[ALSA] hda-intel - Add POWER_SAVE option
Added CONFIG_SND_HDA_POWER_SAVE kconfig. It's an experimental option
to achieve an aggressive power-saving. With this option, the driver
will turn on/off the power of each codec and controller chip dynamically
on demand.
The patch introduces a new module option 'power_save'. It specifies
the second of time-out for automatic power-down. As default, it's
10 seconds. Setting 0 means to suppress the power-saving feature.
The codec may have analog-input loopbacks, which are usually represented
by mixer elements such as 'Mic Playback Switch' or 'CD Playback Switch'.
When these are on, we cannot turn off the mixer and the codec chip has
to be kept on. For bookkeeping these states, a new codec-callback is
introduced.
For the bus-controller side, a new callback pm_notify is introduced,
which can be used to turn on/off the contoller appropriately.
Note that this power-saving might cause slight click-noise at
power-on/off. Also, it might take some time to wake up the codec, and
might even drop some tones at the very beginning. This seems to be the
side-effect of turning off the controller chip.
This turn-off of the controller can be disabled by undefining
HDA_POWER_SAVE_RESET_CONTOLLER in hda_intel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index ff7a689..9554140 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -581,6 +581,14 @@
Say Y here to enable the generic HD-audio codec parser
in snd-hda-intel driver.
+config SND_HDA_POWER_SAVE
+ bool "Aggressive power-saving on HD-audio"
+ depends on SND_HDA_INTEL && EXPERIMENTAL
+ help
+ Say Y here to enable more aggressive power-saving mode on
+ HD-audio driver. The power-saving timeout can be configured
+ via power_save option or over sysfs on-the-fly.
+
config SND_HDSP
tristate "RME Hammerfall DSP Audio"
depends on SND
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0435293..9a3b728 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -33,6 +33,13 @@
#include "hda_local.h"
#include <sound/hda_hwdep.h>
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* define this option here to hide as static */
+static int power_save = 10;
+module_param(power_save, int, 0644);
+MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
+ "(in second, 0 = disable).");
+#endif
/*
* vendor / preset table
@@ -60,6 +67,13 @@
#include "hda_patch.h"
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_power_work(struct work_struct *work);
+static void hda_keep_power_on(struct hda_codec *codec);
+#else
+static inline void hda_keep_power_on(struct hda_codec *codec) {}
+#endif
+
/**
* snd_hda_codec_read - send a command and get the response
* @codec: the HDA codec
@@ -77,12 +91,14 @@
unsigned int verb, unsigned int parm)
{
unsigned int res;
+ snd_hda_power_up(codec);
mutex_lock(&codec->bus->cmd_mutex);
if (!codec->bus->ops.command(codec, nid, direct, verb, parm))
res = codec->bus->ops.get_response(codec);
else
res = (unsigned int)-1;
mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
return res;
}
@@ -102,9 +118,11 @@
unsigned int verb, unsigned int parm)
{
int err;
+ snd_hda_power_up(codec);
mutex_lock(&codec->bus->cmd_mutex);
err = codec->bus->ops.command(codec, nid, direct, verb, parm);
mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
return err;
}
@@ -505,6 +523,9 @@
{
if (!codec)
return;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+#endif
list_del(&codec->list);
codec->bus->caddr_tbl[codec->addr] = NULL;
if (codec->patch_ops.free)
@@ -551,6 +572,15 @@
init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
+ /* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
+ * the caller has to power down appropriatley after initialization
+ * phase.
+ */
+ hda_keep_power_on(codec);
+#endif
+
list_add_tail(&codec->list, &bus->codec_list);
bus->caddr_tbl[codec_addr] = codec;
@@ -855,7 +885,7 @@
return ret;
}
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
/* resume the all amp commands from the cache */
void snd_hda_codec_resume_amp(struct hda_codec *codec)
{
@@ -879,7 +909,7 @@
}
}
}
-#endif /* CONFIG_PM */
+#endif /* SND_HDA_NEEDS_RESUME */
/*
* AMP control callbacks
@@ -945,6 +975,7 @@
long *valp = ucontrol->value.integer.value;
int change = 0;
+ snd_hda_power_up(codec);
if (chs & 1) {
change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
0x7f, *valp);
@@ -953,6 +984,7 @@
if (chs & 2)
change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
0x7f, *valp);
+ snd_hda_power_down(codec);
return change;
}
@@ -1025,6 +1057,7 @@
long *valp = ucontrol->value.integer.value;
int change = 0;
+ snd_hda_power_up(codec);
if (chs & 1) {
change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
HDA_AMP_MUTE,
@@ -1035,7 +1068,11 @@
change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
HDA_AMP_MUTE,
*valp ? 0 : HDA_AMP_MUTE);
-
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (codec->patch_ops.check_power_status)
+ codec->patch_ops.check_power_status(codec, nid);
+#endif
+ snd_hda_power_down(codec);
return change;
}
@@ -1502,7 +1539,7 @@
return 0;
}
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
/*
* command cache
*/
@@ -1528,6 +1565,7 @@
int direct, unsigned int verb, unsigned int parm)
{
int err;
+ snd_hda_power_up(codec);
mutex_lock(&codec->bus->cmd_mutex);
err = codec->bus->ops.command(codec, nid, direct, verb, parm);
if (!err) {
@@ -1538,6 +1576,7 @@
c->val = parm;
}
mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
return err;
}
@@ -1572,7 +1611,7 @@
snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb,
seq->param);
}
-#endif /* CONFIG_PM */
+#endif /* SND_HDA_NEEDS_RESUME */
/*
* set power state of the codec
@@ -1580,24 +1619,70 @@
static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
- hda_nid_t nid, nid_start;
- int nodes;
+ hda_nid_t nid;
+ int i;
snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE,
power_state);
- nodes = snd_hda_get_sub_nodes(codec, fg, &nid_start);
- for (nid = nid_start; nid < nodes + nid_start; nid++) {
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
if (get_wcaps(codec, nid) & AC_WCAP_POWER)
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_POWER_STATE,
power_state);
}
- if (power_state == AC_PWRST_D0)
+ if (power_state == AC_PWRST_D0) {
+ unsigned long end_time;
+ int state;
msleep(10);
+ /* wait until the codec reachs to D0 */
+ end_time = jiffies + msecs_to_jiffies(500);
+ do {
+ state = snd_hda_codec_read(codec, fg, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+ if (state == power_state)
+ break;
+ msleep(1);
+ } while (time_after_eq(end_time, jiffies));
+ }
}
+#ifdef SND_HDA_NEEDS_RESUME
+/*
+ * call suspend and power-down; used both from PM and power-save
+ */
+static void hda_call_codec_suspend(struct hda_codec *codec)
+{
+ if (codec->patch_ops.suspend)
+ codec->patch_ops.suspend(codec, PMSG_SUSPEND);
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D3);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+#endif
+}
+
+/*
+ * kick up codec; used both from PM and power-save
+ */
+static void hda_call_codec_resume(struct hda_codec *codec)
+{
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D0);
+ if (codec->patch_ops.resume)
+ codec->patch_ops.resume(codec);
+ else {
+ codec->patch_ops.init(codec);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ }
+}
+#endif /* SND_HDA_NEEDS_RESUME */
+
/**
* snd_hda_build_controls - build mixer controls
@@ -1611,28 +1696,24 @@
{
struct hda_codec *codec;
- /* build controls */
list_for_each_entry(codec, &bus->codec_list, list) {
- int err;
- if (!codec->patch_ops.build_controls)
- continue;
- err = codec->patch_ops.build_controls(codec);
+ int err = 0;
+ /* fake as if already powered-on */
+ hda_keep_power_on(codec);
+ /* then fire up */
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D0);
+ /* continue to initialize... */
+ if (codec->patch_ops.init)
+ err = codec->patch_ops.init(codec);
+ if (!err && codec->patch_ops.build_controls)
+ err = codec->patch_ops.build_controls(codec);
+ snd_hda_power_down(codec);
if (err < 0)
return err;
}
- /* initialize */
- list_for_each_entry(codec, &bus->codec_list, list) {
- int err;
- hda_set_power_state(codec,
- codec->afg ? codec->afg : codec->mfg,
- AC_PWRST_D0);
- if (!codec->patch_ops.init)
- continue;
- err = codec->patch_ops.init(codec);
- if (err < 0)
- return err;
- }
return 0;
}
@@ -2078,7 +2159,7 @@
*/
int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
{
- int err;
+ int err;
for (; knew->name; knew++) {
struct snd_kcontrol *kctl;
@@ -2101,6 +2182,89 @@
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+ unsigned int power_state);
+
+static void hda_power_work(struct work_struct *work)
+{
+ struct hda_codec *codec =
+ container_of(work, struct hda_codec, power_work.work);
+
+ if (!codec->power_on || codec->power_count)
+ return;
+
+ hda_call_codec_suspend(codec);
+ codec->power_on = 0;
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+}
+
+static void hda_keep_power_on(struct hda_codec *codec)
+{
+ codec->power_count++;
+ codec->power_on = 1;
+}
+
+void snd_hda_power_up(struct hda_codec *codec)
+{
+ codec->power_count++;
+ if (codec->power_on)
+ return;
+
+ codec->power_on = 1;
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+ hda_call_codec_resume(codec);
+ cancel_delayed_work(&codec->power_work);
+}
+
+void snd_hda_power_down(struct hda_codec *codec)
+{
+ --codec->power_count;
+ if (!codec->power_on)
+ return;
+ if (power_save)
+ schedule_delayed_work(&codec->power_work,
+ msecs_to_jiffies(power_save * 1000));
+}
+
+int snd_hda_check_amp_list_power(struct hda_codec *codec,
+ struct hda_loopback_check *check,
+ hda_nid_t nid)
+{
+ struct hda_amp_list *p;
+ int ch, v;
+
+ if (!check->amplist)
+ return 0;
+ for (p = check->amplist; p->nid; p++) {
+ if (p->nid == nid)
+ break;
+ }
+ if (!p->nid)
+ return 0; /* nothing changed */
+
+ for (p = check->amplist; p->nid; p++) {
+ 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) {
+ if (!check->power_on) {
+ check->power_on = 1;
+ snd_hda_power_up(codec);
+ }
+ return 1;
+ }
+ }
+ }
+ if (check->power_on) {
+ check->power_on = 0;
+ snd_hda_power_down(codec);
+ }
+ return 0;
+}
+#endif
/*
* Channel mode helper
@@ -2605,41 +2769,32 @@
{
struct hda_codec *codec;
- /* FIXME: should handle power widget capabilities */
list_for_each_entry(codec, &bus->codec_list, list) {
- if (codec->patch_ops.suspend)
- codec->patch_ops.suspend(codec, state);
- hda_set_power_state(codec,
- codec->afg ? codec->afg : codec->mfg,
- AC_PWRST_D3);
+ hda_call_codec_suspend(codec);
}
return 0;
}
+#ifndef CONFIG_SND_HDA_POWER_SAVE
/**
* snd_hda_resume - resume the codecs
* @bus: the HDA bus
* @state: resume state
*
* Returns 0 if successful.
+ *
+ * This fucntion is defined only when POWER_SAVE isn't set.
+ * In the power-save mode, the codec is resumed dynamically.
*/
int snd_hda_resume(struct hda_bus *bus)
{
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
- hda_set_power_state(codec,
- codec->afg ? codec->afg : codec->mfg,
- AC_PWRST_D0);
- if (codec->patch_ops.resume)
- codec->patch_ops.resume(codec);
- else {
- codec->patch_ops.init(codec);
- snd_hda_codec_resume_amp(codec);
- snd_hda_codec_resume_cache(codec);
- }
+ hda_call_codec_resume(codec);
}
return 0;
}
+#endif /* !CONFIG_SND_HDA_POWER_SAVE */
#endif
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 92938d2..1ffffaa 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -26,6 +26,10 @@
#include <sound/pcm.h>
#include <sound/hwdep.h>
+#if defined(CONFIG_PM) || defined(CONFIG_SND_HDA_POWER_SAVE)
+#define SND_HDA_NEEDS_RESUME /* resume control code is required */
+#endif
+
/*
* nodes
*/
@@ -412,6 +416,10 @@
unsigned int (*get_response)(struct hda_codec *codec);
/* free the private data */
void (*private_free)(struct hda_bus *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ /* notify power-up/down from codec to contoller */
+ void (*pm_notify)(struct hda_codec *codec);
+#endif
};
/* template to pass to the bus constructor */
@@ -473,10 +481,13 @@
int (*init)(struct hda_codec *codec);
void (*free)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
int (*suspend)(struct hda_codec *codec, pm_message_t state);
int (*resume)(struct hda_codec *codec);
#endif
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
+#endif
};
/* record for amp information cache */
@@ -573,6 +584,12 @@
unsigned int spdif_in_enable; /* SPDIF input enable? */
struct snd_hwdep *hwdep; /* assigned hwdep device */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ int power_on; /* current (global) power-state */
+ int power_count; /* current (global) power refcount */
+ struct delayed_work power_work; /* delayed task for powerdown */
+#endif
};
/* direction */
@@ -617,7 +634,7 @@
int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex);
/* cached write */
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
int direct, unsigned int verb, unsigned int parm);
void snd_hda_sequence_write_cache(struct hda_codec *codec,
@@ -662,4 +679,15 @@
int snd_hda_resume(struct hda_bus *bus);
#endif
+/*
+ * power saving
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+void snd_hda_power_up(struct hda_codec *codec);
+void snd_hda_power_down(struct hda_codec *codec);
+#else
+static inline void snd_hda_power_up(struct hda_codec *codec) {}
+static inline void snd_hda_power_down(struct hda_codec *codec) {}
+#endif
+
#endif /* __SOUND_HDA_CODEC_H */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 91cd9b9e..819c804 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -70,6 +70,13 @@
struct hda_pcm pcm_rec; /* PCM information */
struct list_head nid_list; /* list of widgets */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define MAX_LOOPBACK_AMPS 7
+ struct hda_loopback_check loopback;
+ int num_loopbacks;
+ struct hda_amp_list loopback_list[MAX_LOOPBACK_AMPS + 1];
+#endif
};
/*
@@ -682,11 +689,33 @@
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void add_input_loopback(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_amp_list *p;
+
+ if (spec->num_loopbacks >= MAX_LOOPBACK_AMPS) {
+ snd_printk(KERN_ERR "hda_generic: Too many loopback ctls\n");
+ return;
+ }
+ p = &spec->loopback_list[spec->num_loopbacks++];
+ p->nid = nid;
+ p->dir = dir;
+ p->idx = idx;
+ spec->loopback.amplist = spec->loopback_list;
+}
+#else
+#define add_input_loopback(codec,nid,dir,idx)
+#endif
+
/*
* create mixer controls if possible
*/
static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
- unsigned int index, const char *type, const char *dir_sfx)
+ unsigned int index, const char *type,
+ const char *dir_sfx, int is_loopback)
{
char name[32];
int err;
@@ -700,6 +729,8 @@
if ((node->wid_caps & AC_WCAP_IN_AMP) &&
(node->amp_in_caps & AC_AMPCAP_MUTE)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT);
+ if (is_loopback)
+ add_input_loopback(codec, node->nid, HDA_INPUT, index);
snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
return err;
@@ -707,6 +738,8 @@
} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
(node->amp_out_caps & AC_AMPCAP_MUTE)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT);
+ if (is_loopback)
+ add_input_loopback(codec, node->nid, HDA_OUTPUT, 0);
snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
return err;
@@ -765,7 +798,7 @@
for (i = 0; i < spec->pcm_vol_nodes; i++) {
err = create_mixer(codec, spec->pcm_vol[i].node,
spec->pcm_vol[i].index,
- names[i], "Playback");
+ names[i], "Playback", 0);
if (err < 0)
return err;
}
@@ -782,7 +815,7 @@
case 1:
return create_mixer(codec, spec->pcm_vol[0].node,
spec->pcm_vol[0].index,
- "Master", "Playback");
+ "Master", "Playback", 0);
case 2:
if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER)
return create_output_mixers(codec, types_speaker);
@@ -818,7 +851,7 @@
if (spec->input_mux.num_items == 1) {
err = create_mixer(codec, adc_node,
spec->input_mux.items[0].index,
- NULL, "Capture");
+ NULL, "Capture", 0);
if (err < 0)
return err;
return 0;
@@ -884,7 +917,8 @@
return err;
else if (err >= 1) {
if (err == 1) {
- err = create_mixer(codec, node, i, type, "Playback");
+ err = create_mixer(codec, node, i, type,
+ "Playback", 1);
if (err < 0)
return err;
if (err > 0)
@@ -1020,6 +1054,14 @@
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int generic_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct hda_gspec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
/*
*/
@@ -1027,6 +1069,9 @@
.build_controls = build_generic_controls,
.build_pcms = build_generic_pcms,
.free = snd_hda_generic_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = generic_check_power_status,
+#endif
};
/*
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ebb442d..7be3a9b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -75,6 +75,7 @@
module_param(enable_msi, int, 0);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
+/* power_save option is defined in hda_codec.c */
/* just for backward compatibility */
static int enable;
@@ -102,6 +103,18 @@
#define SFX "hda-intel: "
/*
+ * build flags
+ */
+
+/*
+ * reset the HD-audio controller in power save mode.
+ * this may give more power-saving, but will take longer time to
+ * wake up.
+ */
+#define HDA_POWER_SAVE_RESET_CONTROLLER
+
+
+/*
* registers
*/
#define ICH6_REG_GCAP 0x00
@@ -345,6 +358,7 @@
/* flags */
int position_fix;
+ unsigned int running :1;
unsigned int initialized :1;
unsigned int single_cmd :1;
unsigned int polling_mode :1;
@@ -665,6 +679,9 @@
return azx_rirb_get_response(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void azx_power_notify(struct hda_codec *codec);
+#endif
/* reset codec link */
static int azx_reset(struct azx *chip)
@@ -790,19 +807,12 @@
/*
- * initialize the chip
+ * reset and start the controller registers
*/
static void azx_init_chip(struct azx *chip)
{
- unsigned char reg;
-
- /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
- * TCSEL == Traffic Class Select Register, which sets PCI express QOS
- * Ensuring these bits are 0 clears playback static on some HD Audio
- * codecs
- */
- pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, ®);
- pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, reg & 0xf8);
+ if (chip->initialized)
+ return;
/* reset controller */
azx_reset(chip);
@@ -819,22 +829,45 @@
azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
+ chip->initialized = 1;
+}
+
+/*
+ * initialize the PCI registers
+ */
+/* update bits in a PCI register byte */
+static void update_pci_byte(struct pci_dev *pci, unsigned int reg,
+ unsigned char mask, unsigned char val)
+{
+ unsigned char data;
+
+ pci_read_config_byte(pci, reg, &data);
+ data &= ~mask;
+ data |= (val & mask);
+ pci_write_config_byte(pci, reg, data);
+}
+
+static void azx_init_pci(struct azx *chip)
+{
+ /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
+ * TCSEL == Traffic Class Select Register, which sets PCI express QOS
+ * Ensuring these bits are 0 clears playback static on some HD Audio
+ * codecs
+ */
+ update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0);
+
switch (chip->driver_type) {
case AZX_DRIVER_ATI:
/* For ATI SB450 azalia HD audio, we need to enable snoop */
- pci_read_config_byte(chip->pci,
- ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
- ®);
- pci_write_config_byte(chip->pci,
- ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
- (reg & 0xf8) |
- ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+ update_pci_byte(chip->pci,
+ ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
+ 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP);
break;
case AZX_DRIVER_NVIDIA:
/* For NVIDIA HDA, enable snoop */
- pci_read_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR, ®);
- pci_write_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR,
- (reg & 0xf0) | NVIDIA_HDA_ENABLE_COHBITS);
+ update_pci_byte(chip->pci,
+ NVIDIA_HDA_TRANSREG_ADDR,
+ 0x0f, NVIDIA_HDA_ENABLE_COHBITS);
break;
}
}
@@ -1007,6 +1040,9 @@
bus_temp.pci = chip->pci;
bus_temp.ops.command = azx_send_cmd;
bus_temp.ops.get_response = azx_get_response;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ bus_temp.ops.pm_notify = azx_power_notify;
+#endif
err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus);
if (err < 0)
@@ -1128,9 +1164,11 @@
128);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
128);
+ snd_hda_power_up(apcm->codec);
err = hinfo->ops.open(hinfo, apcm->codec, substream);
if (err < 0) {
azx_release_device(azx_dev);
+ snd_hda_power_down(apcm->codec);
mutex_unlock(&chip->open_mutex);
return err;
}
@@ -1159,6 +1197,7 @@
spin_unlock_irqrestore(&chip->reg_lock, flags);
azx_release_device(azx_dev);
hinfo->ops.close(hinfo, apcm->codec, substream);
+ snd_hda_power_down(apcm->codec);
mutex_unlock(&chip->open_mutex);
return 0;
}
@@ -1459,6 +1498,48 @@
}
+static void azx_stop_chip(struct azx *chip)
+{
+ if (chip->initialized)
+ return;
+
+ /* disable interrupts */
+ azx_int_disable(chip);
+ azx_int_clear(chip);
+
+ /* disable CORB/RIRB */
+ azx_free_cmd_io(chip);
+
+ /* disable position buffer */
+ azx_writel(chip, DPLBASE, 0);
+ azx_writel(chip, DPUBASE, 0);
+
+ chip->initialized = 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* power-up/down the controller */
+static void azx_power_notify(struct hda_codec *codec)
+{
+ struct azx *chip = codec->bus->private_data;
+ struct hda_codec *c;
+ int power_on = 0;
+
+ list_for_each_entry(c, &codec->bus->codec_list, list) {
+ if (c->power_on) {
+ power_on = 1;
+ break;
+ }
+ }
+ if (power_on)
+ azx_init_chip(chip);
+#ifdef HDA_POWER_SAVE_RESET_CONTROLLER
+ else if (chip->running)
+ azx_stop_chip(chip);
+#endif
+}
+#endif /* CONFIG_SND_HDA_POWER_SAVE */
+
#ifdef CONFIG_PM
/*
* power management
@@ -1473,7 +1554,7 @@
for (i = 0; i < chip->pcm_devs; i++)
snd_pcm_suspend_all(chip->pcm[i]);
snd_hda_suspend(chip->bus, state);
- azx_free_cmd_io(chip);
+ azx_stop_chip(chip);
if (chip->irq >= 0) {
synchronize_irq(chip->irq);
free_irq(chip->irq, chip);
@@ -1506,8 +1587,12 @@
chip->msi = 0;
if (azx_acquire_irq(chip, 1) < 0)
return -EIO;
+ azx_init_pci(chip);
+#ifndef CONFIG_SND_HDA_POWER_SAVE
+ /* the explicit resume is needed only when POWER_SAVE isn't set */
azx_init_chip(chip);
snd_hda_resume(chip->bus);
+#endif
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
}
@@ -1521,20 +1606,9 @@
{
if (chip->initialized) {
int i;
-
for (i = 0; i < chip->num_streams; i++)
azx_stream_stop(chip, &chip->azx_dev[i]);
-
- /* disable interrupts */
- azx_int_disable(chip);
- azx_int_clear(chip);
-
- /* disable CORB/RIRB */
- azx_free_cmd_io(chip);
-
- /* disable position buffer */
- azx_writel(chip, DPLBASE, 0);
- azx_writel(chip, DPUBASE, 0);
+ azx_stop_chip(chip);
}
if (chip->irq >= 0) {
@@ -1720,10 +1794,9 @@
azx_init_stream(chip);
/* initialize chip */
+ azx_init_pci(chip);
azx_init_chip(chip);
- chip->initialized = 1;
-
/* codec detection */
if (!chip->codec_mask) {
snd_printk(KERN_ERR SFX "no codecs found!\n");
@@ -1750,6 +1823,19 @@
return err;
}
+static void power_down_all_codecs(struct azx *chip)
+{
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ /* The codecs were powered up in snd_hda_codec_new().
+ * Now all initialization done, so turn them down if possible
+ */
+ struct hda_codec *codec;
+ list_for_each_entry(codec, &chip->bus->codec_list, list) {
+ snd_hda_power_down(codec);
+ }
+#endif
+}
+
static int __devinit azx_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
@@ -1800,6 +1886,8 @@
}
pci_set_drvdata(pci, card);
+ chip->running = 1;
+ power_down_all_codecs(chip);
return err;
}
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 35ea0cf..a79d0ed 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -86,7 +86,7 @@
int direction, int idx, int mask, int val);
int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
int dir, int idx, int mask, int val);
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
void snd_hda_codec_resume_amp(struct hda_codec *codec);
#endif
@@ -366,4 +366,27 @@
*/
int snd_hda_create_hwdep(struct hda_codec *codec);
+/*
+ * power-management
+ */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+void snd_hda_schedule_power_save(struct hda_codec *codec);
+
+struct hda_amp_list {
+ hda_nid_t nid;
+ unsigned char dir;
+ unsigned char idx;
+};
+
+struct hda_loopback_check {
+ struct hda_amp_list *amplist;
+ int power_on;
+};
+
+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 */
+
#endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ccd1918..e94944f 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -262,6 +262,7 @@
if (! codec->afg)
return;
+ snd_hda_power_up(codec);
snd_iprintf(buffer, "Default PCM:\n");
print_pcm_caps(buffer, codec, codec->afg);
snd_iprintf(buffer, "Default Amp-In caps: ");
@@ -272,6 +273,7 @@
nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
if (! nid || nodes < 0) {
snd_iprintf(buffer, "Invalid AFG subtree\n");
+ snd_hda_power_down(codec);
return;
}
for (i = 0; i < nodes; i++, nid++) {
@@ -359,6 +361,7 @@
snd_iprintf(buffer, "\n");
}
}
+ snd_hda_power_down(codec);
}
/*
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index f9390a5..53cfa0d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -73,6 +73,10 @@
struct snd_kcontrol_new *kctl_alloc;
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[4];
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
};
/*
@@ -144,6 +148,14 @@
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
/*
* Analog playback callbacks
*/
@@ -323,6 +335,9 @@
.build_pcms = ad198x_build_pcms,
.init = ad198x_init,
.free = ad198x_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = ad198x_check_power_status,
+#endif
};
@@ -736,6 +751,17 @@
{}
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1986a_loopbacks[] = {
+ { 0x13, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x14, HDA_OUTPUT, 0 }, /* Phone */
+ { 0x15, HDA_OUTPUT, 0 }, /* CD */
+ { 0x16, HDA_OUTPUT, 0 }, /* Aux */
+ { 0x17, HDA_OUTPUT, 0 }, /* Line */
+ { } /* end */
+};
+#endif
+
static int patch_ad1986a(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -759,6 +785,9 @@
spec->mixers[0] = ad1986a_mixers;
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1986a_init_verbs;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1986a_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -944,6 +973,13 @@
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1983_loopbacks[] = {
+ { 0x12, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x13, HDA_OUTPUT, 0 }, /* Line */
+ { } /* end */
+};
+#endif
static int patch_ad1983(struct hda_codec *codec)
{
@@ -968,6 +1004,9 @@
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1983_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1983_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -1091,6 +1130,17 @@
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1981_loopbacks[] = {
+ { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
+ { 0x13, HDA_OUTPUT, 0 }, /* Line */
+ { 0x1b, HDA_OUTPUT, 0 }, /* Aux */
+ { 0x1c, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x1d, HDA_OUTPUT, 0 }, /* CD */
+ { } /* end */
+};
+#endif
+
/*
* Patch for HP nx6320
*
@@ -1350,6 +1400,9 @@
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1981_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1981_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -2103,6 +2156,15 @@
snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1988_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Line */
+ { 0x20, HDA_INPUT, 4 }, /* Mic */
+ { 0x20, HDA_INPUT, 6 }, /* CD */
+ { } /* end */
+};
+#endif
/*
* Automatic parse of I/O pins from the BIOS configuration
@@ -2647,6 +2709,9 @@
codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
break;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1988_loopbacks;
+#endif
return 0;
}
@@ -2803,6 +2868,16 @@
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1884_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 2 }, /* CD */
+ { 0x20, HDA_INPUT, 4 }, /* Docking */
+ { } /* end */
+};
+#endif
+
static int patch_ad1884(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -2827,6 +2902,9 @@
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1884_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1884_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -3208,6 +3286,16 @@
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1882_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 4 }, /* Line */
+ { 0x20, HDA_INPUT, 6 }, /* CD */
+ { } /* end */
+};
+#endif
+
/* models */
enum {
AD1882_3STACK,
@@ -3246,6 +3334,9 @@
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1882_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1882_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ebbabeb..b3d3916 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -240,6 +240,10 @@
/* for pin sensing */
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
};
/*
@@ -264,6 +268,9 @@
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*init_hook)(struct hda_codec *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_amp_list *loopbacks;
+#endif
};
@@ -621,6 +628,9 @@
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = preset->loopbacks;
+#endif
}
/* Enable GPIO mask and set output */
@@ -1287,11 +1297,13 @@
* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)
@@ -1836,8 +1848,8 @@
{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_UNMUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+ {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},
@@ -1939,7 +1951,7 @@
{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_UNMUTE(7)},
+ {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},
@@ -1979,6 +1991,24 @@
alc880_lg_lw_automute(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static 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 struct hda_amp_list alc880_lg_loopbacks[] = {
+ { 0x0b, HDA_INPUT, 1 },
+ { 0x0b, HDA_INPUT, 6 },
+ { 0x0b, HDA_INPUT, 7 },
+ { } /* end */
+};
+#endif
+
/*
* Common callbacks
*/
@@ -2005,6 +2035,14 @@
spec->unsol_event(codec, res);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
/*
* Analog playback callbacks
*/
@@ -2236,6 +2274,9 @@
.init = alc_init,
.free = alc_free,
.unsol_event = alc_unsol_event,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = alc_check_power_status,
+#endif
};
@@ -2860,6 +2901,9 @@
.input_mux = &alc880_lg_capture_source,
.unsol_event = alc880_lg_unsol_event,
.init_hook = alc880_lg_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .loopbacks = alc880_lg_loopbacks,
+#endif
},
[ALC880_LG_LW] = {
.mixers = { alc880_lg_lw_mixer },
@@ -3343,6 +3387,10 @@
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;
}
@@ -3691,12 +3739,12 @@
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
- /* mute CD */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- /* mute Line In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- /* mute Mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* 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)},
@@ -3741,12 +3789,12 @@
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
- /* unmute CD */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* unmute Line In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- /* unmute Mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ /* 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))},
@@ -3791,12 +3839,12 @@
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
- /* unmute CD */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* unmute Line In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- /* unmute Mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ /* 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))},
@@ -4418,11 +4466,12 @@
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)
@@ -4499,6 +4548,17 @@
alc260_auto_init_analog_input(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static 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
+
/*
* ALC260 configurations
*/
@@ -4698,6 +4758,10 @@
codec->patch_ops = alc_patch_ops;
if (board_config == ALC260_AUTO)
spec->init_hook = alc260_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc260_loopbacks;
+#endif
return 0;
}
@@ -5223,17 +5287,17 @@
{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
+ /* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)
@@ -5322,6 +5386,10 @@
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc882_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc882_pcm_analog_playback alc880_pcm_analog_playback
#define alc882_pcm_analog_capture alc880_pcm_analog_capture
@@ -5659,6 +5727,10 @@
codec->patch_ops = alc_patch_ops;
if (board_config == ALC882_AUTO)
spec->init_hook = alc882_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc882_loopbacks;
+#endif
return 0;
}
@@ -6242,11 +6314,12 @@
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ /* mute analog input loopbacks */
+ {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)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -6515,17 +6588,17 @@
{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
+ /* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)
@@ -6588,6 +6661,10 @@
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc883_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc883_pcm_analog_playback alc880_pcm_analog_playback
#define alc883_pcm_analog_capture alc880_pcm_analog_capture
@@ -7029,6 +7106,10 @@
codec->patch_ops = alc_patch_ops;
if (board_config == ALC883_AUTO)
spec->init_hook = alc883_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc883_loopbacks;
+#endif
return 0;
}
@@ -7186,17 +7267,17 @@
{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
+ /* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)
@@ -7565,17 +7646,17 @@
{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
+ /* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)
@@ -7626,19 +7707,19 @@
{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
+ /* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
+ {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)
@@ -7713,20 +7794,20 @@
{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
+ /* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+ {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)
*/
@@ -7796,6 +7877,10 @@
{ }
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc262_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc262_pcm_analog_playback alc880_pcm_analog_playback
#define alc262_pcm_analog_capture alc880_pcm_analog_capture
@@ -8098,6 +8183,10 @@
codec->patch_ops = alc_patch_ops;
if (board_config == ALC262_AUTO)
spec->init_hook = alc262_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc262_loopbacks;
+#endif
return 0;
}
@@ -8507,6 +8596,10 @@
alc268_auto_init_analog_input(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc883_loopbacks alc880_loopbacks
+#endif
+
/*
* configuration and preset
*/
@@ -9556,6 +9649,16 @@
alc861_auto_init_analog_input(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static 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
@@ -9753,6 +9856,10 @@
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO)
spec->init_hook = alc861_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861_loopbacks;
+#endif
return 0;
}
@@ -10035,11 +10142,11 @@
* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)},
@@ -10266,6 +10373,10 @@
alc861vd_dallas_automute(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc861vd_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc861vd_pcm_analog_playback alc880_pcm_analog_playback
#define alc861vd_pcm_analog_capture alc880_pcm_analog_capture
@@ -10688,6 +10799,10 @@
if (board_config == ALC861VD_AUTO)
spec->init_hook = alc861vd_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861vd_loopbacks;
+#endif
return 0;
}
@@ -10968,11 +11083,11 @@
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -11041,11 +11156,11 @@
* 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_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(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)
@@ -11132,6 +11247,10 @@
alc662_lenovo_101e_ispeaker_automute(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc662_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc662_pcm_analog_playback alc880_pcm_analog_playback
@@ -11534,6 +11653,10 @@
codec->patch_ops = alc_patch_ops;
if (board_config == ALC662_AUTO)
spec->init_hook = alc662_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc662_loopbacks;
+#endif
return 0;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index bf5d91b..4a98139 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1946,7 +1946,7 @@
}
}
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
static int stac92xx_resume(struct hda_codec *codec)
{
stac92xx_set_config_regs(codec);
@@ -1963,7 +1963,7 @@
.init = stac92xx_init,
.free = stac92xx_free,
.unsol_event = stac92xx_unsol_event,
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
.resume = stac92xx_resume,
#endif
};
@@ -2460,7 +2460,7 @@
.build_pcms = stac92xx_build_pcms,
.init = stac92xx_init,
.free = stac92xx_free,
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
.resume = stac92xx_resume,
#endif
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 6c734f0..33b5e1f 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -115,6 +115,10 @@
struct snd_kcontrol_new *kctl_alloc;
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[4];
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
};
static hda_nid_t vt1708_adc_nids[2] = {
@@ -305,15 +309,15 @@
{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
+ /* 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 */
- {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)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* master */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x19 - 0x1b)
@@ -543,6 +547,14 @@
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct via_spec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
/*
*/
static struct hda_codec_ops via_patch_ops = {
@@ -550,6 +562,9 @@
.build_pcms = via_build_pcms,
.init = via_init,
.free = via_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = via_check_power_status,
+#endif
};
/* fill in the dac_nids table from the parsed pin configuration */
@@ -738,6 +753,16 @@
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1708_loopbacks[] = {
+ { 0x17, HDA_INPUT, 1 },
+ { 0x17, HDA_INPUT, 2 },
+ { 0x17, HDA_INPUT, 3 },
+ { 0x17, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
static int vt1708_parse_auto_config(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -831,6 +856,9 @@
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
return 0;
}
@@ -871,15 +899,15 @@
{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
+ /* Mute 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)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* unmute master */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output selector (0x1a, 0x1b, 0x29)
@@ -1227,6 +1255,16 @@
return 1;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static 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)
{
struct via_spec *spec;
@@ -1269,6 +1307,9 @@
codec->patch_ops = via_patch_ops;
codec->patch_ops.init = via_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1709_loopbacks;
+#endif
return 0;
}
@@ -1359,6 +1400,9 @@
codec->patch_ops = via_patch_ops;
codec->patch_ops.init = via_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1709_loopbacks;
+#endif
return 0;
}