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-rw-r--r--sound/arm/aaci.c285
-rw-r--r--sound/arm/aaci.h9
-rw-r--r--sound/core/jack.c1
-rw-r--r--sound/pci/au88x0/au88x0_core.c14
-rw-r--r--sound/pci/hda/hda_intel.c1
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_conexant.c68
-rw-r--r--sound/pci/hda/patch_hdmi.c5
-rw-r--r--sound/pci/hda/patch_realtek.c9
-rw-r--r--sound/pci/hda/patch_sigmatel.c15
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/soc/codecs/cx20442.c2
-rw-r--r--sound/soc/codecs/wm8903.c2
-rw-r--r--sound/soc/codecs/wm8903.h2
-rw-r--r--sound/soc/codecs/wm8978.c14
-rw-r--r--sound/soc/codecs/wm8994.c249
-rw-r--r--sound/soc/codecs/wm9081.c5
-rw-r--r--sound/soc/codecs/wm_hubs.c3
-rw-r--r--sound/soc/fsl/fsl_dma.c9
-rw-r--r--sound/soc/fsl/fsl_ssi.c9
-rw-r--r--sound/soc/fsl/mpc5200_dma.c24
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c9
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c9
-rw-r--r--sound/soc/imx/eukrea-tlv320.c2
-rw-r--r--sound/soc/omap/am3517evm.c2
-rw-r--r--sound/soc/pxa/e740_wm9705.c4
-rw-r--r--sound/soc/pxa/e750_wm9705.c4
-rw-r--r--sound/soc/pxa/e800_wm9712.c4
-rw-r--r--sound/soc/pxa/em-x270.c4
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c4
-rw-r--r--sound/soc/pxa/palm27x.c4
-rw-r--r--sound/soc/pxa/tosa.c4
-rw-r--r--sound/soc/pxa/zylonite.c4
-rw-r--r--sound/soc/soc-dapm.c25
-rw-r--r--sound/sparc/amd7930.c8
-rw-r--r--sound/sparc/cs4231.c16
-rw-r--r--sound/sparc/dbri.c8
-rw-r--r--sound/usb/caiaq/audio.c2
-rw-r--r--sound/usb/caiaq/midi.c2
-rw-r--r--sound/usb/card.c4
-rw-r--r--sound/usb/pcm.c7
-rw-r--r--sound/usb/usbaudio.h1
42 files changed, 570 insertions, 287 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 7c1fc64cb53d..d0cead38d5fb 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -210,6 +210,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
if (mask & ISR_RXINTR) {
struct aaci_runtime *aacirun = &aaci->capture;
+ bool period_elapsed = false;
void *ptr;
if (!aacirun->substream || !aacirun->start) {
@@ -222,15 +223,12 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
ptr = aacirun->ptr;
do {
- unsigned int len = aacirun->fifosz;
+ unsigned int len = aacirun->fifo_bytes;
u32 val;
if (aacirun->bytes <= 0) {
aacirun->bytes += aacirun->period;
- aacirun->ptr = ptr;
- spin_unlock(&aacirun->lock);
- snd_pcm_period_elapsed(aacirun->substream);
- spin_lock(&aacirun->lock);
+ period_elapsed = true;
}
if (!(aacirun->cr & CR_EN))
break;
@@ -260,6 +258,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
aacirun->ptr = ptr;
spin_unlock(&aacirun->lock);
+
+ if (period_elapsed)
+ snd_pcm_period_elapsed(aacirun->substream);
}
if (mask & ISR_URINTR) {
@@ -269,6 +270,7 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
if (mask & ISR_TXINTR) {
struct aaci_runtime *aacirun = &aaci->playback;
+ bool period_elapsed = false;
void *ptr;
if (!aacirun->substream || !aacirun->start) {
@@ -281,15 +283,12 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
ptr = aacirun->ptr;
do {
- unsigned int len = aacirun->fifosz;
+ unsigned int len = aacirun->fifo_bytes;
u32 val;
if (aacirun->bytes <= 0) {
aacirun->bytes += aacirun->period;
- aacirun->ptr = ptr;
- spin_unlock(&aacirun->lock);
- snd_pcm_period_elapsed(aacirun->substream);
- spin_lock(&aacirun->lock);
+ period_elapsed = true;
}
if (!(aacirun->cr & CR_EN))
break;
@@ -319,6 +318,9 @@ static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
aacirun->ptr = ptr;
spin_unlock(&aacirun->lock);
+
+ if (period_elapsed)
+ snd_pcm_period_elapsed(aacirun->substream);
}
}
@@ -361,7 +363,7 @@ static struct snd_pcm_hardware aaci_hw_info = {
/* rates are setup from the AC'97 codec */
.channels_min = 2,
- .channels_max = 6,
+ .channels_max = 2,
.buffer_bytes_max = 64 * 1024,
.period_bytes_min = 256,
.period_bytes_max = PAGE_SIZE,
@@ -369,12 +371,46 @@ static struct snd_pcm_hardware aaci_hw_info = {
.periods_max = PAGE_SIZE / 16,
};
-static int __aaci_pcm_open(struct aaci *aaci,
- struct snd_pcm_substream *substream,
- struct aaci_runtime *aacirun)
+/*
+ * We can support two and four channel audio. Unfortunately
+ * six channel audio requires a non-standard channel ordering:
+ * 2 -> FL(3), FR(4)
+ * 4 -> FL(3), FR(4), SL(7), SR(8)
+ * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required)
+ * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual)
+ * This requires an ALSA configuration file to correct.
+ */
+static int aaci_rule_channels(struct snd_pcm_hw_params *p,
+ struct snd_pcm_hw_rule *rule)
+{
+ static unsigned int channel_list[] = { 2, 4, 6 };
+ struct aaci *aaci = rule->private;
+ unsigned int mask = 1 << 0, slots;
+
+ /* pcms[0] is the our 5.1 PCM instance. */
+ slots = aaci->ac97_bus->pcms[0].r[0].slots;
+ if (slots & (1 << AC97_SLOT_PCM_SLEFT)) {
+ mask |= 1 << 1;
+ if (slots & (1 << AC97_SLOT_LFE))
+ mask |= 1 << 2;
+ }
+
+ return snd_interval_list(hw_param_interval(p, rule->var),
+ ARRAY_SIZE(channel_list), channel_list, mask);
+}
+
+static int aaci_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- int ret;
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun;
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ aacirun = &aaci->playback;
+ } else {
+ aacirun = &aaci->capture;
+ }
aacirun->substream = substream;
runtime->private_data = aacirun;
@@ -382,27 +418,37 @@ static int __aaci_pcm_open(struct aaci *aaci,
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);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw.channels_max = 6;
+
+ /* Add rule describing channel dependency. */
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ aaci_rule_channels, aaci,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (ret)
+ return ret;
+
+ if (aacirun->pcm->r[1].slots)
+ snd_ac97_pcm_double_rate_rules(runtime);
+ }
/*
- * FIXME: ALSA specifies fifo_size in bytes. If we're in normal
- * mode, each 32-bit word contains one sample. If we're in
- * compact mode, each 32-bit word contains two samples, effectively
- * halving the FIFO size. However, we don't know for sure which
- * we'll be using at this point. We set this to the lower limit.
+ * ALSA wants the byte-size of the FIFOs. As we only support
+ * 16-bit samples, this is twice the FIFO depth irrespective
+ * of whether it's in compact mode or not.
*/
- runtime->hw.fifo_size = aaci->fifosize * 2;
-
- ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED,
- DRIVER_NAME, aaci);
- if (ret)
- goto out;
-
- return 0;
+ runtime->hw.fifo_size = aaci->fifo_depth * 2;
+
+ mutex_lock(&aaci->irq_lock);
+ if (!aaci->users++) {
+ ret = request_irq(aaci->dev->irq[0], aaci_irq,
+ IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci);
+ if (ret != 0)
+ aaci->users--;
+ }
+ mutex_unlock(&aaci->irq_lock);
- out:
return ret;
}
@@ -418,7 +464,11 @@ static int aaci_pcm_close(struct snd_pcm_substream *substream)
WARN_ON(aacirun->cr & CR_EN);
aacirun->substream = NULL;
- free_irq(aaci->dev->irq[0], aaci);
+
+ mutex_lock(&aaci->irq_lock);
+ if (!--aaci->users)
+ free_irq(aaci->dev->irq[0], aaci);
+ mutex_unlock(&aaci->irq_lock);
return 0;
}
@@ -444,12 +494,21 @@ static int aaci_pcm_hw_free(struct snd_pcm_substream *substream)
return 0;
}
+/* Channel to slot mask */
+static const u32 channels_to_slotmask[] = {
+ [2] = CR_SL3 | CR_SL4,
+ [4] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8,
+ [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9,
+};
+
static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
- struct aaci_runtime *aacirun,
struct snd_pcm_hw_params *params)
{
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+ unsigned int channels = params_channels(params);
+ unsigned int rate = params_rate(params);
+ int dbl = rate > 48000;
int err;
- struct aaci *aaci = substream->private_data;
aaci_pcm_hw_free(substream);
if (aacirun->pcm_open) {
@@ -457,22 +516,28 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
aacirun->pcm_open = 0;
}
+ /* channels is already limited to 2, 4, or 6 by aaci_rule_channels */
+ if (dbl && channels != 2)
+ return -EINVAL;
+
err = snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(params));
if (err >= 0) {
- unsigned int rate = params_rate(params);
- int dbl = rate > 48000;
+ struct aaci *aaci = substream->private_data;
- err = snd_ac97_pcm_open(aacirun->pcm, rate,
- params_channels(params),
+ err = snd_ac97_pcm_open(aacirun->pcm, rate, channels,
aacirun->pcm->r[dbl].slots);
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;
+ aacirun->cr |= channels_to_slotmask[channels + dbl * 2];
+
+ /*
+ * fifo_bytes is the number of bytes we transfer to/from
+ * the FIFO, including padding. So that's x4. As we're
+ * in compact mode, the FIFO is half the size.
+ */
+ aacirun->fifo_bytes = aaci->fifo_depth * 4 / 2;
}
return err;
@@ -483,11 +548,11 @@ 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->period = snd_pcm_lib_period_bytes(substream);
aacirun->start = runtime->dma_area;
aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream);
aacirun->ptr = aacirun->start;
- aacirun->period =
- aacirun->bytes = frames_to_bytes(runtime, runtime->period_size);
+ aacirun->bytes = aacirun->period;
return 0;
}
@@ -505,89 +570,6 @@ static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream)
/*
* Playback specific ALSA stuff
*/
-static const u32 channels_to_txmask[] = {
- [2] = CR_SL3 | CR_SL4,
- [4] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8,
- [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9,
-};
-
-/*
- * We can support two and four channel audio. Unfortunately
- * six channel audio requires a non-standard channel ordering:
- * 2 -> FL(3), FR(4)
- * 4 -> FL(3), FR(4), SL(7), SR(8)
- * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required)
- * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual)
- * This requires an ALSA configuration file to correct.
- */
-static unsigned int channel_list[] = { 2, 4, 6 };
-
-static int
-aaci_rule_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule)
-{
- struct aaci *aaci = rule->private;
- unsigned int chan_mask = 1 << 0, slots;
-
- /*
- * pcms[0] is the our 5.1 PCM instance.
- */
- slots = aaci->ac97_bus->pcms[0].r[0].slots;
- if (slots & (1 << AC97_SLOT_PCM_SLEFT)) {
- chan_mask |= 1 << 1;
- if (slots & (1 << AC97_SLOT_LFE))
- chan_mask |= 1 << 2;
- }
-
- return snd_interval_list(hw_param_interval(p, rule->var),
- ARRAY_SIZE(channel_list), channel_list,
- chan_mask);
-}
-
-static int aaci_pcm_open(struct snd_pcm_substream *substream)
-{
- struct aaci *aaci = substream->private_data;
- int ret;
-
- /*
- * Add rule describing channel dependency.
- */
- ret = snd_pcm_hw_rule_add(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- aaci_rule_channels, aaci,
- SNDRV_PCM_HW_PARAM_CHANNELS, -1);
- if (ret)
- return ret;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- ret = __aaci_pcm_open(aaci, substream, &aaci->playback);
- } else {
- ret = __aaci_pcm_open(aaci, substream, &aaci->capture);
- }
- return ret;
-}
-
-static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct aaci_runtime *aacirun = substream->runtime->private_data;
- unsigned int channels = params_channels(params);
- int ret;
-
- WARN_ON(channels >= ARRAY_SIZE(channels_to_txmask) ||
- !channels_to_txmask[channels]);
-
- ret = aaci_pcm_hw_params(substream, aacirun, params);
-
- /*
- * Enable FIFO, compact mode, 16 bits per sample.
- * FIXME: double rate slots?
- */
- if (ret >= 0)
- aacirun->cr |= channels_to_txmask[channels];
-
- return ret;
-}
-
static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun)
{
u32 ie;
@@ -657,27 +639,13 @@ static struct snd_pcm_ops aaci_playback_ops = {
.open = aaci_pcm_open,
.close = aaci_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
- .hw_params = aaci_pcm_playback_hw_params,
+ .hw_params = aaci_pcm_hw_params,
.hw_free = aaci_pcm_hw_free,
.prepare = aaci_pcm_prepare,
.trigger = aaci_pcm_playback_trigger,
.pointer = aaci_pcm_pointer,
};
-static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct aaci_runtime *aacirun = substream->runtime->private_data;
- int ret;
-
- ret = aaci_pcm_hw_params(substream, aacirun, params);
- if (ret >= 0)
- /* Line in record: slot 3 and 4 */
- aacirun->cr |= CR_SL3 | CR_SL4;
-
- return ret;
-}
-
static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun)
{
u32 ie;
@@ -774,7 +742,7 @@ static struct snd_pcm_ops aaci_capture_ops = {
.open = aaci_pcm_open,
.close = aaci_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
- .hw_params = aaci_pcm_capture_hw_params,
+ .hw_params = aaci_pcm_hw_params,
.hw_free = aaci_pcm_hw_free,
.prepare = aaci_pcm_capture_prepare,
.trigger = aaci_pcm_capture_trigger,
@@ -941,12 +909,13 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver));
strlcpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname));
snprintf(card->longname, sizeof(card->longname),
- "%s at 0x%016llx, irq %d",
- card->shortname, (unsigned long long)dev->res.start,
- dev->irq[0]);
+ "%s PL%03x rev%u at 0x%08llx, irq %d",
+ card->shortname, amba_part(dev), amba_rev(dev),
+ (unsigned long long)dev->res.start, dev->irq[0]);
aaci = card->private_data;
mutex_init(&aaci->ac97_sem);
+ mutex_init(&aaci->irq_lock);
aaci->card = card;
aaci->dev = dev;
@@ -984,6 +953,10 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
struct aaci_runtime *aacirun = &aaci->playback;
int i;
+ /*
+ * Enable the channel, but don't assign it to any slots, so
+ * it won't empty onto the AC'97 link.
+ */
writel(CR_FEN | CR_SZ16 | CR_EN, aacirun->base + AACI_TXCR);
for (i = 0; !(readl(aacirun->base + AACI_SR) & SR_TXFF) && i < 4096; i++)
@@ -1002,7 +975,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
writel(aaci->maincr, aaci->base + AACI_MAINCR);
/*
- * If we hit 4096, we failed. Go back to the specified
+ * If we hit 4096 entries, we failed. Go back to the specified
* fifo depth.
*/
if (i == 4096)
@@ -1011,7 +984,8 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
return i;
}
-static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
+static int __devinit aaci_probe(struct amba_device *dev,
+ const struct amba_id *id)
{
struct aaci *aaci;
int ret, i;
@@ -1067,11 +1041,12 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
/*
* Size the FIFOs (must be multiple of 16).
+ * This is the number of entries in the FIFO.
*/
- aaci->fifosize = aaci_size_fifo(aaci);
- if (aaci->fifosize & 15) {
- printk(KERN_WARNING "AACI: fifosize = %d not supported\n",
- aaci->fifosize);
+ aaci->fifo_depth = aaci_size_fifo(aaci);
+ if (aaci->fifo_depth & 15) {
+ printk(KERN_WARNING "AACI: FIFO depth %d not supported\n",
+ aaci->fifo_depth);
ret = -ENODEV;
goto out;
}
@@ -1084,8 +1059,8 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
ret = snd_card_register(aaci->card);
if (ret == 0) {
- dev_info(&dev->dev, "%s, fifo %d\n", aaci->card->longname,
- aaci->fifosize);
+ dev_info(&dev->dev, "%s\n", aaci->card->longname);
+ dev_info(&dev->dev, "FIFO %u entries\n", aaci->fifo_depth);
amba_set_drvdata(dev, aaci->card);
return ret;
}
diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h
index 6a4a2eebdda1..5791bd5bd2ab 100644
--- a/sound/arm/aaci.h
+++ b/sound/arm/aaci.h
@@ -210,6 +210,8 @@ struct aaci_runtime {
u32 cr;
struct snd_pcm_substream *substream;
+ unsigned int period; /* byte size of a "period" */
+
/*
* PIO support
*/
@@ -217,15 +219,16 @@ struct aaci_runtime {
void *end;
void *ptr;
int bytes;
- unsigned int period;
- unsigned int fifosz;
+ unsigned int fifo_bytes;
};
struct aaci {
struct amba_device *dev;
struct snd_card *card;
void __iomem *base;
- unsigned int fifosize;
+ unsigned int fifo_depth;
+ unsigned int users;
+ struct mutex irq_lock;
/* AC'97 */
struct mutex ac97_sem;
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 4902ae568730..53b53e97c896 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
fail_input:
input_free_device(jack->input_dev);
+ kfree(jack->id);
kfree(jack);
return err;
}
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f356e0f..16c0bdfbb164 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
{
stream_t *dma = &vortex->dma_adb[adbdma];
- int temp;
+ int temp, page, delta;
temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
- temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
- return temp;
+ page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+ if (dma->nr_periods >= 4)
+ delta = (page - dma->period_real) & 3;
+ else {
+ delta = (page - dma->period_real);
+ if (delta < 0)
+ delta += dma->nr_periods;
+ }
+ return (dma->period_virt + delta) * dma->period_bytes
+ + (temp & (dma->period_bytes - 1));
}
static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0baffcdee8f9..fcedad9a5fef 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index a07b031090d8..067982f4f182 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1039,9 +1039,11 @@ static struct hda_verb cs_errata_init_verbs[] = {
{0x11, AC_VERB_SET_PROC_COEF, 0x0008},
{0x11, AC_VERB_SET_PROC_STATE, 0x00},
+#if 0 /* Don't to set to D3 as we are in power-up sequence */
{0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */
{0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */
/*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */
+#endif
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fbe97d32140d..4d5004e693f0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3114,6 +3114,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
@@ -3410,7 +3412,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
}
}
spec->multiout.dac_nids = spec->private_dac_nids;
- spec->multiout.max_channels = nums * 2;
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
if (cfg->hp_outs > 0)
spec->auto_mute = 1;
@@ -3729,9 +3731,9 @@ static int cx_auto_init(struct hda_codec *codec)
return 0;
}
-static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
+static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
const char *dir, int cidx,
- hda_nid_t nid, int hda_dir)
+ hda_nid_t nid, int hda_dir, int amp_idx)
{
static char name[32];
static struct snd_kcontrol_new knew[] = {
@@ -3743,7 +3745,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
for (i = 0; i < 2; i++) {
struct snd_kcontrol *kctl;
- knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir);
+ knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+ hda_dir);
knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
knew[i].index = cidx;
snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]);
@@ -3759,6 +3762,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
return 0;
}
+#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \
+ cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+
#define cx_auto_add_pb_volume(codec, nid, str, idx) \
cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
@@ -3808,29 +3814,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
struct conexant_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
static const char *prev_label;
- int i, err, cidx;
+ int i, err, cidx, conn_len;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
+ int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
+ int adc_nid = spec->adc_nids[0];
+
+ conn_len = snd_hda_get_connections(codec, adc_nid, conn,
+ HDA_MAX_CONNECTIONS);
+ if (conn_len < 0)
+ return conn_len;
+
+ multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
+ if (!multi_adc_volume) {
+ err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
+ HDA_INPUT);
+ if (err < 0)
+ return err;
+ }
- err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0],
- HDA_INPUT);
- if (err < 0)
- return err;
prev_label = NULL;
cidx = 0;
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
const char *label;
- if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+ int j;
+ int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
+ if (!pin_amp && !multi_adc_volume)
continue;
+
label = hda_get_autocfg_input_label(codec, cfg, i);
if (label == prev_label)
cidx++;
else
cidx = 0;
prev_label = label;
- err = cx_auto_add_volume(codec, label, " Capture", cidx,
- nid, HDA_INPUT);
- if (err < 0)
- return err;
+
+ if (pin_amp) {
+ err = cx_auto_add_volume(codec, label, " Boost", cidx,
+ nid, HDA_INPUT);
+ if (err < 0)
+ return err;
+ }
+
+ if (!multi_adc_volume)
+ continue;
+ for (j = 0; j < conn_len; j++) {
+ if (conn[j] == nid) {
+ err = cx_auto_add_volume_idx(codec, label,
+ " Capture", cidx, adc_nid, HDA_INPUT, j);
+ if (err < 0)
+ return err;
+ break;
+ }
+ }
}
return 0;
}
@@ -3902,6 +3939,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_cxt5066 },
{ .id = 0x14f15069, .name = "CX20585",
.patch = patch_cxt5066 },
+ { .id = 0x14f1506e, .name = "CX20590",
+ .patch = patch_cxt5066 },
{ .id = 0x14f15097, .name = "CX20631",
.patch = patch_conexant_auto },
{ .id = 0x14f15098, .name = "CX20632",
@@ -3928,6 +3967,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066");
MODULE_ALIAS("snd-hda-codec-id:14f15067");
MODULE_ALIAS("snd-hda-codec-id:14f15068");
MODULE_ALIAS("snd-hda-codec-id:14f15069");
+MODULE_ALIAS("snd-hda-codec-id:14f1506e");
MODULE_ALIAS("snd-hda-codec-id:14f15097");
MODULE_ALIAS("snd-hda-codec-id:14f15098");
MODULE_ALIAS("snd-hda-codec-id:14f150a1");
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index a58767736727..ec0fa2dd0a27 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1634,6 +1634,9 @@ static struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
+{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
+{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
+/* 17 is known to be absent */
{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 },
@@ -1676,6 +1679,8 @@ MODULE_ALIAS("snd-hda-codec-id:10de0011");
MODULE_ALIAS("snd-hda-codec-id:10de0012");
MODULE_ALIAS("snd-hda-codec-id:10de0013");
MODULE_ALIAS("snd-hda-codec-id:10de0014");
+MODULE_ALIAS("snd-hda-codec-id:10de0015");
+MODULE_ALIAS("snd-hda-codec-id:10de0016");
MODULE_ALIAS("snd-hda-codec-id:10de0018");
MODULE_ALIAS("snd-hda-codec-id:10de0019");
MODULE_ALIAS("snd-hda-codec-id:10de001a");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3328a259a242..4261bb8eec1d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1133,11 +1133,8 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl)
nid = spec->autocfg.hp_pins[i];
if (!nid)
break;
- if (snd_hda_jack_detect(codec, nid)) {
- spec->jack_present = 1;
- break;
- }
- alc_report_jack(codec, spec->autocfg.hp_pins[i]);
+ alc_report_jack(codec, nid);
+ spec->jack_present |= snd_hda_jack_detect(codec, nid);
}
mute = spec->jack_present ? HDA_AMP_MUTE : 0;
@@ -15015,7 +15012,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
- SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82Jv", ALC269_AMIC),
+ SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9ea48b425d0b..bd7b123f6440 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -586,7 +586,12 @@ static hda_nid_t stac92hd83xxx_pin_nids[10] = {
0x0f, 0x10, 0x11, 0x1f, 0x20,
};
-static hda_nid_t stac92hd88xxx_pin_nids[10] = {
+static hda_nid_t stac92hd87xxx_pin_nids[6] = {
+ 0x0a, 0x0b, 0x0c, 0x0d,
+ 0x0f, 0x11,
+};
+
+static hda_nid_t stac92hd88xxx_pin_nids[8] = {
0x0a, 0x0b, 0x0c, 0x0d,
0x0f, 0x11, 0x1f, 0x20,
};
@@ -5430,12 +5435,13 @@ again:
switch (codec->vendor_id) {
case 0x111d76d1:
case 0x111d76d9:
+ case 0x111d76e5:
spec->dmic_nids = stac92hd87b_dmic_nids;
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd87b_dmic_nids,
STAC92HD87B_NUM_DMICS);
- spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids);
- spec->pin_nids = stac92hd88xxx_pin_nids;
+ spec->num_pins = ARRAY_SIZE(stac92hd87xxx_pin_nids);
+ spec->pin_nids = stac92hd87xxx_pin_nids;
spec->mono_nid = 0;
spec->num_pwrs = 0;
break;
@@ -5443,6 +5449,7 @@ again:
case 0x111d7667:
case 0x111d7668:
case 0x111d7669:
+ case 0x111d76e3:
spec->num_dmics = stac92xx_connected_ports(codec,
stac92hd88xxx_dmic_nids,
STAC92HD88XXX_NUM_DMICS);
@@ -6387,6 +6394,8 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index a76c3260d941..63b0054200a8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
hda_nid_t nid = cfg->inputs[i].pin;
if (spec->smart51_enabled && is_smart51_pins(spec, nid))
ctl = PIN_OUT;
- else if (i == AUTO_PIN_MIC)
+ else if (cfg->inputs[i].type == AUTO_PIN_MIC)
ctl = PIN_VREF50;
else
ctl = PIN_IN;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index bb4bf65b9e7e..0bb424af956f 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,7 +367,7 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
return 0;
}
-static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+static const u8 cx20442_reg;
static struct snd_soc_codec_driver cx20442_codec_dev = {
.probe = cx20442_codec_probe,
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 987476a5895f..017d99ceb42e 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1482,7 +1482,7 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
WM8903_MICDET_EINT | WM8903_MICSHRT_EINT,
irq_mask);
- if (det && shrt) {
+ if (det || shrt) {
/* Enable mic detection, this may not have been set through
* platform data (eg, if the defaults are OK). */
snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index e8490f3edd03..e3ec2433b215 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -165,7 +165,7 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec,
#define WM8903_VMID_RES_50K 2
#define WM8903_VMID_RES_250K 3
-#define WM8903_VMID_RES_5K 4
+#define WM8903_VMID_RES_5K 6
/*
* R8 (0x08) - Analogue DAC 0
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 4bbc3442703f..8dfb0a0da673 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -145,18 +145,18 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = {
SOC_SINGLE("DAC Playback Limiter Threshold",
WM8978_DAC_LIMITER_2, 4, 7, 0),
SOC_SINGLE("DAC Playback Limiter Boost",
- WM8978_DAC_LIMITER_2, 0, 15, 0),
+ WM8978_DAC_LIMITER_2, 0, 12, 0),
SOC_ENUM("ALC Enable Switch", alc1),
SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0),
SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0),
- SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0),
+ SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0),
SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0),
SOC_ENUM("ALC Capture Mode", alc3),
- SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0),
- SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0),
+ SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0),
SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0),
SOC_SINGLE("ALC Capture Noise Gate Threshold",
@@ -211,8 +211,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = {
WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1),
/* DAC / ADC oversampling */
- SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0),
- SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0),
+ SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL,
+ 5, 1, 0),
+ SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL,
+ 5, 1, 0),
};
/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 37b8aa8a680f..c6c958ee5d59 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -107,6 +107,12 @@ struct wm8994_priv {
int revision;
struct wm8994_pdata *pdata;
+
+ unsigned int aif1clk_enable:1;
+ unsigned int aif2clk_enable:1;
+
+ unsigned int aif1clk_disable:1;
+ unsigned int aif2clk_disable:1;
};
static int wm8994_readable(unsigned int reg)
@@ -1004,6 +1010,110 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
}
}
+static int late_enable_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (wm8994->aif1clk_enable) {
+ snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+ WM8994_AIF1CLK_ENA_MASK,
+ WM8994_AIF1CLK_ENA);
+ wm8994->aif1clk_enable = 0;
+ }
+ if (wm8994->aif2clk_enable) {
+ snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+ WM8994_AIF2CLK_ENA_MASK,
+ WM8994_AIF2CLK_ENA);
+ wm8994->aif2clk_enable = 0;
+ }
+ break;
+ }
+
+ return 0;
+}
+
+static int late_disable_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ if (wm8994->aif1clk_disable) {
+ snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+ WM8994_AIF1CLK_ENA_MASK, 0);
+ wm8994->aif1clk_disable = 0;
+ }
+ if (wm8994->aif2clk_disable) {
+ snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+ WM8994_AIF2CLK_ENA_MASK, 0);
+ wm8994->aif2clk_disable = 0;
+ }
+ break;
+ }
+
+ return 0;
+}
+
+static int aif1clk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ wm8994->aif1clk_enable = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ wm8994->aif1clk_disable = 1;
+ break;
+ }
+
+ return 0;
+}
+
+static int aif2clk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ wm8994->aif2clk_enable = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ wm8994->aif2clk_disable = 1;
+ break;
+ }
+
+ return 0;
+}
+
+static int adc_mux_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ late_enable_ev(w, kcontrol, event);
+ return 0;
+}
+
+static int dac_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int mask = 1 << w->shift;
+
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+ mask, mask);
+ return 0;
+}
+
static const char *hp_mux_text[] = {
"Mixer",
"DAC",
@@ -1272,6 +1382,59 @@ static const struct soc_enum aif2dacr_src_enum =
static const struct snd_kcontrol_new aif2dacr_src_mux =
SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
+static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+
+SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
+};
+
+static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0)
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
+SND_SOC_DAPM_DAC_E("DAC2L", NULL, SND_SOC_NOPM, 3, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC2R", NULL, SND_SOC_NOPM, 2, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1L", NULL, SND_SOC_NOPM, 1, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = {
+SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
+SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
+SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
+SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
+SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
+SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+};
+
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("DMIC1DAT"),
SND_SOC_DAPM_INPUT("DMIC2DAT"),
@@ -1284,9 +1447,6 @@ SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
-
SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
0, WM8994_POWER_MANAGEMENT_4, 9, 0),
SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
@@ -1369,14 +1529,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0),
SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0),
SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0),
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
-
-SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
-SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
-SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
-SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
-
SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
@@ -1516,14 +1668,12 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" },
/* DAC1 inputs */
- { "DAC1L", NULL, "DAC1L Mixer" },
{ "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
{ "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
{ "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" },
- { "DAC1R", NULL, "DAC1R Mixer" },
{ "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
{ "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
@@ -1532,7 +1682,6 @@ static const struct snd_soc_dapm_route intercon[] = {
/* DAC2/AIF2 outputs */
{ "AIF2ADCL", NULL, "AIF2DAC2L Mixer" },
- { "DAC2L", NULL, "AIF2DAC2L Mixer" },
{ "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
{ "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
@@ -1540,7 +1689,6 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" },
{ "AIF2ADCR", NULL, "AIF2DAC2R Mixer" },
- { "DAC2R", NULL, "AIF2DAC2R Mixer" },
{ "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
{ "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
@@ -1584,6 +1732,24 @@ static const struct snd_soc_dapm_route intercon[] = {
{ "Right Headphone Mux", "DAC", "DAC1R" },
};
+static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = {
+ { "DAC1L", NULL, "Late DAC1L Enable PGA" },
+ { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" },
+ { "DAC1R", NULL, "Late DAC1R Enable PGA" },
+ { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" },
+ { "DAC2L", NULL, "Late DAC2L Enable PGA" },
+ { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" },
+ { "DAC2R", NULL, "Late DAC2R Enable PGA" },
+ { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" }
+};
+
+static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = {
+ { "DAC1L", NULL, "DAC1L Mixer" },
+ { "DAC1R", NULL, "DAC1R Mixer" },
+ { "DAC2L", NULL, "AIF2DAC2L Mixer" },
+ { "DAC2R", NULL, "AIF2DAC2R Mixer" },
+};
+
static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
{ "AIF1DACDAT", NULL, "AIF2DACDAT" },
{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
@@ -2514,6 +2680,22 @@ static int wm8994_resume(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int i, ret;
+ unsigned int val, mask;
+
+ if (wm8994->revision < 4) {
+ /* force a HW read */
+ val = wm8994_reg_read(codec->control_data,
+ WM8994_POWER_MANAGEMENT_5);
+
+ /* modify the cache only */
+ codec->cache_only = 1;
+ mask = WM8994_DAC1R_ENA | WM8994_DAC1L_ENA |
+ WM8994_DAC2R_ENA | WM8994_DAC2L_ENA;
+ val &= mask;
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+ mask, val);
+ codec->cache_only = 0;
+ }
/* Restore the registers */
ret = snd_soc_cache_sync(codec);
@@ -2847,11 +3029,10 @@ static void wm8958_default_micdet(u16 status, void *data)
report |= SND_JACK_BTN_5;
done:
- snd_soc_jack_report(wm8994->micdet[0].jack,
+ snd_soc_jack_report(wm8994->micdet[0].jack, report,
SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 |
SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 |
- SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT,
- report);
+ SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT);
}
/**
@@ -3125,10 +3306,31 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
case WM8994:
snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
ARRAY_SIZE(wm8994_specific_dapm_widgets));
+ if (wm8994->revision < 4) {
+ snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
+ ARRAY_SIZE(wm8994_lateclk_revd_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
+ ARRAY_SIZE(wm8994_adc_revd_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets,
+ ARRAY_SIZE(wm8994_dac_revd_widgets));
+ } else {
+ snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+ ARRAY_SIZE(wm8994_lateclk_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+ ARRAY_SIZE(wm8994_adc_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+ ARRAY_SIZE(wm8994_dac_widgets));
+ }
break;
case WM8958:
snd_soc_add_controls(codec, wm8958_snd_controls,
ARRAY_SIZE(wm8958_snd_controls));
+ snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+ ARRAY_SIZE(wm8994_lateclk_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+ ARRAY_SIZE(wm8994_adc_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+ ARRAY_SIZE(wm8994_dac_widgets));
snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
ARRAY_SIZE(wm8958_dapm_widgets));
break;
@@ -3143,12 +3345,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
- if (wm8994->revision < 4)
+ if (wm8994->revision < 4) {
snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
ARRAY_SIZE(wm8994_revd_intercon));
-
+ snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
+ ARRAY_SIZE(wm8994_lateclk_revd_intercon));
+ } else {
+ snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+ ARRAY_SIZE(wm8994_lateclk_intercon));
+ }
break;
case WM8958:
+ snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+ ARRAY_SIZE(wm8994_lateclk_intercon));
snd_soc_dapm_add_routes(dapm, wm8958_intercon,
ARRAY_SIZE(wm8958_intercon));
break;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 43825b2102a5..cce704c275c6 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -15,6 +15,7 @@
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
+#include <linux/device.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
@@ -1341,6 +1342,10 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c,
wm9081->control_type = SND_SOC_I2C;
wm9081->control_data = i2c;
+ if (dev_get_platdata(&i2c->dev))
+ memcpy(&wm9081->retune, dev_get_platdata(&i2c->dev),
+ sizeof(wm9081->retune));
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm9081, &wm9081_dai, 1);
if (ret < 0)
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 613df5db0b32..516892706063 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -674,6 +674,9 @@ SND_SOC_DAPM_OUTPUT("LINEOUT2N"),
};
static const struct snd_soc_dapm_route analogue_routes[] = {
+ { "MICBIAS1", NULL, "CLK_SYS" },
+ { "MICBIAS2", NULL, "CLK_SYS" },
+
{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 4cf98c03af22..15dac0f20cd8 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -896,8 +896,7 @@ static struct snd_pcm_ops fsl_dma_ops = {
.pointer = fsl_dma_pointer,
};
-static int __devinit fsl_soc_dma_probe(struct platform_device *pdev,
- const struct of_device_id *match)
+static int __devinit fsl_soc_dma_probe(struct platform_device *pdev)
{
struct dma_object *dma;
struct device_node *np = pdev->dev.of_node;
@@ -979,7 +978,7 @@ static const struct of_device_id fsl_soc_dma_ids[] = {
};
MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids);
-static struct of_platform_driver fsl_soc_dma_driver = {
+static struct platform_driver fsl_soc_dma_driver = {
.driver = {
.name = "fsl-pcm-audio",
.owner = THIS_MODULE,
@@ -993,12 +992,12 @@ static int __init fsl_soc_dma_init(void)
{
pr_info("Freescale Elo DMA ASoC PCM Driver\n");
- return of_register_platform_driver(&fsl_soc_dma_driver);
+ return platform_driver_register(&fsl_soc_dma_driver);
}
static void __exit fsl_soc_dma_exit(void)
{
- of_unregister_platform_driver(&fsl_soc_dma_driver);
+ platform_driver_unregister(&fsl_soc_dma_driver);
}
module_init(fsl_soc_dma_init);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 4cc167a7aeb8..313e0ccedd5b 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -624,8 +624,7 @@ static void make_lowercase(char *s)
}
}
-static int __devinit fsl_ssi_probe(struct platform_device *pdev,
- const struct of_device_id *match)
+static int __devinit fsl_ssi_probe(struct platform_device *pdev)
{
struct fsl_ssi_private *ssi_private;
int ret = 0;
@@ -774,7 +773,7 @@ static const struct of_device_id fsl_ssi_ids[] = {
};
MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
-static struct of_platform_driver fsl_ssi_driver = {
+static struct platform_driver fsl_ssi_driver = {
.driver = {
.name = "fsl-ssi-dai",
.owner = THIS_MODULE,
@@ -788,12 +787,12 @@ static int __init fsl_ssi_init(void)
{
printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n");
- return of_register_platform_driver(&fsl_ssi_driver);
+ return platform_driver_register(&fsl_ssi_driver);
}
static void __exit fsl_ssi_exit(void)
{
- of_unregister_platform_driver(&fsl_ssi_driver);
+ platform_driver_unregister(&fsl_ssi_driver);
}
module_init(fsl_ssi_init);
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index f92dca07cd35..fff695ccdd3e 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -368,8 +368,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
.pcm_free = &psc_dma_free,
};
-static int mpc5200_hpcd_probe(struct of_device *op,
- const struct of_device_id *match)
+static int mpc5200_hpcd_probe(struct of_device *op)
{
phys_addr_t fifo;
struct psc_dma *psc_dma;
@@ -511,32 +510,31 @@ static int mpc5200_hpcd_remove(struct of_device *op)
}
static struct of_device_id mpc5200_hpcd_match[] = {
- {
- .compatible = "fsl,mpc5200-pcm",
- },
+ { .compatible = "fsl,mpc5200-pcm", },
{}
};
MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match);
-static struct of_platform_driver mpc5200_hpcd_of_driver = {
- .owner = THIS_MODULE,
- .name = "mpc5200-pcm-audio",
- .match_table = mpc5200_hpcd_match,
+static struct platform_driver mpc5200_hpcd_of_driver = {
.probe = mpc5200_hpcd_probe,
.remove = mpc5200_hpcd_remove,
+ .dev = {
+ .owner = THIS_MODULE,
+ .name = "mpc5200-pcm-audio",
+ .of_match_table = mpc5200_hpcd_match,
+ }
};
static int __init mpc5200_hpcd_init(void)
{
- return of_register_platform_driver(&mpc5200_hpcd_of_driver);
+ return platform_driver_register(&mpc5200_hpcd_of_driver);
}
+module_init(mpc5200_hpcd_init);
static void __exit mpc5200_hpcd_exit(void)
{
- of_unregister_platform_driver(&mpc5200_hpcd_of_driver);
+ platform_driver_unregister(&mpc5200_hpcd_of_driver);
}
-
-module_init(mpc5200_hpcd_init);
module_exit(mpc5200_hpcd_exit);
MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 40acc8e2b1ca..ad36b095bb79 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -272,8 +272,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = {
* - Probe/remove operations
* - OF device match table
*/
-static int __devinit psc_ac97_of_probe(struct platform_device *op,
- const struct of_device_id *match)
+static int __devinit psc_ac97_of_probe(struct platform_device *op)
{
int rc;
struct snd_ac97 ac97;
@@ -316,7 +315,7 @@ static struct of_device_id psc_ac97_match[] __devinitdata = {
};
MODULE_DEVICE_TABLE(of, psc_ac97_match);
-static struct of_platform_driver psc_ac97_driver = {
+static struct platform_driver psc_ac97_driver = {
.probe = psc_ac97_of_probe,
.remove = __devexit_p(psc_ac97_of_remove),
.driver = {
@@ -332,13 +331,13 @@ static struct of_platform_driver psc_ac97_driver = {
*/
static int __init psc_ac97_init(void)
{
- return of_register_platform_driver(&psc_ac97_driver);
+ return platform_driver_register(&psc_ac97_driver);
}
module_init(psc_ac97_init);
static void __exit psc_ac97_exit(void)
{
- of_unregister_platform_driver(&psc_ac97_driver);
+ platform_driver_unregister(&psc_ac97_driver);
}
module_exit(psc_ac97_exit);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9018fa5bf0db..87cf2a5c2b2c 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -150,8 +150,7 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{
* - Probe/remove operations
* - OF device match table
*/
-static int __devinit psc_i2s_of_probe(struct platform_device *op,
- const struct of_device_id *match)
+static int __devinit psc_i2s_of_probe(struct platform_device *op)
{
int rc;
struct psc_dma *psc_dma;
@@ -213,7 +212,7 @@ static struct of_device_id psc_i2s_match[] __devinitdata = {
};
MODULE_DEVICE_TABLE(of, psc_i2s_match);
-static struct of_platform_driver psc_i2s_driver = {
+static struct platform_driver psc_i2s_driver = {
.probe = psc_i2s_of_probe,
.remove = __devexit_p(psc_i2s_of_remove),
.driver = {
@@ -229,13 +228,13 @@ static struct of_platform_driver psc_i2s_driver = {
*/
static int __init psc_i2s_init(void)
{
- return of_register_platform_driver(&psc_i2s_driver);
+ return platform_driver_register(&psc_i2s_driver);
}
module_init(psc_i2s_init);
static void __exit psc_i2s_exit(void)
{
- of_unregister_platform_driver(&psc_i2s_driver);
+ platform_driver_unregister(&psc_i2s_driver);
}
module_exit(psc_i2s_exit);
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index e20c9e1457c0..1e9bccae4e80 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -79,7 +79,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "imx-pcm-audio.0",
+ .platform_name = "imx-fiq-pcm-audio.0",
.codec_name = "tlv320aic23-codec.0-001a",
.cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 161750443ebc..73dde4a1adc3 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = {
.cpu_dai_name ="omap-mcbsp-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "omap-pcm-audio",
- .codec_name = "tlv320aic23-codec",
+ .codec_name = "tlv320aic23-codec.2-001a",
.init = am3517evm_aic23_init,
.ops = &am3517evm_ops,
};
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 28333e7d9c50..dc65650a6fa1 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -117,7 +117,7 @@ static struct snd_soc_dai_link e740_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
@@ -126,7 +126,7 @@ static struct snd_soc_dai_link e740_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9705-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 01bf31675c55..51897fcd911b 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -99,7 +99,7 @@ static struct snd_soc_dai_link e750_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
@@ -109,7 +109,7 @@ static struct snd_soc_dai_link e750_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9705-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index c6a37c6ef23b..053ed208e59f 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -89,7 +89,7 @@ static struct snd_soc_dai_link e800_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -98,7 +98,7 @@ static struct snd_soc_dai_link e800_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index fc22e6eefc98..b13a4252812d 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 0d70fc8c12bd..38ca6759907e 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -162,7 +162,7 @@ static struct snd_soc_dai_link mioa701_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9713-hifi",
.codec_name = "wm9713-codec",
.init = mioa701_wm9713_init,
@@ -172,7 +172,7 @@ static struct snd_soc_dai_link mioa701_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9713-aux",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 857db96d4a4f..504e4004f004 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -132,7 +132,7 @@ static struct snd_soc_dai_link palm27x_dai[] = {
{
.name = "AC97 HiFi",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.codec_name = "wm9712-codec",
.platform_name = "pxa-pcm-audio",
@@ -141,7 +141,7 @@ static struct snd_soc_dai_link palm27x_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9712-aux",
.codec_name = "wm9712-codec",
.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f75804ef0897..4b6e5d608b42 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -219,7 +219,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -229,7 +229,7 @@ static struct snd_soc_dai_link tosa_dai[] = {
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index b222a7d72027..25bba108fea3 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -166,7 +166,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.stream_name = "AC97 HiFi",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_name = "wm9713-hifi",
.init = zylonite_wm9713_init,
},
@@ -175,7 +175,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
.stream_name = "AC97 Aux",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_name = "wm9713-aux",
},
{
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8194f150bab7..1790f83ee665 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -712,7 +712,15 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
!path->connected(path->source, path->sink))
continue;
- if (path->sink && path->sink->power_check &&
+ if (!path->sink)
+ continue;
+
+ if (path->sink->force) {
+ power = 1;
+ break;
+ }
+
+ if (path->sink->power_check &&
path->sink->power_check(path->sink)) {
power = 1;
break;
@@ -933,7 +941,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
}
if (!list_empty(&pending))
- dapm_seq_run_coalesced(dapm, &pending);
+ dapm_seq_run_coalesced(cur_dapm, &pending);
}
static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
@@ -1627,6 +1635,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_widget *w;
+ unsigned int val;
list_for_each_entry(w, &dapm->card->widgets, list)
{
@@ -1675,6 +1684,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
case snd_soc_dapm_post:
break;
}
+
+ /* Read the initial power state from the device */
+ if (w->reg >= 0) {
+ val = snd_soc_read(w->codec, w->reg);
+ val &= 1 << w->shift;
+ if (w->invert)
+ val = !val;
+
+ if (val)
+ w->power = 1;
+ }
+
w->new = 1;
}
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index f8bcfc30f800..ad7d4d7d9237 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -1002,7 +1002,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
return 0;
}
-static int __devinit amd7930_sbus_probe(struct platform_device *op, const struct of_device_id *match)
+static int __devinit amd7930_sbus_probe(struct platform_device *op)
{
struct resource *rp = &op->resource[0];
static int dev_num;
@@ -1064,7 +1064,7 @@ static const struct of_device_id amd7930_match[] = {
{},
};
-static struct of_platform_driver amd7930_sbus_driver = {
+static struct platform_driver amd7930_sbus_driver = {
.driver = {
.name = "audio",
.owner = THIS_MODULE,
@@ -1075,7 +1075,7 @@ static struct of_platform_driver amd7930_sbus_driver = {
static int __init amd7930_init(void)
{
- return of_register_platform_driver(&amd7930_sbus_driver);
+ return platform_driver_register(&amd7930_sbus_driver);
}
static void __exit amd7930_exit(void)
@@ -1092,7 +1092,7 @@ static void __exit amd7930_exit(void)
amd7930_list = NULL;
- of_unregister_platform_driver(&amd7930_sbus_driver);
+ platform_driver_unregister(&amd7930_sbus_driver);
}
module_init(amd7930_init);
diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c
index c276086c3b57..0e618f82808c 100644
--- a/sound/sparc/cs4231.c
+++ b/sound/sparc/cs4231.c
@@ -1856,7 +1856,7 @@ static int __devinit snd_cs4231_sbus_create(struct snd_card *card,
return 0;
}
-static int __devinit cs4231_sbus_probe(struct platform_device *op, const struct of_device_id *match)
+static int __devinit cs4231_sbus_probe(struct platform_device *op)
{
struct resource *rp = &op->resource[0];
struct snd_card *card;
@@ -2048,7 +2048,7 @@ static int __devinit snd_cs4231_ebus_create(struct snd_card *card,
return 0;
}
-static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct of_device_id *match)
+static int __devinit cs4231_ebus_probe(struct platform_device *op)
{
struct snd_card *card;
int err;
@@ -2072,16 +2072,16 @@ static int __devinit cs4231_ebus_probe(struct platform_device *op, const struct
}
#endif
-static int __devinit cs4231_probe(struct platform_device *op, const struct of_device_id *match)
+static int __devinit cs4231_probe(struct platform_device *op)
{
#ifdef EBUS_SUPPORT
if (!strcmp(op->dev.of_node->parent->name, "ebus"))
- return cs4231_ebus_probe(op, match);
+ return cs4231_ebus_probe(op);
#endif
#ifdef SBUS_SUPPORT
if (!strcmp(op->dev.of_node->parent->name, "sbus") ||
!strcmp(op->dev.of_node->parent->name, "sbi"))
- return cs4231_sbus_probe(op, match);
+ return cs4231_sbus_probe(op);
#endif
return -ENODEV;
}
@@ -2108,7 +2108,7 @@ static const struct of_device_id cs4231_match[] = {
MODULE_DEVICE_TABLE(of, cs4231_match);
-static struct of_platform_driver cs4231_driver = {
+static struct platform_driver cs4231_driver = {
.driver = {
.name = "audio",
.owner = THIS_MODULE,
@@ -2120,12 +2120,12 @@ static struct of_platform_driver cs4231_driver = {
static int __init cs4231_init(void)
{
- return of_register_platform_driver(&cs4231_driver);
+ return platform_driver_register(&cs4231_driver);
}
static void __exit cs4231_exit(void)
{
- of_unregister_platform_driver(&cs4231_driver);
+ platform_driver_unregister(&cs4231_driver);
}
module_init(cs4231_init);
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index 39cd5d69d051..73f9cbacc077 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -2592,7 +2592,7 @@ static void snd_dbri_free(struct snd_dbri *dbri)
(void *)dbri->dma, dbri->dma_dvma);
}
-static int __devinit dbri_probe(struct platform_device *op, const struct of_device_id *match)
+static int __devinit dbri_probe(struct platform_device *op)
{
struct snd_dbri *dbri;
struct resource *rp;
@@ -2686,7 +2686,7 @@ static const struct of_device_id dbri_match[] = {
MODULE_DEVICE_TABLE(of, dbri_match);
-static struct of_platform_driver dbri_sbus_driver = {
+static struct platform_driver dbri_sbus_driver = {
.driver = {
.name = "dbri",
.owner = THIS_MODULE,
@@ -2699,12 +2699,12 @@ static struct of_platform_driver dbri_sbus_driver = {
/* Probe for the dbri chip and then attach the driver. */
static int __init dbri_init(void)
{
- return of_register_platform_driver(&dbri_sbus_driver);
+ return platform_driver_register(&dbri_sbus_driver);
}
static void __exit dbri_exit(void)
{
- of_unregister_platform_driver(&dbri_sbus_driver);
+ platform_driver_unregister(&dbri_sbus_driver);
}
module_init(dbri_init);
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 68b97477577b..66eabafb1c24 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
}
dev->pcm->private_data = dev;
- strcpy(dev->pcm->name, dev->product_name);
+ strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name));
memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 2f218c77fff2..a1a47088fd0c 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device)
if (ret < 0)
return ret;
- strcpy(rmidi->name, device->product_name);
+ strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name));
rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX;
rmidi->private_data = device;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 800f7cb4f251..c0f8270bc199 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -323,6 +323,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
return -ENOMEM;
}
+ mutex_init(&chip->shutdown_mutex);
chip->index = idx;
chip->dev = dev;
chip->card = card;
@@ -531,6 +532,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
chip = ptr;
card = chip->card;
mutex_lock(&register_mutex);
+ mutex_lock(&chip->shutdown_mutex);
chip->shutdown = 1;
chip->num_interfaces--;
if (chip->num_interfaces <= 0) {
@@ -548,9 +550,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
snd_usb_mixer_disconnect(p);
}
usb_chip[chip->index] = NULL;
+ mutex_unlock(&chip->shutdown_mutex);
mutex_unlock(&register_mutex);
snd_card_free_when_closed(card);
} else {
+ mutex_unlock(&chip->shutdown_mutex);
mutex_unlock(&register_mutex);
}
}
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 4132522ac90f..e3f680526cb5 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -361,6 +361,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
}
if (changed) {
+ mutex_lock(&subs->stream->chip->shutdown_mutex);
/* format changed */
snd_usb_release_substream_urbs(subs, 0);
/* influenced: period_bytes, channels, rate, format, */
@@ -368,6 +369,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
params_rate(hw_params),
snd_pcm_format_physical_width(params_format(hw_params)) *
params_channels(hw_params));
+ mutex_unlock(&subs->stream->chip->shutdown_mutex);
}
return ret;
@@ -385,8 +387,9 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream)
subs->cur_audiofmt = NULL;
subs->cur_rate = 0;
subs->period_bytes = 0;
- if (!subs->stream->chip->shutdown)
- snd_usb_release_substream_urbs(subs, 0);
+ mutex_lock(&subs->stream->chip->shutdown_mutex);
+ snd_usb_release_substream_urbs(subs, 0);
+ mutex_unlock(&subs->stream->chip->shutdown_mutex);
return snd_pcm_lib_free_vmalloc_buffer(substream);
}
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index db3eb21627ee..6e66fffe87f5 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -36,6 +36,7 @@ struct snd_usb_audio {
struct snd_card *card;
u32 usb_id;
int shutdown;
+ struct mutex shutdown_mutex;
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
int num_interfaces;
int num_suspended_intf;