+}
+
+/* create capture and input-src controls for dynamic ADC-switch case */
+static int create_dyn_adc_ctls(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ struct snd_kcontrol_new *knew;
+ int err;
+
+ /* set up the bind capture ctls */
+ err = init_bind_ctl(spec, &spec->bind_cap_vol, &snd_hda_bind_vol);
+ if (err < 0)
+ return err;
+ err = init_bind_ctl(spec, &spec->bind_cap_sw, &snd_hda_bind_sw);
+ if (err < 0)
+ return err;
+
+ /* create capture mixer elements */
+ knew = via_clone_control(spec, &via_bind_cap_vol_ctl);
+ if (!knew)
+ return -ENOMEM;
+ knew->private_value = (long)spec->bind_cap_vol;
+
+ knew = via_clone_control(spec, &via_bind_cap_sw_ctl);
+ if (!knew)
+ return -ENOMEM;
+ knew->private_value = (long)spec->bind_cap_sw;
+
+ /* input-source control */
+ err = create_input_src_ctls(codec, 1);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/* parse and create capture-related stuff */
+static int via_auto_create_analog_input_ctls(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+ err = parse_analog_inputs(codec);
+ if (err < 0)
+ return err;
+ if (spec->dyn_adc_switch)
+ err = create_dyn_adc_ctls(codec);
+ else
+ err = create_multi_adc_ctls(codec);
+ if (err < 0)
+ return err;
+ err = create_loopback_ctls(codec);
+ if (err < 0)
+ return err;
+ err = create_mic_boost_ctls(codec);
+ if (err < 0)
+ return err;
+ return 0;
+}
static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid)
{
@@ -2095,7 +2591,7 @@
return;
}
-static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol,
+static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
@@ -2103,13 +2599,13 @@
if (spec->codec_type != VT1708)
return 0;
- spec->vt1708_jack_detectect =
+ spec->vt1708_jack_detect =
!((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1);
- ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect;
+ ucontrol->value.integer.value[0] = spec->vt1708_jack_detect;
return 0;
}
-static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol,
+static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
@@ -2118,98 +2614,150 @@
if (spec->codec_type != VT1708)
return 0;
- spec->vt1708_jack_detectect = ucontrol->value.integer.value[0];
+ spec->vt1708_jack_detect = ucontrol->value.integer.value[0];
change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8))
- == !spec->vt1708_jack_detectect;
- if (spec->vt1708_jack_detectect) {
+ == !spec->vt1708_jack_detect;
+ if (spec->vt1708_jack_detect) {
mute_aa_path(codec, 1);
notify_aa_path_ctls(codec);
}
return change;
}
-static const struct snd_kcontrol_new vt1708_jack_detectect[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Jack Detect",
- .count = 1,
- .info = snd_ctl_boolean_mono_info,
- .get = vt1708_jack_detectect_get,
- .put = vt1708_jack_detectect_put,
- },
- {} /* end */
+static const struct snd_kcontrol_new vt1708_jack_detect_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Jack Detect",
+ .count = 1,
+ .info = snd_ctl_boolean_mono_info,
+ .get = vt1708_jack_detect_get,
+ .put = vt1708_jack_detect_put,
};
-static int vt1708_parse_auto_config(struct hda_codec *codec)
+static void fill_dig_outs(struct hda_codec *codec);
+static void fill_dig_in(struct hda_codec *codec);
+
+static int via_parse_auto_config(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
int err;
- /* Add HP and CD pin config connect bit re-config action */
- vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID);
- vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID);
-
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
if (err < 0)
return err;
- err = vt1708_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
+ return -EINVAL;
- err = vt1708_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ err = via_auto_create_multi_out_ctls(codec);
if (err < 0)
return err;
- err = vt1708_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ err = via_auto_create_hp_ctls(codec, spec->autocfg.hp_pins[0]);
if (err < 0)
return err;
- err = vt1708_auto_create_analog_input_ctls(codec, &spec->autocfg);
+ err = via_auto_create_speaker_ctls(codec);
if (err < 0)
return err;
- /* add jack detect on/off control */
- err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect);
+ err = via_auto_create_loopback_switch(codec);
+ if (err < 0)
+ return err;
+ err = via_auto_create_analog_input_ctls(codec);
if (err < 0)
return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
- if (spec->autocfg.dig_outs)
- spec->multiout.dig_out_nid = VT1708_DIGOUT_NID;
- spec->dig_in_pin = VT1708_DIGIN_PIN;
- if (spec->autocfg.dig_in_pin)
- spec->dig_in_nid = VT1708_DIGIN_NID;
+ fill_dig_outs(codec);
+ fill_dig_in(codec);
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
- spec->init_verbs[spec->num_iverbs++] = vt1708_volume_init_verbs;
- spec->input_mux = &spec->private_imux[0];
+ if (spec->hp_dac_nid && spec->hp_mix_path.depth) {
+ err = via_hp_build(codec);
+ if (err < 0)
+ return err;
+ }
- if (spec->hp_mux)
- via_hp_build(codec);
+ err = via_smart51_build(codec);
+ if (err < 0)
+ return err;
- via_smart51_build(spec);
+ /* assign slave outs */
+ if (spec->slave_dig_outs[0])
+ codec->slave_dig_outs = spec->slave_dig_outs;
+
return 1;
}
-/* init callback for auto-configuration model -- overriding the default init */
-static int via_auto_init(struct hda_codec *codec)
+static void via_auto_init_dig_outs(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
+ if (spec->multiout.dig_out_nid)
+ init_output_pin(codec, spec->autocfg.dig_out_pins[0], PIN_OUT);
+ if (spec->slave_dig_outs[0])
+ init_output_pin(codec, spec->autocfg.dig_out_pins[1], PIN_OUT);
+}
- via_init(codec);
+static void via_auto_init_dig_in(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ if (!spec->dig_in_nid)
+ return;
+ snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+}
+
+/* initialize the unsolicited events */
+static void via_auto_init_unsol_event(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int ev;
+ int i;
+
+ if (cfg->hp_pins[0] && is_jack_detectable(codec, cfg->hp_pins[0]))
+ snd_hda_codec_write(codec, cfg->hp_pins[0], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT);
+
+ if (cfg->speaker_pins[0])
+ ev = VIA_LINE_EVENT;
+ else
+ ev = 0;
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (cfg->line_out_pins[i] &&
+ is_jack_detectable(codec, cfg->line_out_pins[i]))
+ snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ev | VIA_JACK_EVENT);
+ }
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (is_jack_detectable(codec, cfg->inputs[i].pin))
+ snd_hda_codec_write(codec, cfg->inputs[i].pin, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_JACK_EVENT);
+ }
+}
+
+static int via_init(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_iverbs; i++)
+ snd_hda_sequence_write(codec, spec->init_verbs[i]);
+
via_auto_init_multi_out(codec);
via_auto_init_hp_out(codec);
+ via_auto_init_speaker_out(codec);
via_auto_init_analog_input(codec);
+ via_auto_init_dig_outs(codec);
+ via_auto_init_dig_in(codec);
- if (VT2002P_COMPATIBLE(spec)) {
- via_hp_bind_automute(codec);
- } else {
- via_hp_automute(codec);
- via_speaker_automute(codec);
- }
+ via_auto_init_unsol_event(codec);
+
+ via_hp_automute(codec);
return 0;
}
@@ -2266,437 +2814,36 @@
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x17;
+
+ /* Add HP and CD pin config connect bit re-config action */
+ vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID);
+ vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID);
+
/* automatic parse from the BIOS config */
- err = vt1708_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
}
+ /* add jack detect on/off control */
+ if (!via_clone_control(spec, &vt1708_jack_detect_ctl))
+ return -ENOMEM;
- spec->stream_name_analog = "VT1708 Analog";
- spec->stream_analog_playback = &vt1708_pcm_analog_playback;
/* disable 32bit format on VT1708 */
if (codec->vendor_id == 0x11061708)
spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback;
- spec->stream_analog_capture = &vt1708_pcm_analog_capture;
- spec->stream_name_digital = "VT1708 Digital";
- spec->stream_digital_playback = &vt1708_pcm_digital_playback;
- spec->stream_digital_capture = &vt1708_pcm_digital_capture;
-
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1708_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids);
- get_mux_nids(codec);
- spec->mixers[spec->num_mixers] = vt1708_capture_mixer;
- spec->num_mixers++;
- }
+ spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs;
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1708_loopbacks;
-#endif
INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state);
return 0;
}
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1709_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x15, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x16, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x16, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb vt1709_uniwill_init_verbs[] = {
- {0x20, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
- { }
-};
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb vt1709_10ch_volume_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: AOW0=0, CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- /*
- * Set up output selector (0x1a, 0x1b, 0x29)
- */
- /* set vol=0 to output mixers */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /*
- * Unmute PW3 and PW4
- */
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Set input of PW4 as MW0 */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0},
- /* PW9 Output enable */
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- { }
-};
-
-static const struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 10,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- },
-};
-
-static const struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 6,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- },
-};
-
-static const struct hda_pcm_stream vt1709_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x14, /* NID to query formats and rates */
- .ops = {
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream vt1709_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close
- },
-};
-
-static const struct hda_pcm_stream vt1709_pcm_digital_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
-};
-
-static int vt1709_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int i;
- hda_nid_t nid;
-
- if (cfg->line_outs == 4) /* 10 channels */
- spec->multiout.num_dacs = cfg->line_outs+1; /* AOW0~AOW4 */
- else if (cfg->line_outs == 3) /* 6 channels */
- spec->multiout.num_dacs = cfg->line_outs; /* AOW0~AOW2 */
-
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- if (cfg->line_outs == 4) { /* 10 channels */
- for (i = 0; i < cfg->line_outs; i++) {
- nid = cfg->line_out_pins[i];
- if (nid) {
- /* config dac list */
- switch (i) {
- case AUTO_SEQ_FRONT:
- /* AOW0 */
- spec->private_dac_nids[i] = 0x10;
- break;
- case AUTO_SEQ_CENLFE:
- /* AOW2 */
- spec->private_dac_nids[i] = 0x12;
- break;
- case AUTO_SEQ_SURROUND:
- /* AOW3 */
- spec->private_dac_nids[i] = 0x11;
- break;
- case AUTO_SEQ_SIDE:
- /* AOW1 */
- spec->private_dac_nids[i] = 0x27;
- break;
- default:
- break;
- }
- }
- }
- spec->private_dac_nids[cfg->line_outs] = 0x28; /* AOW4 */
-
- } else if (cfg->line_outs == 3) { /* 6 channels */
- for (i = 0; i < cfg->line_outs; i++) {
- nid = cfg->line_out_pins[i];
- if (nid) {
- /* config dac list */
- switch (i) {
- case AUTO_SEQ_FRONT:
- /* AOW0 */
- spec->private_dac_nids[i] = 0x10;
- break;
- case AUTO_SEQ_CENLFE:
- /* AOW2 */
- spec->private_dac_nids[i] = 0x12;
- break;
- case AUTO_SEQ_SURROUND:
- /* AOW1 */
- spec->private_dac_nids[i] = 0x11;
- break;
- default:
- break;
- }
- }
- }
- }
-
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- char name[32];
- static const char * const chname[4] = {
- "Front", "Surround", "C/LFE", "Side"
- };
- hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29};
- int i, err;
-
- for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
- nid = cfg->line_out_pins[i];
-
- if (!nid)
- continue;
-
- nid_vol = nid_vols[i];
-
- if (i == AUTO_SEQ_CENLFE) {
- /* Center/LFE */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_FRONT) {
- /* ADD control to mixer index 0 */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
-
- /* add control to PW3 */
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_SURROUND) {
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_SIDE) {
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
-
- return 0;
-}
-
-static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
-
- if (!pin)
- return 0;
-
- if (spec->multiout.num_dacs == 5) /* 10 channels */
- spec->multiout.hp_nid = VT1709_HP_DAC_NID;
- else if (spec->multiout.num_dacs == 3) /* 6 channels */
- spec->multiout.hp_nid = 0;
- spec->hp_independent_mode_index = 1;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt1709_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- static const hda_nid_t pin_idxs[] = { 0xff, 0x23, 0x1d, 0x1e, 0x21 };
- return vt_auto_create_analog_input_ctls(codec, cfg, 0x18, pin_idxs,
- ARRAY_SIZE(pin_idxs));
-}
-
-static int vt1709_parse_auto_config(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int err;
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
- if (err < 0)
- return err;
- err = vt1709_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
-
- err = vt1709_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt1709_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = vt1709_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- if (spec->autocfg.dig_outs)
- spec->multiout.dig_out_nid = VT1709_DIGOUT_NID;
- spec->dig_in_pin = VT1709_DIGIN_PIN;
- if (spec->autocfg.dig_in_pin)
- spec->dig_in_nid = VT1709_DIGIN_NID;
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- via_smart51_build(spec);
- return 1;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1709_loopbacks[] = {
- { 0x18, HDA_INPUT, 1 },
- { 0x18, HDA_INPUT, 2 },
- { 0x18, HDA_INPUT, 3 },
- { 0x18, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
-static int patch_vt1709_10ch(struct hda_codec *codec)
+static int patch_vt1709(struct hda_codec *codec)
{
struct via_spec *spec;
int err;
@@ -2706,527 +2853,18 @@
if (spec == NULL)
return -ENOMEM;
- err = vt1709_parse_auto_config(codec);
+ spec->aa_mix_nid = 0x18;
+
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration. "
- "Using genenic mode...\n");
- }
-
- spec->init_verbs[spec->num_iverbs++] = vt1709_10ch_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs;
-
- spec->stream_name_analog = "VT1709 Analog";
- spec->stream_analog_playback = &vt1709_10ch_pcm_analog_playback;
- spec->stream_analog_capture = &vt1709_pcm_analog_capture;
-
- spec->stream_name_digital = "VT1709 Digital";
- spec->stream_digital_playback = &vt1709_pcm_digital_playback;
- spec->stream_digital_capture = &vt1709_pcm_digital_capture;
-
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1709_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
- get_mux_nids(codec);
- spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
- spec->num_mixers++;
}
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1709_loopbacks;
-#endif
-
return 0;
}
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb vt1709_6ch_volume_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: AOW0=0, CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- /*
- * Set up output selector (0x1a, 0x1b, 0x29)
- */
- /* set vol=0 to output mixers */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /*
- * Unmute PW3 and PW4
- */
- {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Set input of PW4 as MW0 */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0},
- /* PW9 Output enable */
- {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- { }
-};
-
-static int patch_vt1709_6ch(struct hda_codec *codec)
-{
- struct via_spec *spec;
- int err;
-
- /* create a codec specific record */
- spec = via_new_spec(codec);
- if (spec == NULL)
- return -ENOMEM;
-
- err = vt1709_parse_auto_config(codec);
- if (err < 0) {
- via_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration. "
- "Using genenic mode...\n");
- }
-
- spec->init_verbs[spec->num_iverbs++] = vt1709_6ch_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs;
-
- spec->stream_name_analog = "VT1709 Analog";
- spec->stream_analog_playback = &vt1709_6ch_pcm_analog_playback;
- spec->stream_analog_capture = &vt1709_pcm_analog_capture;
-
- spec->stream_name_digital = "VT1709 Digital";
- spec->stream_digital_playback = &vt1709_pcm_digital_playback;
- spec->stream_digital_capture = &vt1709_pcm_digital_capture;
-
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1709_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
- get_mux_nids(codec);
- spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
- spec->num_mixers++;
- }
-
- codec->patch_ops = via_patch_ops;
-
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1709_loopbacks;
-#endif
- return 0;
-}
-
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1708B_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb vt1708B_8ch_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- /*
- * Set up output mixers
- */
- /* set vol=0 to output mixers */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Setup default input to PW4 */
- {0x1d, AC_VERB_SET_CONNECT_SEL, 0},
- /* PW9 Output enable */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* PW10 Input enable */
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- { }
-};
-
-static const struct hda_verb vt1708B_4ch_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- /*
- * Set up output mixers
- */
- /* set vol=0 to output mixers */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Setup default input of PW4 to MW0 */
- {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* PW9 Output enable */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* PW10 Input enable */
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- { }
-};
-
-static const struct hda_verb vt1708B_uniwill_init_verbs[] = {
- {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static int via_pcm_open_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- int idle = substream->pstr->substream_opened == 1
- && substream->ref_count == 0;
-
- analog_low_current_mode(codec, idle);
- return 0;
-}
-
-static const struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 8,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close
- },
-};
-
-static const struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 4,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream vt1708B_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x13, /* NID to query formats and rates */
- .ops = {
- .open = via_pcm_open_close,
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup,
- .close = via_pcm_open_close
- },
-};
-
-static const struct hda_pcm_stream vt1708B_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare,
- .cleanup = via_dig_playback_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream vt1708B_pcm_digital_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
-};
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt1708B_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int i;
- hda_nid_t nid;
-
- spec->multiout.num_dacs = cfg->line_outs;
-
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- for (i = 0; i < 4; i++) {
- nid = cfg->line_out_pins[i];
- if (nid) {
- /* config dac list */
- switch (i) {
- case AUTO_SEQ_FRONT:
- spec->private_dac_nids[i] = 0x10;
- break;
- case AUTO_SEQ_CENLFE:
- spec->private_dac_nids[i] = 0x24;
- break;
- case AUTO_SEQ_SURROUND:
- spec->private_dac_nids[i] = 0x11;
- break;
- case AUTO_SEQ_SIDE:
- spec->private_dac_nids[i] = 0x25;
- break;
- }
- }
- }
-
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- char name[32];
- static const char * const chname[4] = {
- "Front", "Surround", "C/LFE", "Side"
- };
- hda_nid_t nid_vols[] = {0x16, 0x18, 0x26, 0x27};
- hda_nid_t nid, nid_vol = 0;
- int i, err;
-
- for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
- nid = cfg->line_out_pins[i];
-
- if (!nid)
- continue;
-
- nid_vol = nid_vols[i];
-
- if (i == AUTO_SEQ_CENLFE) {
- /* Center/LFE */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_FRONT) {
- /* add control to mixer index 0 */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
-
- /* add control to PW3 */
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
-
- return 0;
-}
-
-static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
-
- if (!pin)
- return 0;
-
- spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */
- spec->hp_independent_mode_index = 1;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- create_hp_imux(spec);
-
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt1708B_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- static const hda_nid_t pin_idxs[] = { 0xff, 0x1f, 0x1a, 0x1b, 0x1e };
- return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs,
- ARRAY_SIZE(pin_idxs));
-}
-
-static int vt1708B_parse_auto_config(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int err;
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
- if (err < 0)
- return err;
- err = vt1708B_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
-
- err = vt1708B_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt1708B_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = vt1708B_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- if (spec->autocfg.dig_outs)
- spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID;
- spec->dig_in_pin = VT1708B_DIGIN_PIN;
- if (spec->autocfg.dig_in_pin)
- spec->dig_in_nid = VT1708B_DIGIN_NID;
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- via_smart51_build(spec);
- return 1;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1708B_loopbacks[] = {
- { 0x16, HDA_INPUT, 1 },
- { 0x16, HDA_INPUT, 2 },
- { 0x16, HDA_INPUT, 3 },
- { 0x16, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
{
@@ -3309,157 +2947,37 @@
}
static int patch_vt1708S(struct hda_codec *codec);
-static int patch_vt1708B_8ch(struct hda_codec *codec)
+static int patch_vt1708B(struct hda_codec *codec)
{
struct via_spec *spec;
int err;
if (get_codec_type(codec) == VT1708BCE)
return patch_vt1708S(codec);
- /* create a codec specific record */
- spec = via_new_spec(codec);
- if (spec == NULL)
- return -ENOMEM;
-
- /* automatic parse from the BIOS config */
- err = vt1708B_parse_auto_config(codec);
- if (err < 0) {
- via_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
- }
-
- spec->init_verbs[spec->num_iverbs++] = vt1708B_8ch_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs;
-
- spec->stream_name_analog = "VT1708B Analog";
- spec->stream_analog_playback = &vt1708B_8ch_pcm_analog_playback;
- spec->stream_analog_capture = &vt1708B_pcm_analog_capture;
-
- spec->stream_name_digital = "VT1708B Digital";
- spec->stream_digital_playback = &vt1708B_pcm_digital_playback;
- spec->stream_digital_capture = &vt1708B_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1708B_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
- get_mux_nids(codec);
- spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
- spec->num_mixers++;
- }
-
- codec->patch_ops = via_patch_ops;
-
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1708B_loopbacks;
-#endif
-
- spec->set_widgets_power_state = set_widgets_power_state_vt1708B;
-
- return 0;
-}
-
-static int patch_vt1708B_4ch(struct hda_codec *codec)
-{
- struct via_spec *spec;
- int err;
/* create a codec specific record */
spec = via_new_spec(codec);
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x16;
+
/* automatic parse from the BIOS config */
- err = vt1708B_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
- }
-
- spec->init_verbs[spec->num_iverbs++] = vt1708B_4ch_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs;
-
- spec->stream_name_analog = "VT1708B Analog";
- spec->stream_analog_playback = &vt1708B_4ch_pcm_analog_playback;
- spec->stream_analog_capture = &vt1708B_pcm_analog_capture;
-
- spec->stream_name_digital = "VT1708B Digital";
- spec->stream_digital_playback = &vt1708B_pcm_digital_playback;
- spec->stream_digital_capture = &vt1708B_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1708B_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
- get_mux_nids(codec);
- spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
- spec->num_mixers++;
}
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1708B_loopbacks;
-#endif
-
spec->set_widgets_power_state = set_widgets_power_state_vt1708B;
return 0;
}
/* Patch for VT1708S */
-
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1708S_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0,
- HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb vt1708S_volume_init_verbs[] = {
- /* Unmute ADC0-1 and set the default input to mic-in */
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the
- * analog-loopback mixer widget */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- /* Setup default input of PW4 to MW0 */
- {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* PW9, PW10 Output enable */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+static const struct hda_verb vt1708S_init_verbs[] = {
/* Enable Mic Boost Volume backdoor */
{0x1, 0xf98, 0x1},
/* don't bybass mixer */
@@ -3467,277 +2985,6 @@
{ }
};
-static const struct hda_verb vt1708S_uniwill_init_verbs[] = {
- {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static const struct hda_verb vt1705_uniwill_init_verbs[] = {
- {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static const struct hda_pcm_stream vt1708S_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 8,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close
- },
-};
-
-static const struct hda_pcm_stream vt1705_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 6,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close
- },
-};
-
-static const struct hda_pcm_stream vt1708S_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x13, /* NID to query formats and rates */
- .ops = {
- .open = via_pcm_open_close,
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup,
- .close = via_pcm_open_close
- },
-};
-
-static const struct hda_pcm_stream vt1708S_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare,
- .cleanup = via_dig_playback_pcm_cleanup
- },
-};
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt1708S_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int i;
- hda_nid_t nid;
-
- spec->multiout.num_dacs = cfg->line_outs;
-
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- for (i = 0; i < 4; i++) {
- nid = cfg->line_out_pins[i];
- if (nid) {
- /* config dac list */
- switch (i) {
- case AUTO_SEQ_FRONT:
- spec->private_dac_nids[i] = 0x10;
- break;
- case AUTO_SEQ_CENLFE:
- if (spec->codec->vendor_id == 0x11064397)
- spec->private_dac_nids[i] = 0x25;
- else
- spec->private_dac_nids[i] = 0x24;
- break;
- case AUTO_SEQ_SURROUND:
- spec->private_dac_nids[i] = 0x11;
- break;
- case AUTO_SEQ_SIDE:
- spec->private_dac_nids[i] = 0x25;
- break;
- }
- }
- }
-
- /* for Smart 5.1, line/mic inputs double as output pins */
- if (cfg->line_outs == 1) {
- spec->multiout.num_dacs = 3;
- spec->private_dac_nids[AUTO_SEQ_SURROUND] = 0x11;
- if (spec->codec->vendor_id == 0x11064397)
- spec->private_dac_nids[AUTO_SEQ_CENLFE] = 0x25;
- else
- spec->private_dac_nids[AUTO_SEQ_CENLFE] = 0x24;
- }
-
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int vt1708S_auto_create_multi_out_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- struct via_spec *spec = codec->spec;
- char name[32];
- static const char * const chname[4] = {
- "Front", "Surround", "C/LFE", "Side"
- };
- hda_nid_t nid_vols[2][4] = { {0x10, 0x11, 0x24, 0x25},
- {0x10, 0x11, 0x25, 0} };
- hda_nid_t nid_mutes[2][4] = { {0x1C, 0x18, 0x26, 0x27},
- {0x1C, 0x18, 0x27, 0} };
- hda_nid_t nid, nid_vol, nid_mute;
- int i, err;
-
- for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
- nid = cfg->line_out_pins[i];
-
- /* for Smart 5.1, there are always at least six channels */
- if (!nid && i > AUTO_SEQ_CENLFE)
- continue;
-
- if (codec->vendor_id == 0x11064397) {
- nid_vol = nid_vols[1][i];
- nid_mute = nid_mutes[1][i];
- } else {
- nid_vol = nid_vols[0][i];
- nid_mute = nid_mutes[0][i];
- }
- if (!nid_vol && !nid_mute)
- continue;
-
- if (i == AUTO_SEQ_CENLFE) {
- /* Center/LFE */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_mute,
- 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_mute,
- 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_FRONT) {
- /* add control to mixer index 0 */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x16, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x16, 3, 0,
- HDA_INPUT));
- if (err < 0)
- return err;
-
- /* Front */
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_mute,
- 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_mute,
- 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
-
- return 0;
-}
-
-static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
-
- if (!pin)
- return 0;
-
- spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */
- spec->hp_independent_mode_index = 1;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- create_hp_imux(spec);
-
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt1708S_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- static const hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff };
- return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs,
- ARRAY_SIZE(pin_idxs));
-}
-
/* fill out digital output widgets; one for master and one for slave outputs */
static void fill_dig_outs(struct hda_codec *codec)
{
@@ -3763,56 +3010,33 @@
}
}
-static int vt1708S_parse_auto_config(struct hda_codec *codec)
+static void fill_dig_in(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
- int err;
+ hda_nid_t dig_nid;
+ int i, err;
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
- if (err < 0)
- return err;
- err = vt1708S_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
+ if (!spec->autocfg.dig_in_pin)
+ return;
- err = vt1708S_auto_create_multi_out_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = vt1708S_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- fill_dig_outs(codec);
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- via_smart51_build(spec);
- return 1;
+ dig_nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, dig_nid++) {
+ unsigned int wcaps = get_wcaps(codec, dig_nid);
+ if (get_wcaps_type(wcaps) != AC_WID_AUD_IN)
+ continue;
+ if (!(wcaps & AC_WCAP_DIGITAL))
+ continue;
+ if (!(wcaps & AC_WCAP_CONN_LIST))
+ continue;
+ err = get_connection_index(codec, dig_nid,
+ spec->autocfg.dig_in_pin);
+ if (err >= 0) {
+ spec->dig_in_nid = dig_nid;
+ break;
+ }
+ }
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1708S_loopbacks[] = {
- { 0x16, HDA_INPUT, 1 },
- { 0x16, HDA_INPUT, 2 },
- { 0x16, HDA_INPUT, 3 },
- { 0x16, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin,
int offset, int num_steps, int step_size)
{
@@ -3833,62 +3057,21 @@
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x16;
+ override_mic_boost(codec, 0x1a, 0, 3, 40);
+ override_mic_boost(codec, 0x1e, 0, 3, 40);
+
/* automatic parse from the BIOS config */
- err = vt1708S_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
}
- spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs;
- if (codec->vendor_id == 0x11064397)
- spec->init_verbs[spec->num_iverbs++] =
- vt1705_uniwill_init_verbs;
- else
- spec->init_verbs[spec->num_iverbs++] =
- vt1708S_uniwill_init_verbs;
-
- if (codec->vendor_id == 0x11060440)
- spec->stream_name_analog = "VT1818S Analog";
- else if (codec->vendor_id == 0x11064397)
- spec->stream_name_analog = "VT1705 Analog";
- else
- spec->stream_name_analog = "VT1708S Analog";
- if (codec->vendor_id == 0x11064397)
- spec->stream_analog_playback = &vt1705_pcm_analog_playback;
- else
- spec->stream_analog_playback = &vt1708S_pcm_analog_playback;
- spec->stream_analog_capture = &vt1708S_pcm_analog_capture;
-
- if (codec->vendor_id == 0x11060440)
- spec->stream_name_digital = "VT1818S Digital";
- else if (codec->vendor_id == 0x11064397)
- spec->stream_name_digital = "VT1705 Digital";
- else
- spec->stream_name_digital = "VT1708S Digital";
- spec->stream_digital_playback = &vt1708S_pcm_digital_playback;
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1708S_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids);
- get_mux_nids(codec);
- override_mic_boost(codec, 0x1a, 0, 3, 40);
- override_mic_boost(codec, 0x1e, 0, 3, 40);
- spec->mixers[spec->num_mixers] = vt1708S_capture_mixer;
- spec->num_mixers++;
- }
+ spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs;
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1708S_loopbacks;
-#endif
-
/* correct names for VT1708BCE */
if (get_codec_type(codec) == VT1708BCE) {
kfree(codec->chip_name);
@@ -3896,13 +3079,6 @@
snprintf(codec->bus->card->mixername,
sizeof(codec->bus->card->mixername),
"%s %s", codec->vendor_name, codec->chip_name);
- spec->stream_name_analog = "VT1708BCE Analog";
- spec->stream_name_digital = "VT1708BCE Digital";
- }
- /* correct names for VT1818S */
- if (codec->vendor_id == 0x11060440) {
- spec->stream_name_analog = "VT1818S Analog";
- spec->stream_name_digital = "VT1818S Digital";
}
/* correct names for VT1705 */
if (codec->vendor_id == 0x11064397) {
@@ -3918,55 +3094,7 @@
/* Patch for VT1702 */
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1702_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x20, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x1F, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x1F, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Digital Mic Boost Capture Volume", 0x1E, 0x0,
- HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb vt1702_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1F, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: Mic1 = 1, Line = 1, Mic2 = 3 */
- {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* Setup default input of PW4 to MW0 */
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x1},
- /* PW6 PW7 Output enable */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+static const struct hda_verb vt1702_init_verbs[] = {
/* mixer enable */
{0x1, 0xF88, 0x3},
/* GPIO 0~2 */
@@ -3974,202 +3102,6 @@
{ }
};
-static const struct hda_verb vt1702_uniwill_init_verbs[] = {
- {0x17, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static const struct hda_pcm_stream vt1702_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close
- },
-};
-
-static const struct hda_pcm_stream vt1702_pcm_analog_capture = {
- .substreams = 3,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x12, /* NID to query formats and rates */
- .ops = {
- .open = via_pcm_open_close,
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup,
- .close = via_pcm_open_close
- },
-};
-
-static const struct hda_pcm_stream vt1702_pcm_digital_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare,
- .cleanup = via_dig_playback_pcm_cleanup
- },
-};
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt1702_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- if (cfg->line_out_pins[0]) {
- /* config dac list */
- spec->private_dac_nids[0] = 0x10;
- }
-
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int vt1702_auto_create_line_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int err;
-
- if (!cfg->line_out_pins[0])
- return -1;
-
- /* add control to mixer index 0 */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
-
- /* Front */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- return 0;
-}
-
-static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err, i;
- struct hda_input_mux *imux;
- static const char * const texts[] = { "ON", "OFF", NULL};
- if (!pin)
- return 0;
- spec->multiout.hp_nid = 0x1D;
- spec->hp_independent_mode_index = 0;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x1D, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- imux = &spec->private_imux[1];
-
- /* for hp mode select */
- for (i = 0; texts[i]; i++)
- snd_hda_add_imux_item(imux, texts[i], i, NULL);
-
- spec->hp_mux = &spec->private_imux[1];
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt1702_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- static const hda_nid_t pin_idxs[] = { 0x14, 0x15, 0x18, 0xff };
- return vt_auto_create_analog_input_ctls(codec, cfg, 0x1a, pin_idxs,
- ARRAY_SIZE(pin_idxs));
-}
-
-static int vt1702_parse_auto_config(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int err;
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
- if (err < 0)
- return err;
- err = vt1702_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
-
- err = vt1702_auto_create_line_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- /* limit AA path volume to 0 dB */
- snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT,
- (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
- (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
- (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) |
- (1 << AC_AMPCAP_MUTE_SHIFT));
- err = vt1702_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- fill_dig_outs(codec);
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- return 1;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1702_loopbacks[] = {
- { 0x1A, HDA_INPUT, 1 },
- { 0x1A, HDA_INPUT, 2 },
- { 0x1A, HDA_INPUT, 3 },
- { 0x1A, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
static void set_widgets_power_state_vt1702(struct hda_codec *codec)
{
int imux_is_smixer =
@@ -4211,393 +3143,41 @@
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x1a;
+
+ /* limit AA path volume to 0 dB */
+ snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
+
/* automatic parse from the BIOS config */
- err = vt1702_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
}
- spec->init_verbs[spec->num_iverbs++] = vt1702_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1702_uniwill_init_verbs;
-
- spec->stream_name_analog = "VT1702 Analog";
- spec->stream_analog_playback = &vt1702_pcm_analog_playback;
- spec->stream_analog_capture = &vt1702_pcm_analog_capture;
-
- spec->stream_name_digital = "VT1702 Digital";
- spec->stream_digital_playback = &vt1702_pcm_digital_playback;
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1702_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids);
- get_mux_nids(codec);
- spec->mixers[spec->num_mixers] = vt1702_capture_mixer;
- spec->num_mixers++;
- }
+ spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs;
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1702_loopbacks;
-#endif
-
spec->set_widgets_power_state = set_widgets_power_state_vt1702;
return 0;
}
/* Patch for VT1718S */
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1718S_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0,
- HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- .name = "Input Source",
- .count = 2,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb vt1718S_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
+static const struct hda_verb vt1718S_init_verbs[] = {
/* Enable MW0 adjust Gain 5 */
{0x1, 0xfb2, 0x10},
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- /* PW9 PW10 Output enable */
- {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
- {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
- /* PW11 Input enable */
- {0x2f, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_IN_EN},
/* Enable Boost Volume backdoor */
{0x1, 0xf88, 0x8},
- /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
- {0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
- {0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
+
{ }
};
-
-static const struct hda_verb vt1718S_uniwill_init_verbs[] = {
- {0x28, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
- {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x27, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static const struct hda_pcm_stream vt1718S_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 10,
- .nid = 0x8, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt1718S_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_pcm_open_close,
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt1718S_pcm_digital_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare,
- .cleanup = via_dig_playback_pcm_cleanup
- },
-};
-
-static const struct hda_pcm_stream vt1718S_pcm_digital_capture = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
-};
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt1718S_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int i;
- hda_nid_t nid;
-
- spec->multiout.num_dacs = cfg->line_outs;
-
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- for (i = 0; i < 4; i++) {
- nid = cfg->line_out_pins[i];
- if (nid) {
- /* config dac list */
- switch (i) {
- case AUTO_SEQ_FRONT:
- spec->private_dac_nids[i] = 0x8;
- break;
- case AUTO_SEQ_CENLFE:
- spec->private_dac_nids[i] = 0xa;
- break;
- case AUTO_SEQ_SURROUND:
- spec->private_dac_nids[i] = 0x9;
- break;
- case AUTO_SEQ_SIDE:
- spec->private_dac_nids[i] = 0xb;
- break;
- }
- }
- }
-
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- char name[32];
- static const char * const chname[4] = {
- "Front", "Surround", "C/LFE", "Side"
- };
- hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb};
- hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27};
- hda_nid_t nid, nid_vol, nid_mute = 0;
- int i, err;
-
- for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
- nid = cfg->line_out_pins[i];
-
- if (!nid)
- continue;
- nid_vol = nid_vols[i];
- nid_mute = nid_mutes[i];
-
- if (i == AUTO_SEQ_CENLFE) {
- /* Center/LFE */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_FRONT) {
- /* add control to mixer index 0 */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
- HDA_INPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
- HDA_INPUT));
- if (err < 0)
- return err;
- /* Front */
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
-
- if (!pin)
- return 0;
-
- spec->multiout.hp_nid = 0xc; /* AOW4 */
- spec->hp_independent_mode_index = 1;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(0xc, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- create_hp_imux(spec);
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt1718S_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- static const hda_nid_t pin_idxs[] = { 0x2c, 0x2b, 0x2a, 0x29, 0, 0xff };
- return vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs,
- ARRAY_SIZE(pin_idxs));
-}
-
-static int vt1718S_parse_auto_config(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int err;
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
-
- if (err < 0)
- return err;
- err = vt1718S_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
-
- err = vt1718S_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = vt1718S_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- fill_dig_outs(codec);
-
- if (spec->autocfg.dig_in_pin && codec->vendor_id == 0x11060428)
- spec->dig_in_nid = 0x13;
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- via_smart51_build(spec);
-
- return 1;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1718S_loopbacks[] = {
- { 0x21, HDA_INPUT, 1 },
- { 0x21, HDA_INPUT, 2 },
- { 0x21, HDA_INPUT, 3 },
- { 0x21, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -4664,6 +3244,41 @@
}
}
+/* Add a connection to the primary DAC from AA-mixer for some codecs
+ * This isn't listed from the raw info, but the chip has a secret connection.
+ */
+static int add_secret_dac_path(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int i, nums;
+ hda_nid_t conn[8];
+ hda_nid_t nid;
+
+ if (!spec->aa_mix_nid)
+ return 0;
+ nums = snd_hda_get_connections(codec, spec->aa_mix_nid, conn,
+ ARRAY_SIZE(conn) - 1);
+ for (i = 0; i < nums; i++) {
+ if (get_wcaps_type(get_wcaps(codec, conn[i])) == AC_WID_AUD_OUT)
+ return 0;
+ }
+
+ /* find the primary DAC and add to the connection list */
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ unsigned int caps = get_wcaps(codec, nid);
+ if (get_wcaps_type(caps) == AC_WID_AUD_OUT &&
+ !(caps & AC_WCAP_DIGITAL)) {
+ conn[nums++] = nid;
+ return snd_hda_override_conn_list(codec,
+ spec->aa_mix_nid,
+ nums, conn);
+ }
+ }
+ return 0;
+}
+
+
static int patch_vt1718S(struct hda_codec *codec)
{
struct via_spec *spec;
@@ -4674,57 +3289,22 @@
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x21;
+ override_mic_boost(codec, 0x2b, 0, 3, 40);
+ override_mic_boost(codec, 0x29, 0, 3, 40);
+ add_secret_dac_path(codec);
+
/* automatic parse from the BIOS config */
- err = vt1718S_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
}
- spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs;
-
- if (codec->vendor_id == 0x11060441)
- spec->stream_name_analog = "VT2020 Analog";
- else if (codec->vendor_id == 0x11064441)
- spec->stream_name_analog = "VT1828S Analog";
- else
- spec->stream_name_analog = "VT1718S Analog";
- spec->stream_analog_playback = &vt1718S_pcm_analog_playback;
- spec->stream_analog_capture = &vt1718S_pcm_analog_capture;
-
- if (codec->vendor_id == 0x11060441)
- spec->stream_name_digital = "VT2020 Digital";
- else if (codec->vendor_id == 0x11064441)
- spec->stream_name_digital = "VT1828S Digital";
- else
- spec->stream_name_digital = "VT1718S Digital";
- spec->stream_digital_playback = &vt1718S_pcm_digital_playback;
- if (codec->vendor_id == 0x11060428 || codec->vendor_id == 0x11060441)
- spec->stream_digital_capture = &vt1718S_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1718S_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids);
- get_mux_nids(codec);
- override_mic_boost(codec, 0x2b, 0, 3, 40);
- override_mic_boost(codec, 0x29, 0, 3, 40);
- spec->mixers[spec->num_mixers] = vt1718S_capture_mixer;
- spec->num_mixers++;
- }
+ spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs;
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1718S_loopbacks;
-#endif
-
spec->set_widgets_power_state = set_widgets_power_state_vt1718S;
return 0;
@@ -4770,26 +3350,6 @@
return 1;
}
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1716S_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0,
- HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .count = 1,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = {
HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT),
{
@@ -4811,45 +3371,7 @@
{ } /* end */
};
-static const struct hda_verb vt1716S_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* MUX Indices: Stereo Mixer = 5 */
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x5},
-
- /* Setup default input of PW4 to MW0 */
- {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0},
-
- /* Setup default input of SW1 as MW0 */
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x1},
-
- /* Setup default input of SW4 as AOW0 */
- {0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
-
- /* PW9 PW10 Output enable */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-
- /* Unmute SW1, PW12 */
- {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* PW12 Output enable */
- {0x2a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+static const struct hda_verb vt1716S_init_verbs[] = {
/* Enable Boost Volume backdoor */
{0x1, 0xf8a, 0x80},
/* don't bybass mixer */
@@ -4859,272 +3381,6 @@
{ }
};
-
-static const struct hda_verb vt1716S_uniwill_init_verbs[] = {
- {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_MONO_EVENT | VIA_JACK_EVENT},
- {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static const struct hda_pcm_stream vt1716S_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 6,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt1716S_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x13, /* NID to query formats and rates */
- .ops = {
- .open = via_pcm_open_close,
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt1716S_pcm_digital_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare,
- .cleanup = via_dig_playback_pcm_cleanup
- },
-};
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt1716S_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{ int i;
- hda_nid_t nid;
-
- spec->multiout.num_dacs = cfg->line_outs;
-
- spec->multiout.dac_nids = spec->private_dac_nids;
-
- for (i = 0; i < 3; i++) {
- nid = cfg->line_out_pins[i];
- if (nid) {
- /* config dac list */
- switch (i) {
- case AUTO_SEQ_FRONT:
- spec->private_dac_nids[i] = 0x10;
- break;
- case AUTO_SEQ_CENLFE:
- spec->private_dac_nids[i] = 0x25;
- break;
- case AUTO_SEQ_SURROUND:
- spec->private_dac_nids[i] = 0x11;
- break;
- }
- }
- }
-
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- char name[32];
- static const char * const chname[3] = {
- "Front", "Surround", "C/LFE"
- };
- hda_nid_t nid_vols[] = {0x10, 0x11, 0x25};
- hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27};
- hda_nid_t nid, nid_vol, nid_mute;
- int i, err;
-
- for (i = 0; i <= AUTO_SEQ_CENLFE; i++) {
- nid = cfg->line_out_pins[i];
-
- if (!nid)
- continue;
-
- nid_vol = nid_vols[i];
- nid_mute = nid_mutes[i];
-
- if (i == AUTO_SEQ_CENLFE) {
- err = via_add_control(
- spec, VIA_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(
- spec, VIA_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (i == AUTO_SEQ_FRONT) {
-
- err = via_add_control(
- spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
-
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- sprintf(name, "%s Playback Volume", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", chname[i]);
- err = via_add_control(
- spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
-
- if (!pin)
- return 0;
-
- spec->multiout.hp_nid = 0x25; /* AOW3 */
- spec->hp_independent_mode_index = 1;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- create_hp_imux(spec);
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt1716S_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- static const hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff };
- return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs,
- ARRAY_SIZE(pin_idxs));
-}
-
-static int vt1716S_parse_auto_config(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int err;
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
- if (err < 0)
- return err;
- err = vt1716S_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
-
- err = vt1716S_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = vt1716S_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- fill_dig_outs(codec);
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- via_smart51_build(spec);
-
- return 1;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1716S_loopbacks[] = {
- { 0x16, HDA_INPUT, 1 },
- { 0x16, HDA_INPUT, 2 },
- { 0x16, HDA_INPUT, 3 },
- { 0x16, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -5228,35 +3484,18 @@
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x16;
+ override_mic_boost(codec, 0x1a, 0, 3, 40);
+ override_mic_boost(codec, 0x1e, 0, 3, 40);
+
/* automatic parse from the BIOS config */
- err = vt1716S_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
}
- spec->init_verbs[spec->num_iverbs++] = vt1716S_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1716S_uniwill_init_verbs;
-
- spec->stream_name_analog = "VT1716S Analog";
- spec->stream_analog_playback = &vt1716S_pcm_analog_playback;
- spec->stream_analog_capture = &vt1716S_pcm_analog_capture;
-
- spec->stream_name_digital = "VT1716S Digital";
- spec->stream_digital_playback = &vt1716S_pcm_digital_playback;
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1716S_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1716S_adc_nids);
- get_mux_nids(codec);
- override_mic_boost(codec, 0x1a, 0, 3, 40);
- override_mic_boost(codec, 0x1e, 0, 3, 40);
- spec->mixers[spec->num_mixers] = vt1716S_capture_mixer;
- spec->num_mixers++;
- }
+ spec->init_verbs[spec->num_iverbs++] = vt1716S_init_verbs;
spec->mixers[spec->num_mixers] = vt1716s_dmic_mixer;
spec->num_mixers++;
@@ -5265,353 +3504,31 @@
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1716S_loopbacks;
-#endif
-
spec->set_widgets_power_state = set_widgets_power_state_vt1716S;
return 0;
}
/* for vt2002P */
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt2002P_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0,
- HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb vt2002P_volume_init_verbs[] = {
+static const struct hda_verb vt2002P_init_verbs[] = {
/* Class-D speaker related verbs */
{0x1, 0xfe0, 0x4},
{0x1, 0xfe9, 0x80},
{0x1, 0xfe2, 0x22},
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* MUX Indices: Mic = 0 */
- {0x1e, AC_VERB_SET_CONNECT_SEL, 0},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* PW9 Output enable */
- {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
-
/* Enable Boost Volume backdoor */
{0x1, 0xfb9, 0x24},
-
- /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* set MUX0/1/4/8 = 0 (AOW0) */
- {0x34, AC_VERB_SET_CONNECT_SEL, 0},
- {0x35, AC_VERB_SET_CONNECT_SEL, 0},
- {0x37, AC_VERB_SET_CONNECT_SEL, 0},
- {0x3b, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* set PW0 index=0 (MW0) */
- {0x24, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Enable AOW0 to MW9 */
- {0x1, 0xfb8, 0x88},
- { }
-};
-static const struct hda_verb vt1802_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* MUX Indices: Mic = 0 */
- {0x1e, AC_VERB_SET_CONNECT_SEL, 0},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* PW9 Output enable */
- {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
-
- /* Enable Boost Volume backdoor */
- {0x1, 0xfb9, 0x24},
-
- /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* set MUX0/1/4/8 = 0 (AOW0) */
- {0x34, AC_VERB_SET_CONNECT_SEL, 0},
- {0x35, AC_VERB_SET_CONNECT_SEL, 0},
- {0x38, AC_VERB_SET_CONNECT_SEL, 0},
- {0x3c, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* set PW0 index=0 (MW0) */
- {0x24, AC_VERB_SET_CONNECT_SEL, 0},
-
/* Enable AOW0 to MW9 */
{0x1, 0xfb8, 0x88},
{ }
};
-
-static const struct hda_verb vt2002P_uniwill_init_verbs[] = {
- {0x25, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
- {0x26, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
- {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+static const struct hda_verb vt1802_init_verbs[] = {
+ /* Enable Boost Volume backdoor */
+ {0x1, 0xfb9, 0x24},
+ /* Enable AOW0 to MW9 */
+ {0x1, 0xfb8, 0x88},
{ }
};
-static const struct hda_verb vt1802_uniwill_init_verbs[] = {
- {0x25, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
- {0x28, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
- {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static const struct hda_pcm_stream vt2002P_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x8, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt2002P_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_pcm_open_close,
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt2002P_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare,
- .cleanup = via_dig_playback_pcm_cleanup
- },
-};
-
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt2002P_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = spec->private_dac_nids;
- if (cfg->line_out_pins[0])
- spec->private_dac_nids[0] = 0x8;
- return 0;
-}
-
-/* add playback controls from the parsed DAC table */
-static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int err;
- hda_nid_t sw_nid;
-
- if (!cfg->line_out_pins[0])
- return -1;
-
- if (spec->codec_type == VT1802)
- sw_nid = 0x28;
- else
- sw_nid = 0x26;
-
- /* Line-Out: PortE */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
- "Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(sw_nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- return 0;
-}
-
-static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
-
- if (!pin)
- return 0;
-
- spec->multiout.hp_nid = 0x9;
- spec->hp_independent_mode_index = 1;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(
- spec->multiout.hp_nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- create_hp_imux(spec);
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt2002P_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- struct via_spec *spec = codec->spec;
- struct hda_input_mux *imux = &spec->private_imux[0];
- static const hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0xff };
- int err;
-
- err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs,
- ARRAY_SIZE(pin_idxs));
- if (err < 0)
- return err;
- /* build volume/mute control of loopback */
- err = via_new_analog_input(spec, "Stereo Mixer", 0, 3, 0x21);
- if (err < 0)
- return err;
-
- /* for digital mic select */
- snd_hda_add_imux_item(imux, "Digital Mic", 4, NULL);
-
- return 0;
-}
-
-static int vt2002P_parse_auto_config(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int err;
-
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
- if (err < 0)
- return err;
-
- err = vt2002P_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
-
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
- return 0; /* can't find valid BIOS pin config */
-
- err = vt2002P_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = vt2002P_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- fill_dig_outs(codec);
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- return 1;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt2002P_loopbacks[] = {
- { 0x21, HDA_INPUT, 0 },
- { 0x21, HDA_INPUT, 1 },
- { 0x21, HDA_INPUT, 2 },
- { } /* end */
-};
-#endif
static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
{
@@ -5735,334 +3652,39 @@
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x21;
+ override_mic_boost(codec, 0x2b, 0, 3, 40);
+ override_mic_boost(codec, 0x29, 0, 3, 40);
+ add_secret_dac_path(codec);
+
/* automatic parse from the BIOS config */
- err = vt2002P_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
}
if (spec->codec_type == VT1802)
- spec->init_verbs[spec->num_iverbs++] =
- vt1802_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1802_init_verbs;
else
- spec->init_verbs[spec->num_iverbs++] =
- vt2002P_volume_init_verbs;
-
- if (spec->codec_type == VT1802)
- spec->init_verbs[spec->num_iverbs++] =
- vt1802_uniwill_init_verbs;
- else
- spec->init_verbs[spec->num_iverbs++] =
- vt2002P_uniwill_init_verbs;
-
- if (spec->codec_type == VT1802)
- spec->stream_name_analog = "VT1802 Analog";
- else
- spec->stream_name_analog = "VT2002P Analog";
- spec->stream_analog_playback = &vt2002P_pcm_analog_playback;
- spec->stream_analog_capture = &vt2002P_pcm_analog_capture;
-
- if (spec->codec_type == VT1802)
- spec->stream_name_digital = "VT1802 Digital";
- else
- spec->stream_name_digital = "VT2002P Digital";
- spec->stream_digital_playback = &vt2002P_pcm_digital_playback;
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt2002P_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt2002P_adc_nids);
- get_mux_nids(codec);
- override_mic_boost(codec, 0x2b, 0, 3, 40);
- override_mic_boost(codec, 0x29, 0, 3, 40);
- spec->mixers[spec->num_mixers] = vt2002P_capture_mixer;
- spec->num_mixers++;
- }
+ spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs;
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt2002P_loopbacks;
-#endif
-
spec->set_widgets_power_state = set_widgets_power_state_vt2002P;
return 0;
}
/* for vt1812 */
-/* capture mixer elements */
-static const struct snd_kcontrol_new vt1812_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Boost Capture Volume", 0x29, 0x0,
- HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- .name = "Input Source",
- .count = 2,
- .info = via_mux_enum_info,
- .get = via_mux_enum_get,
- .put = via_mux_enum_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb vt1812_volume_init_verbs[] = {
- /*
- * Unmute ADC0-1 and set the default input to mic-in
- */
- {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- */
- /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* MUX Indices: Mic = 0 */
- {0x1e, AC_VERB_SET_CONNECT_SEL, 0},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* PW9 Output enable */
- {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
-
+static const struct hda_verb vt1812_init_verbs[] = {
/* Enable Boost Volume backdoor */
{0x1, 0xfb9, 0x24},
-
- /* MW0/1/4/13/15: un-mute index 0 (MUXx), un-mute index 1 (MW9) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* set MUX0/1/4/13/15 = 0 (AOW0) */
- {0x34, AC_VERB_SET_CONNECT_SEL, 0},
- {0x35, AC_VERB_SET_CONNECT_SEL, 0},
- {0x38, AC_VERB_SET_CONNECT_SEL, 0},
- {0x3c, AC_VERB_SET_CONNECT_SEL, 0},
- {0x3d, AC_VERB_SET_CONNECT_SEL, 0},
-
/* Enable AOW0 to MW9 */
{0x1, 0xfb8, 0xa8},
{ }
};
-
-static const struct hda_verb vt1812_uniwill_init_verbs[] = {
- {0x33, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
- {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT },
- {0x28, AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
- {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
- { }
-};
-
-static const struct hda_pcm_stream vt1812_pcm_analog_playback = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x8, /* NID to query formats and rates */
- .ops = {
- .open = via_playback_pcm_open,
- .prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt1812_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = 0x10, /* NID to query formats and rates */
- .ops = {
- .open = via_pcm_open_close,
- .prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup,
- .close = via_pcm_open_close,
- },
-};
-
-static const struct hda_pcm_stream vt1812_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- /* NID is set in via_build_pcms */
- .ops = {
- .open = via_dig_playback_pcm_open,
- .close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare,
- .cleanup = via_dig_playback_pcm_cleanup
- },
-};
-/* fill in the dac_nids table from the parsed pin configuration */
-static int vt1812_auto_fill_dac_nids(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- spec->multiout.num_dacs = 1;
- spec->multiout.dac_nids = spec->private_dac_nids;
- if (cfg->line_out_pins[0])
- spec->private_dac_nids[0] = 0x8;
- return 0;
-}
-
-
-/* add playback controls from the parsed DAC table */
-static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int err;
-
- if (!cfg->line_out_pins[0])
- return -1;
-
- /* Line-Out: PortE */
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
- "Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- return 0;
-}
-
-static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
-{
- int err;
-
- if (!pin)
- return 0;
-
- spec->multiout.hp_nid = 0x9;
- spec->hp_independent_mode_index = 1;
-
-
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(
- spec->multiout.hp_nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
-
- create_hp_imux(spec);
- return 0;
-}
-
-/* create playback/capture controls for input pins */
-static int vt1812_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- struct via_spec *spec = codec->spec;
- struct hda_input_mux *imux = &spec->private_imux[0];
- static const hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0, 0, 0xff };
- int err;
-
- err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs,
- ARRAY_SIZE(pin_idxs));
- if (err < 0)
- return err;
-
- /* build volume/mute control of loopback */
- err = via_new_analog_input(spec, "Stereo Mixer", 0, 5, 0x21);
- if (err < 0)
- return err;
-
- /* for digital mic select */
- snd_hda_add_imux_item(imux, "Digital Mic", 6, NULL);
-
- return 0;
-}
-
-static int vt1812_parse_auto_config(struct hda_codec *codec)
-{
- struct via_spec *spec = codec->spec;
- int err;
-
-
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
- if (err < 0)
- return err;
- fill_dig_outs(codec);
- err = vt1812_auto_fill_dac_nids(spec, &spec->autocfg);
- if (err < 0)
- return err;
-
- if (!spec->autocfg.line_outs && !spec->autocfg.hp_outs)
- return 0; /* can't find valid BIOS pin config */
-
- err = vt1812_auto_create_multi_out_ctls(spec, &spec->autocfg);
- if (err < 0)
- return err;
- err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
- if (err < 0)
- return err;
- err = vt1812_auto_create_analog_input_ctls(codec, &spec->autocfg);
- if (err < 0)
- return err;
-
- spec->multiout.max_channels = spec->multiout.num_dacs * 2;
-
- fill_dig_outs(codec);
-
- if (spec->kctls.list)
- spec->mixers[spec->num_mixers++] = spec->kctls.list;
-
- spec->input_mux = &spec->private_imux[0];
-
- if (spec->hp_mux)
- via_hp_build(codec);
-
- return 1;
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list vt1812_loopbacks[] = {
- { 0x21, HDA_INPUT, 0 },
- { 0x21, HDA_INPUT, 1 },
- { 0x21, HDA_INPUT, 2 },
- { } /* end */
-};
-#endif
-
static void set_widgets_power_state_vt1812(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -6166,47 +3788,22 @@
if (spec == NULL)
return -ENOMEM;
+ spec->aa_mix_nid = 0x21;
+ override_mic_boost(codec, 0x2b, 0, 3, 40);
+ override_mic_boost(codec, 0x29, 0, 3, 40);
+ add_secret_dac_path(codec);
+
/* automatic parse from the BIOS config */
- err = vt1812_parse_auto_config(codec);
+ err = via_parse_auto_config(codec);
if (err < 0) {
via_free(codec);
return err;
- } else if (!err) {
- printk(KERN_INFO "hda_codec: Cannot set up configuration "
- "from BIOS. Using genenic mode...\n");
}
-
- spec->init_verbs[spec->num_iverbs++] = vt1812_volume_init_verbs;
- spec->init_verbs[spec->num_iverbs++] = vt1812_uniwill_init_verbs;
-
- spec->stream_name_analog = "VT1812 Analog";
- spec->stream_analog_playback = &vt1812_pcm_analog_playback;
- spec->stream_analog_capture = &vt1812_pcm_analog_capture;
-
- spec->stream_name_digital = "VT1812 Digital";
- spec->stream_digital_playback = &vt1812_pcm_digital_playback;
-
-
- if (!spec->adc_nids && spec->input_mux) {
- spec->adc_nids = vt1812_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(vt1812_adc_nids);
- get_mux_nids(codec);
- override_mic_boost(codec, 0x2b, 0, 3, 40);
- override_mic_boost(codec, 0x29, 0, 3, 40);
- spec->mixers[spec->num_mixers] = vt1812_capture_mixer;
- spec->num_mixers++;
- }
+ spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs;
codec->patch_ops = via_patch_ops;
- codec->patch_ops.init = via_auto_init;
- codec->patch_ops.unsol_event = via_unsol_event;
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->loopback.amplist = vt1812_loopbacks;
-#endif
-
spec->set_widgets_power_state = set_widgets_power_state_vt1812;
return 0;
}
@@ -6220,37 +3817,37 @@
{ .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708},
{ .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708},
{ .id = 0x1106e710, .name = "VT1709 10-Ch",
- .patch = patch_vt1709_10ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e711, .name = "VT1709 10-Ch",
- .patch = patch_vt1709_10ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e712, .name = "VT1709 10-Ch",
- .patch = patch_vt1709_10ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e713, .name = "VT1709 10-Ch",
- .patch = patch_vt1709_10ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e714, .name = "VT1709 6-Ch",
- .patch = patch_vt1709_6ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e715, .name = "VT1709 6-Ch",
- .patch = patch_vt1709_6ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e716, .name = "VT1709 6-Ch",
- .patch = patch_vt1709_6ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e717, .name = "VT1709 6-Ch",
- .patch = patch_vt1709_6ch},
+ .patch = patch_vt1709},
{ .id = 0x1106e720, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B_8ch},
+ .patch = patch_vt1708B},
{ .id = 0x1106e721, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B_8ch},
+ .patch = patch_vt1708B},
{ .id = 0x1106e722, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B_8ch},
+ .patch = patch_vt1708B},
{ .id = 0x1106e723, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B_8ch},
+ .patch = patch_vt1708B},
{ .id = 0x1106e724, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B_4ch},
+ .patch = patch_vt1708B},
{ .id = 0x1106e725, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B_4ch},
+ .patch = patch_vt1708B},
{ .id = 0x1106e726, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B_4ch},
+ .patch = patch_vt1708B},
{ .id = 0x1106e727, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B_4ch},
+ .patch = patch_vt1708B},
{ .id = 0x11060397, .name = "VT1708S",
.patch = patch_vt1708S},
{ .id = 0x11061397, .name = "VT1708S",
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index f4594d7..be06fb3 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2607,7 +2607,7 @@
ice->profi_port = pci_resource_start(pci, 3);
if (request_irq(pci->irq, snd_ice1712_interrupt, IRQF_SHARED,
- "ICE1712", ice)) {
+ KBUILD_MODNAME, ice)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_ice1712_free(ice);
return -EIO;
@@ -2802,7 +2802,7 @@
}
static struct pci_driver driver = {
- .name = "ICE1712",
+ .name = KBUILD_MODNAME,
.id_table = snd_ice1712_ids,
.probe = snd_ice1712_probe,
.remove = __devexit_p(snd_ice1712_remove),
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index c1498fa..c2b7f8b 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2509,7 +2509,7 @@
ice->profi_port = pci_resource_start(pci, 1);
if (request_irq(pci->irq, snd_vt1724_interrupt,
- IRQF_SHARED, "ICE1724", ice)) {
+ IRQF_SHARED, KBUILD_MODNAME, ice)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_vt1724_free(ice);
return -EIO;
@@ -2802,7 +2802,7 @@
#endif
static struct pci_driver driver = {
- .name = "ICE1724",
+ .name = KBUILD_MODNAME,
.id_table = snd_vt1724_ids,
.probe = snd_vt1724_probe,
.remove = __devexit_p(snd_vt1724_remove),
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 6c896db..6a5b387 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1884,6 +1884,12 @@
},
{
.subvendor = 0x1028,
+ .subdevice = 0x0189,
+ .name = "Dell Inspiron 9300",
+ .type = AC97_TUNE_HP_MUTE_LED
+ },
+ {
+ .subvendor = 0x1028,
.subdevice = 0x0191,
.name = "Dell Inspiron 8600",
.type = AC97_TUNE_HP_ONLY
@@ -2647,7 +2653,7 @@
pci_set_master(pci);
snd_intel8x0_chip_init(chip, 0);
if (request_irq(pci->irq, snd_intel8x0_interrupt,
- IRQF_SHARED, card->shortname, chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "intel8x0: unable to grab IRQ %d, "
"disabling device\n", pci->irq);
snd_card_disconnect(card);
@@ -3106,7 +3112,7 @@
/* request irq after initializaing int_sta_mask, etc */
if (request_irq(pci->irq, snd_intel8x0_interrupt,
- IRQF_SHARED, card->shortname, chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_intel8x0_free(chip);
return -EBUSY;
@@ -3266,7 +3272,7 @@
}
static struct pci_driver driver = {
- .name = "Intel ICH",
+ .name = KBUILD_MODNAME,
.id_table = snd_intel8x0_ids,
.probe = snd_intel8x0_probe,
.remove = __devexit_p(snd_intel8x0_remove),
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index f3353b4..7c16164 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1047,7 +1047,7 @@
}
pci_set_master(pci);
if (request_irq(pci->irq, snd_intel8x0m_interrupt,
- IRQF_SHARED, card->shortname, chip)) {
+ IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "intel8x0m: unable to grab IRQ %d, "
"disabling device\n", pci->irq);
snd_card_disconnect(card);
@@ -1174,7 +1174,7 @@
port_inited:
if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
- card->shortname, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_intel8x0m_free(chip);
return -EBUSY;
@@ -1325,7 +1325,7 @@
}
static struct pci_driver driver = {
- .name = "Intel ICH Modem",
+ .name = KBUILD_MODNAME,
.id_table = snd_intel8x0m_ids,
.probe = snd_intel8x0m_probe,
.remove = __devexit_p(snd_intel8x0m_remove),
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 6d79570..fc1d573 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2241,7 +2241,7 @@
err = request_irq(pci->irq, snd_korg1212_interrupt,
IRQF_SHARED,
- "korg1212", korg1212);
+ KBUILD_MODNAME, korg1212);
if (err) {
snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq);
@@ -2477,7 +2477,7 @@
}
static struct pci_driver driver = {
- .name = "korg1212",
+ .name = KBUILD_MODNAME,
.id_table = snd_korg1212_ids,
.probe = snd_korg1212_probe,
.remove = __devexit_p(snd_korg1212_remove),
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 2692e5a..3e92e5b 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -648,7 +648,7 @@
goto errout;
if (request_irq(pci->irq, lola_interrupt, IRQF_SHARED,
- DRVNAME, chip)) {
+ KBUILD_MODNAME, chip)) {
printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq);
err = -EBUSY;
goto errout;
@@ -771,7 +771,7 @@
/* pci_driver definition */
static struct pci_driver driver = {
- .name = DRVNAME,
+ .name = KBUILD_MODNAME,
.id_table = lola_ids,
.probe = lola_probe,
.remove = __devexit_p(lola_remove),
diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h
index d5708e2..f0b1000 100644
--- a/sound/pci/lola/lola.h
+++ b/sound/pci/lola/lola.h
@@ -480,7 +480,7 @@
/* count values in the Vendor Specific Mixer Widget's Audio Widget Capabilities */
#define LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(res) ((res >> 2) & 0x1f)
-#define LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(res) ((res >> 7) & 0x1f)
+#define LOLA_MIXER_DEST_REC_OUTPUT_SEPARATION(res) ((res >> 7) & 0x1f)
int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb,
unsigned int data, unsigned int extdata);
diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c
index 5d518f1..6b8d648 100644
--- a/sound/pci/lola/lola_mixer.c
+++ b/sound/pci/lola/lola_mixer.c
@@ -144,40 +144,61 @@
chip->mixer.dest_stream_ins = chip->pcm[CAPT].num_streams;
chip->mixer.dest_phys_outs = chip->pin[PLAY].num_pins;
- /* mixer matrix can have unused areas between PhysIn and
+ /* mixer matrix may have unused areas between PhysIn and
* Play or Record and PhysOut zones
*/
chip->mixer.src_stream_out_ofs = chip->mixer.src_phys_ins +
LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(val);
chip->mixer.dest_phys_out_ofs = chip->mixer.dest_stream_ins +
- LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(val);
+ LOLA_MIXER_DEST_REC_OUTPUT_SEPARATION(val);
- /* example : MixerMatrix of LoLa881
- * 0-------8------16-------8------16
- * | | | | |
- * | INPUT | | INPUT | |
- * | -> |unused | -> |unused |
- * | RECORD| | OUTPUT| |
- * | | | | |
- * 8--------------------------------
- * | | | | |
- * | | | | |
- * |unused |unused |unused |unused |
- * | | | | |
- * | | | | |
- * 16-------------------------------
- * | | | | |
- * | PLAY | | PLAY | |
- * | -> |unused | -> |unused |
- * | RECORD| | OUTPUT| |
- * | | | | |
- * 8--------------------------------
- * | | | | |
- * | | | | |
- * |unused |unused |unused |unused |
- * | | | | |
- * | | | | |
- * 16-------------------------------
+ /* example : MixerMatrix of LoLa881 (LoLa16161 uses unused zones)
+ * +-+ 0-------8------16-------8------16
+ * | | | | | | |
+ * |s| | INPUT | | INPUT | |
+ * | |->| -> |unused | -> |unused |
+ * |r| |CAPTURE| | OUTPUT| |
+ * | | | MIX | | MIX | |
+ * |c| 8--------------------------------
+ * | | | | | | |
+ * | | | | | | |
+ * |g| |unused |unused |unused |unused |
+ * | | | | | | |
+ * |a| | | | | |
+ * | | 16-------------------------------
+ * |i| | | | | |
+ * | | | PLAYBK| | PLAYBK| |
+ * |n|->| -> |unused | -> |unused |
+ * | | |CAPTURE| | OUTPUT| |
+ * | | | MIX | | MIX | |
+ * |a| 8--------------------------------
+ * |r| | | | | |
+ * |r| | | | | |
+ * |a| |unused |unused |unused |unused |
+ * |y| | | | | |
+ * | | | | | | |
+ * +++ 16--|---------------|------------
+ * +---V---------------V-----------+
+ * | dest_mix_gain_enable array |
+ * +-------------------------------+
+ */
+ /* example : MixerMatrix of LoLa280
+ * +-+ 0-------8-2
+ * | | | | |
+ * |s| | INPUT | | INPUT
+ * |r|->| -> | | ->
+ * |c| |CAPTURE| | <- OUTPUT
+ * | | | MIX | | MIX
+ * |g| 8----------
+ * |a| | | |
+ * |i| | PLAYBK| | PLAYBACK
+ * |n|->| -> | | ->
+ * | | |CAPTURE| | <- OUTPUT
+ * |a| | MIX | | MIX
+ * |r| 8---|----|-
+ * |r| +---V----V-------------------+
+ * |a| | dest_mix_gain_enable array |
+ * |y| +----------------------------+
*/
if (chip->mixer.src_stream_out_ofs > MAX_AUDIO_INOUT_COUNT ||
chip->mixer.dest_phys_out_ofs > MAX_STREAM_IN_COUNT) {
@@ -192,6 +213,9 @@
(((1U << chip->mixer.dest_phys_outs) - 1)
<< chip->mixer.dest_phys_out_ofs);
+ snd_printdd("Mixer src_mask=%x, dest_mask=%x\n",
+ chip->mixer.src_mask, chip->mixer.dest_mask);
+
return 0;
}
@@ -202,12 +226,19 @@
if (!(chip->mixer.src_mask & (1 << id)))
return -EINVAL;
- writew(gain, &chip->mixer.array->src_gain[id]);
oldval = val = readl(&chip->mixer.array->src_gain_enable);
if (on)
val |= (1 << id);
else
val &= ~(1 << id);
+ /* test if values unchanged */
+ if ((val == oldval) &&
+ (gain == readw(&chip->mixer.array->src_gain[id])))
+ return 0;
+
+ snd_printdd("lola_mixer_set_src_gain (id=%d, gain=%d) enable=%x\n",
+ id, gain, val);
+ writew(gain, &chip->mixer.array->src_gain[id]);
writel(val, &chip->mixer.array->src_gain_enable);
lola_codec_flush(chip);
/* inform micro-controller about the new source gain */
@@ -269,6 +300,7 @@
src, dest);
}
+#if 0 /* not used */
static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id,
unsigned int mask, unsigned short *gains)
{
@@ -289,6 +321,7 @@
return lola_codec_write(chip, chip->mixer.nid,
LOLA_VERB_SET_DESTINATION_GAIN, id, 0);
}
+#endif /* not used */
/*
*/
@@ -376,6 +409,8 @@
return 0;
if (external_call)
lola_codec_flush(chip);
+ snd_printdd("set_analog_volume (dir=%d idx=%d, volume=%d)\n",
+ dir, idx, val);
err = lola_codec_write(chip, pin->nid,
LOLA_VERB_SET_AMP_GAIN_MUTE, val, 0);
if (err < 0)
@@ -427,23 +462,40 @@
{
int i;
- /* all src on */
+ /* all sample rate converters on */
lola_set_src_config(chip, (1 << chip->pin[CAPT].num_pins) - 1, false);
- /* clear all matrix */
+ /* clear all mixer matrix settings */
memset_io(chip->mixer.array, 0, sizeof(*chip->mixer.array));
- /* set src gain to 0dB */
+ /* inform firmware about all updated matrix columns - capture part */
+ for (i = 0; i < chip->mixer.dest_stream_ins; i++)
+ lola_codec_write(chip, chip->mixer.nid,
+ LOLA_VERB_SET_DESTINATION_GAIN,
+ i, 0);
+ /* inform firmware about all updated matrix columns - output part */
+ for (i = 0; i < chip->mixer.dest_phys_outs; i++)
+ lola_codec_write(chip, chip->mixer.nid,
+ LOLA_VERB_SET_DESTINATION_GAIN,
+ chip->mixer.dest_phys_out_ofs + i, 0);
+
+ /* set all digital input source (master) gains to 0dB */
for (i = 0; i < chip->mixer.src_phys_ins; i++)
lola_mixer_set_src_gain(chip, i, 336, true); /* 0dB */
+
+ /* set all digital playback source (master) gains to 0dB */
for (i = 0; i < chip->mixer.src_stream_outs; i++)
lola_mixer_set_src_gain(chip,
i + chip->mixer.src_stream_out_ofs,
336, true); /* 0dB */
- /* set 1:1 dest gain */
+ /* set gain value 0dB diagonally in matrix - part INPUT -> CAPTURE */
for (i = 0; i < chip->mixer.dest_stream_ins; i++) {
int src = i % chip->mixer.src_phys_ins;
lola_mixer_set_mapping_gain(chip, src, i, 336, true);
}
+ /* set gain value 0dB diagonally in matrix , part PLAYBACK -> OUTPUT
+ * (LoLa280 : playback channel 0,2,4,6 linked to output channel 0)
+ * (LoLa280 : playback channel 1,3,5,7 linked to output channel 1)
+ */
for (i = 0; i < chip->mixer.src_stream_outs; i++) {
int src = chip->mixer.src_stream_out_ofs + i;
int dst = chip->mixer.dest_phys_out_ofs +
@@ -693,6 +745,7 @@
snd_ctl_new1(&lola_src_gain_mixer, chip));
}
+#if 0 /* not used */
/*
* destination gain (matrix-like) mixer
*/
@@ -781,6 +834,7 @@
return snd_ctl_add(chip->card,
snd_ctl_new1(&lola_dest_gain_mixer, chip));
}
+#endif /* not used */
/*
*/
@@ -798,14 +852,16 @@
if (err < 0)
return err;
err = create_src_gain_mixer(chip, chip->mixer.src_phys_ins, 0,
- "Line Source Gain Volume");
+ "Digital Capture Volume");
if (err < 0)
return err;
err = create_src_gain_mixer(chip, chip->mixer.src_stream_outs,
chip->mixer.src_stream_out_ofs,
- "Stream Source Gain Volume");
+ "Digital Playback Volume");
if (err < 0)
return err;
+#if 0
+/* FIXME: buggy mixer matrix handling */
err = create_dest_gain_mixer(chip,
chip->mixer.src_phys_ins, 0,
chip->mixer.dest_stream_ins, 0,
@@ -834,6 +890,6 @@
"Stream Playback Volume");
if (err < 0)
return err;
-
+#endif /* FIXME */
return init_mixer_values(chip);
}
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index 1bd7a54..04ae84b 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -762,7 +762,6 @@
static int __devinit lx_init_dsp(struct lx6464es *chip)
{
int err;
- u8 mac_address[6];
int i;
snd_printdd("->lx_init_dsp\n");
@@ -787,11 +786,11 @@
/** \todo the mac address should be ready by not, but it isn't,
* so we wait for it */
for (i = 0; i != 1000; ++i) {
- err = lx_dsp_get_mac(chip, mac_address);
+ err = lx_dsp_get_mac(chip);
if (err)
return err;
- if (mac_address[0] || mac_address[1] || mac_address[2] ||
- mac_address[3] || mac_address[4] || mac_address[5])
+ if (chip->mac_address[0] || chip->mac_address[1] || chip->mac_address[2] ||
+ chip->mac_address[3] || chip->mac_address[4] || chip->mac_address[5])
goto mac_ready;
msleep(1);
}
@@ -800,8 +799,8 @@
mac_ready:
snd_printd(LXP "mac address ready read after: %dms\n", i);
snd_printk(LXP "mac address: %02X.%02X.%02X.%02X.%02X.%02X\n",
- mac_address[0], mac_address[1], mac_address[2],
- mac_address[3], mac_address[4], mac_address[5]);
+ chip->mac_address[0], chip->mac_address[1], chip->mac_address[2],
+ chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
err = lx_init_get_version_features(chip);
if (err)
@@ -1031,7 +1030,7 @@
chip->port_dsp_bar = pci_ioremap_bar(pci, 2);
err = request_irq(pci->irq, lx_interrupt, IRQF_SHARED,
- card_name, chip);
+ KBUILD_MODNAME, chip);
if (err) {
snd_printk(KERN_ERR LXP "unable to grab IRQ %d\n", pci->irq);
goto request_irq_failed;
@@ -1108,8 +1107,14 @@
goto out_free;
}
- strcpy(card->driver, "lx6464es");
- strcpy(card->shortname, "Digigram LX6464ES");
+ strcpy(card->driver, "LX6464ES");
+ sprintf(card->id, "LX6464ES_%02X%02X%02X",
+ chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
+
+ sprintf(card->shortname, "LX6464ES %02X.%02X.%02X.%02X.%02X.%02X",
+ chip->mac_address[0], chip->mac_address[1], chip->mac_address[2],
+ chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
+
sprintf(card->longname, "%s at 0x%lx, 0x%p, irq %i",
card->shortname, chip->port_plx,
chip->port_dsp_bar, chip->irq);
@@ -1137,7 +1142,7 @@
static struct pci_driver driver = {
- .name = "Digigram LX6464ES",
+ .name = KBUILD_MODNAME,
.id_table = snd_lx6464es_ids,
.probe = snd_lx6464es_probe,
.remove = __devexit_p(snd_lx6464es_remove),
diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h
index aea621e..e2a124a 100644
--- a/sound/pci/lx6464es/lx6464es.h
+++ b/sound/pci/lx6464es/lx6464es.h
@@ -69,6 +69,8 @@
struct pci_dev *pci;
int irq;
+ u8 mac_address[6];
+
spinlock_t lock; /* interrupt spinlock */
struct mutex setup_mutex; /* mutex used in hw_params, open
* and close */
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index 617f98b..5c8717e 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -424,7 +424,7 @@
return ret;
}
-int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address)
+int lx_dsp_get_mac(struct lx6464es *chip)
{
u32 macmsb, maclsb;
@@ -432,12 +432,12 @@
maclsb = lx_dsp_reg_read(chip, eReg_ADMACESLSB) & 0x00FFFFFF;
/* todo: endianess handling */
- mac_address[5] = ((u8 *)(&maclsb))[0];
- mac_address[4] = ((u8 *)(&maclsb))[1];
- mac_address[3] = ((u8 *)(&maclsb))[2];
- mac_address[2] = ((u8 *)(&macmsb))[0];
- mac_address[1] = ((u8 *)(&macmsb))[1];
- mac_address[0] = ((u8 *)(&macmsb))[2];
+ chip->mac_address[5] = ((u8 *)(&maclsb))[0];
+ chip->mac_address[4] = ((u8 *)(&maclsb))[1];
+ chip->mac_address[3] = ((u8 *)(&maclsb))[2];
+ chip->mac_address[2] = ((u8 *)(&macmsb))[0];
+ chip->mac_address[1] = ((u8 *)(&macmsb))[1];
+ chip->mac_address[0] = ((u8 *)(&macmsb))[2];
return 0;
}
diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h
index 6bd9cbb..1dd5629 100644
--- a/sound/pci/lx6464es/lx_core.h
+++ b/sound/pci/lx6464es/lx_core.h
@@ -116,7 +116,7 @@
int lx_dsp_get_clock_frequency(struct lx6464es *chip, u32 *rfreq);
int lx_dsp_set_granularity(struct lx6464es *chip, u32 gran);
int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data);
-int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address);
+int lx_dsp_get_mac(struct lx6464es *chip);
/* low-level pipe handling */
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 3c40d72..0378126 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -850,11 +850,10 @@
struct input_dev *input_dev;
char phys[64]; /* physical device path */
#else
- spinlock_t ac97_lock;
struct snd_kcontrol *master_switch;
struct snd_kcontrol *master_volume;
- struct tasklet_struct hwvol_tq;
#endif
+ struct work_struct hwvol_work;
unsigned int in_suspend;
@@ -1609,13 +1608,10 @@
(without wrap around) in response to volume button presses and then
generating an interrupt. The pair of counters is stored in bits 1-3 and 5-7
of a byte wide register. The meaning of bits 0 and 4 is unknown. */
-static void snd_m3_update_hw_volume(unsigned long private_data)
+static void snd_m3_update_hw_volume(struct work_struct *work)
{
- struct snd_m3 *chip = (struct snd_m3 *) private_data;
+ struct snd_m3 *chip = container_of(work, struct snd_m3, hwvol_work);
int x, val;
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- unsigned long flags;
-#endif
/* Figure out which volume control button was pushed,
based on differences from the default register
@@ -1645,21 +1641,13 @@
if (!chip->master_switch || !chip->master_volume)
return;
- /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
- spin_lock_irqsave(&chip->ac97_lock, flags);
-
- val = chip->ac97->regs[AC97_MASTER_VOL];
+ val = snd_ac97_read(chip->ac97, AC97_MASTER);
switch (x) {
case 0x88:
/* The counters have not changed, yet we've received a HV
interrupt. According to tests run by various people this
happens when pressing the mute button. */
val ^= 0x8000;
- chip->ac97->regs[AC97_MASTER_VOL] = val;
- outw(val, chip->iobase + CODEC_DATA);
- outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &chip->master_switch->id);
break;
case 0xaa:
/* counters increased by 1 -> volume up */
@@ -1667,11 +1655,6 @@
val--;
if ((val & 0x7f00) > 0)
val -= 0x0100;
- chip->ac97->regs[AC97_MASTER_VOL] = val;
- outw(val, chip->iobase + CODEC_DATA);
- outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &chip->master_volume->id);
break;
case 0x66:
/* counters decreased by 1 -> volume down */
@@ -1679,14 +1662,11 @@
val++;
if ((val & 0x7f00) < 0x1f00)
val += 0x0100;
- chip->ac97->regs[AC97_MASTER_VOL] = val;
- outw(val, chip->iobase + CODEC_DATA);
- outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
- &chip->master_volume->id);
break;
}
- spin_unlock_irqrestore(&chip->ac97_lock, flags);
+ if (snd_ac97_update(chip->ac97, AC97_MASTER, val))
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_switch->id);
#else
if (!chip->input_dev)
return;
@@ -1730,11 +1710,7 @@
return IRQ_NONE;
if (status & HV_INT_PENDING)
-#ifdef CONFIG_SND_MAESTRO3_INPUT
- snd_m3_update_hw_volume((unsigned long)chip);
-#else
- tasklet_schedule(&chip->hwvol_tq);
-#endif
+ schedule_work(&chip->hwvol_work);
/*
* ack an assp int if its running
@@ -2000,24 +1976,14 @@
snd_m3_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
struct snd_m3 *chip = ac97->private_data;
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- unsigned long flags;
-#endif
unsigned short data = 0xffff;
if (snd_m3_ac97_wait(chip))
goto fail;
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- spin_lock_irqsave(&chip->ac97_lock, flags);
-#endif
snd_m3_outb(chip, 0x80 | (reg & 0x7f), CODEC_COMMAND);
if (snd_m3_ac97_wait(chip))
- goto fail_unlock;
+ goto fail;
data = snd_m3_inw(chip, CODEC_DATA);
-fail_unlock:
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- spin_unlock_irqrestore(&chip->ac97_lock, flags);
-#endif
fail:
return data;
}
@@ -2026,20 +1992,11 @@
snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
{
struct snd_m3 *chip = ac97->private_data;
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- unsigned long flags;
-#endif
if (snd_m3_ac97_wait(chip))
return;
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- spin_lock_irqsave(&chip->ac97_lock, flags);
-#endif
snd_m3_outw(chip, val, CODEC_DATA);
snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- spin_unlock_irqrestore(&chip->ac97_lock, flags);
-#endif
}
@@ -2458,6 +2415,7 @@
struct m3_dma *s;
int i;
+ cancel_work_sync(&chip->hwvol_work);
#ifdef CONFIG_SND_MAESTRO3_INPUT
if (chip->input_dev)
input_unregister_device(chip->input_dev);
@@ -2511,6 +2469,7 @@
return 0;
chip->in_suspend = 1;
+ cancel_work_sync(&chip->hwvol_work);
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
snd_pcm_suspend_all(chip->pcm);
snd_ac97_suspend(chip->ac97);
@@ -2667,9 +2626,6 @@
}
spin_lock_init(&chip->reg_lock);
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- spin_lock_init(&chip->ac97_lock);
-#endif
switch (pci->device) {
case PCI_DEVICE_ID_ESS_ALLEGRO:
@@ -2683,6 +2639,7 @@
chip->card = card;
chip->pci = pci;
chip->irq = -1;
+ INIT_WORK(&chip->hwvol_work, snd_m3_update_hw_volume);
chip->external_amp = enable_amp;
if (amp_gpio >= 0 && amp_gpio <= 0x0f)
@@ -2752,12 +2709,8 @@
snd_m3_hv_init(chip);
-#ifndef CONFIG_SND_MAESTRO3_INPUT
- tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip);
-#endif
-
if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED,
- card->driver, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_m3_free(chip);
return -ENOMEM;
@@ -2885,7 +2838,7 @@
}
static struct pci_driver driver = {
- .name = "Maestro3",
+ .name = KBUILD_MODNAME,
.id_table = snd_m3_ids,
.probe = snd_m3_probe,
.remove = __devexit_p(snd_m3_remove),
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 6c3fd4d..dbee599 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1268,7 +1268,7 @@
}
if (request_irq(pci->irq, snd_mixart_interrupt, IRQF_SHARED,
- CARD_NAME, mgr)) {
+ KBUILD_MODNAME, mgr)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_mixart_free(mgr);
return -EBUSY;
@@ -1381,7 +1381,7 @@
}
static struct pci_driver driver = {
- .name = "Digigram miXart",
+ .name = KBUILD_MODNAME,
.id_table = snd_mixart_ids,
.probe = snd_mixart_probe,
.remove = __devexit_p(snd_mixart_remove),
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 5a60492..83ea7a7 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -465,7 +465,7 @@
mutex_lock(&chip->irq_mutex);
if (chip->irq < 0) {
if (request_irq(chip->pci->irq, chip->interrupt, IRQF_SHARED,
- chip->card->driver, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->pci->irq);
mutex_unlock(&chip->irq_mutex);
return -EBUSY;
@@ -1743,7 +1743,7 @@
static struct pci_driver driver = {
- .name = "NeoMagic 256",
+ .name = KBUILD_MODNAME,
.id_table = snd_nm256_ids,
.probe = snd_nm256_probe,
.remove = __devexit_p(snd_nm256_remove),
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index d7e8ddd..218d985 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -859,7 +859,7 @@
}
static struct pci_driver oxygen_driver = {
- .name = "CMI8788",
+ .name = KBUILD_MODNAME,
.id_table = oxygen_ids,
.probe = generic_oxygen_probe,
.remove = __devexit_p(oxygen_pci_remove),
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 70b7398..82311fc 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -655,7 +655,7 @@
chip->model.init(chip);
err = request_irq(pci->irq, oxygen_interrupt, IRQF_SHARED,
- DRIVER, chip);
+ KBUILD_MODNAME, chip);
if (err < 0) {
snd_printk(KERN_ERR "cannot grab interrupt %d\n", pci->irq);
goto err_card;
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
index d5533e3..cc0bcd9 100644
--- a/sound/pci/oxygen/oxygen_pcm.c
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -168,12 +168,6 @@
if (err < 0)
return err;
}
- if (channel == PCM_MULTICH) {
- err = snd_pcm_hw_constraint_minmax
- (runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 0, 8192000);
- if (err < 0)
- return err;
- }
snd_pcm_set_sync(substream);
chip->streams[channel] = substream;
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 469010a..773db79 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -88,7 +88,7 @@
}
static struct pci_driver xonar_driver = {
- .name = "AV200",
+ .name = KBUILD_MODNAME,
.id_table = xonar_ids,
.probe = xonar_probe,
.remove = __devexit_p(oxygen_pci_remove),
diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c
index 54cad38..32d096c 100644
--- a/sound/pci/oxygen/xonar_pcm179x.c
+++ b/sound/pci/oxygen/xonar_pcm179x.c
@@ -327,8 +327,10 @@
{
struct xonar_pcm179x *data = chip->model_data;
- data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = PCM1796_MUTE |
+ data->pcm1796_regs[0][18 - PCM1796_REG_BASE] =
PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD;
+ if (!data->broken_i2c)
+ data->pcm1796_regs[0][18 - PCM1796_REG_BASE] |= PCM1796_MUTE;
data->pcm1796_regs[0][19 - PCM1796_REG_BASE] =
PCM1796_FLT_SHARP | PCM1796_ATS_1;
data->pcm1796_regs[0][20 - PCM1796_REG_BASE] =
@@ -1123,6 +1125,7 @@
chip->model.control_filter = xonar_st_h6_control_filter;
chip->model.dac_channels_pcm = 8;
chip->model.dac_channels_mixer = 8;
+ chip->model.dac_volume_min = 255;
chip->model.dac_mclks = OXYGEN_MCLKS(256, 128, 128);
break;
}
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 95cfde2..046578d 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1501,7 +1501,7 @@
mgr->irq = -1;
if (request_irq(pci->irq, pcxhr_interrupt, IRQF_SHARED,
- card_name, mgr)) {
+ KBUILD_MODNAME, mgr)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
pcxhr_free(mgr);
return -EBUSY;
@@ -1608,7 +1608,7 @@
}
static struct pci_driver driver = {
- .name = "Digigram pcxhr",
+ .name = KBUILD_MODNAME,
.id_table = pcxhr_ids,
.probe = pcxhr_probe,
.remove = __devexit_p(pcxhr_remove),
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index ad5202e..e34ae14 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1890,7 +1890,7 @@
UNSET_AIE(hwport);
if (request_irq(pci->irq, snd_riptide_interrupt, IRQF_SHARED,
- "RIPTIDE", chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "Riptide: unable to grab IRQ %d\n",
pci->irq);
snd_riptide_free(chip);
@@ -2176,7 +2176,7 @@
}
static struct pci_driver driver = {
- .name = "RIPTIDE",
+ .name = KBUILD_MODNAME,
.id_table = snd_riptide_ids,
.probe = snd_card_riptide_probe,
.remove = __devexit_p(snd_card_riptide_remove),
@@ -2188,7 +2188,7 @@
#ifdef SUPPORT_JOYSTICK
static struct pci_driver joystick_driver = {
- .name = "Riptide Joystick",
+ .name = KBUILD_MODNAME "-joystick",
.id_table = snd_riptide_joystick_ids,
.probe = snd_riptide_joystick_probe,
.remove = __devexit_p(snd_riptide_joystick_remove),
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 3c04524..6be77a2 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1355,7 +1355,7 @@
}
if (request_irq(pci->irq, snd_rme32_interrupt, IRQF_SHARED,
- "RME32", rme32)) {
+ KBUILD_MODNAME, rme32)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
return -EBUSY;
}
@@ -1985,7 +1985,7 @@
}
static struct pci_driver driver = {
- .name = "RME Digi32",
+ .name = KBUILD_MODNAME,
.id_table = snd_rme32_ids,
.probe = snd_rme32_probe,
.remove = __devexit_p(snd_rme32_remove),
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 9ff247f..409e5b8 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -1561,7 +1561,7 @@
}
if (request_irq(pci->irq, snd_rme96_interrupt, IRQF_SHARED,
- "RME96", rme96)) {
+ KBUILD_MODNAME, rme96)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
return -EBUSY;
}
@@ -2396,7 +2396,7 @@
}
static struct pci_driver driver = {
- .name = "RME Digi96",
+ .name = KBUILD_MODNAME,
.id_table = snd_rme96_ids,
.probe = snd_rme96_probe,
.remove = __devexit_p(snd_rme96_remove),
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 2d83324..1c6d1e1 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5482,7 +5482,7 @@
}
if (request_irq(pci->irq, snd_hdsp_interrupt, IRQF_SHARED,
- "hdsp", hdsp)) {
+ KBUILD_MODNAME, hdsp)) {
snd_printk(KERN_ERR "Hammerfall-DSP: unable to use IRQ %d\n", pci->irq);
return -EBUSY;
}
@@ -5637,7 +5637,7 @@
}
static struct pci_driver driver = {
- .name = "RME Hammerfall DSP",
+ .name = KBUILD_MODNAME,
.id_table = snd_hdsp_ids,
.probe = snd_hdsp_probe,
.remove = __devexit_p(snd_hdsp_remove),
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index c8e402f..af130ee 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6441,7 +6441,7 @@
hdspm->port + io_extent - 1);
if (request_irq(pci->irq, snd_hdspm_interrupt,
- IRQF_SHARED, "hdspm", hdspm)) {
+ IRQF_SHARED, KBUILD_MODNAME, hdspm)) {
snd_printk(KERN_ERR "HDSPM: unable to use IRQ %d\n", pci->irq);
return -EBUSY;
}
@@ -6779,7 +6779,7 @@
}
static struct pci_driver driver = {
- .name = "RME Hammerfall DSP MADI",
+ .name = KBUILD_MODNAME,
.id_table = snd_hdspm_ids,
.probe = snd_hdspm_probe,
.remove = __devexit_p(snd_hdspm_remove),
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index c492af5..1c7bc1e 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -2479,7 +2479,7 @@
}
if (request_irq(pci->irq, snd_rme9652_interrupt, IRQF_SHARED,
- "rme9652", rme9652)) {
+ KBUILD_MODNAME, rme9652)) {
snd_printk(KERN_ERR "unable to request IRQ %d\n", pci->irq);
return -EBUSY;
}
@@ -2632,7 +2632,7 @@
}
static struct pci_driver driver = {
- .name = "RME Digi9652 (Hammerfall)",
+ .name = KBUILD_MODNAME,
.id_table = snd_rme9652_ids,
.probe = snd_rme9652_probe,
.remove = __devexit_p(snd_rme9652_remove),
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 2b5c7a95..bcf6152 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1235,7 +1235,7 @@
}
if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
- card->shortname, sis)) {
+ KBUILD_MODNAME, sis)) {
printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq);
goto error;
}
@@ -1341,7 +1341,7 @@
goto error_out_cleanup;
if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
- card->shortname, sis)) {
+ KBUILD_MODNAME, sis)) {
printk(KERN_ERR "unable to allocate irq %d\n", sis->irq);
goto error_out_cleanup;
}
@@ -1436,7 +1436,7 @@
}
static struct pci_driver sis7019_driver = {
- .name = "SiS7019",
+ .name = KBUILD_MODNAME,
.id_table = snd_sis7019_ids,
.probe = snd_sis7019_probe,
.remove = __devexit_p(snd_sis7019_remove),
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 337b9fa..2571a67 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1294,7 +1294,7 @@
sonic->game_port = pci_resource_start(pci, 4);
if (request_irq(pci->irq, snd_sonicvibes_interrupt, IRQF_SHARED,
- "S3 SonicVibes", sonic)) {
+ KBUILD_MODNAME, sonic)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_sonicvibes_free(sonic);
return -EBUSY;
@@ -1530,7 +1530,7 @@
}
static struct pci_driver driver = {
- .name = "S3 SonicVibes",
+ .name = KBUILD_MODNAME,
.id_table = snd_sonic_ids,
.probe = snd_sonic_probe,
.remove = __devexit_p(snd_sonic_remove),
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 6d05818..d8a128f 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -172,7 +172,7 @@
}
static struct pci_driver driver = {
- .name = "Trident4DWaveAudio",
+ .name = KBUILD_MODNAME,
.id_table = snd_trident_ids,
.probe = snd_trident_probe,
.remove = __devexit_p(snd_trident_remove),
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 2870a4f..5bd57a7 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -3598,7 +3598,7 @@
trident->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_trident_interrupt, IRQF_SHARED,
- "Trident Audio", trident)) {
+ KBUILD_MODNAME, trident)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_trident_free(trident);
return -EBUSY;
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 8c5f8b5..f03fd62 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2377,7 +2377,7 @@
chip_type == TYPE_VIA8233 ?
snd_via8233_interrupt : snd_via686_interrupt,
IRQF_SHARED,
- card->driver, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_via82xx_free(chip);
return -EBUSY;
@@ -2611,7 +2611,7 @@
}
static struct pci_driver driver = {
- .name = "VIA 82xx Audio",
+ .name = KBUILD_MODNAME,
.id_table = snd_via82xx_ids,
.probe = snd_via82xx_probe,
.remove = __devexit_p(snd_via82xx_remove),
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index f7e8bbbe..a386dd9 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1129,7 +1129,7 @@
}
chip->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_via82xx_interrupt, IRQF_SHARED,
- card->driver, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_via82xx_free(chip);
return -EBUSY;
@@ -1224,7 +1224,7 @@
}
static struct pci_driver driver = {
- .name = "VIA 82xx Modem",
+ .name = KBUILD_MODNAME,
.id_table = snd_via82xx_modem_ids,
.probe = snd_via82xx_probe,
.remove = __devexit_p(snd_via82xx_remove),
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 99a9a81..5342d5e 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -169,7 +169,7 @@
vx->port[i] = pci_resource_start(pci, i + 1);
if (request_irq(pci->irq, snd_vx_irq_handler, IRQF_SHARED,
- CARD_NAME, chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_vx222_free(chip);
return -EBUSY;
@@ -290,7 +290,7 @@
#endif
static struct pci_driver driver = {
- .name = "Digigram VX222",
+ .name = KBUILD_MODNAME,
.id_table = snd_vx222_ids,
.probe = snd_vx222_probe,
.remove = __devexit_p(snd_vx222_remove),
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 80c6821..511d576 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -345,7 +345,7 @@
}
static struct pci_driver driver = {
- .name = "Yamaha DS-1 PCI",
+ .name = KBUILD_MODNAME,
.id_table = snd_ymfpci_ids,
.probe = snd_card_ymfpci_probe,
.remove = __devexit_p(snd_card_ymfpci_remove),
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index c94c051..f3260e6 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -2380,7 +2380,7 @@
return -EBUSY;
}
if (request_irq(pci->irq, snd_ymfpci_interrupt, IRQF_SHARED,
- "YMFPCI", chip)) {
+ KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_ymfpci_free(chip);
return -EBUSY;
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index ce33be0..66488a7 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -223,7 +223,7 @@
if (ret)
goto failed;
- ret = pcmcia_request_exclusive_irq(link, pdacf_interrupt);
+ ret = pcmcia_request_irq(link, pdacf_interrupt);
if (ret)
goto failed;
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index d9ef21d..31777d1 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -229,7 +229,7 @@
if (ret)
goto failed;
- ret = pcmcia_request_exclusive_irq(link, snd_vx_irq_handler);
+ ret = pcmcia_request_irq(link, snd_vx_irq_handler);
if (ret)
goto failed;
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 1ed61c5..4f91387 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,5 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
+snd-soc-core-objs += soc-pcm.o soc-io.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index d0e7532..f81d4c3 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -364,9 +364,11 @@
\*--------------------------------------------------------------------------*/
static u64 atmel_pcm_dmamask = 0xffffffff;
-static int atmel_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
@@ -382,7 +384,7 @@
}
if (dai->driver->capture.channels_min) {
- pr_debug("at32-pcm:"
+ pr_debug("atmel-pcm:"
"Allocating PCM capture DMA buffer\n");
ret = atmel_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h
index 2597329..5e0a95e 100644
--- a/sound/soc/atmel/atmel-pcm.h
+++ b/sound/soc/atmel/atmel-pcm.h
@@ -60,7 +60,7 @@
* This structure, shared between the PCM driver and the interface,
* contains all information required by the PCM driver to perform the
* PDC DMA operation. All fields except dma_intr_handler() are initialized
- * by the interface. The dms_intr_handler() pointer is set by the PCM
+ * by the interface. The dma_intr_handler() pointer is set by the PCM
* driver and called by the interface SSC interrupt handler if it is
* non-NULL.
*/
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index eda955b..7122509 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -402,7 +402,7 @@
if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
&& bits > 16) {
printk(KERN_WARNING
- "atmel_ssc_dai: sample size %d"
+ "atmel_ssc_dai: sample size %d "
"is too large for I2S\n", bits);
return -EINVAL;
}
@@ -838,10 +838,8 @@
}
ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id);
- if (!ssc_pdev) {
- ssc_free(ssc);
+ if (!ssc_pdev)
return -ENOMEM;
- }
/* If we can grab the SSC briefly to parent the DAI device off it */
ssc = ssc_request(ssc_id);
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 95572d2..bad3aa1 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -92,6 +92,7 @@
};
static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
static int mclk_on;
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 10fdd28..20bb53a 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -319,10 +319,11 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int au1xpsc_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ae40359..fe9d548 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -10,13 +10,36 @@
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio support for BF52x ezkit"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
select SND_BF5XX_SOC_I2S
select SND_SOC_SSM2602
- select I2C
help
Say Y if you want to add support for SoC audio on BF527-EZKIT.
+config SND_SOC_BFIN_EVAL_ADAU1701
+ tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
+ depends on SND_BF5XX_I2S
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_ADAU1701
+ select I2C
+ help
+ Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ
+ board connected to one of the Blackfin evaluation boards like the
+ BF5XX-STAMP or BF5XX-EZKIT.
+
+config SND_SOC_BFIN_EVAL_ADAV80X
+ tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
+ depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_ADAV80X
+ help
+ Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
+ EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
+ like the BF5XX-STAMP or BF5XX-EZKIT.
+
+ Note: This driver assumes that the ADAV80X digital record and playback
+ interfaces are connected to the first SPORT port on the BF5XX board.
+
config SND_BF5XX_SOC_AD73311
tristate "SoC AD73311 Audio support for Blackfin"
depends on SND_BF5XX_I2S
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 49af3f3..6018bf5 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -21,9 +21,13 @@
snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
snd-ad193x-objs := bf5xx-ad193x.o
+snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
+snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 98b44b3..9e59f68 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -418,9 +418,11 @@
static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
-int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
pr_debug("%s enter\n", __func__);
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index f1fd95b..61ddf94 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -168,7 +168,7 @@
snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware);
- ret = snd_pcm_hw_constraint_integer(runtime, \
+ ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
goto out;
@@ -257,9 +257,11 @@
static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
-int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
pr_debug("%s enter\n", __func__);
@@ -304,8 +306,8 @@
static struct platform_driver bfin_i2s_pcm_driver = {
.driver = {
- .name = "bfin-i2s-pcm-audio",
- .owner = THIS_MODULE,
+ .name = "bfin-i2s-pcm-audio",
+ .owner = THIS_MODULE,
},
.probe = bfin_i2s_soc_platform_probe,
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c
index 07cfc7a..c95cc03 100644
--- a/sound/soc/blackfin/bf5xx-tdm-pcm.c
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c
@@ -283,9 +283,11 @@
static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
-static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
new file mode 100644
index 0000000..e5550ac
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -0,0 +1,139 @@
+/*
+ * Machine driver for EVAL-ADAU1701MINIZ on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/adau1701.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adau1701_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adau1701_dapm_routes[] = {
+ { "Speaker", NULL, "OUT0" },
+ { "Speaker", NULL, "OUT1" },
+ { "Line Out", NULL, "OUT2" },
+ { "Line Out", NULL, "OUT3" },
+
+ { "IN0", NULL, "Line In" },
+ { "IN1", NULL, "Line In" },
+};
+
+static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000,
+ SND_SOC_CLOCK_IN);
+
+ return ret;
+}
+
+static struct snd_soc_ops bfin_eval_adau1701_ops = {
+ .hw_params = bfin_eval_adau1701_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = {
+ {
+ .name = "adau1701",
+ .stream_name = "adau1701",
+ .cpu_dai_name = "bfin-i2s.0",
+ .codec_dai_name = "adau1701",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .codec_name = "adau1701.0-0034",
+ .ops = &bfin_eval_adau1701_ops,
+ },
+ {
+ .name = "adau1701",
+ .stream_name = "adau1701",
+ .cpu_dai_name = "bfin-i2s.1",
+ .codec_dai_name = "adau1701",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .codec_name = "adau1701.0-0034",
+ .ops = &bfin_eval_adau1701_ops,
+ },
+};
+
+static struct snd_soc_card bfin_eval_adau1701 = {
+ .name = "bfin-eval-adau1701",
+ .dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM],
+ .num_links = 1,
+
+ .dapm_widgets = bfin_eval_adau1701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1701_dapm_widgets),
+ .dapm_routes = bfin_eval_adau1701_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1701_dapm_routes),
+};
+
+static int bfin_eval_adau1701_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &bfin_eval_adau1701;
+
+ card->dev = &pdev->dev;
+
+ return snd_soc_register_card(&bfin_eval_adau1701);
+}
+
+static int __devexit bfin_eval_adau1701_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver bfin_eval_adau1701_driver = {
+ .driver = {
+ .name = "bfin-eval-adau1701",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bfin_eval_adau1701_probe,
+ .remove = __devexit_p(bfin_eval_adau1701_remove),
+};
+
+static int __init bfin_eval_adau1701_init(void)
+{
+ return platform_driver_register(&bfin_eval_adau1701_driver);
+}
+module_init(bfin_eval_adau1701_init);
+
+static void __exit bfin_eval_adau1701_exit(void)
+{
+ platform_driver_unregister(&bfin_eval_adau1701_driver);
+}
+module_exit(bfin_eval_adau1701_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bfin-eval-adau1701");
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
new file mode 100644
index 0000000..8d014d0
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -0,0 +1,173 @@
+/*
+ * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include "../codecs/adav80x.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = {
+ { "Line Out", NULL, "VOUTL" },
+ { "Line Out", NULL, "VOUTR" },
+
+ { "VINL", NULL, "Line In" },
+ { "VINR", NULL, "Line In" },
+};
+
+static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL,
+ 27000000, params_rate(params) * 256);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1,
+ params_rate(params) * 256, SND_SOC_CLOCK_IN);
+
+ return ret;
+}
+
+static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0,
+ SND_SOC_CLOCK_OUT);
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0,
+ SND_SOC_CLOCK_OUT);
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0,
+ SND_SOC_CLOCK_OUT);
+
+ snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0);
+
+ return 0;
+}
+
+static struct snd_soc_ops bfin_eval_adav80x_ops = {
+ .hw_params = bfin_eval_adav80x_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = {
+ {
+ .name = "adav80x",
+ .stream_name = "ADAV80x HiFi",
+ .cpu_dai_name = "bfin-i2s.0",
+ .codec_dai_name = "adav80x-hifi",
+ .platform_name = "bfin-i2s-pcm-audio",
+ .init = bfin_eval_adav80x_codec_init,
+ .ops = &bfin_eval_adav80x_ops,
+ },
+};
+
+static struct snd_soc_card bfin_eval_adav80x = {
+ .name = "bfin-eval-adav80x",
+ .dai_link = bfin_eval_adav80x_dais,
+ .num_links = ARRAY_SIZE(bfin_eval_adav80x_dais),
+
+ .dapm_widgets = bfin_eval_adav80x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets),
+ .dapm_routes = bfin_eval_adav80x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(bfin_eval_adav80x_dapm_routes),
+};
+
+enum bfin_eval_adav80x_type {
+ BFIN_EVAL_ADAV801,
+ BFIN_EVAL_ADAV803,
+};
+
+static int bfin_eval_adav80x_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &bfin_eval_adav80x;
+ const char *codec_name;
+
+ switch (platform_get_device_id(pdev)->driver_data) {
+ case BFIN_EVAL_ADAV801:
+ codec_name = "spi0.1";
+ break;
+ case BFIN_EVAL_ADAV803:
+ codec_name = "adav803.0-0034";
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ bfin_eval_adav80x_dais[0].codec_name = codec_name;
+
+ card->dev = &pdev->dev;
+
+ return snd_soc_register_card(&bfin_eval_adav80x);
+}
+
+static int __devexit bfin_eval_adav80x_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static const struct platform_device_id bfin_eval_adav80x_ids[] = {
+ { "bfin-eval-adav801", BFIN_EVAL_ADAV801 },
+ { "bfin-eval-adav803", BFIN_EVAL_ADAV803 },
+ { },
+};
+MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids);
+
+static struct platform_driver bfin_eval_adav80x_driver = {
+ .driver = {
+ .name = "bfin-eval-adav80x",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = bfin_eval_adav80x_probe,
+ .remove = __devexit_p(bfin_eval_adav80x_remove),
+ .id_table = bfin_eval_adav80x_ids,
+};
+
+static int __init bfin_eval_adav80x_init(void)
+{
+ return platform_driver_register(&bfin_eval_adav80x_driver);
+}
+module_init(bfin_eval_adav80x_init);
+
+static void __exit bfin_eval_adav80x_exit(void)
+{
+ platform_driver_unregister(&bfin_eval_adav80x_driver);
+}
+module_exit(bfin_eval_adav80x_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 98175a0..36a030f 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -17,6 +17,7 @@
select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311
+ select SND_SOC_ADAV80X
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
@@ -42,6 +43,7 @@
select SND_SOC_SN95031 if INTEL_SCU_IPC
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_STA32X if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -71,6 +73,7 @@
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8770 if SPI_MASTER
select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8782
select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8900 if I2C
select SND_SOC_WM8903 if I2C
@@ -84,6 +87,7 @@
select SND_SOC_WM8971 if I2C
select SND_SOC_WM8974 if I2C
select SND_SOC_WM8978 if I2C
+ select SND_SOC_WM8983 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8990 if I2C
@@ -130,7 +134,14 @@
config SND_SOC_AD73311
tristate
-
+
+config SND_SOC_ADAU1701
+ select SIGMA
+ tristate
+
+config SND_SOC_ADAV80X
+ tristate
+
config SND_SOC_ADS117X
tristate
@@ -216,6 +227,9 @@
config SND_SOC_SSM2602
tristate
+config SND_SOC_STA32X
+ tristate
+
config SND_SOC_STAC9766
tristate
@@ -299,6 +313,9 @@
config SND_SOC_WM8776
tristate
+config SND_SOC_WM8782
+ tristate
+
config SND_SOC_WM8804
tristate
@@ -338,6 +355,9 @@
config SND_SOC_WM8978
tristate
+config SND_SOC_WM8983
+ tristate
+
config SND_SOC_WM8985
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fd85584..da9990f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -4,6 +4,8 @@
snd-soc-ad193x-objs := ad193x.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
+snd-soc-adau1701-objs := adau1701.o
+snd-soc-adav80x-objs := adav80x.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
@@ -28,6 +30,7 @@
snd-soc-sn95031-objs := sn95031.o
snd-soc-spdif-objs := spdif_transciever.o
snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-sta32x-objs := sta32x.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -55,6 +58,7 @@
snd-soc-wm8753-objs := wm8753.o
snd-soc-wm8770-objs := wm8770.o
snd-soc-wm8776-objs := wm8776.o
+snd-soc-wm8782-objs := wm8782.o
snd-soc-wm8804-objs := wm8804.o
snd-soc-wm8900-objs := wm8900.o
snd-soc-wm8903-objs := wm8903.o
@@ -68,6 +72,7 @@
snd-soc-wm8971-objs := wm8971.o
snd-soc-wm8974-objs := wm8974.o
snd-soc-wm8978-objs := wm8978.o
+snd-soc-wm8983-objs := wm8983.o
snd-soc-wm8985-objs := wm8985.o
snd-soc-wm8988-objs := wm8988.o
snd-soc-wm8990-objs := wm8990.o
@@ -95,6 +100,8 @@
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o
+obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
@@ -120,6 +127,7 @@
obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
@@ -147,6 +155,7 @@
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o
obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o
+obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o
obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o
obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
@@ -160,6 +169,7 @@
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o
obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o
+obj-$(CONFIG_SND_SOC_WM8983) += snd-soc-wm8983.o
obj-$(CONFIG_SND_SOC_WM8985) += snd-soc-wm8985.o
obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 754c496..4e5c572 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -1,19 +1,10 @@
-/*
- * File: sound/soc/codecs/ad1836.c
- * Author: Barry Song <Barry.Song@analog.com>
+ /*
+ * Audio Codec driver supporting:
+ * AD1835A, AD1836, AD1837A, AD1838A, AD1839A
*
- * Created: Aug 04 2009
- * Description: Driver for AD1836 sound chip
+ * Copyright 2009-2011 Analog Devices Inc.
*
- * Modified:
- * Copyright 2009 Analog Devices Inc.
- *
- * Bugs: Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * Licensed under the GPL-2 or later.
*/
#include <linux/init.h>
@@ -30,10 +21,15 @@
#include <linux/spi/spi.h>
#include "ad1836.h"
+enum ad1836_type {
+ AD1835,
+ AD1836,
+ AD1838,
+};
+
/* codec private data */
struct ad1836_priv {
- enum snd_soc_control_type control_type;
- void *control_data;
+ enum ad1836_type type;
};
/*
@@ -44,29 +40,60 @@
static const struct soc_enum ad1836_deemp_enum =
SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
-static const struct snd_kcontrol_new ad1836_snd_controls[] = {
- /* DAC volume control */
- SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL,
- AD1836_DAC_R1_VOL, 0, 0x3FF, 0),
- SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL,
- AD1836_DAC_R2_VOL, 0, 0x3FF, 0),
- SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL,
- AD1836_DAC_R3_VOL, 0, 0x3FF, 0),
+#define AD1836_DAC_VOLUME(x) \
+ SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \
+ AD1836_DAC_R_VOL(x), 0, 0x3FF, 0)
- /* ADC switch control */
- SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE,
- AD1836_ADCR1_MUTE, 1, 1),
- SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE,
- AD1836_ADCR2_MUTE, 1, 1),
+#define AD1836_DAC_SWITCH(x) \
+ SOC_DOUBLE("DAC" #x " Playback Switch", AD1836_DAC_CTRL2, \
+ AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
- /* DAC switch control */
- SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE,
- AD1836_DACR1_MUTE, 1, 1),
- SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE,
- AD1836_DACR2_MUTE, 1, 1),
- SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE,
- AD1836_DACR3_MUTE, 1, 1),
+#define AD1836_ADC_SWITCH(x) \
+ SOC_DOUBLE("ADC" #x " Capture Switch", AD1836_ADC_CTRL2, \
+ AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
+static const struct snd_kcontrol_new ad183x_dac_controls[] = {
+ AD1836_DAC_VOLUME(1),
+ AD1836_DAC_SWITCH(1),
+ AD1836_DAC_VOLUME(2),
+ AD1836_DAC_SWITCH(2),
+ AD1836_DAC_VOLUME(3),
+ AD1836_DAC_SWITCH(3),
+ AD1836_DAC_VOLUME(4),
+ AD1836_DAC_SWITCH(4),
+};
+
+static const struct snd_soc_dapm_widget ad183x_dac_dapm_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("DAC1OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC2OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC3OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+};
+
+static const struct snd_soc_dapm_route ad183x_dac_routes[] = {
+ { "DAC1OUT", NULL, "DAC" },
+ { "DAC2OUT", NULL, "DAC" },
+ { "DAC3OUT", NULL, "DAC" },
+ { "DAC4OUT", NULL, "DAC" },
+};
+
+static const struct snd_kcontrol_new ad183x_adc_controls[] = {
+ AD1836_ADC_SWITCH(1),
+ AD1836_ADC_SWITCH(2),
+ AD1836_ADC_SWITCH(3),
+};
+
+static const struct snd_soc_dapm_widget ad183x_adc_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("ADC1IN"),
+ SND_SOC_DAPM_INPUT("ADC2IN"),
+};
+
+static const struct snd_soc_dapm_route ad183x_adc_routes[] = {
+ { "ADC", NULL, "ADC1IN" },
+ { "ADC", NULL, "ADC2IN" },
+};
+
+static const struct snd_kcontrol_new ad183x_controls[] = {
/* ADC high-pass filter */
SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1,
AD1836_ADC_HIGHPASS_FILTER, 1, 0),
@@ -75,27 +102,24 @@
SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum),
};
-static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget ad183x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1,
AD1836_DAC_POWERDOWN, 1),
SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1,
AD1836_ADC_POWERDOWN, 1, NULL, 0),
- SND_SOC_DAPM_OUTPUT("DAC1OUT"),
- SND_SOC_DAPM_OUTPUT("DAC2OUT"),
- SND_SOC_DAPM_OUTPUT("DAC3OUT"),
- SND_SOC_DAPM_INPUT("ADC1IN"),
- SND_SOC_DAPM_INPUT("ADC2IN"),
};
-static const struct snd_soc_dapm_route audio_paths[] = {
+static const struct snd_soc_dapm_route ad183x_dapm_routes[] = {
{ "DAC", NULL, "ADC_PWR" },
{ "ADC", NULL, "ADC_PWR" },
- { "DAC1OUT", "DAC1 Switch", "DAC" },
- { "DAC2OUT", "DAC2 Switch", "DAC" },
- { "DAC3OUT", "DAC3 Switch", "DAC" },
- { "ADC", "ADC1 Switch", "ADC1IN" },
- { "ADC", "ADC2 Switch", "ADC2IN" },
+};
+
+static const DECLARE_TLV_DB_SCALE(ad1836_in_tlv, 0, 300, 0);
+
+static const struct snd_kcontrol_new ad1836_controls[] = {
+ SOC_DOUBLE_TLV("ADC2 Capture Volume", AD1836_ADC_CTRL1, 3, 0, 4, 0,
+ ad1836_in_tlv),
};
/*
@@ -165,64 +189,69 @@
return 0;
}
-#ifdef CONFIG_PM
-static int ad1836_soc_suspend(struct snd_soc_codec *codec,
- pm_message_t state)
-{
- /* reset clock control mode */
- u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
- adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
-
- return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
-}
-
-static int ad1836_soc_resume(struct snd_soc_codec *codec)
-{
- /* restore clock control mode */
- u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
- adc_ctrl2 |= AD1836_ADC_AUX;
-
- return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
-}
-#else
-#define ad1836_soc_suspend NULL
-#define ad1836_soc_resume NULL
-#endif
-
static struct snd_soc_dai_ops ad1836_dai_ops = {
.hw_params = ad1836_hw_params,
.set_fmt = ad1836_set_dai_fmt,
};
-/* codec DAI instance */
-static struct snd_soc_dai_driver ad1836_dai = {
- .name = "ad1836-hifi",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 2,
- .channels_max = 6,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
- .capture = {
- .stream_name = "Capture",
- .channels_min = 2,
- .channels_max = 4,
- .rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
- },
- .ops = &ad1836_dai_ops,
+#define AD183X_DAI(_name, num_dacs, num_adcs) \
+{ \
+ .name = _name "-hifi", \
+ .playback = { \
+ .stream_name = "Playback", \
+ .channels_min = 2, \
+ .channels_max = (num_dacs) * 2, \
+ .rates = SNDRV_PCM_RATE_48000, \
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
+ }, \
+ .capture = { \
+ .stream_name = "Capture", \
+ .channels_min = 2, \
+ .channels_max = (num_adcs) * 2, \
+ .rates = SNDRV_PCM_RATE_48000, \
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
+ }, \
+ .ops = &ad1836_dai_ops, \
+}
+
+static struct snd_soc_dai_driver ad183x_dais[] = {
+ [AD1835] = AD183X_DAI("ad1835", 4, 1),
+ [AD1836] = AD183X_DAI("ad1836", 3, 2),
+ [AD1838] = AD183X_DAI("ad1838", 3, 1),
};
+#ifdef CONFIG_PM
+static int ad1836_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ /* reset clock control mode */
+ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+ AD1836_ADC_SERFMT_MASK, 0);
+}
+
+static int ad1836_resume(struct snd_soc_codec *codec)
+{
+ /* restore clock control mode */
+ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+ AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX);
+}
+#else
+#define ad1836_suspend NULL
+#define ad1836_resume NULL
+#endif
+
static int ad1836_probe(struct snd_soc_codec *codec)
{
struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int num_dacs, num_adcs;
int ret = 0;
+ int i;
- codec->control_data = ad1836->control_data;
+ num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2;
+ num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2;
+
ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI);
if (ret < 0) {
dev_err(codec->dev, "failed to set cache I/O: %d\n",
@@ -239,21 +268,46 @@
snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100);
/* unmute adc channles, adc aux mode */
snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180);
- /* left/right diff:PGA/MUX */
- snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
/* volume */
- snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF);
- snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF);
+ for (i = 1; i <= num_dacs; ++i) {
+ snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF);
+ snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF);
+ }
- snd_soc_add_controls(codec, ad1836_snd_controls,
- ARRAY_SIZE(ad1836_snd_controls));
- snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets,
- ARRAY_SIZE(ad1836_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
+ if (ad1836->type == AD1836) {
+ /* left/right diff:PGA/MUX */
+ snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
+ ret = snd_soc_add_controls(codec, ad1836_controls,
+ ARRAY_SIZE(ad1836_controls));
+ if (ret)
+ return ret;
+ } else {
+ snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00);
+ }
+
+ ret = snd_soc_add_controls(codec, ad183x_dac_controls, num_dacs * 2);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_add_controls(codec, ad183x_adc_controls, num_adcs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, ad183x_dac_dapm_widgets, num_dacs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, ad183x_adc_dapm_widgets, num_adcs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, ad183x_dac_routes, num_dacs);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, ad183x_adc_routes, num_adcs);
+ if (ret)
+ return ret;
return ret;
}
@@ -262,19 +316,24 @@
static int ad1836_remove(struct snd_soc_codec *codec)
{
/* reset clock control mode */
- u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
- adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
-
- return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
+ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+ AD1836_ADC_SERFMT_MASK, 0);
}
static struct snd_soc_codec_driver soc_codec_dev_ad1836 = {
- .probe = ad1836_probe,
- .remove = ad1836_remove,
- .suspend = ad1836_soc_suspend,
- .resume = ad1836_soc_resume,
+ .probe = ad1836_probe,
+ .remove = ad1836_remove,
+ .suspend = ad1836_suspend,
+ .resume = ad1836_resume,
.reg_cache_size = AD1836_NUM_REGS,
.reg_word_size = sizeof(u16),
+
+ .controls = ad183x_controls,
+ .num_controls = ARRAY_SIZE(ad183x_controls),
+ .dapm_widgets = ad183x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ad183x_dapm_widgets),
+ .dapm_routes = ad183x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes),
};
static int __devinit ad1836_spi_probe(struct spi_device *spi)
@@ -286,12 +345,12 @@
if (ad1836 == NULL)
return -ENOMEM;
+ ad1836->type = spi_get_device_id(spi)->driver_data;
+
spi_set_drvdata(spi, ad1836);
- ad1836->control_data = spi;
- ad1836->control_type = SND_SOC_SPI;
ret = snd_soc_register_codec(&spi->dev,
- &soc_codec_dev_ad1836, &ad1836_dai, 1);
+ &soc_codec_dev_ad1836, &ad183x_dais[ad1836->type], 1);
if (ret < 0)
kfree(ad1836);
return ret;
@@ -303,27 +362,29 @@
kfree(spi_get_drvdata(spi));
return 0;
}
+static const struct spi_device_id ad1836_ids[] = {
+ { "ad1835", AD1835 },
+ { "ad1836", AD1836 },
+ { "ad1837", AD1835 },
+ { "ad1838", AD1838 },
+ { "ad1839", AD1838 },
+ { },
+};
+MODULE_DEVICE_TABLE(spi, ad1836_ids);
static struct spi_driver ad1836_spi_driver = {
.driver = {
- .name = "ad1836-codec",
+ .name = "ad1836",
.owner = THIS_MODULE,
},
.probe = ad1836_spi_probe,
.remove = __devexit_p(ad1836_spi_remove),
+ .id_table = ad1836_ids,
};
static int __init ad1836_init(void)
{
- int ret;
-
- ret = spi_register_driver(&ad1836_spi_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n",
- ret);
- }
-
- return ret;
+ return spi_register_driver(&ad1836_spi_driver);
}
module_init(ad1836_init);
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 9d6a3f8..444747f 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -1,19 +1,10 @@
/*
- * File: sound/soc/codecs/ad1836.h
- * Based on:
- * Author: Barry Song <Barry.Song@analog.com>
+ * Audio Codec driver supporting:
+ * AD1835A, AD1836, AD1837A, AD1838A, AD1839A
*
- * Created: Aug 04, 2009
- * Description: definitions for AD1836 registers
+ * Copyright 2009-2011 Analog Devices Inc.
*
- * Modified:
- *
- * Bugs: Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * Licensed under the GPL-2 or later.
*/
#ifndef __AD1836_H__
@@ -21,39 +12,30 @@
#define AD1836_DAC_CTRL1 0
#define AD1836_DAC_POWERDOWN 2
-#define AD1836_DAC_SERFMT_MASK 0xE0
+#define AD1836_DAC_SERFMT_MASK 0xE0
#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5)
#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5)
#define AD1836_DAC_WORD_LEN_MASK 0x18
#define AD1836_DAC_WORD_LEN_OFFSET 3
#define AD1836_DAC_CTRL2 1
-#define AD1836_DACL1_MUTE 0
-#define AD1836_DACR1_MUTE 1
-#define AD1836_DACL2_MUTE 2
-#define AD1836_DACR2_MUTE 3
-#define AD1836_DACL3_MUTE 4
-#define AD1836_DACR3_MUTE 5
-#define AD1836_DAC_L1_VOL 2
-#define AD1836_DAC_R1_VOL 3
-#define AD1836_DAC_L2_VOL 4
-#define AD1836_DAC_R2_VOL 5
-#define AD1836_DAC_L3_VOL 6
-#define AD1836_DAC_R3_VOL 7
+/* These macros are one-based. So AD183X_MUTE_LEFT(1) will return the mute bit
+ * for the first ADC/DAC */
+#define AD1836_MUTE_LEFT(x) (((x) * 2) - 2)
+#define AD1836_MUTE_RIGHT(x) (((x) * 2) - 1)
+
+#define AD1836_DAC_L_VOL(x) ((x) * 2)
+#define AD1836_DAC_R_VOL(x) (1 + ((x) * 2))
#define AD1836_ADC_CTRL1 12
#define AD1836_ADC_POWERDOWN 7
#define AD1836_ADC_HIGHPASS_FILTER 8
#define AD1836_ADC_CTRL2 13
-#define AD1836_ADCL1_MUTE 0
-#define AD1836_ADCR1_MUTE 1
-#define AD1836_ADCL2_MUTE 2
-#define AD1836_ADCR2_MUTE 3
#define AD1836_ADC_WORD_LEN_MASK 0x30
#define AD1836_ADC_WORD_OFFSET 5
-#define AD1836_ADC_SERFMT_MASK (7 << 6)
+#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
#define AD1836_ADC_AUX (0x6 << 6)
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
new file mode 100644
index 0000000..2758d5f
--- /dev/null
+++ b/sound/soc/codecs/adau1701.c
@@ -0,0 +1,549 @@
+/*
+ * Driver for ADAU1701 SigmaDSP processor
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ * based on an inital version by Cliff Cai <cliff.cai@analog.com>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/sigma.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "adau1701.h"
+
+#define ADAU1701_DSPCTRL 0x1c
+#define ADAU1701_SEROCTL 0x1e
+#define ADAU1701_SERICTL 0x1f
+
+#define ADAU1701_AUXNPOW 0x22
+
+#define ADAU1701_OSCIPOW 0x26
+#define ADAU1701_DACSET 0x27
+
+#define ADAU1701_NUM_REGS 0x28
+
+#define ADAU1701_DSPCTRL_CR (1 << 2)
+#define ADAU1701_DSPCTRL_DAM (1 << 3)
+#define ADAU1701_DSPCTRL_ADM (1 << 4)
+#define ADAU1701_DSPCTRL_SR_48 0x00
+#define ADAU1701_DSPCTRL_SR_96 0x01
+#define ADAU1701_DSPCTRL_SR_192 0x02
+#define ADAU1701_DSPCTRL_SR_MASK 0x03
+
+#define ADAU1701_SEROCTL_INV_LRCLK 0x2000
+#define ADAU1701_SEROCTL_INV_BCLK 0x1000
+#define ADAU1701_SEROCTL_MASTER 0x0800
+
+#define ADAU1701_SEROCTL_OBF16 0x0000
+#define ADAU1701_SEROCTL_OBF8 0x0200
+#define ADAU1701_SEROCTL_OBF4 0x0400
+#define ADAU1701_SEROCTL_OBF2 0x0600
+#define ADAU1701_SEROCTL_OBF_MASK 0x0600
+
+#define ADAU1701_SEROCTL_OLF1024 0x0000
+#define ADAU1701_SEROCTL_OLF512 0x0080
+#define ADAU1701_SEROCTL_OLF256 0x0100
+#define ADAU1701_SEROCTL_OLF_MASK 0x0180
+
+#define ADAU1701_SEROCTL_MSB_DEALY1 0x0000
+#define ADAU1701_SEROCTL_MSB_DEALY0 0x0004
+#define ADAU1701_SEROCTL_MSB_DEALY8 0x0008
+#define ADAU1701_SEROCTL_MSB_DEALY12 0x000c
+#define ADAU1701_SEROCTL_MSB_DEALY16 0x0010
+#define ADAU1701_SEROCTL_MSB_DEALY_MASK 0x001c
+
+#define ADAU1701_SEROCTL_WORD_LEN_24 0x0000
+#define ADAU1701_SEROCTL_WORD_LEN_20 0x0001
+#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010
+#define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003
+
+#define ADAU1701_AUXNPOW_VBPD 0x40
+#define ADAU1701_AUXNPOW_VRPD 0x20
+
+#define ADAU1701_SERICTL_I2S 0
+#define ADAU1701_SERICTL_LEFTJ 1
+#define ADAU1701_SERICTL_TDM 2
+#define ADAU1701_SERICTL_RIGHTJ_24 3
+#define ADAU1701_SERICTL_RIGHTJ_20 4
+#define ADAU1701_SERICTL_RIGHTJ_18 5
+#define ADAU1701_SERICTL_RIGHTJ_16 6
+#define ADAU1701_SERICTL_MODE_MASK 7
+#define ADAU1701_SERICTL_INV_BCLK BIT(3)
+#define ADAU1701_SERICTL_INV_LRCLK BIT(4)
+
+#define ADAU1701_OSCIPOW_OPD 0x04
+#define ADAU1701_DACSET_DACINIT 1
+
+#define ADAU1701_FIRMWARE "adau1701.bin"
+
+struct adau1701 {
+ unsigned int dai_fmt;
+};
+
+static const struct snd_kcontrol_new adau1701_controls[] = {
+ SOC_SINGLE("Master Capture Switch", ADAU1701_DSPCTRL, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget adau1701_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC0", "Playback", ADAU1701_AUXNPOW, 3, 1),
+ SND_SOC_DAPM_DAC("DAC1", "Playback", ADAU1701_AUXNPOW, 2, 1),
+ SND_SOC_DAPM_DAC("DAC2", "Playback", ADAU1701_AUXNPOW, 1, 1),
+ SND_SOC_DAPM_DAC("DAC3", "Playback", ADAU1701_AUXNPOW, 0, 1),
+ SND_SOC_DAPM_ADC("ADC", "Capture", ADAU1701_AUXNPOW, 7, 1),
+
+ SND_SOC_DAPM_OUTPUT("OUT0"),
+ SND_SOC_DAPM_OUTPUT("OUT1"),
+ SND_SOC_DAPM_OUTPUT("OUT2"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_INPUT("IN0"),
+ SND_SOC_DAPM_INPUT("IN1"),
+};
+
+static const struct snd_soc_dapm_route adau1701_dapm_routes[] = {
+ { "OUT0", NULL, "DAC0" },
+ { "OUT1", NULL, "DAC1" },
+ { "OUT2", NULL, "DAC2" },
+ { "OUT3", NULL, "DAC3" },
+
+ { "ADC", NULL, "IN0" },
+ { "ADC", NULL, "IN1" },
+};
+
+static unsigned int adau1701_register_size(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ case ADAU1701_DSPCTRL:
+ case ADAU1701_SEROCTL:
+ case ADAU1701_AUXNPOW:
+ case ADAU1701_OSCIPOW:
+ case ADAU1701_DACSET:
+ return 2;
+ case ADAU1701_SERICTL:
+ return 1;
+ }
+
+ dev_err(codec->dev, "Unsupported register address: %d\n", reg);
+ return 0;
+}
+
+static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ unsigned int i;
+ unsigned int size;
+ uint8_t buf[4];
+ int ret;
+
+ size = adau1701_register_size(codec, reg);
+ if (size == 0)
+ return -EINVAL;
+
+ snd_soc_cache_write(codec, reg, value);
+
+ buf[0] = 0x08;
+ buf[1] = reg;
+
+ for (i = size + 1; i >= 2; --i) {
+ buf[i] = value;
+ value >>= 8;
+ }
+
+ ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2);
+ if (ret == size + 2)
+ return 0;
+ else if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ unsigned int value;
+ unsigned int ret;
+
+ ret = snd_soc_cache_read(codec, reg, &value);
+ if (ret)
+ return ret;
+
+ return value;
+}
+
+static int adau1701_load_firmware(struct snd_soc_codec *codec)
+{
+ return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE);
+}
+
+static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
+ snd_pcm_format_t format)
+{
+ struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+ unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK;
+ unsigned int val;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAU1701_SEROCTL_WORD_LEN_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAU1701_SEROCTL_WORD_LEN_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAU1701_SEROCTL_WORD_LEN_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) {
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val |= ADAU1701_SEROCTL_MSB_DEALY16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val |= ADAU1701_SEROCTL_MSB_DEALY12;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val |= ADAU1701_SEROCTL_MSB_DEALY8;
+ break;
+ }
+ mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val);
+
+ return 0;
+}
+
+static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
+ snd_pcm_format_t format)
+{
+ struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
+ return 0;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAU1701_SERICTL_RIGHTJ_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAU1701_SERICTL_RIGHTJ_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAU1701_SERICTL_RIGHTJ_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_SERICTL,
+ ADAU1701_SERICTL_MODE_MASK, val);
+
+ return 0;
+}
+
+static int adau1701_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ snd_pcm_format_t format;
+ unsigned int val;
+
+ switch (params_rate(params)) {
+ case 192000:
+ val = ADAU1701_DSPCTRL_SR_192;
+ break;
+ case 96000:
+ val = ADAU1701_DSPCTRL_SR_96;
+ break;
+ case 48000:
+ val = ADAU1701_DSPCTRL_SR_48;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_DSPCTRL,
+ ADAU1701_DSPCTRL_SR_MASK, val);
+
+ format = params_format(params);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return adau1701_set_playback_pcm_format(codec, format);
+ else
+ return adau1701_set_capture_pcm_format(codec, format);
+}
+
+static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+ unsigned int serictl = 0x00, seroctl = 0x00;
+ bool invert_lrclk;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* master, 64-bits per sample, 1 frame per sample */
+ seroctl |= ADAU1701_SEROCTL_MASTER | ADAU1701_SEROCTL_OBF16
+ | ADAU1701_SEROCTL_OLF1024;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ invert_lrclk = true;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ invert_lrclk = false;
+ serictl |= ADAU1701_SERICTL_INV_BCLK;
+ seroctl |= ADAU1701_SEROCTL_INV_BCLK;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ invert_lrclk = true;
+ serictl |= ADAU1701_SERICTL_INV_BCLK;
+ seroctl |= ADAU1701_SEROCTL_INV_BCLK;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ serictl |= ADAU1701_SERICTL_LEFTJ;
+ seroctl |= ADAU1701_SEROCTL_MSB_DEALY0;
+ invert_lrclk = !invert_lrclk;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ serictl |= ADAU1701_SERICTL_RIGHTJ_24;
+ seroctl |= ADAU1701_SEROCTL_MSB_DEALY8;
+ invert_lrclk = !invert_lrclk;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (invert_lrclk) {
+ seroctl |= ADAU1701_SEROCTL_INV_LRCLK;
+ serictl |= ADAU1701_SERICTL_INV_LRCLK;
+ }
+
+ adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+ snd_soc_write(codec, ADAU1701_SERICTL, serictl);
+ snd_soc_update_bits(codec, ADAU1701_SEROCTL,
+ ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl);
+
+ return 0;
+}
+
+static int adau1701_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* Enable VREF and VREF buffer */
+ snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* Disable VREF and VREF buffer */
+ snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int mask = ADAU1701_DSPCTRL_DAM;
+ unsigned int val;
+
+ if (mute)
+ val = 0;
+ else
+ val = mask;
+
+ snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val);
+
+ return 0;
+}
+
+static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ unsigned int freq, int dir)
+{
+ unsigned int val;
+
+ switch (clk_id) {
+ case ADAU1701_CLK_SRC_OSC:
+ val = 0x0;
+ break;
+ case ADAU1701_CLK_SRC_MCLK:
+ val = ADAU1701_OSCIPOW_OPD;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val);
+
+ return 0;
+}
+
+#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_192000)
+
+#define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static const struct snd_soc_dai_ops adau1701_dai_ops = {
+ .set_fmt = adau1701_set_dai_fmt,
+ .hw_params = adau1701_hw_params,
+ .digital_mute = adau1701_digital_mute,
+};
+
+static struct snd_soc_dai_driver adau1701_dai = {
+ .name = "adau1701",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = ADAU1701_RATES,
+ .formats = ADAU1701_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = ADAU1701_RATES,
+ .formats = ADAU1701_FORMATS,
+ },
+ .ops = &adau1701_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int adau1701_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ codec->dapm.idle_bias_off = 1;
+
+ ret = adau1701_load_firmware(codec);
+ if (ret)
+ dev_warn(codec->dev, "Failed to load firmware\n");
+
+ snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT);
+ snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver adau1701_codec_drv = {
+ .probe = adau1701_probe,
+ .set_bias_level = adau1701_set_bias_level,
+
+ .reg_cache_size = ADAU1701_NUM_REGS,
+ .reg_word_size = sizeof(u16),
+
+ .controls = adau1701_controls,
+ .num_controls = ARRAY_SIZE(adau1701_controls),
+ .dapm_widgets = adau1701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adau1701_dapm_widgets),
+ .dapm_routes = adau1701_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes),
+
+ .write = adau1701_write,
+ .read = adau1701_read,
+
+ .set_sysclk = adau1701_set_sysclk,
+};
+
+static __devinit int adau1701_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct adau1701 *adau1701;
+ int ret;
+
+ adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL);
+ if (!adau1701)
+ return -ENOMEM;
+
+ i2c_set_clientdata(client, adau1701);
+ ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv,
+ &adau1701_dai, 1);
+ if (ret < 0)
+ kfree(adau1701);
+
+ return ret;
+}
+
+static __devexit int adau1701_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ kfree(i2c_get_clientdata(client));
+ return 0;
+}
+
+static const struct i2c_device_id adau1701_i2c_id[] = {
+ { "adau1701", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id);
+
+static struct i2c_driver adau1701_i2c_driver = {
+ .driver = {
+ .name = "adau1701",
+ .owner = THIS_MODULE,
+ },
+ .probe = adau1701_i2c_probe,
+ .remove = __devexit_p(adau1701_i2c_remove),
+ .id_table = adau1701_i2c_id,
+};
+
+static int __init adau1701_init(void)
+{
+ return i2c_add_driver(&adau1701_i2c_driver);
+}
+module_init(adau1701_init);
+
+static void __exit adau1701_exit(void)
+{
+ i2c_del_driver(&adau1701_i2c_driver);
+}
+module_exit(adau1701_exit);
+
+MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver");
+MODULE_AUTHOR("Cliff Cai <cliff.cai@analog.com>");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1701.h b/sound/soc/codecs/adau1701.h
new file mode 100644
index 0000000..8d0949a
--- /dev/null
+++ b/sound/soc/codecs/adau1701.h
@@ -0,0 +1,17 @@
+/*
+ * header file for ADAU1701 SigmaDSP processor
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAU1701_H
+#define _ADAU1701_H
+
+enum adau1701_clk_src {
+ ADAU1701_CLK_SRC_OSC,
+ ADAU1701_CLK_SRC_MCLK,
+};
+
+#endif
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
new file mode 100644
index 0000000..300c04b
--- /dev/null
+++ b/sound/soc/codecs/adav80x.c
@@ -0,0 +1,951 @@
+/*
+ * ADAV80X Audio Codec driver supporting ADAV801, ADAV803
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Yi Li <yi.li@analog.com>
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+#define ADAV80X_PLAYBACK_CTRL 0x04
+#define ADAV80X_AUX_IN_CTRL 0x05
+#define ADAV80X_REC_CTRL 0x06
+#define ADAV80X_AUX_OUT_CTRL 0x07
+#define ADAV80X_DPATH_CTRL1 0x62
+#define ADAV80X_DPATH_CTRL2 0x63
+#define ADAV80X_DAC_CTRL1 0x64
+#define ADAV80X_DAC_CTRL2 0x65
+#define ADAV80X_DAC_CTRL3 0x66
+#define ADAV80X_DAC_L_VOL 0x68
+#define ADAV80X_DAC_R_VOL 0x69
+#define ADAV80X_PGA_L_VOL 0x6c
+#define ADAV80X_PGA_R_VOL 0x6d
+#define ADAV80X_ADC_CTRL1 0x6e
+#define ADAV80X_ADC_CTRL2 0x6f
+#define ADAV80X_ADC_L_VOL 0x70
+#define ADAV80X_ADC_R_VOL 0x71
+#define ADAV80X_PLL_CTRL1 0x74
+#define ADAV80X_PLL_CTRL2 0x75
+#define ADAV80X_ICLK_CTRL1 0x76
+#define ADAV80X_ICLK_CTRL2 0x77
+#define ADAV80X_PLL_CLK_SRC 0x78
+#define ADAV80X_PLL_OUTE 0x7a
+
+#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00
+#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll))
+#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll))
+
+#define ADAV80X_ICLK_CTRL1_DAC_SRC(src) ((src) << 5)
+#define ADAV80X_ICLK_CTRL1_ADC_SRC(src) ((src) << 2)
+#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src) (src)
+#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src) ((src) << 3)
+
+#define ADAV80X_PLL_CTRL1_PLLDIV 0x10
+#define ADAV80X_PLL_CTRL1_PLLPD(pll) (0x04 << (pll))
+#define ADAV80X_PLL_CTRL1_XTLPD 0x02
+
+#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4))
+
+#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00)
+#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08)
+#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c)
+
+#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02)
+#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01)
+#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f)
+
+#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80
+#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00
+#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80
+
+#define ADAV80X_DAC_CTRL1_PD 0x80
+
+#define ADAV80X_DAC_CTRL2_DIV1 0x00
+#define ADAV80X_DAC_CTRL2_DIV1_5 0x10
+#define ADAV80X_DAC_CTRL2_DIV2 0x20
+#define ADAV80X_DAC_CTRL2_DIV3 0x30
+#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30
+
+#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00
+#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40
+#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80
+#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0
+
+#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00
+#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01
+#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02
+#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03
+#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01
+
+#define ADAV80X_CAPTURE_MODE_MASTER 0x20
+#define ADAV80X_CAPTURE_WORD_LEN24 0x00
+#define ADAV80X_CAPTURE_WORD_LEN20 0x04
+#define ADAV80X_CAPTRUE_WORD_LEN18 0x08
+#define ADAV80X_CAPTURE_WORD_LEN16 0x0c
+#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c
+
+#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00
+#define ADAV80X_CAPTURE_MODE_I2S 0x01
+#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03
+#define ADAV80X_CAPTURE_MODE_MASK 0x03
+
+#define ADAV80X_PLAYBACK_MODE_MASTER 0x10
+#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00
+#define ADAV80X_PLAYBACK_MODE_I2S 0x01
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07
+#define ADAV80X_PLAYBACK_MODE_MASK 0x07
+
+#define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x))
+
+static u8 adav80x_default_regs[] = {
+ 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00,
+ 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37,
+ 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b,
+ 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00,
+ 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee,
+ 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f,
+ 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x52, 0x00,
+};
+
+struct adav80x {
+ enum snd_soc_control_type control_type;
+
+ enum adav80x_clk_src clk_src;
+ unsigned int sysclk;
+ enum adav80x_pll_src pll_src;
+
+ unsigned int dai_fmt[2];
+ unsigned int rate;
+ bool deemph;
+ bool sysclk_pd[3];
+};
+
+static const char *adav80x_mux_text[] = {
+ "ADC",
+ "Playback",
+ "Aux Playback",
+};
+
+static const unsigned int adav80x_mux_values[] = {
+ 0, 2, 3,
+};
+
+#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \
+ SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \
+ ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \
+ adav80x_mux_values)
+
+static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0);
+static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3);
+static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3);
+
+static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl =
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum);
+static const struct snd_kcontrol_new adav80x_capture_mux_ctrl =
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum);
+static const struct snd_kcontrol_new adav80x_dac_mux_ctrl =
+ SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum);
+
+#define ADAV80X_MUX(name, ctrl) \
+ SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1),
+ SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1),
+
+ SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl),
+ ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl),
+ ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl),
+
+ SND_SOC_DAPM_INPUT("VINR"),
+ SND_SOC_DAPM_INPUT("VINL"),
+ SND_SOC_DAPM_OUTPUT("VOUTR"),
+ SND_SOC_DAPM_OUTPUT("VOUTL"),
+
+ SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0),
+};
+
+static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = source->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ const char *clk;
+
+ switch (adav80x->clk_src) {
+ case ADAV80X_CLK_PLL1:
+ clk = "PLL1";
+ break;
+ case ADAV80X_CLK_PLL2:
+ clk = "PLL2";
+ break;
+ case ADAV80X_CLK_XTAL:
+ clk = "OSC";
+ break;
+ default:
+ return 0;
+ }
+
+ return strcmp(source->name, clk) == 0;
+}
+
+static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink)
+{
+ struct snd_soc_codec *codec = source->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL;
+}
+
+
+static const struct snd_soc_dapm_route adav80x_dapm_routes[] = {
+ { "DAC Select", "ADC", "ADC" },
+ { "DAC Select", "Playback", "AIFIN" },
+ { "DAC Select", "Aux Playback", "AIFAUXIN" },
+ { "DAC", NULL, "DAC Select" },
+
+ { "Capture Select", "ADC", "ADC" },
+ { "Capture Select", "Playback", "AIFIN" },
+ { "Capture Select", "Aux Playback", "AIFAUXIN" },
+ { "AIFOUT", NULL, "Capture Select" },
+
+ { "Aux Capture Select", "ADC", "ADC" },
+ { "Aux Capture Select", "Playback", "AIFIN" },
+ { "Aux Capture Select", "Aux Playback", "AIFAUXIN" },
+ { "AIFAUXOUT", NULL, "Aux Capture Select" },
+
+ { "VOUTR", NULL, "DAC" },
+ { "VOUTL", NULL, "DAC" },
+
+ { "Left PGA", NULL, "VINL" },
+ { "Right PGA", NULL, "VINR" },
+ { "ADC", NULL, "Left PGA" },
+ { "ADC", NULL, "Right PGA" },
+
+ { "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check },
+ { "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check },
+ { "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check },
+ { "PLL1", NULL, "OSC", adav80x_dapm_pll_check },
+ { "PLL2", NULL, "OSC", adav80x_dapm_pll_check },
+
+ { "ADC", NULL, "SYSCLK" },
+ { "DAC", NULL, "SYSCLK" },
+ { "AIFOUT", NULL, "SYSCLK" },
+ { "AIFAUXOUT", NULL, "SYSCLK" },
+ { "AIFIN", NULL, "SYSCLK" },
+ { "AIFAUXIN", NULL, "SYSCLK" },
+};
+
+static int adav80x_set_deemph(struct snd_soc_codec *codec)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adav80x->deemph) {
+ switch (adav80x->rate) {
+ case 32000:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_32;
+ break;
+ case 44100:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_44;
+ break;
+ case 48000:
+ case 64000:
+ case 88200:
+ case 96000:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_48;
+ break;
+ default:
+ val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+ break;
+ }
+ } else {
+ val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+ }
+
+ return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+ ADAV80X_DAC_CTRL2_DEEMPH_MASK, val);
+}
+
+static int adav80x_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int deemph = ucontrol->value.enumerated.item[0];
+
+ if (deemph > 1)
+ return -EINVAL;
+
+ adav80x->deemph = deemph;
+
+ return adav80x_set_deemph(codec);
+}
+
+static int adav80x_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = adav80x->deemph;
+ return 0;
+};
+
+static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0);
+
+static const struct snd_kcontrol_new adav80x_controls[] = {
+ SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL,
+ ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+ SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL,
+ ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+
+ SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL,
+ ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv),
+
+ SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0),
+ SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1),
+
+ SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0),
+
+ SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0,
+ adav80x_get_deemph, adav80x_put_deemph),
+};
+
+static unsigned int adav80x_port_ctrl_regs[2][2] = {
+ { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, },
+ { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL },
+};
+
+static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int capture = 0x00;
+ unsigned int playback = 0x00;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ capture |= ADAV80X_CAPTURE_MODE_MASTER;
+ playback |= ADAV80X_PLAYBACK_MODE_MASTER;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ capture |= ADAV80X_CAPTURE_MODE_I2S;
+ playback |= ADAV80X_PLAYBACK_MODE_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ capture |= ADAV80X_CAPTURE_MODE_LEFT_J;
+ playback |= ADAV80X_PLAYBACK_MODE_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ capture |= ADAV80X_CAPTURE_MODE_RIGHT_J;
+ playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+ ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER,
+ capture);
+ snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback);
+
+ adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+ return 0;
+}
+
+static int adav80x_set_adc_clock(struct snd_soc_codec *codec,
+ unsigned int sample_rate)
+{
+ unsigned int val;
+
+ if (sample_rate <= 48000)
+ val = ADAV80X_ADC_CTRL1_MODULATOR_128FS;
+ else
+ val = ADAV80X_ADC_CTRL1_MODULATOR_64FS;
+
+ snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1,
+ ADAV80X_ADC_CTRL1_MODULATOR_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
+ unsigned int sample_rate)
+{
+ unsigned int val;
+
+ if (sample_rate <= 48000)
+ val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS;
+ else
+ val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS;
+
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+ ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK,
+ val);
+
+ return 0;
+}
+
+static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
+ struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+ unsigned int val;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAV80X_CAPTURE_WORD_LEN16;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ val = ADAV80X_CAPTRUE_WORD_LEN18;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAV80X_CAPTURE_WORD_LEN20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAV80X_CAPTURE_WORD_LEN24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+ ADAV80X_CAPTURE_WORD_LEN_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
+ struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int val;
+
+ if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J)
+ return 0;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16;
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1],
+ ADAV80X_PLAYBACK_MODE_MASK, val);
+
+ return 0;
+}
+
+static int adav80x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int rate = params_rate(params);
+
+ if (rate * 256 != adav80x->sysclk)
+ return -EINVAL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ adav80x_set_playback_pcm_format(codec, dai,
+ params_format(params));
+ adav80x_set_dac_clock(codec, rate);
+ } else {
+ adav80x_set_capture_pcm_format(codec, dai,
+ params_format(params));
+ adav80x_set_adc_clock(codec, rate);
+ }
+ adav80x->rate = rate;
+ adav80x_set_deemph(codec);
+
+ return 0;
+}
+
+static int adav80x_set_sysclk(struct snd_soc_codec *codec,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (dir == SND_SOC_CLOCK_IN) {
+ switch (clk_id) {
+ case ADAV80X_CLK_XIN:
+ case ADAV80X_CLK_XTAL:
+ case ADAV80X_CLK_MCLKI:
+ case ADAV80X_CLK_PLL1:
+ case ADAV80X_CLK_PLL2:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adav80x->sysclk = freq;
+
+ if (adav80x->clk_src != clk_id) {
+ unsigned int iclk_ctrl1, iclk_ctrl2;
+
+ adav80x->clk_src = clk_id;
+ if (clk_id == ADAV80X_CLK_XTAL)
+ clk_id = ADAV80X_CLK_XIN;
+
+ iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) |
+ ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) |
+ ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id);
+ iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id);
+
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1);
+ snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2);
+
+ snd_soc_dapm_sync(&codec->dapm);
+ }
+ } else {
+ unsigned int mask;
+
+ switch (clk_id) {
+ case ADAV80X_CLK_SYSCLK1:
+ case ADAV80X_CLK_SYSCLK2:
+ case ADAV80X_CLK_SYSCLK3:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ clk_id -= ADAV80X_CLK_SYSCLK1;
+ mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id);
+
+ if (freq == 0) {
+ snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask);
+ adav80x->sysclk_pd[clk_id] = true;
+ } else {
+ snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0);
+ adav80x->sysclk_pd[clk_id] = false;
+ }
+
+ if (adav80x->sysclk_pd[0])
+ snd_soc_dapm_disable_pin(&codec->dapm, "PLL1");
+ else
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+
+ if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2])
+ snd_soc_dapm_disable_pin(&codec->dapm, "PLL2");
+ else
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+ snd_soc_dapm_sync(&codec->dapm);
+ }
+
+ return 0;
+}
+
+static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
+{
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ unsigned int pll_ctrl1 = 0;
+ unsigned int pll_ctrl2 = 0;
+ unsigned int pll_src;
+
+ switch (source) {
+ case ADAV80X_PLL_SRC_XTAL:
+ case ADAV80X_PLL_SRC_XIN:
+ case ADAV80X_PLL_SRC_MCLKI:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (!freq_out)
+ return 0;
+
+ switch (freq_in) {
+ case 27000000:
+ break;
+ case 54000000:
+ if (source == ADAV80X_PLL_SRC_XIN) {
+ pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
+ break;
+ }
+ default:
+ return -EINVAL;
+ }
+
+ if (freq_out > 12288000) {
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id);
+ freq_out /= 2;
+ }
+
+ /* freq_out = sample_rate * 256 */
+ switch (freq_out) {
+ case 8192000:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id);
+ break;
+ case 11289600:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id);
+ break;
+ case 12288000:
+ pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV,
+ pll_ctrl1);
+ snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2,
+ ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2);
+
+ if (source != adav80x->pll_src) {
+ if (source == ADAV80X_PLL_SRC_MCLKI)
+ pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id);
+ else
+ pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id);
+
+ snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC,
+ ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src);
+
+ adav80x->pll_src = source;
+
+ snd_soc_dapm_sync(&codec->dapm);
+ }
+
+ return 0;
+}
+
+static int adav80x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ unsigned int mask = ADAV80X_DAC_CTRL1_PD;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+/* Enforce the same sample rate on all audio interfaces */
+static int adav80x_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (!codec->active || !adav80x->rate)
+ return 0;
+
+ return snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+}
+
+static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ if (!codec->active)
+ adav80x->rate = 0;
+}
+
+static const struct snd_soc_dai_ops adav80x_dai_ops = {
+ .set_fmt = adav80x_set_dai_fmt,
+ .hw_params = adav80x_hw_params,
+ .startup = adav80x_dai_startup,
+ .shutdown = adav80x_dai_shutdown,
+};
+
+#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver adav80x_dais[] = {
+ {
+ .name = "adav80x-hifi",
+ .id = 0,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_PLAYBACK_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_CAPTURE_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .ops = &adav80x_dai_ops,
+ },
+ {
+ .name = "adav80x-aux",
+ .id = 1,
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_PLAYBACK_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Aux Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ADAV80X_CAPTURE_RATES,
+ .formats = ADAV80X_FORMATS,
+ },
+ .ops = &adav80x_dai_ops,
+ },
+};
+
+static int adav80x_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type);
+ if (ret) {
+ dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ /* Force PLLs on for SYSCLK output */
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+ /* Power down S/PDIF receiver, since it is currently not supported */
+ snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20);
+ /* Disable DAC zero flag */
+ snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6);
+
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int adav80x_resume(struct snd_soc_codec *codec)
+{
+ adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ codec->cache_sync = 1;
+ snd_soc_cache_sync(codec);
+
+ return 0;
+}
+
+static int adav80x_remove(struct snd_soc_codec *codec)
+{
+ return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static struct snd_soc_codec_driver adav80x_codec_driver = {
+ .probe = adav80x_probe,
+ .remove = adav80x_remove,
+ .suspend = adav80x_suspend,
+ .resume = adav80x_resume,
+ .set_bias_level = adav80x_set_bias_level,
+
+ .set_pll = adav80x_set_pll,
+ .set_sysclk = adav80x_set_sysclk,
+
+ .reg_word_size = sizeof(u8),
+ .reg_cache_size = ARRAY_SIZE(adav80x_default_regs),
+ .reg_cache_default = adav80x_default_regs,
+
+ .controls = adav80x_controls,
+ .num_controls = ARRAY_SIZE(adav80x_controls),
+ .dapm_widgets = adav80x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets),
+ .dapm_routes = adav80x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
+};
+
+static int __devinit adav80x_bus_probe(struct device *dev,
+ enum snd_soc_control_type control_type)
+{
+ struct adav80x *adav80x;
+ int ret;
+
+ adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
+ if (!adav80x)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, adav80x);
+ adav80x->control_type = control_type;
+
+ ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
+ adav80x_dais, ARRAY_SIZE(adav80x_dais));
+ if (ret)
+ kfree(adav80x);
+
+ return ret;
+}
+
+static int __devexit adav80x_bus_remove(struct device *dev)
+{
+ snd_soc_unregister_codec(dev);
+ kfree(dev_get_drvdata(dev));
+ return 0;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit adav80x_spi_probe(struct spi_device *spi)
+{
+ return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
+}
+
+static int __devexit adav80x_spi_remove(struct spi_device *spi)
+{
+ return adav80x_bus_remove(&spi->dev);
+}
+
+static struct spi_driver adav80x_spi_driver = {
+ .driver = {
+ .name = "adav801",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_spi_probe,
+ .remove = __devexit_p(adav80x_spi_remove),
+};
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct i2c_device_id adav80x_id[] = {
+ { "adav803", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adav80x_id);
+
+static int __devinit adav80x_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ return adav80x_bus_probe(&client->dev, SND_SOC_I2C);
+}
+
+static int __devexit adav80x_i2c_remove(struct i2c_client *client)
+{
+ return adav80x_bus_remove(&client->dev);
+}
+
+static struct i2c_driver adav80x_i2c_driver = {
+ .driver = {
+ .name = "adav803",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_i2c_probe,
+ .remove = __devexit_p(adav80x_i2c_remove),
+ .id_table = adav80x_id,
+};
+#endif
+
+static int __init adav80x_init(void)
+{
+ int ret = 0;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&adav80x_i2c_driver);
+ if (ret)
+ return ret;
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&adav80x_spi_driver);
+#endif
+
+ return ret;
+}
+module_init(adav80x_init);
+
+static void __exit adav80x_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&adav80x_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&adav80x_spi_driver);
+#endif
+}
+module_exit(adav80x_exit);
+
+MODULE_DESCRIPTION("ASoC ADAV80x driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h
new file mode 100644
index 0000000..adb0fc7
--- /dev/null
+++ b/sound/soc/codecs/adav80x.h
@@ -0,0 +1,35 @@
+/*
+ * header file for ADAV80X parts
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAV80X_H
+#define _ADAV80X_H
+
+enum adav80x_pll_src {
+ ADAV80X_PLL_SRC_XIN,
+ ADAV80X_PLL_SRC_XTAL,
+ ADAV80X_PLL_SRC_MCLKI,
+};
+
+enum adav80x_pll {
+ ADAV80X_PLL1 = 0,
+ ADAV80X_PLL2 = 1,
+};
+
+enum adav80x_clk_src {
+ ADAV80X_CLK_XIN = 0,
+ ADAV80X_CLK_MCLKI = 1,
+ ADAV80X_CLK_PLL1 = 2,
+ ADAV80X_CLK_PLL2 = 3,
+ ADAV80X_CLK_XTAL = 6,
+
+ ADAV80X_CLK_SYSCLK1 = 6,
+ ADAV80X_CLK_SYSCLK2 = 7,
+ ADAV80X_CLK_SYSCLK3 = 8,
+};
+
+#endif
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index ed96f247c..7a64e58 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -457,7 +457,7 @@
.set_sysclk = ak4641_set_dai_sysclk,
};
-struct snd_soc_dai_driver ak4641_dai[] = {
+static struct snd_soc_dai_driver ak4641_dai[] = {
{
.name = "ak4641-hifi",
.id = 1,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0206a17..6cc8678 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -636,10 +636,7 @@
#endif /* CONFIG_PM */
/*
- * ASoC codec device structure
- *
- * Assign this variable to the codec_dev field of the machine driver's
- * snd_soc_device structure.
+ * ASoC codec driver structure
*/
static const struct snd_soc_codec_driver soc_codec_device_cs4270 = {
.probe = cs4270_probe,
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 4173b67..ac65a2d 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1397,8 +1397,6 @@
if (freq == max98088->sysclk)
return 0;
- max98088->sysclk = freq; /* remember current sysclk */
-
/* Setup clocks for slave mode, and using the PLL
* PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
* 0x02 (when master clk is 20MHz to 30MHz)..
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index e1d282d..668434d 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1517,8 +1517,6 @@
if (freq == max98095->sysclk)
return 0;
- max98095->sysclk = freq; /* remember current sysclk */
-
/* Setup clocks for slave mode, and using the PLL
* PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
* 0x02 (when master clk is 20MHz to 40MHz)..
@@ -2261,11 +2259,11 @@
ret = snd_soc_read(codec, M98095_0FF_REV_ID);
if (ret < 0) {
- dev_err(codec->dev, "Failed to read device revision: %d\n",
+ dev_err(codec->dev, "Failure reading hardware revision: %d\n",
ret);
goto err_access;
}
- dev_info(codec->dev, "revision %c\n", ret + 'A');
+ dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A');
snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV);
@@ -2342,8 +2340,8 @@
max98095->control_data = i2c;
max98095->pdata = i2c->dev.platform_data;
- ret = snd_soc_register_codec(&i2c->dev,
- &soc_codec_dev_max98095, &max98095_dai[0], 3);
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095,
+ max98095_dai, ARRAY_SIZE(max98095_dai));
if (ret < 0)
kfree(max98095);
return ret;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
new file mode 100644
index 0000000..409d89d
--- /dev/null
+++ b/sound/soc/codecs/sta32x.c
@@ -0,0 +1,917 @@
+/*
+ * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on code from:
+ * Wolfson Microelectronics PLC.
+ * Mark Brown <broonie@opensource.wolfsonmicro.com>
+ * Freescale Semiconductor, Inc.
+ * Timur Tabi <timur@freescale.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "sta32x.h"
+
+#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | \
+ SNDRV_PCM_RATE_192000)
+
+#define STA32X_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE)
+
+/* Power-up register defaults */
+static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = {
+ 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60,
+ 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69,
+ 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d,
+ 0xc0, 0xf3, 0x33, 0x00, 0x0c,
+};
+
+/* regulator power supply names */
+static const char *sta32x_supply_names[] = {
+ "Vdda", /* analog supply, 3.3VV */
+ "Vdd3", /* digital supply, 3.3V */
+ "Vcc" /* power amp spply, 10V - 36V */
+};
+
+/* codec private data */
+struct sta32x_priv {
+ struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
+ struct snd_soc_codec *codec;
+
+ unsigned int mclk;
+ unsigned int format;
+};
+
+static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1);
+static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0);
+
+static const char *sta32x_drc_ac[] = {
+ "Anti-Clipping", "Dynamic Range Compression" };
+static const char *sta32x_auto_eq_mode[] = {
+ "User", "Preset", "Loudness" };
+static const char *sta32x_auto_gc_mode[] = {
+ "User", "AC no clipping", "AC limited clipping (10%)",
+ "DRC nighttime listening mode" };
+static const char *sta32x_auto_xo_mode[] = {
+ "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz",
+ "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" };
+static const char *sta32x_preset_eq_mode[] = {
+ "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft",
+ "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1",
+ "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2",
+ "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7",
+ "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12",
+ "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" };
+static const char *sta32x_limiter_select[] = {
+ "Limiter Disabled", "Limiter #1", "Limiter #2" };
+static const char *sta32x_limiter_attack_rate[] = {
+ "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024",
+ "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752",
+ "0.0645", "0.0564", "0.0501", "0.0451" };
+static const char *sta32x_limiter_release_rate[] = {
+ "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299",
+ "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137",
+ "0.0134", "0.0117", "0.0110", "0.0104" };
+
+static const unsigned int sta32x_limiter_ac_attack_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0),
+ 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0),
+};
+
+static const unsigned int sta32x_limiter_ac_release_tlv[] = {
+ TLV_DB_RANGE_HEAD(5),
+ 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0),
+ 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0),
+ 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0),
+};
+
+static const unsigned int sta32x_limiter_drc_attack_tlv[] = {
+ TLV_DB_RANGE_HEAD(3),
+ 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0),
+ 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0),
+ 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0),
+};
+
+static const unsigned int sta32x_limiter_drc_release_tlv[] = {
+ TLV_DB_RANGE_HEAD(5),
+ 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
+ 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0),
+ 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0),
+ 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0),
+ 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+};
+
+static const struct soc_enum sta32x_drc_ac_enum =
+ SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
+ 2, sta32x_drc_ac);
+static const struct soc_enum sta32x_auto_eq_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
+ 3, sta32x_auto_eq_mode);
+static const struct soc_enum sta32x_auto_gc_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
+ 4, sta32x_auto_gc_mode);
+static const struct soc_enum sta32x_auto_xo_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
+ 16, sta32x_auto_xo_mode);
+static const struct soc_enum sta32x_preset_eq_enum =
+ SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
+ 32, sta32x_preset_eq_mode);
+static const struct soc_enum sta32x_limiter_ch1_enum =
+ SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
+ 3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter_ch2_enum =
+ SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
+ 3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter_ch3_enum =
+ SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
+ 3, sta32x_limiter_select);
+static const struct soc_enum sta32x_limiter1_attack_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT,
+ 16, sta32x_limiter_attack_rate);
+static const struct soc_enum sta32x_limiter2_attack_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT,
+ 16, sta32x_limiter_attack_rate);
+static const struct soc_enum sta32x_limiter1_release_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT,
+ 16, sta32x_limiter_release_rate);
+static const struct soc_enum sta32x_limiter2_release_rate_enum =
+ SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT,
+ 16, sta32x_limiter_release_rate);
+
+/* byte array controls for setting biquad, mixer, scaling coefficients;
+ * for biquads all five coefficients need to be set in one go,
+ * mixer and pre/postscale coefs can be set individually;
+ * each coef is 24bit, the bytes are ordered in the same way
+ * as given in the STA32x data sheet (big endian; b1, b2, a1, a2, b0)
+ */
+
+static int sta32x_coefficient_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int numcoef = kcontrol->private_value >> 16;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = 3 * numcoef;
+ return 0;
+}
+
+static int sta32x_coefficient_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int numcoef = kcontrol->private_value >> 16;
+ int index = kcontrol->private_value & 0xffff;
+ unsigned int cfud;
+ int i;
+
+ /* preserve reserved bits in STA32X_CFUD */
+ cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0;
+ /* chip documentation does not say if the bits are self clearing,
+ * so do it explicitly */
+ snd_soc_write(codec, STA32X_CFUD, cfud);
+
+ snd_soc_write(codec, STA32X_CFADDR2, index);
+ if (numcoef == 1)
+ snd_soc_write(codec, STA32X_CFUD, cfud | 0x04);
+ else if (numcoef == 5)
+ snd_soc_write(codec, STA32X_CFUD, cfud | 0x08);
+ else
+ return -EINVAL;
+ for (i = 0; i < 3 * numcoef; i++)
+ ucontrol->value.bytes.data[i] =
+ snd_soc_read(codec, STA32X_B1CF1 + i);
+
+ return 0;
+}
+
+static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int numcoef = kcontrol->private_value >> 16;
+ int index = kcontrol->private_value & 0xffff;
+ unsigned int cfud;
+ int i;
+
+ /* preserve reserved bits in STA32X_CFUD */
+ cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0;
+ /* chip documentation does not say if the bits are self clearing,
+ * so do it explicitly */
+ snd_soc_write(codec, STA32X_CFUD, cfud);
+
+ snd_soc_write(codec, STA32X_CFADDR2, index);
+ for (i = 0; i < 3 * numcoef; i++)
+ snd_soc_write(codec, STA32X_B1CF1 + i,
+ ucontrol->value.bytes.data[i]);
+ if (numcoef == 1)
+ snd_soc_write(codec, STA32X_CFUD, cfud | 0x01);
+ else if (numcoef == 5)
+ snd_soc_write(codec, STA32X_CFUD, cfud | 0x02);
+ else
+ return -EINVAL;
+
+ return 0;
+}
+
+#define SINGLE_COEF(xname, index) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = sta32x_coefficient_info, \
+ .get = sta32x_coefficient_get,\
+ .put = sta32x_coefficient_put, \
+ .private_value = index | (1 << 16) }
+
+#define BIQUAD_COEFS(xname, index) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = sta32x_coefficient_info, \
+ .get = sta32x_coefficient_get,\
+ .put = sta32x_coefficient_put, \
+ .private_value = index | (5 << 16) }
+
+static const struct snd_kcontrol_new sta32x_snd_controls[] = {
+SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv),
+SOC_SINGLE("Master Switch", STA32X_MMUTE, 0, 1, 1),
+SOC_SINGLE("Ch1 Switch", STA32X_MMUTE, 1, 1, 1),
+SOC_SINGLE("Ch2 Switch", STA32X_MMUTE, 2, 1, 1),
+SOC_SINGLE("Ch3 Switch", STA32X_MMUTE, 3, 1, 1),
+SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv),
+SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0),
+SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum),
+SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0),
+SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0),
+SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0),
+SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0),
+SOC_ENUM("Automode EQ", sta32x_auto_eq_enum),
+SOC_ENUM("Automode GC", sta32x_auto_gc_enum),
+SOC_ENUM("Automode XO", sta32x_auto_xo_enum),
+SOC_ENUM("Preset EQ", sta32x_preset_eq_enum),
+SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0),
+SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0),
+SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum),
+SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum),
+SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum),
+SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv),
+SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv),
+SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum),
+SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum),
+SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+
+/* depending on mode, the attack/release thresholds have
+ * two different enum definitions; provide both
+ */
+SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_ac_attack_tlv),
+SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_ac_attack_tlv),
+SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_ac_release_tlv),
+SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_ac_release_tlv),
+SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_drc_attack_tlv),
+SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT,
+ 16, 0, sta32x_limiter_drc_attack_tlv),
+SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_drc_release_tlv),
+SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT,
+ 16, 0, sta32x_limiter_drc_release_tlv),
+
+BIQUAD_COEFS("Ch1 - Biquad 1", 0),
+BIQUAD_COEFS("Ch1 - Biquad 2", 5),
+BIQUAD_COEFS("Ch1 - Biquad 3", 10),
+BIQUAD_COEFS("Ch1 - Biquad 4", 15),
+BIQUAD_COEFS("Ch2 - Biquad 1", 20),
+BIQUAD_COEFS("Ch2 - Biquad 2", 25),
+BIQUAD_COEFS("Ch2 - Biquad 3", 30),
+BIQUAD_COEFS("Ch2 - Biquad 4", 35),
+BIQUAD_COEFS("High-pass", 40),
+BIQUAD_COEFS("Low-pass", 45),
+SINGLE_COEF("Ch1 - Prescale", 50),
+SINGLE_COEF("Ch2 - Prescale", 51),
+SINGLE_COEF("Ch1 - Postscale", 52),
+SINGLE_COEF("Ch2 - Postscale", 53),
+SINGLE_COEF("Ch3 - Postscale", 54),
+SINGLE_COEF("Thermal warning - Postscale", 55),
+SINGLE_COEF("Ch1 - Mix 1", 56),
+SINGLE_COEF("Ch1 - Mix 2", 57),
+SINGLE_COEF("Ch2 - Mix 1", 58),
+SINGLE_COEF("Ch2 - Mix 2", 59),
+SINGLE_COEF("Ch3 - Mix 1", 60),
+SINGLE_COEF("Ch3 - Mix 2", 61),
+};
+
+static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("LEFT"),
+SND_SOC_DAPM_OUTPUT("RIGHT"),
+SND_SOC_DAPM_OUTPUT("SUB"),
+};
+
+static const struct snd_soc_dapm_route sta32x_dapm_routes[] = {
+ { "LEFT", NULL, "DAC" },
+ { "RIGHT", NULL, "DAC" },
+ { "SUB", NULL, "DAC" },
+};
+
+/* MCLK interpolation ratio per fs */
+static struct {
+ int fs;
+ int ir;
+} interpolation_ratios[] = {
+ { 32000, 0 },
+ { 44100, 0 },
+ { 48000, 0 },
+ { 88200, 1 },
+ { 96000, 1 },
+ { 176400, 2 },
+ { 192000, 2 },
+};
+
+/* MCLK to fs clock ratios */
+static struct {
+ int ratio;
+ int mcs;
+} mclk_ratios[3][7] = {
+ { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 },
+ { 128, 4 }, { 576, 5 }, { 0, 0 } },
+ { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } },
+ { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } },
+};
+
+
+/**
+ * sta32x_set_dai_sysclk - configure MCLK
+ * @codec_dai: the codec DAI
+ * @clk_id: the clock ID (ignored)
+ * @freq: the MCLK input frequency
+ * @dir: the clock direction (ignored)
+ *
+ * The value of MCLK is used to determine which sample rates are supported
+ * by the STA32X, based on the mclk_ratios table.
+ *
+ * This function must be called by the machine driver's 'startup' function,
+ * otherwise the list of supported sample rates will not be available in
+ * time for ALSA.
+ *
+ * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause
+ * theoretically possible sample rates to be enabled. Call it again with a
+ * proper value set one the external clock is set (most probably you would do
+ * that from a machine's driver 'hw_param' hook.
+ */
+static int sta32x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ int i, j, ir, fs;
+ unsigned int rates = 0;
+ unsigned int rate_min = -1;
+ unsigned int rate_max = 0;
+
+ pr_debug("mclk=%u\n", freq);
+ sta32x->mclk = freq;
+
+ if (sta32x->mclk) {
+ for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) {
+ ir = interpolation_ratios[i].ir;
+ fs = interpolation_ratios[i].fs;
+ for (j = 0; mclk_ratios[ir][j].ratio; j++) {
+ if (mclk_ratios[ir][j].ratio * fs == freq) {
+ rates |= snd_pcm_rate_to_rate_bit(fs);
+ if (fs < rate_min)
+ rate_min = fs;
+ if (fs > rate_max)
+ rate_max = fs;
+ }
+ }
+ }
+ /* FIXME: soc should support a rate list */
+ rates &= ~SNDRV_PCM_RATE_KNOT;
+
+ if (!rates) {
+ dev_err(codec->dev, "could not find a valid sample rate\n");
+ return -EINVAL;
+ }
+ } else {
+ /* enable all possible rates */
+ rates = STA32X_RATES;
+ rate_min = 32000;
+ rate_max = 192000;
+ }
+
+ codec_dai->driver->playback.rates = rates;
+ codec_dai->driver->playback.rate_min = rate_min;
+ codec_dai->driver->playback.rate_max = rate_max;
+ return 0;
+}
+
+/**
+ * sta32x_set_dai_fmt - configure the codec for the selected audio format
+ * @codec_dai: the codec DAI
+ * @fmt: a SND_SOC_DAIFMT_x value indicating the data format
+ *
+ * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the
+ * codec accordingly.
+ */
+static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ u8 confb = snd_soc_read(codec, STA32X_CONFB);
+
+ pr_debug("\n");
+ confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ confb |= STA32X_CONFB_C2IM;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ confb |= STA32X_CONFB_C1IM;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, STA32X_CONFB, confb);
+ return 0;
+}
+
+/**
+ * sta32x_hw_params - program the STA32X with the given hardware parameters.
+ * @substream: the audio stream
+ * @params: the hardware parameters to set
+ * @dai: the SOC DAI (ignored)
+ *
+ * This function programs the hardware with the values provided.
+ * Specifically, the sample rate and the data format.
+ */
+static int sta32x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ unsigned int rate;
+ int i, mcs = -1, ir = -1;
+ u8 confa, confb;
+
+ rate = params_rate(params);
+ pr_debug("rate: %u\n", rate);
+ for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++)
+ if (interpolation_ratios[i].fs == rate)
+ ir = interpolation_ratios[i].ir;
+ if (ir < 0)
+ return -EINVAL;
+ for (i = 0; mclk_ratios[ir][i].ratio; i++)
+ if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk)
+ mcs = mclk_ratios[ir][i].mcs;
+ if (mcs < 0)
+ return -EINVAL;
+
+ confa = snd_soc_read(codec, STA32X_CONFA);
+ confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK);
+ confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT);
+
+ confb = snd_soc_read(codec, STA32X_CONFB);
+ confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB);
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S24_BE:
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ case SNDRV_PCM_FORMAT_S24_3BE:
+ pr_debug("24bit\n");
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32_LE:
+ case SNDRV_PCM_FORMAT_S32_BE:
+ pr_debug("24bit or 32bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0x1;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0x2;
+ break;
+ }
+
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ case SNDRV_PCM_FORMAT_S20_3BE:
+ pr_debug("20bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x4;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0x5;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0x6;
+ break;
+ }
+
+ break;
+ case SNDRV_PCM_FORMAT_S18_3LE:
+ case SNDRV_PCM_FORMAT_S18_3BE:
+ pr_debug("18bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x8;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0x9;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0xa;
+ break;
+ }
+
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_S16_BE:
+ pr_debug("16bit\n");
+ switch (sta32x->format) {
+ case SND_SOC_DAIFMT_I2S:
+ confb |= 0x0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ confb |= 0xd;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ confb |= 0xe;
+ break;
+ }
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, STA32X_CONFA, confa);
+ snd_soc_write(codec, STA32X_CONFB, confb);
+ return 0;
+}
+
+/**
+ * sta32x_set_bias_level - DAPM callback
+ * @codec: the codec device
+ * @level: DAPM power level
+ *
+ * This is called by ALSA to put the codec into low power mode
+ * or to wake it up. If the codec is powered off completely
+ * all registers must be restored after power on.
+ */
+static int sta32x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int ret;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("level = %d\n", level);
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* Full power on */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ snd_soc_cache_sync(codec);
+ }
+
+ /* Power up to mute */
+ /* FIXME */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD);
+
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* The chip runs through the power down sequence for us. */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
+ STA32X_CONFF_PWDN);
+ msleep(300);
+
+ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+static struct snd_soc_dai_ops sta32x_dai_ops = {
+ .hw_params = sta32x_hw_params,
+ .set_sysclk = sta32x_set_dai_sysclk,
+ .set_fmt = sta32x_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver sta32x_dai = {
+ .name = "STA32X",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = STA32X_RATES,
+ .formats = STA32X_FORMATS,
+ },
+ .ops = &sta32x_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int sta32x_resume(struct snd_soc_codec *codec)
+{
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+#else
+#define sta32x_suspend NULL
+#define sta32x_resume NULL
+#endif
+
+static int sta32x_probe(struct snd_soc_codec *codec)
+{
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ int i, ret = 0;
+
+ sta32x->codec = codec;
+
+ /* regulators */
+ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++)
+ sta32x->supplies[i].supply = sta32x_supply_names[i];
+
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+ goto err;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
+ sta32x->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+ goto err_get;
+ }
+
+ /* Tell ASoC what kind of I/O to use to read the registers. ASoC will
+ * then do the I2C transactions itself.
+ */
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret);
+ return ret;
+ }
+
+ /* read reg reset values into cache */
+ for (i = 0; i < STA32X_REGISTER_COUNT; i++)
+ snd_soc_cache_write(codec, i, sta32x_regs[i]);
+
+ /* preserve reset values of reserved register bits */
+ snd_soc_cache_write(codec, STA32X_CONFC,
+ codec->hw_read(codec, STA32X_CONFC));
+ snd_soc_cache_write(codec, STA32X_CONFE,
+ codec->hw_read(codec, STA32X_CONFE));
+ snd_soc_cache_write(codec, STA32X_CONFF,
+ codec->hw_read(codec, STA32X_CONFF));
+ snd_soc_cache_write(codec, STA32X_MMUTE,
+ codec->hw_read(codec, STA32X_MMUTE));
+ snd_soc_cache_write(codec, STA32X_AUTO1,
+ codec->hw_read(codec, STA32X_AUTO1));
+ snd_soc_cache_write(codec, STA32X_AUTO3,
+ codec->hw_read(codec, STA32X_AUTO3));
+ snd_soc_cache_write(codec, STA32X_C3CFG,
+ codec->hw_read(codec, STA32X_C3CFG));
+
+ /* FIXME enable thermal warning adjustment and recovery */
+ snd_soc_update_bits(codec, STA32X_CONFA,
+ STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0);
+
+ /* FIXME select 2.1 mode */
+ snd_soc_update_bits(codec, STA32X_CONFF,
+ STA32X_CONFF_OCFG_MASK,
+ 1 << STA32X_CONFF_OCFG_SHIFT);
+
+ /* FIXME channel to output mapping */
+ snd_soc_update_bits(codec, STA32X_C1CFG,
+ STA32X_CxCFG_OM_MASK,
+ 0 << STA32X_CxCFG_OM_SHIFT);
+ snd_soc_update_bits(codec, STA32X_C2CFG,
+ STA32X_CxCFG_OM_MASK,
+ 1 << STA32X_CxCFG_OM_SHIFT);
+ snd_soc_update_bits(codec, STA32X_C3CFG,
+ STA32X_CxCFG_OM_MASK,
+ 2 << STA32X_CxCFG_OM_SHIFT);
+
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ /* Bias level configuration will have done an extra enable */
+ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+ return 0;
+
+err_get:
+ regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+err:
+ return ret;
+}
+
+static int sta32x_remove(struct snd_soc_codec *codec)
+{
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+
+ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+ return 0;
+}
+
+static int sta32x_reg_is_volatile(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ case STA32X_CONFA ... STA32X_L2ATRT:
+ case STA32X_MPCC1 ... STA32X_FDRC2:
+ return 0;
+ }
+ return 1;
+}
+
+static const struct snd_soc_codec_driver sta32x_codec = {
+ .probe = sta32x_probe,
+ .remove = sta32x_remove,
+ .suspend = sta32x_suspend,
+ .resume = sta32x_resume,
+ .reg_cache_size = STA32X_REGISTER_COUNT,
+ .reg_word_size = sizeof(u8),
+ .volatile_register = sta32x_reg_is_volatile,
+ .set_bias_level = sta32x_set_bias_level,
+ .controls = sta32x_snd_controls,
+ .num_controls = ARRAY_SIZE(sta32x_snd_controls),
+ .dapm_widgets = sta32x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets),
+ .dapm_routes = sta32x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes),
+};
+
+static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct sta32x_priv *sta32x;
+ int ret;
+
+ sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL);
+ if (!sta32x)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, sta32x);
+
+ ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static __devexit int sta32x_i2c_remove(struct i2c_client *client)
+{
+ struct sta32x_priv *sta32x = i2c_get_clientdata(client);
+ struct snd_soc_codec *codec = sta32x->codec;
+
+ if (codec)
+ sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+ if (codec) {
+ snd_soc_unregister_codec(&client->dev);
+ snd_soc_codec_set_drvdata(codec, NULL);
+ }
+
+ kfree(sta32x);
+ return 0;
+}
+
+static const struct i2c_device_id sta32x_i2c_id[] = {
+ { "sta326", 0 },
+ { "sta328", 0 },
+ { "sta329", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id);
+
+static struct i2c_driver sta32x_i2c_driver = {
+ .driver = {
+ .name = "sta32x",
+ .owner = THIS_MODULE,
+ },
+ .probe = sta32x_i2c_probe,
+ .remove = __devexit_p(sta32x_i2c_remove),
+ .id_table = sta32x_i2c_id,
+};
+
+static int __init sta32x_init(void)
+{
+ return i2c_add_driver(&sta32x_i2c_driver);
+}
+module_init(sta32x_init);
+
+static void __exit sta32x_exit(void)
+{
+ i2c_del_driver(&sta32x_i2c_driver);
+}
+module_exit(sta32x_exit);
+
+MODULE_DESCRIPTION("ASoC STA32X driver");
+MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
new file mode 100644
index 0000000..b97ee5a
--- /dev/null
+++ b/sound/soc/codecs/sta32x.h
@@ -0,0 +1,210 @@
+/*
+ * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on code from:
+ * Wolfson Microelectronics PLC.
+ * Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+#ifndef _ASOC_STA_32X_H
+#define _ASOC_STA_32X_H
+
+/* STA326 register addresses */
+
+#define STA32X_REGISTER_COUNT 0x2d
+
+#define STA32X_CONFA 0x00
+#define STA32X_CONFB 0x01
+#define STA32X_CONFC 0x02
+#define STA32X_CONFD 0x03
+#define STA32X_CONFE 0x04
+#define STA32X_CONFF 0x05
+#define STA32X_MMUTE 0x06
+#define STA32X_MVOL 0x07
+#define STA32X_C1VOL 0x08
+#define STA32X_C2VOL 0x09
+#define STA32X_C3VOL 0x0a
+#define STA32X_AUTO1 0x0b
+#define STA32X_AUTO2 0x0c
+#define STA32X_AUTO3 0x0d
+#define STA32X_C1CFG 0x0e
+#define STA32X_C2CFG 0x0f
+#define STA32X_C3CFG 0x10
+#define STA32X_TONE 0x11
+#define STA32X_L1AR 0x12
+#define STA32X_L1ATRT 0x13
+#define STA32X_L2AR 0x14
+#define STA32X_L2ATRT 0x15
+#define STA32X_CFADDR2 0x16
+#define STA32X_B1CF1 0x17
+#define STA32X_B1CF2 0x18
+#define STA32X_B1CF3 0x19
+#define STA32X_B2CF1 0x1a
+#define STA32X_B2CF2 0x1b
+#define STA32X_B2CF3 0x1c
+#define STA32X_A1CF1 0x1d
+#define STA32X_A1CF2 0x1e
+#define STA32X_A1CF3 0x1f
+#define STA32X_A2CF1 0x20
+#define STA32X_A2CF2 0x21
+#define STA32X_A2CF3 0x22
+#define STA32X_B0CF1 0x23
+#define STA32X_B0CF2 0x24
+#define STA32X_B0CF3 0x25
+#define STA32X_CFUD 0x26
+#define STA32X_MPCC1 0x27
+#define STA32X_MPCC2 0x28
+/* Reserved 0x29 */
+/* Reserved 0x2a */
+#define STA32X_Reserved 0x2a
+#define STA32X_FDRC1 0x2b
+#define STA32X_FDRC2 0x2c
+/* Reserved 0x2d */
+
+
+/* STA326 register field definitions */
+
+/* 0x00 CONFA */
+#define STA32X_CONFA_MCS_MASK 0x03
+#define STA32X_CONFA_MCS_SHIFT 0
+#define STA32X_CONFA_IR_MASK 0x18
+#define STA32X_CONFA_IR_SHIFT 3
+#define STA32X_CONFA_TWRB 0x20
+#define STA32X_CONFA_TWAB 0x40
+#define STA32X_CONFA_FDRB 0x80
+
+/* 0x01 CONFB */
+#define STA32X_CONFB_SAI_MASK 0x0f
+#define STA32X_CONFB_SAI_SHIFT 0
+#define STA32X_CONFB_SAIFB 0x10
+#define STA32X_CONFB_DSCKE 0x20
+#define STA32X_CONFB_C1IM 0x40
+#define STA32X_CONFB_C2IM 0x80
+
+/* 0x02 CONFC */
+#define STA32X_CONFC_OM_MASK 0x03
+#define STA32X_CONFC_OM_SHIFT 0
+#define STA32X_CONFC_CSZ_MASK 0x7c
+#define STA32X_CONFC_CSZ_SHIFT 2
+
+/* 0x03 CONFD */
+#define STA32X_CONFD_HPB 0x01
+#define STA32X_CONFD_HPB_SHIFT 0
+#define STA32X_CONFD_DEMP 0x02
+#define STA32X_CONFD_DEMP_SHIFT 1
+#define STA32X_CONFD_DSPB 0x04
+#define STA32X_CONFD_DSPB_SHIFT 2
+#define STA32X_CONFD_PSL 0x08
+#define STA32X_CONFD_PSL_SHIFT 3
+#define STA32X_CONFD_BQL 0x10
+#define STA32X_CONFD_BQL_SHIFT 4
+#define STA32X_CONFD_DRC 0x20
+#define STA32X_CONFD_DRC_SHIFT 5
+#define STA32X_CONFD_ZDE 0x40
+#define STA32X_CONFD_ZDE_SHIFT 6
+#define STA32X_CONFD_MME 0x80
+#define STA32X_CONFD_MME_SHIFT 7
+
+/* 0x04 CONFE */
+#define STA32X_CONFE_MPCV 0x01
+#define STA32X_CONFE_MPCV_SHIFT 0
+#define STA32X_CONFE_MPC 0x02
+#define STA32X_CONFE_MPC_SHIFT 1
+#define STA32X_CONFE_AME 0x08
+#define STA32X_CONFE_AME_SHIFT 3
+#define STA32X_CONFE_PWMS 0x10
+#define STA32X_CONFE_PWMS_SHIFT 4
+#define STA32X_CONFE_ZCE 0x40
+#define STA32X_CONFE_ZCE_SHIFT 6
+#define STA32X_CONFE_SVE 0x80
+#define STA32X_CONFE_SVE_SHIFT 7
+
+/* 0x05 CONFF */
+#define STA32X_CONFF_OCFG_MASK 0x03
+#define STA32X_CONFF_OCFG_SHIFT 0
+#define STA32X_CONFF_IDE 0x04
+#define STA32X_CONFF_IDE_SHIFT 3
+#define STA32X_CONFF_BCLE 0x08
+#define STA32X_CONFF_ECLE 0x20
+#define STA32X_CONFF_PWDN 0x40
+#define STA32X_CONFF_EAPD 0x80
+
+/* 0x06 MMUTE */
+#define STA32X_MMUTE_MMUTE 0x01
+
+/* 0x0b AUTO1 */
+#define STA32X_AUTO1_AMEQ_MASK 0x03
+#define STA32X_AUTO1_AMEQ_SHIFT 0
+#define STA32X_AUTO1_AMV_MASK 0xc0
+#define STA32X_AUTO1_AMV_SHIFT 2
+#define STA32X_AUTO1_AMGC_MASK 0x30
+#define STA32X_AUTO1_AMGC_SHIFT 4
+#define STA32X_AUTO1_AMPS 0x80
+
+/* 0x0c AUTO2 */
+#define STA32X_AUTO2_AMAME 0x01
+#define STA32X_AUTO2_AMAM_MASK 0x0e
+#define STA32X_AUTO2_AMAM_SHIFT 1
+#define STA32X_AUTO2_XO_MASK 0xf0
+#define STA32X_AUTO2_XO_SHIFT 4
+
+/* 0x0d AUTO3 */
+#define STA32X_AUTO3_PEQ_MASK 0x1f
+#define STA32X_AUTO3_PEQ_SHIFT 0
+
+/* 0x0e 0x0f 0x10 CxCFG */
+#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */
+#define STA32X_CxCFG_TCB_SHIFT 0
+#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */
+#define STA32X_CxCFG_EQBP_SHIFT 1
+#define STA32X_CxCFG_VBP 0x03
+#define STA32X_CxCFG_VBP_SHIFT 2
+#define STA32X_CxCFG_BO 0x04
+#define STA32X_CxCFG_LS_MASK 0x30
+#define STA32X_CxCFG_LS_SHIFT 4
+#define STA32X_CxCFG_OM_MASK 0xc0
+#define STA32X_CxCFG_OM_SHIFT 6
+
+/* 0x11 TONE */
+#define STA32X_TONE_BTC_SHIFT 0
+#define STA32X_TONE_TTC_SHIFT 4
+
+/* 0x12 0x13 0x14 0x15 limiter attack/release */
+#define STA32X_LxA_SHIFT 0
+#define STA32X_LxR_SHIFT 4
+
+/* 0x26 CFUD */
+#define STA32X_CFUD_W1 0x01
+#define STA32X_CFUD_WA 0x02
+#define STA32X_CFUD_R1 0x04
+#define STA32X_CFUD_RA 0x08
+
+
+/* biquad filter coefficient table offsets */
+#define STA32X_C1_BQ_BASE 0
+#define STA32X_C2_BQ_BASE 20
+#define STA32X_CH_BQ_NUM 4
+#define STA32X_BQ_NUM_COEF 5
+#define STA32X_XO_HP_BQ_BASE 40
+#define STA32X_XO_LP_BQ_BASE 45
+#define STA32X_C1_PRESCALE 50
+#define STA32X_C2_PRESCALE 51
+#define STA32X_C1_POSTSCALE 52
+#define STA32X_C2_POSTSCALE 53
+#define STA32X_C3_POSTSCALE 54
+#define STA32X_TW_POSTSCALE 55
+#define STA32X_C1_MIX1 56
+#define STA32X_C1_MIX2 57
+#define STA32X_C2_MIX1 58
+#define STA32X_C2_MIX2 59
+#define STA32X_C3_MIX1 60
+#define STA32X_C3_MIX2 61
+
+#endif /* _ASOC_STA_32X_H */
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 789453d..0963c4c 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -226,11 +226,13 @@
#define RDAC_ENUM 1
#define LHPCOM_ENUM 2
#define RHPCOM_ENUM 3
-#define LINE1L_ENUM 4
-#define LINE1R_ENUM 5
-#define LINE2L_ENUM 6
-#define LINE2R_ENUM 7
-#define ADC_HPF_ENUM 8
+#define LINE1L_2_L_ENUM 4
+#define LINE1L_2_R_ENUM 5
+#define LINE1R_2_L_ENUM 6
+#define LINE1R_2_R_ENUM 7
+#define LINE2L_ENUM 8
+#define LINE2R_ENUM 9
+#define ADC_HPF_ENUM 10
static const struct soc_enum aic3x_enum[] = {
SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
@@ -238,6 +240,8 @@
SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux),
SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux),
SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+ SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+ SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
@@ -490,12 +494,16 @@
};
/* Left Line1 Mux */
-static const struct snd_kcontrol_new aic3x_left_line1_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_ENUM]);
+static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]);
+static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]);
/* Right Line1 Mux */
-static const struct snd_kcontrol_new aic3x_right_line1_mux_controls =
-SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_ENUM]);
+static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]);
+static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]);
/* Left Line2 Mux */
static const struct snd_kcontrol_new aic3x_left_line2_mux_controls =
@@ -535,9 +543,9 @@
&aic3x_left_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
- &aic3x_left_line1_mux_controls),
+ &aic3x_left_line1l_mux_controls),
SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0,
- &aic3x_left_line1_mux_controls),
+ &aic3x_left_line1r_mux_controls),
SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line2_mux_controls),
@@ -548,9 +556,9 @@
&aic3x_right_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0,
- &aic3x_right_line1_mux_controls),
+ &aic3x_right_line1l_mux_controls),
SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
- &aic3x_right_line1_mux_controls),
+ &aic3x_right_line1r_mux_controls),
SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_line2_mux_controls),
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 4c33663..cd63bba 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -954,9 +954,9 @@
/*
* MICGAIN volume control:
- * from -6 to 30 dB in 6 dB steps
+ * from 6 to 30 dB in 6 dB steps
*/
-static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0);
+static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0);
/*
* AFMGAIN volume control:
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
new file mode 100644
index 0000000..a2a09f8
--- /dev/null
+++ b/sound/soc/codecs/wm8782.c
@@ -0,0 +1,80 @@
+/*
+ * sound/soc/codecs/wm8782.c
+ * simple, strap-pin configured 24bit 2ch ADC
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * based on ad73311.c
+ * Copyright: Analog Device Inc.
+ * Author: Cliff Cai <cliff.cai@analog.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver wm8782_dai = {
+ .name = "wm8782",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ /* For configurations with FSAMPEN=0 */
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8782;
+
+static __devinit int wm8782_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev,
+ &soc_codec_dev_wm8782, &wm8782_dai, 1);
+}
+
+static int __devexit wm8782_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver wm8782_codec_driver = {
+ .driver = {
+ .name = "wm8782",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8782_probe,
+ .remove = wm8782_remove,
+};
+
+static int __init wm8782_init(void)
+{
+ return platform_driver_register(&wm8782_codec_driver);
+}
+module_init(wm8782_init);
+
+static void __exit wm8782_exit(void)
+{
+ platform_driver_unregister(&wm8782_codec_driver);
+}
+module_exit(wm8782_exit);
+
+MODULE_DESCRIPTION("ASoC WM8782 driver");
+MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 449ea09..082040e 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1167,6 +1167,7 @@
ret = wm8900_set_fll(codec, 0, fll_in, fll_out);
if (ret != 0) {
dev_err(codec->dev, "Failed to restart FLL\n");
+ kfree(cache);
return ret;
}
}
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9b3bba4..b085575 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2560,6 +2560,7 @@
static const struct i2c_device_id wm8904_i2c_id[] = {
{ "wm8904", WM8904 },
{ "wm8912", WM8912 },
+ { "wm8918", WM8904 }, /* Actually a subset, updates to follow */
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id);
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index e2ab4fa..423baa9 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -41,14 +41,12 @@
#define HPOUT2L 4
#define HPOUT2R 8
-#define WM8915_NUM_SUPPLIES 6
+#define WM8915_NUM_SUPPLIES 4
static const char *wm8915_supply_names[WM8915_NUM_SUPPLIES] = {
- "DCVDD",
"DBVDD",
"AVDD1",
"AVDD2",
"CPVDD",
- "MICVDD",
};
struct wm8915_priv {
@@ -57,6 +55,7 @@
int ldo1ena;
int sysclk;
+ int sysclk_src;
int fll_src;
int fll_fref;
@@ -76,6 +75,7 @@
struct wm8915_pdata pdata;
int rx_rate[WM8915_AIFS];
+ int bclk_rate[WM8915_AIFS];
/* Platform dependant ReTune mobile configuration */
int num_retune_mobile_texts;
@@ -113,8 +113,6 @@
WM8915_REGULATOR_EVENT(1)
WM8915_REGULATOR_EVENT(2)
WM8915_REGULATOR_EVENT(3)
-WM8915_REGULATOR_EVENT(4)
-WM8915_REGULATOR_EVENT(5)
static const u16 wm8915_reg[WM8915_MAX_REGISTER] = {
[WM8915_SOFTWARE_RESET] = 0x8915,
@@ -1565,6 +1563,50 @@
return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915);
}
+static const int bclk_divs[] = {
+ 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
+};
+
+static void wm8915_update_bclk(struct snd_soc_codec *codec)
+{
+ struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+ int aif, best, cur_val, bclk_rate, bclk_reg, i;
+
+ /* Don't bother if we're in a low frequency idle mode that
+ * can't support audio.
+ */
+ if (wm8915->sysclk < 64000)
+ return;
+
+ for (aif = 0; aif < WM8915_AIFS; aif++) {
+ switch (aif) {
+ case 0:
+ bclk_reg = WM8915_AIF1_BCLK;
+ break;
+ case 1:
+ bclk_reg = WM8915_AIF2_BCLK;
+ break;
+ }
+
+ bclk_rate = wm8915->bclk_rate[aif];
+
+ /* Pick a divisor for BCLK as close as we can get to ideal */
+ best = 0;
+ for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+ cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
+ if (cur_val < 0) /* BCLK table is sorted */
+ break;
+ best = i;
+ }
+ bclk_rate = wm8915->sysclk / bclk_divs[best];
+ dev_dbg(codec->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
+ bclk_divs[best], bclk_rate);
+
+ snd_soc_update_bits(codec, bclk_reg,
+ WM8915_AIF1_BCLK_DIV_MASK, best);
+ }
+}
+
static int wm8915_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -1717,10 +1759,6 @@
return 0;
}
-static const int bclk_divs[] = {
- 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
-};
-
static const int dsp_divs[] = {
48000, 32000, 16000, 8000
};
@@ -1731,17 +1769,11 @@
{
struct snd_soc_codec *codec = dai->codec;
struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
- int bits, i, bclk_rate, best, cur_val;
+ int bits, i, bclk_rate;
int aifdata = 0;
- int bclk = 0;
int lrclk = 0;
int dsp = 0;
- int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift;
-
- if (!wm8915->sysclk) {
- dev_err(codec->dev, "SYSCLK not configured\n");
- return -EINVAL;
- }
+ int aifdata_reg, lrclk_reg, dsp_shift;
switch (dai->id) {
case 0:
@@ -1753,7 +1785,6 @@
aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1;
lrclk_reg = WM8915_AIF1_TX_LRCLK_1;
}
- bclk_reg = WM8915_AIF1_BCLK;
dsp_shift = 0;
break;
case 1:
@@ -1765,7 +1796,6 @@
aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1;
lrclk_reg = WM8915_AIF2_TX_LRCLK_1;
}
- bclk_reg = WM8915_AIF2_BCLK;
dsp_shift = WM8915_DSP2_DIV_SHIFT;
break;
default:
@@ -1779,6 +1809,9 @@
return bclk_rate;
}
+ wm8915->bclk_rate[dai->id] = bclk_rate;
+ wm8915->rx_rate[dai->id] = params_rate(params);
+
/* Needs looking at for TDM */
bits = snd_pcm_format_width(params_format(params));
if (bits < 0)
@@ -1796,18 +1829,7 @@
}
dsp |= i << dsp_shift;
- /* Pick a divisor for BCLK as close as we can get to ideal */
- best = 0;
- for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
- cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate;
- if (cur_val < 0) /* BCLK table is sorted */
- break;
- best = i;
- }
- bclk_rate = wm8915->sysclk / bclk_divs[best];
- dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
- bclk_divs[best], bclk_rate);
- bclk |= best;
+ wm8915_update_bclk(codec);
lrclk = bclk_rate / params_rate(params);
dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
@@ -1817,14 +1839,11 @@
WM8915_AIF1TX_WL_MASK |
WM8915_AIF1TX_SLOT_LEN_MASK,
aifdata);
- snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk);
snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK,
lrclk);
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2,
WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp);
- wm8915->rx_rate[dai->id] = params_rate(params);
-
return 0;
}
@@ -1838,6 +1857,9 @@
int src;
int old;
+ if (freq == wm8915->sysclk && clk_id == wm8915->sysclk_src)
+ return 0;
+
/* Disable SYSCLK while we reconfigure */
old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
@@ -1882,6 +1904,8 @@
return -EINVAL;
}
+ wm8915_update_bclk(codec);
+
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK,
src << WM8915_SYSCLK_SRC_SHIFT | ratediv);
@@ -1889,6 +1913,8 @@
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
WM8915_SYSCLK_ENA, old);
+ wm8915->sysclk_src = clk_id;
+
return 0;
}
@@ -2007,6 +2033,7 @@
unsigned int Fref, unsigned int Fout)
{
struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
+ struct i2c_client *i2c = to_i2c_client(codec->dev);
struct _fll_div fll_div;
unsigned long timeout;
int ret, reg;
@@ -2093,7 +2120,18 @@
else
timeout = msecs_to_jiffies(2);
- wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+ /* Allow substantially longer if we've actually got the IRQ */
+ if (i2c->irq)
+ timeout *= 1000;
+
+ ret = wait_for_completion_timeout(&wm8915->fll_lock, timeout);
+
+ if (ret == 0 && i2c->irq) {
+ dev_err(codec->dev, "Timed out waiting for FLL\n");
+ ret = -ETIMEDOUT;
+ } else {
+ ret = 0;
+ }
dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
@@ -2101,7 +2139,7 @@
wm8915->fll_fout = Fout;
wm8915->fll_src = source;
- return 0;
+ return ret;
}
#ifdef CONFIG_GPIOLIB
@@ -2293,6 +2331,12 @@
SND_JACK_HEADSET | SND_JACK_BTN_0);
wm8915->jack_mic = true;
wm8915->detecting = false;
+
+ /* Increase poll rate to give better responsiveness
+ * for buttons */
+ snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+ WM8915_MICD_RATE_MASK,
+ 5 << WM8915_MICD_RATE_SHIFT);
}
/* If we detected a lower impedence during initial startup
@@ -2333,15 +2377,17 @@
SND_JACK_HEADPHONE,
SND_JACK_HEADSET |
SND_JACK_BTN_0);
+
+ /* Increase the detection rate a bit for
+ * responsiveness.
+ */
+ snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
+ WM8915_MICD_RATE_MASK,
+ 7 << WM8915_MICD_RATE_SHIFT);
+
wm8915->detecting = false;
}
}
-
- /* Increase poll rate to give better responsiveness for buttons */
- if (!wm8915->detecting)
- snd_soc_update_bits(codec, WM8915_MIC_DETECT_1,
- WM8915_MICD_RATE_MASK,
- 5 << WM8915_MICD_RATE_SHIFT);
}
static irqreturn_t wm8915_irq(int irq, void *data)
@@ -2383,6 +2429,20 @@
}
}
+static irqreturn_t wm8915_edge_irq(int irq, void *data)
+{
+ irqreturn_t ret = IRQ_NONE;
+ irqreturn_t val;
+
+ do {
+ val = wm8915_irq(irq, data);
+ if (val != IRQ_NONE)
+ ret = val;
+ } while (val != IRQ_NONE);
+
+ return ret;
+}
+
static void wm8915_retune_mobile_pdata(struct snd_soc_codec *codec)
{
struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec);
@@ -2482,8 +2542,6 @@
wm8915->disable_nb[1].notifier_call = wm8915_regulator_event_1;
wm8915->disable_nb[2].notifier_call = wm8915_regulator_event_2;
wm8915->disable_nb[3].notifier_call = wm8915_regulator_event_3;
- wm8915->disable_nb[4].notifier_call = wm8915_regulator_event_4;
- wm8915->disable_nb[5].notifier_call = wm8915_regulator_event_5;
/* This should really be moved into the regulator core */
for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) {
@@ -2709,8 +2767,14 @@
irq_flags |= IRQF_ONESHOT;
- ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
- irq_flags, "wm8915", codec);
+ if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING))
+ ret = request_threaded_irq(i2c->irq, NULL,
+ wm8915_edge_irq,
+ irq_flags, "wm8915", codec);
+ else
+ ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq,
+ irq_flags, "wm8915", codec);
+
if (ret == 0) {
/* Unmask the interrupt */
snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 25580e3..056daa0 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -297,8 +297,6 @@
if (ret)
goto error_ret;
ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- if (ret)
- goto error_ret;
error_ret:
return ret;
@@ -683,8 +681,6 @@
}
}
ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- if (ret)
- goto error_ret;
error_ret:
return ret;
@@ -730,9 +726,6 @@
if (ret)
return ret;
ret = wm8940_add_widgets(codec);
- if (ret)
- return ret;
-
return ret;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 5e05eed..8499c56 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -78,6 +78,8 @@
#ifdef CONFIG_GPIOLIB
struct gpio_chip gpio_chip;
#endif
+
+ int irq;
};
/* We can't use the same notifier block for more than one supply and
@@ -1982,6 +1984,7 @@
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
/* The VU bits for the headphones are in a different register to the mute
* bits and only take effect on the PGA if it is actually powered.
@@ -2119,6 +2122,18 @@
SOC_SINGLE_TLV("Speaker Boost Volume", WM8962_CLASS_D_CONTROL_2, 0, 7, 0,
classd_tlv),
+
+SOC_SINGLE("EQ Switch", WM8962_EQ1, WM8962_EQ_ENA_SHIFT, 1, 0),
+SOC_DOUBLE_R_TLV("EQ1 Volume", WM8962_EQ2, WM8962_EQ22,
+ WM8962_EQL_B1_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ2 Volume", WM8962_EQ2, WM8962_EQ22,
+ WM8962_EQL_B2_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ3 Volume", WM8962_EQ2, WM8962_EQ22,
+ WM8962_EQL_B3_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
+ WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv),
+SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
+ WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
};
static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2184,6 +2199,8 @@
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+ unsigned long timeout;
int src;
int fll;
@@ -2203,9 +2220,19 @@
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- if (fll)
+ if (fll) {
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_ENA, WM8962_FLL_ENA);
+ if (wm8962->irq) {
+ timeout = msecs_to_jiffies(5);
+ timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+ timeout);
+
+ if (timeout == 0)
+ dev_err(codec->dev,
+ "Timed out starting FLL\n");
+ }
+ }
break;
case SND_SOC_DAPM_POST_PMD:
@@ -2763,18 +2790,44 @@
1, -1, 2, 3, 4, -1, 6, 8, -1, 12, 16, 24, -1, 32, 32, 32
};
+static const int sysclk_rates[] = {
+ 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
+};
+
static void wm8962_configure_bclk(struct snd_soc_codec *codec)
{
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
int dspclk, i;
int clocking2 = 0;
+ int clocking4 = 0;
int aif2 = 0;
- if (!wm8962->bclk) {
- dev_dbg(codec->dev, "No BCLK rate configured\n");
+ if (!wm8962->sysclk_rate) {
+ dev_dbg(codec->dev, "No SYSCLK configured\n");
return;
}
+ if (!wm8962->bclk || !wm8962->lrclk) {
+ dev_dbg(codec->dev, "No audio clocks configured\n");
+ return;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
+ if (sysclk_rates[i] == wm8962->sysclk_rate / wm8962->lrclk) {
+ clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
+ break;
+ }
+ }
+
+ if (i == ARRAY_SIZE(sysclk_rates)) {
+ dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
+ wm8962->sysclk_rate / wm8962->lrclk);
+ return;
+ }
+
+ snd_soc_update_bits(codec, WM8962_CLOCKING_4,
+ WM8962_SYSCLK_RATE_MASK, clocking4);
+
dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
if (dspclk < 0) {
dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
@@ -2844,6 +2897,8 @@
/* VMID 2*50k */
snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
WM8962_VMID_SEL_MASK, 0x80);
+
+ wm8962_configure_bclk(codec);
break;
case SND_SOC_BIAS_STANDBY:
@@ -2876,8 +2931,6 @@
snd_soc_update_bits(codec, WM8962_CLOCKING2,
WM8962_CLKREG_OVD,
WM8962_CLKREG_OVD);
-
- wm8962_configure_bclk(codec);
}
/* VMID 2*250k */
@@ -2918,10 +2971,6 @@
{ 96000, 6 },
};
-static const int sysclk_rates[] = {
- 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536,
-};
-
static int wm8962_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -2929,41 +2978,27 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
- int rate = params_rate(params);
int i;
int aif0 = 0;
int adctl3 = 0;
- int clocking4 = 0;
wm8962->bclk = snd_soc_params_to_bclk(params);
wm8962->lrclk = params_rate(params);
for (i = 0; i < ARRAY_SIZE(sr_vals); i++) {
- if (sr_vals[i].rate == rate) {
+ if (sr_vals[i].rate == wm8962->lrclk) {
adctl3 |= sr_vals[i].reg;
break;
}
}
if (i == ARRAY_SIZE(sr_vals)) {
- dev_err(codec->dev, "Unsupported rate %dHz\n", rate);
+ dev_err(codec->dev, "Unsupported rate %dHz\n", wm8962->lrclk);
return -EINVAL;
}
- if (rate % 8000 == 0)
+ if (wm8962->lrclk % 8000 == 0)
adctl3 |= WM8962_SAMPLE_RATE_INT_MODE;
- for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) {
- if (sysclk_rates[i] == wm8962->sysclk_rate / rate) {
- clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT;
- break;
- }
- }
- if (i == ARRAY_SIZE(sysclk_rates)) {
- dev_err(codec->dev, "Unsupported sysclk ratio %d\n",
- wm8962->sysclk_rate / rate);
- return -EINVAL;
- }
-
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
@@ -2985,8 +3020,6 @@
snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_3,
WM8962_SAMPLE_RATE_INT_MODE |
WM8962_SAMPLE_RATE_MASK, adctl3);
- snd_soc_update_bits(codec, WM8962_CLOCKING_4,
- WM8962_SYSCLK_RATE_MASK, clocking4);
wm8962_configure_bclk(codec);
@@ -3261,16 +3294,31 @@
dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
- /* This should be a massive overestimate */
- timeout = msecs_to_jiffies(1);
+ ret = 0;
- wait_for_completion_timeout(&wm8962->fll_lock, timeout);
+ if (fll1 & WM8962_FLL_ENA) {
+ /* This should be a massive overestimate but go even
+ * higher if we'll error out
+ */
+ if (wm8962->irq)
+ timeout = msecs_to_jiffies(5);
+ else
+ timeout = msecs_to_jiffies(1);
+
+ timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+ timeout);
+
+ if (timeout == 0 && wm8962->irq) {
+ dev_err(codec->dev, "FLL lock timed out");
+ ret = -ETIMEDOUT;
+ }
+ }
wm8962->fll_fref = Fref;
wm8962->fll_fout = Fout;
wm8962->fll_src = source;
- return 0;
+ return ret;
}
static int wm8962_mute(struct snd_soc_dai *dai, int mute)
@@ -3731,8 +3779,6 @@
int ret;
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
- struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
- dev);
u16 *reg_cache = codec->reg_cache;
int i, trigger, irq_pol;
bool dmicclk, dmicdat;
@@ -3871,6 +3917,9 @@
snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME,
WM8962_HPOUT_VU, WM8962_HPOUT_VU);
+ /* Stereo control for EQ */
+ snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0);
+
wm8962_add_widgets(codec);
/* Save boards having to disable DMIC when not in use */
@@ -3899,7 +3948,7 @@
wm8962_init_beep(codec);
wm8962_init_gpio(codec);
- if (i2c->irq) {
+ if (wm8962->irq) {
if (pdata && pdata->irq_active_low) {
trigger = IRQF_TRIGGER_LOW;
irq_pol = WM8962_IRQ_POL;
@@ -3911,12 +3960,13 @@
snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL,
WM8962_IRQ_POL, irq_pol);
- ret = request_threaded_irq(i2c->irq, NULL, wm8962_irq,
+ ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq,
trigger | IRQF_ONESHOT,
"wm8962", codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to request IRQ %d: %d\n",
- i2c->irq, ret);
+ wm8962->irq, ret);
+ wm8962->irq = 0;
/* Non-fatal */
} else {
/* Enable some IRQs by default */
@@ -3941,12 +3991,10 @@
static int wm8962_remove(struct snd_soc_codec *codec)
{
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
- struct i2c_client *i2c = container_of(codec->dev, struct i2c_client,
- dev);
int i;
- if (i2c->irq)
- free_irq(i2c->irq, codec);
+ if (wm8962->irq)
+ free_irq(wm8962->irq, codec);
cancel_delayed_work_sync(&wm8962->mic_work);
@@ -3986,6 +4034,8 @@
i2c_set_clientdata(i2c, wm8962);
+ wm8962->irq = i2c->irq;
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8962, &wm8962_dai, 1);
if (ret < 0)
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
new file mode 100644
index 0000000..17f04ec
--- /dev/null
+++ b/sound/soc/codecs/wm8983.c
@@ -0,0 +1,1203 @@
+/*
+ * wm8983.c -- WM8983 ALSA SoC Audio driver
+ *
+ * Copyright 2011 Wolfson Microelectronics plc
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8983.h"
+
+static const u16 wm8983_reg_defs[WM8983_MAX_REGISTER + 1] = {
+ [0x00] = 0x0000, /* R0 - Software Reset */
+ [0x01] = 0x0000, /* R1 - Power management 1 */
+ [0x02] = 0x0000, /* R2 - Power management 2 */
+ [0x03] = 0x0000, /* R3 - Power management 3 */
+ [0x04] = 0x0050, /* R4 - Audio Interface */
+ [0x05] = 0x0000, /* R5 - Companding control */
+ [0x06] = 0x0140, /* R6 - Clock Gen control */
+ [0x07] = 0x0000, /* R7 - Additional control */
+ [0x08] = 0x0000, /* R8 - GPIO Control */
+ [0x09] = 0x0000, /* R9 - Jack Detect Control 1 */
+ [0x0A] = 0x0000, /* R10 - DAC Control */
+ [0x0B] = 0x00FF, /* R11 - Left DAC digital Vol */
+ [0x0C] = 0x00FF, /* R12 - Right DAC digital vol */
+ [0x0D] = 0x0000, /* R13 - Jack Detect Control 2 */
+ [0x0E] = 0x0100, /* R14 - ADC Control */
+ [0x0F] = 0x00FF, /* R15 - Left ADC Digital Vol */
+ [0x10] = 0x00FF, /* R16 - Right ADC Digital Vol */
+ [0x12] = 0x012C, /* R18 - EQ1 - low shelf */
+ [0x13] = 0x002C, /* R19 - EQ2 - peak 1 */
+ [0x14] = 0x002C, /* R20 - EQ3 - peak 2 */
+ [0x15] = 0x002C, /* R21 - EQ4 - peak 3 */
+ [0x16] = 0x002C, /* R22 - EQ5 - high shelf */
+ [0x18] = 0x0032, /* R24 - DAC Limiter 1 */
+ [0x19] = 0x0000, /* R25 - DAC Limiter 2 */
+ [0x1B] = 0x0000, /* R27 - Notch Filter 1 */
+ [0x1C] = 0x0000, /* R28 - Notch Filter 2 */
+ [0x1D] = 0x0000, /* R29 - Notch Filter 3 */
+ [0x1E] = 0x0000, /* R30 - Notch Filter 4 */
+ [0x20] = 0x0038, /* R32 - ALC control 1 */
+ [0x21] = 0x000B, /* R33 - ALC control 2 */
+ [0x22] = 0x0032, /* R34 - ALC control 3 */
+ [0x23] = 0x0000, /* R35 - Noise Gate */
+ [0x24] = 0x0008, /* R36 - PLL N */
+ [0x25] = 0x000C, /* R37 - PLL K 1 */
+ [0x26] = 0x0093, /* R38 - PLL K 2 */
+ [0x27] = 0x00E9, /* R39 - PLL K 3 */
+ [0x29] = 0x0000, /* R41 - 3D control */
+ [0x2A] = 0x0000, /* R42 - OUT4 to ADC */
+ [0x2B] = 0x0000, /* R43 - Beep control */
+ [0x2C] = 0x0033, /* R44 - Input ctrl */
+ [0x2D] = 0x0010, /* R45 - Left INP PGA gain ctrl */
+ [0x2E] = 0x0010, /* R46 - Right INP PGA gain ctrl */
+ [0x2F] = 0x0100, /* R47 - Left ADC BOOST ctrl */
+ [0x30] = 0x0100, /* R48 - Right ADC BOOST ctrl */
+ [0x31] = 0x0002, /* R49 - Output ctrl */
+ [0x32] = 0x0001, /* R50 - Left mixer ctrl */
+ [0x33] = 0x0001, /* R51 - Right mixer ctrl */
+ [0x34] = 0x0039, /* R52 - LOUT1 (HP) volume ctrl */
+ [0x35] = 0x0039, /* R53 - ROUT1 (HP) volume ctrl */
+ [0x36] = 0x0039, /* R54 - LOUT2 (SPK) volume ctrl */
+ [0x37] = 0x0039, /* R55 - ROUT2 (SPK) volume ctrl */
+ [0x38] = 0x0001, /* R56 - OUT3 mixer ctrl */
+ [0x39] = 0x0001, /* R57 - OUT4 (MONO) mix ctrl */
+ [0x3D] = 0x0000 /* R61 - BIAS CTRL */
+};
+
+static const struct wm8983_reg_access {
+ u16 read; /* Mask of readable bits */
+ u16 write; /* Mask of writable bits */
+} wm8983_access_masks[WM8983_MAX_REGISTER + 1] = {
+ [0x00] = { 0x0000, 0x01FF }, /* R0 - Software Reset */
+ [0x01] = { 0x0000, 0x01FF }, /* R1 - Power management 1 */
+ [0x02] = { 0x0000, 0x01FF }, /* R2 - Power management 2 */
+ [0x03] = { 0x0000, 0x01EF }, /* R3 - Power management 3 */
+ [0x04] = { 0x0000, 0x01FF }, /* R4 - Audio Interface */
+ [0x05] = { 0x0000, 0x003F }, /* R5 - Companding control */
+ [0x06] = { 0x0000, 0x01FD }, /* R6 - Clock Gen control */
+ [0x07] = { 0x0000, 0x000F }, /* R7 - Additional control */
+ [0x08] = { 0x0000, 0x003F }, /* R8 - GPIO Control */
+ [0x09] = { 0x0000, 0x0070 }, /* R9 - Jack Detect Control 1 */
+ [0x0A] = { 0x0000, 0x004F }, /* R10 - DAC Control */
+ [0x0B] = { 0x0000, 0x01FF }, /* R11 - Left DAC digital Vol */
+ [0x0C] = { 0x0000, 0x01FF }, /* R12 - Right DAC digital vol */
+ [0x0D] = { 0x0000, 0x00FF }, /* R13 - Jack Detect Control 2 */
+ [0x0E] = { 0x0000, 0x01FB }, /* R14 - ADC Control */
+ [0x0F] = { 0x0000, 0x01FF }, /* R15 - Left ADC Digital Vol */
+ [0x10] = { 0x0000, 0x01FF }, /* R16 - Right ADC Digital Vol */
+ [0x12] = { 0x0000, 0x017F }, /* R18 - EQ1 - low shelf */
+ [0x13] = { 0x0000, 0x017F }, /* R19 - EQ2 - peak 1 */
+ [0x14] = { 0x0000, 0x017F }, /* R20 - EQ3 - peak 2 */
+ [0x15] = { 0x0000, 0x017F }, /* R21 - EQ4 - peak 3 */
+ [0x16] = { 0x0000, 0x007F }, /* R22 - EQ5 - high shelf */
+ [0x18] = { 0x0000, 0x01FF }, /* R24 - DAC Limiter 1 */
+ [0x19] = { 0x0000, 0x007F }, /* R25 - DAC Limiter 2 */
+ [0x1B] = { 0x0000, 0x01FF }, /* R27 - Notch Filter 1 */
+ [0x1C] = { 0x0000, 0x017F }, /* R28 - Notch Filter 2 */
+ [0x1D] = { 0x0000, 0x017F }, /* R29 - Notch Filter 3 */
+ [0x1E] = { 0x0000, 0x017F }, /* R30 - Notch Filter 4 */
+ [0x20] = { 0x0000, 0x01BF }, /* R32 - ALC control 1 */
+ [0x21] = { 0x0000, 0x00FF }, /* R33 - ALC control 2 */
+ [0x22] = { 0x0000, 0x01FF }, /* R34 - ALC control 3 */
+ [0x23] = { 0x0000, 0x000F }, /* R35 - Noise Gate */
+ [0x24] = { 0x0000, 0x001F }, /* R36 - PLL N */
+ [0x25] = { 0x0000, 0x003F }, /* R37 - PLL K 1 */
+ [0x26] = { 0x0000, 0x01FF }, /* R38 - PLL K 2 */
+ [0x27] = { 0x0000, 0x01FF }, /* R39 - PLL K 3 */
+ [0x29] = { 0x0000, 0x000F }, /* R41 - 3D control */
+ [0x2A] = { 0x0000, 0x01E7 }, /* R42 - OUT4 to ADC */
+ [0x2B] = { 0x0000, 0x01BF }, /* R43 - Beep control */
+ [0x2C] = { 0x0000, 0x0177 }, /* R44 - Input ctrl */
+ [0x2D] = { 0x0000, 0x01FF }, /* R45 - Left INP PGA gain ctrl */
+ [0x2E] = { 0x0000, 0x01FF }, /* R46 - Right INP PGA gain ctrl */
+ [0x2F] = { 0x0000, 0x0177 }, /* R47 - Left ADC BOOST ctrl */
+ [0x30] = { 0x0000, 0x0177 }, /* R48 - Right ADC BOOST ctrl */
+ [0x31] = { 0x0000, 0x007F }, /* R49 - Output ctrl */
+ [0x32] = { 0x0000, 0x01FF }, /* R50 - Left mixer ctrl */
+ [0x33] = { 0x0000, 0x01FF }, /* R51 - Right mixer ctrl */
+ [0x34] = { 0x0000, 0x01FF }, /* R52 - LOUT1 (HP) volume ctrl */
+ [0x35] = { 0x0000, 0x01FF }, /* R53 - ROUT1 (HP) volume ctrl */
+ [0x36] = { 0x0000, 0x01FF }, /* R54 - LOUT2 (SPK) volume ctrl */
+ [0x37] = { 0x0000, 0x01FF }, /* R55 - ROUT2 (SPK) volume ctrl */
+ [0x38] = { 0x0000, 0x004F }, /* R56 - OUT3 mixer ctrl */
+ [0x39] = { 0x0000, 0x00FF }, /* R57 - OUT4 (MONO) mix ctrl */
+ [0x3D] = { 0x0000, 0x0100 } /* R61 - BIAS CTRL */
+};
+
+/* vol/gain update regs */
+static const int vol_update_regs[] = {
+ WM8983_LEFT_DAC_DIGITAL_VOL,
+ WM8983_RIGHT_DAC_DIGITAL_VOL,
+ WM8983_LEFT_ADC_DIGITAL_VOL,
+ WM8983_RIGHT_ADC_DIGITAL_VOL,
+ WM8983_LOUT1_HP_VOLUME_CTRL,
+ WM8983_ROUT1_HP_VOLUME_CTRL,
+ WM8983_LOUT2_SPK_VOLUME_CTRL,
+ WM8983_ROUT2_SPK_VOLUME_CTRL,
+ WM8983_LEFT_INP_PGA_GAIN_CTRL,
+ WM8983_RIGHT_INP_PGA_GAIN_CTRL
+};
+
+struct wm8983_priv {
+ enum snd_soc_control_type control_type;
+ u32 sysclk;
+ u32 bclk;
+};
+
+static const struct {
+ int div;
+ int ratio;
+} fs_ratios[] = {
+ { 10, 128 },
+ { 15, 192 },
+ { 20, 256 },
+ { 30, 384 },
+ { 40, 512 },
+ { 60, 768 },
+ { 80, 1024 },
+ { 120, 1536 }
+};
+
+static const int srates[] = { 48000, 32000, 24000, 16000, 12000, 8000 };
+
+static const int bclk_divs[] = {
+ 1, 2, 4, 8, 16, 32
+};
+
+static int eqmode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+static int eqmode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(lim_thresh_tlv, -600, 100, 0);
+static const DECLARE_TLV_DB_SCALE(lim_boost_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(alc_min_tlv, -1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(alc_max_tlv, -675, 600, 0);
+static const DECLARE_TLV_DB_SCALE(alc_tar_tlv, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(pga_vol_tlv, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(aux_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
+
+static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" };
+static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7,
+ alc_sel_text);
+
+static const char *alc_mode_text[] = { "ALC", "Limiter" };
+static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8,
+ alc_mode_text);
+
+static const char *filter_mode_text[] = { "Audio", "Application" };
+static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7,
+ filter_mode_text);
+
+static const char *eq_bw_text[] = { "Narrow", "Wide" };
+static const char *eqmode_text[] = { "Capture", "Playback" };
+static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
+
+static const char *eq1_cutoff_text[] = {
+ "80Hz", "105Hz", "135Hz", "175Hz"
+};
+static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5,
+ eq1_cutoff_text);
+static const char *eq2_cutoff_text[] = {
+ "230Hz", "300Hz", "385Hz", "500Hz"
+};
+static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text);
+static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5,
+ eq2_cutoff_text);
+static const char *eq3_cutoff_text[] = {
+ "650Hz", "850Hz", "1.1kHz", "1.4kHz"
+};
+static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text);
+static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5,
+ eq3_cutoff_text);
+static const char *eq4_cutoff_text[] = {
+ "1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"
+};
+static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text);
+static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5,
+ eq4_cutoff_text);
+static const char *eq5_cutoff_text[] = {
+ "5.3kHz", "6.9kHz", "9kHz", "11.7kHz"
+};
+static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5,
+ eq5_cutoff_text);
+
+static const char *speaker_mode_text[] = { "Class A/B", "Class D" };
+static const SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text);
+
+static const char *depth_3d_text[] = {
+ "Off",
+ "6.67%",
+ "13.3%",
+ "20%",
+ "26.7%",
+ "33.3%",
+ "40%",
+ "46.6%",
+ "53.3%",
+ "60%",
+ "66.7%",
+ "73.3%",
+ "80%",
+ "86.7%",
+ "93.3%",
+ "100%"
+};
+static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0,
+ depth_3d_text);
+
+static const struct snd_kcontrol_new wm8983_snd_controls[] = {
+ SOC_SINGLE("Digital Loopback Switch", WM8983_COMPANDING_CONTROL,
+ 0, 1, 0),
+
+ SOC_ENUM("ALC Capture Function", alc_sel),
+ SOC_SINGLE_TLV("ALC Capture Max Volume", WM8983_ALC_CONTROL_1,
+ 3, 7, 0, alc_max_tlv),
+ SOC_SINGLE_TLV("ALC Capture Min Volume", WM8983_ALC_CONTROL_1,
+ 0, 7, 0, alc_min_tlv),
+ SOC_SINGLE_TLV("ALC Capture Target Volume", WM8983_ALC_CONTROL_2,
+ 0, 15, 0, alc_tar_tlv),
+ SOC_SINGLE("ALC Capture Attack", WM8983_ALC_CONTROL_3, 0, 10, 0),
+ SOC_SINGLE("ALC Capture Hold", WM8983_ALC_CONTROL_2, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Decay", WM8983_ALC_CONTROL_3, 4, 10, 0),
+ SOC_ENUM("ALC Mode", alc_mode),
+ SOC_SINGLE("ALC Capture NG Switch", WM8983_NOISE_GATE,
+ 3, 1, 0),
+ SOC_SINGLE("ALC Capture NG Threshold", WM8983_NOISE_GATE,
+ 0, 7, 1),
+
+ SOC_DOUBLE_R_TLV("Capture Volume", WM8983_LEFT_ADC_DIGITAL_VOL,
+ WM8983_RIGHT_ADC_DIGITAL_VOL, 0, 255, 0, adc_tlv),
+ SOC_DOUBLE_R("Capture PGA ZC Switch", WM8983_LEFT_INP_PGA_GAIN_CTRL,
+ WM8983_RIGHT_INP_PGA_GAIN_CTRL, 7, 1, 0),
+ SOC_DOUBLE_R_TLV("Capture PGA Volume", WM8983_LEFT_INP_PGA_GAIN_CTRL,
+ WM8983_RIGHT_INP_PGA_GAIN_CTRL, 0, 63, 0, pga_vol_tlv),
+
+ SOC_DOUBLE_R_TLV("Capture PGA Boost Volume",
+ WM8983_LEFT_ADC_BOOST_CTRL, WM8983_RIGHT_ADC_BOOST_CTRL,
+ 8, 1, 0, pga_boost_tlv),
+
+ SOC_DOUBLE("ADC Inversion Switch", WM8983_ADC_CONTROL, 0, 1, 1, 0),
+ SOC_SINGLE("ADC 128x Oversampling Switch", WM8983_ADC_CONTROL, 8, 1, 0),
+
+ SOC_DOUBLE_R_TLV("Playback Volume", WM8983_LEFT_DAC_DIGITAL_VOL,
+ WM8983_RIGHT_DAC_DIGITAL_VOL, 0, 255, 0, dac_tlv),
+
+ SOC_SINGLE("DAC Playback Limiter Switch", WM8983_DAC_LIMITER_1, 8, 1, 0),
+ SOC_SINGLE("DAC Playback Limiter Decay", WM8983_DAC_LIMITER_1, 4, 10, 0),
+ SOC_SINGLE("DAC Playback Limiter Attack", WM8983_DAC_LIMITER_1, 0, 11, 0),
+ SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8983_DAC_LIMITER_2,
+ 4, 7, 1, lim_thresh_tlv),
+ SOC_SINGLE_TLV("DAC Playback Limiter Boost Volume", WM8983_DAC_LIMITER_2,
+ 0, 12, 0, lim_boost_tlv),
+ SOC_DOUBLE("DAC Inversion Switch", WM8983_DAC_CONTROL, 0, 1, 1, 0),
+ SOC_SINGLE("DAC Auto Mute Switch", WM8983_DAC_CONTROL, 2, 1, 0),
+ SOC_SINGLE("DAC 128x Oversampling Switch", WM8983_DAC_CONTROL, 3, 1, 0),
+
+ SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8983_LOUT1_HP_VOLUME_CTRL,
+ WM8983_ROUT1_HP_VOLUME_CTRL, 0, 63, 0, out_tlv),
+ SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8983_LOUT1_HP_VOLUME_CTRL,
+ WM8983_ROUT1_HP_VOLUME_CTRL, 7, 1, 0),
+ SOC_DOUBLE_R("Headphone Switch", WM8983_LOUT1_HP_VOLUME_CTRL,
+ WM8983_ROUT1_HP_VOLUME_CTRL, 6, 1, 1),
+
+ SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8983_LOUT2_SPK_VOLUME_CTRL,
+ WM8983_ROUT2_SPK_VOLUME_CTRL, 0, 63, 0, out_tlv),
+ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8983_LOUT2_SPK_VOLUME_CTRL,
+ WM8983_ROUT2_SPK_VOLUME_CTRL, 7, 1, 0),
+ SOC_DOUBLE_R("Speaker Switch", WM8983_LOUT2_SPK_VOLUME_CTRL,
+ WM8983_ROUT2_SPK_VOLUME_CTRL, 6, 1, 1),
+
+ SOC_SINGLE("OUT3 Switch", WM8983_OUT3_MIXER_CTRL,
+ 6, 1, 1),
+
+ SOC_SINGLE("OUT4 Switch", WM8983_OUT4_MONO_MIX_CTRL,
+ 6, 1, 1),
+
+ SOC_SINGLE("High Pass Filter Switch", WM8983_ADC_CONTROL, 8, 1, 0),
+ SOC_ENUM("High Pass Filter Mode", filter_mode),
+ SOC_SINGLE("High Pass Filter Cutoff", WM8983_ADC_CONTROL, 4, 7, 0),
+
+ SOC_DOUBLE_R_TLV("Aux Bypass Volume",
+ WM8983_LEFT_MIXER_CTRL, WM8983_RIGHT_MIXER_CTRL, 6, 7, 0,
+ aux_tlv),
+
+ SOC_DOUBLE_R_TLV("Input PGA Bypass Volume",
+ WM8983_LEFT_MIXER_CTRL, WM8983_RIGHT_MIXER_CTRL, 2, 7, 0,
+ bypass_tlv),
+
+ SOC_ENUM_EXT("Equalizer Function", eqmode, eqmode_get, eqmode_put),
+ SOC_ENUM("EQ1 Cutoff", eq1_cutoff),
+ SOC_SINGLE_TLV("EQ1 Volume", WM8983_EQ1_LOW_SHELF, 0, 24, 1, eq_tlv),
+ SOC_ENUM("EQ2 Bandwith", eq2_bw),
+ SOC_ENUM("EQ2 Cutoff", eq2_cutoff),
+ SOC_SINGLE_TLV("EQ2 Volume", WM8983_EQ2_PEAK_1, 0, 24, 1, eq_tlv),
+ SOC_ENUM("EQ3 Bandwith", eq3_bw),
+ SOC_ENUM("EQ3 Cutoff", eq3_cutoff),
+ SOC_SINGLE_TLV("EQ3 Volume", WM8983_EQ3_PEAK_2, 0, 24, 1, eq_tlv),
+ SOC_ENUM("EQ4 Bandwith", eq4_bw),
+ SOC_ENUM("EQ4 Cutoff", eq4_cutoff),
+ SOC_SINGLE_TLV("EQ4 Volume", WM8983_EQ4_PEAK_3, 0, 24, 1, eq_tlv),
+ SOC_ENUM("EQ5 Cutoff", eq5_cutoff),
+ SOC_SINGLE_TLV("EQ5 Volume", WM8983_EQ5_HIGH_SHELF, 0, 24, 1, eq_tlv),
+
+ SOC_ENUM("3D Depth", depth_3d),
+
+ SOC_ENUM("Speaker Mode", speaker_mode)
+};
+
+static const struct snd_kcontrol_new left_out_mixer[] = {
+ SOC_DAPM_SINGLE("Line Switch", WM8983_LEFT_MIXER_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("Aux Switch", WM8983_LEFT_MIXER_CTRL, 5, 1, 0),
+ SOC_DAPM_SINGLE("PCM Switch", WM8983_LEFT_MIXER_CTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_out_mixer[] = {
+ SOC_DAPM_SINGLE("Line Switch", WM8983_RIGHT_MIXER_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("Aux Switch", WM8983_RIGHT_MIXER_CTRL, 5, 1, 0),
+ SOC_DAPM_SINGLE("PCM Switch", WM8983_RIGHT_MIXER_CTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_input_mixer[] = {
+ SOC_DAPM_SINGLE("L2 Switch", WM8983_INPUT_CTRL, 2, 1, 0),
+ SOC_DAPM_SINGLE("MicN Switch", WM8983_INPUT_CTRL, 1, 1, 0),
+ SOC_DAPM_SINGLE("MicP Switch", WM8983_INPUT_CTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_input_mixer[] = {
+ SOC_DAPM_SINGLE("R2 Switch", WM8983_INPUT_CTRL, 6, 1, 0),
+ SOC_DAPM_SINGLE("MicN Switch", WM8983_INPUT_CTRL, 5, 1, 0),
+ SOC_DAPM_SINGLE("MicP Switch", WM8983_INPUT_CTRL, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_boost_mixer[] = {
+ SOC_DAPM_SINGLE_TLV("L2 Volume", WM8983_LEFT_ADC_BOOST_CTRL,
+ 4, 7, 0, boost_tlv),
+ SOC_DAPM_SINGLE_TLV("AUXL Volume", WM8983_LEFT_ADC_BOOST_CTRL,
+ 0, 7, 0, boost_tlv)
+};
+
+static const struct snd_kcontrol_new out3_mixer[] = {
+ SOC_DAPM_SINGLE("LMIX2OUT3 Switch", WM8983_OUT3_MIXER_CTRL,
+ 1, 1, 0),
+ SOC_DAPM_SINGLE("LDAC2OUT3 Switch", WM8983_OUT3_MIXER_CTRL,
+ 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new out4_mixer[] = {
+ SOC_DAPM_SINGLE("LMIX2OUT4 Switch", WM8983_OUT4_MONO_MIX_CTRL,
+ 4, 1, 0),
+ SOC_DAPM_SINGLE("RMIX2OUT4 Switch", WM8983_OUT4_MONO_MIX_CTRL,
+ 1, 1, 0),
+ SOC_DAPM_SINGLE("LDAC2OUT4 Switch", WM8983_OUT4_MONO_MIX_CTRL,
+ 3, 1, 0),
+ SOC_DAPM_SINGLE("RDAC2OUT4 Switch", WM8983_OUT4_MONO_MIX_CTRL,
+ 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_boost_mixer[] = {
+ SOC_DAPM_SINGLE_TLV("R2 Volume", WM8983_RIGHT_ADC_BOOST_CTRL,
+ 4, 7, 0, boost_tlv),
+ SOC_DAPM_SINGLE_TLV("AUXR Volume", WM8983_RIGHT_ADC_BOOST_CTRL,
+ 0, 7, 0, boost_tlv)
+};
+
+static const struct snd_soc_dapm_widget wm8983_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8983_POWER_MANAGEMENT_3,
+ 0, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8983_POWER_MANAGEMENT_3,
+ 1, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8983_POWER_MANAGEMENT_2,
+ 0, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8983_POWER_MANAGEMENT_2,
+ 1, 0),
+
+ SND_SOC_DAPM_MIXER("Left Output Mixer", WM8983_POWER_MANAGEMENT_3,
+ 2, 0, left_out_mixer, ARRAY_SIZE(left_out_mixer)),
+ SND_SOC_DAPM_MIXER("Right Output Mixer", WM8983_POWER_MANAGEMENT_3,
+ 3, 0, right_out_mixer, ARRAY_SIZE(right_out_mixer)),
+
+ SND_SOC_DAPM_MIXER("Left Input Mixer", WM8983_POWER_MANAGEMENT_2,
+ 2, 0, left_input_mixer, ARRAY_SIZE(left_input_mixer)),
+ SND_SOC_DAPM_MIXER("Right Input Mixer", WM8983_POWER_MANAGEMENT_2,
+ 3, 0, right_input_mixer, ARRAY_SIZE(right_input_mixer)),
+
+ SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8983_POWER_MANAGEMENT_2,
+ 4, 0, left_boost_mixer, ARRAY_SIZE(left_boost_mixer)),
+ SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8983_POWER_MANAGEMENT_2,
+ 5, 0, right_boost_mixer, ARRAY_SIZE(right_boost_mixer)),
+
+ SND_SOC_DAPM_MIXER("OUT3 Mixer", WM8983_POWER_MANAGEMENT_1,
+ 6, 0, out3_mixer, ARRAY_SIZE(out3_mixer)),
+
+ SND_SOC_DAPM_MIXER("OUT4 Mixer", WM8983_POWER_MANAGEMENT_1,
+ 7, 0, out4_mixer, ARRAY_SIZE(out4_mixer)),
+
+ SND_SOC_DAPM_PGA("Left Capture PGA", WM8983_LEFT_INP_PGA_GAIN_CTRL,
+ 6, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Capture PGA", WM8983_RIGHT_INP_PGA_GAIN_CTRL,
+ 6, 1, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Left Headphone Out", WM8983_POWER_MANAGEMENT_2,
+ 7, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Headphone Out", WM8983_POWER_MANAGEMENT_2,
+ 8, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("Left Speaker Out", WM8983_POWER_MANAGEMENT_3,
+ 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Speaker Out", WM8983_POWER_MANAGEMENT_3,
+ 6, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("OUT3 Out", WM8983_POWER_MANAGEMENT_3,
+ 7, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("OUT4 Out", WM8983_POWER_MANAGEMENT_3,
+ 8, 0, NULL, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8983_POWER_MANAGEMENT_1, 4, 0),
+
+ SND_SOC_DAPM_INPUT("LIN"),
+ SND_SOC_DAPM_INPUT("LIP"),
+ SND_SOC_DAPM_INPUT("RIN"),
+ SND_SOC_DAPM_INPUT("RIP"),
+ SND_SOC_DAPM_INPUT("AUXL"),
+ SND_SOC_DAPM_INPUT("AUXR"),
+ SND_SOC_DAPM_INPUT("L2"),
+ SND_SOC_DAPM_INPUT("R2"),
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+ SND_SOC_DAPM_OUTPUT("SPKL"),
+ SND_SOC_DAPM_OUTPUT("SPKR"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_OUTPUT("OUT4")
+};
+
+static const struct snd_soc_dapm_route wm8983_audio_map[] = {
+ { "OUT3 Mixer", "LMIX2OUT3 Switch", "Left Output Mixer" },
+ { "OUT3 Mixer", "LDAC2OUT3 Switch", "Left DAC" },
+
+ { "OUT3 Out", NULL, "OUT3 Mixer" },
+ { "OUT3", NULL, "OUT3 Out" },
+
+ { "OUT4 Mixer", "LMIX2OUT4 Switch", "Left Output Mixer" },
+ { "OUT4 Mixer", "RMIX2OUT4 Switch", "Right Output Mixer" },
+ { "OUT4 Mixer", "LDAC2OUT4 Switch", "Left DAC" },
+ { "OUT4 Mixer", "RDAC2OUT4 Switch", "Right DAC" },
+
+ { "OUT4 Out", NULL, "OUT4 Mixer" },
+ { "OUT4", NULL, "OUT4 Out" },
+
+ { "Right Output Mixer", "PCM Switch", "Right DAC" },
+ { "Right Output Mixer", "Aux Switch", "AUXR" },
+ { "Right Output Mixer", "Line Switch", "Right Boost Mixer" },
+
+ { "Left Output Mixer", "PCM Switch", "Left DAC" },
+ { "Left Output Mixer", "Aux Switch", "AUXL" },
+ { "Left Output Mixer", "Line Switch", "Left Boost Mixer" },
+
+ { "Right Headphone Out", NULL, "Right Output Mixer" },
+ { "HPR", NULL, "Right Headphone Out" },
+
+ { "Left Headphone Out", NULL, "Left Output Mixer" },
+ { "HPL", NULL, "Left Headphone Out" },
+
+ { "Right Speaker Out", NULL, "Right Output Mixer" },
+ { "SPKR", NULL, "Right Speaker Out" },
+
+ { "Left Speaker Out", NULL, "Left Output Mixer" },
+ { "SPKL", NULL, "Left Speaker Out" },
+
+ { "Right ADC", NULL, "Right Boost Mixer" },
+
+ { "Right Boost Mixer", "AUXR Volume", "AUXR" },
+ { "Right Boost Mixer", NULL, "Right Capture PGA" },
+ { "Right Boost Mixer", "R2 Volume", "R2" },
+
+ { "Left ADC", NULL, "Left Boost Mixer" },
+
+ { "Left Boost Mixer", "AUXL Volume", "AUXL" },
+ { "Left Boost Mixer", NULL, "Left Capture PGA" },
+ { "Left Boost Mixer", "L2 Volume", "L2" },
+
+ { "Right Capture PGA", NULL, "Right Input Mixer" },
+ { "Left Capture PGA", NULL, "Left Input Mixer" },
+
+ { "Right Input Mixer", "R2 Switch", "R2" },
+ { "Right Input Mixer", "MicN Switch", "RIN" },
+ { "Right Input Mixer", "MicP Switch", "RIP" },
+
+ { "Left Input Mixer", "L2 Switch", "L2" },
+ { "Left Input Mixer", "MicN Switch", "LIN" },
+ { "Left Input Mixer", "MicP Switch", "LIP" },
+};
+
+static int eqmode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg;
+
+ reg = snd_soc_read(codec, WM8983_EQ1_LOW_SHELF);
+ if (reg & WM8983_EQ3DMODE)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+static int eqmode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int regpwr2, regpwr3;
+ unsigned int reg_eq;
+
+ if (ucontrol->value.integer.value[0] != 0
+ && ucontrol->value.integer.value[0] != 1)
+ return -EINVAL;
+
+ reg_eq = snd_soc_read(codec, WM8983_EQ1_LOW_SHELF);
+ switch ((reg_eq & WM8983_EQ3DMODE) >> WM8983_EQ3DMODE_SHIFT) {
+ case 0:
+ if (!ucontrol->value.integer.value[0])
+ return 0;
+ break;
+ case 1:
+ if (ucontrol->value.integer.value[0])
+ return 0;
+ break;
+ }
+
+ regpwr2 = snd_soc_read(codec, WM8983_POWER_MANAGEMENT_2);
+ regpwr3 = snd_soc_read(codec, WM8983_POWER_MANAGEMENT_3);
+ /* disable the DACs and ADCs */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_2,
+ WM8983_ADCENR_MASK | WM8983_ADCENL_MASK, 0);
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_3,
+ WM8983_DACENR_MASK | WM8983_DACENL_MASK, 0);
+ /* set the desired eqmode */
+ snd_soc_update_bits(codec, WM8983_EQ1_LOW_SHELF,
+ WM8983_EQ3DMODE_MASK,
+ ucontrol->value.integer.value[0]
+ << WM8983_EQ3DMODE_SHIFT);
+ /* restore DAC/ADC configuration */
+ snd_soc_write(codec, WM8983_POWER_MANAGEMENT_2, regpwr2);
+ snd_soc_write(codec, WM8983_POWER_MANAGEMENT_3, regpwr3);
+ return 0;
+}
+
+static int wm8983_readable(struct snd_soc_codec *codec, unsigned int reg)
+{
+ if (reg > WM8983_MAX_REGISTER)
+ return 0;
+
+ return wm8983_access_masks[reg].read != 0;
+}
+
+static int wm8983_dac_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ return snd_soc_update_bits(codec, WM8983_DAC_CONTROL,
+ WM8983_SOFTMUTE_MASK,
+ !!mute << WM8983_SOFTMUTE_SHIFT);
+}
+
+static int wm8983_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 format, master, bcp, lrp;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ format = 0x2;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ format = 0x0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ format = 0x1;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ format = 0x3;
+ break;
+ default:
+ dev_err(dai->dev, "Unknown dai format\n");
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, WM8983_AUDIO_INTERFACE,
+ WM8983_FMT_MASK, format << WM8983_FMT_SHIFT);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ master = 0;
+ break;
+ default:
+ dev_err(dai->dev, "Unknown master/slave configuration\n");
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, WM8983_CLOCK_GEN_CONTROL,
+ WM8983_MS_MASK, master << WM8983_MS_SHIFT);
+
+ /* FIXME: We don't currently support DSP A/B modes */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ dev_err(dai->dev, "DSP A/B modes are not supported\n");
+ return -EINVAL;
+ default:
+ break;
+ }
+
+ bcp = lrp = 0;
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ bcp = lrp = 1;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ bcp = 1;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ lrp = 1;
+ break;
+ default:
+ dev_err(dai->dev, "Unknown polarity configuration\n");
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, WM8983_AUDIO_INTERFACE,
+ WM8983_LRCP_MASK, lrp << WM8983_LRCP_SHIFT);
+ snd_soc_update_bits(codec, WM8983_AUDIO_INTERFACE,
+ WM8983_BCP_MASK, bcp << WM8983_BCP_SHIFT);
+ return 0;
+}
+
+static int wm8983_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int i;
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8983_priv *wm8983 = snd_soc_codec_get_drvdata(codec);
+ u16 blen, srate_idx;
+ u32 tmp;
+ int srate_best;
+ int ret;
+
+ ret = snd_soc_params_to_bclk(params);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to convert params to bclk: %d\n", ret);
+ return ret;
+ }
+
+ wm8983->bclk = ret;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ blen = 0x0;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ blen = 0x1;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ blen = 0x2;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ blen = 0x3;
+ break;
+ default:
+ dev_err(dai->dev, "Unsupported word length %u\n",
+ params_format(params));
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, WM8983_AUDIO_INTERFACE,
+ WM8983_WL_MASK, blen << WM8983_WL_SHIFT);
+
+ /*
+ * match to the nearest possible sample rate and rely
+ * on the array index to configure the SR register
+ */
+ srate_idx = 0;
+ srate_best = abs(srates[0] - params_rate(params));
+ for (i = 1; i < ARRAY_SIZE(srates); ++i) {
+ if (abs(srates[i] - params_rate(params)) >= srate_best)
+ continue;
+ srate_idx = i;
+ srate_best = abs(srates[i] - params_rate(params));
+ }
+
+ dev_dbg(dai->dev, "Selected SRATE = %d\n", srates[srate_idx]);
+ snd_soc_update_bits(codec, WM8983_ADDITIONAL_CONTROL,
+ WM8983_SR_MASK, srate_idx << WM8983_SR_SHIFT);
+
+ dev_dbg(dai->dev, "Target BCLK = %uHz\n", wm8983->bclk);
+ dev_dbg(dai->dev, "SYSCLK = %uHz\n", wm8983->sysclk);
+
+ for (i = 0; i < ARRAY_SIZE(fs_ratios); ++i) {
+ if (wm8983->sysclk / params_rate(params)
+ == fs_ratios[i].ratio)
+ break;
+ }
+
+ if (i == ARRAY_SIZE(fs_ratios)) {
+ dev_err(dai->dev, "Unable to configure MCLK ratio %u/%u\n",
+ wm8983->sysclk, params_rate(params));
+ return -EINVAL;
+ }
+
+ dev_dbg(dai->dev, "MCLK ratio = %dfs\n", fs_ratios[i].ratio);
+ snd_soc_update_bits(codec, WM8983_CLOCK_GEN_CONTROL,
+ WM8983_MCLKDIV_MASK, i << WM8983_MCLKDIV_SHIFT);
+
+ /* select the appropriate bclk divider */
+ tmp = (wm8983->sysclk / fs_ratios[i].div) * 10;
+ for (i = 0; i < ARRAY_SIZE(bclk_divs); ++i) {
+ if (wm8983->bclk == tmp / bclk_divs[i])
+ break;
+ }
+
+ if (i == ARRAY_SIZE(bclk_divs)) {
+ dev_err(dai->dev, "No matching BCLK divider found\n");
+ return -EINVAL;
+ }
+
+ dev_dbg(dai->dev, "BCLK div = %d\n", i);
+ snd_soc_update_bits(codec, WM8983_CLOCK_GEN_CONTROL,
+ WM8983_BCLKDIV_MASK, i << WM8983_BCLKDIV_SHIFT);
+
+ return 0;
+}
+
+struct pll_div {
+ u32 div2:1;
+ u32 n:4;
+ u32 k:24;
+};
+
+#define FIXED_PLL_SIZE ((1ULL << 24) * 10)
+static int pll_factors(struct pll_div *pll_div, unsigned int target,
+ unsigned int source)
+{
+ u64 Kpart;
+ unsigned long int K, Ndiv, Nmod;
+
+ pll_div->div2 = 0;
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div->div2 = 1;
+ Ndiv = target / source;
+ }
+
+ if (Ndiv < 6 || Ndiv > 12) {
+ printk(KERN_ERR "%s: WM8983 N value is not within"
+ " the recommended range: %lu\n", __func__, Ndiv);
+ return -EINVAL;
+ }
+ pll_div->n = Ndiv;
+
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (u64)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xffffffff;
+ if ((K % 10) >= 5)
+ K += 5;
+ K /= 10;
+ pll_div->k = K;
+ return 0;
+}
+
+static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id,
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ int ret;
+ struct snd_soc_codec *codec;
+ struct pll_div pll_div;
+
+ codec = dai->codec;
+ if (freq_in && freq_out) {
+ ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in);
+ if (ret)
+ return ret;
+ }
+
+ /* disable the PLL before re-programming it */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1,
+ WM8983_PLLEN_MASK, 0);
+
+ if (!freq_in || !freq_out)
+ return 0;
+
+ /* set PLLN and PRESCALE */
+ snd_soc_write(codec, WM8983_PLL_N,
+ (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT)
+ | pll_div.n);
+ /* set PLLK */
+ snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff);
+ snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff);
+ snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18));
+ /* enable the PLL */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1,
+ WM8983_PLLEN_MASK, WM8983_PLLEN);
+ return 0;
+}
+
+static int wm8983_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8983_priv *wm8983 = snd_soc_codec_get_drvdata(codec);
+
+ switch (clk_id) {
+ case WM8983_CLKSRC_MCLK:
+ snd_soc_update_bits(codec, WM8983_CLOCK_GEN_CONTROL,
+ WM8983_CLKSEL_MASK, 0);
+ break;
+ case WM8983_CLKSRC_PLL:
+ snd_soc_update_bits(codec, WM8983_CLOCK_GEN_CONTROL,
+ WM8983_CLKSEL_MASK, WM8983_CLKSEL);
+ break;
+ default:
+ dev_err(dai->dev, "Unknown clock source: %d\n", clk_id);
+ return -EINVAL;
+ }
+
+ wm8983->sysclk = freq;
+ return 0;
+}
+
+static int wm8983_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ /* VMID at 100k */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1,
+ WM8983_VMIDSEL_MASK,
+ 1 << WM8983_VMIDSEL_SHIFT);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = snd_soc_cache_sync(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+ /* enable anti-pop features */
+ snd_soc_update_bits(codec, WM8983_OUT4_TO_ADC,
+ WM8983_POBCTRL_MASK | WM8983_DELEN_MASK,
+ WM8983_POBCTRL | WM8983_DELEN);
+ /* enable thermal shutdown */
+ snd_soc_update_bits(codec, WM8983_OUTPUT_CTRL,
+ WM8983_TSDEN_MASK, WM8983_TSDEN);
+ /* enable BIASEN */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1,
+ WM8983_BIASEN_MASK, WM8983_BIASEN);
+ /* VMID at 100k */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1,
+ WM8983_VMIDSEL_MASK,
+ 1 << WM8983_VMIDSEL_SHIFT);
+ msleep(250);
+ /* disable anti-pop features */
+ snd_soc_update_bits(codec, WM8983_OUT4_TO_ADC,
+ WM8983_POBCTRL_MASK |
+ WM8983_DELEN_MASK, 0);
+ }
+
+ /* VMID at 500k */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1,
+ WM8983_VMIDSEL_MASK,
+ 2 << WM8983_VMIDSEL_SHIFT);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* disable thermal shutdown */
+ snd_soc_update_bits(codec, WM8983_OUTPUT_CTRL,
+ WM8983_TSDEN_MASK, 0);
+ /* disable VMIDSEL and BIASEN */
+ snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1,
+ WM8983_VMIDSEL_MASK | WM8983_BIASEN_MASK,
+ 0);
+ /* wait for VMID to discharge */
+ msleep(100);
+ snd_soc_write(codec, WM8983_POWER_MANAGEMENT_1, 0);
+ snd_soc_write(codec, WM8983_POWER_MANAGEMENT_2, 0);
+ snd_soc_write(codec, WM8983_POWER_MANAGEMENT_3, 0);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm8983_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8983_resume(struct snd_soc_codec *codec)
+{
+ wm8983_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+#else
+#define wm8983_suspend NULL
+#define wm8983_resume NULL
+#endif
+
+static int wm8983_remove(struct snd_soc_codec *codec)
+{
+ wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8983_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct wm8983_priv *wm8983 = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8983->control_type);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0x8983);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
+ return ret;
+ }
+
+ /* set the vol/gain update bits */
+ for (i = 0; i < ARRAY_SIZE(vol_update_regs); ++i)
+ snd_soc_update_bits(codec, vol_update_regs[i],
+ 0x100, 0x100);
+
+ /* mute all outputs and set PGAs to minimum gain */
+ for (i = WM8983_LOUT1_HP_VOLUME_CTRL;
+ i <= WM8983_OUT4_MONO_MIX_CTRL; ++i)
+ snd_soc_update_bits(codec, i, 0x40, 0x40);
+
+ /* enable soft mute */
+ snd_soc_update_bits(codec, WM8983_DAC_CONTROL,
+ WM8983_SOFTMUTE_MASK,
+ WM8983_SOFTMUTE);
+
+ /* enable BIASCUT */
+ snd_soc_update_bits(codec, WM8983_BIAS_CTRL,
+ WM8983_BIASCUT, WM8983_BIASCUT);
+ return 0;
+}
+
+static struct snd_soc_dai_ops wm8983_dai_ops = {
+ .digital_mute = wm8983_dac_mute,
+ .hw_params = wm8983_hw_params,
+ .set_fmt = wm8983_set_fmt,
+ .set_sysclk = wm8983_set_sysclk,
+ .set_pll = wm8983_set_pll
+};
+
+#define WM8983_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver wm8983_dai = {
+ .name = "wm8983-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = WM8983_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = WM8983_FORMATS,
+ },
+ .ops = &wm8983_dai_ops,
+ .symmetric_rates = 1
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8983 = {
+ .probe = wm8983_probe,
+ .remove = wm8983_remove,
+ .suspend = wm8983_suspend,
+ .resume = wm8983_resume,
+ .set_bias_level = wm8983_set_bias_level,
+ .reg_cache_size = ARRAY_SIZE(wm8983_reg_defs),
+ .reg_word_size = sizeof(u16),
+ .reg_cache_default = wm8983_reg_defs,
+ .controls = wm8983_snd_controls,
+ .num_controls = ARRAY_SIZE(wm8983_snd_controls),
+ .dapm_widgets = wm8983_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm8983_dapm_widgets),
+ .dapm_routes = wm8983_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(wm8983_audio_map),
+ .readable_register = wm8983_readable
+};
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8983_spi_probe(struct spi_device *spi)
+{
+ struct wm8983_priv *wm8983;
+ int ret;
+
+ wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL);
+ if (!wm8983)
+ return -ENOMEM;
+
+ wm8983->control_type = SND_SOC_SPI;
+ spi_set_drvdata(spi, wm8983);
+
+ ret = snd_soc_register_codec(&spi->dev,
+ &soc_codec_dev_wm8983, &wm8983_dai, 1);
+ if (ret < 0)
+ kfree(wm8983);
+ return ret;
+}
+
+static int __devexit wm8983_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ kfree(spi_get_drvdata(spi));
+ return 0;
+}
+
+static struct spi_driver wm8983_spi_driver = {
+ .driver = {
+ .name = "wm8983",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8983_spi_probe,
+ .remove = __devexit_p(wm8983_spi_remove)
+};
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int wm8983_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8983_priv *wm8983;
+ int ret;
+
+ wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL);
+ if (!wm8983)
+ return -ENOMEM;
+
+ wm8983->control_type = SND_SOC_I2C;
+ i2c_set_clientdata(i2c, wm8983);
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_wm8983, &wm8983_dai, 1);
+ if (ret < 0)
+ kfree(wm8983);
+ return ret;
+}
+
+static __devexit int wm8983_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ kfree(i2c_get_clientdata(client));
+ return 0;
+}
+
+static const struct i2c_device_id wm8983_i2c_id[] = {
+ { "wm8983", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8983_i2c_id);
+
+static struct i2c_driver wm8983_i2c_driver = {
+ .driver = {
+ .name = "wm8983",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8983_i2c_probe,
+ .remove = __devexit_p(wm8983_i2c_remove),
+ .id_table = wm8983_i2c_id
+};
+#endif
+
+static int __init wm8983_modinit(void)
+{
+ int ret = 0;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&wm8983_i2c_driver);
+ if (ret) {
+ printk(KERN_ERR "Failed to register wm8983 I2C driver: %d\n",
+ ret);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&wm8983_spi_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register wm8983 SPI driver: %d\n",
+ ret);
+ }
+#endif
+ return ret;
+}
+module_init(wm8983_modinit);
+
+static void __exit wm8983_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8983_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8983_spi_driver);
+#endif
+}
+module_exit(wm8983_exit);
+
+MODULE_DESCRIPTION("ASoC WM8983 driver");
+MODULE_AUTHOR("Dimitris Papastamos <dp@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8983.h b/sound/soc/codecs/wm8983.h
new file mode 100644
index 0000000..71ee619
--- /dev/null
+++ b/sound/soc/codecs/wm8983.h
@@ -0,0 +1,1029 @@
+/*
+ * wm8983.h -- WM8983 ALSA SoC Audio driver
+ *
+ * Copyright 2011 Wolfson Microelectronics plc
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8983_H
+#define _WM8983_H
+
+/*
+ * Register values.
+ */
+#define WM8983_SOFTWARE_RESET 0x00
+#define WM8983_POWER_MANAGEMENT_1 0x01
+#define WM8983_POWER_MANAGEMENT_2 0x02
+#define WM8983_POWER_MANAGEMENT_3 0x03
+#define WM8983_AUDIO_INTERFACE 0x04
+#define WM8983_COMPANDING_CONTROL 0x05
+#define WM8983_CLOCK_GEN_CONTROL 0x06
+#define WM8983_ADDITIONAL_CONTROL 0x07
+#define WM8983_GPIO_CONTROL 0x08
+#define WM8983_JACK_DETECT_CONTROL_1 0x09
+#define WM8983_DAC_CONTROL 0x0A
+#define WM8983_LEFT_DAC_DIGITAL_VOL 0x0B
+#define WM8983_RIGHT_DAC_DIGITAL_VOL 0x0C
+#define WM8983_JACK_DETECT_CONTROL_2 0x0D
+#define WM8983_ADC_CONTROL 0x0E
+#define WM8983_LEFT_ADC_DIGITAL_VOL 0x0F
+#define WM8983_RIGHT_ADC_DIGITAL_VOL 0x10
+#define WM8983_EQ1_LOW_SHELF 0x12
+#define WM8983_EQ2_PEAK_1 0x13
+#define WM8983_EQ3_PEAK_2 0x14
+#define WM8983_EQ4_PEAK_3 0x15
+#define WM8983_EQ5_HIGH_SHELF 0x16
+#define WM8983_DAC_LIMITER_1 0x18
+#define WM8983_DAC_LIMITER_2 0x19
+#define WM8983_NOTCH_FILTER_1 0x1B
+#define WM8983_NOTCH_FILTER_2 0x1C
+#define WM8983_NOTCH_FILTER_3 0x1D
+#define WM8983_NOTCH_FILTER_4 0x1E
+#define WM8983_ALC_CONTROL_1 0x20
+#define WM8983_ALC_CONTROL_2 0x21
+#define WM8983_ALC_CONTROL_3 0x22
+#define WM8983_NOISE_GATE 0x23
+#define WM8983_PLL_N 0x24
+#define WM8983_PLL_K_1 0x25
+#define WM8983_PLL_K_2 0x26
+#define WM8983_PLL_K_3 0x27
+#define WM8983_3D_CONTROL 0x29
+#define WM8983_OUT4_TO_ADC 0x2A
+#define WM8983_BEEP_CONTROL 0x2B
+#define WM8983_INPUT_CTRL 0x2C
+#define WM8983_LEFT_INP_PGA_GAIN_CTRL 0x2D
+#define WM8983_RIGHT_INP_PGA_GAIN_CTRL 0x2E
+#define WM8983_LEFT_ADC_BOOST_CTRL 0x2F
+#define WM8983_RIGHT_ADC_BOOST_CTRL 0x30
+#define WM8983_OUTPUT_CTRL 0x31
+#define WM8983_LEFT_MIXER_CTRL 0x32
+#define WM8983_RIGHT_MIXER_CTRL 0x33
+#define WM8983_LOUT1_HP_VOLUME_CTRL 0x34
+#define WM8983_ROUT1_HP_VOLUME_CTRL 0x35
+#define WM8983_LOUT2_SPK_VOLUME_CTRL 0x36
+#define WM8983_ROUT2_SPK_VOLUME_CTRL 0x37
+#define WM8983_OUT3_MIXER_CTRL 0x38
+#define WM8983_OUT4_MONO_MIX_CTRL 0x39
+#define WM8983_BIAS_CTRL 0x3D
+
+#define WM8983_REGISTER_COUNT 59
+#define WM8983_MAX_REGISTER 0x3F
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Software Reset
+ */
+#define WM8983_SOFTWARE_RESET_MASK 0x01FF /* SOFTWARE_RESET - [8:0] */
+#define WM8983_SOFTWARE_RESET_SHIFT 0 /* SOFTWARE_RESET - [8:0] */
+#define WM8983_SOFTWARE_RESET_WIDTH 9 /* SOFTWARE_RESET - [8:0] */
+
+/*
+ * R1 (0x01) - Power management 1
+ */
+#define WM8983_BUFDCOPEN 0x0100 /* BUFDCOPEN */
+#define WM8983_BUFDCOPEN_MASK 0x0100 /* BUFDCOPEN */
+#define WM8983_BUFDCOPEN_SHIFT 8 /* BUFDCOPEN */
+#define WM8983_BUFDCOPEN_WIDTH 1 /* BUFDCOPEN */
+#define WM8983_OUT4MIXEN 0x0080 /* OUT4MIXEN */
+#define WM8983_OUT4MIXEN_MASK 0x0080 /* OUT4MIXEN */
+#define WM8983_OUT4MIXEN_SHIFT 7 /* OUT4MIXEN */
+#define WM8983_OUT4MIXEN_WIDTH 1 /* OUT4MIXEN */
+#define WM8983_OUT3MIXEN 0x0040 /* OUT3MIXEN */
+#define WM8983_OUT3MIXEN_MASK 0x0040 /* OUT3MIXEN */
+#define WM8983_OUT3MIXEN_SHIFT 6 /* OUT3MIXEN */
+#define WM8983_OUT3MIXEN_WIDTH 1 /* OUT3MIXEN */
+#define WM8983_PLLEN 0x0020 /* PLLEN */
+#define WM8983_PLLEN_MASK 0x0020 /* PLLEN */
+#define WM8983_PLLEN_SHIFT 5 /* PLLEN */
+#define WM8983_PLLEN_WIDTH 1 /* PLLEN */
+#define WM8983_MICBEN 0x0010 /* MICBEN */
+#define WM8983_MICBEN_MASK 0x0010 /* MICBEN */
+#define WM8983_MICBEN_SHIFT 4 /* MICBEN */
+#define WM8983_MICBEN_WIDTH 1 /* MICBEN */
+#define WM8983_BIASEN 0x0008 /* BIASEN */
+#define WM8983_BIASEN_MASK 0x0008 /* BIASEN */
+#define WM8983_BIASEN_SHIFT 3 /* BIASEN */
+#define WM8983_BIASEN_WIDTH 1 /* BIASEN */
+#define WM8983_BUFIOEN 0x0004 /* BUFIOEN */
+#define WM8983_BUFIOEN_MASK 0x0004 /* BUFIOEN */
+#define WM8983_BUFIOEN_SHIFT 2 /* BUFIOEN */
+#define WM8983_BUFIOEN_WIDTH 1 /* BUFIOEN */
+#define WM8983_VMIDSEL_MASK 0x0003 /* VMIDSEL - [1:0] */
+#define WM8983_VMIDSEL_SHIFT 0 /* VMIDSEL - [1:0] */
+#define WM8983_VMIDSEL_WIDTH 2 /* VMIDSEL - [1:0] */
+
+/*
+ * R2 (0x02) - Power management 2
+ */
+#define WM8983_ROUT1EN 0x0100 /* ROUT1EN */
+#define WM8983_ROUT1EN_MASK 0x0100 /* ROUT1EN */
+#define WM8983_ROUT1EN_SHIFT 8 /* ROUT1EN */
+#define WM8983_ROUT1EN_WIDTH 1 /* ROUT1EN */
+#define WM8983_LOUT1EN 0x0080 /* LOUT1EN */
+#define WM8983_LOUT1EN_MASK 0x0080 /* LOUT1EN */
+#define WM8983_LOUT1EN_SHIFT 7 /* LOUT1EN */
+#define WM8983_LOUT1EN_WIDTH 1 /* LOUT1EN */
+#define WM8983_SLEEP 0x0040 /* SLEEP */
+#define WM8983_SLEEP_MASK 0x0040 /* SLEEP */
+#define WM8983_SLEEP_SHIFT 6 /* SLEEP */
+#define WM8983_SLEEP_WIDTH 1 /* SLEEP */
+#define WM8983_BOOSTENR 0x0020 /* BOOSTENR */
+#define WM8983_BOOSTENR_MASK 0x0020 /* BOOSTENR */
+#define WM8983_BOOSTENR_SHIFT 5 /* BOOSTENR */
+#define WM8983_BOOSTENR_WIDTH 1 /* BOOSTENR */
+#define WM8983_BOOSTENL 0x0010 /* BOOSTENL */
+#define WM8983_BOOSTENL_MASK 0x0010 /* BOOSTENL */
+#define WM8983_BOOSTENL_SHIFT 4 /* BOOSTENL */
+#define WM8983_BOOSTENL_WIDTH 1 /* BOOSTENL */
+#define WM8983_INPGAENR 0x0008 /* INPGAENR */
+#define WM8983_INPGAENR_MASK 0x0008 /* INPGAENR */
+#define WM8983_INPGAENR_SHIFT 3 /* INPGAENR */
+#define WM8983_INPGAENR_WIDTH 1 /* INPGAENR */
+#define WM8983_INPPGAENL 0x0004 /* INPPGAENL */
+#define WM8983_INPPGAENL_MASK 0x0004 /* INPPGAENL */
+#define WM8983_INPPGAENL_SHIFT 2 /* INPPGAENL */
+#define WM8983_INPPGAENL_WIDTH 1 /* INPPGAENL */
+#define WM8983_ADCENR 0x0002 /* ADCENR */
+#define WM8983_ADCENR_MASK 0x0002 /* ADCENR */
+#define WM8983_ADCENR_SHIFT 1 /* ADCENR */
+#define WM8983_ADCENR_WIDTH 1 /* ADCENR */
+#define WM8983_ADCENL 0x0001 /* ADCENL */
+#define WM8983_ADCENL_MASK 0x0001 /* ADCENL */
+#define WM8983_ADCENL_SHIFT 0 /* ADCENL */
+#define WM8983_ADCENL_WIDTH 1 /* ADCENL */
+
+/*
+ * R3 (0x03) - Power management 3
+ */
+#define WM8983_OUT4EN 0x0100 /* OUT4EN */
+#define WM8983_OUT4EN_MASK 0x0100 /* OUT4EN */
+#define WM8983_OUT4EN_SHIFT 8 /* OUT4EN */
+#define WM8983_OUT4EN_WIDTH 1 /* OUT4EN */
+#define WM8983_OUT3EN 0x0080 /* OUT3EN */
+#define WM8983_OUT3EN_MASK 0x0080 /* OUT3EN */
+#define WM8983_OUT3EN_SHIFT 7 /* OUT3EN */
+#define WM8983_OUT3EN_WIDTH 1 /* OUT3EN */
+#define WM8983_LOUT2EN 0x0040 /* LOUT2EN */
+#define WM8983_LOUT2EN_MASK 0x0040 /* LOUT2EN */
+#define WM8983_LOUT2EN_SHIFT 6 /* LOUT2EN */
+#define WM8983_LOUT2EN_WIDTH 1 /* LOUT2EN */
+#define WM8983_ROUT2EN 0x0020 /* ROUT2EN */
+#define WM8983_ROUT2EN_MASK 0x0020 /* ROUT2EN */
+#define WM8983_ROUT2EN_SHIFT 5 /* ROUT2EN */
+#define WM8983_ROUT2EN_WIDTH 1 /* ROUT2EN */
+#define WM8983_RMIXEN 0x0008 /* RMIXEN */
+#define WM8983_RMIXEN_MASK 0x0008 /* RMIXEN */
+#define WM8983_RMIXEN_SHIFT 3 /* RMIXEN */
+#define WM8983_RMIXEN_WIDTH 1 /* RMIXEN */
+#define WM8983_LMIXEN 0x0004 /* LMIXEN */
+#define WM8983_LMIXEN_MASK 0x0004 /* LMIXEN */
+#define WM8983_LMIXEN_SHIFT 2 /* LMIXEN */
+#define WM8983_LMIXEN_WIDTH 1 /* LMIXEN */
+#define WM8983_DACENR 0x0002 /* DACENR */
+#define WM8983_DACENR_MASK 0x0002 /* DACENR */
+#define WM8983_DACENR_SHIFT 1 /* DACENR */
+#define WM8983_DACENR_WIDTH 1 /* DACENR */
+#define WM8983_DACENL 0x0001 /* DACENL */
+#define WM8983_DACENL_MASK 0x0001 /* DACENL */
+#define WM8983_DACENL_SHIFT 0 /* DACENL */
+#define WM8983_DACENL_WIDTH 1 /* DACENL */
+
+/*
+ * R4 (0x04) - Audio Interface
+ */
+#define WM8983_BCP 0x0100 /* BCP */
+#define WM8983_BCP_MASK 0x0100 /* BCP */
+#define WM8983_BCP_SHIFT 8 /* BCP */
+#define WM8983_BCP_WIDTH 1 /* BCP */
+#define WM8983_LRCP 0x0080 /* LRCP */
+#define WM8983_LRCP_MASK 0x0080 /* LRCP */
+#define WM8983_LRCP_SHIFT 7 /* LRCP */
+#define WM8983_LRCP_WIDTH 1 /* LRCP */
+#define WM8983_WL_MASK 0x0060 /* WL - [6:5] */
+#define WM8983_WL_SHIFT 5 /* WL - [6:5] */
+#define WM8983_WL_WIDTH 2 /* WL - [6:5] */
+#define WM8983_FMT_MASK 0x0018 /* FMT - [4:3] */
+#define WM8983_FMT_SHIFT 3 /* FMT - [4:3] */
+#define WM8983_FMT_WIDTH 2 /* FMT - [4:3] */
+#define WM8983_DLRSWAP 0x0004 /* DLRSWAP */
+#define WM8983_DLRSWAP_MASK 0x0004 /* DLRSWAP */
+#define WM8983_DLRSWAP_SHIFT 2 /* DLRSWAP */
+#define WM8983_DLRSWAP_WIDTH 1 /* DLRSWAP */
+#define WM8983_ALRSWAP 0x0002 /* ALRSWAP */
+#define WM8983_ALRSWAP_MASK 0x0002 /* ALRSWAP */
+#define WM8983_ALRSWAP_SHIFT 1 /* ALRSWAP */
+#define WM8983_ALRSWAP_WIDTH 1 /* ALRSWAP */
+#define WM8983_MONO 0x0001 /* MONO */
+#define WM8983_MONO_MASK 0x0001 /* MONO */
+#define WM8983_MONO_SHIFT 0 /* MONO */
+#define WM8983_MONO_WIDTH 1 /* MONO */
+
+/*
+ * R5 (0x05) - Companding control
+ */
+#define WM8983_WL8 0x0020 /* WL8 */
+#define WM8983_WL8_MASK 0x0020 /* WL8 */
+#define WM8983_WL8_SHIFT 5 /* WL8 */
+#define WM8983_WL8_WIDTH 1 /* WL8 */
+#define WM8983_DAC_COMP_MASK 0x0018 /* DAC_COMP - [4:3] */
+#define WM8983_DAC_COMP_SHIFT 3 /* DAC_COMP - [4:3] */
+#define WM8983_DAC_COMP_WIDTH 2 /* DAC_COMP - [4:3] */
+#define WM8983_ADC_COMP_MASK 0x0006 /* ADC_COMP - [2:1] */
+#define WM8983_ADC_COMP_SHIFT 1 /* ADC_COMP - [2:1] */
+#define WM8983_ADC_COMP_WIDTH 2 /* ADC_COMP - [2:1] */
+#define WM8983_LOOPBACK 0x0001 /* LOOPBACK */
+#define WM8983_LOOPBACK_MASK 0x0001 /* LOOPBACK */
+#define WM8983_LOOPBACK_SHIFT 0 /* LOOPBACK */
+#define WM8983_LOOPBACK_WIDTH 1 /* LOOPBACK */
+
+/*
+ * R6 (0x06) - Clock Gen control
+ */
+#define WM8983_CLKSEL 0x0100 /* CLKSEL */
+#define WM8983_CLKSEL_MASK 0x0100 /* CLKSEL */
+#define WM8983_CLKSEL_SHIFT 8 /* CLKSEL */
+#define WM8983_CLKSEL_WIDTH 1 /* CLKSEL */
+#define WM8983_MCLKDIV_MASK 0x00E0 /* MCLKDIV - [7:5] */
+#define WM8983_MCLKDIV_SHIFT 5 /* MCLKDIV - [7:5] */
+#define WM8983_MCLKDIV_WIDTH 3 /* MCLKDIV - [7:5] */
+#define WM8983_BCLKDIV_MASK 0x001C /* BCLKDIV - [4:2] */
+#define WM8983_BCLKDIV_SHIFT 2 /* BCLKDIV - [4:2] */
+#define WM8983_BCLKDIV_WIDTH 3 /* BCLKDIV - [4:2] */
+#define WM8983_MS 0x0001 /* MS */
+#define WM8983_MS_MASK 0x0001 /* MS */
+#define WM8983_MS_SHIFT 0 /* MS */
+#define WM8983_MS_WIDTH 1 /* MS */
+
+/*
+ * R7 (0x07) - Additional control
+ */
+#define WM8983_SR_MASK 0x000E /* SR - [3:1] */
+#define WM8983_SR_SHIFT 1 /* SR - [3:1] */
+#define WM8983_SR_WIDTH 3 /* SR - [3:1] */
+#define WM8983_SLOWCLKEN 0x0001 /* SLOWCLKEN */
+#define WM8983_SLOWCLKEN_MASK 0x0001 /* SLOWCLKEN */
+#define WM8983_SLOWCLKEN_SHIFT 0 /* SLOWCLKEN */
+#define WM8983_SLOWCLKEN_WIDTH 1 /* SLOWCLKEN */
+
+/*
+ * R8 (0x08) - GPIO Control
+ */
+#define WM8983_OPCLKDIV_MASK 0x0030 /* OPCLKDIV - [5:4] */
+#define WM8983_OPCLKDIV_SHIFT 4 /* OPCLKDIV - [5:4] */
+#define WM8983_OPCLKDIV_WIDTH 2 /* OPCLKDIV - [5:4] */
+#define WM8983_GPIO1POL 0x0008 /* GPIO1POL */
+#define WM8983_GPIO1POL_MASK 0x0008 /* GPIO1POL */
+#define WM8983_GPIO1POL_SHIFT 3 /* GPIO1POL */
+#define WM8983_GPIO1POL_WIDTH 1 /* GPIO1POL */
+#define WM8983_GPIO1SEL_MASK 0x0007 /* GPIO1SEL - [2:0] */
+#define WM8983_GPIO1SEL_SHIFT 0 /* GPIO1SEL - [2:0] */
+#define WM8983_GPIO1SEL_WIDTH 3 /* GPIO1SEL - [2:0] */
+
+/*
+ * R9 (0x09) - Jack Detect Control 1
+ */
+#define WM8983_JD_VMID1 0x0100 /* JD_VMID1 */
+#define WM8983_JD_VMID1_MASK 0x0100 /* JD_VMID1 */
+#define WM8983_JD_VMID1_SHIFT 8 /* JD_VMID1 */
+#define WM8983_JD_VMID1_WIDTH 1 /* JD_VMID1 */
+#define WM8983_JD_VMID0 0x0080 /* JD_VMID0 */
+#define WM8983_JD_VMID0_MASK 0x0080 /* JD_VMID0 */
+#define WM8983_JD_VMID0_SHIFT 7 /* JD_VMID0 */
+#define WM8983_JD_VMID0_WIDTH 1 /* JD_VMID0 */
+#define WM8983_JD_EN 0x0040 /* JD_EN */
+#define WM8983_JD_EN_MASK 0x0040 /* JD_EN */
+#define WM8983_JD_EN_SHIFT 6 /* JD_EN */
+#define WM8983_JD_EN_WIDTH 1 /* JD_EN */
+#define WM8983_JD_SEL_MASK 0x0030 /* JD_SEL - [5:4] */
+#define WM8983_JD_SEL_SHIFT 4 /* JD_SEL - [5:4] */
+#define WM8983_JD_SEL_WIDTH 2 /* JD_SEL - [5:4] */
+
+/*
+ * R10 (0x0A) - DAC Control
+ */
+#define WM8983_SOFTMUTE 0x0040 /* SOFTMUTE */
+#define WM8983_SOFTMUTE_MASK 0x0040 /* SOFTMUTE */
+#define WM8983_SOFTMUTE_SHIFT 6 /* SOFTMUTE */
+#define WM8983_SOFTMUTE_WIDTH 1 /* SOFTMUTE */
+#define WM8983_DACOSR128 0x0008 /* DACOSR128 */
+#define WM8983_DACOSR128_MASK 0x0008 /* DACOSR128 */
+#define WM8983_DACOSR128_SHIFT 3 /* DACOSR128 */
+#define WM8983_DACOSR128_WIDTH 1 /* DACOSR128 */
+#define WM8983_AMUTE 0x0004 /* AMUTE */
+#define WM8983_AMUTE_MASK 0x0004 /* AMUTE */
+#define WM8983_AMUTE_SHIFT 2 /* AMUTE */
+#define WM8983_AMUTE_WIDTH 1 /* AMUTE */
+#define WM8983_DACRPOL 0x0002 /* DACRPOL */
+#define WM8983_DACRPOL_MASK 0x0002 /* DACRPOL */
+#define WM8983_DACRPOL_SHIFT 1 /* DACRPOL */
+#define WM8983_DACRPOL_WIDTH 1 /* DACRPOL */
+#define WM8983_DACLPOL 0x0001 /* DACLPOL */
+#define WM8983_DACLPOL_MASK 0x0001 /* DACLPOL */
+#define WM8983_DACLPOL_SHIFT 0 /* DACLPOL */
+#define WM8983_DACLPOL_WIDTH 1 /* DACLPOL */
+
+/*
+ * R11 (0x0B) - Left DAC digital Vol
+ */
+#define WM8983_DACVU 0x0100 /* DACVU */
+#define WM8983_DACVU_MASK 0x0100 /* DACVU */
+#define WM8983_DACVU_SHIFT 8 /* DACVU */
+#define WM8983_DACVU_WIDTH 1 /* DACVU */
+#define WM8983_DACLVOL_MASK 0x00FF /* DACLVOL - [7:0] */
+#define WM8983_DACLVOL_SHIFT 0 /* DACLVOL - [7:0] */
+#define WM8983_DACLVOL_WIDTH 8 /* DACLVOL - [7:0] */
+
+/*
+ * R12 (0x0C) - Right DAC digital vol
+ */
+#define WM8983_DACVU 0x0100 /* DACVU */
+#define WM8983_DACVU_MASK 0x0100 /* DACVU */
+#define WM8983_DACVU_SHIFT 8 /* DACVU */
+#define WM8983_DACVU_WIDTH 1 /* DACVU */
+#define WM8983_DACRVOL_MASK 0x00FF /* DACRVOL - [7:0] */
+#define WM8983_DACRVOL_SHIFT 0 /* DACRVOL - [7:0] */
+#define WM8983_DACRVOL_WIDTH 8 /* DACRVOL - [7:0] */
+
+/*
+ * R13 (0x0D) - Jack Detect Control 2
+ */
+#define WM8983_JD_EN1_MASK 0x00F0 /* JD_EN1 - [7:4] */
+#define WM8983_JD_EN1_SHIFT 4 /* JD_EN1 - [7:4] */
+#define WM8983_JD_EN1_WIDTH 4 /* JD_EN1 - [7:4] */
+#define WM8983_JD_EN0_MASK 0x000F /* JD_EN0 - [3:0] */
+#define WM8983_JD_EN0_SHIFT 0 /* JD_EN0 - [3:0] */
+#define WM8983_JD_EN0_WIDTH 4 /* JD_EN0 - [3:0] */
+
+/*
+ * R14 (0x0E) - ADC Control
+ */
+#define WM8983_HPFEN 0x0100 /* HPFEN */
+#define WM8983_HPFEN_MASK 0x0100 /* HPFEN */
+#define WM8983_HPFEN_SHIFT 8 /* HPFEN */
+#define WM8983_HPFEN_WIDTH 1 /* HPFEN */
+#define WM8983_HPFAPP 0x0080 /* HPFAPP */
+#define WM8983_HPFAPP_MASK 0x0080 /* HPFAPP */
+#define WM8983_HPFAPP_SHIFT 7 /* HPFAPP */
+#define WM8983_HPFAPP_WIDTH 1 /* HPFAPP */
+#define WM8983_HPFCUT_MASK 0x0070 /* HPFCUT - [6:4] */
+#define WM8983_HPFCUT_SHIFT 4 /* HPFCUT - [6:4] */
+#define WM8983_HPFCUT_WIDTH 3 /* HPFCUT - [6:4] */
+#define WM8983_ADCOSR128 0x0008 /* ADCOSR128 */
+#define WM8983_ADCOSR128_MASK 0x0008 /* ADCOSR128 */
+#define WM8983_ADCOSR128_SHIFT 3 /* ADCOSR128 */
+#define WM8983_ADCOSR128_WIDTH 1 /* ADCOSR128 */
+#define WM8983_ADCRPOL 0x0002 /* ADCRPOL */
+#define WM8983_ADCRPOL_MASK 0x0002 /* ADCRPOL */
+#define WM8983_ADCRPOL_SHIFT 1 /* ADCRPOL */
+#define WM8983_ADCRPOL_WIDTH 1 /* ADCRPOL */
+#define WM8983_ADCLPOL 0x0001 /* ADCLPOL */
+#define WM8983_ADCLPOL_MASK 0x0001 /* ADCLPOL */
+#define WM8983_ADCLPOL_SHIFT 0 /* ADCLPOL */
+#define WM8983_ADCLPOL_WIDTH 1 /* ADCLPOL */
+
+/*
+ * R15 (0x0F) - Left ADC Digital Vol
+ */
+#define WM8983_ADCVU 0x0100 /* ADCVU */
+#define WM8983_ADCVU_MASK 0x0100 /* ADCVU */
+#define WM8983_ADCVU_SHIFT 8 /* ADCVU */
+#define WM8983_ADCVU_WIDTH 1 /* ADCVU */
+#define WM8983_ADCLVOL_MASK 0x00FF /* ADCLVOL - [7:0] */
+#define WM8983_ADCLVOL_SHIFT 0 /* ADCLVOL - [7:0] */
+#define WM8983_ADCLVOL_WIDTH 8 /* ADCLVOL - [7:0] */
+
+/*
+ * R16 (0x10) - Right ADC Digital Vol
+ */
+#define WM8983_ADCVU 0x0100 /* ADCVU */
+#define WM8983_ADCVU_MASK 0x0100 /* ADCVU */
+#define WM8983_ADCVU_SHIFT 8 /* ADCVU */
+#define WM8983_ADCVU_WIDTH 1 /* ADCVU */
+#define WM8983_ADCRVOL_MASK 0x00FF /* ADCRVOL - [7:0] */
+#define WM8983_ADCRVOL_SHIFT 0 /* ADCRVOL - [7:0] */
+#define WM8983_ADCRVOL_WIDTH 8 /* ADCRVOL - [7:0] */
+
+/*
+ * R18 (0x12) - EQ1 - low shelf
+ */
+#define WM8983_EQ3DMODE 0x0100 /* EQ3DMODE */
+#define WM8983_EQ3DMODE_MASK 0x0100 /* EQ3DMODE */
+#define WM8983_EQ3DMODE_SHIFT 8 /* EQ3DMODE */
+#define WM8983_EQ3DMODE_WIDTH 1 /* EQ3DMODE */
+#define WM8983_EQ1C_MASK 0x0060 /* EQ1C - [6:5] */
+#define WM8983_EQ1C_SHIFT 5 /* EQ1C - [6:5] */
+#define WM8983_EQ1C_WIDTH 2 /* EQ1C - [6:5] */
+#define WM8983_EQ1G_MASK 0x001F /* EQ1G - [4:0] */
+#define WM8983_EQ1G_SHIFT 0 /* EQ1G - [4:0] */
+#define WM8983_EQ1G_WIDTH 5 /* EQ1G - [4:0] */
+
+/*
+ * R19 (0x13) - EQ2 - peak 1
+ */
+#define WM8983_EQ2BW 0x0100 /* EQ2BW */
+#define WM8983_EQ2BW_MASK 0x0100 /* EQ2BW */
+#define WM8983_EQ2BW_SHIFT 8 /* EQ2BW */
+#define WM8983_EQ2BW_WIDTH 1 /* EQ2BW */
+#define WM8983_EQ2C_MASK 0x0060 /* EQ2C - [6:5] */
+#define WM8983_EQ2C_SHIFT 5 /* EQ2C - [6:5] */
+#define WM8983_EQ2C_WIDTH 2 /* EQ2C - [6:5] */
+#define WM8983_EQ2G_MASK 0x001F /* EQ2G - [4:0] */
+#define WM8983_EQ2G_SHIFT 0 /* EQ2G - [4:0] */
+#define WM8983_EQ2G_WIDTH 5 /* EQ2G - [4:0] */
+
+/*
+ * R20 (0x14) - EQ3 - peak 2
+ */
+#define WM8983_EQ3BW 0x0100 /* EQ3BW */
+#define WM8983_EQ3BW_MASK 0x0100 /* EQ3BW */
+#define WM8983_EQ3BW_SHIFT 8 /* EQ3BW */
+#define WM8983_EQ3BW_WIDTH 1 /* EQ3BW */
+#define WM8983_EQ3C_MASK 0x0060 /* EQ3C - [6:5] */
+#define WM8983_EQ3C_SHIFT 5 /* EQ3C - [6:5] */
+#define WM8983_EQ3C_WIDTH 2 /* EQ3C - [6:5] */
+#define WM8983_EQ3G_MASK 0x001F /* EQ3G - [4:0] */
+#define WM8983_EQ3G_SHIFT 0 /* EQ3G - [4:0] */
+#define WM8983_EQ3G_WIDTH 5 /* EQ3G - [4:0] */
+
+/*
+ * R21 (0x15) - EQ4 - peak 3
+ */
+#define WM8983_EQ4BW 0x0100 /* EQ4BW */
+#define WM8983_EQ4BW_MASK 0x0100 /* EQ4BW */
+#define WM8983_EQ4BW_SHIFT 8 /* EQ4BW */
+#define WM8983_EQ4BW_WIDTH 1 /* EQ4BW */
+#define WM8983_EQ4C_MASK 0x0060 /* EQ4C - [6:5] */
+#define WM8983_EQ4C_SHIFT 5 /* EQ4C - [6:5] */
+#define WM8983_EQ4C_WIDTH 2 /* EQ4C - [6:5] */
+#define WM8983_EQ4G_MASK 0x001F /* EQ4G - [4:0] */
+#define WM8983_EQ4G_SHIFT 0 /* EQ4G - [4:0] */
+#define WM8983_EQ4G_WIDTH 5 /* EQ4G - [4:0] */
+
+/*
+ * R22 (0x16) - EQ5 - high shelf
+ */
+#define WM8983_EQ5C_MASK 0x0060 /* EQ5C - [6:5] */
+#define WM8983_EQ5C_SHIFT 5 /* EQ5C - [6:5] */
+#define WM8983_EQ5C_WIDTH 2 /* EQ5C - [6:5] */
+#define WM8983_EQ5G_MASK 0x001F /* EQ5G - [4:0] */
+#define WM8983_EQ5G_SHIFT 0 /* EQ5G - [4:0] */
+#define WM8983_EQ5G_WIDTH 5 /* EQ5G - [4:0] */
+
+/*
+ * R24 (0x18) - DAC Limiter 1
+ */
+#define WM8983_LIMEN 0x0100 /* LIMEN */
+#define WM8983_LIMEN_MASK 0x0100 /* LIMEN */
+#define WM8983_LIMEN_SHIFT 8 /* LIMEN */
+#define WM8983_LIMEN_WIDTH 1 /* LIMEN */
+#define WM8983_LIMDCY_MASK 0x00F0 /* LIMDCY - [7:4] */
+#define WM8983_LIMDCY_SHIFT 4 /* LIMDCY - [7:4] */
+#define WM8983_LIMDCY_WIDTH 4 /* LIMDCY - [7:4] */
+#define WM8983_LIMATK_MASK 0x000F /* LIMATK - [3:0] */
+#define WM8983_LIMATK_SHIFT 0 /* LIMATK - [3:0] */
+#define WM8983_LIMATK_WIDTH 4 /* LIMATK - [3:0] */
+
+/*
+ * R25 (0x19) - DAC Limiter 2
+ */
+#define WM8983_LIMLVL_MASK 0x0070 /* LIMLVL - [6:4] */
+#define WM8983_LIMLVL_SHIFT 4 /* LIMLVL - [6:4] */
+#define WM8983_LIMLVL_WIDTH 3 /* LIMLVL - [6:4] */
+#define WM8983_LIMBOOST_MASK 0x000F /* LIMBOOST - [3:0] */
+#define WM8983_LIMBOOST_SHIFT 0 /* LIMBOOST - [3:0] */
+#define WM8983_LIMBOOST_WIDTH 4 /* LIMBOOST - [3:0] */
+
+/*
+ * R27 (0x1B) - Notch Filter 1
+ */
+#define WM8983_NFU 0x0100 /* NFU */
+#define WM8983_NFU_MASK 0x0100 /* NFU */
+#define WM8983_NFU_SHIFT 8 /* NFU */
+#define WM8983_NFU_WIDTH 1 /* NFU */
+#define WM8983_NFEN 0x0080 /* NFEN */
+#define WM8983_NFEN_MASK 0x0080 /* NFEN */
+#define WM8983_NFEN_SHIFT 7 /* NFEN */
+#define WM8983_NFEN_WIDTH 1 /* NFEN */
+#define WM8983_NFA0_13_7_MASK 0x007F /* NFA0(13:7) - [6:0] */
+#define WM8983_NFA0_13_7_SHIFT 0 /* NFA0(13:7) - [6:0] */
+#define WM8983_NFA0_13_7_WIDTH 7 /* NFA0(13:7) - [6:0] */
+
+/*
+ * R28 (0x1C) - Notch Filter 2
+ */
+#define WM8983_NFU 0x0100 /* NFU */
+#define WM8983_NFU_MASK 0x0100 /* NFU */
+#define WM8983_NFU_SHIFT 8 /* NFU */
+#define WM8983_NFU_WIDTH 1 /* NFU */
+#define WM8983_NFA0_6_0_MASK 0x007F /* NFA0(6:0) - [6:0] */
+#define WM8983_NFA0_6_0_SHIFT 0 /* NFA0(6:0) - [6:0] */
+#define WM8983_NFA0_6_0_WIDTH 7 /* NFA0(6:0) - [6:0] */
+
+/*
+ * R29 (0x1D) - Notch Filter 3
+ */
+#define WM8983_NFU 0x0100 /* NFU */
+#define WM8983_NFU_MASK 0x0100 /* NFU */
+#define WM8983_NFU_SHIFT 8 /* NFU */
+#define WM8983_NFU_WIDTH 1 /* NFU */
+#define WM8983_NFA1_13_7_MASK 0x007F /* NFA1(13:7) - [6:0] */
+#define WM8983_NFA1_13_7_SHIFT 0 /* NFA1(13:7) - [6:0] */
+#define WM8983_NFA1_13_7_WIDTH 7 /* NFA1(13:7) - [6:0] */
+
+/*
+ * R30 (0x1E) - Notch Filter 4
+ */
+#define WM8983_NFU 0x0100 /* NFU */
+#define WM8983_NFU_MASK 0x0100 /* NFU */
+#define WM8983_NFU_SHIFT 8 /* NFU */
+#define WM8983_NFU_WIDTH 1 /* NFU */
+#define WM8983_NFA1_6_0_MASK 0x007F /* NFA1(6:0) - [6:0] */
+#define WM8983_NFA1_6_0_SHIFT 0 /* NFA1(6:0) - [6:0] */
+#define WM8983_NFA1_6_0_WIDTH 7 /* NFA1(6:0) - [6:0] */
+
+/*
+ * R32 (0x20) - ALC control 1
+ */
+#define WM8983_ALCSEL_MASK 0x0180 /* ALCSEL - [8:7] */
+#define WM8983_ALCSEL_SHIFT 7 /* ALCSEL - [8:7] */
+#define WM8983_ALCSEL_WIDTH 2 /* ALCSEL - [8:7] */
+#define WM8983_ALCMAX_MASK 0x0038 /* ALCMAX - [5:3] */
+#define WM8983_ALCMAX_SHIFT 3 /* ALCMAX - [5:3] */
+#define WM8983_ALCMAX_WIDTH 3 /* ALCMAX - [5:3] */
+#define WM8983_ALCMIN_MASK 0x0007 /* ALCMIN - [2:0] */
+#define WM8983_ALCMIN_SHIFT 0 /* ALCMIN - [2:0] */
+#define WM8983_ALCMIN_WIDTH 3 /* ALCMIN - [2:0] */
+
+/*
+ * R33 (0x21) - ALC control 2
+ */
+#define WM8983_ALCHLD_MASK 0x00F0 /* ALCHLD - [7:4] */
+#define WM8983_ALCHLD_SHIFT 4 /* ALCHLD - [7:4] */
+#define WM8983_ALCHLD_WIDTH 4 /* ALCHLD - [7:4] */
+#define WM8983_ALCLVL_MASK 0x000F /* ALCLVL - [3:0] */
+#define WM8983_ALCLVL_SHIFT 0 /* ALCLVL - [3:0] */
+#define WM8983_ALCLVL_WIDTH 4 /* ALCLVL - [3:0] */
+
+/*
+ * R34 (0x22) - ALC control 3
+ */
+#define WM8983_ALCMODE 0x0100 /* ALCMODE */
+#define WM8983_ALCMODE_MASK 0x0100 /* ALCMODE */
+#define WM8983_ALCMODE_SHIFT 8 /* ALCMODE */
+#define WM8983_ALCMODE_WIDTH 1 /* ALCMODE */
+#define WM8983_ALCDCY_MASK 0x00F0 /* ALCDCY - [7:4] */
+#define WM8983_ALCDCY_SHIFT 4 /* ALCDCY - [7:4] */
+#define WM8983_ALCDCY_WIDTH 4 /* ALCDCY - [7:4] */
+#define WM8983_ALCATK_MASK 0x000F /* ALCATK - [3:0] */
+#define WM8983_ALCATK_SHIFT 0 /* ALCATK - [3:0] */
+#define WM8983_ALCATK_WIDTH 4 /* ALCATK - [3:0] */
+
+/*
+ * R35 (0x23) - Noise Gate
+ */
+#define WM8983_NGEN 0x0008 /* NGEN */
+#define WM8983_NGEN_MASK 0x0008 /* NGEN */
+#define WM8983_NGEN_SHIFT 3 /* NGEN */
+#define WM8983_NGEN_WIDTH 1 /* NGEN */
+#define WM8983_NGTH_MASK 0x0007 /* NGTH - [2:0] */
+#define WM8983_NGTH_SHIFT 0 /* NGTH - [2:0] */
+#define WM8983_NGTH_WIDTH 3 /* NGTH - [2:0] */
+
+/*
+ * R36 (0x24) - PLL N
+ */
+#define WM8983_PLL_PRESCALE 0x0010 /* PLL_PRESCALE */
+#define WM8983_PLL_PRESCALE_MASK 0x0010 /* PLL_PRESCALE */
+#define WM8983_PLL_PRESCALE_SHIFT 4 /* PLL_PRESCALE */
+#define WM8983_PLL_PRESCALE_WIDTH 1 /* PLL_PRESCALE */
+#define WM8983_PLLN_MASK 0x000F /* PLLN - [3:0] */
+#define WM8983_PLLN_SHIFT 0 /* PLLN - [3:0] */
+#define WM8983_PLLN_WIDTH 4 /* PLLN - [3:0] */
+
+/*
+ * R37 (0x25) - PLL K 1
+ */
+#define WM8983_PLLK_23_18_MASK 0x003F /* PLLK(23:18) - [5:0] */
+#define WM8983_PLLK_23_18_SHIFT 0 /* PLLK(23:18) - [5:0] */
+#define WM8983_PLLK_23_18_WIDTH 6 /* PLLK(23:18) - [5:0] */
+
+/*
+ * R38 (0x26) - PLL K 2
+ */
+#define WM8983_PLLK_17_9_MASK 0x01FF /* PLLK(17:9) - [8:0] */
+#define WM8983_PLLK_17_9_SHIFT 0 /* PLLK(17:9) - [8:0] */
+#define WM8983_PLLK_17_9_WIDTH 9 /* PLLK(17:9) - [8:0] */
+
+/*
+ * R39 (0x27) - PLL K 3
+ */
+#define WM8983_PLLK_8_0_MASK 0x01FF /* PLLK(8:0) - [8:0] */
+#define WM8983_PLLK_8_0_SHIFT 0 /* PLLK(8:0) - [8:0] */
+#define WM8983_PLLK_8_0_WIDTH 9 /* PLLK(8:0) - [8:0] */
+
+/*
+ * R41 (0x29) - 3D control
+ */
+#define WM8983_DEPTH3D_MASK 0x000F /* DEPTH3D - [3:0] */
+#define WM8983_DEPTH3D_SHIFT 0 /* DEPTH3D - [3:0] */
+#define WM8983_DEPTH3D_WIDTH 4 /* DEPTH3D - [3:0] */
+
+/*
+ * R42 (0x2A) - OUT4 to ADC
+ */
+#define WM8983_OUT4_2ADCVOL_MASK 0x01C0 /* OUT4_2ADCVOL - [8:6] */
+#define WM8983_OUT4_2ADCVOL_SHIFT 6 /* OUT4_2ADCVOL - [8:6] */
+#define WM8983_OUT4_2ADCVOL_WIDTH 3 /* OUT4_2ADCVOL - [8:6] */
+#define WM8983_OUT4_2LNR 0x0020 /* OUT4_2LNR */
+#define WM8983_OUT4_2LNR_MASK 0x0020 /* OUT4_2LNR */
+#define WM8983_OUT4_2LNR_SHIFT 5 /* OUT4_2LNR */
+#define WM8983_OUT4_2LNR_WIDTH 1 /* OUT4_2LNR */
+#define WM8983_POBCTRL 0x0004 /* POBCTRL */
+#define WM8983_POBCTRL_MASK 0x0004 /* POBCTRL */
+#define WM8983_POBCTRL_SHIFT 2 /* POBCTRL */
+#define WM8983_POBCTRL_WIDTH 1 /* POBCTRL */
+#define WM8983_DELEN 0x0002 /* DELEN */
+#define WM8983_DELEN_MASK 0x0002 /* DELEN */
+#define WM8983_DELEN_SHIFT 1 /* DELEN */
+#define WM8983_DELEN_WIDTH 1 /* DELEN */
+#define WM8983_OUT1DEL 0x0001 /* OUT1DEL */
+#define WM8983_OUT1DEL_MASK 0x0001 /* OUT1DEL */
+#define WM8983_OUT1DEL_SHIFT 0 /* OUT1DEL */
+#define WM8983_OUT1DEL_WIDTH 1 /* OUT1DEL */
+
+/*
+ * R43 (0x2B) - Beep control
+ */
+#define WM8983_BYPL2RMIX 0x0100 /* BYPL2RMIX */
+#define WM8983_BYPL2RMIX_MASK 0x0100 /* BYPL2RMIX */
+#define WM8983_BYPL2RMIX_SHIFT 8 /* BYPL2RMIX */
+#define WM8983_BYPL2RMIX_WIDTH 1 /* BYPL2RMIX */
+#define WM8983_BYPR2LMIX 0x0080 /* BYPR2LMIX */
+#define WM8983_BYPR2LMIX_MASK 0x0080 /* BYPR2LMIX */
+#define WM8983_BYPR2LMIX_SHIFT 7 /* BYPR2LMIX */
+#define WM8983_BYPR2LMIX_WIDTH 1 /* BYPR2LMIX */
+#define WM8983_MUTERPGA2INV 0x0020 /* MUTERPGA2INV */
+#define WM8983_MUTERPGA2INV_MASK 0x0020 /* MUTERPGA2INV */
+#define WM8983_MUTERPGA2INV_SHIFT 5 /* MUTERPGA2INV */
+#define WM8983_MUTERPGA2INV_WIDTH 1 /* MUTERPGA2INV */
+#define WM8983_INVROUT2 0x0010 /* INVROUT2 */
+#define WM8983_INVROUT2_MASK 0x0010 /* INVROUT2 */
+#define WM8983_INVROUT2_SHIFT 4 /* INVROUT2 */
+#define WM8983_INVROUT2_WIDTH 1 /* INVROUT2 */
+#define WM8983_BEEPVOL_MASK 0x000E /* BEEPVOL - [3:1] */
+#define WM8983_BEEPVOL_SHIFT 1 /* BEEPVOL - [3:1] */
+#define WM8983_BEEPVOL_WIDTH 3 /* BEEPVOL - [3:1] */
+#define WM8983_BEEPEN 0x0001 /* BEEPEN */
+#define WM8983_BEEPEN_MASK 0x0001 /* BEEPEN */
+#define WM8983_BEEPEN_SHIFT 0 /* BEEPEN */
+#define WM8983_BEEPEN_WIDTH 1 /* BEEPEN */
+
+/*
+ * R44 (0x2C) - Input ctrl
+ */
+#define WM8983_MBVSEL 0x0100 /* MBVSEL */
+#define WM8983_MBVSEL_MASK 0x0100 /* MBVSEL */
+#define WM8983_MBVSEL_SHIFT 8 /* MBVSEL */
+#define WM8983_MBVSEL_WIDTH 1 /* MBVSEL */
+#define WM8983_R2_2INPPGA 0x0040 /* R2_2INPPGA */
+#define WM8983_R2_2INPPGA_MASK 0x0040 /* R2_2INPPGA */
+#define WM8983_R2_2INPPGA_SHIFT 6 /* R2_2INPPGA */
+#define WM8983_R2_2INPPGA_WIDTH 1 /* R2_2INPPGA */
+#define WM8983_RIN2INPPGA 0x0020 /* RIN2INPPGA */
+#define WM8983_RIN2INPPGA_MASK 0x0020 /* RIN2INPPGA */
+#define WM8983_RIN2INPPGA_SHIFT 5 /* RIN2INPPGA */
+#define WM8983_RIN2INPPGA_WIDTH 1 /* RIN2INPPGA */
+#define WM8983_RIP2INPPGA 0x0010 /* RIP2INPPGA */
+#define WM8983_RIP2INPPGA_MASK 0x0010 /* RIP2INPPGA */
+#define WM8983_RIP2INPPGA_SHIFT 4 /* RIP2INPPGA */
+#define WM8983_RIP2INPPGA_WIDTH 1 /* RIP2INPPGA */
+#define WM8983_L2_2INPPGA 0x0004 /* L2_2INPPGA */
+#define WM8983_L2_2INPPGA_MASK 0x0004 /* L2_2INPPGA */
+#define WM8983_L2_2INPPGA_SHIFT 2 /* L2_2INPPGA */
+#define WM8983_L2_2INPPGA_WIDTH 1 /* L2_2INPPGA */
+#define WM8983_LIN2INPPGA 0x0002 /* LIN2INPPGA */
+#define WM8983_LIN2INPPGA_MASK 0x0002 /* LIN2INPPGA */
+#define WM8983_LIN2INPPGA_SHIFT 1 /* LIN2INPPGA */
+#define WM8983_LIN2INPPGA_WIDTH 1 /* LIN2INPPGA */
+#define WM8983_LIP2INPPGA 0x0001 /* LIP2INPPGA */
+#define WM8983_LIP2INPPGA_MASK 0x0001 /* LIP2INPPGA */
+#define WM8983_LIP2INPPGA_SHIFT 0 /* LIP2INPPGA */
+#define WM8983_LIP2INPPGA_WIDTH 1 /* LIP2INPPGA */
+
+/*
+ * R45 (0x2D) - Left INP PGA gain ctrl
+ */
+#define WM8983_INPGAVU 0x0100 /* INPGAVU */
+#define WM8983_INPGAVU_MASK 0x0100 /* INPGAVU */
+#define WM8983_INPGAVU_SHIFT 8 /* INPGAVU */
+#define WM8983_INPGAVU_WIDTH 1 /* INPGAVU */
+#define WM8983_INPPGAZCL 0x0080 /* INPPGAZCL */
+#define WM8983_INPPGAZCL_MASK 0x0080 /* INPPGAZCL */
+#define WM8983_INPPGAZCL_SHIFT 7 /* INPPGAZCL */
+#define WM8983_INPPGAZCL_WIDTH 1 /* INPPGAZCL */
+#define WM8983_INPPGAMUTEL 0x0040 /* INPPGAMUTEL */
+#define WM8983_INPPGAMUTEL_MASK 0x0040 /* INPPGAMUTEL */
+#define WM8983_INPPGAMUTEL_SHIFT 6 /* INPPGAMUTEL */
+#define WM8983_INPPGAMUTEL_WIDTH 1 /* INPPGAMUTEL */
+#define WM8983_INPPGAVOLL_MASK 0x003F /* INPPGAVOLL - [5:0] */
+#define WM8983_INPPGAVOLL_SHIFT 0 /* INPPGAVOLL - [5:0] */
+#define WM8983_INPPGAVOLL_WIDTH 6 /* INPPGAVOLL - [5:0] */
+
+/*
+ * R46 (0x2E) - Right INP PGA gain ctrl
+ */
+#define WM8983_INPGAVU 0x0100 /* INPGAVU */
+#define WM8983_INPGAVU_MASK 0x0100 /* INPGAVU */
+#define WM8983_INPGAVU_SHIFT 8 /* INPGAVU */
+#define WM8983_INPGAVU_WIDTH 1 /* INPGAVU */
+#define WM8983_INPPGAZCR 0x0080 /* INPPGAZCR */
+#define WM8983_INPPGAZCR_MASK 0x0080 /* INPPGAZCR */
+#define WM8983_INPPGAZCR_SHIFT 7 /* INPPGAZCR */
+#define WM8983_INPPGAZCR_WIDTH 1 /* INPPGAZCR */
+#define WM8983_INPPGAMUTER 0x0040 /* INPPGAMUTER */
+#define WM8983_INPPGAMUTER_MASK 0x0040 /* INPPGAMUTER */
+#define WM8983_INPPGAMUTER_SHIFT 6 /* INPPGAMUTER */
+#define WM8983_INPPGAMUTER_WIDTH 1 /* INPPGAMUTER */
+#define WM8983_INPPGAVOLR_MASK 0x003F /* INPPGAVOLR - [5:0] */
+#define WM8983_INPPGAVOLR_SHIFT 0 /* INPPGAVOLR - [5:0] */
+#define WM8983_INPPGAVOLR_WIDTH 6 /* INPPGAVOLR - [5:0] */
+
+/*
+ * R47 (0x2F) - Left ADC BOOST ctrl
+ */
+#define WM8983_PGABOOSTL 0x0100 /* PGABOOSTL */
+#define WM8983_PGABOOSTL_MASK 0x0100 /* PGABOOSTL */
+#define WM8983_PGABOOSTL_SHIFT 8 /* PGABOOSTL */
+#define WM8983_PGABOOSTL_WIDTH 1 /* PGABOOSTL */
+#define WM8983_L2_2BOOSTVOL_MASK 0x0070 /* L2_2BOOSTVOL - [6:4] */
+#define WM8983_L2_2BOOSTVOL_SHIFT 4 /* L2_2BOOSTVOL - [6:4] */
+#define WM8983_L2_2BOOSTVOL_WIDTH 3 /* L2_2BOOSTVOL - [6:4] */
+#define WM8983_AUXL2BOOSTVOL_MASK 0x0007 /* AUXL2BOOSTVOL - [2:0] */
+#define WM8983_AUXL2BOOSTVOL_SHIFT 0 /* AUXL2BOOSTVOL - [2:0] */
+#define WM8983_AUXL2BOOSTVOL_WIDTH 3 /* AUXL2BOOSTVOL - [2:0] */
+
+/*
+ * R48 (0x30) - Right ADC BOOST ctrl
+ */
+#define WM8983_PGABOOSTR 0x0100 /* PGABOOSTR */
+#define WM8983_PGABOOSTR_MASK 0x0100 /* PGABOOSTR */
+#define WM8983_PGABOOSTR_SHIFT 8 /* PGABOOSTR */
+#define WM8983_PGABOOSTR_WIDTH 1 /* PGABOOSTR */
+#define WM8983_R2_2BOOSTVOL_MASK 0x0070 /* R2_2BOOSTVOL - [6:4] */
+#define WM8983_R2_2BOOSTVOL_SHIFT 4 /* R2_2BOOSTVOL - [6:4] */
+#define WM8983_R2_2BOOSTVOL_WIDTH 3 /* R2_2BOOSTVOL - [6:4] */
+#define WM8983_AUXR2BOOSTVOL_MASK 0x0007 /* AUXR2BOOSTVOL - [2:0] */
+#define WM8983_AUXR2BOOSTVOL_SHIFT 0 /* AUXR2BOOSTVOL - [2:0] */
+#define WM8983_AUXR2BOOSTVOL_WIDTH 3 /* AUXR2BOOSTVOL - [2:0] */
+
+/*
+ * R49 (0x31) - Output ctrl
+ */
+#define WM8983_DACL2RMIX 0x0040 /* DACL2RMIX */
+#define WM8983_DACL2RMIX_MASK 0x0040 /* DACL2RMIX */
+#define WM8983_DACL2RMIX_SHIFT 6 /* DACL2RMIX */
+#define WM8983_DACL2RMIX_WIDTH 1 /* DACL2RMIX */
+#define WM8983_DACR2LMIX 0x0020 /* DACR2LMIX */
+#define WM8983_DACR2LMIX_MASK 0x0020 /* DACR2LMIX */
+#define WM8983_DACR2LMIX_SHIFT 5 /* DACR2LMIX */
+#define WM8983_DACR2LMIX_WIDTH 1 /* DACR2LMIX */
+#define WM8983_OUT4BOOST 0x0010 /* OUT4BOOST */
+#define WM8983_OUT4BOOST_MASK 0x0010 /* OUT4BOOST */
+#define WM8983_OUT4BOOST_SHIFT 4 /* OUT4BOOST */
+#define WM8983_OUT4BOOST_WIDTH 1 /* OUT4BOOST */
+#define WM8983_OUT3BOOST 0x0008 /* OUT3BOOST */
+#define WM8983_OUT3BOOST_MASK 0x0008 /* OUT3BOOST */
+#define WM8983_OUT3BOOST_SHIFT 3 /* OUT3BOOST */
+#define WM8983_OUT3BOOST_WIDTH 1 /* OUT3BOOST */
+#define WM8983_SPKBOOST 0x0004 /* SPKBOOST */
+#define WM8983_SPKBOOST_MASK 0x0004 /* SPKBOOST */
+#define WM8983_SPKBOOST_SHIFT 2 /* SPKBOOST */
+#define WM8983_SPKBOOST_WIDTH 1 /* SPKBOOST */
+#define WM8983_TSDEN 0x0002 /* TSDEN */
+#define WM8983_TSDEN_MASK 0x0002 /* TSDEN */
+#define WM8983_TSDEN_SHIFT 1 /* TSDEN */
+#define WM8983_TSDEN_WIDTH 1 /* TSDEN */
+#define WM8983_VROI 0x0001 /* VROI */
+#define WM8983_VROI_MASK 0x0001 /* VROI */
+#define WM8983_VROI_SHIFT 0 /* VROI */
+#define WM8983_VROI_WIDTH 1 /* VROI */
+
+/*
+ * R50 (0x32) - Left mixer ctrl
+ */
+#define WM8983_AUXLMIXVOL_MASK 0x01C0 /* AUXLMIXVOL - [8:6] */
+#define WM8983_AUXLMIXVOL_SHIFT 6 /* AUXLMIXVOL - [8:6] */
+#define WM8983_AUXLMIXVOL_WIDTH 3 /* AUXLMIXVOL - [8:6] */
+#define WM8983_AUXL2LMIX 0x0020 /* AUXL2LMIX */
+#define WM8983_AUXL2LMIX_MASK 0x0020 /* AUXL2LMIX */
+#define WM8983_AUXL2LMIX_SHIFT 5 /* AUXL2LMIX */
+#define WM8983_AUXL2LMIX_WIDTH 1 /* AUXL2LMIX */
+#define WM8983_BYPLMIXVOL_MASK 0x001C /* BYPLMIXVOL - [4:2] */
+#define WM8983_BYPLMIXVOL_SHIFT 2 /* BYPLMIXVOL - [4:2] */
+#define WM8983_BYPLMIXVOL_WIDTH 3 /* BYPLMIXVOL - [4:2] */
+#define WM8983_BYPL2LMIX 0x0002 /* BYPL2LMIX */
+#define WM8983_BYPL2LMIX_MASK 0x0002 /* BYPL2LMIX */
+#define WM8983_BYPL2LMIX_SHIFT 1 /* BYPL2LMIX */
+#define WM8983_BYPL2LMIX_WIDTH 1 /* BYPL2LMIX */
+#define WM8983_DACL2LMIX 0x0001 /* DACL2LMIX */
+#define WM8983_DACL2LMIX_MASK 0x0001 /* DACL2LMIX */
+#define WM8983_DACL2LMIX_SHIFT 0 /* DACL2LMIX */
+#define WM8983_DACL2LMIX_WIDTH 1 /* DACL2LMIX */
+
+/*
+ * R51 (0x33) - Right mixer ctrl
+ */
+#define WM8983_AUXRMIXVOL_MASK 0x01C0 /* AUXRMIXVOL - [8:6] */
+#define WM8983_AUXRMIXVOL_SHIFT 6 /* AUXRMIXVOL - [8:6] */
+#define WM8983_AUXRMIXVOL_WIDTH 3 /* AUXRMIXVOL - [8:6] */
+#define WM8983_AUXR2RMIX 0x0020 /* AUXR2RMIX */
+#define WM8983_AUXR2RMIX_MASK 0x0020 /* AUXR2RMIX */
+#define WM8983_AUXR2RMIX_SHIFT 5 /* AUXR2RMIX */
+#define WM8983_AUXR2RMIX_WIDTH 1 /* AUXR2RMIX */
+#define WM8983_BYPRMIXVOL_MASK 0x001C /* BYPRMIXVOL - [4:2] */
+#define WM8983_BYPRMIXVOL_SHIFT 2 /* BYPRMIXVOL - [4:2] */
+#define WM8983_BYPRMIXVOL_WIDTH 3 /* BYPRMIXVOL - [4:2] */
+#define WM8983_BYPR2RMIX 0x0002 /* BYPR2RMIX */
+#define WM8983_BYPR2RMIX_MASK 0x0002 /* BYPR2RMIX */
+#define WM8983_BYPR2RMIX_SHIFT 1 /* BYPR2RMIX */
+#define WM8983_BYPR2RMIX_WIDTH 1 /* BYPR2RMIX */
+#define WM8983_DACR2RMIX 0x0001 /* DACR2RMIX */
+#define WM8983_DACR2RMIX_MASK 0x0001 /* DACR2RMIX */
+#define WM8983_DACR2RMIX_SHIFT 0 /* DACR2RMIX */
+#define WM8983_DACR2RMIX_WIDTH 1 /* DACR2RMIX */
+
+/*
+ * R52 (0x34) - LOUT1 (HP) volume ctrl
+ */
+#define WM8983_OUT1VU 0x0100 /* OUT1VU */
+#define WM8983_OUT1VU_MASK 0x0100 /* OUT1VU */
+#define WM8983_OUT1VU_SHIFT 8 /* OUT1VU */
+#define WM8983_OUT1VU_WIDTH 1 /* OUT1VU */
+#define WM8983_LOUT1ZC 0x0080 /* LOUT1ZC */
+#define WM8983_LOUT1ZC_MASK 0x0080 /* LOUT1ZC */
+#define WM8983_LOUT1ZC_SHIFT 7 /* LOUT1ZC */
+#define WM8983_LOUT1ZC_WIDTH 1 /* LOUT1ZC */
+#define WM8983_LOUT1MUTE 0x0040 /* LOUT1MUTE */
+#define WM8983_LOUT1MUTE_MASK 0x0040 /* LOUT1MUTE */
+#define WM8983_LOUT1MUTE_SHIFT 6 /* LOUT1MUTE */
+#define WM8983_LOUT1MUTE_WIDTH 1 /* LOUT1MUTE */
+#define WM8983_LOUT1VOL_MASK 0x003F /* LOUT1VOL - [5:0] */
+#define WM8983_LOUT1VOL_SHIFT 0 /* LOUT1VOL - [5:0] */
+#define WM8983_LOUT1VOL_WIDTH 6 /* LOUT1VOL - [5:0] */
+
+/*
+ * R53 (0x35) - ROUT1 (HP) volume ctrl
+ */
+#define WM8983_OUT1VU 0x0100 /* OUT1VU */
+#define WM8983_OUT1VU_MASK 0x0100 /* OUT1VU */
+#define WM8983_OUT1VU_SHIFT 8 /* OUT1VU */
+#define WM8983_OUT1VU_WIDTH 1 /* OUT1VU */
+#define WM8983_ROUT1ZC 0x0080 /* ROUT1ZC */
+#define WM8983_ROUT1ZC_MASK 0x0080 /* ROUT1ZC */
+#define WM8983_ROUT1ZC_SHIFT 7 /* ROUT1ZC */
+#define WM8983_ROUT1ZC_WIDTH 1 /* ROUT1ZC */
+#define WM8983_ROUT1MUTE 0x0040 /* ROUT1MUTE */
+#define WM8983_ROUT1MUTE_MASK 0x0040 /* ROUT1MUTE */
+#define WM8983_ROUT1MUTE_SHIFT 6 /* ROUT1MUTE */
+#define WM8983_ROUT1MUTE_WIDTH 1 /* ROUT1MUTE */
+#define WM8983_ROUT1VOL_MASK 0x003F /* ROUT1VOL - [5:0] */
+#define WM8983_ROUT1VOL_SHIFT 0 /* ROUT1VOL - [5:0] */
+#define WM8983_ROUT1VOL_WIDTH 6 /* ROUT1VOL - [5:0] */
+
+/*
+ * R54 (0x36) - LOUT2 (SPK) volume ctrl
+ */
+#define WM8983_OUT2VU 0x0100 /* OUT2VU */
+#define WM8983_OUT2VU_MASK 0x0100 /* OUT2VU */
+#define WM8983_OUT2VU_SHIFT 8 /* OUT2VU */
+#define WM8983_OUT2VU_WIDTH 1 /* OUT2VU */
+#define WM8983_LOUT2ZC 0x0080 /* LOUT2ZC */
+#define WM8983_LOUT2ZC_MASK 0x0080 /* LOUT2ZC */
+#define WM8983_LOUT2ZC_SHIFT 7 /* LOUT2ZC */
+#define WM8983_LOUT2ZC_WIDTH 1 /* LOUT2ZC */
+#define WM8983_LOUT2MUTE 0x0040 /* LOUT2MUTE */
+#define WM8983_LOUT2MUTE_MASK 0x0040 /* LOUT2MUTE */
+#define WM8983_LOUT2MUTE_SHIFT 6 /* LOUT2MUTE */
+#define WM8983_LOUT2MUTE_WIDTH 1 /* LOUT2MUTE */
+#define WM8983_LOUT2VOL_MASK 0x003F /* LOUT2VOL - [5:0] */
+#define WM8983_LOUT2VOL_SHIFT 0 /* LOUT2VOL - [5:0] */
+#define WM8983_LOUT2VOL_WIDTH 6 /* LOUT2VOL - [5:0] */
+
+/*
+ * R55 (0x37) - ROUT2 (SPK) volume ctrl
+ */
+#define WM8983_OUT2VU 0x0100 /* OUT2VU */
+#define WM8983_OUT2VU_MASK 0x0100 /* OUT2VU */
+#define WM8983_OUT2VU_SHIFT 8 /* OUT2VU */
+#define WM8983_OUT2VU_WIDTH 1 /* OUT2VU */
+#define WM8983_ROUT2ZC 0x0080 /* ROUT2ZC */
+#define WM8983_ROUT2ZC_MASK 0x0080 /* ROUT2ZC */
+#define WM8983_ROUT2ZC_SHIFT 7 /* ROUT2ZC */
+#define WM8983_ROUT2ZC_WIDTH 1 /* ROUT2ZC */
+#define WM8983_ROUT2MUTE 0x0040 /* ROUT2MUTE */
+#define WM8983_ROUT2MUTE_MASK 0x0040 /* ROUT2MUTE */
+#define WM8983_ROUT2MUTE_SHIFT 6 /* ROUT2MUTE */
+#define WM8983_ROUT2MUTE_WIDTH 1 /* ROUT2MUTE */
+#define WM8983_ROUT2VOL_MASK 0x003F /* ROUT2VOL - [5:0] */
+#define WM8983_ROUT2VOL_SHIFT 0 /* ROUT2VOL - [5:0] */
+#define WM8983_ROUT2VOL_WIDTH 6 /* ROUT2VOL - [5:0] */
+
+/*
+ * R56 (0x38) - OUT3 mixer ctrl
+ */
+#define WM8983_OUT3MUTE 0x0040 /* OUT3MUTE */
+#define WM8983_OUT3MUTE_MASK 0x0040 /* OUT3MUTE */
+#define WM8983_OUT3MUTE_SHIFT 6 /* OUT3MUTE */
+#define WM8983_OUT3MUTE_WIDTH 1 /* OUT3MUTE */
+#define WM8983_OUT4_2OUT3 0x0008 /* OUT4_2OUT3 */
+#define WM8983_OUT4_2OUT3_MASK 0x0008 /* OUT4_2OUT3 */
+#define WM8983_OUT4_2OUT3_SHIFT 3 /* OUT4_2OUT3 */
+#define WM8983_OUT4_2OUT3_WIDTH 1 /* OUT4_2OUT3 */
+#define WM8983_BYPL2OUT3 0x0004 /* BYPL2OUT3 */
+#define WM8983_BYPL2OUT3_MASK 0x0004 /* BYPL2OUT3 */
+#define WM8983_BYPL2OUT3_SHIFT 2 /* BYPL2OUT3 */
+#define WM8983_BYPL2OUT3_WIDTH 1 /* BYPL2OUT3 */
+#define WM8983_LMIX2OUT3 0x0002 /* LMIX2OUT3 */
+#define WM8983_LMIX2OUT3_MASK 0x0002 /* LMIX2OUT3 */
+#define WM8983_LMIX2OUT3_SHIFT 1 /* LMIX2OUT3 */
+#define WM8983_LMIX2OUT3_WIDTH 1 /* LMIX2OUT3 */
+#define WM8983_LDAC2OUT3 0x0001 /* LDAC2OUT3 */
+#define WM8983_LDAC2OUT3_MASK 0x0001 /* LDAC2OUT3 */
+#define WM8983_LDAC2OUT3_SHIFT 0 /* LDAC2OUT3 */
+#define WM8983_LDAC2OUT3_WIDTH 1 /* LDAC2OUT3 */
+
+/*
+ * R57 (0x39) - OUT4 (MONO) mix ctrl
+ */
+#define WM8983_OUT3_2OUT4 0x0080 /* OUT3_2OUT4 */
+#define WM8983_OUT3_2OUT4_MASK 0x0080 /* OUT3_2OUT4 */
+#define WM8983_OUT3_2OUT4_SHIFT 7 /* OUT3_2OUT4 */
+#define WM8983_OUT3_2OUT4_WIDTH 1 /* OUT3_2OUT4 */
+#define WM8983_OUT4MUTE 0x0040 /* OUT4MUTE */
+#define WM8983_OUT4MUTE_MASK 0x0040 /* OUT4MUTE */
+#define WM8983_OUT4MUTE_SHIFT 6 /* OUT4MUTE */
+#define WM8983_OUT4MUTE_WIDTH 1 /* OUT4MUTE */
+#define WM8983_OUT4ATTN 0x0020 /* OUT4ATTN */
+#define WM8983_OUT4ATTN_MASK 0x0020 /* OUT4ATTN */
+#define WM8983_OUT4ATTN_SHIFT 5 /* OUT4ATTN */
+#define WM8983_OUT4ATTN_WIDTH 1 /* OUT4ATTN */
+#define WM8983_LMIX2OUT4 0x0010 /* LMIX2OUT4 */
+#define WM8983_LMIX2OUT4_MASK 0x0010 /* LMIX2OUT4 */
+#define WM8983_LMIX2OUT4_SHIFT 4 /* LMIX2OUT4 */
+#define WM8983_LMIX2OUT4_WIDTH 1 /* LMIX2OUT4 */
+#define WM8983_LDAC2OUT4 0x0008 /* LDAC2OUT4 */
+#define WM8983_LDAC2OUT4_MASK 0x0008 /* LDAC2OUT4 */
+#define WM8983_LDAC2OUT4_SHIFT 3 /* LDAC2OUT4 */
+#define WM8983_LDAC2OUT4_WIDTH 1 /* LDAC2OUT4 */
+#define WM8983_BYPR2OUT4 0x0004 /* BYPR2OUT4 */
+#define WM8983_BYPR2OUT4_MASK 0x0004 /* BYPR2OUT4 */
+#define WM8983_BYPR2OUT4_SHIFT 2 /* BYPR2OUT4 */
+#define WM8983_BYPR2OUT4_WIDTH 1 /* BYPR2OUT4 */
+#define WM8983_RMIX2OUT4 0x0002 /* RMIX2OUT4 */
+#define WM8983_RMIX2OUT4_MASK 0x0002 /* RMIX2OUT4 */
+#define WM8983_RMIX2OUT4_SHIFT 1 /* RMIX2OUT4 */
+#define WM8983_RMIX2OUT4_WIDTH 1 /* RMIX2OUT4 */
+#define WM8983_RDAC2OUT4 0x0001 /* RDAC2OUT4 */
+#define WM8983_RDAC2OUT4_MASK 0x0001 /* RDAC2OUT4 */
+#define WM8983_RDAC2OUT4_SHIFT 0 /* RDAC2OUT4 */
+#define WM8983_RDAC2OUT4_WIDTH 1 /* RDAC2OUT4 */
+
+/*
+ * R61 (0x3D) - BIAS CTRL
+ */
+#define WM8983_BIASCUT 0x0100 /* BIASCUT */
+#define WM8983_BIASCUT_MASK 0x0100 /* BIASCUT */
+#define WM8983_BIASCUT_SHIFT 8 /* BIASCUT */
+#define WM8983_BIASCUT_WIDTH 1 /* BIASCUT */
+#define WM8983_HALFIPBIAS 0x0080 /* HALFIPBIAS */
+#define WM8983_HALFIPBIAS_MASK 0x0080 /* HALFIPBIAS */
+#define WM8983_HALFIPBIAS_SHIFT 7 /* HALFIPBIAS */
+#define WM8983_HALFIPBIAS_WIDTH 1 /* HALFIPBIAS */
+#define WM8983_VBBIASTST_MASK 0x0060 /* VBBIASTST - [6:5] */
+#define WM8983_VBBIASTST_SHIFT 5 /* VBBIASTST - [6:5] */
+#define WM8983_VBBIASTST_WIDTH 2 /* VBBIASTST - [6:5] */
+#define WM8983_BUFBIAS_MASK 0x0018 /* BUFBIAS - [4:3] */
+#define WM8983_BUFBIAS_SHIFT 3 /* BUFBIAS - [4:3] */
+#define WM8983_BUFBIAS_WIDTH 2 /* BUFBIAS - [4:3] */
+#define WM8983_ADCBIAS_MASK 0x0006 /* ADCBIAS - [2:1] */
+#define WM8983_ADCBIAS_SHIFT 1 /* ADCBIAS - [2:1] */
+#define WM8983_ADCBIAS_WIDTH 2 /* ADCBIAS - [2:1] */
+#define WM8983_HALFOPBIAS 0x0001 /* HALFOPBIAS */
+#define WM8983_HALFOPBIAS_MASK 0x0001 /* HALFOPBIAS */
+#define WM8983_HALFOPBIAS_SHIFT 0 /* HALFOPBIAS */
+#define WM8983_HALFOPBIAS_WIDTH 1 /* HALFOPBIAS */
+
+enum clk_src {
+ WM8983_CLKSRC_MCLK,
+ WM8983_CLKSRC_PLL
+};
+
+#endif /* _WM8983_H */
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 9e5ff78..6e85b88 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -876,7 +876,7 @@
left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-
+SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
};
static const struct snd_soc_dapm_route routes[] = {
@@ -1434,6 +1434,7 @@
wm8993->hubs_data.hp_startup_mode = 1;
wm8993->hubs_data.dcs_codes = -2;
+ wm8993->hubs_data.series_startup = 1;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
if (ret != 0) {
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 83014a7..09e680a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -195,10 +195,6 @@
aif + 1, rate);
}
- if (rate && rate < 3000000)
- dev_warn(codec->dev, "AIF%dCLK is %dHz, should be >=3MHz for optimal performance\n",
- aif + 1, rate);
-
wm8994->aifclk[aif] = rate;
snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset,
@@ -1146,13 +1142,33 @@
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+
+SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
+ left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer),
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
+ right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer),
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
};
static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = {
SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0)
+SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
+ left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
+SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
+ right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
};
static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
@@ -1282,14 +1298,6 @@
SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0),
SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0),
-SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
-SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
-
-SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
- left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
-SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
- right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-
SND_SOC_DAPM_POST("Debug log", post_ev),
};
@@ -1624,6 +1632,7 @@
int reg_offset, ret;
struct fll_div fll;
u16 reg, aif1, aif2;
+ unsigned long timeout;
aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
& WM8994_AIF1CLK_ENA;
@@ -1705,6 +1714,9 @@
(fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) |
(src - 1));
+ /* Clear any pending completion from a previous failure */
+ try_wait_for_completion(&wm8994->fll_locked[id]);
+
/* Enable (with fractional mode if required) */
if (freq_out) {
if (fll.k)
@@ -1715,7 +1727,15 @@
WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
reg);
- msleep(5);
+ if (wm8994->fll_locked_irq) {
+ timeout = wait_for_completion_timeout(&wm8994->fll_locked[id],
+ msecs_to_jiffies(10));
+ if (timeout == 0)
+ dev_warn(codec->dev,
+ "Timed out waiting for FLL lock\n");
+ } else {
+ msleep(5);
+ }
}
wm8994->fll[id].in = freq_in;
@@ -1733,6 +1753,14 @@
return 0;
}
+static irqreturn_t wm8994_fll_locked_irq(int irq, void *data)
+{
+ struct completion *completion = data;
+
+ complete(completion);
+
+ return IRQ_HANDLED;
+}
static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 };
@@ -2272,6 +2300,33 @@
return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1);
}
+static void wm8994_aif_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int rate_reg = 0;
+
+ switch (dai->id) {
+ case 1:
+ rate_reg = WM8994_AIF1_RATE;
+ break;
+ case 2:
+ rate_reg = WM8994_AIF1_RATE;
+ break;
+ default:
+ break;
+ }
+
+ /* If the DAI is idle then configure the divider tree for the
+ * lowest output rate to save a little power if the clock is
+ * still active (eg, because it is system clock).
+ */
+ if (rate_reg && !dai->playback_active && !dai->capture_active)
+ snd_soc_update_bits(codec, rate_reg,
+ WM8994_AIF1_SR_MASK |
+ WM8994_AIF1CLK_RATE_MASK, 0x9);
+}
+
static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -2338,6 +2393,7 @@
.set_sysclk = wm8994_set_dai_sysclk,
.set_fmt = wm8994_set_dai_fmt,
.hw_params = wm8994_hw_params,
+ .shutdown = wm8994_aif_shutdown,
.digital_mute = wm8994_aif_mute,
.set_pll = wm8994_set_fll,
.set_tristate = wm8994_set_tristate,
@@ -2347,6 +2403,7 @@
.set_sysclk = wm8994_set_dai_sysclk,
.set_fmt = wm8994_set_dai_fmt,
.hw_params = wm8994_hw_params,
+ .shutdown = wm8994_aif_shutdown,
.digital_mute = wm8994_aif_mute,
.set_pll = wm8994_set_fll,
.set_tristate = wm8994_set_tristate,
@@ -2850,6 +2907,15 @@
return IRQ_HANDLED;
}
+static irqreturn_t wm8994_fifo_error(int irq, void *data)
+{
+ struct snd_soc_codec *codec = data;
+
+ dev_err(codec->dev, "FIFO error\n");
+
+ return IRQ_HANDLED;
+}
+
static int wm8994_codec_probe(struct snd_soc_codec *codec)
{
struct wm8994 *control;
@@ -2868,6 +2934,9 @@
wm8994->pdata = dev_get_platdata(codec->dev->parent);
wm8994->codec = codec;
+ for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
+ init_completion(&wm8994->fll_locked[i]);
+
if (wm8994->pdata && wm8994->pdata->micdet_irq)
wm8994->micdet_irq = wm8994->pdata->micdet_irq;
else if (wm8994->pdata && wm8994->pdata->irq_base)
@@ -2906,6 +2975,7 @@
wm8994->hubs.dcs_codes = -5;
wm8994->hubs.hp_startup_mode = 1;
wm8994->hubs.dcs_readback_mode = 1;
+ wm8994->hubs.series_startup = 1;
break;
default:
wm8994->hubs.dcs_readback_mode = 1;
@@ -2920,6 +2990,15 @@
break;
}
+ wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR,
+ wm8994_fifo_error, "FIFO error", codec);
+
+ ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
+ wm_hubs_dcs_done, "DC servo done",
+ &wm8994->hubs);
+ if (ret == 0)
+ wm8994->hubs.dcs_done_irq = true;
+
switch (control->type) {
case WM8994:
if (wm8994->micdet_irq) {
@@ -2976,6 +3055,16 @@
}
}
+ wm8994->fll_locked_irq = true;
+ for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) {
+ ret = wm8994_request_irq(codec->control_data,
+ WM8994_IRQ_FLL1_LOCK + i,
+ wm8994_fll_locked_irq, "FLL lock",
+ &wm8994->fll_locked[i]);
+ if (ret != 0)
+ wm8994->fll_locked_irq = false;
+ }
+
/* Remember if AIFnLRCLK is configured as a GPIO. This should be
* configured on init - if a system wants to do this dynamically
* at runtime we can deal with that then.
@@ -3051,10 +3140,18 @@
1 << WM8994_AIF2DAC_3D_GAIN_SHIFT,
1 << WM8994_AIF2DAC_3D_GAIN_SHIFT);
- /* Unconditionally enable AIF1 ADC TDM mode; it only affects
- * behaviour on idle TDM clock cycles. */
- snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1,
- WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM);
+ /* Unconditionally enable AIF1 ADC TDM mode on chips which can
+ * use this; it only affects behaviour on idle TDM clock
+ * cycles. */
+ switch (control->type) {
+ case WM8994:
+ case WM8958:
+ snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1,
+ WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM);
+ break;
+ default:
+ break;
+ }
wm8994_update_class_w(codec);
@@ -3153,6 +3250,12 @@
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
if (wm8994->micdet_irq)
free_irq(wm8994->micdet_irq, wm8994);
+ for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i,
+ &wm8994->fll_locked[i]);
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
+ &wm8994->hubs);
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec);
err:
kfree(wm8994);
return ret;
@@ -3162,11 +3265,20 @@
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = codec->control_data;
+ int i;
wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
pm_runtime_disable(codec->dev);
+ for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i,
+ &wm8994->fll_locked[i]);
+
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
+ &wm8994->hubs);
+ wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec);
+
switch (control->type) {
case WM8994:
if (wm8994->micdet_irq)
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 0a1db04..1ab2266 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -11,6 +11,7 @@
#include <sound/soc.h>
#include <linux/firmware.h>
+#include <linux/completion.h>
#include "wm_hubs.h"
@@ -79,6 +80,8 @@
int mclk[2];
int aifclk[2];
struct wm8994_fll_config fll[2], fll_suspend[2];
+ struct completion fll_locked[2];
+ bool fll_locked_irq;
int dac_rates[2];
int lrclk_shared[2];
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 91c6b39..a469132 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -727,7 +727,7 @@
SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
SND_SOC_DAPM_OUTPUT("SPKN"),
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 9e370d1..4cc2d56 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -63,8 +63,10 @@
static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
{
+ struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
unsigned int reg;
int count = 0;
+ int timeout;
unsigned int val;
val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1;
@@ -74,18 +76,39 @@
dev_dbg(codec->dev, "Waiting for DC servo...\n");
+ if (hubs->dcs_done_irq)
+ timeout = 4;
+ else
+ timeout = 400;
+
do {
count++;
- msleep(1);
+
+ if (hubs->dcs_done_irq)
+ wait_for_completion_timeout(&hubs->dcs_done,
+ msecs_to_jiffies(250));
+ else
+ msleep(1);
+
reg = snd_soc_read(codec, WM8993_DC_SERVO_0);
dev_dbg(codec->dev, "DC servo: %x\n", reg);
- } while (reg & op && count < 400);
+ } while (reg & op && count < timeout);
if (reg & op)
dev_err(codec->dev, "Timed out waiting for DC Servo %x\n",
op);
}
+irqreturn_t wm_hubs_dcs_done(int irq, void *data)
+{
+ struct wm_hubs_data *hubs = data;
+
+ complete(&hubs->dcs_done);
+
+ return IRQ_HANDLED;
+}
+EXPORT_SYMBOL_GPL(wm_hubs_dcs_done);
+
/*
* Startup calibration of the DC servo
*/
@@ -107,8 +130,7 @@
return;
}
- /* Devices not using a DCS code correction have startup mode */
- if (hubs->dcs_codes) {
+ if (hubs->series_startup) {
/* Set for 32 series updates */
snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
WM8993_DCS_SERIES_NO_01_MASK,
@@ -134,9 +156,9 @@
break;
case 1:
reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
- reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+ reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
- reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+ reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
break;
default:
WARN(1, "Unknown DCS readback method\n");
@@ -150,13 +172,13 @@
dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
hubs->dcs_codes);
- /* HPOUT1L */
- offset = reg_l;
+ /* HPOUT1R */
+ offset = reg_r;
offset += hubs->dcs_codes;
dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
- /* HPOUT1R */
- offset = reg_r;
+ /* HPOUT1L */
+ offset = reg_l;
offset += hubs->dcs_codes;
dcs_cfg |= (u8)offset;
@@ -168,8 +190,8 @@
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
} else {
- dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
- dcs_cfg |= reg_r;
+ dcs_cfg = reg_r << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ dcs_cfg |= reg_l;
}
/* Save the callibrated offset if we're in class W mode and
@@ -195,7 +217,7 @@
/* If we're applying an offset correction then updating the
* callibration would be likely to introduce further offsets. */
- if (hubs->dcs_codes)
+ if (hubs->dcs_codes || hubs->no_series_update)
return ret;
/* Only need to do this if the outputs are active */
@@ -599,9 +621,6 @@
SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0,
in2r_pga, ARRAY_SIZE(in2r_pga)),
-/* Dummy widgets to represent differential paths */
-SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
-
SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0,
mixinl, ARRAY_SIZE(mixinl)),
SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0,
@@ -867,8 +886,11 @@
int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
int lineout1_diff, int lineout2_diff)
{
+ struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ init_completion(&hubs->dcs_done);
+
snd_soc_dapm_add_routes(dapm, analogue_routes,
ARRAY_SIZE(analogue_routes));
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index f8a5e97..676b125 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -14,6 +14,9 @@
#ifndef _WM_HUBS_H
#define _WM_HUBS_H
+#include <linux/completion.h>
+#include <linux/interrupt.h>
+
struct snd_soc_codec;
extern const unsigned int wm_hubs_spkmix_tlv[];
@@ -23,9 +26,14 @@
int dcs_codes;
int dcs_readback_mode;
int hp_startup_mode;
+ int series_startup;
+ int no_series_update;
bool class_w;
u16 class_w_dcs;
+
+ bool dcs_done_irq;
+ struct completion dcs_done;
};
extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
@@ -36,4 +44,6 @@
int jd_scthr, int jd_thr,
int micbias1_lvl, int micbias2_lvl);
+extern irqreturn_t wm_hubs_dcs_done(int irq, void *data);
+
#endif
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 9d35b8c..a49e667 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -46,11 +46,28 @@
}
#endif
+#define DAVINCI_PCM_FMTBITS (\
+ SNDRV_PCM_FMTBIT_S8 |\
+ SNDRV_PCM_FMTBIT_U8 |\
+ SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_U16_LE |\
+ SNDRV_PCM_FMTBIT_U16_BE |\
+ SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S24_BE |\
+ SNDRV_PCM_FMTBIT_U24_LE |\
+ SNDRV_PCM_FMTBIT_U24_BE |\
+ SNDRV_PCM_FMTBIT_S32_LE |\
+ SNDRV_PCM_FMTBIT_S32_BE |\
+ SNDRV_PCM_FMTBIT_U32_LE |\
+ SNDRV_PCM_FMTBIT_U32_BE)
+
static struct snd_pcm_hardware pcm_hardware_playback = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME|
+ SNDRV_PCM_INFO_BATCH),
+ .formats = DAVINCI_PCM_FMTBITS,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
@@ -59,7 +76,7 @@
.rate_min = 8000,
.rate_max = 96000,
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 384,
.buffer_bytes_max = 128 * 1024,
.period_bytes_min = 32,
.period_bytes_max = 8 * 1024,
@@ -71,8 +88,9 @@
static struct snd_pcm_hardware pcm_hardware_capture = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE),
- .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_BATCH),
+ .formats = DAVINCI_PCM_FMTBITS,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
@@ -81,7 +99,7 @@
.rate_min = 8000,
.rate_max = 96000,
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 384,
.buffer_bytes_max = 128 * 1024,
.period_bytes_min = 32,
.period_bytes_max = 8 * 1024,
@@ -139,6 +157,22 @@
struct edmacc_param ram_params;
};
+static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ prtd->period++;
+ if (unlikely(prtd->period >= runtime->periods))
+ prtd->period = 0;
+}
+
+static void davinci_pcm_period_reset(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+
+ prtd->period = 0;
+}
/*
* Not used with ping/pong
*/
@@ -199,10 +233,6 @@
else
edma_set_transfer_params(link, acnt, fifo_level, count,
fifo_level, ABSYNC);
-
- prtd->period++;
- if (unlikely(prtd->period >= runtime->periods))
- prtd->period = 0;
}
static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
@@ -217,12 +247,13 @@
return;
if (snd_pcm_running(substream)) {
+ spin_lock(&prtd->lock);
if (prtd->ram_channel < 0) {
/* No ping/pong must fix up link dma data*/
- spin_lock(&prtd->lock);
davinci_pcm_enqueue_dma(substream);
- spin_unlock(&prtd->lock);
}
+ davinci_pcm_period_elapsed(substream);
+ spin_unlock(&prtd->lock);
snd_pcm_period_elapsed(substream);
}
}
@@ -425,7 +456,8 @@
edma_read_slot(link, &prtd->asp_params);
prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN);
- prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f);
+ prtd->asp_params.opt |= TCCHEN |
+ EDMA_TCC(prtd->ram_channel & 0x3f);
edma_write_slot(link, &prtd->asp_params);
/* pong */
@@ -439,7 +471,7 @@
prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f));
/* interrupt after every pong completion */
prtd->asp_params.opt |= TCINTEN | TCCHEN |
- EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel));
+ EDMA_TCC(prtd->ram_channel & 0x3f);
edma_write_slot(link, &prtd->asp_params);
/* ram */
@@ -527,6 +559,13 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
+ edma_start(prtd->asp_channel);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ prtd->ram_channel >= 0) {
+ /* copy 1st iram buffer */
+ edma_start(prtd->ram_channel);
+ }
+ break;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
edma_resume(prtd->asp_channel);
@@ -550,6 +589,7 @@
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ davinci_pcm_period_reset(substream);
if (prtd->ram_channel >= 0) {
int ret = ping_pong_dma_setup(substream);
if (ret < 0)
@@ -565,21 +605,31 @@
print_buf_info(prtd->asp_link[0], "asp_link[0]");
print_buf_info(prtd->asp_link[1], "asp_link[1]");
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* copy 1st iram buffer */
- edma_start(prtd->ram_channel);
- }
- edma_start(prtd->asp_channel);
+ /*
+ * There is a phase offset of 2 periods between the position
+ * used by dma setup and the position reported in the pointer
+ * function.
+ *
+ * The phase offset, when not using ping-pong buffers, is due to
+ * the two consecutive calls to davinci_pcm_enqueue_dma() below.
+ *
+ * Whereas here, with ping-pong buffers, the phase is due to
+ * there being an entire buffer transfer complete before the
+ * first dma completion event triggers davinci_pcm_dma_irq().
+ */
+ davinci_pcm_period_elapsed(substream);
+ davinci_pcm_period_elapsed(substream);
+
return 0;
}
- prtd->period = 0;
davinci_pcm_enqueue_dma(substream);
+ davinci_pcm_period_elapsed(substream);
/* Copy self-linked parameter RAM entry into master channel */
edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
edma_write_slot(prtd->asp_channel, &prtd->asp_params);
davinci_pcm_enqueue_dma(substream);
- edma_start(prtd->asp_channel);
+ davinci_pcm_period_elapsed(substream);
return 0;
}
@@ -591,51 +641,23 @@
struct davinci_runtime_data *prtd = runtime->private_data;
unsigned int offset;
int asp_count;
- dma_addr_t asp_src, asp_dst;
+ unsigned int period_size = snd_pcm_lib_period_bytes(substream);
+ /*
+ * There is a phase offset of 2 periods between the position used by dma
+ * setup and the position reported in the pointer function. Either +2 in
+ * the dma setup or -2 here in the pointer function (with wrapping,
+ * both) accounts for this offset -- choose the latter since it makes
+ * the first-time setup clearer.
+ */
spin_lock(&prtd->lock);
- if (prtd->ram_channel >= 0) {
- int ram_count;
- int mod_ram;
- dma_addr_t ram_src, ram_dst;
- unsigned int period_size = snd_pcm_lib_period_bytes(substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* reading ram before asp should be safe
- * as long as the asp transfers less than a ping size
- * of bytes between the 2 reads
- */
- edma_get_position(prtd->ram_channel,
- &ram_src, &ram_dst);
- edma_get_position(prtd->asp_channel,
- &asp_src, &asp_dst);
- asp_count = asp_src - prtd->asp_params.src;
- ram_count = ram_src - prtd->ram_params.src;
- mod_ram = ram_count % period_size;
- mod_ram -= asp_count;
- if (mod_ram < 0)
- mod_ram += period_size;
- else if (mod_ram == 0) {
- if (snd_pcm_running(substream))
- mod_ram += period_size;
- }
- ram_count -= mod_ram;
- if (ram_count < 0)
- ram_count += period_size * runtime->periods;
- } else {
- edma_get_position(prtd->ram_channel,
- &ram_src, &ram_dst);
- ram_count = ram_dst - prtd->ram_params.dst;
- }
- asp_count = ram_count;
- } else {
- edma_get_position(prtd->asp_channel, &asp_src, &asp_dst);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- asp_count = asp_src - runtime->dma_addr;
- else
- asp_count = asp_dst - runtime->dma_addr;
- }
+ asp_count = prtd->period - 2;
spin_unlock(&prtd->lock);
+ if (asp_count < 0)
+ asp_count += runtime->periods;
+ asp_count *= period_size;
+
offset = bytes_to_frames(runtime, asp_count);
if (offset >= runtime->buffer_size)
offset = 0;
@@ -811,9 +833,11 @@
static u64 davinci_pcm_dmamask = 0xffffffff;
-static int davinci_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret;
if (!card->dev->dma_mask)
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index a07f99c..dd7ac53 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -283,9 +283,11 @@
static u64 ep93xx_pcm_dmamask = 0xffffffff;
-static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 6680c0b..732208c 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -294,9 +294,11 @@
* Regardless of where the memory is actually allocated, since the device can
* technically DMA to any 36-bit address, we do need to set the DMA mask to 36.
*/
-static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
static u64 fsl_dma_dmamask = DMA_BIT_MASK(36);
int ret;
@@ -939,7 +941,7 @@
iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
if (iprop)
- dma->ssi_fifo_depth = *iprop;
+ dma->ssi_fifo_depth = be32_to_cpup(iprop);
else
/* Older 8610 DTs didn't have the fifo-depth property */
dma->ssi_fifo_depth = 8;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 313e0cc..d48afea 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -678,7 +678,12 @@
kfree(ssi_private);
return ret;
}
- ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start);
+ ssi_private->ssi = of_iomap(np, 0);
+ if (!ssi_private->ssi) {
+ dev_err(&pdev->dev, "could not map device resources\n");
+ kfree(ssi_private);
+ return -ENOMEM;
+ }
ssi_private->ssi_phys = res.start;
ssi_private->irq = irq_of_parse_and_map(np, 0);
@@ -691,7 +696,7 @@
/* Determine the FIFO depth. */
iprop = of_get_property(np, "fsl,fifo-depth", NULL);
if (iprop)
- ssi_private->fifo_depth = *iprop;
+ ssi_private->fifo_depth = be32_to_cpup(iprop);
else
/* Older 8610 DTs didn't have the fifo-depth property */
ssi_private->fifo_depth = 8;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index fff695c..19ad0c1 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -299,10 +299,11 @@
};
static u64 psc_dma_dmamask = 0xffffffff;
-static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int psc_dma_new(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
size_t size = psc_dma_hardware.buffer_bytes_max;
int rc = 0;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index c16c6b2..a192979 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -233,7 +233,7 @@
if (!iprop)
return -1;
- return *iprop;
+ return be32_to_cpup(iprop);
}
/**
@@ -258,7 +258,7 @@
if (!iprop)
return -EINVAL;
- addr = *iprop;
+ addr = be32_to_cpup(iprop);
bus = get_parent_cell_index(np);
if (bus < 0)
@@ -305,7 +305,7 @@
return -EINVAL;
}
- *dma_channel_id = *iprop;
+ *dma_channel_id = be32_to_cpup(iprop);
*dma_id = get_parent_cell_index(dma_channel_np);
of_node_put(dma_channel_np);
@@ -379,7 +379,7 @@
ret = -EINVAL;
goto error;
}
- machine_data->ssi_id = *iprop;
+ machine_data->ssi_id = be32_to_cpup(iprop);
/* Get the serial format and clock direction. */
sprop = of_get_property(np, "fsl,mode", NULL);
@@ -405,7 +405,7 @@
ret = -EINVAL;
goto error;
}
- machine_data->clk_frequency = *iprop;
+ machine_data->clk_frequency = be32_to_cpup(iprop);
} else if (strcasecmp(sprop, "i2s-master") == 0) {
machine_data->dai_format = SND_SOC_DAIFMT_I2S;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 66e0b68..8fa4d5f 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -232,7 +232,7 @@
iprop = of_get_property(parent, "cell-index", NULL);
if (iprop)
- ret = *iprop;
+ ret = be32_to_cpup(iprop);
of_node_put(parent);
@@ -261,7 +261,7 @@
if (!iprop)
return -EINVAL;
- addr = *iprop;
+ addr = be32_to_cpup(iprop);
bus = get_parent_cell_index(np);
if (bus < 0)
@@ -308,7 +308,7 @@
return -EINVAL;
}
- *dma_channel_id = *iprop;
+ *dma_channel_id = be32_to_cpup(iprop);
*dma_id = get_parent_cell_index(dma_channel_np);
of_node_put(dma_channel_np);
@@ -379,7 +379,7 @@
ret = -EINVAL;
goto error;
}
- mdata->ssi_id = *iprop;
+ mdata->ssi_id = be32_to_cpup(iprop);
/* Get the serial format and clock direction. */
sprop = of_get_property(np, "fsl,mode", NULL);
@@ -405,7 +405,7 @@
ret = -EINVAL;
goto error;
}
- mdata->clk_frequency = *iprop;
+ mdata->clk_frequency = be32_to_cpup(iprop);
} else if (strcasecmp(sprop, "i2s-master") == 0) {
mdata->dai_format = SND_SOC_DAIFMT_I2S;
mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 413b78d..309c59e 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -238,12 +238,14 @@
static int ssi_irq = 0;
-static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret;
- ret = imx_pcm_new(card, dai, pcm);
+ ret = imx_pcm_new(rtd);
if (ret)
return ret;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 61fceb0..10a8e27 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -388,10 +388,11 @@
static u64 imx_pcm_dmamask = DMA_BIT_MASK(32);
-int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
-
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index dc8a875..0a84cec 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -225,8 +225,7 @@
struct imx_ssi *ssi);
int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma);
-int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm);
+int imx_pcm_new(struct snd_soc_pcm_runtime *rtd);
void imx_pcm_free(struct snd_pcm *pcm);
/*
diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c
index fb1483f..a7c9578 100644
--- a/sound/soc/jz4740/jz4740-pcm.c
+++ b/sound/soc/jz4740/jz4740-pcm.c
@@ -299,9 +299,11 @@
static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32);
-int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index e13c6ce..cd33de1 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -312,9 +312,11 @@
return 0;
}
-static int kirkwood_dma_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret;
if (!card->dev->dma_mask)
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index 5a946b4..3e78260 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -402,9 +402,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int retval = 0;
pr_debug("sst_pcm_new called\n");
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index dac6732..9c0edad 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -356,7 +356,7 @@
nuc900_audio->irq_num = platform_get_irq(pdev, 0);
if (!nuc900_audio->irq_num) {
ret = -EBUSY;
- goto out2;
+ goto out3;
}
nuc900_ac97_data = nuc900_audio;
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 8263f56..d589ef1 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -315,9 +315,12 @@
}
static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32);
-static int nuc900_dma_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
+
if (!card->dev->dma_mask)
card->dev->dma_mask = &nuc900_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 99054cf..fe83d0d 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -9,6 +9,9 @@
config SND_OMAP_SOC_MCPDM
tristate
+config SND_OMAP_SOC_HDMI
+ tristate
+
config SND_OMAP_SOC_N810
tristate "SoC Audio support for Nokia N810"
depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
@@ -100,6 +103,14 @@
Say Y if you want to add support for SoC audio on Texas Instruments
SDP4430.
+config SND_OMAP_SOC_OMAP4_HDMI
+ tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"
+ depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4
+ select SND_OMAP_SOC_HDMI
+ help
+ Say Y if you want to add support for SoC HDMI audio on Texas Instruments
+ OMAP4 chips
+
config SND_OMAP_SOC_OMAP3_PANDORA
tristate "SoC Audio support for OMAP3 Pandora"
depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 6c2c87e..59e2c8d 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -2,10 +2,12 @@
snd-soc-omap-objs := omap-pcm.o
snd-soc-omap-mcbsp-objs := omap-mcbsp.o
snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o
+snd-soc-omap-hdmi-objs := omap-hdmi.o
obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
+obj-$(CONFIG_SND_OMAP_SOC_HDMI) += snd-soc-omap-hdmi.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
@@ -21,6 +23,7 @@
snd-soc-omap3beagle-objs := omap3beagle.o
snd-soc-zoom2-objs := zoom2.o
snd-soc-igep0020-objs := igep0020.o
+snd-soc-omap4-hdmi-objs := omap4-hdmi-card.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
@@ -36,3 +39,4 @@
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP4_HDMI) += snd-soc-omap4-hdmi.o
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 462cbcb..b40095a 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -427,7 +427,8 @@
/* Board specific codec bias level control */
static int ams_delta_set_bias_level(struct snd_soc_card *card,
- enum snd_soc_bias_level level)
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
{
struct snd_soc_codec *codec = card->rtd->codec;
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
new file mode 100644
index 0000000..36c6eae
--- /dev/null
+++ b/sound/soc/omap/omap-hdmi.c
@@ -0,0 +1,158 @@
+/*
+ * omap-hdmi.c
+ *
+ * OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors.
+ * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Authors: Jorge Candelaria <jorge.candelaria@ti.com>
+ * Ricardo Neri <ricardo.neri@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/dma.h>
+#include "omap-pcm.h"
+#include "omap-hdmi.h"
+
+#define DRV_NAME "hdmi-audio-dai"
+
+static struct omap_pcm_dma_data omap_hdmi_dai_dma_params = {
+ .name = "HDMI playback",
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+};
+
+static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ int err;
+ /*
+ * Make sure that the period bytes are multiple of the DMA packet size.
+ * Largest packet size we use is 32 32-bit words = 128 bytes
+ */
+ err = snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int err = 0;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ omap_hdmi_dai_dma_params.packet_size = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ omap_hdmi_dai_dma_params.packet_size = 32;
+ break;
+ default:
+ err = -EINVAL;
+ }
+
+ omap_hdmi_dai_dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
+
+ snd_soc_dai_set_dma_data(dai, substream,
+ &omap_hdmi_dai_dma_params);
+
+ return err;
+}
+
+static struct snd_soc_dai_ops omap_hdmi_dai_ops = {
+ .startup = omap_hdmi_dai_startup,
+ .hw_params = omap_hdmi_dai_hw_params,
+};
+
+static struct snd_soc_dai_driver omap_hdmi_dai = {
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = OMAP_HDMI_RATES,
+ .formats = OMAP_HDMI_FORMATS,
+ },
+ .ops = &omap_hdmi_dai_ops,
+};
+
+static __devinit int omap_hdmi_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct resource *hdmi_rsrc;
+
+ hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!hdmi_rsrc) {
+ dev_err(&pdev->dev, "Cannot obtain IORESOURCE_MEM HDMI\n");
+ return -EINVAL;
+ }
+
+ omap_hdmi_dai_dma_params.port_addr = hdmi_rsrc->start
+ + OMAP_HDMI_AUDIO_DMA_PORT;
+
+ hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!hdmi_rsrc) {
+ dev_err(&pdev->dev, "Cannot obtain IORESOURCE_DMA HDMI\n");
+ return -EINVAL;
+ }
+
+ omap_hdmi_dai_dma_params.dma_req = hdmi_rsrc->start;
+
+ ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai);
+ return ret;
+}
+
+static int __devexit omap_hdmi_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver hdmi_dai_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+ .probe = omap_hdmi_probe,
+ .remove = __devexit_p(omap_hdmi_remove),
+};
+
+static int __init hdmi_dai_init(void)
+{
+ return platform_driver_register(&hdmi_dai_driver);
+}
+module_init(hdmi_dai_init);
+
+static void __exit hdmi_dai_exit(void)
+{
+ platform_driver_unregister(&hdmi_dai_driver);
+}
+module_exit(hdmi_dai_exit);
+
+MODULE_AUTHOR("Jorge Candelaria <jorge.candelaria@ti.com>");
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("OMAP HDMI SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/omap/omap-hdmi.h b/sound/soc/omap/omap-hdmi.h
new file mode 100644
index 0000000..34c298d
--- /dev/null
+++ b/sound/soc/omap/omap-hdmi.h
@@ -0,0 +1,36 @@
+/*
+ * omap-hdmi.h
+ *
+ * Definitions for OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors.
+ * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Authors: Jorge Candelaria <jorge.candelaria@ti.com>
+ * Ricardo Neri <ricardo.neri@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_HDMI_H__
+#define __OMAP_HDMI_H__
+
+#define OMAP_HDMI_AUDIO_DMA_PORT 0x8c
+
+#define OMAP_HDMI_RATES (SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+#define OMAP_HDMI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+#endif
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e6a6b99..b2f5751 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -366,9 +366,11 @@
}
}
-static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c
new file mode 100644
index 0000000..9f32615
--- /dev/null
+++ b/sound/soc/omap/omap4-hdmi-card.c
@@ -0,0 +1,129 @@
+/*
+ * omap4-hdmi-card.c
+ *
+ * OMAP ALSA SoC machine driver for TI OMAP4 HDMI
+ * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Ricardo Neri <ricardo.neri@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <video/omapdss.h>
+
+#define DRV_NAME "omap4-hdmi-audio"
+
+static int omap4_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ int i;
+ struct omap_overlay_manager *mgr = NULL;
+ struct device *dev = substream->pcm->card->dev;
+
+ /* Find DSS HDMI device */
+ for (i = 0; i < omap_dss_get_num_overlay_managers(); i++) {
+ mgr = omap_dss_get_overlay_manager(i);
+ if (mgr && mgr->device
+ && mgr->device->type == OMAP_DISPLAY_TYPE_HDMI)
+ break;
+ }
+
+ if (i == omap_dss_get_num_overlay_managers()) {
+ dev_err(dev, "HDMI display device not found!\n");
+ return -ENODEV;
+ }
+
+ /* Make sure HDMI is power-on to avoid L3 interconnect errors */
+ if (mgr->device->state != OMAP_DSS_DISPLAY_ACTIVE) {
+ dev_err(dev, "HDMI display is not active!\n");
+ return -EIO;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops omap4_hdmi_dai_ops = {
+ .hw_params = omap4_hdmi_dai_hw_params,
+};
+
+static struct snd_soc_dai_link omap4_hdmi_dai = {
+ .name = "HDMI",
+ .stream_name = "HDMI",
+ .cpu_dai_name = "hdmi-audio-dai",
+ .platform_name = "omap-pcm-audio",
+ .codec_name = "omapdss_hdmi",
+ .codec_dai_name = "hdmi-audio-codec",
+ .ops = &omap4_hdmi_dai_ops,
+};
+
+static struct snd_soc_card snd_soc_omap4_hdmi = {
+ .name = "OMAP4HDMI",
+ .dai_link = &omap4_hdmi_dai,
+ .num_links = 1,
+};
+
+static __devinit int omap4_hdmi_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &snd_soc_omap4_hdmi;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ card->dev = NULL;
+ return ret;
+ }
+ return 0;
+}
+
+static int __devexit omap4_hdmi_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ card->dev = NULL;
+ return 0;
+}
+
+static struct platform_driver omap4_hdmi_driver = {
+ .driver = {
+ .name = "omap4-hdmi-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = omap4_hdmi_probe,
+ .remove = __devexit_p(omap4_hdmi_remove),
+};
+
+static int __init omap4_hdmi_init(void)
+{
+ return platform_driver_register(&omap4_hdmi_driver);
+}
+module_init(omap4_hdmi_init);
+
+static void __exit omap4_hdmi_exit(void)
+{
+ platform_driver_unregister(&omap4_hdmi_driver);
+}
+module_exit(omap4_hdmi_exit);
+
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("OMAP4 HDMI machine ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index fab20a5..c430600 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -85,9 +85,10 @@
static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32);
-static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index ab3ccae..80c85fd 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -443,10 +443,11 @@
static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32);
-static int s6000_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime)
{
- struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+ struct snd_card *card = runtime->card->snd_card;
+ struct snd_soc_dai *dai = runtime->cpu_dai;
+ struct snd_pcm *pcm = runtime->pcm;
struct s6000_pcm_dma_params *params;
int res;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index d155cbb..54b0e4b 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -158,7 +158,7 @@
config SND_SOC_SAMSUNG_SMDK_SPDIF
tristate "SoC S/PDIF Audio support for SMDK"
- depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310)
select SND_SAMSUNG_SPDIF
help
Say Y if you want to add support for SoC S/PDIF audio on the SMDK.
@@ -171,9 +171,23 @@
help
Say Y if you want to add support for SoC audio on the SMDK.
+config SND_SOC_SMDK_WM8994_PCM
+ tristate "SoC PCM Audio support for WM8994 on SMDK"
+ depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310)
+ select SND_SOC_WM8994
+ select SND_SAMSUNG_PCM
+ help
+ Say Y if you want to add support for SoC audio on the SMDK
+
config SND_SOC_SPEYSIDE
tristate "Audio support for Wolfson Speyside"
depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
select SND_SAMSUNG_I2S
select SND_SOC_WM8915
select SND_SOC_WM9081
+
+config SND_SOC_SPEYSIDE_WM8962
+ tristate "Audio support for Wolfson Speyside with WM8962"
+ depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+ select SND_SAMSUNG_I2S
+ select SND_SOC_WM8962
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 683843a..9eb3b12 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -35,7 +35,9 @@
snd-soc-goni-wm8994-objs := goni_wm8994.o
snd-soc-smdk-spdif-objs := smdk_spdif.o
snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o
+snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o
snd-soc-speyside-objs := speyside.o
+snd-soc-speyside-wm8962-objs := speyside_wm8962.o
obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -54,4 +56,6 @@
obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o
obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o
obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o
+obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o
obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o
+obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 5cb3b88..9465588 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -425,9 +425,11 @@
static u64 dma_mask = DMA_BIT_MASK(32);
-static int dma_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
pr_debug("Entered %s\n", __func__);
diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h
new file mode 100644
index 0000000..c0e6d9a
--- /dev/null
+++ b/sound/soc/samsung/i2s-regs.h
@@ -0,0 +1,143 @@
+/*
+ * linux/sound/soc/samsung/i2s-regs.h
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd.
+ * http://www.samsung.com
+ *
+ * Samsung I2S driver's register header
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __SND_SOC_SAMSUNG_I2S_REGS_H
+#define __SND_SOC_SAMSUNG_I2S_REGS_H
+
+#define I2SCON 0x0
+#define I2SMOD 0x4
+#define I2SFIC 0x8
+#define I2SPSR 0xc
+#define I2STXD 0x10
+#define I2SRXD 0x14
+#define I2SFICS 0x18
+#define I2STXDS 0x1c
+#define I2SAHB 0x20
+#define I2SSTR0 0x24
+#define I2SSIZE 0x28
+#define I2STRNCNT 0x2c
+#define I2SLVL0ADDR 0x30
+#define I2SLVL1ADDR 0x34
+#define I2SLVL2ADDR 0x38
+#define I2SLVL3ADDR 0x3c
+
+#define CON_RSTCLR (1 << 31)
+#define CON_FRXOFSTATUS (1 << 26)
+#define CON_FRXORINTEN (1 << 25)
+#define CON_FTXSURSTAT (1 << 24)
+#define CON_FTXSURINTEN (1 << 23)
+#define CON_TXSDMA_PAUSE (1 << 20)
+#define CON_TXSDMA_ACTIVE (1 << 18)
+
+#define CON_FTXURSTATUS (1 << 17)
+#define CON_FTXURINTEN (1 << 16)
+#define CON_TXFIFO2_EMPTY (1 << 15)
+#define CON_TXFIFO1_EMPTY (1 << 14)
+#define CON_TXFIFO2_FULL (1 << 13)
+#define CON_TXFIFO1_FULL (1 << 12)
+
+#define CON_LRINDEX (1 << 11)
+#define CON_TXFIFO_EMPTY (1 << 10)
+#define CON_RXFIFO_EMPTY (1 << 9)
+#define CON_TXFIFO_FULL (1 << 8)
+#define CON_RXFIFO_FULL (1 << 7)
+#define CON_TXDMA_PAUSE (1 << 6)
+#define CON_RXDMA_PAUSE (1 << 5)
+#define CON_TXCH_PAUSE (1 << 4)
+#define CON_RXCH_PAUSE (1 << 3)
+#define CON_TXDMA_ACTIVE (1 << 2)
+#define CON_RXDMA_ACTIVE (1 << 1)
+#define CON_ACTIVE (1 << 0)
+
+#define MOD_OPCLK_CDCLK_OUT (0 << 30)
+#define MOD_OPCLK_CDCLK_IN (1 << 30)
+#define MOD_OPCLK_BCLK_OUT (2 << 30)
+#define MOD_OPCLK_PCLK (3 << 30)
+#define MOD_OPCLK_MASK (3 << 30)
+#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */
+
+#define MOD_BLCS_SHIFT 26
+#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT)
+#define MOD_BLCP_SHIFT 24
+#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT)
+
+#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */
+#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */
+#define MOD_C1DD_HHALF (1 << 19)
+#define MOD_C1DD_LHALF (1 << 18)
+#define MOD_DC2_EN (1 << 17)
+#define MOD_DC1_EN (1 << 16)
+#define MOD_BLC_16BIT (0 << 13)
+#define MOD_BLC_8BIT (1 << 13)
+#define MOD_BLC_24BIT (2 << 13)
+#define MOD_BLC_MASK (3 << 13)
+
+#define MOD_IMS_SYSMUX (1 << 10)
+#define MOD_SLAVE (1 << 11)
+#define MOD_TXONLY (0 << 8)
+#define MOD_RXONLY (1 << 8)
+#define MOD_TXRX (2 << 8)
+#define MOD_MASK (3 << 8)
+#define MOD_LR_LLOW (0 << 7)
+#define MOD_LR_RLOW (1 << 7)
+#define MOD_SDF_IIS (0 << 5)
+#define MOD_SDF_MSB (1 << 5)
+#define MOD_SDF_LSB (2 << 5)
+#define MOD_SDF_MASK (3 << 5)
+#define MOD_RCLK_256FS (0 << 3)
+#define MOD_RCLK_512FS (1 << 3)
+#define MOD_RCLK_384FS (2 << 3)
+#define MOD_RCLK_768FS (3 << 3)
+#define MOD_RCLK_MASK (3 << 3)
+#define MOD_BCLK_32FS (0 << 1)
+#define MOD_BCLK_48FS (1 << 1)
+#define MOD_BCLK_16FS (2 << 1)
+#define MOD_BCLK_24FS (3 << 1)
+#define MOD_BCLK_MASK (3 << 1)
+#define MOD_8BIT (1 << 0)
+
+#define MOD_CDCLKCON (1 << 12)
+
+#define PSR_PSREN (1 << 15)
+
+#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf)
+#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf)
+
+#define FIC_TXFLUSH (1 << 15)
+#define FIC_RXFLUSH (1 << 7)
+
+#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf)
+#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf)
+#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f)
+
+#define AHB_INTENLVL0 (1 << 24)
+#define AHB_LVL0INT (1 << 20)
+#define AHB_CLRLVL0INT (1 << 16)
+#define AHB_DMARLD (1 << 5)
+#define AHB_INTMASK (1 << 3)
+#define AHB_DMAEN (1 << 0)
+#define AHB_LVLINTMASK (0xf << 20)
+
+#define I2SSIZE_TRNMSK (0xffff)
+#define I2SSIZE_SHIFT (16)
+
+#endif /* __SND_SOC_SAMSUNG_I2S_REGS_H */
+
+
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 992a732..1568eea 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -22,109 +22,7 @@
#include "dma.h"
#include "i2s.h"
-
-#define I2SCON 0x0
-#define I2SMOD 0x4
-#define I2SFIC 0x8
-#define I2SPSR 0xc
-#define I2STXD 0x10
-#define I2SRXD 0x14
-#define I2SFICS 0x18
-#define I2STXDS 0x1c
-
-#define CON_RSTCLR (1 << 31)
-#define CON_FRXOFSTATUS (1 << 26)
-#define CON_FRXORINTEN (1 << 25)
-#define CON_FTXSURSTAT (1 << 24)
-#define CON_FTXSURINTEN (1 << 23)
-#define CON_TXSDMA_PAUSE (1 << 20)
-#define CON_TXSDMA_ACTIVE (1 << 18)
-
-#define CON_FTXURSTATUS (1 << 17)
-#define CON_FTXURINTEN (1 << 16)
-#define CON_TXFIFO2_EMPTY (1 << 15)
-#define CON_TXFIFO1_EMPTY (1 << 14)
-#define CON_TXFIFO2_FULL (1 << 13)
-#define CON_TXFIFO1_FULL (1 << 12)
-
-#define CON_LRINDEX (1 << 11)
-#define CON_TXFIFO_EMPTY (1 << 10)
-#define CON_RXFIFO_EMPTY (1 << 9)
-#define CON_TXFIFO_FULL (1 << 8)
-#define CON_RXFIFO_FULL (1 << 7)
-#define CON_TXDMA_PAUSE (1 << 6)
-#define CON_RXDMA_PAUSE (1 << 5)
-#define CON_TXCH_PAUSE (1 << 4)
-#define CON_RXCH_PAUSE (1 << 3)
-#define CON_TXDMA_ACTIVE (1 << 2)
-#define CON_RXDMA_ACTIVE (1 << 1)
-#define CON_ACTIVE (1 << 0)
-
-#define MOD_OPCLK_CDCLK_OUT (0 << 30)
-#define MOD_OPCLK_CDCLK_IN (1 << 30)
-#define MOD_OPCLK_BCLK_OUT (2 << 30)
-#define MOD_OPCLK_PCLK (3 << 30)
-#define MOD_OPCLK_MASK (3 << 30)
-#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */
-
-#define MOD_BLCS_SHIFT 26
-#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT)
-#define MOD_BLCP_SHIFT 24
-#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT)
-
-#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */
-#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */
-#define MOD_C1DD_HHALF (1 << 19)
-#define MOD_C1DD_LHALF (1 << 18)
-#define MOD_DC2_EN (1 << 17)
-#define MOD_DC1_EN (1 << 16)
-#define MOD_BLC_16BIT (0 << 13)
-#define MOD_BLC_8BIT (1 << 13)
-#define MOD_BLC_24BIT (2 << 13)
-#define MOD_BLC_MASK (3 << 13)
-
-#define MOD_IMS_SYSMUX (1 << 10)
-#define MOD_SLAVE (1 << 11)
-#define MOD_TXONLY (0 << 8)
-#define MOD_RXONLY (1 << 8)
-#define MOD_TXRX (2 << 8)
-#define MOD_MASK (3 << 8)
-#define MOD_LR_LLOW (0 << 7)
-#define MOD_LR_RLOW (1 << 7)
-#define MOD_SDF_IIS (0 << 5)
-#define MOD_SDF_MSB (1 << 5)
-#define MOD_SDF_LSB (2 << 5)
-#define MOD_SDF_MASK (3 << 5)
-#define MOD_RCLK_256FS (0 << 3)
-#define MOD_RCLK_512FS (1 << 3)
-#define MOD_RCLK_384FS (2 << 3)
-#define MOD_RCLK_768FS (3 << 3)
-#define MOD_RCLK_MASK (3 << 3)
-#define MOD_BCLK_32FS (0 << 1)
-#define MOD_BCLK_48FS (1 << 1)
-#define MOD_BCLK_16FS (2 << 1)
-#define MOD_BCLK_24FS (3 << 1)
-#define MOD_BCLK_MASK (3 << 1)
-#define MOD_8BIT (1 << 0)
-
-#define MOD_CDCLKCON (1 << 12)
-
-#define PSR_PSREN (1 << 15)
-
-#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf)
-#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf)
-
-#define FIC_TXFLUSH (1 << 15)
-#define FIC_RXFLUSH (1 << 7)
-#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf)
-#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf)
-#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f)
+#include "i2s-regs.h"
#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index e7c1009..45fbe2b 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -8,6 +8,7 @@
*/
#include "../codecs/wm8994.h"
+#include <sound/pcm_params.h>
/*
* Default CFG switch settings to use this driver:
@@ -44,7 +45,9 @@
int ret;
/* AIF1CLK should be >=3MHz for optimal performance */
- if (params_rate(params) == 8000 || params_rate(params) == 11025)
+ if (params_format(params) == SNDRV_PCM_FORMAT_S24_LE)
+ pll_out = params_rate(params) * 384;
+ else if (params_rate(params) == 8000 || params_rate(params) == 11025)
pll_out = params_rate(params) * 512;
else
pll_out = params_rate(params) * 256;
diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c
new file mode 100644
index 0000000..5f21116
--- /dev/null
+++ b/sound/soc/samsung/smdk_wm8994pcm.c
@@ -0,0 +1,176 @@
+/*
+ * sound/soc/samsung/smdk_wm8994pcm.c
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd
+ * http://www.samsung.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/wm8994.h"
+#include "dma.h"
+#include "pcm.h"
+
+/*
+ * Board Settings:
+ * o '1' means 'ON'
+ * o '0' means 'OFF'
+ * o 'X' means 'Don't care'
+ *
+ * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111
+ */
+
+/*
+ * Configure audio route as :-
+ * $ amixer sset 'DAC1' on,on
+ * $ amixer sset 'Right Headphone Mux' 'DAC'
+ * $ amixer sset 'Left Headphone Mux' 'DAC'
+ * $ amixer sset 'DAC1R Mixer AIF1.1' on
+ * $ amixer sset 'DAC1L Mixer AIF1.1' on
+ * $ amixer sset 'IN2L' on
+ * $ amixer sset 'IN2L PGA IN2LN' on
+ * $ amixer sset 'MIXINL IN2L' on
+ * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on
+ * $ amixer sset 'IN2R' on
+ * $ amixer sset 'IN2R PGA IN2RN' on
+ * $ amixer sset 'MIXINR IN2R' on
+ * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on
+ */
+
+/* SMDK has a 16.9344MHZ crystal attached to WM8994 */
+#define SMDK_WM8994_FREQ 16934400
+
+static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned long mclk_freq;
+ int rfs, ret;
+
+ switch(params_rate(params)) {
+ case 8000:
+ rfs = 512;
+ break;
+ default:
+ dev_err(cpu_dai->dev, "%s:%d Sampling Rate %u not supported!\n",
+ __func__, __LINE__, params_rate(params));
+ return -EINVAL;
+ }
+
+ mclk_freq = params_rate(params) * rfs;
+
+ /* Set the codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B
+ | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* Set the cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B
+ | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
+ mclk_freq, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
+ SMDK_WM8994_FREQ, mclk_freq);
+ if (ret < 0)
+ return ret;
+
+ /* Set PCM source clock on CPU */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX,
+ mclk_freq, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* Set SCLK_DIV for making bclk */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops smdk_wm8994_pcm_ops = {
+ .hw_params = smdk_wm8994_pcm_hw_params,
+};
+
+static struct snd_soc_dai_link smdk_dai[] = {
+ {
+ .name = "WM8994 PAIF PCM",
+ .stream_name = "Primary PCM",
+ .cpu_dai_name = "samsung-pcm.0",
+ .codec_dai_name = "wm8994-aif1",
+ .platform_name = "samsung-audio",
+ .codec_name = "wm8994-codec",
+ .ops = &smdk_wm8994_pcm_ops,
+ },
+};
+
+static struct snd_soc_card smdk_pcm = {
+ .name = "SMDK-PCM",
+ .dai_link = smdk_dai,
+ .num_links = 1,
+};
+
+static int __devinit snd_smdk_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+
+ smdk_pcm.dev = &pdev->dev;
+ ret = snd_soc_register_card(&smdk_pcm);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit snd_smdk_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&smdk_pcm);
+ platform_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+static struct platform_driver snd_smdk_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "samsung-smdk-pcm",
+ },
+ .probe = snd_smdk_probe,
+ .remove = __devexit_p(snd_smdk_remove),
+};
+
+static int __init smdk_audio_init(void)
+{
+ return platform_driver_register(&snd_smdk_driver);
+}
+
+module_init(smdk_audio_init);
+
+static void __exit smdk_audio_exit(void)
+{
+ platform_driver_unregister(&snd_smdk_driver);
+}
+
+module_exit(smdk_audio_exit);
+
+MODULE_AUTHOR("Sangbeom Kim, <sbkim73@samsung.com>");
+MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 360a333..d6dee4d 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -20,24 +20,29 @@
#define WM8915_HPSEL_GPIO 214
static int speyside_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_STANDBY:
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK1,
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK2,
32768, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK1,
+ ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK2,
0, 0, 0);
if (ret < 0) {
pr_err("Failed to stop FLL\n");
return ret;
}
+ break;
default:
break;
@@ -46,6 +51,45 @@
return 0;
}
+static int speyside_set_bias_level_post(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+ ret = snd_soc_dai_set_pll(codec_dai, 0,
+ WM8915_FLL_MCLK2,
+ 32768, 48000 * 256);
+ if (ret < 0) {
+ pr_err("Failed to start FLL\n");
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8915_SYSCLK_FLL,
+ 48000 * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ card->dapm.bias_level = level;
+
+ return 0;
+}
+
static int speyside_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -66,16 +110,6 @@
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK1,
- 32768, 256 * 48000);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_FLL,
- 256 * 48000, SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
return 0;
}
@@ -127,7 +161,7 @@
struct snd_soc_codec *codec = rtd->codec;
int ret;
- ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0);
+ ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK2, 32768, 0);
if (ret < 0)
return ret;
@@ -267,6 +301,7 @@
.num_configs = ARRAY_SIZE(speyside_codec_conf),
.set_bias_level = speyside_set_bias_level,
+ .set_bias_level_post = speyside_set_bias_level_post,
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
new file mode 100644
index 0000000..8ac42bf
--- /dev/null
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -0,0 +1,264 @@
+/*
+ * Speyside with WM8962 audio support
+ *
+ * Copyright 2011 Wolfson Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <linux/gpio.h>
+
+#include "../codecs/wm8962.h"
+
+static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ WM8962_FLL_MCLK, 32768,
+ 44100 * 256);
+ if (ret < 0)
+ pr_err("Failed to start FLL: %d\n", ret);
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_FLL,
+ 44100 * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err("Failed to set SYSCLK: %d\n");
+ return ret;
+ }
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_STANDBY:
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ 32768, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err("Failed to switch away from FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ 0, 0, 0);
+ if (ret < 0) {
+ pr_err("Failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ dapm->bias_level = level;
+
+ return 0;
+}
+
+static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops speyside_wm8962_ops = {
+ .hw_params = speyside_wm8962_hw_params,
+};
+
+static struct snd_soc_dai_link speyside_wm8962_dai[] = {
+ {
+ .name = "CPU",
+ .stream_name = "CPU",
+ .cpu_dai_name = "samsung-i2s.0",
+ .codec_dai_name = "wm8962",
+ .platform_name = "samsung-audio",
+ .codec_name = "wm8962.1-001a",
+ .ops = &speyside_wm8962_ops,
+ },
+};
+
+static const struct snd_kcontrol_new controls[] = {
+ SOC_DAPM_PIN_SWITCH("Main Speaker"),
+};
+
+static struct snd_soc_dapm_widget widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+
+ SND_SOC_DAPM_SPK("Main Speaker", NULL),
+};
+
+static struct snd_soc_dapm_route audio_paths[] = {
+ { "Headphone", NULL, "HPOUTL" },
+ { "Headphone", NULL, "HPOUTR" },
+
+ { "Main Speaker", NULL, "SPKOUTL" },
+ { "Main Speaker", NULL, "SPKOUTR" },
+
+ { "MICBIAS", NULL, "Headset Mic" },
+ { "IN4L", NULL, "MICBIAS" },
+ { "IN4R", NULL, "MICBIAS" },
+
+ { "MICBIAS", NULL, "DMIC" },
+ { "DMICDAT", NULL, "MICBIAS" },
+};
+
+static struct snd_soc_jack speyside_wm8962_headset;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin speyside_wm8962_headset_pins[] = {
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static int speyside_wm8962_late_probe(struct snd_soc_card *card)
+{
+ struct snd_soc_codec *codec = card->rtd[0].codec;
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ 32768, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_jack_new(codec, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0,
+ &speyside_wm8962_headset);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_jack_add_pins(&speyside_wm8962_headset,
+ ARRAY_SIZE(speyside_wm8962_headset_pins),
+ speyside_wm8962_headset_pins);
+ if (ret)
+ return ret;
+
+ wm8962_mic_detect(codec, &speyside_wm8962_headset);
+
+ return 0;
+}
+
+static struct snd_soc_card speyside_wm8962 = {
+ .name = "Speyside WM8962",
+ .dai_link = speyside_wm8962_dai,
+ .num_links = ARRAY_SIZE(speyside_wm8962_dai),
+
+ .set_bias_level = speyside_wm8962_set_bias_level,
+ .set_bias_level_post = speyside_wm8962_set_bias_level_post,
+
+ .controls = controls,
+ .num_controls = ARRAY_SIZE(controls),
+ .dapm_widgets = widgets,
+ .num_dapm_widgets = ARRAY_SIZE(widgets),
+ .dapm_routes = audio_paths,
+ .num_dapm_routes = ARRAY_SIZE(audio_paths),
+
+ .late_probe = speyside_wm8962_late_probe,
+};
+
+static __devinit int speyside_wm8962_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &speyside_wm8962;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit speyside_wm8962_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver speyside_wm8962_driver = {
+ .driver = {
+ .name = "speyside-wm8962",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = speyside_wm8962_probe,
+ .remove = __devexit_p(speyside_wm8962_remove),
+};
+
+static int __init speyside_wm8962_audio_init(void)
+{
+ return platform_driver_register(&speyside_wm8962_driver);
+}
+module_init(speyside_wm8962_audio_init);
+
+static void __exit speyside_wm8962_audio_exit(void)
+{
+ platform_driver_unregister(&speyside_wm8962_driver);
+}
+module_exit(speyside_wm8962_audio_exit);
+
+MODULE_DESCRIPTION("Speyside WM8962 audio support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:speyside-wm8962");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index c326d29..db74005 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -327,10 +327,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int camelot_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_pcm *pcm = rtd->pcm;
+
/* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
* in MMAP mode (i.e. aplay -M)
*/
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 4a9da6b..8e112cc 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -118,10 +118,38 @@
/*
* FSI driver use below type name for variable
*
- * xxx_len : data length
- * xxx_width : data width
- * xxx_offset : data offset
* xxx_num : number of data
+ * xxx_pos : position of data
+ * xxx_capa : capacity of data
+ */
+
+/*
+ * period/frame/sample image
+ *
+ * ex) PCM (2ch)
+ *
+ * period pos period pos
+ * [n] [n + 1]
+ * |<-------------------- period--------------------->|
+ * ==|============================================ ... =|==
+ * | |
+ * ||<----- frame ----->|<------ frame ----->| ... |
+ * |+--------------------+--------------------+- ... |
+ * ||[ sample ][ sample ]|[ sample ][ sample ]| ... |
+ * |+--------------------+--------------------+- ... |
+ * ==|============================================ ... =|==
+ */
+
+/*
+ * FSI FIFO image
+ *
+ * | |
+ * | |
+ * | [ sample ] |
+ * | [ sample ] |
+ * | [ sample ] |
+ * | [ sample ] |
+ * --> go to codecs
*/
/*
@@ -131,12 +159,11 @@
struct fsi_stream {
struct snd_pcm_substream *substream;
- int fifo_max_num;
-
- int buff_offset;
- int buff_len;
- int period_len;
- int period_num;
+ int fifo_sample_capa; /* sample capacity of FSI FIFO */
+ int buff_sample_capa; /* sample capacity of ALSA buffer */
+ int buff_sample_pos; /* sample position of ALSA buffer */
+ int period_samples; /* sample number / 1 period */
+ int period_pos; /* current period position */
int uerr_num;
int oerr_num;
@@ -149,17 +176,14 @@
struct fsi_stream playback;
struct fsi_stream capture;
+ u32 do_fmt;
+ u32 di_fmt;
+
int chan_num:16;
int clk_master:1;
+ int spdif:1;
long rate;
-
- /* for suspend/resume */
- u32 saved_do_fmt;
- u32 saved_di_fmt;
- u32 saved_ckg1;
- u32 saved_ckg2;
- u32 saved_out_sel;
};
struct fsi_core {
@@ -180,14 +204,6 @@
struct fsi_core *core;
struct sh_fsi_platform_info *info;
spinlock_t lock;
-
- /* for suspend/resume */
- u32 saved_a_mclk;
- u32 saved_b_mclk;
- u32 saved_iemsk;
- u32 saved_imsk;
- u32 saved_clk_rst;
- u32 saved_soft_rst;
};
/*
@@ -271,6 +287,11 @@
return fsi->master->base == fsi->base;
}
+static int fsi_is_spdif(struct fsi_priv *fsi)
+{
+ return fsi->spdif;
+}
+
static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -342,28 +363,59 @@
return shift;
}
-static void fsi_stream_push(struct fsi_priv *fsi,
- int is_play,
- struct snd_pcm_substream *substream,
- u32 buffer_len,
- u32 period_len)
+static int fsi_frame2sample(struct fsi_priv *fsi, int frames)
+{
+ return frames * fsi->chan_num;
+}
+
+static int fsi_sample2frame(struct fsi_priv *fsi, int samples)
+{
+ return samples / fsi->chan_num;
+}
+
+static int fsi_stream_is_working(struct fsi_priv *fsi,
+ int is_play)
{
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+ struct fsi_master *master = fsi_get_master(fsi);
+ unsigned long flags;
+ int ret;
+ spin_lock_irqsave(&master->lock, flags);
+ ret = !!io->substream;
+ spin_unlock_irqrestore(&master->lock, flags);
+
+ return ret;
+}
+
+static void fsi_stream_push(struct fsi_priv *fsi,
+ int is_play,
+ struct snd_pcm_substream *substream)
+{
+ struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsi_master *master = fsi_get_master(fsi);
+ unsigned long flags;
+
+ spin_lock_irqsave(&master->lock, flags);
io->substream = substream;
- io->buff_len = buffer_len;
- io->buff_offset = 0;
- io->period_len = period_len;
- io->period_num = 0;
+ io->buff_sample_capa = fsi_frame2sample(fsi, runtime->buffer_size);
+ io->buff_sample_pos = 0;
+ io->period_samples = fsi_frame2sample(fsi, runtime->period_size);
+ io->period_pos = 0;
io->oerr_num = -1; /* ignore 1st err */
io->uerr_num = -1; /* ignore 1st err */
+ spin_unlock_irqrestore(&master->lock, flags);
}
static void fsi_stream_pop(struct fsi_priv *fsi, int is_play)
{
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
struct snd_soc_dai *dai = fsi_get_dai(io->substream);
+ struct fsi_master *master = fsi_get_master(fsi);
+ unsigned long flags;
+ spin_lock_irqsave(&master->lock, flags);
if (io->oerr_num > 0)
dev_err(dai->dev, "over_run = %d\n", io->oerr_num);
@@ -372,47 +424,27 @@
dev_err(dai->dev, "under_run = %d\n", io->uerr_num);
io->substream = NULL;
- io->buff_len = 0;
- io->buff_offset = 0;
- io->period_len = 0;
- io->period_num = 0;
+ io->buff_sample_capa = 0;
+ io->buff_sample_pos = 0;
+ io->period_samples = 0;
+ io->period_pos = 0;
io->oerr_num = 0;
io->uerr_num = 0;
+ spin_unlock_irqrestore(&master->lock, flags);
}
-static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
+static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, int is_play)
{
u32 status;
- int data_num;
+ int frames;
status = is_play ?
fsi_reg_read(fsi, DOFF_ST) :
fsi_reg_read(fsi, DIFF_ST);
- data_num = 0x1ff & (status >> 8);
- data_num *= fsi->chan_num;
+ frames = 0x1ff & (status >> 8);
- return data_num;
-}
-
-static int fsi_len2num(int len, int width)
-{
- return len / width;
-}
-
-#define fsi_num2offset(a, b) fsi_num2len(a, b)
-static int fsi_num2len(int num, int width)
-{
- return num * width;
-}
-
-static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play)
-{
- struct fsi_stream *io = fsi_get_stream(fsi, is_play);
- struct snd_pcm_substream *substream = io->substream;
- struct snd_pcm_runtime *runtime = substream->runtime;
-
- return frames_to_bytes(runtime, 1) / fsi->chan_num;
+ return fsi_frame2sample(fsi, frames);
}
static void fsi_count_fifo_err(struct fsi_priv *fsi)
@@ -444,8 +476,10 @@
{
int is_play = fsi_stream_is_play(stream);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
+ struct snd_pcm_runtime *runtime = io->substream->runtime;
- return io->substream->runtime->dma_area + io->buff_offset;
+ return runtime->dma_area +
+ samples_to_bytes(runtime, io->buff_sample_pos);
}
static void fsi_dma_soft_push16(struct fsi_priv *fsi, int num)
@@ -559,37 +593,94 @@
/*
* clock function
*/
-#define fsi_module_init(m, d) __fsi_module_clk_ctrl(m, d, 1)
-#define fsi_module_kill(m, d) __fsi_module_clk_ctrl(m, d, 0)
-static void __fsi_module_clk_ctrl(struct fsi_master *master,
- struct device *dev,
- int enable)
-{
- pm_runtime_get_sync(dev);
-
- if (enable) {
- /* enable only SR */
- fsi_master_mask_set(master, SOFT_RST, FSISR, FSISR);
- fsi_master_mask_set(master, SOFT_RST, PASR | PBSR, 0);
- } else {
- /* clear all registers */
- fsi_master_mask_set(master, SOFT_RST, FSISR, 0);
- }
-
- pm_runtime_put_sync(dev);
-}
-
-#define fsi_port_start(f) __fsi_port_clk_ctrl(f, 1)
-#define fsi_port_stop(f) __fsi_port_clk_ctrl(f, 0)
-static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int enable)
+static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi,
+ long rate, int enable)
{
struct fsi_master *master = fsi_get_master(fsi);
- u32 soft = fsi_is_port_a(fsi) ? PASR : PBSR;
- u32 clk = fsi_is_port_a(fsi) ? CRA : CRB;
- int is_master = fsi_is_clk_master(fsi);
+ set_rate_func set_rate = fsi_get_info_set_rate(master);
+ int fsi_ver = master->core->ver;
+ int ret;
- fsi_master_mask_set(master, SOFT_RST, soft, (enable) ? soft : 0);
- if (is_master)
+ ret = set_rate(dev, fsi_is_port_a(fsi), rate, enable);
+ if (ret < 0) /* error */
+ return ret;
+
+ if (!enable)
+ return 0;
+
+ if (ret > 0) {
+ u32 data = 0;
+
+ switch (ret & SH_FSI_ACKMD_MASK) {
+ default:
+ /* FALL THROUGH */
+ case SH_FSI_ACKMD_512:
+ data |= (0x0 << 12);
+ break;
+ case SH_FSI_ACKMD_256:
+ data |= (0x1 << 12);
+ break;
+ case SH_FSI_ACKMD_128:
+ data |= (0x2 << 12);
+ break;
+ case SH_FSI_ACKMD_64:
+ data |= (0x3 << 12);
+ break;
+ case SH_FSI_ACKMD_32:
+ if (fsi_ver < 2)
+ dev_err(dev, "unsupported ACKMD\n");
+ else
+ data |= (0x4 << 12);
+ break;
+ }
+
+ switch (ret & SH_FSI_BPFMD_MASK) {
+ default:
+ /* FALL THROUGH */
+ case SH_FSI_BPFMD_32:
+ data |= (0x0 << 8);
+ break;
+ case SH_FSI_BPFMD_64:
+ data |= (0x1 << 8);
+ break;
+ case SH_FSI_BPFMD_128:
+ data |= (0x2 << 8);
+ break;
+ case SH_FSI_BPFMD_256:
+ data |= (0x3 << 8);
+ break;
+ case SH_FSI_BPFMD_512:
+ data |= (0x4 << 8);
+ break;
+ case SH_FSI_BPFMD_16:
+ if (fsi_ver < 2)
+ dev_err(dev, "unsupported ACKMD\n");
+ else
+ data |= (0x7 << 8);
+ break;
+ }
+
+ fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
+ udelay(10);
+ ret = 0;
+ }
+
+ return ret;
+}
+
+#define fsi_port_start(f, i) __fsi_port_clk_ctrl(f, i, 1)
+#define fsi_port_stop(f, i) __fsi_port_clk_ctrl(f, i, 0)
+static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable)
+{
+ struct fsi_master *master = fsi_get_master(fsi);
+ u32 clk = fsi_is_port_a(fsi) ? CRA : CRB;
+
+ if (enable)
+ fsi_irq_enable(fsi, is_play);
+ else
+ fsi_irq_disable(fsi, is_play);
+
+ if (fsi_is_clk_master(fsi))
fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
}
@@ -598,18 +689,19 @@
*/
static void fsi_fifo_init(struct fsi_priv *fsi,
int is_play,
- struct snd_soc_dai *dai)
+ struct device *dev)
{
struct fsi_master *master = fsi_get_master(fsi);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
u32 shift, i;
+ int frame_capa;
/* get on-chip RAM capacity */
shift = fsi_master_read(master, FIFO_SZ);
shift >>= fsi_get_port_shift(fsi, is_play);
shift &= FIFO_SZ_MASK;
- io->fifo_max_num = 256 << shift;
- dev_dbg(dai->dev, "fifo = %d words\n", io->fifo_max_num);
+ frame_capa = 256 << shift;
+ dev_dbg(dev, "fifo = %d words\n", frame_capa);
/*
* The maximum number of sample data varies depending
@@ -631,9 +723,11 @@
* 8 channels: 32 ( 32 x 8 = 256)
*/
for (i = 1; i < fsi->chan_num; i <<= 1)
- io->fifo_max_num >>= 1;
- dev_dbg(dai->dev, "%d channel %d store\n",
- fsi->chan_num, io->fifo_max_num);
+ frame_capa >>= 1;
+ dev_dbg(dev, "%d channel %d store\n",
+ fsi->chan_num, frame_capa);
+
+ io->fifo_sample_capa = fsi_frame2sample(fsi, frame_capa);
/*
* set interrupt generation factor
@@ -654,10 +748,10 @@
struct snd_pcm_substream *substream = NULL;
int is_play = fsi_stream_is_play(stream);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
- int data_residue_num;
- int data_num;
- int data_num_max;
- int ch_width;
+ int sample_residues;
+ int sample_width;
+ int samples;
+ int samples_max;
int over_period;
void (*fn)(struct fsi_priv *fsi, int size);
@@ -673,36 +767,35 @@
/* FSI FIFO has limit.
* So, this driver can not send periods data at a time
*/
- if (io->buff_offset >=
- fsi_num2offset(io->period_num + 1, io->period_len)) {
+ if (io->buff_sample_pos >=
+ io->period_samples * (io->period_pos + 1)) {
over_period = 1;
- io->period_num = (io->period_num + 1) % runtime->periods;
+ io->period_pos = (io->period_pos + 1) % runtime->periods;
- if (0 == io->period_num)
- io->buff_offset = 0;
+ if (0 == io->period_pos)
+ io->buff_sample_pos = 0;
}
- /* get 1 channel data width */
- ch_width = fsi_get_frame_width(fsi, is_play);
+ /* get 1 sample data width */
+ sample_width = samples_to_bytes(runtime, 1);
- /* get residue data number of alsa */
- data_residue_num = fsi_len2num(io->buff_len - io->buff_offset,
- ch_width);
+ /* get number of residue samples */
+ sample_residues = io->buff_sample_capa - io->buff_sample_pos;
if (is_play) {
/*
* for play-back
*
- * data_num_max : number of FSI fifo free space
- * data_num : number of ALSA residue data
+ * samples_max : number of FSI fifo free samples space
+ * samples : number of ALSA residue samples
*/
- data_num_max = io->fifo_max_num * fsi->chan_num;
- data_num_max -= fsi_get_fifo_data_num(fsi, is_play);
+ samples_max = io->fifo_sample_capa;
+ samples_max -= fsi_get_current_fifo_samples(fsi, is_play);
- data_num = data_residue_num;
+ samples = sample_residues;
- switch (ch_width) {
+ switch (sample_width) {
case 2:
fn = fsi_dma_soft_push16;
break;
@@ -716,13 +809,13 @@
/*
* for capture
*
- * data_num_max : number of ALSA free space
- * data_num : number of data in FSI fifo
+ * samples_max : number of ALSA free samples space
+ * samples : number of samples in FSI fifo
*/
- data_num_max = data_residue_num;
- data_num = fsi_get_fifo_data_num(fsi, is_play);
+ samples_max = sample_residues;
+ samples = fsi_get_current_fifo_samples(fsi, is_play);
- switch (ch_width) {
+ switch (sample_width) {
case 2:
fn = fsi_dma_soft_pop16;
break;
@@ -734,12 +827,12 @@
}
}
- data_num = min(data_num, data_num_max);
+ samples = min(samples, samples_max);
- fn(fsi, data_num);
+ fn(fsi, samples);
- /* update buff_offset */
- io->buff_offset += fsi_num2offset(data_num, ch_width);
+ /* update buff_sample_pos */
+ io->buff_sample_pos += samples;
if (over_period)
snd_pcm_period_elapsed(substream);
@@ -788,16 +881,20 @@
* dai ops
*/
-static int fsi_dai_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+static int fsi_hw_startup(struct fsi_priv *fsi,
+ int is_play,
+ struct device *dev)
{
- struct fsi_priv *fsi = fsi_get_priv(substream);
u32 flags = fsi_get_info_flags(fsi);
- u32 data;
- int is_play = fsi_is_play(substream);
+ u32 data = 0;
- pm_runtime_get_sync(dai->dev);
+ pm_runtime_get_sync(dev);
+ /* clock setting */
+ if (fsi_is_clk_master(fsi))
+ data = DIMD | DOMD;
+
+ fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
/* clock inversion (CKG2) */
data = 0;
@@ -812,54 +909,70 @@
fsi_reg_write(fsi, CKG2, data);
+ /* set format */
+ fsi_reg_write(fsi, DO_FMT, fsi->do_fmt);
+ fsi_reg_write(fsi, DI_FMT, fsi->di_fmt);
+
+ /* spdif ? */
+ if (fsi_is_spdif(fsi)) {
+ fsi_spdif_clk_ctrl(fsi, 1);
+ fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+ }
+
/* irq clear */
fsi_irq_disable(fsi, is_play);
fsi_irq_clear_status(fsi);
/* fifo init */
- fsi_fifo_init(fsi, is_play, dai);
+ fsi_fifo_init(fsi, is_play, dev);
return 0;
}
+static void fsi_hw_shutdown(struct fsi_priv *fsi,
+ int is_play,
+ struct device *dev)
+{
+ if (fsi_is_clk_master(fsi))
+ fsi_set_master_clk(dev, fsi, fsi->rate, 0);
+
+ pm_runtime_put_sync(dev);
+}
+
+static int fsi_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get_priv(substream);
+ int is_play = fsi_is_play(substream);
+
+ return fsi_hw_startup(fsi, is_play, dai->dev);
+}
+
static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
int is_play = fsi_is_play(substream);
- struct fsi_master *master = fsi_get_master(fsi);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
- fsi_irq_disable(fsi, is_play);
-
- if (fsi_is_clk_master(fsi))
- set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
-
+ fsi_hw_shutdown(fsi, is_play, dai->dev);
fsi->rate = 0;
-
- pm_runtime_put_sync(dai->dev);
}
static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
int is_play = fsi_is_play(substream);
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- fsi_stream_push(fsi, is_play, substream,
- frames_to_bytes(runtime, runtime->buffer_size),
- frames_to_bytes(runtime, runtime->period_size));
+ fsi_stream_push(fsi, is_play, substream);
ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi);
- fsi_irq_enable(fsi, is_play);
- fsi_port_start(fsi);
+ fsi_port_start(fsi, is_play);
break;
case SNDRV_PCM_TRIGGER_STOP:
- fsi_port_stop(fsi);
- fsi_irq_disable(fsi, is_play);
+ fsi_port_stop(fsi, is_play);
fsi_stream_pop(fsi, is_play);
break;
}
@@ -884,8 +997,8 @@
return -EINVAL;
}
- fsi_reg_write(fsi, DO_FMT, data);
- fsi_reg_write(fsi, DI_FMT, data);
+ fsi->do_fmt = data;
+ fsi->di_fmt = data;
return 0;
}
@@ -900,11 +1013,10 @@
data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
fsi->chan_num = 2;
- fsi_spdif_clk_ctrl(fsi, 1);
- fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
+ fsi->spdif = 1;
- fsi_reg_write(fsi, DO_FMT, data);
- fsi_reg_write(fsi, DI_FMT, data);
+ fsi->do_fmt = data;
+ fsi->di_fmt = data;
return 0;
}
@@ -915,32 +1027,24 @@
struct fsi_master *master = fsi_get_master(fsi);
set_rate_func set_rate = fsi_get_info_set_rate(master);
u32 flags = fsi_get_info_flags(fsi);
- u32 data = 0;
int ret;
- pm_runtime_get_sync(dai->dev);
-
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- data = DIMD | DOMD;
fsi->clk_master = 1;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
- ret = -EINVAL;
- goto set_fmt_exit;
+ return -EINVAL;
}
if (fsi_is_clk_master(fsi) && !set_rate) {
dev_err(dai->dev, "platform doesn't have set_rate\n");
- ret = -EINVAL;
- goto set_fmt_exit;
+ return -EINVAL;
}
- fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data);
-
/* set format */
switch (flags & SH_FSI_FMT_MASK) {
case SH_FSI_FMT_DAI:
@@ -953,9 +1057,6 @@
ret = -EINVAL;
}
-set_fmt_exit:
- pm_runtime_put_sync(dai->dev);
-
return ret;
}
@@ -964,79 +1065,19 @@
struct snd_soc_dai *dai)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
- struct fsi_master *master = fsi_get_master(fsi);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
- int fsi_ver = master->core->ver;
long rate = params_rate(params);
int ret;
if (!fsi_is_clk_master(fsi))
return 0;
- ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
- if (ret < 0) /* error */
+ ret = fsi_set_master_clk(dai->dev, fsi, rate, 1);
+ if (ret < 0)
return ret;
fsi->rate = rate;
- if (ret > 0) {
- u32 data = 0;
-
- switch (ret & SH_FSI_ACKMD_MASK) {
- default:
- /* FALL THROUGH */
- case SH_FSI_ACKMD_512:
- data |= (0x0 << 12);
- break;
- case SH_FSI_ACKMD_256:
- data |= (0x1 << 12);
- break;
- case SH_FSI_ACKMD_128:
- data |= (0x2 << 12);
- break;
- case SH_FSI_ACKMD_64:
- data |= (0x3 << 12);
- break;
- case SH_FSI_ACKMD_32:
- if (fsi_ver < 2)
- dev_err(dai->dev, "unsupported ACKMD\n");
- else
- data |= (0x4 << 12);
- break;
- }
-
- switch (ret & SH_FSI_BPFMD_MASK) {
- default:
- /* FALL THROUGH */
- case SH_FSI_BPFMD_32:
- data |= (0x0 << 8);
- break;
- case SH_FSI_BPFMD_64:
- data |= (0x1 << 8);
- break;
- case SH_FSI_BPFMD_128:
- data |= (0x2 << 8);
- break;
- case SH_FSI_BPFMD_256:
- data |= (0x3 << 8);
- break;
- case SH_FSI_BPFMD_512:
- data |= (0x4 << 8);
- break;
- case SH_FSI_BPFMD_16:
- if (fsi_ver < 2)
- dev_err(dai->dev, "unsupported ACKMD\n");
- else
- data |= (0x7 << 8);
- break;
- }
-
- fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
- udelay(10);
- ret = 0;
- }
return ret;
-
}
static struct snd_soc_dai_ops fsi_dai_ops = {
@@ -1097,16 +1138,14 @@
static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
- long location;
+ int samples_pos = io->buff_sample_pos - 1;
- location = (io->buff_offset - 1);
- if (location < 0)
- location = 0;
+ if (samples_pos < 0)
+ samples_pos = 0;
- return bytes_to_frames(runtime, location);
+ return fsi_sample2frame(fsi, samples_pos);
}
static struct snd_pcm_ops fsi_pcm_ops = {
@@ -1129,10 +1168,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int fsi_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_pcm *pcm = rtd->pcm;
+
/*
* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
* in MMAP mode (i.e. aplay -M)
@@ -1246,8 +1285,6 @@
pm_runtime_enable(&pdev->dev);
dev_set_drvdata(&pdev->dev, master);
- fsi_module_init(master, &pdev->dev);
-
ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
id_entry->name, master);
if (ret) {
@@ -1290,8 +1327,6 @@
master = dev_get_drvdata(&pdev->dev);
- fsi_module_kill(master, &pdev->dev);
-
free_irq(master->irq, master);
pm_runtime_disable(&pdev->dev);
@@ -1305,53 +1340,43 @@
}
static void __fsi_suspend(struct fsi_priv *fsi,
- struct device *dev,
- set_rate_func set_rate)
+ int is_play,
+ struct device *dev)
{
- fsi->saved_do_fmt = fsi_reg_read(fsi, DO_FMT);
- fsi->saved_di_fmt = fsi_reg_read(fsi, DI_FMT);
- fsi->saved_ckg1 = fsi_reg_read(fsi, CKG1);
- fsi->saved_ckg2 = fsi_reg_read(fsi, CKG2);
- fsi->saved_out_sel = fsi_reg_read(fsi, OUT_SEL);
+ if (!fsi_stream_is_working(fsi, is_play))
+ return;
- if (fsi_is_clk_master(fsi))
- set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 0);
+ fsi_port_stop(fsi, is_play);
+ fsi_hw_shutdown(fsi, is_play, dev);
}
static void __fsi_resume(struct fsi_priv *fsi,
- struct device *dev,
- set_rate_func set_rate)
+ int is_play,
+ struct device *dev)
{
- fsi_reg_write(fsi, DO_FMT, fsi->saved_do_fmt);
- fsi_reg_write(fsi, DI_FMT, fsi->saved_di_fmt);
- fsi_reg_write(fsi, CKG1, fsi->saved_ckg1);
- fsi_reg_write(fsi, CKG2, fsi->saved_ckg2);
- fsi_reg_write(fsi, OUT_SEL, fsi->saved_out_sel);
+ if (!fsi_stream_is_working(fsi, is_play))
+ return;
- if (fsi_is_clk_master(fsi))
- set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 1);
+ fsi_hw_startup(fsi, is_play, dev);
+
+ if (fsi_is_clk_master(fsi) && fsi->rate)
+ fsi_set_master_clk(dev, fsi, fsi->rate, 1);
+
+ fsi_port_start(fsi, is_play);
+
}
static int fsi_suspend(struct device *dev)
{
struct fsi_master *master = dev_get_drvdata(dev);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
+ struct fsi_priv *fsia = &master->fsia;
+ struct fsi_priv *fsib = &master->fsib;
- pm_runtime_get_sync(dev);
+ __fsi_suspend(fsia, 1, dev);
+ __fsi_suspend(fsia, 0, dev);
- __fsi_suspend(&master->fsia, dev, set_rate);
- __fsi_suspend(&master->fsib, dev, set_rate);
-
- master->saved_a_mclk = fsi_core_read(master, a_mclk);
- master->saved_b_mclk = fsi_core_read(master, b_mclk);
- master->saved_iemsk = fsi_core_read(master, iemsk);
- master->saved_imsk = fsi_core_read(master, imsk);
- master->saved_clk_rst = fsi_master_read(master, CLK_RST);
- master->saved_soft_rst = fsi_master_read(master, SOFT_RST);
-
- fsi_module_kill(master, dev);
-
- pm_runtime_put_sync(dev);
+ __fsi_suspend(fsib, 1, dev);
+ __fsi_suspend(fsib, 0, dev);
return 0;
}
@@ -1359,23 +1384,14 @@
static int fsi_resume(struct device *dev)
{
struct fsi_master *master = dev_get_drvdata(dev);
- set_rate_func set_rate = fsi_get_info_set_rate(master);
+ struct fsi_priv *fsia = &master->fsia;
+ struct fsi_priv *fsib = &master->fsib;
- pm_runtime_get_sync(dev);
+ __fsi_resume(fsia, 1, dev);
+ __fsi_resume(fsia, 0, dev);
- fsi_module_init(master, dev);
-
- fsi_master_mask_set(master, SOFT_RST, 0xffff, master->saved_soft_rst);
- fsi_master_mask_set(master, CLK_RST, 0xffff, master->saved_clk_rst);
- fsi_core_mask_set(master, a_mclk, 0xffff, master->saved_a_mclk);
- fsi_core_mask_set(master, b_mclk, 0xffff, master->saved_b_mclk);
- fsi_core_mask_set(master, iemsk, 0xffff, master->saved_iemsk);
- fsi_core_mask_set(master, imsk, 0xffff, master->saved_imsk);
-
- __fsi_resume(&master->fsia, dev, set_rate);
- __fsi_resume(&master->fsib, dev, set_rate);
-
- pm_runtime_put_sync(dev);
+ __fsi_resume(fsib, 1, dev);
+ __fsi_resume(fsib, 0, dev);
return 0;
}
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index a423bab..f8f6816 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -527,10 +527,11 @@
return bytes_to_frames(ss->runtime, ptr);
}
-static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int siu_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
/* card->dev == socdev->dev, see snd_soc_new_pcms() */
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
struct siu_info *info = siu_i2s_data;
struct platform_device *pdev = to_platform_device(card->dev);
int ret;
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 039b953..d9f8ade 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -20,422 +20,6 @@
#include <trace/events/asoc.h>
-#ifdef CONFIG_SPI_MASTER
-static int do_spi_write(void *control, const char *data, int len)
-{
- struct spi_device *spi = control;
- int ret;
-
- ret = spi_write(spi, data, len);
- if (ret < 0)
- return ret;
-
- return len;
-}
-#endif
-
-static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value, const void *data, int len)
-{
- int ret;
-
- if (!snd_soc_codec_volatile_register(codec, reg) &&
- reg < codec->driver->reg_cache_size &&
- !codec->cache_bypass) {
- ret = snd_soc_cache_write(codec, reg, value);
- if (ret < 0)
- return -1;
- }
-
- if (codec->cache_only) {
- codec->cache_sync = 1;
- return 0;
- }
-
- ret = codec->hw_write(codec->control_data, data, len);
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-
-static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg)
-{
- int ret;
- unsigned int val;
-
- if (reg >= codec->driver->reg_cache_size ||
- snd_soc_codec_volatile_register(codec, reg) ||
- codec->cache_bypass) {
- if (codec->cache_only)
- return -1;
-
- BUG_ON(!codec->hw_read);
- return codec->hw_read(codec, reg);
- }
-
- ret = snd_soc_cache_read(codec, reg, &val);
- if (ret < 0)
- return -1;
- return val;
-}
-
-static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 data;
-
- data = cpu_to_be16((reg << 12) | (value & 0xffffff));
-
- return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- return do_hw_write(codec, reg, value, data, 2);
-}
-
-static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- reg &= 0xff;
- data[0] = reg;
- data[1] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 2);
-}
-
-static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
-
- data[0] = reg;
- data[1] = (value >> 8) & 0xff;
- data[2] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int do_i2c_read(struct snd_soc_codec *codec,
- void *reg, int reglen,
- void *data, int datalen)
-{
- struct i2c_msg xfer[2];
- int ret;
- struct i2c_client *client = codec->control_data;
-
- /* Write register */
- xfer[0].addr = client->addr;
- xfer[0].flags = 0;
- xfer[0].len = reglen;
- xfer[0].buf = reg;
-
- /* Read data */
- xfer[1].addr = client->addr;
- xfer[1].flags = I2C_M_RD;
- xfer[1].len = datalen;
- xfer[1].buf = data;
-
- ret = i2c_transfer(client->adapter, xfer, 2);
- if (ret == 2)
- return 0;
- else if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_8_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u8 reg = r;
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 1, &data, 2);
- if (ret < 0)
- return 0;
- return (data >> 8) | ((data & 0xff) << 8);
-}
-#else
-#define snd_soc_8_16_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = r;
- u8 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 1);
- if (ret < 0)
- return 0;
- return data;
-}
-#else
-#define snd_soc_16_8_read_i2c NULL
-#endif
-
-static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
-
- data[0] = (reg >> 8) & 0xff;
- data[1] = reg & 0xff;
- data[2] = value;
-
- return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
- unsigned int r)
-{
- u16 reg = cpu_to_be16(r);
- u16 data;
- int ret;
-
- ret = do_i2c_read(codec, ®, 2, &data, 2);
- if (ret < 0)
- return 0;
- return be16_to_cpu(data);
-}
-#else
-#define snd_soc_16_16_read_i2c NULL
-#endif
-
-static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return do_hw_read(codec, reg);
-}
-
-static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[4];
-
- data[0] = (reg >> 8) & 0xff;
- data[1] = reg & 0xff;
- data[2] = (value >> 8) & 0xff;
- data[3] = value & 0xff;
-
- return do_hw_write(codec, reg, value, data, 4);
-}
-
-/* Primitive bulk write support for soc-cache. The data pointed to by
- * `data' needs to already be in the form the hardware expects
- * including any leading register specific data. Any data written
- * through this function will not go through the cache as it only
- * handles writing to volatile or out of bounds registers.
- */
-static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
- const void *data, size_t len)
-{
- int ret;
-
- /* To ensure that we don't get out of sync with the cache, check
- * whether the base register is volatile or if we've directly asked
- * to bypass the cache. Out of bounds registers are considered
- * volatile.
- */
- if (!codec->cache_bypass
- && !snd_soc_codec_volatile_register(codec, reg)
- && reg < codec->driver->reg_cache_size)
- return -EINVAL;
-
- switch (codec->control_type) {
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- case SND_SOC_I2C:
- ret = i2c_master_send(codec->control_data, data, len);
- break;
-#endif
-#if defined(CONFIG_SPI_MASTER)
- case SND_SOC_SPI:
- ret = spi_write(codec->control_data, data, len);
- break;
-#endif
- default:
- BUG();
- }
-
- if (ret == len)
- return 0;
- if (ret < 0)
- return ret;
- else
- return -EIO;
-}
-
-static struct {
- int addr_bits;
- int data_bits;
- int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
- unsigned int (*read)(struct snd_soc_codec *, unsigned int);
- unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
-} io_types[] = {
- {
- .addr_bits = 4, .data_bits = 12,
- .write = snd_soc_4_12_write, .read = snd_soc_4_12_read,
- },
- {
- .addr_bits = 7, .data_bits = 9,
- .write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
- },
- {
- .addr_bits = 8, .data_bits = 8,
- .write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
- .i2c_read = snd_soc_8_8_read_i2c,
- },
- {
- .addr_bits = 8, .data_bits = 16,
- .write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
- .i2c_read = snd_soc_8_16_read_i2c,
- },
- {
- .addr_bits = 16, .data_bits = 8,
- .write = snd_soc_16_8_write, .read = snd_soc_16_8_read,
- .i2c_read = snd_soc_16_8_read_i2c,
- },
- {
- .addr_bits = 16, .data_bits = 16,
- .write = snd_soc_16_16_write, .read = snd_soc_16_16_read,
- .i2c_read = snd_soc_16_16_read_i2c,
- },
-};
-
-/**
- * snd_soc_codec_set_cache_io: Set up standard I/O functions.
- *
- * @codec: CODEC to configure.
- * @addr_bits: Number of bits of register address data.
- * @data_bits: Number of bits of data per register.
- * @control: Control bus used.
- *
- * Register formats are frequently shared between many I2C and SPI
- * devices. In order to promote code reuse the ASoC core provides
- * some standard implementations of CODEC read and write operations
- * which can be set up using this function.
- *
- * The caller is responsible for allocating and initialising the
- * actual cache.
- *
- * Note that at present this code cannot be used by CODECs with
- * volatile registers.
- */
-int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
- int addr_bits, int data_bits,
- enum snd_soc_control_type control)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(io_types); i++)
- if (io_types[i].addr_bits == addr_bits &&
- io_types[i].data_bits == data_bits)
- break;
- if (i == ARRAY_SIZE(io_types)) {
- printk(KERN_ERR
- "No I/O functions for %d bit address %d bit data\n",
- addr_bits, data_bits);
- return -EINVAL;
- }
-
- codec->write = io_types[i].write;
- codec->read = io_types[i].read;
- codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
-
- switch (control) {
- case SND_SOC_I2C:
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
- codec->hw_write = (hw_write_t)i2c_master_send;
-#endif
- if (io_types[i].i2c_read)
- codec->hw_read = io_types[i].i2c_read;
-
- codec->control_data = container_of(codec->dev,
- struct i2c_client,
- dev);
- break;
-
- case SND_SOC_SPI:
-#ifdef CONFIG_SPI_MASTER
- codec->hw_write = do_spi_write;
-#endif
-
- codec->control_data = container_of(codec->dev,
- struct spi_device,
- dev);
- break;
- }
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
-
static bool snd_soc_set_cache_val(void *base, unsigned int idx,
unsigned int val, unsigned int word_size)
{
@@ -483,31 +67,86 @@
}
struct snd_soc_rbtree_node {
- struct rb_node node;
- unsigned int reg;
- unsigned int value;
- unsigned int defval;
+ struct rb_node node; /* the actual rbtree node holding this block */
+ unsigned int base_reg; /* base register handled by this block */
+ unsigned int word_size; /* number of bytes needed to represent the register index */
+ void *block; /* block of adjacent registers */
+ unsigned int blklen; /* number of registers available in the block */
} __attribute__ ((packed));
struct snd_soc_rbtree_ctx {
struct rb_root root;
+ struct snd_soc_rbtree_node *cached_rbnode;
};
+static inline void snd_soc_rbtree_get_base_top_reg(
+ struct snd_soc_rbtree_node *rbnode,
+ unsigned int *base, unsigned int *top)
+{
+ *base = rbnode->base_reg;
+ *top = rbnode->base_reg + rbnode->blklen - 1;
+}
+
+static unsigned int snd_soc_rbtree_get_register(
+ struct snd_soc_rbtree_node *rbnode, unsigned int idx)
+{
+ unsigned int val;
+
+ switch (rbnode->word_size) {
+ case 1: {
+ u8 *p = rbnode->block;
+ val = p[idx];
+ return val;
+ }
+ case 2: {
+ u16 *p = rbnode->block;
+ val = p[idx];
+ return val;
+ }
+ default:
+ BUG();
+ break;
+ }
+ return -1;
+}
+
+static void snd_soc_rbtree_set_register(struct snd_soc_rbtree_node *rbnode,
+ unsigned int idx, unsigned int val)
+{
+ switch (rbnode->word_size) {
+ case 1: {
+ u8 *p = rbnode->block;
+ p[idx] = val;
+ break;
+ }
+ case 2: {
+ u16 *p = rbnode->block;
+ p[idx] = val;
+ break;
+ }
+ default:
+ BUG();
+ break;
+ }
+}
+
static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup(
struct rb_root *root, unsigned int reg)
{
struct rb_node *node;
struct snd_soc_rbtree_node *rbnode;
+ unsigned int base_reg, top_reg;
node = root->rb_node;
while (node) {
rbnode = container_of(node, struct snd_soc_rbtree_node, node);
- if (rbnode->reg < reg)
- node = node->rb_left;
- else if (rbnode->reg > reg)
- node = node->rb_right;
- else
+ snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+ if (reg >= base_reg && reg <= top_reg)
return rbnode;
+ else if (reg > top_reg)
+ node = node->rb_right;
+ else if (reg < base_reg)
+ node = node->rb_left;
}
return NULL;
@@ -518,19 +157,28 @@
{
struct rb_node **new, *parent;
struct snd_soc_rbtree_node *rbnode_tmp;
+ unsigned int base_reg_tmp, top_reg_tmp;
+ unsigned int base_reg;
parent = NULL;
new = &root->rb_node;
while (*new) {
rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node,
node);
+ /* base and top registers of the current rbnode */
+ snd_soc_rbtree_get_base_top_reg(rbnode_tmp, &base_reg_tmp,
+ &top_reg_tmp);
+ /* base register of the rbnode to be added */
+ base_reg = rbnode->base_reg;
parent = *new;
- if (rbnode_tmp->reg < rbnode->reg)
- new = &((*new)->rb_left);
- else if (rbnode_tmp->reg > rbnode->reg)
- new = &((*new)->rb_right);
- else
+ /* if this register has already been inserted, just return */
+ if (base_reg >= base_reg_tmp &&
+ base_reg <= top_reg_tmp)
return 0;
+ else if (base_reg > top_reg_tmp)
+ new = &((*new)->rb_right);
+ else if (base_reg < base_reg_tmp)
+ new = &((*new)->rb_left);
}
/* insert the node into the rbtree */
@@ -545,58 +193,146 @@
struct snd_soc_rbtree_ctx *rbtree_ctx;
struct rb_node *node;
struct snd_soc_rbtree_node *rbnode;
- unsigned int val;
+ unsigned int regtmp;
+ unsigned int val, def;
int ret;
+ int i;
rbtree_ctx = codec->reg_cache;
for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) {
rbnode = rb_entry(node, struct snd_soc_rbtree_node, node);
- if (rbnode->value == rbnode->defval)
- continue;
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, rbnode->reg));
- ret = snd_soc_cache_read(codec, rbnode->reg, &val);
- if (ret)
- return ret;
- codec->cache_bypass = 1;
- ret = snd_soc_write(codec, rbnode->reg, val);
- codec->cache_bypass = 0;
- if (ret)
- return ret;
- dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
- rbnode->reg, val);
+ for (i = 0; i < rbnode->blklen; ++i) {
+ regtmp = rbnode->base_reg + i;
+ WARN_ON(codec->writable_register &&
+ codec->writable_register(codec, regtmp));
+ val = snd_soc_rbtree_get_register(rbnode, i);
+ def = snd_soc_get_cache_val(codec->reg_def_copy, i,
+ rbnode->word_size);
+ if (val == def)
+ continue;
+
+ codec->cache_bypass = 1;
+ ret = snd_soc_write(codec, regtmp, val);
+ codec->cache_bypass = 0;
+ if (ret)
+ return ret;
+ dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
+ regtmp, val);
+ }
}
return 0;
}
+static int snd_soc_rbtree_insert_to_block(struct snd_soc_rbtree_node *rbnode,
+ unsigned int pos, unsigned int reg,
+ unsigned int value)
+{
+ u8 *blk;
+
+ blk = krealloc(rbnode->block,
+ (rbnode->blklen + 1) * rbnode->word_size, GFP_KERNEL);
+ if (!blk)
+ return -ENOMEM;
+
+ /* insert the register value in the correct place in the rbnode block */
+ memmove(blk + (pos + 1) * rbnode->word_size,
+ blk + pos * rbnode->word_size,
+ (rbnode->blklen - pos) * rbnode->word_size);
+
+ /* update the rbnode block, its size and the base register */
+ rbnode->block = blk;
+ rbnode->blklen++;
+ if (!pos)
+ rbnode->base_reg = reg;
+
+ snd_soc_rbtree_set_register(rbnode, pos, value);
+ return 0;
+}
+
static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
struct snd_soc_rbtree_ctx *rbtree_ctx;
- struct snd_soc_rbtree_node *rbnode;
+ struct snd_soc_rbtree_node *rbnode, *rbnode_tmp;
+ struct rb_node *node;
+ unsigned int val;
+ unsigned int reg_tmp;
+ unsigned int base_reg, top_reg;
+ unsigned int pos;
+ int i;
+ int ret;
rbtree_ctx = codec->reg_cache;
+ /* look up the required register in the cached rbnode */
+ rbnode = rbtree_ctx->cached_rbnode;
+ if (rbnode) {
+ snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+ if (reg >= base_reg && reg <= top_reg) {
+ reg_tmp = reg - base_reg;
+ val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ if (val == value)
+ return 0;
+ snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
+ return 0;
+ }
+ }
+ /* if we can't locate it in the cached rbnode we'll have
+ * to traverse the rbtree looking for it.
+ */
rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
if (rbnode) {
- if (rbnode->value == value)
+ reg_tmp = reg - rbnode->base_reg;
+ val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ if (val == value)
return 0;
- rbnode->value = value;
+ snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
+ rbtree_ctx->cached_rbnode = rbnode;
} else {
/* bail out early, no need to create the rbnode yet */
if (!value)
return 0;
- /*
- * for uninitialized registers whose value is changed
- * from the default zero, create an rbnode and insert
- * it into the tree.
+ /* look for an adjacent register to the one we are about to add */
+ for (node = rb_first(&rbtree_ctx->root); node;
+ node = rb_next(node)) {
+ rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node);
+ for (i = 0; i < rbnode_tmp->blklen; ++i) {
+ reg_tmp = rbnode_tmp->base_reg + i;
+ if (abs(reg_tmp - reg) != 1)
+ continue;
+ /* decide where in the block to place our register */
+ if (reg_tmp + 1 == reg)
+ pos = i + 1;
+ else
+ pos = i;
+ ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos,
+ reg, value);
+ if (ret)
+ return ret;
+ rbtree_ctx->cached_rbnode = rbnode_tmp;
+ return 0;
+ }
+ }
+ /* we did not manage to find a place to insert it in an existing
+ * block so create a new rbnode with a single register in its block.
+ * This block will get populated further if any other adjacent
+ * registers get modified in the future.
*/
rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL);
if (!rbnode)
return -ENOMEM;
- rbnode->reg = reg;
- rbnode->value = value;
+ rbnode->blklen = 1;
+ rbnode->base_reg = reg;
+ rbnode->word_size = codec->driver->reg_word_size;
+ rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size,
+ GFP_KERNEL);
+ if (!rbnode->block) {
+ kfree(rbnode);
+ return -ENOMEM;
+ }
+ snd_soc_rbtree_set_register(rbnode, 0, value);
snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode);
+ rbtree_ctx->cached_rbnode = rbnode;
}
return 0;
@@ -607,11 +343,28 @@
{
struct snd_soc_rbtree_ctx *rbtree_ctx;
struct snd_soc_rbtree_node *rbnode;
+ unsigned int base_reg, top_reg;
+ unsigned int reg_tmp;
rbtree_ctx = codec->reg_cache;
+ /* look up the required register in the cached rbnode */
+ rbnode = rbtree_ctx->cached_rbnode;
+ if (rbnode) {
+ snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
+ if (reg >= base_reg && reg <= top_reg) {
+ reg_tmp = reg - base_reg;
+ *value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ return 0;
+ }
+ }
+ /* if we can't locate it in the cached rbnode we'll have
+ * to traverse the rbtree looking for it.
+ */
rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
if (rbnode) {
- *value = rbnode->value;
+ reg_tmp = reg - rbnode->base_reg;
+ *value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
+ rbtree_ctx->cached_rbnode = rbnode;
} else {
/* uninitialized registers default to 0 */
*value = 0;
@@ -637,6 +390,7 @@
rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node);
next = rb_next(&rbtree_node->node);
rb_erase(&rbtree_node->node, &rbtree_ctx->root);
+ kfree(rbtree_node->block);
kfree(rbtree_node);
}
@@ -649,10 +403,9 @@
static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
{
- struct snd_soc_rbtree_node *rbtree_node;
struct snd_soc_rbtree_ctx *rbtree_ctx;
- unsigned int val;
unsigned int word_size;
+ unsigned int val;
int i;
int ret;
@@ -662,32 +415,27 @@
rbtree_ctx = codec->reg_cache;
rbtree_ctx->root = RB_ROOT;
+ rbtree_ctx->cached_rbnode = NULL;
if (!codec->reg_def_copy)
return 0;
- /*
- * populate the rbtree with the initialized registers. All other
- * registers will be inserted when they are first modified.
- */
word_size = codec->driver->reg_word_size;
for (i = 0; i < codec->driver->reg_cache_size; ++i) {
- val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size);
+ val = snd_soc_get_cache_val(codec->reg_def_copy, i,
+ word_size);
if (!val)
continue;
- rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL);
- if (!rbtree_node) {
- ret = -ENOMEM;
- snd_soc_cache_exit(codec);
- break;
- }
- rbtree_node->reg = i;
- rbtree_node->value = val;
- rbtree_node->defval = val;
- snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node);
+ ret = snd_soc_rbtree_cache_write(codec, i, val);
+ if (ret)
+ goto err;
}
return 0;
+
+err:
+ snd_soc_cache_exit(codec);
+ return ret;
}
#ifdef CONFIG_SND_SOC_CACHE_LZO
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b194be0..e44267f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -44,7 +44,6 @@
#define NAME_SIZE 32
-static DEFINE_MUTEX(pcm_mutex);
static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
#ifdef CONFIG_DEBUG_FS
@@ -58,7 +57,7 @@
static LIST_HEAD(platform_list);
static LIST_HEAD(codec_list);
-static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -485,552 +484,6 @@
}
#endif
-static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- if (!codec_dai->driver->symmetric_rates &&
- !cpu_dai->driver->symmetric_rates &&
- !rtd->dai_link->symmetric_rates)
- return 0;
-
- /* This can happen if multiple streams are starting simultaneously -
- * the second can need to get its constraints before the first has
- * picked a rate. Complain and allow the application to carry on.
- */
- if (!rtd->rate) {
- dev_warn(&rtd->dev,
- "Not enforcing symmetric_rates due to race\n");
- return 0;
- }
-
- dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
-
- ret = snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- rtd->rate, rtd->rate);
- if (ret < 0) {
- dev_err(&rtd->dev,
- "Unable to apply rate symmetry constraint: %d\n", ret);
- return ret;
- }
-
- return 0;
-}
-
-/*
- * Called by ALSA when a PCM substream is opened, the runtime->hw record is
- * then initialized and any private data can be allocated. This also calls
- * startup for the cpu DAI, platform, machine and codec DAI.
- */
-static int soc_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
- struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver;
- int ret = 0;
-
- mutex_lock(&pcm_mutex);
-
- /* startup the audio subsystem */
- if (cpu_dai->driver->ops->startup) {
- ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't open interface %s\n",
- cpu_dai->name);
- goto out;
- }
- }
-
- if (platform->driver->ops && platform->driver->ops->open) {
- ret = platform->driver->ops->open(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
- goto platform_err;
- }
- }
-
- if (codec_dai->driver->ops->startup) {
- ret = codec_dai->driver->ops->startup(substream, codec_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't open codec %s\n",
- codec_dai->name);
- goto codec_dai_err;
- }
- }
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
- ret = rtd->dai_link->ops->startup(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name);
- goto machine_err;
- }
- }
-
- /* Check that the codec and cpu DAIs are compatible */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- runtime->hw.rate_min =
- max(codec_dai_drv->playback.rate_min,
- cpu_dai_drv->playback.rate_min);
- runtime->hw.rate_max =
- min(codec_dai_drv->playback.rate_max,
- cpu_dai_drv->playback.rate_max);
- runtime->hw.channels_min =
- max(codec_dai_drv->playback.channels_min,
- cpu_dai_drv->playback.channels_min);
- runtime->hw.channels_max =
- min(codec_dai_drv->playback.channels_max,
- cpu_dai_drv->playback.channels_max);
- runtime->hw.formats =
- codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats;
- runtime->hw.rates =
- codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates;
- if (codec_dai_drv->playback.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= cpu_dai_drv->playback.rates;
- if (cpu_dai_drv->playback.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= codec_dai_drv->playback.rates;
- } else {
- runtime->hw.rate_min =
- max(codec_dai_drv->capture.rate_min,
- cpu_dai_drv->capture.rate_min);
- runtime->hw.rate_max =
- min(codec_dai_drv->capture.rate_max,
- cpu_dai_drv->capture.rate_max);
- runtime->hw.channels_min =
- max(codec_dai_drv->capture.channels_min,
- cpu_dai_drv->capture.channels_min);
- runtime->hw.channels_max =
- min(codec_dai_drv->capture.channels_max,
- cpu_dai_drv->capture.channels_max);
- runtime->hw.formats =
- codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats;
- runtime->hw.rates =
- codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates;
- if (codec_dai_drv->capture.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= cpu_dai_drv->capture.rates;
- if (cpu_dai_drv->capture.rates
- & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
- runtime->hw.rates |= codec_dai_drv->capture.rates;
- }
-
- ret = -EINVAL;
- snd_pcm_limit_hw_rates(runtime);
- if (!runtime->hw.rates) {
- printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
- codec_dai->name, cpu_dai->name);
- goto config_err;
- }
- if (!runtime->hw.formats) {
- printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
- codec_dai->name, cpu_dai->name);
- goto config_err;
- }
- if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
- runtime->hw.channels_min > runtime->hw.channels_max) {
- printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
- codec_dai->name, cpu_dai->name);
- goto config_err;
- }
-
- /* Symmetry only applies if we've already got an active stream. */
- if (cpu_dai->active || codec_dai->active) {
- ret = soc_pcm_apply_symmetry(substream);
- if (ret != 0)
- goto config_err;
- }
-
- pr_debug("asoc: %s <-> %s info:\n",
- codec_dai->name, cpu_dai->name);
- pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
- pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
- runtime->hw.channels_max);
- pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
- runtime->hw.rate_max);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
- cpu_dai->active++;
- codec_dai->active++;
- rtd->codec->active++;
- mutex_unlock(&pcm_mutex);
- return 0;
-
-config_err:
- if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
- rtd->dai_link->ops->shutdown(substream);
-
-machine_err:
- if (codec_dai->driver->ops->shutdown)
- codec_dai->driver->ops->shutdown(substream, codec_dai);
-
-codec_dai_err:
- if (platform->driver->ops && platform->driver->ops->close)
- platform->driver->ops->close(substream);
-
-platform_err:
- if (cpu_dai->driver->ops->shutdown)
- cpu_dai->driver->ops->shutdown(substream, cpu_dai);
-out:
- mutex_unlock(&pcm_mutex);
- return ret;
-}
-
-/*
- * Power down the audio subsystem pmdown_time msecs after close is called.
- * This is to ensure there are no pops or clicks in between any music tracks
- * due to DAPM power cycling.
- */
-static void close_delayed_work(struct work_struct *work)
-{
- struct snd_soc_pcm_runtime *rtd =
- container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
- mutex_lock(&pcm_mutex);
-
- pr_debug("pop wq checking: %s status: %s waiting: %s\n",
- codec_dai->driver->playback.stream_name,
- codec_dai->playback_active ? "active" : "inactive",
- codec_dai->pop_wait ? "yes" : "no");
-
- /* are we waiting on this codec DAI stream */
- if (codec_dai->pop_wait == 1) {
- codec_dai->pop_wait = 0;
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->playback.stream_name,
- SND_SOC_DAPM_STREAM_STOP);
- }
-
- mutex_unlock(&pcm_mutex);
-}
-
-/*
- * Called by ALSA when a PCM substream is closed. Private data can be
- * freed here. The cpu DAI, codec DAI, machine and platform are also
- * shutdown.
- */
-static int soc_codec_close(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&pcm_mutex);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
-
- cpu_dai->active--;
- codec_dai->active--;
- codec->active--;
-
- /* Muting the DAC suppresses artifacts caused during digital
- * shutdown, for example from stopping clocks.
- */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dai_digital_mute(codec_dai, 1);
-
- if (cpu_dai->driver->ops->shutdown)
- cpu_dai->driver->ops->shutdown(substream, cpu_dai);
-
- if (codec_dai->driver->ops->shutdown)
- codec_dai->driver->ops->shutdown(substream, codec_dai);
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
- rtd->dai_link->ops->shutdown(substream);
-
- if (platform->driver->ops && platform->driver->ops->close)
- platform->driver->ops->close(substream);
- cpu_dai->runtime = NULL;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* start delayed pop wq here for playback streams */
- codec_dai->pop_wait = 1;
- schedule_delayed_work(&rtd->delayed_work,
- msecs_to_jiffies(rtd->pmdown_time));
- } else {
- /* capture streams can be powered down now */
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->capture.stream_name,
- SND_SOC_DAPM_STREAM_STOP);
- }
-
- mutex_unlock(&pcm_mutex);
- return 0;
-}
-
-/*
- * Called by ALSA when the PCM substream is prepared, can set format, sample
- * rate, etc. This function is non atomic and can be called multiple times,
- * it can refer to the runtime info.
- */
-static int soc_pcm_prepare(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret = 0;
-
- mutex_lock(&pcm_mutex);
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
- ret = rtd->dai_link->ops->prepare(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: machine prepare error\n");
- goto out;
- }
- }
-
- if (platform->driver->ops && platform->driver->ops->prepare) {
- ret = platform->driver->ops->prepare(substream);
- if (ret < 0) {
- printk(KERN_ERR "asoc: platform prepare error\n");
- goto out;
- }
- }
-
- if (codec_dai->driver->ops->prepare) {
- ret = codec_dai->driver->ops->prepare(substream, codec_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: codec DAI prepare error\n");
- goto out;
- }
- }
-
- if (cpu_dai->driver->ops->prepare) {
- ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: cpu DAI prepare error\n");
- goto out;
- }
- }
-
- /* cancel any delayed stream shutdown that is pending */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
- codec_dai->pop_wait) {
- codec_dai->pop_wait = 0;
- cancel_delayed_work(&rtd->delayed_work);
- }
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->playback.stream_name,
- SND_SOC_DAPM_STREAM_START);
- else
- snd_soc_dapm_stream_event(rtd,
- codec_dai->driver->capture.stream_name,
- SND_SOC_DAPM_STREAM_START);
-
- snd_soc_dai_digital_mute(codec_dai, 0);
-
-out:
- mutex_unlock(&pcm_mutex);
- return ret;
-}
-
-/*
- * Called by ALSA when the hardware params are set by application. This
- * function can also be called multiple times and can allocate buffers
- * (using snd_pcm_lib_* ). It's non-atomic.
- */
-static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret = 0;
-
- mutex_lock(&pcm_mutex);
-
- if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
- ret = rtd->dai_link->ops->hw_params(substream, params);
- if (ret < 0) {
- printk(KERN_ERR "asoc: machine hw_params failed\n");
- goto out;
- }
- }
-
- if (codec_dai->driver->ops->hw_params) {
- ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't set codec %s hw params\n",
- codec_dai->name);
- goto codec_err;
- }
- }
-
- if (cpu_dai->driver->ops->hw_params) {
- ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
- if (ret < 0) {
- printk(KERN_ERR "asoc: interface %s hw params failed\n",
- cpu_dai->name);
- goto interface_err;
- }
- }
-
- if (platform->driver->ops && platform->driver->ops->hw_params) {
- ret = platform->driver->ops->hw_params(substream, params);
- if (ret < 0) {
- printk(KERN_ERR "asoc: platform %s hw params failed\n",
- platform->name);
- goto platform_err;
- }
- }
-
- rtd->rate = params_rate(params);
-
-out:
- mutex_unlock(&pcm_mutex);
- return ret;
-
-platform_err:
- if (cpu_dai->driver->ops->hw_free)
- cpu_dai->driver->ops->hw_free(substream, cpu_dai);
-
-interface_err:
- if (codec_dai->driver->ops->hw_free)
- codec_dai->driver->ops->hw_free(substream, codec_dai);
-
-codec_err:
- if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
- rtd->dai_link->ops->hw_free(substream);
-
- mutex_unlock(&pcm_mutex);
- return ret;
-}
-
-/*
- * Frees resources allocated by hw_params, can be called multiple times
- */
-static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&pcm_mutex);
-
- /* apply codec digital mute */
- if (!codec->active)
- snd_soc_dai_digital_mute(codec_dai, 1);
-
- /* free any machine hw params */
- if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
- rtd->dai_link->ops->hw_free(substream);
-
- /* free any DMA resources */
- if (platform->driver->ops && platform->driver->ops->hw_free)
- platform->driver->ops->hw_free(substream);
-
- /* now free hw params for the DAIs */
- if (codec_dai->driver->ops->hw_free)
- codec_dai->driver->ops->hw_free(substream, codec_dai);
-
- if (cpu_dai->driver->ops->hw_free)
- cpu_dai->driver->ops->hw_free(substream, cpu_dai);
-
- mutex_unlock(&pcm_mutex);
- return 0;
-}
-
-static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int ret;
-
- if (codec_dai->driver->ops->trigger) {
- ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
- if (ret < 0)
- return ret;
- }
-
- if (platform->driver->ops && platform->driver->ops->trigger) {
- ret = platform->driver->ops->trigger(substream, cmd);
- if (ret < 0)
- return ret;
- }
-
- if (cpu_dai->driver->ops->trigger) {
- ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
- if (ret < 0)
- return ret;
- }
- return 0;
-}
-
-/*
- * soc level wrapper for pointer callback
- * If cpu_dai, codec_dai, platform driver has the delay callback, than
- * the runtime->delay will be updated accordingly.
- */
-static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_uframes_t offset = 0;
- snd_pcm_sframes_t delay = 0;
-
- if (platform->driver->ops && platform->driver->ops->pointer)
- offset = platform->driver->ops->pointer(substream);
-
- if (cpu_dai->driver->ops->delay)
- delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
-
- if (codec_dai->driver->ops->delay)
- delay += codec_dai->driver->ops->delay(substream, codec_dai);
-
- if (platform->driver->delay)
- delay += platform->driver->delay(substream, codec_dai);
-
- runtime->delay = delay;
-
- return offset;
-}
-
-/* ASoC PCM operations */
-static struct snd_pcm_ops soc_pcm_ops = {
- .open = soc_pcm_open,
- .close = soc_codec_close,
- .hw_params = soc_pcm_hw_params,
- .hw_free = soc_pcm_hw_free,
- .prepare = soc_pcm_prepare,
- .trigger = soc_pcm_trigger,
- .pointer = soc_pcm_pointer,
-};
-
#ifdef CONFIG_PM_SLEEP
/* powers down audio subsystem for suspend */
int snd_soc_suspend(struct device *dev)
@@ -1256,7 +709,7 @@
int snd_soc_resume(struct device *dev)
{
struct snd_soc_card *card = dev_get_drvdata(dev);
- int i;
+ int i, ac97_control = 0;
/* AC97 devices might have other drivers hanging off them so
* need to resume immediately. Other drivers don't have that
@@ -1265,14 +718,15 @@
*/
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
- if (cpu_dai->driver->ac97_control) {
- dev_dbg(dev, "Resuming AC97 immediately\n");
- soc_resume_deferred(&card->deferred_resume_work);
- } else {
- dev_dbg(dev, "Scheduling resume work\n");
- if (!schedule_work(&card->deferred_resume_work))
- dev_err(dev, "resume work item may be lost\n");
- }
+ ac97_control |= cpu_dai->driver->ac97_control;
+ }
+ if (ac97_control) {
+ dev_dbg(dev, "Resuming AC97 immediately\n");
+ soc_resume_deferred(&card->deferred_resume_work);
+ } else {
+ dev_dbg(dev, "Scheduling resume work\n");
+ if (!schedule_work(&card->deferred_resume_work))
+ dev_err(dev, "resume work item may be lost\n");
}
return 0;
@@ -1393,7 +847,7 @@
module_put(codec->dev->driver->owner);
}
-static void soc_remove_dai_link(struct snd_soc_card *card, int num)
+static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
{
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
struct snd_soc_codec *codec = rtd->codec;
@@ -1410,7 +864,8 @@
}
/* remove the CODEC DAI */
- if (codec_dai && codec_dai->probed) {
+ if (codec_dai && codec_dai->probed &&
+ codec_dai->driver->remove_order == order) {
if (codec_dai->driver->remove) {
err = codec_dai->driver->remove(codec_dai);
if (err < 0)
@@ -1421,7 +876,8 @@
}
/* remove the platform */
- if (platform && platform->probed) {
+ if (platform && platform->probed &&
+ platform->driver->remove_order == order) {
if (platform->driver->remove) {
err = platform->driver->remove(platform);
if (err < 0)
@@ -1433,11 +889,13 @@
}
/* remove the CODEC */
- if (codec && codec->probed)
+ if (codec && codec->probed &&
+ codec->driver->remove_order == order)
soc_remove_codec(codec);
/* remove the cpu_dai */
- if (cpu_dai && cpu_dai->probed) {
+ if (cpu_dai && cpu_dai->probed &&
+ cpu_dai->driver->remove_order == order) {
if (cpu_dai->driver->remove) {
err = cpu_dai->driver->remove(cpu_dai);
if (err < 0)
@@ -1451,11 +909,13 @@
static void soc_remove_dai_links(struct snd_soc_card *card)
{
- int i;
+ int dai, order;
- for (i = 0; i < card->num_rtd; i++)
- soc_remove_dai_link(card, i);
-
+ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+ order++) {
+ for (dai = 0; dai < card->num_rtd; dai++)
+ soc_remove_dai_link(card, dai, order);
+ }
card->num_rtd = 0;
}
@@ -1526,6 +986,52 @@
return ret;
}
+static int soc_probe_platform(struct snd_soc_card *card,
+ struct snd_soc_platform *platform)
+{
+ int ret = 0;
+ const struct snd_soc_platform_driver *driver = platform->driver;
+
+ platform->card = card;
+ platform->dapm.card = card;
+
+ if (!try_module_get(platform->dev->driver->owner))
+ return -ENODEV;
+
+ if (driver->dapm_widgets)
+ snd_soc_dapm_new_controls(&platform->dapm,
+ driver->dapm_widgets, driver->num_dapm_widgets);
+
+ if (driver->probe) {
+ ret = driver->probe(platform);
+ if (ret < 0) {
+ dev_err(platform->dev,
+ "asoc: failed to probe platform %s: %d\n",
+ platform->name, ret);
+ goto err_probe;
+ }
+ }
+
+ if (driver->controls)
+ snd_soc_add_platform_controls(platform, driver->controls,
+ driver->num_controls);
+ if (driver->dapm_routes)
+ snd_soc_dapm_add_routes(&platform->dapm, driver->dapm_routes,
+ driver->num_dapm_routes);
+
+ /* mark platform as probed and add to card platform list */
+ platform->probed = 1;
+ list_add(&platform->card_list, &card->platform_dev_list);
+ list_add(&platform->dapm.list, &card->dapm_list);
+
+ return 0;
+
+err_probe:
+ module_put(platform->dev->driver->owner);
+
+ return ret;
+}
+
static void rtd_release(struct device *dev) {}
static int soc_post_component_init(struct snd_soc_card *card,
@@ -1572,6 +1078,7 @@
rtd->dev.parent = card->dev;
rtd->dev.release = rtd_release;
rtd->dev.init_name = name;
+ mutex_init(&rtd->pcm_mutex);
ret = device_register(&rtd->dev);
if (ret < 0) {
dev_err(card->dev,
@@ -1596,7 +1103,7 @@
return 0;
}
-static int soc_probe_dai_link(struct snd_soc_card *card, int num)
+static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
{
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
@@ -1605,7 +1112,8 @@
struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
int ret;
- dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num);
+ dev_dbg(card->dev, "probe %s dai link %d late %d\n",
+ card->name, num, order);
/* config components */
codec_dai->codec = codec;
@@ -1617,7 +1125,8 @@
rtd->pmdown_time = pmdown_time;
/* probe the cpu_dai */
- if (!cpu_dai->probed) {
+ if (!cpu_dai->probed &&
+ cpu_dai->driver->probe_order == order) {
if (!try_module_get(cpu_dai->dev->driver->owner))
return -ENODEV;
@@ -1636,33 +1145,23 @@
}
/* probe the CODEC */
- if (!codec->probed) {
+ if (!codec->probed &&
+ codec->driver->probe_order == order) {
ret = soc_probe_codec(card, codec);
if (ret < 0)
return ret;
}
/* probe the platform */
- if (!platform->probed) {
- if (!try_module_get(platform->dev->driver->owner))
- return -ENODEV;
-
- if (platform->driver->probe) {
- ret = platform->driver->probe(platform);
- if (ret < 0) {
- printk(KERN_ERR "asoc: failed to probe platform %s\n",
- platform->name);
- module_put(platform->dev->driver->owner);
- return ret;
- }
- }
- /* mark platform as probed and add to card platform list */
- platform->probed = 1;
- list_add(&platform->card_list, &card->platform_dev_list);
+ if (!platform->probed &&
+ platform->driver->probe_order == order) {
+ ret = soc_probe_platform(card, platform);
+ if (ret < 0)
+ return ret;
}
/* probe the CODEC DAI */
- if (!codec_dai->probed) {
+ if (!codec_dai->probed && codec_dai->driver->probe_order == order) {
if (codec_dai->driver->probe) {
ret = codec_dai->driver->probe(codec_dai);
if (ret < 0) {
@@ -1677,8 +1176,9 @@
list_add(&codec_dai->card_list, &card->dai_dev_list);
}
- /* DAPM dai link stream work */
- INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+ /* complete DAI probe during last probe */
+ if (order != SND_SOC_COMP_ORDER_LAST)
+ return 0;
ret = soc_post_component_init(card, codec, num, 0);
if (ret)
@@ -1817,7 +1317,7 @@
struct snd_soc_codec *codec;
struct snd_soc_codec_conf *codec_conf;
enum snd_soc_compress_type compress_type;
- int ret, i;
+ int ret, i, order;
mutex_lock(&card->mutex);
@@ -1895,12 +1395,16 @@
goto card_probe_error;
}
- for (i = 0; i < card->num_links; i++) {
- ret = soc_probe_dai_link(card, i);
- if (ret < 0) {
- pr_err("asoc: failed to instantiate card %s: %d\n",
+ /* early DAI link probe */
+ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+ order++) {
+ for (i = 0; i < card->num_links; i++) {
+ ret = soc_probe_dai_link(card, i, order);
+ if (ret < 0) {
+ pr_err("asoc: failed to instantiate card %s: %d\n",
card->name, ret);
- goto probe_dai_err;
+ goto probe_dai_err;
+ }
}
}
@@ -2096,67 +1600,6 @@
.remove = soc_remove,
};
-/* create a new pcm */
-static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_pcm *pcm;
- char new_name[64];
- int ret = 0, playback = 0, capture = 0;
-
- /* check client and interface hw capabilities */
- snprintf(new_name, sizeof(new_name), "%s %s-%d",
- rtd->dai_link->stream_name, codec_dai->name, num);
-
- if (codec_dai->driver->playback.channels_min)
- playback = 1;
- if (codec_dai->driver->capture.channels_min)
- capture = 1;
-
- dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
- ret = snd_pcm_new(rtd->card->snd_card, new_name,
- num, playback, capture, &pcm);
- if (ret < 0) {
- printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
- return ret;
- }
-
- rtd->pcm = pcm;
- pcm->private_data = rtd;
- if (platform->driver->ops) {
- soc_pcm_ops.mmap = platform->driver->ops->mmap;
- soc_pcm_ops.pointer = platform->driver->ops->pointer;
- soc_pcm_ops.ioctl = platform->driver->ops->ioctl;
- soc_pcm_ops.copy = platform->driver->ops->copy;
- soc_pcm_ops.silence = platform->driver->ops->silence;
- soc_pcm_ops.ack = platform->driver->ops->ack;
- soc_pcm_ops.page = platform->driver->ops->page;
- }
-
- if (playback)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
-
- if (capture)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
-
- if (platform->driver->pcm_new) {
- ret = platform->driver->pcm_new(rtd->card->snd_card,
- codec_dai, pcm);
- if (ret < 0) {
- pr_err("asoc: platform pcm constructor failed\n");
- return ret;
- }
- }
-
- pcm->private_free = platform->driver->pcm_free;
- printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
- cpu_dai->name);
- return ret;
-}
-
/**
* snd_soc_codec_volatile_register: Report if a register is volatile.
*
@@ -2211,6 +1654,38 @@
}
EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register);
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+ unsigned int reg)
+{
+ unsigned int ret;
+
+ if (!platform->driver->read) {
+ dev_err(platform->dev, "platform has no read back\n");
+ return -1;
+ }
+
+ ret = platform->driver->read(platform, reg);
+ dev_dbg(platform->dev, "read %x => %x\n", reg, ret);
+ trace_snd_soc_preg_read(platform, reg, ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_platform_read);
+
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+ unsigned int reg, unsigned int val)
+{
+ if (!platform->driver->write) {
+ dev_err(platform->dev, "platform has no write back\n");
+ return -1;
+ }
+
+ dev_dbg(platform->dev, "write %x = %x\n", reg, val);
+ trace_snd_soc_preg_write(platform, reg, val);
+ return platform->driver->write(platform, reg, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_platform_write);
+
/**
* snd_soc_new_ac97_codec - initailise AC97 device
* @codec: audio codec
@@ -2323,7 +1798,7 @@
return ret;
old = ret;
- new = (old & ~mask) | value;
+ new = (old & ~mask) | (value & mask);
change = old != new;
if (change) {
ret = snd_soc_write(codec, reg, new);
@@ -2490,6 +1965,36 @@
EXPORT_SYMBOL_GPL(snd_soc_add_controls);
/**
+ * snd_soc_add_platform_controls - add an array of controls to a platform.
+ * Convienience function to add a list of controls.
+ *
+ * @platform: platform to add controls to
+ * @controls: array of controls to add
+ * @num_controls: number of elements in the array
+ *
+ * Return 0 for success, else error.
+ */
+int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
+ const struct snd_kcontrol_new *controls, int num_controls)
+{
+ struct snd_card *card = platform->card->snd_card;
+ int err, i;
+
+ for (i = 0; i < num_controls; i++) {
+ const struct snd_kcontrol_new *control = &controls[i];
+ err = snd_ctl_add(card, snd_soc_cnew(control, platform,
+ control->name, NULL));
+ if (err < 0) {
+ dev_err(platform->dev, "Failed to add %s %d\n",control->name, err);
+ return err;
+ }
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls);
+
+/**
* snd_soc_info_enum_double - enumerated double mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
@@ -3633,6 +3138,8 @@
platform->dev = dev;
platform->driver = platform_drv;
+ platform->dapm.dev = dev;
+ platform->dapm.platform = platform;
mutex_lock(&client_mutex);
list_add(&platform->list, &platform_list);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 32ab7fc..fbfcda0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -124,6 +124,51 @@
return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
}
+static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg)
+{
+ if (w->codec)
+ return snd_soc_read(w->codec, reg);
+ else if (w->platform)
+ return snd_soc_platform_read(w->platform, reg);
+
+ dev_err(w->dapm->dev, "no valid widget read method\n");
+ return -1;
+}
+
+static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val)
+{
+ if (w->codec)
+ return snd_soc_write(w->codec, reg, val);
+ else if (w->platform)
+ return snd_soc_platform_write(w->platform, reg, val);
+
+ dev_err(w->dapm->dev, "no valid widget write method\n");
+ return -1;
+}
+
+static int soc_widget_update_bits(struct snd_soc_dapm_widget *w,
+ unsigned short reg, unsigned int mask, unsigned int value)
+{
+ int change;
+ unsigned int old, new;
+ int ret;
+
+ ret = soc_widget_read(w, reg);
+ if (ret < 0)
+ return ret;
+
+ old = ret;
+ new = (old & ~mask) | (value & mask);
+ change = old != new;
+ if (change) {
+ ret = soc_widget_write(w, reg, new);
+ if (ret < 0)
+ return ret;
+ }
+
+ return change;
+}
+
/**
* snd_soc_dapm_set_bias_level - set the bias level for the system
* @dapm: DAPM context
@@ -139,39 +184,26 @@
struct snd_soc_card *card = dapm->card;
int ret = 0;
- switch (level) {
- case SND_SOC_BIAS_ON:
- dev_dbg(dapm->dev, "Setting full bias\n");
- break;
- case SND_SOC_BIAS_PREPARE:
- dev_dbg(dapm->dev, "Setting bias prepare\n");
- break;
- case SND_SOC_BIAS_STANDBY:
- dev_dbg(dapm->dev, "Setting standby bias\n");
- break;
- case SND_SOC_BIAS_OFF:
- dev_dbg(dapm->dev, "Setting bias off\n");
- break;
- default:
- dev_err(dapm->dev, "Setting invalid bias %d\n", level);
- return -EINVAL;
- }
-
trace_snd_soc_bias_level_start(card, level);
if (card && card->set_bias_level)
- ret = card->set_bias_level(card, level);
- if (ret == 0) {
- if (dapm->codec && dapm->codec->driver->set_bias_level)
- ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
+ ret = card->set_bias_level(card, dapm, level);
+ if (ret != 0)
+ goto out;
+
+ if (dapm->codec) {
+ if (dapm->codec->driver->set_bias_level)
+ ret = dapm->codec->driver->set_bias_level(dapm->codec,
+ level);
else
dapm->bias_level = level;
}
- if (ret == 0) {
- if (card && card->set_bias_level_post)
- ret = card->set_bias_level_post(card, level);
- }
+ if (ret != 0)
+ goto out;
+ if (card && card->set_bias_level_post)
+ ret = card->set_bias_level_post(card, dapm, level);
+out:
trace_snd_soc_bias_level_done(card, level);
return ret;
@@ -194,7 +226,7 @@
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- val = snd_soc_read(w->codec, reg);
+ val = soc_widget_read(w, reg);
val = (val >> shift) & mask;
if ((invert && !val) || (!invert && val))
@@ -209,8 +241,8 @@
int val, item, bitmask;
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
- ;
- val = snd_soc_read(w->codec, e->reg);
+ ;
+ val = soc_widget_read(w, e->reg);
item = (val >> e->shift_l) & (bitmask - 1);
p->connect = 0;
@@ -240,7 +272,7 @@
w->kcontrol_news[i].private_value;
int val, item;
- val = snd_soc_read(w->codec, e->reg);
+ val = soc_widget_read(w, e->reg);
val = (val >> e->shift_l) & e->mask;
for (item = 0; item < e->max; item++) {
if (val == e->values[item])
@@ -606,6 +638,9 @@
}
list_for_each_entry(path, &widget->sinks, list_source) {
+ if (path->weak)
+ continue;
+
if (path->walked)
continue;
@@ -656,6 +691,9 @@
}
list_for_each_entry(path, &widget->sources, list_sink) {
+ if (path->weak)
+ continue;
+
if (path->walked)
continue;
@@ -681,7 +719,7 @@
else
val = w->off_val;
- snd_soc_update_bits(w->codec, -(w->reg + 1),
+ soc_widget_update_bits(w, -(w->reg + 1),
w->mask << w->shift, val << w->shift);
return 0;
@@ -737,6 +775,9 @@
/* Check if one of our outputs is connected */
list_for_each_entry(path, &w->sinks, list_source) {
+ if (path->weak)
+ continue;
+
if (path->connected &&
!path->connected(path->source, path->sink))
continue;
@@ -885,11 +926,17 @@
}
if (reg >= 0) {
+ /* Any widget will do, they should all be updating the
+ * same register.
+ */
+ w = list_first_entry(pending, struct snd_soc_dapm_widget,
+ power_list);
+
pop_dbg(dapm->dev, card->pop_time,
"pop test : Applying 0x%x/0x%x to %x in %dms\n",
value, mask, reg, card->pop_time);
pop_wait(card->pop_time);
- snd_soc_update_bits(dapm->codec, reg, mask, value);
+ soc_widget_update_bits(w, reg, mask, value);
}
list_for_each_entry(w, pending, power_list) {
@@ -942,7 +989,7 @@
INIT_LIST_HEAD(&pending);
cur_sort = -1;
- cur_subseq = -1;
+ cur_subseq = INT_MIN;
cur_reg = SND_SOC_NOPM;
cur_dapm = NULL;
}
@@ -1041,16 +1088,17 @@
struct snd_soc_dapm_context *d = data;
int ret;
- if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
+ /* If we're off and we're not supposed to be go into STANDBY */
+ if (d->bias_level == SND_SOC_BIAS_OFF &&
+ d->target_bias_level != SND_SOC_BIAS_OFF) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
if (ret != 0)
dev_err(d->dev,
"Failed to turn on bias: %d\n", ret);
}
- /* If we're changing to all on or all off then prepare */
- if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
- (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
+ /* Prepare for a STADDBY->ON or ON->STANDBY transition */
+ if (d->bias_level != d->target_bias_level) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
if (ret != 0)
dev_err(d->dev,
@@ -1067,7 +1115,9 @@
int ret;
/* If we just powered the last thing off drop to standby bias */
- if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
+ if (d->bias_level == SND_SOC_BIAS_PREPARE &&
+ (d->target_bias_level == SND_SOC_BIAS_STANDBY ||
+ d->target_bias_level == SND_SOC_BIAS_OFF)) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
if (ret != 0)
dev_err(d->dev, "Failed to apply standby bias: %d\n",
@@ -1075,14 +1125,16 @@
}
/* If we're in standby and can support bias off then do that */
- if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) {
+ if (d->bias_level == SND_SOC_BIAS_STANDBY &&
+ d->target_bias_level == SND_SOC_BIAS_OFF) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF);
if (ret != 0)
dev_err(d->dev, "Failed to turn off bias: %d\n", ret);
}
/* If we just powered up then move to active bias */
- if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
+ if (d->bias_level == SND_SOC_BIAS_PREPARE &&
+ d->target_bias_level == SND_SOC_BIAS_ON) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON);
if (ret != 0)
dev_err(d->dev, "Failed to apply active bias: %d\n",
@@ -1107,13 +1159,19 @@
LIST_HEAD(up_list);
LIST_HEAD(down_list);
LIST_HEAD(async_domain);
+ enum snd_soc_bias_level bias;
int power;
trace_snd_soc_dapm_start(card);
- list_for_each_entry(d, &card->dapm_list, list)
- if (d->n_widgets || d->codec == NULL)
- d->dev_power = 0;
+ list_for_each_entry(d, &card->dapm_list, list) {
+ if (d->n_widgets || d->codec == NULL) {
+ if (d->idle_bias_off)
+ d->target_bias_level = SND_SOC_BIAS_OFF;
+ else
+ d->target_bias_level = SND_SOC_BIAS_STANDBY;
+ }
+ }
/* Check which widgets we need to power and store them in
* lists indicating if they should be powered up or down.
@@ -1135,8 +1193,27 @@
power = w->power_check(w);
else
power = 1;
- if (power)
- w->dapm->dev_power = 1;
+
+ if (power) {
+ d = w->dapm;
+
+ /* Supplies and micbiases only bring
+ * the context up to STANDBY as unless
+ * something else is active and
+ * passing audio they generally don't
+ * require full power.
+ */
+ switch (w->id) {
+ case snd_soc_dapm_supply:
+ case snd_soc_dapm_micbias:
+ if (d->target_bias_level < SND_SOC_BIAS_STANDBY)
+ d->target_bias_level = SND_SOC_BIAS_STANDBY;
+ break;
+ default:
+ d->target_bias_level = SND_SOC_BIAS_ON;
+ break;
+ }
+ }
if (w->power == power)
continue;
@@ -1160,24 +1237,19 @@
switch (event) {
case SND_SOC_DAPM_STREAM_START:
case SND_SOC_DAPM_STREAM_RESUME:
- dapm->dev_power = 1;
+ dapm->target_bias_level = SND_SOC_BIAS_ON;
break;
case SND_SOC_DAPM_STREAM_STOP:
- dapm->dev_power = !!dapm->codec->active;
+ if (dapm->codec->active)
+ dapm->target_bias_level = SND_SOC_BIAS_ON;
+ else
+ dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
break;
case SND_SOC_DAPM_STREAM_SUSPEND:
- dapm->dev_power = 0;
+ dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
break;
case SND_SOC_DAPM_STREAM_NOP:
- switch (dapm->bias_level) {
- case SND_SOC_BIAS_STANDBY:
- case SND_SOC_BIAS_OFF:
- dapm->dev_power = 0;
- break;
- default:
- dapm->dev_power = 1;
- break;
- }
+ dapm->target_bias_level = dapm->bias_level;
break;
default:
break;
@@ -1185,12 +1257,12 @@
}
/* Force all contexts in the card to the same bias state */
- power = 0;
+ bias = SND_SOC_BIAS_OFF;
list_for_each_entry(d, &card->dapm_list, list)
- if (d->dev_power)
- power = 1;
+ if (d->target_bias_level > bias)
+ bias = d->target_bias_level;
list_for_each_entry(d, &card->dapm_list, list)
- d->dev_power = power;
+ d->target_bias_level = bias;
/* Run all the bias changes in parallel */
@@ -1794,6 +1866,84 @@
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
+static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route)
+{
+ struct snd_soc_dapm_widget *source = dapm_find_widget(dapm,
+ route->source,
+ true);
+ struct snd_soc_dapm_widget *sink = dapm_find_widget(dapm,
+ route->sink,
+ true);
+ struct snd_soc_dapm_path *path;
+ int count = 0;
+
+ if (!source) {
+ dev_err(dapm->dev, "Unable to find source %s for weak route\n",
+ route->source);
+ return -ENODEV;
+ }
+
+ if (!sink) {
+ dev_err(dapm->dev, "Unable to find sink %s for weak route\n",
+ route->sink);
+ return -ENODEV;
+ }
+
+ if (route->control || route->connected)
+ dev_warn(dapm->dev, "Ignoring control for weak route %s->%s\n",
+ route->source, route->sink);
+
+ list_for_each_entry(path, &source->sinks, list_source) {
+ if (path->sink == sink) {
+ path->weak = 1;
+ count++;
+ }
+ }
+
+ if (count == 0)
+ dev_err(dapm->dev, "No path found for weak route %s->%s\n",
+ route->source, route->sink);
+ if (count > 1)
+ dev_warn(dapm->dev, "%d paths found for weak route %s->%s\n",
+ count, route->source, route->sink);
+
+ return 0;
+}
+
+/**
+ * snd_soc_dapm_weak_routes - Mark routes between DAPM widgets as weak
+ * @dapm: DAPM context
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * Mark existing routes matching those specified in the passed array
+ * as being weak, meaning that they are ignored for the purpose of
+ * power decisions. The main intended use case is for sidetone paths
+ * which couple audio between other independent paths if they are both
+ * active in order to make the combination work better at the user
+ * level but which aren't intended to be "used".
+ *
+ * Note that CODEC drivers should not use this as sidetone type paths
+ * can frequently also be used as bypass paths.
+ */
+int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_route *route, int num)
+{
+ int i, err;
+ int ret = 0;
+
+ for (i = 0; i < num; i++) {
+ err = snd_soc_dapm_weak_route(dapm, route);
+ if (err)
+ ret = err;
+ route++;
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
+
/**
* snd_soc_dapm_new_widgets - add new dapm widgets
* @dapm: DAPM context
@@ -1865,7 +2015,7 @@
/* Read the initial power state from the device */
if (w->reg >= 0) {
- val = snd_soc_read(w->codec, w->reg);
+ val = soc_widget_read(w, w->reg);
val &= 1 << w->shift;
if (w->invert)
val = !val;
@@ -2353,6 +2503,7 @@
dapm->n_widgets++;
w->dapm = dapm;
w->codec = dapm->codec;
+ w->platform = dapm->platform;
INIT_LIST_HEAD(&w->sources);
INIT_LIST_HEAD(&w->sinks);
INIT_LIST_HEAD(&w->list);
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
new file mode 100644
index 0000000..cca490c
--- /dev/null
+++ b/sound/soc/soc-io.c
@@ -0,0 +1,396 @@
+/*
+ * soc-io.c -- ASoC register I/O helpers
+ *
+ * Copyright 2009-2011 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include <trace/events/asoc.h>
+
+#ifdef CONFIG_SPI_MASTER
+static int do_spi_write(void *control, const char *data, int len)
+{
+ struct spi_device *spi = control;
+ int ret;
+
+ ret = spi_write(spi, data, len);
+ if (ret < 0)
+ return ret;
+
+ return len;
+}
+#endif
+
+static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value, const void *data, int len)
+{
+ int ret;
+
+ if (!snd_soc_codec_volatile_register(codec, reg) &&
+ reg < codec->driver->reg_cache_size &&
+ !codec->cache_bypass) {
+ ret = snd_soc_cache_write(codec, reg, value);
+ if (ret < 0)
+ return -1;
+ }
+
+ if (codec->cache_only) {
+ codec->cache_sync = 1;
+ return 0;
+ }
+
+ ret = codec->hw_write(codec->control_data, data, len);
+ if (ret == len)
+ return 0;
+ if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ int ret;
+ unsigned int val;
+
+ if (reg >= codec->driver->reg_cache_size ||
+ snd_soc_codec_volatile_register(codec, reg) ||
+ codec->cache_bypass) {
+ if (codec->cache_only)
+ return -1;
+
+ BUG_ON(!codec->hw_read);
+ return codec->hw_read(codec, reg);
+ }
+
+ ret = snd_soc_cache_read(codec, reg, &val);
+ if (ret < 0)
+ return -1;
+ return val;
+}
+
+static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 data;
+
+ data = cpu_to_be16((reg << 12) | (value & 0xffffff));
+
+ return do_hw_write(codec, reg, value, &data, 2);
+}
+
+static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 data;
+
+ data = cpu_to_be16((reg << 9) | (value & 0x1ff));
+
+ return do_hw_write(codec, reg, value, &data, 2);
+}
+
+static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ reg &= 0xff;
+ data[0] = reg;
+ data[1] = value & 0xff;
+
+ return do_hw_write(codec, reg, value, data, 2);
+}
+
+static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+ u16 val = cpu_to_be16(value);
+
+ data[0] = reg;
+ memcpy(&data[1], &val, sizeof(val));
+
+ return do_hw_write(codec, reg, value, data, 3);
+}
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int do_i2c_read(struct snd_soc_codec *codec,
+ void *reg, int reglen,
+ void *data, int datalen)
+{
+ struct i2c_msg xfer[2];
+ int ret;
+ struct i2c_client *client = codec->control_data;
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = reglen;
+ xfer[0].buf = reg;
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = datalen;
+ xfer[1].buf = data;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret == 2)
+ return 0;
+ else if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u8 reg = r;
+ u8 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 1, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_8_8_read_i2c NULL
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u8 reg = r;
+ u16 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 1, &data, 2);
+ if (ret < 0)
+ return 0;
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+#else
+#define snd_soc_8_16_read_i2c NULL
+#endif
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u16 reg = r;
+ u8 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 2, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_16_8_read_i2c NULL
+#endif
+
+static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+ u16 rval = cpu_to_be16(reg);
+
+ memcpy(data, &rval, sizeof(rval));
+ data[2] = value;
+
+ return do_hw_write(codec, reg, value, data, 3);
+}
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ u16 reg = cpu_to_be16(r);
+ u16 data;
+ int ret;
+
+ ret = do_i2c_read(codec, ®, 2, &data, 2);
+ if (ret < 0)
+ return 0;
+ return be16_to_cpu(data);
+}
+#else
+#define snd_soc_16_16_read_i2c NULL
+#endif
+
+static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 data[2];
+
+ data[0] = cpu_to_be16(reg);
+ data[1] = cpu_to_be16(value);
+
+ return do_hw_write(codec, reg, value, data, sizeof(data));
+}
+
+/* Primitive bulk write support for soc-cache. The data pointed to by
+ * `data' needs to already be in the form the hardware expects
+ * including any leading register specific data. Any data written
+ * through this function will not go through the cache as it only
+ * handles writing to volatile or out of bounds registers.
+ */
+static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
+ const void *data, size_t len)
+{
+ int ret;
+
+ /* To ensure that we don't get out of sync with the cache, check
+ * whether the base register is volatile or if we've directly asked
+ * to bypass the cache. Out of bounds registers are considered
+ * volatile.
+ */
+ if (!codec->cache_bypass
+ && !snd_soc_codec_volatile_register(codec, reg)
+ && reg < codec->driver->reg_cache_size)
+ return -EINVAL;
+
+ switch (codec->control_type) {
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+ case SND_SOC_I2C:
+ ret = i2c_master_send(to_i2c_client(codec->dev), data, len);
+ break;
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ case SND_SOC_SPI:
+ ret = spi_write(to_spi_device(codec->dev), data, len);
+ break;
+#endif
+ default:
+ BUG();
+ }
+
+ if (ret == len)
+ return 0;
+ if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+static struct {
+ int addr_bits;
+ int data_bits;
+ int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
+ unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+ unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+} io_types[] = {
+ {
+ .addr_bits = 4, .data_bits = 12,
+ .write = snd_soc_4_12_write,
+ },
+ {
+ .addr_bits = 7, .data_bits = 9,
+ .write = snd_soc_7_9_write,
+ },
+ {
+ .addr_bits = 8, .data_bits = 8,
+ .write = snd_soc_8_8_write,
+ .i2c_read = snd_soc_8_8_read_i2c,
+ },
+ {
+ .addr_bits = 8, .data_bits = 16,
+ .write = snd_soc_8_16_write,
+ .i2c_read = snd_soc_8_16_read_i2c,
+ },
+ {
+ .addr_bits = 16, .data_bits = 8,
+ .write = snd_soc_16_8_write,
+ .i2c_read = snd_soc_16_8_read_i2c,
+ },
+ {
+ .addr_bits = 16, .data_bits = 16,
+ .write = snd_soc_16_16_write,
+ .i2c_read = snd_soc_16_16_read_i2c,
+ },
+};
+
+/**
+ * snd_soc_codec_set_cache_io: Set up standard I/O functions.
+ *
+ * @codec: CODEC to configure.
+ * @addr_bits: Number of bits of register address data.
+ * @data_bits: Number of bits of data per register.
+ * @control: Control bus used.
+ *
+ * Register formats are frequently shared between many I2C and SPI
+ * devices. In order to promote code reuse the ASoC core provides
+ * some standard implementations of CODEC read and write operations
+ * which can be set up using this function.
+ *
+ * The caller is responsible for allocating and initialising the
+ * actual cache.
+ *
+ * Note that at present this code cannot be used by CODECs with
+ * volatile registers.
+ */
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+ int addr_bits, int data_bits,
+ enum snd_soc_control_type control)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(io_types); i++)
+ if (io_types[i].addr_bits == addr_bits &&
+ io_types[i].data_bits == data_bits)
+ break;
+ if (i == ARRAY_SIZE(io_types)) {
+ printk(KERN_ERR
+ "No I/O functions for %d bit address %d bit data\n",
+ addr_bits, data_bits);
+ return -EINVAL;
+ }
+
+ codec->write = io_types[i].write;
+ codec->read = hw_read;
+ codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
+
+ switch (control) {
+ case SND_SOC_I2C:
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+ codec->hw_write = (hw_write_t)i2c_master_send;
+#endif
+ if (io_types[i].i2c_read)
+ codec->hw_read = io_types[i].i2c_read;
+
+ codec->control_data = container_of(codec->dev,
+ struct i2c_client,
+ dev);
+ break;
+
+ case SND_SOC_SPI:
+#ifdef CONFIG_SPI_MASTER
+ codec->hw_write = do_spi_write;
+#endif
+
+ codec->control_data = container_of(codec->dev,
+ struct spi_device,
+ dev);
+ break;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
+
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
new file mode 100644
index 0000000..b575939
--- /dev/null
+++ b/sound/soc/soc-pcm.c
@@ -0,0 +1,639 @@
+/*
+ * soc-pcm.c -- ALSA SoC PCM
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ * Copyright (C) 2010 Slimlogic Ltd.
+ * Copyright (C) 2010 Texas Instruments Inc.
+ *
+ * Authors: Liam Girdwood <lrg@ti.com>
+ * Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+static DEFINE_MUTEX(pcm_mutex);
+
+static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ if (!codec_dai->driver->symmetric_rates &&
+ !cpu_dai->driver->symmetric_rates &&
+ !rtd->dai_link->symmetric_rates)
+ return 0;
+
+ /* This can happen if multiple streams are starting simultaneously -
+ * the second can need to get its constraints before the first has
+ * picked a rate. Complain and allow the application to carry on.
+ */
+ if (!rtd->rate) {
+ dev_warn(&rtd->dev,
+ "Not enforcing symmetric_rates due to race\n");
+ return 0;
+ }
+
+ dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
+
+ ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ rtd->rate, rtd->rate);
+ if (ret < 0) {
+ dev_err(&rtd->dev,
+ "Unable to apply rate symmetry constraint: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+/*
+ * Called by ALSA when a PCM substream is opened, the runtime->hw record is
+ * then initialized and any private data can be allocated. This also calls
+ * startup for the cpu DAI, platform, machine and codec DAI.
+ */
+static int soc_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+ struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver;
+ int ret = 0;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ /* startup the audio subsystem */
+ if (cpu_dai->driver->ops->startup) {
+ ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open interface %s\n",
+ cpu_dai->name);
+ goto out;
+ }
+ }
+
+ if (platform->driver->ops && platform->driver->ops->open) {
+ ret = platform->driver->ops->open(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
+ goto platform_err;
+ }
+ }
+
+ if (codec_dai->driver->ops->startup) {
+ ret = codec_dai->driver->ops->startup(substream, codec_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open codec %s\n",
+ codec_dai->name);
+ goto codec_dai_err;
+ }
+ }
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
+ ret = rtd->dai_link->ops->startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name);
+ goto machine_err;
+ }
+ }
+
+ /* Check that the codec and cpu DAIs are compatible */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw.rate_min =
+ max(codec_dai_drv->playback.rate_min,
+ cpu_dai_drv->playback.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai_drv->playback.rate_max,
+ cpu_dai_drv->playback.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai_drv->playback.channels_min,
+ cpu_dai_drv->playback.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai_drv->playback.channels_max,
+ cpu_dai_drv->playback.channels_max);
+ runtime->hw.formats =
+ codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats;
+ runtime->hw.rates =
+ codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates;
+ if (codec_dai_drv->playback.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= cpu_dai_drv->playback.rates;
+ if (cpu_dai_drv->playback.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= codec_dai_drv->playback.rates;
+ } else {
+ runtime->hw.rate_min =
+ max(codec_dai_drv->capture.rate_min,
+ cpu_dai_drv->capture.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai_drv->capture.rate_max,
+ cpu_dai_drv->capture.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai_drv->capture.channels_min,
+ cpu_dai_drv->capture.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai_drv->capture.channels_max,
+ cpu_dai_drv->capture.channels_max);
+ runtime->hw.formats =
+ codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats;
+ runtime->hw.rates =
+ codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates;
+ if (codec_dai_drv->capture.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= cpu_dai_drv->capture.rates;
+ if (cpu_dai_drv->capture.rates
+ & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+ runtime->hw.rates |= codec_dai_drv->capture.rates;
+ }
+
+ ret = -EINVAL;
+ snd_pcm_limit_hw_rates(runtime);
+ if (!runtime->hw.rates) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
+ codec_dai->name, cpu_dai->name);
+ goto config_err;
+ }
+ if (!runtime->hw.formats) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
+ codec_dai->name, cpu_dai->name);
+ goto config_err;
+ }
+ if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+ runtime->hw.channels_min > runtime->hw.channels_max) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
+ codec_dai->name, cpu_dai->name);
+ goto config_err;
+ }
+
+ /* Symmetry only applies if we've already got an active stream. */
+ if (cpu_dai->active || codec_dai->active) {
+ ret = soc_pcm_apply_symmetry(substream);
+ if (ret != 0)
+ goto config_err;
+ }
+
+ pr_debug("asoc: %s <-> %s info:\n",
+ codec_dai->name, cpu_dai->name);
+ pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
+ pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+ runtime->hw.channels_max);
+ pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+ runtime->hw.rate_max);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active++;
+ codec_dai->playback_active++;
+ } else {
+ cpu_dai->capture_active++;
+ codec_dai->capture_active++;
+ }
+ cpu_dai->active++;
+ codec_dai->active++;
+ rtd->codec->active++;
+ mutex_unlock(&rtd->pcm_mutex);
+ return 0;
+
+config_err:
+ if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
+ rtd->dai_link->ops->shutdown(substream);
+
+machine_err:
+ if (codec_dai->driver->ops->shutdown)
+ codec_dai->driver->ops->shutdown(substream, codec_dai);
+
+codec_dai_err:
+ if (platform->driver->ops && platform->driver->ops->close)
+ platform->driver->ops->close(substream);
+
+platform_err:
+ if (cpu_dai->driver->ops->shutdown)
+ cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+out:
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+/*
+ * Power down the audio subsystem pmdown_time msecs after close is called.
+ * This is to ensure there are no pops or clicks in between any music tracks
+ * due to DAPM power cycling.
+ */
+static void close_delayed_work(struct work_struct *work)
+{
+ struct snd_soc_pcm_runtime *rtd =
+ container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ pr_debug("pop wq checking: %s status: %s waiting: %s\n",
+ codec_dai->driver->playback.stream_name,
+ codec_dai->playback_active ? "active" : "inactive",
+ codec_dai->pop_wait ? "yes" : "no");
+
+ /* are we waiting on this codec DAI stream */
+ if (codec_dai->pop_wait == 1) {
+ codec_dai->pop_wait = 0;
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+ }
+
+ mutex_unlock(&rtd->pcm_mutex);
+}
+
+/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and platform are also
+ * shutdown.
+ */
+static int soc_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active--;
+ codec_dai->playback_active--;
+ } else {
+ cpu_dai->capture_active--;
+ codec_dai->capture_active--;
+ }
+
+ cpu_dai->active--;
+ codec_dai->active--;
+ codec->active--;
+
+ /* Muting the DAC suppresses artifacts caused during digital
+ * shutdown, for example from stopping clocks.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ if (cpu_dai->driver->ops->shutdown)
+ cpu_dai->driver->ops->shutdown(substream, cpu_dai);
+
+ if (codec_dai->driver->ops->shutdown)
+ codec_dai->driver->ops->shutdown(substream, codec_dai);
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown)
+ rtd->dai_link->ops->shutdown(substream);
+
+ if (platform->driver->ops && platform->driver->ops->close)
+ platform->driver->ops->close(substream);
+ cpu_dai->runtime = NULL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* start delayed pop wq here for playback streams */
+ codec_dai->pop_wait = 1;
+ schedule_delayed_work(&rtd->delayed_work,
+ msecs_to_jiffies(rtd->pmdown_time));
+ } else {
+ /* capture streams can be powered down now */
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+ }
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return 0;
+}
+
+/*
+ * Called by ALSA when the PCM substream is prepared, can set format, sample
+ * rate, etc. This function is non atomic and can be called multiple times,
+ * it can refer to the runtime info.
+ */
+static int soc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret = 0;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
+ ret = rtd->dai_link->ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine prepare error\n");
+ goto out;
+ }
+ }
+
+ if (platform->driver->ops && platform->driver->ops->prepare) {
+ ret = platform->driver->ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform prepare error\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->driver->ops->prepare) {
+ ret = codec_dai->driver->ops->prepare(substream, codec_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: codec DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ if (cpu_dai->driver->ops->prepare) {
+ ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: cpu DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ /* cancel any delayed stream shutdown that is pending */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ codec_dai->pop_wait) {
+ codec_dai->pop_wait = 0;
+ cancel_delayed_work(&rtd->delayed_work);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+
+ snd_soc_dai_digital_mute(codec_dai, 0);
+
+out:
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret = 0;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
+ ret = rtd->dai_link->ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine hw_params failed\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->driver->ops->hw_params) {
+ ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't set codec %s hw params\n",
+ codec_dai->name);
+ goto codec_err;
+ }
+ }
+
+ if (cpu_dai->driver->ops->hw_params) {
+ ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: interface %s hw params failed\n",
+ cpu_dai->name);
+ goto interface_err;
+ }
+ }
+
+ if (platform->driver->ops && platform->driver->ops->hw_params) {
+ ret = platform->driver->ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform %s hw params failed\n",
+ platform->name);
+ goto platform_err;
+ }
+ }
+
+ rtd->rate = params_rate(params);
+
+out:
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+
+platform_err:
+ if (cpu_dai->driver->ops->hw_free)
+ cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+
+interface_err:
+ if (codec_dai->driver->ops->hw_free)
+ codec_dai->driver->ops->hw_free(substream, codec_dai);
+
+codec_err:
+ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
+ rtd->dai_link->ops->hw_free(substream);
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+/*
+ * Frees resources allocated by hw_params, can be called multiple times
+ */
+static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+
+ /* apply codec digital mute */
+ if (!codec->active)
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ /* free any machine hw params */
+ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free)
+ rtd->dai_link->ops->hw_free(substream);
+
+ /* free any DMA resources */
+ if (platform->driver->ops && platform->driver->ops->hw_free)
+ platform->driver->ops->hw_free(substream);
+
+ /* now free hw params for the DAIs */
+ if (codec_dai->driver->ops->hw_free)
+ codec_dai->driver->ops->hw_free(substream, codec_dai);
+
+ if (cpu_dai->driver->ops->hw_free)
+ cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return 0;
+}
+
+static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ if (codec_dai->driver->ops->trigger) {
+ ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (platform->driver->ops && platform->driver->ops->trigger) {
+ ret = platform->driver->ops->trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (cpu_dai->driver->ops->trigger) {
+ ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai);
+ if (ret < 0)
+ return ret;
+ }
+ return 0;
+}
+
+/*
+ * soc level wrapper for pointer callback
+ * If cpu_dai, codec_dai, platform driver has the delay callback, than
+ * the runtime->delay will be updated accordingly.
+ */
+static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t offset = 0;
+ snd_pcm_sframes_t delay = 0;
+
+ if (platform->driver->ops && platform->driver->ops->pointer)
+ offset = platform->driver->ops->pointer(substream);
+
+ if (cpu_dai->driver->ops->delay)
+ delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
+
+ if (codec_dai->driver->ops->delay)
+ delay += codec_dai->driver->ops->delay(substream, codec_dai);
+
+ if (platform->driver->delay)
+ delay += platform->driver->delay(substream, codec_dai);
+
+ runtime->delay = delay;
+
+ return offset;
+}
+
+/* ASoC PCM operations */
+static struct snd_pcm_ops soc_pcm_ops = {
+ .open = soc_pcm_open,
+ .close = soc_pcm_close,
+ .hw_params = soc_pcm_hw_params,
+ .hw_free = soc_pcm_hw_free,
+ .prepare = soc_pcm_prepare,
+ .trigger = soc_pcm_trigger,
+ .pointer = soc_pcm_pointer,
+};
+
+/* create a new pcm */
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_pcm *pcm;
+ char new_name[64];
+ int ret = 0, playback = 0, capture = 0;
+
+ /* check client and interface hw capabilities */
+ snprintf(new_name, sizeof(new_name), "%s %s-%d",
+ rtd->dai_link->stream_name, codec_dai->name, num);
+
+ if (codec_dai->driver->playback.channels_min)
+ playback = 1;
+ if (codec_dai->driver->capture.channels_min)
+ capture = 1;
+
+ dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
+ ret = snd_pcm_new(rtd->card->snd_card, new_name,
+ num, playback, capture, &pcm);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+ return ret;
+ }
+
+ /* DAPM dai link stream work */
+ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+
+ rtd->pcm = pcm;
+ pcm->private_data = rtd;
+ if (platform->driver->ops) {
+ soc_pcm_ops.mmap = platform->driver->ops->mmap;
+ soc_pcm_ops.pointer = platform->driver->ops->pointer;
+ soc_pcm_ops.ioctl = platform->driver->ops->ioctl;
+ soc_pcm_ops.copy = platform->driver->ops->copy;
+ soc_pcm_ops.silence = platform->driver->ops->silence;
+ soc_pcm_ops.ack = platform->driver->ops->ack;
+ soc_pcm_ops.page = platform->driver->ops->page;
+ }
+
+ if (playback)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
+
+ if (capture)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
+
+ if (platform->driver->pcm_new) {
+ ret = platform->driver->pcm_new(rtd);
+ if (ret < 0) {
+ pr_err("asoc: platform pcm constructor failed\n");
+ return ret;
+ }
+ }
+
+ pcm->private_free = platform->driver->pcm_free;
+ printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+ cpu_dai->name);
+ return ret;
+}
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 035d39a..c6af1fd 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -12,6 +12,15 @@
Tegra I2S interface. You will also need to select the individual
machine drivers to support below.
+config SND_SOC_TEGRA_SPDIF
+ tristate
+ depends on SND_SOC_TEGRA
+ default m
+ help
+ Say Y or M if you want to add support for the SPDIF interface.
+ You will also need to select the individual machine drivers to support
+ below.
+
config MACH_HAS_SND_SOC_TEGRA_WM8903
bool
help
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index fa6574d..4d943b3 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -2,12 +2,14 @@
snd-soc-tegra-das-objs := tegra_das.o
snd-soc-tegra-pcm-objs := tegra_pcm.o
snd-soc-tegra-i2s-objs := tegra_i2s.o
+snd-soc-tegra-spdif-objs := tegra_spdif.o
snd-soc-tegra-utils-objs += tegra_asoc_utils.o
obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o
obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o
obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o
obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o
+obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o
# Tegra machine Support
snd-soc-tegra-wm8903-objs := tegra_wm8903.o
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 95f03c1..f36b996 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -354,7 +354,6 @@
static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
{
struct tegra_i2s * i2s;
- char clk_name[12]; /* tegra-i2s.0 */
struct resource *mem, *memregion, *dmareq;
int ret;
@@ -389,8 +388,7 @@
}
dev_set_drvdata(&pdev->dev, i2s);
- snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id);
- i2s->clk_i2s = clk_get_sys(clk_name, NULL);
+ i2s->clk_i2s = clk_get(&pdev->dev, NULL);
if (IS_ERR(i2s->clk_i2s)) {
dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
ret = PTR_ERR(i2s->clk_i2s);
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 3c271f9..ff86e5e 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -322,9 +322,11 @@
static u64 tegra_dma_mask = DMA_BIT_MASK(32);
-static int tegra_pcm_new(struct snd_card *card,
- struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
int ret = 0;
if (!card->dev->dma_mask)
diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
new file mode 100644
index 0000000..abe606b
--- /dev/null
+++ b/sound/soc/tegra/tegra_spdif.c
@@ -0,0 +1,371 @@
+/*
+ * tegra_spdif.c - Tegra SPDIF driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2011 - NVIDIA, Inc.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/debugfs.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/seq_file.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <mach/iomap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "tegra_spdif.h"
+
+#define DRV_NAME "tegra-spdif"
+
+static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg,
+ u32 val)
+{
+ __raw_writel(val, spdif->regs + reg);
+}
+
+static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg)
+{
+ return __raw_readl(spdif->regs + reg);
+}
+
+#ifdef CONFIG_DEBUG_FS
+static int tegra_spdif_show(struct seq_file *s, void *unused)
+{
+#define REG(r) { r, #r }
+ static const struct {
+ int offset;
+ const char *name;
+ } regs[] = {
+ REG(TEGRA_SPDIF_CTRL),
+ REG(TEGRA_SPDIF_STATUS),
+ REG(TEGRA_SPDIF_STROBE_CTRL),
+ REG(TEGRA_SPDIF_DATA_FIFO_CSR),
+ REG(TEGRA_SPDIF_CH_STA_RX_A),
+ REG(TEGRA_SPDIF_CH_STA_RX_B),
+ REG(TEGRA_SPDIF_CH_STA_RX_C),
+ REG(TEGRA_SPDIF_CH_STA_RX_D),
+ REG(TEGRA_SPDIF_CH_STA_RX_E),
+ REG(TEGRA_SPDIF_CH_STA_RX_F),
+ REG(TEGRA_SPDIF_CH_STA_TX_A),
+ REG(TEGRA_SPDIF_CH_STA_TX_B),
+ REG(TEGRA_SPDIF_CH_STA_TX_C),
+ REG(TEGRA_SPDIF_CH_STA_TX_D),
+ REG(TEGRA_SPDIF_CH_STA_TX_E),
+ REG(TEGRA_SPDIF_CH_STA_TX_F),
+ };
+#undef REG
+
+ struct tegra_spdif *spdif = s->private;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(regs); i++) {
+ u32 val = tegra_spdif_read(spdif, regs[i].offset);
+ seq_printf(s, "%s = %08x\n", regs[i].name, val);
+ }
+
+ return 0;
+}
+
+static int tegra_spdif_debug_open(struct inode *inode, struct file *file)
+{
+ return single_open(file, tegra_spdif_show, inode->i_private);
+}
+
+static const struct file_operations tegra_spdif_debug_fops = {
+ .open = tegra_spdif_debug_open,
+ .read = seq_read,
+ .llseek = seq_lseek,
+ .release = single_release,
+};
+
+static void tegra_spdif_debug_add(struct tegra_spdif *spdif)
+{
+ spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO,
+ snd_soc_debugfs_root, spdif,
+ &tegra_spdif_debug_fops);
+}
+
+static void tegra_spdif_debug_remove(struct tegra_spdif *spdif)
+{
+ if (spdif->debug)
+ debugfs_remove(spdif->debug);
+}
+#else
+static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif)
+{
+}
+
+static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif)
+{
+}
+#endif
+
+static int tegra_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct device *dev = substream->pcm->card->dev;
+ struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+ int ret, srate, spdifclock;
+
+ spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK;
+ spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK;
+ spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ srate = params_rate(params);
+ switch (params_rate(params)) {
+ case 32000:
+ spdifclock = 4096000;
+ break;
+ case 44100:
+ spdifclock = 5644800;
+ break;
+ case 48000:
+ spdifclock = 6144000;
+ break;
+ case 88200:
+ spdifclock = 11289600;
+ break;
+ case 96000:
+ spdifclock = 12288000;
+ break;
+ case 176400:
+ spdifclock = 22579200;
+ break;
+ case 192000:
+ spdifclock = 24576000;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = clk_set_rate(spdif->clk_spdif_out, spdifclock);
+ if (ret) {
+ dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void tegra_spdif_start_playback(struct tegra_spdif *spdif)
+{
+ spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN;
+ tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl);
+}
+
+static void tegra_spdif_stop_playback(struct tegra_spdif *spdif)
+{
+ spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN;
+ tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl);
+}
+
+static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (!spdif->clk_refs)
+ clk_enable(spdif->clk_spdif_out);
+ spdif->clk_refs++;
+ tegra_spdif_start_playback(spdif);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ tegra_spdif_stop_playback(spdif);
+ spdif->clk_refs--;
+ if (!spdif->clk_refs)
+ clk_disable(spdif->clk_spdif_out);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int tegra_spdif_probe(struct snd_soc_dai *dai)
+{
+ struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+ dai->capture_dma_data = NULL;
+ dai->playback_dma_data = &spdif->playback_dma_data;
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops tegra_spdif_dai_ops = {
+ .hw_params = tegra_spdif_hw_params,
+ .trigger = tegra_spdif_trigger,
+};
+
+struct snd_soc_dai_driver tegra_spdif_dai = {
+ .name = DRV_NAME,
+ .probe = tegra_spdif_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &tegra_spdif_dai_ops,
+};
+
+static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev)
+{
+ struct tegra_spdif *spdif;
+ struct resource *mem, *memregion, *dmareq;
+ int ret;
+
+ spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL);
+ if (!spdif) {
+ dev_err(&pdev->dev, "Can't allocate tegra_spdif\n");
+ ret = -ENOMEM;
+ goto exit;
+ }
+ dev_set_drvdata(&pdev->dev, spdif);
+
+ spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out");
+ if (IS_ERR(spdif->clk_spdif_out)) {
+ pr_err("Can't retrieve spdif clock\n");
+ ret = PTR_ERR(spdif->clk_spdif_out);
+ goto err_free;
+ }
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem) {
+ dev_err(&pdev->dev, "No memory resource\n");
+ ret = -ENODEV;
+ goto err_clk_put;
+ }
+
+ dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!dmareq) {
+ dev_err(&pdev->dev, "No DMA resource\n");
+ ret = -ENODEV;
+ goto err_clk_put;
+ }
+
+ memregion = request_mem_region(mem->start, resource_size(mem),
+ DRV_NAME);
+ if (!memregion) {
+ dev_err(&pdev->dev, "Memory region already claimed\n");
+ ret = -EBUSY;
+ goto err_clk_put;
+ }
+
+ spdif->regs = ioremap(mem->start, resource_size(mem));
+ if (!spdif->regs) {
+ dev_err(&pdev->dev, "ioremap failed\n");
+ ret = -ENOMEM;
+ goto err_release;
+ }
+
+ spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT;
+ spdif->playback_dma_data.wrap = 4;
+ spdif->playback_dma_data.width = 32;
+ spdif->playback_dma_data.req_sel = dmareq->start;
+
+ ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai);
+ if (ret) {
+ dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
+ ret = -ENOMEM;
+ goto err_unmap;
+ }
+
+ tegra_spdif_debug_add(spdif);
+
+ return 0;
+
+err_unmap:
+ iounmap(spdif->regs);
+err_release:
+ release_mem_region(mem->start, resource_size(mem));
+err_clk_put:
+ clk_put(spdif->clk_spdif_out);
+err_free:
+ kfree(spdif);
+exit:
+ return ret;
+}
+
+static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev)
+{
+ struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev);
+ struct resource *res;
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ tegra_spdif_debug_remove(spdif);
+
+ iounmap(spdif->regs);
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ release_mem_region(res->start, resource_size(res));
+
+ clk_put(spdif->clk_spdif_out);
+
+ kfree(spdif);
+
+ return 0;
+}
+
+static struct platform_driver tegra_spdif_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+ .probe = tegra_spdif_platform_probe,
+ .remove = __devexit_p(tegra_spdif_platform_remove),
+};
+
+static int __init snd_tegra_spdif_init(void)
+{
+ return platform_driver_register(&tegra_spdif_driver);
+}
+module_init(snd_tegra_spdif_init);
+
+static void __exit snd_tegra_spdif_exit(void)
+{
+ platform_driver_unregister(&tegra_spdif_driver);
+}
+module_exit(snd_tegra_spdif_exit);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra SPDIF ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h
new file mode 100644
index 0000000..2e03db4
--- /dev/null
+++ b/sound/soc/tegra/tegra_spdif.h
@@ -0,0 +1,473 @@
+/*
+ * tegra_spdif.h - Definitions for Tegra SPDIF driver
+ *
+ * Author: Stephen Warren <swarren@nvidia.com>
+ * Copyright (C) 2011 - NVIDIA, Inc.
+ *
+ * Based on code copyright/by:
+ * Copyright (c) 2008-2009, NVIDIA Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TEGRA_SPDIF_H__
+#define __TEGRA_SPDIF_H__
+
+#include "tegra_pcm.h"
+
+/* Offsets from TEGRA_SPDIF_BASE */
+
+#define TEGRA_SPDIF_CTRL 0x0
+#define TEGRA_SPDIF_STATUS 0x4
+#define TEGRA_SPDIF_STROBE_CTRL 0x8
+#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C
+#define TEGRA_SPDIF_DATA_OUT 0x40
+#define TEGRA_SPDIF_DATA_IN 0x80
+#define TEGRA_SPDIF_CH_STA_RX_A 0x100
+#define TEGRA_SPDIF_CH_STA_RX_B 0x104
+#define TEGRA_SPDIF_CH_STA_RX_C 0x108
+#define TEGRA_SPDIF_CH_STA_RX_D 0x10C
+#define TEGRA_SPDIF_CH_STA_RX_E 0x110
+#define TEGRA_SPDIF_CH_STA_RX_F 0x114
+#define TEGRA_SPDIF_CH_STA_TX_A 0x140
+#define TEGRA_SPDIF_CH_STA_TX_B 0x144
+#define TEGRA_SPDIF_CH_STA_TX_C 0x148
+#define TEGRA_SPDIF_CH_STA_TX_D 0x14C
+#define TEGRA_SPDIF_CH_STA_TX_E 0x150
+#define TEGRA_SPDIF_CH_STA_TX_F 0x154
+#define TEGRA_SPDIF_USR_STA_RX_A 0x180
+#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0
+
+/* Fields in TEGRA_SPDIF_CTRL */
+
+/* Start capturing from 0=right, 1=left channel */
+#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30)
+
+/* SPDIF receiver(RX) enable */
+#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29)
+
+/* SPDIF Transmitter(TX) enable */
+#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28)
+
+/* Transmit Channel status */
+#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27)
+
+/* Transmit user Data */
+#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26)
+
+/* Interrupt on transmit error */
+#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25)
+
+/* Interrupt on receive error */
+#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24)
+
+/* Interrupt on invalid preamble */
+#define TEGRA_SPDIF_CTRL_IE_P (1 << 23)
+
+/* Interrupt on "B" preamble */
+#define TEGRA_SPDIF_CTRL_IE_B (1 << 22)
+
+/* Interrupt when block of channel status received */
+#define TEGRA_SPDIF_CTRL_IE_C (1 << 21)
+
+/* Interrupt when a valid information unit (IU) is received */
+#define TEGRA_SPDIF_CTRL_IE_U (1 << 20)
+
+/* Interrupt when RX user FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19)
+
+/* Interrupt when TX user FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18)
+
+/* Interrupt when RX data FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17)
+
+/* Interrupt when TX data FIFO attention level is reached */
+#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16)
+
+/* Loopback test mode enable */
+#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15)
+
+/*
+ * Pack data mode:
+ * 0 = Single data (16 bit needs to be padded to match the
+ * interface data bit size).
+ * 1 = Packeted left/right channel data into a single word.
+ */
+#define TEGRA_SPDIF_CTRL_PACK (1 << 14)
+
+/*
+ * 00 = 16bit data
+ * 01 = 20bit data
+ * 10 = 24bit data
+ * 11 = raw data
+ */
+#define TEGRA_SPDIF_BIT_MODE_16BIT 0
+#define TEGRA_SPDIF_BIT_MODE_20BIT 1
+#define TEGRA_SPDIF_BIT_MODE_24BIT 2
+#define TEGRA_SPDIF_BIT_MODE_RAW 3
+
+#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12
+#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT)
+
+/* Fields in TEGRA_SPDIF_STATUS */
+
+/*
+ * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must
+ * write a 1 to the corresponding bit location to clear the status.
+ */
+
+/*
+ * Receiver(RX) shifter is busy receiving data.
+ * This bit is asserted when the receiver first locked onto the
+ * preamble of the data stream after RX_EN is asserted. This bit is
+ * deasserted when either,
+ * (a) the end of a frame is reached after RX_EN is deeasserted, or
+ * (b) the SPDIF data stream becomes inactive.
+ */
+#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29)
+
+/*
+ * Transmitter(TX) shifter is busy transmitting data.
+ * This bit is asserted when TX_EN is asserted.
+ * This bit is deasserted when the end of a frame is reached after
+ * TX_EN is deasserted.
+ */
+#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28)
+
+/*
+ * TX is busy shifting out channel status.
+ * This bit is asserted when both TX_EN and TC_EN are asserted and
+ * data from CH_STA_TX_A register is loaded into the internal shifter.
+ * This bit is deasserted when either,
+ * (a) the end of a frame is reached after TX_EN is deasserted, or
+ * (b) CH_STA_TX_F register is loaded into the internal shifter.
+ */
+#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27)
+
+/*
+ * TX User data FIFO busy.
+ * This bit is asserted when TX_EN and TXU_EN are asserted and
+ * there's data in the TX user FIFO. This bit is deassert when either,
+ * (a) the end of a frame is reached after TX_EN is deasserted, or
+ * (b) there's no data left in the TX user FIFO.
+ */
+#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26)
+
+/* TX FIFO Underrun error status */
+#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25)
+
+/* RX FIFO Overrun error status */
+#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24)
+
+/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */
+#define TEGRA_SPDIF_STATUS_IS_P (1 << 23)
+
+/* B-preamble detection status: 0=not detected, 1=B-preamble detected */
+#define TEGRA_SPDIF_STATUS_IS_B (1 << 22)
+
+/*
+ * RX channel block data receive status:
+ * 0=entire block not recieved yet.
+ * 1=received entire block of channel status,
+ */
+#define TEGRA_SPDIF_STATUS_IS_C (1 << 21)
+
+/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */
+#define TEGRA_SPDIF_STATUS_IS_U (1 << 20)
+
+/*
+ * RX User FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19)
+
+/*
+ * TX User FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18)
+
+/*
+ * RX Data FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17)
+
+/*
+ * TX Data FIFO Status:
+ * 1=attention level reached, 0=attention level not reached.
+ */
+#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16)
+
+/* Fields in TEGRA_SPDIF_STROBE_CTRL */
+
+/*
+ * Indicates the approximate number of detected SPDIFIN clocks within a
+ * bi-phase period.
+ */
+#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16
+#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT)
+
+/* Data strobe mode: 0=Auto-locked 1=Manual locked */
+#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15)
+
+/*
+ * Manual data strobe time within the bi-phase clock period (in terms of
+ * the number of over-sampling clocks).
+ */
+#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8
+#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT)
+
+/*
+ * Manual SPDIFIN bi-phase clock period (in terms of the number of
+ * over-sampling clocks).
+ */
+#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0
+#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT)
+
+/* Fields in SPDIF_DATA_FIFO_CSR */
+
+/* Clear Receiver User FIFO (RX USR.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31)
+
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2
+#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3
+
+/* RU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \
+ (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT)
+
+/* Number of RX USR.FIFO levels with valid data. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT)
+
+/* Clear Transmitter User FIFO (TX USR.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23)
+
+/* TU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \
+ (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT)
+
+/* Number of TX USR.FIFO levels that could be filled. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT)
+
+/* Clear Receiver Data FIFO (RX DATA.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15)
+
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2
+#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3
+
+/* RU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \
+ (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT)
+
+/* Number of RX DATA.FIFO levels with valid data. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8
+#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT)
+
+/* Clear Transmitter Data FIFO (TX DATA.FIFO) */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7)
+
+/* TU FIFO attention level */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \
+ (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \
+ (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT)
+
+/* Number of TX DATA.FIFO levels that could be filled. */
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0
+#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT)
+
+/* Fields in TEGRA_SPDIF_DATA_OUT */
+
+/*
+ * This register has 5 different formats:
+ * 16-bit (BIT_MODE=00, PACK=0)
+ * 20-bit (BIT_MODE=01, PACK=0)
+ * 24-bit (BIT_MODE=10, PACK=0)
+ * raw (BIT_MODE=11, PACK=0)
+ * 16-bit packed (BIT_MODE=00, PACK=1)
+ */
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0
+#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0
+#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31)
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30)
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29)
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0
+#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT)
+
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0
+#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT)
+
+/* Fields in TEGRA_SPDIF_DATA_IN */
+
+/*
+ * This register has 5 different formats:
+ * 16-bit (BIT_MODE=00, PACK=0)
+ * 20-bit (BIT_MODE=01, PACK=0)
+ * 24-bit (BIT_MODE=10, PACK=0)
+ * raw (BIT_MODE=11, PACK=0)
+ * 16-bit packed (BIT_MODE=00, PACK=1)
+ *
+ * Bits 31:24 are common to all modes except 16-bit packed
+ */
+
+#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31)
+#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30)
+#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29)
+#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24
+#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0
+#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0
+#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0
+#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0
+#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT)
+
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0
+#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT)
+
+/* Fields in TEGRA_SPDIF_CH_STA_RX_A */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_B */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_C */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_D */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_E */
+/* Fields in TEGRA_SPDIF_CH_STA_RX_F */
+
+/*
+ * The 6-word receive channel data page buffer holds a block (192 frames) of
+ * channel status information. The order of receive is from LSB to MSB
+ * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A.
+ */
+
+/* Fields in TEGRA_SPDIF_CH_STA_TX_A */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_B */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_C */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_D */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_E */
+/* Fields in TEGRA_SPDIF_CH_STA_TX_F */
+
+/*
+ * The 6-word transmit channel data page buffer holds a block (192 frames) of
+ * channel status information. The order of transmission is from LSB to MSB
+ * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A.
+ */
+
+/* Fields in TEGRA_SPDIF_USR_STA_RX_A */
+
+/*
+ * This 4-word deep FIFO receives user FIFO field information. The order of
+ * receive is from LSB to MSB bit.
+ */
+
+/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */
+
+/*
+ * This 4-word deep FIFO transmits user FIFO field information. The order of
+ * transmission is from LSB to MSB bit.
+ */
+
+struct tegra_spdif {
+ struct clk *clk_spdif_out;
+ int clk_refs;
+ struct tegra_pcm_dma_params capture_dma_data;
+ struct tegra_pcm_dma_params playback_dma_data;
+ void __iomem *regs;
+ struct dentry *debug;
+ u32 reg_ctrl;
+};
+
+#endif
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0d6738a..a42e9ac 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -267,7 +267,7 @@
}
machine->gpio_requested |= GPIO_HP_MUTE;
- gpio_direction_output(pdata->gpio_hp_mute, 0);
+ gpio_direction_output(pdata->gpio_hp_mute, 1);
}
if (gpio_is_valid(pdata->gpio_int_mic_en)) {
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index f4aa4e0..34aa972 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -288,9 +288,10 @@
snd_pcm_lib_preallocate_free_for_all(pcm);
}
-static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
+static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_pcm *pcm = rtd->pcm;
struct platform_device *pdev = to_platform_device(dai->platform->dev);
struct txx9aclc_soc_device *dev;
struct resource *r;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 220c616..781d9e6 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -433,9 +433,10 @@
* only at the first time. the successive calls of this function will
* append the pcm interface to the corresponding card.
*/
-static void *snd_usb_audio_probe(struct usb_device *dev,
- struct usb_interface *intf,
- const struct usb_device_id *usb_id)
+static struct snd_usb_audio *
+snd_usb_audio_probe(struct usb_device *dev,
+ struct usb_interface *intf,
+ const struct usb_device_id *usb_id)
{
const struct snd_usb_audio_quirk *quirk = (const struct snd_usb_audio_quirk *)usb_id->driver_info;
int i, err;
@@ -540,16 +541,15 @@
* we need to take care of counter, since disconnection can be called also
* many times as well as usb_audio_probe().
*/
-static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
+static void snd_usb_audio_disconnect(struct usb_device *dev,
+ struct snd_usb_audio *chip)
{
- struct snd_usb_audio *chip;
struct snd_card *card;
struct list_head *p;
- if (ptr == (void *)-1L)
+ if (chip == (void *)-1L)
return;
- chip = ptr;
card = chip->card;
mutex_lock(®ister_mutex);
mutex_lock(&chip->shutdown_mutex);
@@ -585,7 +585,7 @@
static int usb_audio_probe(struct usb_interface *intf,
const struct usb_device_id *id)
{
- void *chip;
+ struct snd_usb_audio *chip;
chip = snd_usb_audio_probe(interface_to_usbdev(intf), intf, id);
if (chip) {
usb_set_intfdata(intf, chip);
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index b0ef9f5..7c0d21e 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -408,6 +408,8 @@
/* doesn't set the sample rate attribute, but supports it */
fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE;
break;
+ case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */
+ case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */
case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
an older model 77d:223) */
diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index fb5d68f..67bec76 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -645,7 +645,7 @@
err = snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIOD_TIME,
1500000 / ua->packets_per_second,
- 8192000);
+ UINT_MAX);
if (err < 0)
return err;
err = snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 0b2ae8e..dba0b7f 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1677,6 +1677,36 @@
}
}
},
+{
+ USB_DEVICE(0x0582, 0x011e),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "BOSS", */
+ /* .product_name = "BR-800", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
/* Guillemot devices */
{
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 090e193..77762c9 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -369,6 +369,30 @@
return 0;
}
+static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev)
+{
+ int err;
+
+ if (dev->actconfig->desc.bConfigurationValue == 1) {
+ snd_printk(KERN_INFO "usb-audio: "
+ "Fast Track Pro switching to config #2\n");
+ /* This function has to be available by the usb core module.
+ * if it is not avialable the boot quirk has to be left out
+ * and the configuration has to be set by udev or hotplug
+ * rules
+ */
+ err = usb_driver_set_configuration(dev, 2);
+ if (err < 0) {
+ snd_printdd("error usb_driver_set_configuration: %d\n",
+ err);
+ return -ENODEV;
+ }
+ } else
+ snd_printk(KERN_INFO "usb-audio: Fast Track Pro config OK\n");
+
+ return 0;
+}
+
/*
* C-Media CM106/CM106+ have four 16-bit internal registers that are nicely
* documented in the device's data sheet.
@@ -471,16 +495,49 @@
/*
* Setup quirks
*/
-#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */
-#define AUDIOPHILE_SET_DTS 0x02 /* if set, enable DTS Digital Output */
-#define AUDIOPHILE_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
-#define AUDIOPHILE_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */
-#define AUDIOPHILE_SET_DI 0x10 /* if set, enable Digital Input */
-#define AUDIOPHILE_SET_MASK 0x1F /* bit mask for setup value */
-#define AUDIOPHILE_SET_24B_48K_DI 0x19 /* value for 24bits+48KHz+Digital Input */
-#define AUDIOPHILE_SET_24B_48K_NOTDI 0x09 /* value for 24bits+48KHz+No Digital Input */
-#define AUDIOPHILE_SET_16B_48K_DI 0x11 /* value for 16bits+48KHz+Digital Input */
-#define AUDIOPHILE_SET_16B_48K_NOTDI 0x01 /* value for 16bits+48KHz+No Digital Input */
+#define MAUDIO_SET 0x01 /* parse device_setup */
+#define MAUDIO_SET_COMPATIBLE 0x80 /* use only "win-compatible" interfaces */
+#define MAUDIO_SET_DTS 0x02 /* enable DTS Digital Output */
+#define MAUDIO_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
+#define MAUDIO_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */
+#define MAUDIO_SET_DI 0x10 /* enable Digital Input */
+#define MAUDIO_SET_MASK 0x1f /* bit mask for setup value */
+#define MAUDIO_SET_24B_48K_DI 0x19 /* 24bits+48KHz+Digital Input */
+#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48KHz+No Digital Input */
+#define MAUDIO_SET_16B_48K_DI 0x11 /* 16bits+48KHz+Digital Input */
+#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48KHz+No Digital Input */
+
+static int quattro_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno)
+{
+ /* Reset ALL ifaces to 0 altsetting.
+ * Call it for every possible altsetting of every interface.
+ */
+ usb_set_interface(chip->dev, iface, 0);
+ if (chip->setup & MAUDIO_SET) {
+ if (chip->setup & MAUDIO_SET_COMPATIBLE) {
+ if (iface != 1 && iface != 2)
+ return 1; /* skip all interfaces but 1 and 2 */
+ } else {
+ unsigned int mask;
+ if (iface == 1 || iface == 2)
+ return 1; /* skip interfaces 1 and 2 */
+ if ((chip->setup & MAUDIO_SET_96K) && altno != 1)
+ return 1; /* skip this altsetting */
+ mask = chip->setup & MAUDIO_SET_MASK;
+ if (mask == MAUDIO_SET_24B_48K_DI && altno != 2)
+ return 1; /* skip this altsetting */
+ if (mask == MAUDIO_SET_24B_48K_NOTDI && altno != 3)
+ return 1; /* skip this altsetting */
+ if (mask == MAUDIO_SET_16B_48K_NOTDI && altno != 4)
+ return 1; /* skip this altsetting */
+ }
+ }
+ snd_printdd(KERN_INFO
+ "using altsetting %d for interface %d config %d\n",
+ altno, iface, chip->setup);
+ return 0; /* keep this altsetting */
+}
static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
int iface,
@@ -491,30 +548,65 @@
*/
usb_set_interface(chip->dev, iface, 0);
- if (chip->setup & AUDIOPHILE_SET) {
- if ((chip->setup & AUDIOPHILE_SET_DTS)
- && altno != 6)
+ if (chip->setup & MAUDIO_SET) {
+ unsigned int mask;
+ if ((chip->setup & MAUDIO_SET_DTS) && altno != 6)
return 1; /* skip this altsetting */
- if ((chip->setup & AUDIOPHILE_SET_96K)
- && altno != 1)
+ if ((chip->setup & MAUDIO_SET_96K) && altno != 1)
return 1; /* skip this altsetting */
- if ((chip->setup & AUDIOPHILE_SET_MASK) ==
- AUDIOPHILE_SET_24B_48K_DI && altno != 2)
+ mask = chip->setup & MAUDIO_SET_MASK;
+ if (mask == MAUDIO_SET_24B_48K_DI && altno != 2)
return 1; /* skip this altsetting */
- if ((chip->setup & AUDIOPHILE_SET_MASK) ==
- AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3)
+ if (mask == MAUDIO_SET_24B_48K_NOTDI && altno != 3)
return 1; /* skip this altsetting */
- if ((chip->setup & AUDIOPHILE_SET_MASK) ==
- AUDIOPHILE_SET_16B_48K_DI && altno != 4)
+ if (mask == MAUDIO_SET_16B_48K_DI && altno != 4)
return 1; /* skip this altsetting */
- if ((chip->setup & AUDIOPHILE_SET_MASK) ==
- AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5)
+ if (mask == MAUDIO_SET_16B_48K_NOTDI && altno != 5)
return 1; /* skip this altsetting */
}
return 0; /* keep this altsetting */
}
+
+static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno)
+{
+ /* Reset ALL ifaces to 0 altsetting.
+ * Call it for every possible altsetting of every interface.
+ */
+ usb_set_interface(chip->dev, iface, 0);
+
+ /* possible configuration where both inputs and only one output is
+ *used is not supported by the current setup
+ */
+ if (chip->setup & (MAUDIO_SET | MAUDIO_SET_24B)) {
+ if (chip->setup & MAUDIO_SET_96K) {
+ if (altno != 3 && altno != 6)
+ return 1;
+ } else if (chip->setup & MAUDIO_SET_DI) {
+ if (iface == 4)
+ return 1; /* no analog input */
+ if (altno != 2 && altno != 5)
+ return 1; /* enable only altsets 2 and 5 */
+ } else {
+ if (iface == 5)
+ return 1; /* disable digialt input */
+ if (altno != 2 && altno != 5)
+ return 1; /* enalbe only altsets 2 and 5 */
+ }
+ } else {
+ /* keep only 16-Bit mode */
+ if (altno != 1)
+ return 1;
+ }
+
+ snd_printdd(KERN_INFO
+ "using altsetting %d for interface %d config %d\n",
+ altno, iface, chip->setup);
+ return 0; /* keep this altsetting */
+}
+
int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
int iface,
int altno)
@@ -522,6 +614,12 @@
/* audiophile usb: skip altsets incompatible with device_setup */
if (chip->usb_id == USB_ID(0x0763, 0x2003))
return audiophile_skip_setting_quirk(chip, iface, altno);
+ /* quattro usb: skip altsets incompatible with device_setup */
+ if (chip->usb_id == USB_ID(0x0763, 0x2001))
+ return quattro_skip_setting_quirk(chip, iface, altno);
+ /* fasttrackpro usb: skip altsets incompatible with device_setup */
+ if (chip->usb_id == USB_ID(0x0763, 0x2012))
+ return fasttrackpro_skip_setting_quirk(chip, iface, altno);
return 0;
}
@@ -560,6 +658,8 @@
case USB_ID(0x17cc, 0x1010): /* Traktor Audio 6 */
case USB_ID(0x17cc, 0x1020): /* Traktor Audio 10 */
return snd_usb_nativeinstruments_boot_quirk(dev);
+ case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */
+ return snd_usb_fasttrackpro_boot_quirk(dev);
}
return 0;
@@ -570,15 +670,24 @@
*/
int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp)
{
+ /* it depends on altsetting wether the device is big-endian or not */
switch (chip->usb_id) {
case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */
- if (fp->endpoint & USB_DIR_IN)
+ if (fp->altsetting == 2 || fp->altsetting == 3 ||
+ fp->altsetting == 5 || fp->altsetting == 6)
return 1;
break;
case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
if (chip->setup == 0x00 ||
- fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3)
+ fp->altsetting == 1 || fp->altsetting == 2 ||
+ fp->altsetting == 3)
return 1;
+ break;
+ case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro */
+ if (fp->altsetting == 2 || fp->altsetting == 3 ||
+ fp->altsetting == 5 || fp->altsetting == 6)
+ return 1;
+ break;
}
return 0;
}