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-rw-r--r--sound/arm/aaci.c293
-rw-r--r--sound/arm/aaci.h9
-rw-r--r--sound/atmel/ac97c.c5
-rw-r--r--sound/core/hrtimer.c7
-rw-r--r--sound/core/jack.c1
-rw-r--r--sound/drivers/mtpav.c3
-rw-r--r--sound/oss/Makefile4
-rw-r--r--sound/pci/au88x0/au88x0_core.c14
-rw-r--r--sound/pci/azt3328.c38
-rw-r--r--sound/pci/hda/hda_eld.c2
-rw-r--r--sound/pci/hda/hda_intel.c3
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_conexant.c217
-rw-r--r--sound/pci/hda/patch_hdmi.c7
-rw-r--r--sound/pci/hda/patch_realtek.c42
-rw-r--r--sound/pci/hda/patch_sigmatel.c15
-rw-r--r--sound/pci/hda/patch_via.c2
-rw-r--r--sound/pci/oxygen/oxygen.h2
-rw-r--r--sound/pci/oxygen/oxygen_mixer.c2
-rw-r--r--sound/pci/oxygen/xonar_cs43xx.c2
-rw-r--r--sound/pci/oxygen/xonar_dg.c36
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf.h2
-rw-r--r--sound/pcmcia/vx/vxp_ops.c2
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c2
-rw-r--r--sound/soc/blackfin/bf5xx-ssm2602.c2
-rw-r--r--sound/soc/codecs/cq93vc.c2
-rw-r--r--sound/soc/codecs/cx20442.c3
-rw-r--r--sound/soc/codecs/wm8903.c2
-rw-r--r--sound/soc/codecs/wm8903.h2
-rw-r--r--sound/soc/codecs/wm8994.c265
-rw-r--r--sound/soc/codecs/wm8995.c2
-rw-r--r--sound/soc/codecs/wm9081.c5
-rw-r--r--sound/soc/codecs/wm_hubs.c18
-rw-r--r--sound/soc/davinci/davinci-evm.c20
-rw-r--r--sound/soc/imx/eukrea-tlv320.c2
-rw-r--r--sound/soc/omap/ams-delta.c2
-rw-r--r--sound/soc/pxa/corgi.c4
-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/poodle.c2
-rw-r--r--sound/soc/pxa/spitz.c4
-rw-r--r--sound/soc/pxa/tosa.c4
-rw-r--r--sound/soc/pxa/zylonite.c4
-rw-r--r--sound/soc/samsung/neo1973_gta02_wm8753.c6
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c6
-rw-r--r--sound/soc/samsung/s3c24xx_simtec_hermes.c4
-rw-r--r--sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c4
-rw-r--r--sound/soc/samsung/s3c24xx_uda134x.c2
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dapm.c29
-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/mixer.c4
-rw-r--r--sound/usb/pcm.c7
-rw-r--r--sound/usb/quirks-table.h7
-rw-r--r--sound/usb/quirks.c3
-rw-r--r--sound/usb/usbaudio.h1
62 files changed, 786 insertions, 379 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 24d3013c023..d0cead38d5f 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -50,7 +50,11 @@ static void aaci_ac97_select_codec(struct aaci *aaci, struct snd_ac97 *ac97)
if (v & SLFR_1RXV)
readl(aaci->base + AACI_SL1RX);
- writel(maincr, aaci->base + AACI_MAINCR);
+ if (maincr != readl(aaci->base + AACI_MAINCR)) {
+ writel(maincr, aaci->base + AACI_MAINCR);
+ readl(aaci->base + AACI_MAINCR);
+ udelay(1);
+ }
}
/*
@@ -206,6 +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) {
@@ -218,15 +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;
@@ -256,6 +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) {
@@ -265,6 +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) {
@@ -277,15 +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;
@@ -315,6 +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);
}
}
@@ -357,7 +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,
@@ -365,12 +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;
@@ -378,27 +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;
}
@@ -414,7 +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;
}
@@ -440,12 +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) {
@@ -453,22 +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;
@@ -479,11 +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;
}
@@ -501,89 +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;
@@ -653,27 +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;
@@ -770,7 +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,
@@ -937,12 +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;
@@ -980,6 +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++)
@@ -993,10 +970,12 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
* disabling the channel doesn't clear the FIFO.
*/
writel(aaci->maincr & ~MAINCR_IE, aaci->base + AACI_MAINCR);
+ readl(aaci->base + AACI_MAINCR);
+ udelay(1);
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)
@@ -1005,7 +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;
@@ -1061,11 +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;
}
@@ -1078,8 +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 6a4a2eebdda..5791bd5bd2a 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/atmel/ac97c.c b/sound/atmel/ac97c.c
index 10c3a871a12..b310702c646 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -33,9 +33,12 @@
#include <linux/dw_dmac.h>
#include <mach/cpu.h>
-#include <mach/hardware.h>
#include <mach/gpio.h>
+#ifdef CONFIG_ARCH_AT91
+#include <mach/hardware.h>
+#endif
+
#include "ac97c.h"
enum {
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 7730575bfad..b8b31c433d6 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
{
struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
struct snd_timer *t = stime->timer;
+ unsigned long oruns;
if (!atomic_read(&stime->running))
return HRTIMER_NORESTART;
- hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
- snd_timer_interrupt(stime->timer, t->sticks);
+ oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
+ snd_timer_interrupt(stime->timer, t->sticks * oruns);
if (!atomic_read(&stime->running))
return HRTIMER_NORESTART;
@@ -104,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
}
static struct snd_timer_hardware hrtimer_hw = {
- .flags = SNDRV_TIMER_HW_AUTO,
+ .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
.open = snd_hrtimer_open,
.close = snd_hrtimer_close,
.start = snd_hrtimer_start,
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 4902ae56873..53b53e97c89 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/drivers/mtpav.c b/sound/drivers/mtpav.c
index da03597fc89..5c426df8767 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -55,14 +55,13 @@
#include <linux/err.h>
#include <linux/platform_device.h>
#include <linux/ioport.h>
+#include <linux/io.h>
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/rawmidi.h>
#include <linux/delay.h>
-#include <asm/io.h>
-
/*
* globals
*/
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index 96f14dcd0cd..90ffb99c6b1 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
$(obj)/bin2hex pss_synth < $< > $@
else
$(obj)/pss_boot.h:
- ( \
+ $(Q)( \
echo 'static unsigned char * pss_synth = NULL;'; \
echo 'static int pss_synthLen = 0;'; \
) > $@
@@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
$(obj)/hex2hex -i trix_boot < $< > $@
else
$(obj)/trix_boot.h:
- ( \
+ $(Q)( \
echo 'static unsigned char * trix_boot = NULL;'; \
echo 'static int trix_boot_len = 0;'; \
) > $@
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f356e0..16c0bdfbb16 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/azt3328.c b/sound/pci/azt3328.c
index 6117595fc07..573594bf322 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
snd_azf3328_dbgcallenter();
switch (bitrate) {
-#define AZF_FMT_XLATE(in_freq, out_bits) \
- do { \
- case AZF_FREQ_ ## in_freq: \
- freq = SOUNDFORMAT_FREQ_ ## out_bits; \
- break; \
- } while (0);
- AZF_FMT_XLATE(4000, SUSPECTED_4000)
- AZF_FMT_XLATE(4800, SUSPECTED_4800)
- /* the AZF3328 names it "5510" for some strange reason: */
- AZF_FMT_XLATE(5512, 5510)
- AZF_FMT_XLATE(6620, 6620)
- AZF_FMT_XLATE(8000, 8000)
- AZF_FMT_XLATE(9600, 9600)
- AZF_FMT_XLATE(11025, 11025)
- AZF_FMT_XLATE(13240, SUSPECTED_13240)
- AZF_FMT_XLATE(16000, 16000)
- AZF_FMT_XLATE(22050, 22050)
- AZF_FMT_XLATE(32000, 32000)
+ case AZF_FREQ_4000: freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+ case AZF_FREQ_4800: freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+ case AZF_FREQ_5512:
+ /* the AZF3328 names it "5510" for some strange reason */
+ freq = SOUNDFORMAT_FREQ_5510; break;
+ case AZF_FREQ_6620: freq = SOUNDFORMAT_FREQ_6620; break;
+ case AZF_FREQ_8000: freq = SOUNDFORMAT_FREQ_8000; break;
+ case AZF_FREQ_9600: freq = SOUNDFORMAT_FREQ_9600; break;
+ case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break;
+ case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+ case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break;
+ case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break;
+ case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
default:
snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
/* fall-through */
- AZF_FMT_XLATE(44100, 44100)
- AZF_FMT_XLATE(48000, 48000)
- AZF_FMT_XLATE(66200, SUSPECTED_66200)
-#undef AZF_FMT_XLATE
+ case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
+ case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
+ case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
}
/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 4a663471dad..74b0560289c 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
snd_print_pcm_rates(a->rates, buf, sizeof(buf));
if (a->format == AUDIO_CODING_TYPE_LPCM)
- snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
+ snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
else if (a->max_bitrate)
snprintf(buf2, sizeof(buf2),
", max bitrate = %d", a->max_bitrate);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2e91a991eb1..fcedad9a5fe 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),
@@ -2703,7 +2704,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
if (err < 0)
goto out_free;
#ifdef CONFIG_SND_HDA_PATCH_LOADER
- if (patch[dev]) {
+ if (patch[dev] && *patch[dev]) {
snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
patch[dev]);
err = snd_hda_load_patch(chip->bus, patch[dev]);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index a07b031090d..067982f4f18 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 9bb030a469c..4d5004e693f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -85,6 +85,7 @@ struct conexant_spec {
unsigned int auto_mic;
int auto_mic_ext; /* autocfg.inputs[] index for ext mic */
unsigned int need_dac_fix;
+ hda_nid_t slave_dig_outs[2];
/* capture */
unsigned int num_adc_nids;
@@ -127,6 +128,7 @@ struct conexant_spec {
unsigned int ideapad:1;
unsigned int thinkpad:1;
unsigned int hp_laptop:1;
+ unsigned int asus:1;
unsigned int ext_mic_present;
unsigned int recording;
@@ -352,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec)
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
spec->dig_in_nid;
}
+ if (spec->slave_dig_outs[0])
+ codec->slave_dig_outs = spec->slave_dig_outs;
}
return 0;
@@ -403,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec,
struct conexant_spec *spec;
struct conexant_jack *jack;
const char *name;
- int err;
+ int i, err;
spec = codec->spec;
snd_array_init(&spec->jacks, sizeof(*jack), 32);
+
+ jack = spec->jacks.list;
+ for (i = 0; i < spec->jacks.used; i++, jack++)
+ if (jack->nid == nid)
+ return 0 ; /* already present */
+
jack = snd_array_new(&spec->jacks);
name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ;
@@ -2100,7 +2110,7 @@ static int patch_cxt5051(struct hda_codec *codec)
static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-#define CXT5066_SPDIF_OUT 0x21
+static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
/* OLPC's microphone port is DC coupled for use with external sensors,
* therefore we use a 50% mic bias in order to center the input signal with
@@ -2312,6 +2322,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
}
}
+
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_asus_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_jack_detect(codec, 0x1b);
+ snd_printdd("CXT5066: external microphone present=%d\n", present);
+ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 1 : 0);
+}
+
+
/* toggle input of built-in digital mic and mic jack appropriately */
static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
{
@@ -2387,79 +2410,55 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
cxt5066_update_speaker(codec);
}
-/* unsolicited event for jack sensing */
-static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
+/* Dispatch the right mic autoswitch function */
+static void cxt5066_automic(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
- snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
- /* ignore mic events in DC mode; we're always using the jack */
- if (!spec->dc_enable)
- cxt5066_olpc_automic(codec);
- break;
- }
-}
-/* unsolicited event for jack sensing */
-static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res)
-{
- snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
+ if (spec->dell_vostro)
cxt5066_vostro_automic(codec);
- break;
- }
-}
-
-/* unsolicited event for jack sensing */
-static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
-{
- snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
+ else if (spec->ideapad)
cxt5066_ideapad_automic(codec);
- break;
- }
+ else if (spec->thinkpad)
+ cxt5066_thinkpad_automic(codec);
+ else if (spec->hp_laptop)
+ cxt5066_hp_laptop_automic(codec);
+ else if (spec->asus)
+ cxt5066_asus_automic(codec);
}
/* unsolicited event for jack sensing */
-static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
{
- snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+ struct conexant_spec *spec = codec->spec;
+ snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
switch (res >> 26) {
case CONEXANT_HP_EVENT:
cxt5066_hp_automute(codec);
break;
case CONEXANT_MIC_EVENT:
- cxt5066_hp_laptop_automic(codec);
+ /* ignore mic events in DC mode; we're always using the jack */
+ if (!spec->dc_enable)
+ cxt5066_olpc_automic(codec);
break;
}
}
/* unsolicited event for jack sensing */
-static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
{
- snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
+ snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
switch (res >> 26) {
case CONEXANT_HP_EVENT:
cxt5066_hp_automute(codec);
break;
case CONEXANT_MIC_EVENT:
- cxt5066_thinkpad_automic(codec);
+ cxt5066_automic(codec);
break;
}
}
+
static const struct hda_input_mux cxt5066_analog_mic_boost = {
.num_items = 5,
.items = {
@@ -2633,6 +2632,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
spec->recording = 0;
}
+static void conexant_check_dig_outs(struct hda_codec *codec,
+ hda_nid_t *dig_pins,
+ int num_pins)
+{
+ struct conexant_spec *spec = codec->spec;
+ hda_nid_t *nid_loc = &spec->multiout.dig_out_nid;
+ int i;
+
+ for (i = 0; i < num_pins; i++, dig_pins++) {
+ unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins);
+ if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE)
+ continue;
+ if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1)
+ continue;
+ if (spec->slave_dig_outs[0])
+ nid_loc++;
+ else
+ nid_loc = spec->slave_dig_outs;
+ }
+}
+
static struct hda_input_mux cxt5066_capture_source = {
.num_items = 4,
.items = {
@@ -3039,20 +3059,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
/* initialize jack-sensing, too */
static int cxt5066_init(struct hda_codec *codec)
{
- struct conexant_spec *spec = codec->spec;
-
snd_printdd("CXT5066: init\n");
conexant_init(codec);
if (codec->patch_ops.unsol_event) {
cxt5066_hp_automute(codec);
- if (spec->dell_vostro)
- cxt5066_vostro_automic(codec);
- else if (spec->ideapad)
- cxt5066_ideapad_automic(codec);
- else if (spec->thinkpad)
- cxt5066_thinkpad_automic(codec);
- else if (spec->hp_laptop)
- cxt5066_hp_laptop_automic(codec);
+ cxt5066_automic(codec);
}
cxt5066_set_mic_boost(codec);
return 0;
@@ -3080,6 +3091,7 @@ enum {
CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */
CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */
CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */
+ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
CXT5066_HP_LAPTOP, /* HP Laptop */
CXT5066_MODELS
};
@@ -3091,6 +3103,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
[CXT5066_DELL_VOSTRO] = "dell-vostro",
[CXT5066_IDEAPAD] = "ideapad",
[CXT5066_THINKPAD] = "thinkpad",
+ [CXT5066_ASUS] = "asus",
[CXT5066_HP_LAPTOP] = "hp-laptop",
};
@@ -3101,8 +3114,12 @@ 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_HP_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3111,7 +3128,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
{}
};
@@ -3133,7 +3152,8 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->multiout.max_channels = 2;
spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
spec->multiout.dac_nids = cxt5066_dac_nids;
- spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+ conexant_check_dig_outs(codec, cxt5066_digout_pin_nids,
+ ARRAY_SIZE(cxt5066_digout_pin_nids));
spec->num_adc_nids = 1;
spec->adc_nids = cxt5066_adc_nids;
spec->capsrc_nids = cxt5066_capsrc_nids;
@@ -3167,17 +3187,20 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->num_init_verbs++;
spec->dell_automute = 1;
break;
+ case CXT5066_ASUS:
case CXT5066_HP_LAPTOP:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->init_verbs[spec->num_init_verbs] =
cxt5066_init_verbs_hp_laptop;
spec->num_init_verbs++;
- spec->hp_laptop = 1;
+ spec->hp_laptop = board_config == CXT5066_HP_LAPTOP;
+ spec->asus = board_config == CXT5066_ASUS;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
/* no S/PDIF out */
- spec->multiout.dig_out_nid = 0;
+ if (board_config == CXT5066_HP_LAPTOP)
+ spec->multiout.dig_out_nid = 0;
/* input source automatically selected */
spec->input_mux = NULL;
spec->port_d_mode = 0;
@@ -3207,7 +3230,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_DELL_VOSTRO:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_vostro_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->init_verbs[0] = cxt5066_init_verbs_vostro;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
@@ -3224,7 +3247,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_IDEAPAD:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_ideapad_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->init_verbs[0] = cxt5066_init_verbs_ideapad;
@@ -3240,7 +3263,7 @@ static int patch_cxt5066(struct hda_codec *codec)
break;
case CXT5066_THINKPAD:
codec->patch_ops.init = cxt5066_init;
- codec->patch_ops.unsol_event = cxt5066_thinkpad_event;
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->init_verbs[0] = cxt5066_init_verbs_thinkpad;
@@ -3389,7 +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;
@@ -3708,9 +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[] = {
@@ -3722,7 +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]);
@@ -3738,6 +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)
@@ -3787,29 +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;
}
@@ -3881,6 +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",
@@ -3907,6 +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 2d5b83fa8d2..ec0fa2dd0a2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
hdmi_ai->ver = 0x01;
hdmi_ai->len = 0x0a;
hdmi_ai->CC02_CT47 = channels - 1;
+ hdmi_ai->CA = ca;
hdmi_checksum_audio_infoframe(hdmi_ai);
} else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */
struct dp_audio_infoframe *dp_ai;
@@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
dp_ai->len = 0x1b;
dp_ai->ver = 0x11 << 2;
dp_ai->CC02_CT47 = channels - 1;
+ dp_ai->CA = ca;
} else {
snd_printd("HDMI: unknown connection type at pin %d\n",
pin_nid);
@@ -1632,6 +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 },
@@ -1674,6 +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 be4df4c6fd5..4261bb8eec1 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;
@@ -2290,6 +2287,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc888_acer_aspire_4930g_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", 0x0f, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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 Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* 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),
@@ -10359,7 +10379,7 @@ static struct alc_config_preset alc882_presets[] = {
.init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_4930G] = {
- .mixers = { alc888_base_mixer,
+ .mixers = { alc888_acer_aspire_4930g_mixer,
alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
alc888_acer_aspire_4930g_verbs },
@@ -14954,9 +14974,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
- SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
{}
@@ -14990,7 +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),
@@ -18800,6 +18822,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
{}
};
@@ -19492,6 +19515,7 @@ static const struct alc_fixup alc662_fixups[] = {
};
static struct snd_pci_quirk alc662_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9ea48b425d0..bd7b123f644 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 a76c3260d94..63b0054200a 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/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c2ae63d17cd..f53897a708b 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -92,6 +92,8 @@ struct oxygen_model {
void (*update_dac_volume)(struct oxygen *chip);
void (*update_dac_mute)(struct oxygen *chip);
void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed);
+ unsigned int (*adjust_dac_routing)(struct oxygen *chip,
+ unsigned int play_routing);
void (*gpio_changed)(struct oxygen *chip);
void (*uart_input)(struct oxygen *chip);
void (*ac97_switch)(struct oxygen *chip,
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 9bff14d5895..26c7e8bcb22 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip)
(1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
(2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
(3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+ if (chip->model.adjust_dac_routing)
+ reg_value = chip->model.adjust_dac_routing(chip, reg_value);
oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
OXYGEN_PLAY_DAC0_SOURCE_MASK |
OXYGEN_PLAY_DAC1_SOURCE_MASK |
diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c
index 9f72d424969..252719101c4 100644
--- a/sound/pci/oxygen/xonar_cs43xx.c
+++ b/sound/pci/oxygen/xonar_cs43xx.c
@@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip,
unsigned int i;
snd_iprintf(buffer, "\nCS4398: 7?");
- for (i = 2; i <= 8; ++i)
+ for (i = 2; i < 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
snd_iprintf(buffer, "\n");
dump_cs4362a_registers(data, buffer);
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e1fa602eba7..bc6eb58be38 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -24,6 +24,11 @@
*
* SPI 0 -> CS4245
*
+ * I²S 1 -> CS4245
+ * I²S 2 -> CS4361 (center/LFE)
+ * I²S 3 -> CS4361 (surround)
+ * I²S 4 -> CS4361 (front)
+ *
* GPIO 3 <- ?
* GPIO 4 <- headphone detect
* GPIO 5 -> route input jack to line-in (0) or mic-in (1)
@@ -36,6 +41,7 @@
* input 1 <- aux
* input 2 <- front mic
* input 4 <- line/mic
+ * DAC out -> headphones
* aux out -> front panel headphones
*/
@@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip,
cs4245_write_cached(chip, CS4245_ADC_CTRL, value);
}
+static inline unsigned int shift_bits(unsigned int value,
+ unsigned int shift_from,
+ unsigned int shift_to,
+ unsigned int mask)
+{
+ if (shift_from < shift_to)
+ return (value << (shift_to - shift_from)) & mask;
+ else
+ return (value >> (shift_from - shift_to)) & mask;
+}
+
+static unsigned int adjust_dg_dac_routing(struct oxygen *chip,
+ unsigned int play_routing)
+{
+ return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC1_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC2_SOURCE_MASK) |
+ shift_bits(play_routing,
+ OXYGEN_PLAY_DAC0_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC3_SOURCE_SHIFT,
+ OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
static int output_switch_info(struct snd_kcontrol *ctl,
struct snd_ctl_elem_info *info)
{
@@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = {
.resume = dg_resume,
.set_dac_params = set_cs4245_dac_params,
.set_adc_params = set_cs4245_adc_params,
+ .adjust_dac_routing = adjust_dg_dac_routing,
.dump_registers = dump_cs4245_registers,
.model_data_size = sizeof(struct dg),
.device_config = PLAYBACK_0_TO_I2S |
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index bd26e092aea..6ce9ad70029 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -22,7 +22,7 @@
#define __PDAUDIOCF_H
#include <sound/pcm.h>
-#include <asm/io.h>
+#include <linux/io.h>
#include <linux/interrupt.h>
#include <pcmcia/cistpl.h>
#include <pcmcia/ds.h>
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 989e04abb52..fe33e122e37 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -23,8 +23,8 @@
#include <linux/delay.h>
#include <linux/device.h>
#include <linux/firmware.h>
+#include <linux/io.h>
#include <sound/core.h>
-#include <asm/io.h>
#include "vxpocket.h"
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e06b0..5e4d499d843 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
.cpu_dai_name = "atmel-ssc-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "atmel_pcm-audio",
- .codec_name = "tlv320aic23-codec.0-0x1a",
+ .codec_name = "tlv320aic23-codec.0-001a",
.init = afeb9260_tlv320aic23_init,
.ops = &afeb9260_ops,
};
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24c185..ad28663f5bb 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
.cpu_dai_name = "bf5xx-i2s",
.codec_dai_name = "ssm2602-hifi",
.platform_name = "bf5xx-pcm-audio",
- .codec_name = "ssm2602-codec.0-0x1b",
+ .codec_name = "ssm2602-codec.0-001b",
.ops = &bf5xx_ssm2602_ops,
};
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd067f7..347a567b01e 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
static int cq93vc_probe(struct snd_soc_codec *codec)
{
- struct davinci_vc *davinci_vc = codec->dev->platform_data;
+ struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
davinci_vc->cq93vc.codec = codec;
codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e860d22..0bb424af956 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
return 0;
}
+static const u8 cx20442_reg;
+
static struct snd_soc_codec_driver cx20442_codec_dev = {
.probe = cx20442_codec_probe,
.remove = cx20442_codec_remove,
+ .reg_cache_default = &cx20442_reg,
.reg_cache_size = 1,
.reg_word_size = sizeof(u8),
.read = cx20442_read_reg_cache,
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 987476a5895..017d99ceb42 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 e8490f3edd0..e3ec2433b21 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/wm8994.c b/sound/soc/codecs/wm8994.c
index 247a6a99feb..4afbe3b2e44 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("DAC1R", 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"),
@@ -1284,12 +1447,9 @@ 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", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
0, WM8994_POWER_MANAGEMENT_4, 8, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1458,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
0, WM8994_POWER_MANAGEMENT_4, 10, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1505,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1368,14 +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),
@@ -1515,14 +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" },
@@ -1531,7 +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" },
@@ -1539,13 +1689,17 @@ 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" },
{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
/* AIF3 output */
@@ -1578,6 +1732,31 @@ 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" },
+ { "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+ { "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+};
+
static const struct snd_soc_dapm_route wm8994_intercon[] = {
{ "AIF2DACL", NULL, "AIF2DAC Mux" },
{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -2386,7 +2565,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
#define WM8994_RATES SNDRV_PCM_RATE_8000_96000
@@ -2501,6 +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);
@@ -2834,11 +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);
}
/**
@@ -3112,6 +3306,21 @@ 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,
@@ -3129,6 +3338,16 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
case WM8994:
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
+
+ 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, wm8958_intercon,
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 6045cbde492..608c84c5aa8 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
else
val = 0;
- return snd_soc_update_bits(codec, reg, mask, reg);
+ return snd_soc_update_bits(codec, reg, mask, val);
}
/* The size in bits of the FLL divide multiplied by 10
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 43825b2102a..cce704c275c 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 c466982eed2..51689270606 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
static void calibrate_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+ s8 offset;
u16 reg, reg_l, reg_r, dcs_cfg;
/* If we're using a digital only path and have a previously
@@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
hubs->dcs_codes);
/* HPOUT1L */
- if (reg_l + hubs->dcs_codes > 0 &&
- reg_l + hubs->dcs_codes < 0xff)
- reg_l += hubs->dcs_codes;
- dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ offset = reg_l;
+ offset += hubs->dcs_codes;
+ dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1R */
- if (reg_r + hubs->dcs_codes > 0 &&
- reg_r + hubs->dcs_codes < 0xff)
- reg_r += hubs->dcs_codes;
- dcs_cfg |= reg_r;
+ offset = reg_r;
+ offset += hubs->dcs_codes;
+ dcs_cfg |= (u8)offset;
dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
@@ -675,6 +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/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6bacc68..fe7984221eb 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,12 +218,24 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
.ops = &evm_spdif_ops,
},
};
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai_name = "davinci-mcasp.1",
+ .codec_dai_name = "tlv320aic3x-hifi",
+ .codec_name = "tlv320aic3x-codec.1-0018",
+ .platform_name = "davinci-pcm-audio",
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
.codec_dai_name = "tlv320aic3x-hifi",
- .codec_name = "tlv320aic3x-codec.0-001a",
+ .codec_name = "tlv320aic3x-codec.1-0018",
.platform_name = "davinci-pcm-audio",
.init = evm_aic3x_init,
.ops = &evm_ops,
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
static struct snd_soc_card da830_snd_soc_card = {
.name = "DA830/OMAP-L137 EVM",
- .dai_link = &da8xx_evm_dai,
+ .dai_link = &da830_evm_dai,
.num_links = 1,
};
static struct snd_soc_card da850_snd_soc_card = {
.name = "DA850/OMAP-L138 EVM",
- .dai_link = &da8xx_evm_dai,
+ .dai_link = &da850_evm_dai,
.num_links = 1,
};
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index e20c9e1457c..1e9bccae4e8 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/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdcee21..3167be68962 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
/* Set up digital mute if not provided by the codec */
if (!codec_dai->driver->ops) {
codec_dai->driver->ops = &ams_delta_dai_ops;
- } else if (!codec_dai->driver->ops->digital_mute) {
- codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
} else {
ams_delta_ops.startup = ams_delta_startup;
ams_delta_ops.shutdown = ams_delta_shutdown;
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0d5fc..784cff5f67e 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
- .cpu_dai_name = "pxa-is2-dai",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec-0.001a",
+ .codec_name = "wm8731-codec-0.001b",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 28333e7d9c5..dc65650a6fa 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 01bf31675c5..51897fcd911 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 c6a37c6ef23..053ed208e59 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 fc22e6eefc9..b13a4252812 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 0d70fc8c12b..38ca6759907 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 857db96d4a4..504e4004f00 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/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee115e2..a7d4999f9b2 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
.cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8731-codec.0-001a",
+ .codec_name = "wm8731-codec.0-001b",
.init = poodle_wm8731_init,
.ops = &poodle_ops,
};
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69b957..8e157135063 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link spitz_dai = {
.name = "wm8750",
.stream_name = "WM8750",
- .cpu_dai_name = "pxa-is2",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
- .codec_name = "wm8750-codec.0-001a",
+ .codec_name = "wm8750-codec.0-001b",
.init = spitz_wm8750_init,
.ops = &spitz_ops,
};
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index f75804ef089..4b6e5d608b4 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 b222a7d7202..25bba108fea 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/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 3eec610c10f..0d0ae2b9eef 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -397,11 +397,11 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
.stream_name = "WM8753 HiFi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
.init = neo1973_gta02_wm8753_init,
.platform_name = "samsung-audio",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_gta02_hifi_ops,
},
{ /* Voice via BT */
@@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
.ops = &neo1973_gta02_voice_ops,
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.platform_name = "samsung-audio",
},
};
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a24514beb..d20815d5ab2 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -559,9 +559,9 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.name = "WM8753",
.stream_name = "WM8753 HiFi",
.platform_name = "samsung-audio",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
@@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
.platform_name = "samsung-audio",
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
- .codec_name = "wm8753-codec.0-0x1a",
+ .codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_voice_ops,
},
};
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index bb4292e3596..08fcaaa6690 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -94,8 +94,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic33 = {
.name = "tlv320aic33",
.stream_name = "TLV320AIC33",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .codec_name = "tlv320aic3x-codec.0-001a",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index fbba4e36772..116e3e67016 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -85,8 +85,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
static struct snd_soc_dai_link simtec_dai_aic23 = {
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
- .codec_name = "tlv320aic3x-codec.0-0x1a",
- .cpu_dai_name = "s3c24xx-i2s",
+ .codec_name = "tlv320aic3x-codec.0-001a",
+ .cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "tlv320aic3x-hifi",
.platform_name = "samsung-audio",
.init = simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index cdc8ecbcb8e..2c09e93dd56 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
.stream_name = "UDA134X",
.codec_name = "uda134x-hifi",
.codec_dai_name = "uda134x-hifi",
- .cpu_dai_name = "s3c24xx-i2s",
+ .cpu_dai_name = "s3c24xx-iis",
.ops = &s3c24xx_uda134x_ops,
.platform_name = "samsung-audio",
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bac7291b6ff..c3f6f1e7279 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
rtd = &card->rtd_aux[num];
name = aux_dev->name;
}
+ rtd->card = card;
/* machine controls, routes and widgets are not prefixed */
temp = codec->name_prefix;
@@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
/* register the rtd device */
rtd->codec = codec;
- rtd->card = card;
rtd->dev.parent = card->dev;
rtd->dev.release = rtd_release;
rtd->dev.init_name = name;
@@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
goto out;
found:
- if (!try_module_get(codec->dev->driver->owner))
- return -ENODEV;
-
ret = soc_probe_codec(card, codec);
if (ret < 0)
return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730ab563..25e54230cc6 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;
@@ -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;
}
@@ -1742,7 +1763,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- unsigned int val, val_mask;
+ unsigned int val;
int connect, change;
struct snd_soc_dapm_update update;
@@ -1750,13 +1771,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
if (invert)
val = max - val;
- val_mask = mask << shift;
+ mask = mask << shift;
val = val << shift;
mutex_lock(&widget->codec->mutex);
widget->value = val;
- change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+ change = snd_soc_test_bits(widget->codec, reg, mask, val);
if (change) {
if (val)
/* new connection */
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 68b97477577..66eabafb1c2 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 2f218c77fff..a1a47088fd0 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 800f7cb4f25..c0f8270bc19 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/mixer.c b/sound/usb/mixer.c
index 7df89b3d7de..85af6051b52 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -95,7 +95,7 @@ enum {
};
-/*E-mu 0202(0404) eXtension Unit(XU) control*/
+/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/
enum {
USB_XU_CLOCK_RATE = 0xe301,
USB_XU_CLOCK_SOURCE = 0xe302,
@@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
cval->initialized = 1;
} else {
if (type == USB_XU_CLOCK_RATE) {
- /* E-Mu USB 0404/0202/TrackerPre
+ /* E-Mu USB 0404/0202/TrackerPre/0204
* samplerate control quirk
*/
cval->min = 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 4132522ac90..e3f680526cb 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/quirks-table.h b/sound/usb/quirks-table.h
index 35999874d30..921a86fd988 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -79,6 +79,13 @@
.idProduct = 0x3f0a,
.bInterfaceClass = USB_CLASS_AUDIO,
},
+{
+ /* E-Mu 0204 USB */
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x041e,
+ .idProduct = 0x3f19,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
/*
* Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index cf8bf088394..e314cdb8500 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat
}
/*
- * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device,
+ * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device,
* not for interface.
*/
@@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
+ case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
set_format_emu_quirk(subs, fmt);
break;
}
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index db3eb21627e..6e66fffe87f 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;