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authorJaroslav Kysela <perex@perex.cz>2010-01-08 08:26:34 (GMT)
committerJaroslav Kysela <perex@perex.cz>2010-01-08 08:26:34 (GMT)
commit1cb4f624ea38361b6397966470f0a1bed5532483 (patch)
tree418b05ddc854b09d64f7d5ee0c78875e42b5f151 /sound
parent444c1953d496d272208902ff7010dc70d1f887f0 (diff)
parent2c1f1895ef2aa8f0e5497893eff71304aef332e1 (diff)
downloadlinux-fsl-qoriq-1cb4f624ea38361b6397966470f0a1bed5532483.tar.xz
Merge branch 'master' of git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6 into fixes
Diffstat (limited to 'sound')
-rw-r--r--sound/arm/aaci.c180
-rw-r--r--sound/arm/aaci.h2
-rw-r--r--sound/arm/pxa2xx-ac97.c2
-rw-r--r--sound/core/Kconfig1
-rw-r--r--sound/core/pcm_lib.c4
-rw-r--r--sound/core/pcm_native.c8
-rw-r--r--sound/core/pcm_timer.c17
-rw-r--r--sound/isa/gus/gus_mem.c3
-rw-r--r--sound/isa/msnd/msnd_midi.c2
-rw-r--r--sound/isa/sb/emu8000.c6
-rw-r--r--sound/mips/sgio2audio.c2
-rw-r--r--sound/oss/pss.c6
-rw-r--r--sound/pci/cs5535audio/Makefile2
-rw-r--r--sound/pci/cs5535audio/cs5535audio.c1
-rw-r--r--sound/pci/cs5535audio/cs5535audio.h4
-rw-r--r--sound/pci/cs5535audio/cs5535audio_olpc.c26
-rw-r--r--sound/pci/hda/hda_beep.c16
-rw-r--r--sound/pci/hda/hda_codec.c20
-rw-r--r--sound/pci/hda/hda_codec.h1
-rw-r--r--sound/pci/hda/hda_hwdep.c7
-rw-r--r--sound/pci/hda/hda_intel.c22
-rw-r--r--sound/pci/hda/patch_analog.c16
-rw-r--r--sound/pci/hda/patch_cirrus.c22
-rw-r--r--sound/pci/hda/patch_realtek.c75
-rw-r--r--sound/pci/hda/patch_sigmatel.c40
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf.c3
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c2
-rw-r--r--sound/soc/codecs/ac97.c6
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/stac9766.c18
-rw-r--r--sound/soc/codecs/twl4030.c10
-rw-r--r--sound/soc/codecs/wm8350.c25
-rw-r--r--sound/soc/codecs/wm8510.c14
-rw-r--r--sound/soc/codecs/wm8900.c2
-rw-r--r--sound/soc/codecs/wm8940.c14
-rw-r--r--sound/soc/codecs/wm8974.c16
-rw-r--r--sound/soc/codecs/wm9712.c3
-rw-r--r--sound/soc/imx/mx27vis_wm8974.c3
-rw-r--r--sound/soc/omap/sdp3430.c6
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec.c2
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec.h2
-rw-r--r--sound/soc/sh/fsi-ak4642.c30
-rw-r--r--sound/soc/sh/fsi.c2
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/usb/usbaudio.c2
45 files changed, 337 insertions, 312 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1497dce..656e474 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)
@@ -498,6 +441,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
int err;
+ struct aaci *aaci = substream->private_data;
aaci_pcm_hw_free(substream);
if (aacirun->pcm_open) {
@@ -507,18 +451,22 @@ 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 +475,7 @@ 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 =
@@ -613,7 +561,6 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream)
static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
unsigned int channels = params_channels(params);
int ret;
@@ -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;
}
@@ -716,23 +659,14 @@ static struct snd_pcm_ops aaci_playback_ops = {
static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct aaci *aaci = substream->private_data;
struct aaci_runtime *aacirun = substream->runtime->private_data;
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 +674,7 @@ 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 +689,7 @@ 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 +706,11 @@ 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 +739,7 @@ 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 +822,12 @@ 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 +940,6 @@ 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 +966,7 @@ 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 +1026,14 @@ 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/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index b4b48af..5d94118 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -159,7 +159,7 @@ static int pxa2xx_ac97_resume(struct device *dev)
return ret;
}
-static struct dev_pm_ops pxa2xx_ac97_pm_ops = {
+static const struct dev_pm_ops pxa2xx_ac97_pm_ops = {
.suspend = pxa2xx_ac97_suspend,
.resume = pxa2xx_ac97_resume,
};
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index c15682a..475455c 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -5,6 +5,7 @@ config SND_TIMER
config SND_PCM
tristate
select SND_TIMER
+ select GCD
config SND_HWDEP
tristate
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 30f4108..a27545b2 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/core/pcm_native.c b/sound/core/pcm_native.c
index 29ab46a1..25b0641 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1918,13 +1918,13 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
hw->rate_min, hw->rate_max);
- if (err < 0)
- return err;
+ if (err < 0)
+ return err;
err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
hw->period_bytes_min, hw->period_bytes_max);
- if (err < 0)
- return err;
+ if (err < 0)
+ return err;
err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS,
hw->periods_min, hw->periods_max);
diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c
index ca8068b..b01d948 100644
--- a/sound/core/pcm_timer.c
+++ b/sound/core/pcm_timer.c
@@ -20,6 +20,7 @@
*/
#include <linux/time.h>
+#include <linux/gcd.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/timer.h>
@@ -28,22 +29,6 @@
* Timer functions
*/
-/* Greatest common divisor */
-static unsigned long gcd(unsigned long a, unsigned long b)
-{
- unsigned long r;
- if (a < b) {
- r = a;
- a = b;
- b = r;
- }
- while ((r = a % b) != 0) {
- a = b;
- b = r;
- }
- return b;
-}
-
void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream)
{
unsigned long rate, mult, fsize, l, post;
diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c
index 661205c..af888a0 100644
--- a/sound/isa/gus/gus_mem.c
+++ b/sound/isa/gus/gus_mem.c
@@ -127,7 +127,8 @@ static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc,
!share_id[2] && !share_id[3])
return NULL;
for (block = alloc->first; block; block = block->next)
- if (!memcmp(share_id, block->share_id, sizeof(share_id)))
+ if (!memcmp(share_id, block->share_id,
+ sizeof(block->share_id)))
return block;
return NULL;
}
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/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index fda7a94..ccc6422 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -4,9 +4,7 @@
snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
-ifdef CONFIG_MGEODE_LX
snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.o
-endif
# Toplevel Module Dependency
obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 05f56e0..91e7faf 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -389,6 +389,7 @@ probefail_out:
static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
{
+ olpc_quirks_cleanup();
snd_card_free(pci_get_drvdata(pci));
pci_set_drvdata(pci, NULL);
}
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 7a298ac..51966d7 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -99,10 +99,11 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
int snd_cs5535audio_resume(struct pci_dev *pci);
#endif
-#if defined(CONFIG_OLPC) && defined(CONFIG_MGEODE_LX)
+#ifdef CONFIG_OLPC
void __devinit olpc_prequirks(struct snd_card *card,
struct snd_ac97_template *ac97);
int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97);
+void __devexit olpc_quirks_cleanup(void);
void olpc_analog_input(struct snd_ac97 *ac97, int on);
void olpc_mic_bias(struct snd_ac97 *ac97, int on);
@@ -128,6 +129,7 @@ static inline int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
{
return 0;
}
+static inline void olpc_quirks_cleanup(void) { }
static inline void olpc_analog_input(struct snd_ac97 *ac97, int on) { }
static inline void olpc_mic_bias(struct snd_ac97 *ac97, int on) { }
static inline void olpc_capture_open(struct snd_ac97 *ac97) { }
diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c
index 5c68143..50da49b 100644
--- a/sound/pci/cs5535audio/cs5535audio_olpc.c
+++ b/sound/pci/cs5535audio/cs5535audio_olpc.c
@@ -13,10 +13,13 @@
#include <sound/info.h>
#include <sound/control.h>
#include <sound/ac97_codec.h>
+#include <linux/gpio.h>
#include <asm/olpc.h>
#include "cs5535audio.h"
+#define DRV_NAME "cs5535audio-olpc"
+
/*
* OLPC has an additional feature on top of the regular AD1888 codec features.
* It has an Analog Input mode that is switched into (after disabling the
@@ -38,10 +41,7 @@ void olpc_analog_input(struct snd_ac97 *ac97, int on)
}
/* set Analog Input through GPIO */
- if (on)
- geode_gpio_set(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
- else
- geode_gpio_clear(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
+ gpio_set_value(OLPC_GPIO_MIC_AC, on);
}
/*
@@ -73,8 +73,7 @@ static int olpc_dc_info(struct snd_kcontrol *kctl,
static int olpc_dc_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
{
- v->value.integer.value[0] = geode_gpio_isset(OLPC_GPIO_MIC_AC,
- GPIO_OUTPUT_VAL);
+ v->value.integer.value[0] = gpio_get_value(OLPC_GPIO_MIC_AC);
return 0;
}
@@ -153,6 +152,12 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
if (!machine_is_olpc())
return 0;
+ if (gpio_request(OLPC_GPIO_MIC_AC, DRV_NAME)) {
+ printk(KERN_ERR DRV_NAME ": unable to allocate MIC GPIO\n");
+ return -EIO;
+ }
+ gpio_direction_output(OLPC_GPIO_MIC_AC, 0);
+
/* drop the original AD1888 HPF control */
memset(&elem, 0, sizeof(elem));
elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
@@ -169,11 +174,18 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
for (i = 0; i < ARRAY_SIZE(olpc_cs5535audio_ctls); i++) {
err = snd_ctl_add(card, snd_ctl_new1(&olpc_cs5535audio_ctls[i],
ac97->private_data));
- if (err < 0)
+ if (err < 0) {
+ gpio_free(OLPC_GPIO_MIC_AC);
return err;
+ }
}
/* turn off the mic by default */
olpc_mic_bias(ac97, 0);
return 0;
}
+
+void __devexit olpc_quirks_cleanup(void)
+{
+ gpio_free(OLPC_GPIO_MIC_AC);
+}
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 5fe34a8..e4581a42 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -42,7 +42,7 @@ static void snd_hda_generate_beep(struct work_struct *work)
return;
/* generate tone */
- snd_hda_codec_write_cache(codec, beep->nid, 0,
+ snd_hda_codec_write(codec, beep->nid, 0,
AC_VERB_SET_BEEP_CONTROL, beep->tone);
}
@@ -119,7 +119,7 @@ static void snd_hda_do_detach(struct hda_beep *beep)
beep->dev = NULL;
cancel_work_sync(&beep->beep_work);
/* turn off beep for sure */
- snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+ snd_hda_codec_write(beep->codec, beep->nid, 0,
AC_VERB_SET_BEEP_CONTROL, 0);
}
@@ -192,7 +192,7 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
beep->enabled = enable;
if (!enable) {
/* turn off beep */
- snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+ snd_hda_codec_write(beep->codec, beep->nid, 0,
AC_VERB_SET_BEEP_CONTROL, 0);
}
if (beep->mode == HDA_BEEP_MODE_SWREG) {
@@ -239,8 +239,12 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
mutex_init(&beep->mutex);
if (beep->mode == HDA_BEEP_MODE_ON) {
- beep->enabled = 1;
- snd_hda_do_register(&beep->register_work);
+ int err = snd_hda_do_attach(beep);
+ if (err < 0) {
+ kfree(beep);
+ codec->beep = NULL;
+ return err;
+ }
}
return 0;
@@ -253,7 +257,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
if (beep) {
cancel_work_sync(&beep->register_work);
cancel_delayed_work(&beep->unregister_work);
- if (beep->enabled)
+ if (beep->dev)
snd_hda_do_detach(beep);
codec->beep = NULL;
kfree(beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 9cfdb77..f98b47c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1086,11 +1086,6 @@ int snd_hda_codec_configure(struct hda_codec *codec)
if (err < 0)
return err;
}
- /* audio codec should override the mixer name */
- if (codec->afg || !*codec->bus->card->mixername)
- snprintf(codec->bus->card->mixername,
- sizeof(codec->bus->card->mixername),
- "%s %s", codec->vendor_name, codec->chip_name);
if (is_generic_config(codec)) {
err = snd_hda_parse_generic_codec(codec);
@@ -1109,6 +1104,11 @@ int snd_hda_codec_configure(struct hda_codec *codec)
patched:
if (!err && codec->patch_ops.unsol_event)
err = init_unsol_queue(codec->bus);
+ /* audio codec should override the mixer name */
+ if (!err && (codec->afg || !*codec->bus->card->mixername))
+ snprintf(codec->bus->card->mixername,
+ sizeof(codec->bus->card->mixername),
+ "%s %s", codec->vendor_name, codec->chip_name);
return err;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
@@ -1327,11 +1327,13 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
*/
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
{
- u32 pincap = snd_hda_query_pin_caps(codec, nid);
-
- if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
- snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ u32 pincap;
+ if (!codec->no_trigger_sense) {
+ pincap = snd_hda_query_pin_caps(codec, nid);
+ if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ }
return snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 1d541b7..0a770a2 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -817,6 +817,7 @@ struct hda_codec {
unsigned int pin_amp_workaround:1; /* pin out-amp takes index
* (e.g. Conexant codecs)
*/
+ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index d243286..40ccb41 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -24,6 +24,7 @@
#include <linux/compat.h>
#include <linux/mutex.h>
#include <linux/ctype.h>
+#include <linux/string.h>
#include <linux/firmware.h>
#include <sound/core.h>
#include "hda_codec.h"
@@ -428,8 +429,7 @@ static int parse_hints(struct hda_codec *codec, const char *buf)
char *key, *val;
struct hda_hint *hint;
- while (isspace(*buf))
- buf++;
+ buf = skip_spaces(buf);
if (!*buf || *buf == '#' || *buf == '\n')
return 0;
if (*buf == '=')
@@ -444,8 +444,7 @@ static int parse_hints(struct hda_codec *codec, const char *buf)
return -EINVAL;
}
*val++ = 0;
- while (isspace(*val))
- val++;
+ val = skip_spaces(val);
remove_trail_spaces(key);
remove_trail_spaces(val);
hint = get_hint(codec, key);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9b56f93..ec9c348 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -356,6 +356,7 @@ struct azx_dev {
*/
unsigned char stream_tag; /* assigned stream */
unsigned char index; /* stream index */
+ int device; /* last device number assigned to */
unsigned int opened :1;
unsigned int running :1;
@@ -1441,10 +1442,13 @@ static int __devinit azx_codec_configure(struct azx *chip)
*/
/* assign a stream for the PCM */
-static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
+static inline struct azx_dev *
+azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
{
int dev, i, nums;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ struct azx_dev *res = NULL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dev = chip->playback_index_offset;
nums = chip->playback_streams;
} else {
@@ -1453,10 +1457,15 @@ static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
}
for (i = 0; i < nums; i++, dev++)
if (!chip->azx_dev[dev].opened) {
- chip->azx_dev[dev].opened = 1;
- return &chip->azx_dev[dev];
+ res = &chip->azx_dev[dev];
+ if (res->device == substream->pcm->device)
+ break;
}
- return NULL;
+ if (res) {
+ res->opened = 1;
+ res->device = substream->pcm->device;
+ }
+ return res;
}
/* release the assigned stream */
@@ -1505,7 +1514,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
int err;
mutex_lock(&chip->open_mutex);
- azx_dev = azx_assign_device(chip, substream->stream);
+ azx_dev = azx_assign_device(chip, substream);
if (azx_dev == NULL) {
mutex_unlock(&chip->open_mutex);
return -EBUSY;
@@ -2322,6 +2331,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
* white/black-list for enable_msi
*/
static struct snd_pci_quirk msi_black_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
{}
};
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1a36137..69a941c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1186,6 +1186,8 @@ static int patch_ad1986a(struct hda_codec *codec)
*/
spec->multiout.no_share_stream = 1;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -1371,6 +1373,8 @@ static int patch_ad1983(struct hda_codec *codec)
codec->patch_ops = ad198x_patch_ops;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -1813,6 +1817,9 @@ static int patch_ad1981(struct hda_codec *codec)
codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
break;
}
+
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -3118,6 +3125,8 @@ static int patch_ad1988(struct hda_codec *codec)
#endif
spec->vmaster_nid = 0x04;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -3330,6 +3339,8 @@ static int patch_ad1884(struct hda_codec *codec)
codec->patch_ops = ad198x_patch_ops;
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -4287,6 +4298,8 @@ static int patch_ad1884a(struct hda_codec *codec)
break;
}
+ codec->no_trigger_sense = 1;
+
return 0;
}
@@ -4623,6 +4636,9 @@ static int patch_ad1882(struct hda_codec *codec)
spec->mixers[2] = ad1882_6stack_mixers;
break;
}
+
+ codec->no_trigger_sense = 1;
+
return 0;
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 4b200da..fe0423c 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -66,6 +66,7 @@ struct cs_spec {
/* available models */
enum {
CS420X_MBP55,
+ CS420X_IMAC27,
CS420X_AUTO,
CS420X_MODELS
};
@@ -827,7 +828,8 @@ static void cs_automute(struct hda_codec *codec)
AC_VERB_SET_PIN_WIDGET_CONTROL,
hp_present ? 0 : PIN_OUT);
}
- if (spec->board_config == CS420X_MBP55) {
+ if (spec->board_config == CS420X_MBP55 ||
+ spec->board_config == CS420X_IMAC27) {
unsigned int gpio = hp_present ? 0x02 : 0x08;
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, gpio);
@@ -1069,12 +1071,14 @@ static int cs_parse_auto_config(struct hda_codec *codec)
static const char *cs420x_models[CS420X_MODELS] = {
[CS420X_MBP55] = "mbp55",
+ [CS420X_IMAC27] = "imac27",
[CS420X_AUTO] = "auto",
};
static struct snd_pci_quirk cs420x_cfg_tbl[] = {
SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
+ SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),
{} /* terminator */
};
@@ -1097,8 +1101,23 @@ static struct cs_pincfg mbp55_pincfgs[] = {
{} /* terminator */
};
+static struct cs_pincfg imac27_pincfgs[] = {
+ { 0x09, 0x012b4050 },
+ { 0x0a, 0x90100140 },
+ { 0x0b, 0x90100142 },
+ { 0x0c, 0x018b3020 },
+ { 0x0d, 0x90a00110 },
+ { 0x0e, 0x400000f0 },
+ { 0x0f, 0x01cbe030 },
+ { 0x10, 0x014be060 },
+ { 0x12, 0x01ab9070 },
+ { 0x15, 0x400000f0 },
+ {} /* terminator */
+};
+
static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
[CS420X_MBP55] = mbp55_pincfgs,
+ [CS420X_IMAC27] = imac27_pincfgs,
};
static void fix_pincfg(struct hda_codec *codec, int model)
@@ -1128,6 +1147,7 @@ static int patch_cs420x(struct hda_codec *codec)
fix_pincfg(codec, spec->board_config);
switch (spec->board_config) {
+ case CS420X_IMAC27:
case CS420X_MBP55:
/* GPIO1 = headphones */
/* GPIO3 = speakers */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2d3f4f8..c746505 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -337,6 +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;
@@ -388,6 +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
};
@@ -900,6 +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
@@ -1665,9 +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
@@ -1780,6 +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;
@@ -1810,6 +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
*
@@ -3603,12 +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
@@ -3625,6 +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
};
@@ -9381,10 +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),
@@ -9401,6 +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,
@@ -10684,6 +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 = {
@@ -11726,7 +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,
@@ -12863,7 +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;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3d59f83..2291a83 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2104,6 +2104,7 @@ static unsigned int ref9205_pin_configs[12] = {
10280204
1028021F
10280228 (Dell Vostro 1500)
+ 10280229 (Dell Vostro 1700)
*/
static unsigned int dell_9205_m42_pin_configs[12] = {
0x0321101F, 0x03A11020, 0x400003FA, 0x90170310,
@@ -2189,6 +2190,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
"Dell Inspiron", STAC_9205_DELL_M44),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
"Dell Vostro 1500", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229,
+ "Dell Vostro 1700", STAC_9205_DELL_M42),
/* Gateway */
SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
@@ -3779,15 +3782,16 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
err = snd_hda_attach_beep_device(codec, nid);
if (err < 0)
return err;
- /* IDT/STAC codecs have linear beep tone parameter */
- codec->beep->linear_tone = 1;
- /* if no beep switch is available, make its own one */
- caps = query_amp_caps(codec, nid, HDA_OUTPUT);
- if (codec->beep &&
- !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) {
- err = stac92xx_beep_switch_ctl(codec);
- if (err < 0)
- return err;
+ if (codec->beep) {
+ /* IDT/STAC codecs have linear beep tone parameter */
+ codec->beep->linear_tone = 1;
+ /* if no beep switch is available, make its own one */
+ caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+ if (!(caps & AC_AMPCAP_MUTE)) {
+ err = stac92xx_beep_switch_ctl(codec);
+ if (err < 0)
+ return err;
+ }
}
}
#endif
@@ -4449,14 +4453,7 @@ static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
{
if (!nid)
return 0;
- /* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT
- * codecs behave wrongly when SET_PIN_SENSE is triggered, although
- * the pincap gives TRIG_REQ bit.
- */
- if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE)
- return 1;
- return 0;
+ return snd_hda_jack_detect(codec, nid);
}
static void stac92xx_line_out_detect(struct hda_codec *codec,
@@ -4958,6 +4955,7 @@ static int patch_stac9200(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac9200_pin_nids);
spec->pin_nids = stac9200_pin_nids;
@@ -5020,6 +5018,7 @@ static int patch_stac925x(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac925x_pin_nids);
spec->pin_nids = stac925x_pin_nids;
@@ -5104,6 +5103,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->slave_dig_outs = stac92hd73xx_slave_dig_outs;
spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids);
@@ -5251,6 +5251,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
spec->digbeep_nid = 0x21;
@@ -5414,6 +5415,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->patch_ops = stac92xx_patch_ops;
spec->num_pins = STAC92HD71BXX_NUM_PINS;
@@ -5657,6 +5659,7 @@ static int patch_stac922x(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac922x_pin_nids);
spec->pin_nids = stac922x_pin_nids;
@@ -5760,6 +5763,7 @@ static int patch_stac927x(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
codec->slave_dig_outs = stac927x_slave_dig_outs;
spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
@@ -5894,6 +5898,7 @@ static int patch_stac9205(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac9205_pin_nids);
spec->pin_nids = stac9205_pin_nids;
@@ -6049,6 +6054,7 @@ static int patch_stac9872(struct hda_codec *codec)
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
+ codec->no_trigger_sense = 1;
codec->spec = spec;
spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
spec->pin_nids = stac9872_pin_nids;
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 7717e01..edaa729 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -143,7 +143,8 @@ static int snd_pdacf_probe(struct pcmcia_device *link)
link->io.NumPorts1 = 16;
link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_FORCED_PULSE;
- // link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED;
+ /* FIXME: This driver should be updated to allow for dynamic IRQ sharing */
+ /* link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING | IRQ_FORCED_PULSE; */
link->irq.Handler = pdacf_interrupt;
link->conf.Attributes = CONF_ENABLE_IRQ;
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/ac97.c b/sound/soc/codecs/ac97.c
index 69bd0ac..a1bbe16 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n");
+ goto err;
+ }
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0)
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/twl4030.c b/sound/soc/codecs/twl4030.c
index 5f1681f..2a27f7b 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -26,7 +26,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
-#include <linux/i2c/twl4030.h>
+#include <linux/i2c/twl.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -175,7 +175,7 @@ static int twl4030_write(struct snd_soc_codec *codec,
{
twl4030_write_reg_cache(codec, reg, value);
if (likely(reg < TWL4030_REG_SW_SHADOW))
- return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
+ return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
reg);
else
return 0;
@@ -261,7 +261,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
do {
/* this takes a little while, so don't slam i2c */
udelay(2000);
- twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+ twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
TWL4030_REG_ANAMICL);
} while ((i++ < 100) &&
((byte & TWL4030_CNCL_OFFSET_START) ==
@@ -542,7 +542,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \
break; \
case SND_SOC_DAPM_POST_PMD: \
reg_val = twl4030_read_reg_cache(w->codec, reg); \
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \
+ twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \
reg_val & (~mask), \
reg); \
break; \
@@ -679,7 +679,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
twl4030->sysclk) + 1);
/* Bypass the reg_cache to mute the headset */
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+ twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
hs_gain & (~0x0f),
TWL4030_REG_HS_GAIN_SET);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f82125d..ebbf11b 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1340,9 +1340,10 @@ static int wm8350_resume(struct platform_device *pdev)
return 0;
}
-static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
+static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
{
struct wm8350_data *priv = data;
+ struct wm8350 *wm8350 = priv->codec.control_data;
u16 reg;
int report;
int mask;
@@ -1365,7 +1366,7 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
if (!jack->jack) {
dev_warn(wm8350->dev, "Jack interrupt called with no jack\n");
- return;
+ return IRQ_NONE;
}
/* Debounce */
@@ -1378,6 +1379,8 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
report = 0;
snd_soc_jack_report(jack->jack, report, jack->report);
+
+ return IRQ_HANDLED;
}
/**
@@ -1421,9 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
/* Sync status */
- wm8350_hp_jack_handler(wm8350, irq, priv);
-
- wm8350_unmask_irq(wm8350, irq);
+ wm8350_hp_jack_handler(irq, priv);
return 0;
}
@@ -1482,12 +1483,16 @@ static int wm8350_probe(struct platform_device *pdev)
wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+ /* Make sure jack detect is disabled to start off with */
+ wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
+ WM8350_JDL_ENA | WM8350_JDR_ENA);
+
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
- wm8350_hp_jack_handler, priv);
+ wm8350_hp_jack_handler, 0, "Left jack detect",
+ priv);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
- wm8350_hp_jack_handler, priv);
+ wm8350_hp_jack_handler, 0, "Right jack detect",
+ priv);
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
@@ -1516,8 +1521,6 @@ static int wm8350_remove(struct platform_device *pdev)
WM8350_JDL_ENA | WM8350_JDR_ENA);
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 265e68c..af8cb69 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -424,23 +424,23 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
/* filter coefficient */
switch (params_rate(params)) {
- case SNDRV_PCM_RATE_8000:
+ case 8000:
adn |= 0x5 << 1;
break;
- case SNDRV_PCM_RATE_11025:
+ case 11025:
adn |= 0x4 << 1;
break;
- case SNDRV_PCM_RATE_16000:
+ case 16000:
adn |= 0x3 << 1;
break;
- case SNDRV_PCM_RATE_22050:
+ case 22050:
adn |= 0x2 << 1;
break;
- case SNDRV_PCM_RATE_32000:
+ case 32000:
adn |= 0x1 << 1;
break;
- case SNDRV_PCM_RATE_44100:
- case SNDRV_PCM_RATE_48000:
+ case 44100:
+ case 48000:
break;
}
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index c9438dd..dbc368c 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -199,7 +199,7 @@ static void wm8900_reset(struct snd_soc_codec *codec)
snd_soc_write(codec, WM8900_REG_RESET, 0);
memcpy(codec->reg_cache, wm8900_reg_defaults,
- sizeof(codec->reg_cache));
+ sizeof(wm8900_reg_defaults));
}
static int wm8900_hp_event(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 3d850b9..31e39ff 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -378,23 +378,23 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
iface |= (1 << 9);
switch (params_rate(params)) {
- case SNDRV_PCM_RATE_8000:
+ case 8000:
addcntrl |= (0x5 << 1);
break;
- case SNDRV_PCM_RATE_11025:
+ case 11025:
addcntrl |= (0x4 << 1);
break;
- case SNDRV_PCM_RATE_16000:
+ case 16000:
addcntrl |= (0x3 << 1);
break;
- case SNDRV_PCM_RATE_22050:
+ case 22050:
addcntrl |= (0x2 << 1);
break;
- case SNDRV_PCM_RATE_32000:
+ case 32000:
addcntrl |= (0x1 << 1);
break;
- case SNDRV_PCM_RATE_44100:
- case SNDRV_PCM_RATE_48000:
+ case 44100:
+ case 48000:
break;
}
ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 81c57b5..8812751 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;
@@ -482,23 +482,23 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
/* filter coefficient */
switch (params_rate(params)) {
- case SNDRV_PCM_RATE_8000:
+ case 8000:
adn |= 0x5 << 1;
break;
- case SNDRV_PCM_RATE_11025:
+ case 11025:
adn |= 0x4 << 1;
break;
- case SNDRV_PCM_RATE_16000:
+ case 16000:
adn |= 0x3 << 1;
break;
- case SNDRV_PCM_RATE_22050:
+ case 22050:
adn |= 0x2 << 1;
break;
- case SNDRV_PCM_RATE_32000:
+ case 32000:
adn |= 0x1 << 1;
break;
- case SNDRV_PCM_RATE_44100:
- case SNDRV_PCM_RATE_48000:
+ case 44100:
+ case 48000:
break;
}
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 0ac1215..e237bf6 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -463,7 +463,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
{
u16 *cache = codec->reg_cache;
- soc_ac97_ops.write(codec->ac97, reg, val);
+ if (reg < 0x7c)
+ soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9712_reg)))
cache[reg] = val;
diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c
index 0267d2d..07d2a24 100644
--- a/sound/soc/imx/mx27vis_wm8974.c
+++ b/sound/soc/imx/mx27vis_wm8974.c
@@ -180,7 +180,8 @@ static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream)
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
/* disable the PLL */
- return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0);
+ return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG,
+ 0, 0);
}
/*
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index c071f96..3c85c0f 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -24,7 +24,7 @@
#include <linux/clk.h>
#include <linux/platform_device.h>
-#include <linux/i2c/twl4030.h>
+#include <linux/i2c/twl.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -321,11 +321,11 @@ static int __init sdp3430_soc_init(void)
*(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
/* Set TWL4030 GPIO6 as EXTMUTE signal */
- twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
+ twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
TWL4030_INTBR_PMBR1);
pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
- twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
+ twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
TWL4030_INTBR_PMBR1);
ret = platform_device_add(sdp3430_snd_device);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
index d441c3b..4984754 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -312,7 +312,7 @@ int simtec_audio_resume(struct device *dev)
return 0;
}
-struct dev_pm_ops simtec_audio_pmops = {
+const struct dev_pm_ops simtec_audio_pmops = {
.resume = simtec_audio_resume,
};
EXPORT_SYMBOL_GPL(simtec_audio_pmops);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h
index 2714203..e18faee 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.h
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.h
@@ -15,7 +15,7 @@ extern int simtec_audio_core_probe(struct platform_device *pdev,
extern int simtec_audio_remove(struct platform_device *pdev);
#ifdef CONFIG_PM
-extern struct dev_pm_ops simtec_audio_pmops;
+extern const struct dev_pm_ops simtec_audio_pmops;
#define simtec_audio_pm &simtec_audio_pmops
#else
#define simtec_audio_pm NULL
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index c7af097..5263ab1 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -42,42 +42,12 @@ static struct snd_soc_device fsi_snd_devdata = {
.codec_dev = &soc_codec_dev_ak4642,
};
-#define AK4642_BUS 0
-#define AK4642_ADR 0x12
-static int ak4642_add_i2c_device(void)
-{
- struct i2c_board_info info;
- struct i2c_adapter *adapter;
- struct i2c_client *client;
-
- memset(&info, 0, sizeof(struct i2c_board_info));
- info.addr = AK4642_ADR;
- strlcpy(info.type, "ak4642", I2C_NAME_SIZE);
-
- adapter = i2c_get_adapter(AK4642_BUS);
- if (!adapter) {
- printk(KERN_DEBUG "can't get i2c adapter\n");
- return -ENODEV;
- }
-
- client = i2c_new_device(adapter, &info);
- i2c_put_adapter(adapter);
- if (!client) {
- printk(KERN_DEBUG "can't add i2c device\n");
- return -ENODEV;
- }
-
- return 0;
-}
-
static struct platform_device *fsi_snd_device;
static int __init fsi_ak4642_init(void)
{
int ret = -ENOMEM;
- ak4642_add_i2c_device();
-
fsi_snd_device = platform_device_alloc("soc-audio", -1);
if (!fsi_snd_device)
goto out;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 9c49c11..42813b8 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -876,7 +876,7 @@ static int fsi_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
irq = platform_get_irq(pdev, 0);
- if (!res || !irq) {
+ if (!res || (int)irq <= 0) {
dev_err(&pdev->dev, "Not enough FSI platform resources.\n");
ret = -ENODEV;
goto exit;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ef8f282..0a6440c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1236,7 +1236,7 @@ static int soc_poweroff(struct device *dev)
return 0;
}
-static struct dev_pm_ops soc_pm_ops = {
+static const struct dev_pm_ops soc_pm_ops = {
.suspend = soc_suspend,
.resume = soc_resume,
.poweroff = soc_poweroff,
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;