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Message-ID: <s5h7hsgjolz.wl%tiwai@suse.de>
Date: Mon, 21 Dec 2009 17:09:12 +0100
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Andrew Morton <akpm@...ux-foundation.org>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes
Linus,
please pull sound fixes for v2.6.33-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following fixes. Most of them are small individual
fixes. A big chunk is for supporting a few new Realtek codec chips,
which are more or less compatible with older ones.
Thanks!
Takashi
===
Clemens Ladisch (1):
sound: sgio2audio/pdaudiocf/usb-audio: initialize PCM buffer
Daniel T Chen (1):
ALSA: hda: Set Front Mic to input vref 50% for Lenovo 3000 Y410
Einar Rünkaru (2):
ALSA: hda - Fixed internal mic initialization for Dell Vostro 1015
ALSA: hda - Make use of beep device found in Dell Vostro 1015n
Guennadi Liakhovetski (1):
ASoC: wm8974: fix a wrong bit definition
Hector Martin (3):
ALSA: HDA: simplify Aspire 8930G verb array
ALSA: HDA: remove useless mixers on Aspire 8930G
ALSA: HDA: add powersaving hook for Realtek
Jaroslav Kysela (1):
ALSA: hda/realtek: Remove extra .capsrc_nids initialization for ALC889_INTEL
Jon Smirl (1):
ASoC: Fix disable of SPDIF on STAC9766 codec
Julia Lawall (1):
ALSA: Use kzalloc for allocating only one thing
Kailang Yang (1):
ALSA: hda - More ALC663 fixes and support of compatible chips
Krzysztof Helt (2):
ALSA: fix incorrect rounding direction in snd_interval_ratnum()
ALSA: sbawe: fix memory detection
Kuninori Morimoto (1):
ASoC: ak4642: Add default return value in ak4642_modinit
Roel Kluin (1):
sound/oss/pss: Fix test of unsigned in pss_reset_dsp() and pss_download_boot()
Russell King (5):
ALSA: AACI: simplify codec rate information
ALSA: AACI: cleanup aaci_pcm_hw_params
ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params
ALSA: AACI: add double-rate support
ALSA: AACI: switch to per-pcm locking
Takashi Iwai (3):
ALSA: hda - Fix missing capsrc_nids for ALC88x
ALSA: hda - Fix quirk for Maxdata obook4-1
ALSA: aaci - Fix a typo
---
sound/arm/aaci.c | 177 +++++----------
sound/arm/aaci.h | 2 +-
sound/core/pcm_lib.c | 4 +-
sound/isa/msnd/msnd_midi.c | 2 +-
sound/isa/sb/emu8000.c | 6 +-
sound/mips/sgio2audio.c | 2 +-
sound/oss/pss.c | 6 +-
sound/pci/hda/patch_conexant.c | 43 +++-
sound/pci/hda/patch_realtek.c | 387 +++++++++++++++++++++++++++++---
sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 2 +-
sound/soc/codecs/ak4642.c | 2 +-
sound/soc/codecs/stac9766.c | 18 +--
sound/soc/codecs/wm8974.c | 2 +-
sound/usb/usbaudio.c | 2 +-
14 files changed, 467 insertions(+), 188 deletions(-)
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1497dce..c569986 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -172,14 +172,15 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
return v;
}
-static inline void aaci_chan_wait_ready(struct aaci_runtime *aacirun)
+static inline void
+aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask)
{
u32 val;
int timeout = 5000;
do {
val = readl(aacirun->base + AACI_SR);
- } while (val & (SR_TXB|SR_RXB) && timeout--);
+ } while (val & mask && timeout--);
}
@@ -208,8 +209,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
writel(0, aacirun->base + AACI_IE);
return;
}
- ptr = aacirun->ptr;
+ spin_lock(&aacirun->lock);
+
+ ptr = aacirun->ptr;
do {
unsigned int len = aacirun->fifosz;
u32 val;
@@ -217,9 +220,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
if (aacirun->bytes <= 0) {
aacirun->bytes += aacirun->period;
aacirun->ptr = ptr;
- spin_unlock(&aaci->lock);
+ spin_unlock(&aacirun->lock);
snd_pcm_period_elapsed(aacirun->substream);
- spin_lock(&aaci->lock);
+ spin_lock(&aacirun->lock);
}
if (!(aacirun->cr & CR_EN))
break;
@@ -245,7 +248,10 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
ptr = aacirun->start;
}
} while(1);
+
aacirun->ptr = ptr;
+
+ spin_unlock(&aacirun->lock);
}
if (mask & ISR_URINTR) {
@@ -263,6 +269,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
return;
}
+ spin_lock(&aacirun->lock);
+
ptr = aacirun->ptr;
do {
unsigned int len = aacirun->fifosz;
@@ -271,9 +279,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
if (aacirun->bytes <= 0) {
aacirun->bytes += aacirun->period;
aacirun->ptr = ptr;
- spin_unlock(&aaci->lock);
+ spin_unlock(&aacirun->lock);
snd_pcm_period_elapsed(aacirun->substream);
- spin_lock(&aaci->lock);
+ spin_lock(&aacirun->lock);
}
if (!(aacirun->cr & CR_EN))
break;
@@ -301,6 +309,8 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
} while (1);
aacirun->ptr = ptr;
+
+ spin_unlock(&aacirun->lock);
}
}
@@ -310,7 +320,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
u32 mask;
int i;
- spin_lock(&aaci->lock);
mask = readl(aaci->base + AACI_ALLINTS);
if (mask) {
u32 m = mask;
@@ -320,7 +329,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
}
}
}
- spin_unlock(&aaci->lock);
return mask ? IRQ_HANDLED : IRQ_NONE;
}
@@ -330,63 +338,6 @@ static irqreturn_t aaci_irq(int irq, void *devid)
/*
* ALSA support.
*/
-
-struct aaci_stream {
- unsigned char codec_idx;
- unsigned char rate_idx;
-};
-
-static struct aaci_stream aaci_streams[] = {
- [ACSTREAM_FRONT] = {
- .codec_idx = 0,
- .rate_idx = AC97_RATES_FRONT_DAC,
- },
- [ACSTREAM_SURROUND] = {
- .codec_idx = 0,
- .rate_idx = AC97_RATES_SURR_DAC,
- },
- [ACSTREAM_LFE] = {
- .codec_idx = 0,
- .rate_idx = AC97_RATES_LFE_DAC,
- },
-};
-
-static inline unsigned int aaci_rate_mask(struct aaci *aaci, int streamid)
-{
- struct aaci_stream *s = aaci_streams + streamid;
- return aaci->ac97_bus->codec[s->codec_idx]->rates[s->rate_idx];
-}
-
-static unsigned int rate_list[] = {
- 5512, 8000, 11025, 16000, 22050, 32000, 44100,
- 48000, 64000, 88200, 96000, 176400, 192000
-};
-
-/*
- * Double-rate rule: we can support double rate iff channels == 2
- * (unimplemented)
- */
-static int
-aaci_rule_rate_by_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule)
-{
- struct aaci *aaci = rule->private;
- unsigned int rate_mask = SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_5512;
- struct snd_interval *c = hw_param_interval(p, SNDRV_PCM_HW_PARAM_CHANNELS);
-
- switch (c->max) {
- case 6:
- rate_mask &= aaci_rate_mask(aaci, ACSTREAM_LFE);
- case 4:
- rate_mask &= aaci_rate_mask(aaci, ACSTREAM_SURROUND);
- case 2:
- rate_mask &= aaci_rate_mask(aaci, ACSTREAM_FRONT);
- }
-
- return snd_interval_list(hw_param_interval(p, rule->var),
- ARRAY_SIZE(rate_list), rate_list,
- rate_mask);
-}
-
static struct snd_pcm_hardware aaci_hw_info = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
@@ -400,10 +351,7 @@ static struct snd_pcm_hardware aaci_hw_info = {
*/
.formats = SNDRV_PCM_FMTBIT_S16_LE,
- /* should this be continuous or knot? */
- .rates = SNDRV_PCM_RATE_CONTINUOUS,
- .rate_max = 48000,
- .rate_min = 4000,
+ /* rates are setup from the AC'97 codec */
.channels_min = 2,
.channels_max = 6,
.buffer_bytes_max = 64 * 1024,
@@ -423,6 +371,12 @@ static int __aaci_pcm_open(struct aaci *aaci,
aacirun->substream = substream;
runtime->private_data = aacirun;
runtime->hw = aaci_hw_info;
+ runtime->hw.rates = aacirun->pcm->rates;
+ snd_pcm_limit_hw_rates(runtime);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ aacirun->pcm->r[1].slots)
+ snd_ac97_pcm_double_rate_rules(runtime);
/*
* FIXME: ALSA specifies fifo_size in bytes. If we're in normal
@@ -433,17 +387,6 @@ static int __aaci_pcm_open(struct aaci *aaci,
*/
runtime->hw.fifo_size = aaci->fifosize * 2;
- /*
- * Add rule describing hardware rate dependency
- * on the number of channels.
- */
- ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
- aaci_rule_rate_by_channels, aaci,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- SNDRV_PCM_HW_PARAM_RATE, -1);
- if (ret)
- goto out;
-
ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED,
DRIVER_NAME, aaci);
if (ret)
@@ -507,18 +450,22 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
err = snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(params));
- if (err < 0)
- goto out;
+ if (err >= 0) {
+ unsigned int rate = params_rate(params);
+ int dbl = rate > 48000;
- err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
- params_channels(params),
- aacirun->pcm->r[0].slots);
- if (err)
- goto out;
+ err = snd_ac97_pcm_open(aacirun->pcm, rate,
+ params_channels(params),
+ aacirun->pcm->r[dbl].slots);
- aacirun->pcm_open = 1;
+ aacirun->pcm_open = err == 0;
+ aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+ aacirun->fifosz = aaci->fifosize * 4;
+
+ if (aacirun->cr & CR_COMPACT)
+ aacirun->fifosz >>= 1;
+ }
- out:
return err;
}
@@ -527,7 +474,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct aaci_runtime *aacirun = runtime->private_data;
- aacirun->start = (void *)runtime->dma_area;
+ aacirun->start = runtime->dma_area;
aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream);
aacirun->ptr = aacirun->start;
aacirun->period =
@@ -627,14 +574,9 @@ static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
* Enable FIFO, compact mode, 16 bits per sample.
* FIXME: double rate slots?
*/
- if (ret >= 0) {
- aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+ if (ret >= 0)
aacirun->cr |= channels_to_txmask[channels];
- aacirun->fifosz = aaci->fifosize * 4;
- if (aacirun->cr & CR_COMPACT)
- aacirun->fifosz >>= 1;
- }
return ret;
}
@@ -646,7 +588,7 @@ static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun)
ie &= ~(IE_URIE|IE_TXIE);
writel(ie, aacirun->base + AACI_IE);
aacirun->cr &= ~CR_EN;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_TXB);
writel(aacirun->cr, aacirun->base + AACI_TXCR);
}
@@ -654,7 +596,7 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
{
u32 ie;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_TXB);
aacirun->cr |= CR_EN;
ie = readl(aacirun->base + AACI_IE);
@@ -665,12 +607,12 @@ static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
unsigned long flags;
int ret = 0;
- spin_lock_irqsave(&aaci->lock, flags);
+ spin_lock_irqsave(&aacirun->lock, flags);
+
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
aaci_pcm_playback_start(aacirun);
@@ -697,7 +639,8 @@ static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cm
default:
ret = -EINVAL;
}
- spin_unlock_irqrestore(&aaci->lock, flags);
+
+ spin_unlock_irqrestore(&aacirun->lock, flags);
return ret;
}
@@ -721,18 +664,10 @@ static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
int ret;
ret = aaci_pcm_hw_params(substream, aacirun, params);
-
- if (ret >= 0) {
- aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
-
+ if (ret >= 0)
/* Line in record: slot 3 and 4 */
aacirun->cr |= CR_SL3 | CR_SL4;
- aacirun->fifosz = aaci->fifosize * 4;
-
- if (aacirun->cr & CR_COMPACT)
- aacirun->fifosz >>= 1;
- }
return ret;
}
@@ -740,7 +675,7 @@ static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun)
{
u32 ie;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_RXB);
ie = readl(aacirun->base + AACI_IE);
ie &= ~(IE_ORIE | IE_RXIE);
@@ -755,7 +690,7 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun)
{
u32 ie;
- aaci_chan_wait_ready(aacirun);
+ aaci_chan_wait_ready(aacirun, SR_RXB);
#ifdef DEBUG
/* RX Timeout value: bits 28:17 in RXCR */
@@ -772,12 +707,11 @@ static void aaci_pcm_capture_start(struct aaci_runtime *aacirun)
static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
unsigned long flags;
int ret = 0;
- spin_lock_irqsave(&aaci->lock, flags);
+ spin_lock_irqsave(&aacirun->lock, flags);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -806,7 +740,7 @@ static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd
ret = -EINVAL;
}
- spin_unlock_irqrestore(&aaci->lock, flags);
+ spin_unlock_irqrestore(&aacirun->lock, flags);
return ret;
}
@@ -889,6 +823,12 @@ static struct ac97_pcm ac97_defs[] __devinitdata = {
(1 << AC97_SLOT_PCM_SRIGHT) |
(1 << AC97_SLOT_LFE),
},
+ [1] = {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT) |
+ (1 << AC97_SLOT_PCM_LEFT_0) |
+ (1 << AC97_SLOT_PCM_RIGHT_0),
+ },
},
},
[1] = { /* PCM in */
@@ -1001,7 +941,6 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
aaci = card->private_data;
mutex_init(&aaci->ac97_sem);
- spin_lock_init(&aaci->lock);
aaci->card = card;
aaci->dev = dev;
@@ -1028,7 +967,7 @@ static int __devinit aaci_init_pcm(struct aaci *aaci)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops);
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
- NULL, 0, 64 * 104);
+ NULL, 0, 64 * 1024);
}
return ret;
@@ -1088,12 +1027,14 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
/*
* Playback uses AACI channel 0
*/
+ spin_lock_init(&aaci->playback.lock);
aaci->playback.base = aaci->base + AACI_CSCH1;
aaci->playback.fifo = aaci->base + AACI_DR1;
/*
* Capture uses AACI channel 0
*/
+ spin_lock_init(&aaci->capture.lock);
aaci->capture.base = aaci->base + AACI_CSCH1;
aaci->capture.fifo = aaci->base + AACI_DR1;
diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h
index 924f69c..6a4a2ee 100644
--- a/sound/arm/aaci.h
+++ b/sound/arm/aaci.h
@@ -202,6 +202,7 @@
struct aaci_runtime {
void __iomem *base;
void __iomem *fifo;
+ spinlock_t lock;
struct ac97_pcm *pcm;
int pcm_open;
@@ -232,7 +233,6 @@ struct aaci {
struct snd_ac97 *ac97;
u32 maincr;
- spinlock_t lock;
struct aaci_runtime playback;
struct aaci_runtime capture;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 30f4108..a27545b 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -758,7 +758,7 @@ int snd_interval_ratnum(struct snd_interval *i,
int diff;
if (q == 0)
q = 1;
- den = div_down(num, q);
+ den = div_up(num, q);
if (den < rats[k].den_min)
continue;
if (den > rats[k].den_max)
@@ -794,7 +794,7 @@ int snd_interval_ratnum(struct snd_interval *i,
i->empty = 1;
return -EINVAL;
}
- den = div_up(num, q);
+ den = div_down(num, q);
if (den > rats[k].den_max)
continue;
if (den < rats[k].den_min)
diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c
index cb9aa4c..4be562b 100644
--- a/sound/isa/msnd/msnd_midi.c
+++ b/sound/isa/msnd/msnd_midi.c
@@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *card, int device)
err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi);
if (err < 0)
return err;
- mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL);
+ mpu = kzalloc(sizeof(*mpu), GFP_KERNEL);
if (mpu == NULL) {
snd_device_free(card, rmidi);
return -ENOMEM;
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 96678d5..751762f 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -393,8 +393,6 @@ size_dram(struct snd_emu8000 *emu)
while (size < EMU8000_MAX_DRAM) {
- size += 512 * 1024; /* increment 512kbytes */
-
/* Write a unique data on the test address.
* if the address is out of range, the data is written on
* 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is
@@ -414,7 +412,9 @@ size_dram(struct snd_emu8000 *emu)
/*snd_emu8000_read_wait(emu);*/
EMU8000_SMLD_READ(emu); /* discard stale data */
if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2)
- break; /* we must have wrapped around */
+ break; /* no memory at this address */
+
+ size += 512 * 1024; /* increment 512kbytes */
snd_emu8000_read_wait(emu);
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 8691f4c..f1d9d16 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -609,7 +609,7 @@ static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
/* alloc virtual 'dma' area */
if (runtime->dma_area)
vfree(runtime->dma_area);
- runtime->dma_area = vmalloc(size);
+ runtime->dma_area = vmalloc_user(size);
if (runtime->dma_area == NULL)
return -ENOMEM;
runtime->dma_bytes = size;
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 83f5ee2..e19dd5d 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -269,7 +269,7 @@ static int pss_reset_dsp(pss_confdata * devc)
unsigned long i, limit = jiffies + HZ/10;
outw(0x2000, REG(PSS_CONTROL));
- for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+ for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
inw(REG(PSS_CONTROL));
outw(0x0000, REG(PSS_CONTROL));
return 1;
@@ -369,11 +369,11 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size
outw(0, REG(PSS_DATA));
limit = jiffies + HZ/10;
- for (i = 0; i < 32768 && (limit - jiffies >= 0); i++)
+ for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
val = inw(REG(PSS_STATUS));
limit = jiffies + HZ/10;
- for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+ for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++)
{
val = inw(REG(PSS_STATUS));
if (val & 0x4000)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index a09c03c..c578c28 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -29,6 +29,7 @@
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_beep.h"
#define CXT_PIN_DIR_IN 0x00
#define CXT_PIN_DIR_OUT 0x01
@@ -111,6 +112,7 @@ struct conexant_spec {
unsigned int dell_automute;
unsigned int port_d_mode;
unsigned char ext_mic_bias;
+ unsigned int dell_vostro;
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -476,6 +478,7 @@ static void conexant_free(struct hda_codec *codec)
snd_array_free(&spec->jacks);
}
#endif
+ snd_hda_detach_beep_device(codec);
kfree(codec->spec);
}
@@ -2109,9 +2112,12 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int val;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+ int inout = (kcontrol->private_value & 0x100) ?
+ AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT;
- val = snd_hda_codec_read(codec, 0x17, 0,
- AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT);
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE, inout);
ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN;
return 0;
@@ -2123,6 +2129,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
unsigned int idx;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+ int inout = (kcontrol->private_value & 0x100) ?
+ AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT;
if (!imux->num_items)
return 0;
@@ -2130,9 +2139,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- snd_hda_codec_write_cache(codec, 0x17, 0,
+ snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT |
+ AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout |
imux->items[idx].index);
return 1;
@@ -2201,10 +2210,11 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Analog Mic Boost Capture Enum",
+ .name = "Ext Mic Boost Capture Enum",
.info = cxt5066_mic_boost_mux_enum_info,
.get = cxt5066_mic_boost_mux_enum_get,
.put = cxt5066_mic_boost_mux_enum_put,
+ .private_value = 0x17,
},
HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others),
@@ -2212,6 +2222,19 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
{}
};
+static struct snd_kcontrol_new cxt5066_vostro_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Int Mic Boost Capture Enum",
+ .info = cxt5066_mic_boost_mux_enum_info,
+ .get = cxt5066_mic_boost_mux_enum_get,
+ .put = cxt5066_mic_boost_mux_enum_put,
+ .private_value = 0x23 | 0x100,
+ },
+ HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+ {}
+};
+
static struct hda_verb cxt5066_init_verbs[] = {
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
@@ -2397,11 +2420,16 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
/* initialize jack-sensing, too */
static int cxt5066_init(struct hda_codec *codec)
{
+ struct conexant_spec *spec = codec->spec;
+
snd_printdd("CXT5066: init\n");
conexant_init(codec);
if (codec->patch_ops.unsol_event) {
cxt5066_hp_automute(codec);
- cxt5066_automic(codec);
+ if (spec->dell_vostro)
+ cxt5066_vostro_automic(codec);
+ else
+ cxt5066_automic(codec);
}
return 0;
}
@@ -2500,7 +2528,10 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->init_verbs[0] = cxt5066_init_verbs_vostro;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+ spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers;
spec->port_d_mode = 0;
+ spec->dell_vostro = 1;
+ snd_hda_attach_beep_device(codec, 0x13);
/* no S/PDIF out */
spec->multiout.dig_out_nid = 0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index aeed4cc..c746505 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -131,8 +131,8 @@ enum {
enum {
ALC269_BASIC,
ALC269_QUANTA_FL1,
- ALC269_ASUS_EEEPC_P703,
- ALC269_ASUS_EEEPC_P901,
+ ALC269_ASUS_AMIC,
+ ALC269_ASUS_DMIC,
ALC269_FUJITSU,
ALC269_LIFEBOOK,
ALC269_AUTO,
@@ -188,6 +188,8 @@ enum {
ALC663_ASUS_MODE4,
ALC663_ASUS_MODE5,
ALC663_ASUS_MODE6,
+ ALC663_ASUS_MODE7,
+ ALC663_ASUS_MODE8,
ALC272_DELL,
ALC272_DELL_ZM1,
ALC272_SAMSUNG_NC10,
@@ -335,6 +337,9 @@ struct alc_spec {
/* hooks */
void (*init_hook)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ void (*power_hook)(struct hda_codec *codec, int power);
+#endif
/* for pin sensing */
unsigned int sense_updated: 1;
@@ -386,6 +391,7 @@ struct alc_config_preset {
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_amp_list *loopbacks;
+ void (*power_hook)(struct hda_codec *codec, int power);
#endif
};
@@ -898,6 +904,7 @@ static void setup_preset(struct hda_codec *codec,
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->power_hook = preset->power_hook;
spec->loopback.amplist = preset->loopbacks;
#endif
@@ -1663,9 +1670,6 @@ static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
/* some bit here disables the other DACs. Init=0x4900 */
{0x20, AC_VERB_SET_COEF_INDEX, 0x08},
{0x20, AC_VERB_SET_PROC_COEF, 0x0000},
-/* Enable amplifiers */
- {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
/* DMIC fix
* This laptop has a stereo digital microphone. The mics are only 1cm apart
* which makes the stereo useless. However, either the mic or the ALC889
@@ -1778,6 +1782,25 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1808,6 +1831,16 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[2] = 0x1b;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void alc889_power_eapd(struct hda_codec *codec, int power)
+{
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+ snd_hda_codec_write(codec, 0x15, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+}
+#endif
+
/*
* ALC880 3-stack model
*
@@ -3601,12 +3634,29 @@ static void alc_free(struct hda_codec *codec)
snd_hda_detach_beep_device(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc_suspend(struct hda_codec *codec, pm_message_t state)
+{
+ struct alc_spec *spec = codec->spec;
+ if (spec && spec->power_hook)
+ spec->power_hook(codec, 0);
+ return 0;
+}
+#endif
+
#ifdef SND_HDA_NEEDS_RESUME
static int alc_resume(struct hda_codec *codec)
{
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct alc_spec *spec = codec->spec;
+#endif
codec->patch_ops.init(codec);
snd_hda_codec_resume_amp(codec);
snd_hda_codec_resume_cache(codec);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (spec && spec->power_hook)
+ spec->power_hook(codec, 1);
+#endif
return 0;
}
#endif
@@ -3623,6 +3673,7 @@ static struct hda_codec_ops alc_patch_ops = {
.resume = alc_resume,
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .suspend = alc_suspend,
.check_power_status = alc_check_power_status,
#endif
};
@@ -8919,7 +8970,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
- SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
@@ -9282,6 +9333,7 @@ static struct alc_config_preset alc882_presets[] = {
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
@@ -9378,10 +9430,11 @@ static struct alc_config_preset alc882_presets[] = {
.init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_8930G] = {
- .mixers = { alc888_base_mixer,
+ .mixers = { alc889_acer_aspire_8930g_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
- alc889_acer_aspire_8930g_verbs },
+ alc889_acer_aspire_8930g_verbs,
+ alc889_eapd_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
@@ -9398,6 +9451,9 @@ static struct alc_config_preset alc882_presets[] = {
.unsol_event = alc_automute_amp_unsol_event,
.setup = alc889_acer_aspire_8930g_setup,
.init_hook = alc_automute_amp,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .power_hook = alc889_power_eapd,
+#endif
},
[ALC888_ACER_ASPIRE_7730G] = {
.mixers = { alc883_3ST_6ch_mixer,
@@ -9428,6 +9484,7 @@ static struct alc_config_preset alc882_presets[] = {
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
@@ -9489,6 +9546,7 @@ static struct alc_config_preset alc882_presets[] = {
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt),
+ .capsrc_nids = alc883_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_lenovo_101e_capture_source,
@@ -9668,6 +9726,7 @@ static struct alc_config_preset alc882_presets[] = {
alc880_gpio1_init_verbs },
.adc_nids = alc883_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .capsrc_nids = alc883_capsrc_nids,
.dac_nids = alc883_dac_nids,
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.channel_mode = alc889A_mb31_6ch_modes,
@@ -10678,6 +10737,13 @@ static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
{}
};
+static struct hda_verb alc262_lenovo_3000_init_verbs[] = {
+ /* Front Mic pin: input vref at 50% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {}
+};
+
static struct hda_input_mux alc262_fujitsu_capture_source = {
.num_items = 3,
.items = {
@@ -11720,7 +11786,8 @@ static struct alc_config_preset alc262_presets[] = {
[ALC262_LENOVO_3000] = {
.mixers = { alc262_lenovo_3000_mixer },
.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
- alc262_lenovo_3000_unsol_verbs },
+ alc262_lenovo_3000_unsol_verbs,
+ alc262_lenovo_3000_init_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
@@ -12857,7 +12924,7 @@ static int patch_alc268(struct hda_codec *codec)
int board_config;
int i, has_beep, err;
- spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
@@ -13232,10 +13299,12 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
/* toggle speaker-output according to the hp-jack state */
static void alc269_speaker_automute(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
+ unsigned int nid = spec->autocfg.hp_pins[0];
unsigned int present;
unsigned char bits;
- present = snd_hda_jack_detect(codec, 0x15);
+ present = snd_hda_jack_detect(codec, nid);
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -13460,8 +13529,8 @@ static void alc269_auto_init(struct hda_codec *codec)
static const char *alc269_models[ALC269_MODEL_LAST] = {
[ALC269_BASIC] = "basic",
[ALC269_QUANTA_FL1] = "quanta",
- [ALC269_ASUS_EEEPC_P703] = "eeepc-p703",
- [ALC269_ASUS_EEEPC_P901] = "eeepc-p901",
+ [ALC269_ASUS_AMIC] = "asus-amic",
+ [ALC269_ASUS_DMIC] = "asus-dmic",
[ALC269_FUJITSU] = "fujitsu",
[ALC269_LIFEBOOK] = "lifebook",
[ALC269_AUTO] = "auto",
@@ -13470,18 +13539,41 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
static struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
- ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_EEEPC_P703),
- SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_EEEPC_P703),
+ ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80JT", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_ASUS_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_ASUS_AMIC),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
- ALC269_ASUS_EEEPC_P901),
+ ALC269_ASUS_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
- ALC269_ASUS_EEEPC_P901),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_ASUS_EEEPC_P901),
+ ALC269_ASUS_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_ASUS_DMIC),
+ SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_ASUS_DMIC),
SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
{}
@@ -13511,7 +13603,7 @@ static struct alc_config_preset alc269_presets[] = {
.setup = alc269_quanta_fl1_setup,
.init_hook = alc269_quanta_fl1_init_hook,
},
- [ALC269_ASUS_EEEPC_P703] = {
+ [ALC269_ASUS_AMIC] = {
.mixers = { alc269_eeepc_mixer },
.cap_mixer = alc269_epc_capture_mixer,
.init_verbs = { alc269_init_verbs,
@@ -13525,7 +13617,7 @@ static struct alc_config_preset alc269_presets[] = {
.setup = alc269_eeepc_amic_setup,
.init_hook = alc269_eeepc_inithook,
},
- [ALC269_ASUS_EEEPC_P901] = {
+ [ALC269_ASUS_DMIC] = {
.mixers = { alc269_eeepc_mixer },
.cap_mixer = alc269_epc_capture_mixer,
.init_verbs = { alc269_init_verbs,
@@ -16160,6 +16252,52 @@ static struct snd_kcontrol_new alc663_g50v_mixer[] = {
{ } /* end */
};
+static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc663_mode7_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc663_mode8_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
static struct snd_kcontrol_new alc662_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -16447,6 +16585,45 @@ static struct hda_verb alc272_dell_init_verbs[] = {
{}
};
+static struct hda_verb alc663_mode7_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_mode8_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
static struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
@@ -16626,6 +16803,54 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
}
}
+static void alc663_two_hp_m7_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present1, present2;
+
+ present1 = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ present2 = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+
+ if (present1 || present2) {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ } else {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ }
+}
+
+static void alc663_two_hp_m8_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present1, present2;
+
+ present1 = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ present2 = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+
+ if (present1 || present2) {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ } else {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ }
+}
+
static void alc663_m51va_unsol_event(struct hda_codec *codec,
unsigned int res)
{
@@ -16645,7 +16870,7 @@ static void alc663_m51va_setup(struct hda_codec *codec)
spec->ext_mic.pin = 0x18;
spec->ext_mic.mux_idx = 0;
spec->int_mic.pin = 0x12;
- spec->int_mic.mux_idx = 1;
+ spec->int_mic.mux_idx = 9;
spec->auto_mic = 1;
}
@@ -16657,7 +16882,17 @@ static void alc663_m51va_inithook(struct hda_codec *codec)
/* ***************** Mode1 ******************************/
#define alc663_mode1_unsol_event alc663_m51va_unsol_event
-#define alc663_mode1_setup alc663_m51va_setup
+
+static void alc663_mode1_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
+}
+
#define alc663_mode1_inithook alc663_m51va_inithook
/* ***************** Mode2 ******************************/
@@ -16674,7 +16909,7 @@ static void alc662_mode2_unsol_event(struct hda_codec *codec,
}
}
-#define alc662_mode2_setup alc663_m51va_setup
+#define alc662_mode2_setup alc663_mode1_setup
static void alc662_mode2_inithook(struct hda_codec *codec)
{
@@ -16695,7 +16930,7 @@ static void alc663_mode3_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode3_setup alc663_m51va_setup
+#define alc663_mode3_setup alc663_mode1_setup
static void alc663_mode3_inithook(struct hda_codec *codec)
{
@@ -16716,7 +16951,7 @@ static void alc663_mode4_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode4_setup alc663_m51va_setup
+#define alc663_mode4_setup alc663_mode1_setup
static void alc663_mode4_inithook(struct hda_codec *codec)
{
@@ -16737,7 +16972,7 @@ static void alc663_mode5_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode5_setup alc663_m51va_setup
+#define alc663_mode5_setup alc663_mode1_setup
static void alc663_mode5_inithook(struct hda_codec *codec)
{
@@ -16758,7 +16993,7 @@ static void alc663_mode6_unsol_event(struct hda_codec *codec,
}
}
-#define alc663_mode6_setup alc663_m51va_setup
+#define alc663_mode6_setup alc663_mode1_setup
static void alc663_mode6_inithook(struct hda_codec *codec)
{
@@ -16766,6 +17001,50 @@ static void alc663_mode6_inithook(struct hda_codec *codec)
alc_mic_automute(codec);
}
+/* ***************** Mode7 ******************************/
+static void alc663_mode7_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_two_hp_m7_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
+}
+
+#define alc663_mode7_setup alc663_mode1_setup
+
+static void alc663_mode7_inithook(struct hda_codec *codec)
+{
+ alc663_two_hp_m7_speaker_automute(codec);
+ alc_mic_automute(codec);
+}
+
+/* ***************** Mode8 ******************************/
+static void alc663_mode8_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_two_hp_m8_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
+}
+
+#define alc663_mode8_setup alc663_m51va_setup
+
+static void alc663_mode8_inithook(struct hda_codec *codec)
+{
+ alc663_two_hp_m8_speaker_automute(codec);
+ alc_mic_automute(codec);
+}
+
static void alc663_g71v_hp_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -16900,6 +17179,8 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
[ALC663_ASUS_MODE4] = "asus-mode4",
[ALC663_ASUS_MODE5] = "asus-mode5",
[ALC663_ASUS_MODE6] = "asus-mode6",
+ [ALC663_ASUS_MODE7] = "asus-mode7",
+ [ALC663_ASUS_MODE8] = "asus-mode8",
[ALC272_DELL] = "dell",
[ALC272_DELL_ZM1] = "dell-zm1",
[ALC272_SAMSUNG_NC10] = "samsung-nc10",
@@ -16916,12 +17197,22 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
+ SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
@@ -17205,6 +17496,36 @@ static struct alc_config_preset alc662_presets[] = {
.setup = alc663_mode6_setup,
.init_hook = alc663_mode6_inithook,
},
+ [ALC663_ASUS_MODE7] = {
+ .mixers = { alc663_mode7_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc663_mode7_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc663_mode7_unsol_event,
+ .setup = alc663_mode7_setup,
+ .init_hook = alc663_mode7_inithook,
+ },
+ [ALC663_ASUS_MODE8] = {
+ .mixers = { alc663_mode8_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc663_mode8_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc663_mode8_unsol_event,
+ .setup = alc663_mode8_setup,
+ .init_hook = alc663_mode8_inithook,
+ },
[ALC272_DELL] = {
.mixers = { alc663_m51va_mixer },
.cap_mixer = alc272_auto_capture_mixer,
@@ -17688,7 +18009,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
{ .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
{ .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
+ { .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 },
{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
+ { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
.patch = patch_alc861 },
{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index d057e64..5cfa608 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -51,7 +51,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s
return 0; /* already enough large */
vfree(runtime->dma_area);
}
- runtime->dma_area = vmalloc_32(size);
+ runtime->dma_area = vmalloc_32_user(size);
if (! runtime->dma_area)
return -ENOMEM;
runtime->dma_bytes = size;
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index b69861d..3ef16bb 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -470,7 +470,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642);
static int __init ak4642_modinit(void)
{
- int ret;
+ int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&ak4642_i2c_driver);
#endif
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index bbc72c2..81b8c9d 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -191,6 +191,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream,
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x1; /* enable variable rate audio */
+ vra &= ~0x4; /* disable SPDIF output */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
@@ -221,22 +222,6 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream,
return stac9766_ac97_write(codec, reg, runtime->rate);
}
-static int ac97_digital_trigger(struct snd_pcm_substream *substream,
- int cmd, struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
- unsigned short vra;
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_STOP:
- vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
- vra &= !0x04;
- stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
- break;
- }
- return 0;
-}
-
static int stac9766_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
@@ -315,7 +300,6 @@ static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
.prepare = ac97_digital_prepare,
- .trigger = ac97_digital_trigger,
};
struct snd_soc_dai stac9766_dai[] = {
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 81c57b5..a808675 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -47,7 +47,7 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = {
};
#define WM8974_POWER1_BIASEN 0x08
-#define WM8974_POWER1_BUFIOEN 0x10
+#define WM8974_POWER1_BUFIOEN 0x04
struct wm8974_priv {
struct snd_soc_codec codec;
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index b074a59..4963def 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -752,7 +752,7 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s
return 0; /* already large enough */
vfree(runtime->dma_area);
}
- runtime->dma_area = vmalloc(size);
+ runtime->dma_area = vmalloc_user(size);
if (!runtime->dma_area)
return -ENOMEM;
runtime->dma_bytes = size;
--
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