Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for 3.6
A bigger set of updates than I'm entirely comfortable with - things
backed up a bit due to travel. As ever the majority of these are small,
focused updates for specific drivers though there are a couple of core
changes. There's been good exposure in -next.
The AT91 patch fixes a build break.
diff --git a/MAINTAINERS b/MAINTAINERS
index 61ad79e..ed5df6e 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -7670,23 +7670,28 @@
F: Documentation/hwmon/wm83??
F: arch/arm/mach-s3c64xx/mach-crag6410*
F: drivers/clk/clk-wm83*.c
+F: drivers/extcon/extcon-arizona.c
F: drivers/leds/leds-wm83*.c
F: drivers/gpio/gpio-*wm*.c
+F: drivers/gpio/gpio-arizona.c
F: drivers/hwmon/wm83??-hwmon.c
F: drivers/input/misc/wm831x-on.c
F: drivers/input/touchscreen/wm831x-ts.c
F: drivers/input/touchscreen/wm97*.c
-F: drivers/mfd/wm8*.c
+F: drivers/mfd/arizona*
+F: drivers/mfd/wm*.c
F: drivers/power/wm83*.c
F: drivers/rtc/rtc-wm83*.c
F: drivers/regulator/wm8*.c
F: drivers/video/backlight/wm83*_bl.c
F: drivers/watchdog/wm83*_wdt.c
+F: include/linux/mfd/arizona/
F: include/linux/mfd/wm831x/
F: include/linux/mfd/wm8350/
F: include/linux/mfd/wm8400*
F: include/linux/wm97xx.h
F: include/sound/wm????.h
+F: sound/soc/codecs/arizona.?
F: sound/soc/codecs/wm*
WORKQUEUE
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index c75c0d1..cdca2ab 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1075,7 +1075,8 @@
const char *snd_pcm_format_name(snd_pcm_format_t format);
/**
- * Get a string naming the direction of a stream
+ * snd_pcm_stream_str - Get a string naming the direction of a stream
+ * @substream: the pcm substream instance
*/
static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
{
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 0d7b25e..4e1fda7 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -106,7 +106,7 @@
.prepare = pxa2xx_ac97_pcm_prepare,
};
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int pxa2xx_ac97_do_suspend(struct snd_card *card)
{
@@ -243,7 +243,7 @@
.driver = {
.name = "pxa2xx-ac97",
.owner = THIS_MODULE,
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
.pm = &pxa2xx_ac97_pm_ops,
#endif
},
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index eb4ceb7..277ebce 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -452,6 +452,7 @@
dac->regs = ioremap(regs->start, resource_size(regs));
if (!dac->regs) {
dev_dbg(&pdev->dev, "could not remap register memory\n");
+ retval = -ENOMEM;
goto out_free_card;
}
@@ -534,7 +535,7 @@
return retval;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int atmel_abdac_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index bf47025..9052aff 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -278,14 +278,9 @@
if (retval < 0)
return retval;
/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
- if (cpu_is_at32ap7000()) {
- if (retval < 0)
- return retval;
- /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
- if (retval == 1)
- if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
- dw_dma_cyclic_free(chip->dma.rx_chan);
- }
+ if (cpu_is_at32ap7000() && retval == 1)
+ if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
+ dw_dma_cyclic_free(chip->dma.rx_chan);
/* Set restrictions to params. */
mutex_lock(&opened_mutex);
@@ -980,6 +975,7 @@
if (!chip->regs) {
dev_dbg(&pdev->dev, "could not remap register memory\n");
+ retval = -ENOMEM;
goto err_ioremap;
}
@@ -1134,7 +1130,7 @@
return retval;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int atmel_ac97c_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index ec2118d..eb60cb8 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -80,14 +80,12 @@
int maj = imajor(inode);
int ret;
- if (f->f_flags & O_WRONLY)
+ if ((f->f_flags & O_ACCMODE) == O_WRONLY)
dirn = SND_COMPRESS_PLAYBACK;
- else if (f->f_flags & O_RDONLY)
+ else if ((f->f_flags & O_ACCMODE) == O_RDONLY)
dirn = SND_COMPRESS_CAPTURE;
- else {
- pr_err("invalid direction\n");
+ else
return -EINVAL;
- }
if (maj == snd_major)
compr = snd_lookup_minor_data(iminor(inode),
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 1128b35..5a34355 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1176,7 +1176,7 @@
return 0;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int loopback_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index f7d3bfc..54bb664 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1064,7 +1064,7 @@
return 0;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int snd_dummy_suspend(struct device *pdev)
{
struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 6ca59fc..ef17129 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -199,7 +199,7 @@
pcspkr_stop_sound();
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int pcsp_suspend(struct device *dev)
{
struct snd_pcsp *chip = dev_get_drvdata(dev);
@@ -212,7 +212,7 @@
#define PCSP_PM_OPS &pcsp_pm
#else
#define PCSP_PM_OPS NULL
-#endif /* CONFIG_PM */
+#endif /* CONFIG_PM_SLEEP */
static void pcsp_shutdown(struct platform_device *dev)
{
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 2d67c78..f7cdaf5 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -233,7 +233,7 @@
irq[dev], dma8[dev], dma16[dev]);
}
- if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
+ if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) {
snd_card_free(card);
return error;
}
diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c
index 733b014..b2b3c01 100644
--- a/sound/oss/sb_audio.c
+++ b/sound/oss/sb_audio.c
@@ -575,13 +575,15 @@
if (speed > 0)
{
int tmp;
- int s = speed * devc->channels;
+ int s;
if (speed < 5000)
speed = 5000;
if (speed > 44100)
speed = 44100;
+ s = speed * devc->channels;
+
devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff;
tmp = 256 - devc->tconst;
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index f75f5ff..a71d1c1 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -94,7 +94,7 @@
if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
codec_index != CS46XX_SECONDARY_CODEC_INDEX))
- return -EINVAL;
+ return 0xffff;
chip->active_ctrl(chip, 1);
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 8e40262..2f6e9c7 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1725,8 +1725,10 @@
atc_connect_resources(atc);
atc->timer = ct_timer_new(atc);
- if (!atc->timer)
+ if (!atc->timer) {
+ err = -ENOMEM;
goto error1;
+ }
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops);
if (err < 0)
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 0bc2315..0849aac 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -231,16 +231,22 @@
}
EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
+static bool ctl_has_mute(struct snd_kcontrol *kcontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ return query_amp_caps(codec, get_amp_nid(kcontrol),
+ get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE;
+}
+
/* get/put callbacks for beep mute mixer switches */
int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct hda_beep *beep = codec->beep;
- if (beep) {
+ if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
ucontrol->value.integer.value[0] =
- ucontrol->value.integer.value[1] =
- beep->enabled;
+ ucontrol->value.integer.value[1] = beep->enabled;
return 0;
}
return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
@@ -252,9 +258,20 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct hda_beep *beep = codec->beep;
- if (beep)
- snd_hda_enable_beep_device(codec,
- *ucontrol->value.integer.value);
+ if (beep) {
+ u8 chs = get_amp_channels(kcontrol);
+ int enable = 0;
+ long *valp = ucontrol->value.integer.value;
+ if (chs & 1) {
+ enable |= *valp;
+ valp++;
+ }
+ if (chs & 2)
+ enable |= *valp;
+ snd_hda_enable_beep_device(codec, enable);
+ }
+ if (!ctl_has_mute(kcontrol))
+ return 0;
return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88a9c20..1c65cc5 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1209,6 +1209,9 @@
kfree(codec);
}
+static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec,
+ hda_nid_t fg, unsigned int power_state);
+
static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state);
@@ -1317,6 +1320,10 @@
AC_VERB_GET_SUBSYSTEM_ID, 0);
}
+ codec->epss = snd_hda_codec_get_supported_ps(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_EPSS);
+
/* power-up all before initialization */
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
@@ -1386,6 +1393,44 @@
}
EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
+/* update the stream-id if changed */
+static void update_pcm_stream_id(struct hda_codec *codec,
+ struct hda_cvt_setup *p, hda_nid_t nid,
+ u32 stream_tag, int channel_id)
+{
+ unsigned int oldval, newval;
+
+ if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
+ 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);
+ p->stream_tag = stream_tag;
+ p->channel_id = channel_id;
+ }
+}
+
+/* update the format-id if changed */
+static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p,
+ hda_nid_t nid, int format)
+{
+ unsigned int oldval;
+
+ if (p->format_id != format) {
+ oldval = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_STREAM_FORMAT, 0);
+ if (oldval != format) {
+ msleep(1);
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_STREAM_FORMAT,
+ format);
+ }
+ p->format_id = format;
+ }
+}
+
/**
* snd_hda_codec_setup_stream - set up the codec for streaming
* @codec: the CODEC to set up
@@ -1400,7 +1445,6 @@
{
struct hda_codec *c;
struct hda_cvt_setup *p;
- unsigned int oldval, newval;
int type;
int i;
@@ -1413,29 +1457,13 @@
p = get_hda_cvt_setup(codec, nid);
if (!p)
return;
- /* update the stream-id if changed */
- if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
- 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);
- p->stream_tag = stream_tag;
- p->channel_id = channel_id;
- }
- /* update the format-id if changed */
- if (p->format_id != format) {
- oldval = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_STREAM_FORMAT, 0);
- if (oldval != format) {
- msleep(1);
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_STREAM_FORMAT,
- format);
- }
- p->format_id = format;
- }
+
+ if (codec->pcm_format_first)
+ update_pcm_format(codec, p, nid, format);
+ update_pcm_stream_id(codec, p, nid, stream_tag, channel_id);
+ if (!codec->pcm_format_first)
+ update_pcm_format(codec, p, nid, format);
+
p->active = 1;
p->dirty = 0;
@@ -2325,6 +2353,7 @@
}
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
+ memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
snd_hda_jack_tbl_clear(codec);
codec->proc_widget_hook = NULL;
codec->spec = NULL;
@@ -2340,7 +2369,6 @@
codec->num_pcms = 0;
codec->pcm_info = NULL;
codec->preset = NULL;
- memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
codec->slave_dig_outs = NULL;
codec->spdif_status_reset = 0;
module_put(codec->owner);
@@ -3497,7 +3525,7 @@
{
int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE);
- if (sup < 0)
+ if (sup == -1)
return false;
if (sup & power_state)
return true;
@@ -3522,8 +3550,7 @@
/* this delay seems necessary to avoid click noise at power-down */
if (power_state == AC_PWRST_D3) {
/* transition time less than 10ms for power down */
- bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS);
- msleep(epss ? 10 : 100);
+ msleep(codec->epss ? 10 : 100);
}
/* repeat power states setting at most 10 times*/
@@ -4433,6 +4460,8 @@
* then there is no need to go through power up here.
*/
if (codec->power_on) {
+ if (codec->power_transition < 0)
+ codec->power_transition = 0;
spin_unlock(&codec->power_lock);
return;
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index c422d33..e5a7e19 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -861,6 +861,8 @@
unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
unsigned int no_jack_detect:1; /* Machine has no jack-detection */
+ unsigned int pcm_format_first:1; /* PCM format must be set first */
+ unsigned int epss:1; /* supporting EPSS? */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
int power_transition; /* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8aced1..c4763c5 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -151,6 +151,7 @@
"{Intel, CPT},"
"{Intel, PPT},"
"{Intel, LPT},"
+ "{Intel, LPT_LP},"
"{Intel, HPT},"
"{Intel, PBG},"
"{Intel, SCH},"
@@ -2700,6 +2701,8 @@
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, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF),
+ SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
@@ -3270,6 +3273,14 @@
{ PCI_DEVICE(0x8086, 0x8c20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+ /* Lynx Point-LP */
+ { PCI_DEVICE(0x8086, 0x9c20),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+ /* Lynx Point-LP */
+ { PCI_DEVICE(0x8086, 0x9c21),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
/* Haswell */
{ PCI_DEVICE(0x8086, 0x0c0c),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 7e46258..6894ec6 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -412,7 +412,7 @@
if (digi1 & AC_DIG1_EMPHASIS)
snd_iprintf(buffer, " Preemphasis");
if (digi1 & AC_DIG1_COPYRIGHT)
- snd_iprintf(buffer, " Copyright");
+ snd_iprintf(buffer, " Non-Copyright");
if (digi1 & AC_DIG1_NONAUDIO)
snd_iprintf(buffer, " Non-Audio");
if (digi1 & AC_DIG1_PROFESSIONAL)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index d0d3540..49750a9 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -246,7 +246,7 @@
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
}
- if (dac)
+ if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP))
snd_hda_codec_write(codec, dac, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
}
@@ -261,7 +261,7 @@
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
}
- if (adc)
+ if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP))
snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
}
@@ -275,6 +275,10 @@
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+ if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) {
+ snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid);
+ return 0;
+ }
sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
@@ -286,6 +290,10 @@
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+ if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) {
+ snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid);
+ return 0;
+ }
sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
@@ -464,50 +472,17 @@
}
/*
- * 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_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
@@ -515,10 +490,8 @@
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
-
- return 0;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
}
static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
@@ -526,15 +499,20 @@
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->dacs[0]);
-
- return 0;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
}
/*
* Digital out
*/
+static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0132_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
@@ -542,10 +520,8 @@
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format);
-
- return 0;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
}
static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
@@ -553,65 +529,15 @@
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->dig_out);
-
- return 0;
+ return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
}
-/*
- * 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)
+static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ 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;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
/*
@@ -621,6 +547,7 @@
.channels_min = 2,
.channels_max = 2,
.ops = {
+ .open = ca0132_playback_pcm_open,
.prepare = ca0132_playback_pcm_prepare,
.cleanup = ca0132_playback_pcm_cleanup
},
@@ -630,10 +557,6 @@
.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 = {
@@ -641,6 +564,8 @@
.channels_min = 2,
.channels_max = 2,
.ops = {
+ .open = ca0132_dig_playback_pcm_open,
+ .close = ca0132_dig_playback_pcm_close,
.prepare = ca0132_dig_playback_pcm_prepare,
.cleanup = ca0132_dig_playback_pcm_cleanup
},
@@ -650,10 +575,6 @@
.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)
@@ -928,18 +849,16 @@
spec->dig_out);
if (err < 0)
return err;
- err = add_out_volume(codec, spec->dig_out, "IEC958");
+ err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
if (err < 0)
return err;
+ /* spec->multiout.share_spdif = 1; */
}
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");
- if (err < 0)
- return err;
}
return 0;
}
@@ -961,6 +880,9 @@
struct ca0132_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
+ codec->pcm_format_first = 1;
+ codec->no_sticky_stream = 1;
+
/* line-outs */
cfg->line_outs = 1;
cfg->line_out_pins[0] = 0x0b; /* front */
@@ -988,14 +910,24 @@
/* Mic-in */
spec->input_pins[0] = 0x12;
- spec->input_labels[0] = "Mic-In";
+ spec->input_labels[0] = "Mic";
spec->adcs[0] = 0x07;
/* Line-In */
spec->input_pins[1] = 0x11;
- spec->input_labels[1] = "Line-In";
+ spec->input_labels[1] = "Line";
spec->adcs[1] = 0x08;
spec->num_inputs = 2;
+
+ /* SPDIF I/O */
+ spec->dig_out = 0x05;
+ spec->multiout.dig_out_nid = spec->dig_out;
+ cfg->dig_out_pins[0] = 0x0c;
+ cfg->dig_outs = 1;
+ cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+ spec->dig_in = 0x09;
+ cfg->dig_in_pin = 0x0e;
+ cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
}
static void ca0132_init_chip(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 94040cc..3d4722f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1075,7 +1075,7 @@
static const char * const slave_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
- "Headphone", "Speaker", "IEC958",
+ "Headphone", "Speaker", "IEC958", "PCM",
NULL
};
@@ -4272,7 +4272,8 @@
unsigned int gpio;
int i;
- snd_hda_sequence_write(codec, spec->init);
+ if (spec->init)
+ snd_hda_sequence_write(codec, spec->init);
/* power down adcs initially */
if (spec->powerdown_adcs)
@@ -4542,6 +4543,9 @@
struct auto_pin_cfg *cfg = &spec->autocfg;
int i;
+ if (cfg->speaker_outs == 0)
+ return;
+
for (i = 0; i < cfg->line_outs; i++) {
if (presence)
break;
@@ -5530,6 +5534,7 @@
snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e);
}
+ codec->epss = 0; /* longer delay needed for D3 */
codec->no_trigger_sense = 1;
codec->spec = spec;
@@ -5748,7 +5753,6 @@
/* fallthru */
case 0x111d76b4: /* 6 Port without Analog Mixer */
case 0x111d76b5:
- spec->init = stac92hd71bxx_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd71bxx_dmic_nids,
@@ -5773,7 +5777,6 @@
spec->stream_delay = 40; /* 40 milliseconds */
/* disable VSW */
- spec->init = stac92hd71bxx_core_init;
unmute_init++;
snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
@@ -5788,7 +5791,6 @@
/* fallthru */
default:
- spec->init = stac92hd71bxx_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd71bxx_dmic_nids,
@@ -5796,6 +5798,9 @@
break;
}
+ if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB)
+ spec->init = stac92hd71bxx_core_init;
+
if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
snd_hda_sequence_write_cache(codec, unmute_init);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 80d90cb..4307717 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1752,6 +1752,14 @@
{
struct via_spec *spec = codec->spec;
vt1708_stop_hp_work(spec);
+
+ if (spec->codec_type == VT1802) {
+ /* Fix pop noise on headphones */
+ int i;
+ for (i = 0; i < spec->autocfg.hp_outs; i++)
+ snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0);
+ }
+
return 0;
}
#endif
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 764cc93d..075d5aa 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -297,6 +297,7 @@
}
static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
{
@@ -307,7 +308,7 @@
.info = ak4396_dac_vol_info,
.get = ak4396_dac_vol_get,
.put = ak4396_dac_vol_put,
- .tlv = { .p = db_scale_wm_dac },
+ .tlv = { .p = ak4396_db_scale },
},
};
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d1ab437..5579b08 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -851,6 +851,8 @@
/* hardcoded device name & channel count */
err = snd_pcm_new(chip->card, (char *)card_name, 0,
1, 1, &pcm);
+ if (err < 0)
+ return err;
pcm->private_data = chip;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index b8ac871..b12308b 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6585,7 +6585,7 @@
snd_printk(KERN_ERR "HDSPM: "
"unable to kmalloc Mixer memory of %d Bytes\n",
(int)sizeof(struct hdspm_mixer));
- return err;
+ return -ENOMEM;
}
hdspm->port_names_in = NULL;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 512434e..805ab6e 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1377,8 +1377,9 @@
if (rc)
goto error_out_cleanup;
- if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
- sis)) {
+ rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
+ sis);
+ if (rc) {
dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq);
goto error_out_cleanup;
}
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index f5ceb6f..210cafe 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -143,7 +143,7 @@
return 0;
}
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
static int snd_pmac_driver_suspend(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 1aa52ef..9b18b52 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -1040,6 +1040,7 @@
GFP_KERNEL);
if (!the_card.null_buffer_start_vaddr) {
pr_info("%s: nullbuffer alloc failed\n", __func__);
+ ret = -ENOMEM;
goto clean_preallocate;
}
pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__,
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index 318c5ba5..dfb7443 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -413,7 +413,14 @@
void sport_delete(struct sport_device *sport)
{
+ if (sport->tx_desc)
+ dma_free_coherent(NULL, sport->tx_desc_size,
+ sport->tx_desc, 0);
+ if (sport->rx_desc)
+ dma_free_coherent(NULL, sport->rx_desc_size,
+ sport->rx_desc, 0);
sport_free_resource(sport);
+ kfree(sport);
}
EXPORT_SYMBOL(sport_delete);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 6537f16..e33d327 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -128,13 +128,9 @@
ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
-SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
- ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
@@ -236,8 +232,6 @@
ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
@@ -349,10 +343,6 @@
NULL, 0),
SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
NULL, 0),
-SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
- NULL, 0),
-SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
- NULL, 0),
SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
NULL, 0),
@@ -466,8 +456,6 @@
ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
-ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
-ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
@@ -553,8 +541,6 @@
{ name, "EQ4", "EQ4" }, \
{ name, "DRC1L", "DRC1L" }, \
{ name, "DRC1R", "DRC1R" }, \
- { name, "DRC2L", "DRC2L" }, \
- { name, "DRC2R", "DRC2R" }, \
{ name, "LHPF1", "LHPF1" }, \
{ name, "LHPF2", "LHPF2" }, \
{ name, "LHPF3", "LHPF3" }, \
@@ -639,6 +625,15 @@
{ "AIF2 Capture", NULL, "SYSCLK" },
{ "AIF3 Capture", NULL, "SYSCLK" },
+ { "IN1L PGA", NULL, "IN1L" },
+ { "IN1R PGA", NULL, "IN1R" },
+
+ { "IN2L PGA", NULL, "IN2L" },
+ { "IN2R PGA", NULL, "IN2R" },
+
+ { "IN3L PGA", NULL, "IN3L" },
+ { "IN3R PGA", NULL, "IN3R" },
+
ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
@@ -675,8 +670,6 @@
ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
- ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
- ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 8033f70..01ebbcc 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -681,6 +681,18 @@
{ "AIF2 Capture", NULL, "SYSCLK" },
{ "AIF3 Capture", NULL, "SYSCLK" },
+ { "IN1L PGA", NULL, "IN1L" },
+ { "IN1R PGA", NULL, "IN1R" },
+
+ { "IN2L PGA", NULL, "IN2L" },
+ { "IN2R PGA", NULL, "IN2R" },
+
+ { "IN3L PGA", NULL, "IN3L" },
+ { "IN3R PGA", NULL, "IN3R" },
+
+ { "IN4L PGA", NULL, "IN4L" },
+ { "IN4R PGA", NULL, "IN4R" },
+
ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index aa9ce9d..ce67200 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3733,21 +3733,6 @@
regcache_sync(wm8962->regmap);
- regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP,
- WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA,
- WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA);
-
- /* Bias enable at 2*50k for ramp */
- regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
- WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA,
- WM8962_BIAS_ENA | 0x180);
-
- msleep(5);
-
- /* VMID back to 2x250k for standby */
- regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
- WM8962_VMID_SEL_MASK, 0x100);
-
return 0;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 04ef031..6c9eeca 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -4038,6 +4038,8 @@
break;
case WM8958:
if (wm8994->revision < 1) {
+ snd_soc_dapm_add_routes(dapm, wm8994_intercon,
+ ARRAY_SIZE(wm8994_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
ARRAY_SIZE(wm8994_revd_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index f16fb36..c6d2076 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -148,7 +148,7 @@
SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
-SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv),
@@ -272,7 +272,7 @@
/* Mic select */
static const struct snd_kcontrol_new wm9712_mic_src_controls =
-SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
/* diff select */
static const struct snd_kcontrol_new wm9712_diff_sel_controls =
@@ -291,7 +291,9 @@
&wm9712_capture_selectl_controls),
SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectr_controls),
-SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
&wm9712_mic_src_controls),
SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
&wm9712_diff_sel_controls),
@@ -319,6 +321,7 @@
SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
SND_SOC_DAPM_OUTPUT("MONOOUT"),
SND_SOC_DAPM_OUTPUT("HPOUTL"),
@@ -379,6 +382,18 @@
{"Mic PGA", NULL, "MIC1"},
{"Mic PGA", NULL, "MIC2"},
+ /* microphones */
+ {"Differential Mic", NULL, "MIC1"},
+ {"Differential Mic", NULL, "MIC2"},
+ {"Left Mic Select Source", "Mic 1", "MIC1"},
+ {"Left Mic Select Source", "Mic 2", "MIC2"},
+ {"Left Mic Select Source", "Stereo", "MIC1"},
+ {"Left Mic Select Source", "Differential", "Differential Mic"},
+ {"Right Mic Select Source", "Mic 1", "MIC1"},
+ {"Right Mic Select Source", "Mic 2", "MIC2"},
+ {"Right Mic Select Source", "Stereo", "MIC2"},
+ {"Right Mic Select Source", "Differential", "Differential Mic"},
+
/* left capture selector */
{"Left Capture Select", "Mic", "MIC1"},
{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 95441bf..ce5e5cd 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -380,14 +380,20 @@
static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
{
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (dev->txnumevt) /* enable FIFO */
+ if (dev->txnumevt) { /* enable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
FIFO_ENABLE);
+ }
mcasp_start_tx(dev);
} else {
- if (dev->rxnumevt) /* enable FIFO */
+ if (dev->rxnumevt) { /* enable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
FIFO_ENABLE);
+ }
mcasp_start_rx(dev);
}
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 28dd76c..81d7728 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -380,13 +380,14 @@
static struct snd_soc_dai_driver imx_ssi_dai = {
.probe = imx_ssi_dai_probe,
.playback = {
- .channels_min = 1,
+ /* The SSI does not support monaural audio. */
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
index 99a997f..b6fa776 100644
--- a/sound/soc/mxs/Kconfig
+++ b/sound/soc/mxs/Kconfig
@@ -10,7 +10,7 @@
if SND_MXS_SOC
config SND_SOC_MXS_SGTL5000
- tristate "SoC Audio support for i.MX boards with sgtl5000"
+ tristate "SoC Audio support for MXS boards with sgtl5000"
depends on I2C
select SND_SOC_SGTL5000
help
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 34835e8..d33c48b 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -745,7 +745,7 @@
{
const char *signal, *src;
- if (mcbsp->pdata->mux_signal)
+ if (!mcbsp->pdata->mux_signal)
return -EINVAL;
switch (mux) {
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index b7b2a1f..89b0646 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -20,7 +20,7 @@
#include <sound/pcm_params.h>
#include <plat/audio.h>
-#include <plat/dma.h>
+#include <mach/dma.h>
#include "dma.h"
#include "pcm.h"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f81c597..c501af6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -826,7 +826,7 @@
}
if (!rtd->cpu_dai) {
- dev_dbg(card->dev, "CPU DAI %s not registered\n",
+ dev_err(card->dev, "CPU DAI %s not registered\n",
dai_link->cpu_dai_name);
return -EPROBE_DEFER;
}
@@ -857,14 +857,14 @@
}
if (!rtd->codec_dai) {
- dev_dbg(card->dev, "CODEC DAI %s not registered\n",
+ dev_err(card->dev, "CODEC DAI %s not registered\n",
dai_link->codec_dai_name);
return -EPROBE_DEFER;
}
}
if (!rtd->codec) {
- dev_dbg(card->dev, "CODEC %s not registered\n",
+ dev_err(card->dev, "CODEC %s not registered\n",
dai_link->codec_name);
return -EPROBE_DEFER;
}
@@ -888,7 +888,7 @@
rtd->platform = platform;
}
if (!rtd->platform) {
- dev_dbg(card->dev, "platform %s not registered\n",
+ dev_err(card->dev, "platform %s not registered\n",
dai_link->platform_name);
return -EPROBE_DEFER;
}
@@ -1481,6 +1481,8 @@
return 0;
}
+ dev_err(card->dev, "%s not registered\n", aux_dev->codec_name);
+
return -EPROBE_DEFER;
}
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7f8b3b7..0c17293 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -103,7 +103,7 @@
}
/* Report before the DAPM sync to help users updating micbias status */
- blocking_notifier_call_chain(&jack->notifier, status, jack);
+ blocking_notifier_call_chain(&jack->notifier, jack->status, jack);
snd_soc_dapm_sync(dapm);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index d5b5c33..4a469f0 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -553,7 +553,7 @@
struct snd_usb_audio *chip)
{
struct snd_card *card;
- struct list_head *p;
+ struct list_head *p, *n;
if (chip == (void *)-1L)
return;
@@ -570,7 +570,7 @@
snd_usb_stream_disconnect(p);
}
/* release the endpoint resources */
- list_for_each(p, &chip->ep_list) {
+ list_for_each_safe(p, n, &chip->ep_list) {
snd_usb_endpoint_free(p);
}
/* release the midi resources */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 0f647d2..d6e2bb4 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -141,7 +141,7 @@
*
* For implicit feedback, next_packet_size() is unused.
*/
-static int next_packet_size(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
{
unsigned long flags;
int ret;
@@ -177,15 +177,6 @@
ep->retire_data_urb(ep->data_subs, urb);
}
-static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep,
- struct snd_urb_ctx *ctx)
-{
- int i;
-
- for (i = 0; i < ctx->packets; ++i)
- ctx->packet_size[i] = next_packet_size(ep);
-}
-
/*
* Prepare a PLAYBACK urb for submission to the bus.
*/
@@ -370,7 +361,6 @@
goto exit_clear;
}
- prepare_outbound_urb_sizes(ep, ctx);
prepare_outbound_urb(ep, ctx);
} else {
retire_inbound_urb(ep, ctx);
@@ -799,7 +789,9 @@
/**
* snd_usb_endpoint_start: start an snd_usb_endpoint
*
- * @ep: the endpoint to start
+ * @ep: the endpoint to start
+ * @can_sleep: flag indicating whether the operation is executed in
+ * non-atomic context
*
* A call to this function will increment the use count of the endpoint.
* In case it is not already running, the URBs for this endpoint will be
@@ -809,7 +801,7 @@
*
* Returns an error if the URB submission failed, 0 in all other cases.
*/
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep)
{
int err;
unsigned int i;
@@ -822,8 +814,9 @@
return 0;
/* just to be sure */
- deactivate_urbs(ep, 0, 1);
- wait_clear_urbs(ep);
+ deactivate_urbs(ep, 0, can_sleep);
+ if (can_sleep)
+ wait_clear_urbs(ep);
ep->active_mask = 0;
ep->unlink_mask = 0;
@@ -854,7 +847,6 @@
goto __error;
if (usb_pipeout(ep->pipe)) {
- prepare_outbound_urb_sizes(ep, urb->context);
prepare_outbound_urb(ep, urb->context);
} else {
prepare_inbound_urb(ep, urb->context);
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index ee2723f..cbbbdf2 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -13,7 +13,7 @@
struct audioformat *fmt,
struct snd_usb_endpoint *sync_ep);
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep);
void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
int force, int can_sleep, int wait);
int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
@@ -21,6 +21,7 @@
void snd_usb_endpoint_free(struct list_head *head);
int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep);
void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
struct snd_usb_endpoint *sender,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a1298f3..f782ce1 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -212,7 +212,7 @@
}
}
-static int start_endpoints(struct snd_usb_substream *subs)
+static int start_endpoints(struct snd_usb_substream *subs, int can_sleep)
{
int err;
@@ -225,7 +225,7 @@
snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep);
ep->data_subs = subs;
- err = snd_usb_endpoint_start(ep);
+ err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
return err;
@@ -236,10 +236,25 @@
!test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) {
struct snd_usb_endpoint *ep = subs->sync_endpoint;
+ if (subs->data_endpoint->iface != subs->sync_endpoint->iface ||
+ subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) {
+ err = usb_set_interface(subs->dev,
+ subs->sync_endpoint->iface,
+ subs->sync_endpoint->alt_idx);
+ if (err < 0) {
+ snd_printk(KERN_ERR
+ "%d:%d:%d: cannot set interface (%d)\n",
+ subs->dev->devnum,
+ subs->sync_endpoint->iface,
+ subs->sync_endpoint->alt_idx, err);
+ return -EIO;
+ }
+ }
+
snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep);
ep->sync_slave = subs->data_endpoint;
- err = snd_usb_endpoint_start(ep);
+ err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
return err;
@@ -547,7 +562,7 @@
/* for playback, submit the URBs now; otherwise, the first hwptr_done
* updates for all URBs would happen at the same time when starting */
if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
- return start_endpoints(subs);
+ return start_endpoints(subs, 1);
return 0;
}
@@ -1029,6 +1044,7 @@
struct urb *urb)
{
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ struct snd_usb_endpoint *ep = subs->data_endpoint;
struct snd_urb_ctx *ctx = urb->context;
unsigned int counts, frames, bytes;
int i, stride, period_elapsed = 0;
@@ -1040,7 +1056,11 @@
urb->number_of_packets = 0;
spin_lock_irqsave(&subs->lock, flags);
for (i = 0; i < ctx->packets; i++) {
- counts = ctx->packet_size[i];
+ if (ctx->packet_size[i])
+ counts = ctx->packet_size[i];
+ else
+ counts = snd_usb_endpoint_next_packet_size(ep);
+
/* set up descriptor */
urb->iso_frame_desc[i].offset = frames * stride;
urb->iso_frame_desc[i].length = counts * stride;
@@ -1091,7 +1111,16 @@
subs->hwptr_done += bytes;
if (subs->hwptr_done >= runtime->buffer_size * stride)
subs->hwptr_done -= runtime->buffer_size * stride;
+
+ /* update delay with exact number of samples queued */
+ runtime->delay = subs->last_delay;
runtime->delay += frames;
+ subs->last_delay = runtime->delay;
+
+ /* realign last_frame_number */
+ subs->last_frame_number = usb_get_current_frame_number(subs->dev);
+ subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
+
spin_unlock_irqrestore(&subs->lock, flags);
urb->transfer_buffer_length = bytes;
if (period_elapsed)
@@ -1109,12 +1138,32 @@
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
int stride = runtime->frame_bits >> 3;
int processed = urb->transfer_buffer_length / stride;
+ int est_delay;
+
+ /* ignore the delay accounting when procssed=0 is given, i.e.
+ * silent payloads are procssed before handling the actual data
+ */
+ if (!processed)
+ return;
spin_lock_irqsave(&subs->lock, flags);
- if (processed > runtime->delay)
- runtime->delay = 0;
+ est_delay = snd_usb_pcm_delay(subs, runtime->rate);
+ /* update delay with exact number of samples played */
+ if (processed > subs->last_delay)
+ subs->last_delay = 0;
else
- runtime->delay -= processed;
+ subs->last_delay -= processed;
+ runtime->delay = subs->last_delay;
+
+ /*
+ * Report when delay estimate is off by more than 2ms.
+ * The error should be lower than 2ms since the estimate relies
+ * on two reads of a counter updated every ms.
+ */
+ if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
+ snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
+ est_delay, subs->last_delay);
+
spin_unlock_irqrestore(&subs->lock, flags);
}
@@ -1172,7 +1221,7 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- err = start_endpoints(subs);
+ err = start_endpoints(subs, 0);
if (err < 0)
return err;