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Date: Tue, 21 Aug 2012 11:25:40 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lrg@...com>, linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 3.6-rc3
Linus,
please pull sound fixes for v3.6-rc3 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-3.6
The topmost commit is 53e1719f3da0f095b8db1461bd12dd79f3246b84
----------------------------------------------------------------
Sound fixes for 3.6-rc3
This update became slightly bigger than usual for rc3, but most of the
commits are small and trivial. A large chunk is found for HD-audio
ca0132 codec, which is mostly a clean up of the specific code, to make
SPDIF working properly, and also in the new ASoC Arizona driver.
One important fix is for usb-audio Oops fix since 3.5. We still see
some EHCI related bandwidth problem, but usb-audio should be more
stabilized now.
Other than that, a Kconfig fix is spread over files, and various
HD-audio and ASoC fixes as usual, in addition to Julia's error path
fixes.
----------------------------------------------------------------
Chris Rattray (1):
ASoC: wm8994: Add missing dapm routes for WM8958 rev A
Dan Carpenter (2):
ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read()
sound: oss/sb_audio: prevent divide by zero bug
David Henningsson (5):
ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V
ALSA: hda - Fix panned "Beep Playback Switch"
ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch
ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx
ALSA: hda - don't create dysfunctional mixer controls for ca0132
Fabio Estevam (2):
ASoC: mxs: Fix the name of the SoC family
ASoC: imx-ssi: Remove mono support
James Ralston (1):
ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH
Julia Lawall (6):
ALSA: sound/atmel/abdac.c: fix error return code
ALSA: sound/atmel/ac97c.c: fix error return code
ALSA: sound/pci/ctxfi/ctatc.c: fix error return code
ALSA: sound/pci/sis7019.c: fix error return code
ALSA: sound/pci/rme9652/hdspm.c: fix error return code
ALSA: sound/ppc/snd_ps3.c: fix error return code
Mark Brown (9):
ASoC: wm8962: Don't duplicate bias level management in resume
ASoC: core: Upgrade the severity of probe deferral errors to dev_err()
MAINTAINERS: Add entries for Wolfson Arizona class devices
ASoC: wm5102: Add missing input PGA routes
ASoC: wm5110: Add missing input PGA routes
ASoC: jack: Always notify full jack status
ASoC: wm5102: Remove DRC2
ASoC: wm9712: Fix microphone source selection
ASoC: wm9712: Fix inverted capture volume
Mengdong Lin (1):
ALSA : hda - bug fix on checking the supported power states of a codec
Ondrej Zary (1):
ALSA: snd-als100: fix suspend/resume
Peter Ujfalusi (1):
ASoC: omap-mcbsp: Fix 6pin mux configuration
Randy Dunlap (1):
ALSA: fix pcm.h kernel-doc warning and notation
Sachin Kamat (1):
ASoC: Samsung: Fix build error
Scott Jiang (1):
ASoC: bfin: fix memory leak in sport3 controller driver
Takashi Iwai (8):
ALSA: hda - Add codec->pcm_format_first flag
ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items
ALSA: hda - Use the standard PCM ops for CA0132
ALSA: hda - Add missing SPDIF I/O setup for CA0132
ALSA: platform: Check CONFIG_PM_SLEEP instead of CONFIG_PM
ALSA: lx6464es: Add a missing error check
ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream
ALSA: hda - Fix leftover codec->power_transition
Vaibhav Bedia (1):
ASoC: Davinci: McASP: Flush the FIFO before enabling
Wang Xingchao (1):
ALSA: hda - fix Copyright debug message
---
MAINTAINERS | 7 +-
include/sound/pcm.h | 3 +-
sound/arm/pxa2xx-ac97.c | 4 +-
sound/atmel/abdac.c | 3 +-
sound/atmel/ac97c.c | 14 ++-
sound/drivers/aloop.c | 2 +-
sound/drivers/dummy.c | 2 +-
sound/drivers/pcsp/pcsp.c | 4 +-
sound/isa/als100.c | 2 +-
sound/oss/sb_audio.c | 4 +-
sound/pci/cs46xx/cs46xx_lib.c | 2 +-
sound/pci/ctxfi/ctatc.c | 4 +-
sound/pci/hda/hda_beep.c | 29 +++++--
sound/pci/hda/hda_codec.c | 73 ++++++++++------
sound/pci/hda/hda_codec.h | 1 +
sound/pci/hda/hda_intel.c | 9 ++
sound/pci/hda/hda_proc.c | 2 +-
sound/pci/hda/patch_ca0132.c | 174 ++++++++++++--------------------------
sound/pci/hda/patch_sigmatel.c | 9 +-
sound/pci/hda/patch_via.c | 8 ++
sound/pci/lx6464es/lx6464es.c | 2 +
sound/pci/rme9652/hdspm.c | 2 +-
sound/pci/sis7019.c | 5 +-
sound/ppc/powermac.c | 2 +-
sound/ppc/snd_ps3.c | 1 +
sound/soc/blackfin/bf6xx-sport.c | 7 ++
sound/soc/codecs/wm5102.c | 25 ++----
sound/soc/codecs/wm5110.c | 12 +++
sound/soc/codecs/wm8962.c | 15 ----
sound/soc/codecs/wm8994.c | 2 +
sound/soc/codecs/wm9712.c | 21 ++++-
sound/soc/davinci/davinci-mcasp.c | 10 ++-
sound/soc/fsl/imx-ssi.c | 5 +-
sound/soc/mxs/Kconfig | 2 +-
sound/soc/omap/mcbsp.c | 2 +-
sound/soc/samsung/pcm.c | 2 +-
sound/soc/soc-core.c | 10 ++-
sound/soc/soc-jack.c | 2 +-
sound/usb/endpoint.c | 4 -
sound/usb/pcm.c | 3 +
40 files changed, 257 insertions(+), 233 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index 61ad79e..ed5df6e 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -7670,23 +7670,28 @@ S: Supported
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 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
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 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
.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 @@ static struct platform_driver pxa2xx_ac97_driver = {
.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 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
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 @@ out_put_pclk:
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 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream,
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 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
if (!chip->regs) {
dev_dbg(&pdev->dev, "could not remap register memory\n");
+ retval = -ENOMEM;
goto err_ioremap;
}
@@ -1134,7 +1130,7 @@ err_snd_card_new:
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/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 @@ static int __devexit loopback_remove(struct platform_device *devptr)
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 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr)
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 @@ static void pcsp_stop_beep(struct snd_pcsp *chip)
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 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL);
#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 @@ static int __devinit snd_card_als100_probe(int dev,
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 @@ static int jazz16_audio_set_speed(int dev, int speed)
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 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
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 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
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 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
}
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 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
{
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..f560051 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec)
}
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 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
{
struct hda_codec *c;
struct hda_cvt_setup *p;
- unsigned int oldval, newval;
int type;
int i;
@@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
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;
@@ -3497,7 +3518,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg
{
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;
@@ -4433,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
* 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..7fbc1bc 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -861,6 +861,7 @@ struct hda_codec {
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 */
#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..60882c6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, CPT},"
"{Intel, PPT},"
"{Intel, LPT},"
+ "{Intel, LPT_LP},"
"{Intel, HPT},"
"{Intel, PBG},"
"{Intel, SCH},"
@@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ 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 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
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 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
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 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
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 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
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 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
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 @@ exit:
}
/*
- * PCM stuffs
+ * PCM callbacks
*/
-static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid,
- u32 stream_tag,
- int channel_id, int format)
+static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
{
- 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);
+ struct ca0132_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
}
-/*
- * PCM callbacks
- */
static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
@@ -515,10 +490,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
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,
@@ -526,92 +499,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
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_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_open(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->dig_out, stream_tag, 0, format);
-
- return 0;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
-static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->dig_out);
-
- return 0;
-}
-
-/*
- * Analog capture
- */
-static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_setup_stream(codec, spec->adcs[substream->number],
- stream_tag, 0, format);
-
- return 0;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
}
-static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
-
- ca0132_cleanup_stream(codec, spec->adcs[substream->number]);
-
- return 0;
+ return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
}
-/*
- * 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)
+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->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 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
.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 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
- .ops = {
- .prepare = ca0132_capture_pcm_prepare,
- .cleanup = ca0132_capture_pcm_cleanup
- },
};
static struct hda_pcm_stream ca0132_pcm_digital_playback = {
@@ -641,6 +564,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = {
.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 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
- .ops = {
- .prepare = ca0132_dig_capture_pcm_prepare,
- .cleanup = ca0132_dig_capture_pcm_cleanup
- },
};
static int ca0132_build_pcms(struct hda_codec *codec)
@@ -928,18 +849,16 @@ static int ca0132_build_controls(struct hda_codec *codec)
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 @@ static void ca0132_config(struct hda_codec *codec)
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 @@ static void ca0132_config(struct hda_codec *codec)
/* 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..ea5775a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec)
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)
@@ -5748,7 +5749,6 @@ again:
/* 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 +5773,6 @@ again:
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 +5787,6 @@ again:
/* 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 +5794,9 @@ again:
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 @@ static int via_suspend(struct hda_codec *codec)
{
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/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 @@ static int __devinit lx_pcm_create(struct lx6464es *chip)
/* 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 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
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 @@ static int __devinit sis_chip_create(struct snd_card *card,
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 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr)
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 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
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 318c5ba..dfb7443 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create);
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 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
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(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
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 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
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(EQ4, "EQ4"),
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 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
{ 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 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
{ "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 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
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 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
{ "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 @@ static int wm8962_runtime_resume(struct device *dev)
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 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
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("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
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 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]);
/* 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 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
&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("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
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 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = {
{"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 mcasp_start_tx(struct davinci_audio_dev *dev)
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 int imx_ssi_dai_probe(struct snd_soc_dai *dai)
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 @@ menuconfig SND_MXS_SOC
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 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux)
{
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 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
}
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 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
}
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 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
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 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num)
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 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
}
/* 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/endpoint.c b/sound/usb/endpoint.c
index 0f647d2..c411812 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -821,10 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
if (++ep->use_count != 1)
return 0;
- /* just to be sure */
- deactivate_urbs(ep, 0, 1);
- wait_clear_urbs(ep);
-
ep->active_mask = 0;
ep->unlink_mask = 0;
ep->phase = 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a1298f3..62ec808 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -544,6 +544,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
subs->last_frame_number = 0;
runtime->delay = 0;
+ /* clear the pending deactivation on the target EPs */
+ deactivate_endpoints(subs);
+
/* 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)
--
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