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-rw-r--r--Documentation/sound/alsa/soc/jack.txt71
-rw-r--r--sound/arm/pxa2xx-ac97-lib.c15
-rw-r--r--sound/atmel/abdac.c4
-rw-r--r--sound/atmel/ac97c.c128
-rw-r--r--sound/atmel/ac97c.h14
-rw-r--r--sound/core/oss/mixer_oss.c8
-rw-r--r--sound/isa/opl3sa2.c1
-rw-r--r--sound/pci/hda/patch_analog.c2
-rw-r--r--sound/pci/hda/patch_realtek.c5
-rw-r--r--sound/pci/hda/patch_sigmatel.c20
-rw-r--r--sound/ppc/powermac.c2
-rw-r--r--sound/soc/codecs/ak4535.c3
-rw-r--r--sound/soc/codecs/twl4030.c59
-rw-r--r--sound/soc/codecs/twl4030.h1
-rw-r--r--sound/soc/codecs/wm9705.c37
-rw-r--r--sound/soc/fsl/fsl_dma.c29
-rw-r--r--sound/soc/fsl/fsl_ssi.c99
-rw-r--r--sound/soc/omap/omap-mcbsp.c11
-rw-r--r--sound/soc/pxa/Kconfig10
-rw-r--r--sound/soc/pxa/Makefile2
-rw-r--r--sound/soc/pxa/magician.c560
-rw-r--r--sound/soc/pxa/pxa-ssp.c12
-rw-r--r--sound/soc/soc-core.c20
-rw-r--r--sound/usb/usbaudio.c255
24 files changed, 1133 insertions, 235 deletions
diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt
new file mode 100644
index 00000000000..fcf82a41729
--- /dev/null
+++ b/Documentation/sound/alsa/soc/jack.txt
@@ -0,0 +1,71 @@
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h. ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+ user visible jack. In embedded systems it is common for multiple
+ to be present on a single jack but handled by separate bits of
+ hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+ automatically based on the detected jack status (eg, turning off the
+ headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone. Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space. The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack. Each snd_soc_jack has zero or more of these
+which are updated automatically. They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins(). The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update. The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function. Other methods are
+also available, for example integrated into CODECs. One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware. The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 7793d2a511c..0afd1a8226f 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -238,6 +238,8 @@ static inline void pxa_ac97_cold_pxa3xx(void)
bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
{
+ unsigned long gsr;
+
#ifdef CONFIG_PXA25x
if (cpu_is_pxa25x())
pxa_ac97_warm_pxa25x();
@@ -254,10 +256,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
else
#endif
BUG();
-
- if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+ gsr = GSR | gsr_bits;
+ if (!(gsr & (GSR_PCR | GSR_SCR))) {
printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
- __func__, gsr_bits);
+ __func__, gsr);
return false;
}
@@ -268,6 +270,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
{
+ unsigned long gsr;
+
#ifdef CONFIG_PXA25x
if (cpu_is_pxa25x())
pxa_ac97_cold_pxa25x();
@@ -285,9 +289,10 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
#endif
BUG();
- if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+ gsr = GSR | gsr_bits;
+ if (!(gsr & (GSR_PCR | GSR_SCR))) {
printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
- __func__, gsr_bits);
+ __func__, gsr);
return false;
}
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 28b3c7f7cfe..f2f41c85422 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -165,7 +165,7 @@ static struct snd_pcm_hardware atmel_abdac_hw = {
.buffer_bytes_max = 64 * 4096,
.period_bytes_min = 4096,
.period_bytes_max = 4096,
- .periods_min = 4,
+ .periods_min = 6,
.periods_max = 64,
};
@@ -502,7 +502,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, card);
dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n",
- dac->regs, dac->dma.chan->dev->device.bus_id);
+ dac->regs, dev_name(&dac->dma.chan->dev->device));
return retval;
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index dd72e00e5ae..0c0f8771656 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1,5 +1,5 @@
/*
- * Driver for the Atmel AC97C controller
+ * Driver for Atmel AC97C
*
* Copyright (C) 2005-2009 Atmel Corporation
*
@@ -10,6 +10,7 @@
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/bitmap.h>
+#include <linux/device.h>
#include <linux/dmaengine.h>
#include <linux/dma-mapping.h>
#include <linux/init.h>
@@ -65,6 +66,7 @@ struct atmel_ac97c {
/* Serialize access to opened variable */
spinlock_t lock;
void __iomem *regs;
+ int irq;
int opened;
int reset_pin;
};
@@ -150,10 +152,10 @@ static struct snd_pcm_hardware atmel_ac97c_hw = {
.rate_max = 48000,
.channels_min = 1,
.channels_max = 2,
- .buffer_bytes_max = 64 * 4096,
+ .buffer_bytes_max = 2 * 2 * 64 * 2048,
.period_bytes_min = 4096,
.period_bytes_max = 4096,
- .periods_min = 4,
+ .periods_min = 6,
.periods_max = 64,
};
@@ -297,9 +299,11 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
{
struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
- unsigned long word = 0;
+ unsigned long word = ac97c_readl(chip, OCA);
int retval;
+ word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
+
/* assign channels to AC97C channel A */
switch (runtime->channels) {
case 1:
@@ -312,7 +316,6 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
default:
/* TODO: support more than two channels */
return -EINVAL;
- break;
}
ac97c_writel(chip, OCA, word);
@@ -324,13 +327,25 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
word |= AC97C_CMR_CEM_LITTLE;
break;
case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
- default:
word &= ~(AC97C_CMR_CEM_LITTLE);
break;
+ default:
+ word = ac97c_readl(chip, OCA);
+ word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
+ ac97c_writel(chip, OCA, word);
+ return -EINVAL;
}
+ /* Enable underrun interrupt on channel A */
+ word |= AC97C_CSR_UNRUN;
+
ac97c_writel(chip, CAMR, word);
+ /* Enable channel A event interrupt */
+ word = ac97c_readl(chip, IMR);
+ word |= AC97C_SR_CAEVT;
+ ac97c_writel(chip, IER, word);
+
/* set variable rate if needed */
if (runtime->rate != 48000) {
word = ac97c_readl(chip, MR);
@@ -359,9 +374,11 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
{
struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
- unsigned long word = 0;
+ unsigned long word = ac97c_readl(chip, ICA);
int retval;
+ word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
+
/* assign channels to AC97C channel A */
switch (runtime->channels) {
case 1:
@@ -374,7 +391,6 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
default:
/* TODO: support more than two channels */
return -EINVAL;
- break;
}
ac97c_writel(chip, ICA, word);
@@ -386,13 +402,25 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
word |= AC97C_CMR_CEM_LITTLE;
break;
case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
- default:
word &= ~(AC97C_CMR_CEM_LITTLE);
break;
+ default:
+ word = ac97c_readl(chip, ICA);
+ word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT));
+ ac97c_writel(chip, ICA, word);
+ return -EINVAL;
}
+ /* Enable overrun interrupt on channel A */
+ word |= AC97C_CSR_OVRUN;
+
ac97c_writel(chip, CAMR, word);
+ /* Enable channel A event interrupt */
+ word = ac97c_readl(chip, IMR);
+ word |= AC97C_SR_CAEVT;
+ ac97c_writel(chip, IER, word);
+
/* set variable rate if needed */
if (runtime->rate != 48000) {
word = ac97c_readl(chip, MR);
@@ -543,6 +571,43 @@ static struct snd_pcm_ops atmel_ac97_capture_ops = {
.pointer = atmel_ac97c_capture_pointer,
};
+static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
+{
+ struct atmel_ac97c *chip = (struct atmel_ac97c *)dev;
+ irqreturn_t retval = IRQ_NONE;
+ u32 sr = ac97c_readl(chip, SR);
+ u32 casr = ac97c_readl(chip, CASR);
+ u32 cosr = ac97c_readl(chip, COSR);
+
+ if (sr & AC97C_SR_CAEVT) {
+ dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
+ casr & AC97C_CSR_OVRUN ? " OVRUN" : "",
+ casr & AC97C_CSR_RXRDY ? " RXRDY" : "",
+ casr & AC97C_CSR_UNRUN ? " UNRUN" : "",
+ casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
+ casr & AC97C_CSR_TXRDY ? " TXRDY" : "",
+ !casr ? " NONE" : "");
+ retval = IRQ_HANDLED;
+ }
+
+ if (sr & AC97C_SR_COEVT) {
+ dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n",
+ cosr & AC97C_CSR_OVRUN ? " OVRUN" : "",
+ cosr & AC97C_CSR_RXRDY ? " RXRDY" : "",
+ cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
+ cosr & AC97C_CSR_TXRDY ? " TXRDY" : "",
+ !cosr ? " NONE" : "");
+ retval = IRQ_HANDLED;
+ }
+
+ if (retval == IRQ_NONE) {
+ dev_err(&chip->pdev->dev, "spurious interrupt sr 0x%08x "
+ "casr 0x%08x cosr 0x%08x\n", sr, casr, cosr);
+ }
+
+ return retval;
+}
+
static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip)
{
struct snd_pcm *pcm;
@@ -665,17 +730,17 @@ static bool filter(struct dma_chan *chan, void *slave)
static void atmel_ac97c_reset(struct atmel_ac97c *chip)
{
- ac97c_writel(chip, MR, AC97C_MR_WRST);
+ ac97c_writel(chip, MR, 0);
+ ac97c_writel(chip, MR, AC97C_MR_ENA);
+ ac97c_writel(chip, CAMR, 0);
+ ac97c_writel(chip, COMR, 0);
if (gpio_is_valid(chip->reset_pin)) {
gpio_set_value(chip->reset_pin, 0);
/* AC97 v2.2 specifications says minimum 1 us. */
- udelay(10);
+ udelay(2);
gpio_set_value(chip->reset_pin, 1);
}
-
- udelay(1);
- ac97c_writel(chip, MR, AC97C_MR_ENA);
}
static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
@@ -690,6 +755,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
.read = atmel_ac97c_read,
};
int retval;
+ int irq;
regs = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!regs) {
@@ -703,6 +769,12 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
return -ENXIO;
}
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0) {
+ dev_dbg(&pdev->dev, "could not get irq\n");
+ return -ENXIO;
+ }
+
pclk = clk_get(&pdev->dev, "pclk");
if (IS_ERR(pclk)) {
dev_dbg(&pdev->dev, "no peripheral clock\n");
@@ -719,6 +791,13 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
chip = get_chip(card);
+ retval = request_irq(irq, atmel_ac97c_interrupt, 0, "AC97C", chip);
+ if (retval) {
+ dev_dbg(&pdev->dev, "unable to request irq %d\n", irq);
+ goto err_request_irq;
+ }
+ chip->irq = irq;
+
spin_lock_init(&chip->lock);
strcpy(card->driver, "Atmel AC97C");
@@ -747,14 +826,18 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
snd_card_set_dev(card, &pdev->dev);
+ atmel_ac97c_reset(chip);
+
+ /* Enable overrun interrupt from codec channel */
+ ac97c_writel(chip, COMR, AC97C_CSR_OVRUN);
+ ac97c_writel(chip, IER, ac97c_readl(chip, IMR) | AC97C_SR_COEVT);
+
retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus);
if (retval) {
dev_dbg(&pdev->dev, "could not register on ac97 bus\n");
goto err_ac97_bus;
}
- atmel_ac97c_reset(chip);
-
retval = atmel_ac97c_mixer_new(chip);
if (retval) {
dev_dbg(&pdev->dev, "could not register ac97 mixer\n");
@@ -773,7 +856,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
chip->dma.rx_chan = dma_request_channel(mask, filter, dws);
dev_info(&chip->pdev->dev, "using %s for DMA RX\n",
- chip->dma.rx_chan->dev->device.bus_id);
+ dev_name(&chip->dma.rx_chan->dev->device));
set_bit(DMA_RX_CHAN_PRESENT, &chip->flags);
}
@@ -789,7 +872,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
chip->dma.tx_chan = dma_request_channel(mask, filter, dws);
dev_info(&chip->pdev->dev, "using %s for DMA TX\n",
- chip->dma.tx_chan->dev->device.bus_id);
+ dev_name(&chip->dma.tx_chan->dev->device));
set_bit(DMA_TX_CHAN_PRESENT, &chip->flags);
}
@@ -809,7 +892,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
retval = snd_card_register(card);
if (retval) {
dev_dbg(&pdev->dev, "could not register sound card\n");
- goto err_ac97_bus;
+ goto err_dma;
}
platform_set_drvdata(pdev, card);
@@ -836,6 +919,8 @@ err_ac97_bus:
iounmap(chip->regs);
err_ioremap:
+ free_irq(irq, chip);
+err_request_irq:
snd_card_free(card);
err_snd_card_new:
clk_disable(pclk);
@@ -884,9 +969,14 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev)
if (gpio_is_valid(chip->reset_pin))
gpio_free(chip->reset_pin);
+ ac97c_writel(chip, CAMR, 0);
+ ac97c_writel(chip, COMR, 0);
+ ac97c_writel(chip, MR, 0);
+
clk_disable(chip->pclk);
clk_put(chip->pclk);
iounmap(chip->regs);
+ free_irq(chip->irq, chip);
if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags))
dma_release_channel(chip->dma.rx_chan);
diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h
index c17bd582598..ecbba5021c8 100644
--- a/sound/atmel/ac97c.h
+++ b/sound/atmel/ac97c.h
@@ -1,5 +1,5 @@
/*
- * Register definitions for the Atmel AC97C controller
+ * Register definitions for Atmel AC97C
*
* Copyright (C) 2005-2009 Atmel Corporation
*
@@ -17,10 +17,6 @@
#define AC97C_CATHR 0x24
#define AC97C_CASR 0x28
#define AC97C_CAMR 0x2c
-#define AC97C_CBRHR 0x30
-#define AC97C_CBTHR 0x34
-#define AC97C_CBSR 0x38
-#define AC97C_CBMR 0x3c
#define AC97C_CORHR 0x40
#define AC97C_COTHR 0x44
#define AC97C_COSR 0x48
@@ -46,8 +42,10 @@
#define AC97C_MR_VRA (1 << 2)
#define AC97C_CSR_TXRDY (1 << 0)
+#define AC97C_CSR_TXEMPTY (1 << 1)
#define AC97C_CSR_UNRUN (1 << 2)
#define AC97C_CSR_RXRDY (1 << 4)
+#define AC97C_CSR_OVRUN (1 << 5)
#define AC97C_CSR_ENDTX (1 << 10)
#define AC97C_CSR_ENDRX (1 << 14)
@@ -61,11 +59,15 @@
#define AC97C_CMR_DMAEN (1 << 22)
#define AC97C_SR_CAEVT (1 << 3)
+#define AC97C_SR_COEVT (1 << 2)
+#define AC97C_SR_WKUP (1 << 1)
+#define AC97C_SR_SOF (1 << 0)
+#define AC97C_CH_MASK(slot) \
+ (0x7 << (3 * (AC97_SLOT_##slot - 3)))
#define AC97C_CH_ASSIGN(slot, channel) \
(AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3)))
#define AC97C_CHANNEL_NONE 0x0
#define AC97C_CHANNEL_A 0x1
-#define AC97C_CHANNEL_B 0x2
#endif /* __SOUND_ATMEL_AC97C_H */
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index e570649184e..5dcd8a52697 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -703,19 +703,27 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer,
if (left || right) {
if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH)
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0);
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH)
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0);
if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH)
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0);
if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE)
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1);
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE)
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1);
if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE)
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1);
} else {
if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) {
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) {
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0);
} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) {
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0);
} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) {
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) {
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1);
} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) {
snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1);
}
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index ef95279da7a..0481a55334b 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -481,6 +481,7 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1,
OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1),
OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1,
db_scale_5bit_12db_max),
+OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0),
};
static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = {
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 5bb48ee8b6c..38ad3f7b040 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3256,7 +3256,7 @@ static const char *ad1884_slave_vols[] = {
"Mic Playback Volume",
"CD Playback Volume",
"Internal Mic Playback Volume",
- "Docking Mic Playback Volume"
+ "Docking Mic Playback Volume",
/* "Beep Playback Volume", */
"IEC958 Playback Volume",
NULL
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 82097790f6f..f35e58a2d92 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8764,6 +8764,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
{}
};
+static hda_nid_t alc883_slave_dig_outs[] = {
+ ALC1200_DIGOUT_NID, 0,
+};
+
static hda_nid_t alc1200_slave_dig_outs[] = {
ALC883_DIGOUT_NID, 0,
};
@@ -8809,6 +8813,7 @@ static struct alc_config_preset alc883_presets[] = {
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
+ .slave_dig_outs = alc883_slave_dig_outs,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
.channel_mode = alc883_3ST_6ch_intel_modes,
.need_dac_fix = 1,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b5e108aa8f6..61996a2f45d 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4413,6 +4413,24 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
if (spec->num_pwrs > 0)
stac92xx_pin_sense(codec, event->nid);
stac92xx_report_jack(codec, event->nid);
+
+ switch (codec->subsystem_id) {
+ case 0x103c308f:
+ if (event->nid == 0xb) {
+ int pin = AC_PINCTL_IN_EN;
+
+ if (get_pin_presence(codec, 0xa)
+ && get_pin_presence(codec, 0xb))
+ pin |= AC_PINCTL_VREF_80;
+ if (!get_pin_presence(codec, 0xb))
+ pin |= AC_PINCTL_VREF_80;
+
+ /* toggle VREF state based on mic + hp pin
+ * status
+ */
+ stac92xx_auto_set_pinctl(codec, 0x0a, pin);
+ }
+ }
break;
case STAC_VREF_EVENT:
data = snd_hda_codec_read(codec, codec->afg, 0,
@@ -4895,6 +4913,7 @@ again:
switch (codec->vendor_id) {
case 0x111d7604:
case 0x111d7605:
+ case 0x111d76d5:
if (spec->board_config == STAC_92HD83XXX_PWR_REF)
break;
spec->num_pwrs = 0;
@@ -5707,6 +5726,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
{ .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
{ .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx},
{ .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
{ .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
{ .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index 5a929069dce..a2b69b8cff4 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -51,7 +51,7 @@ static struct platform_device *device;
/*
*/
-static int __init snd_pmac_probe(struct platform_device *devptr)
+static int __devinit snd_pmac_probe(struct platform_device *devptr)
{
struct snd_card *card;
struct snd_pmac *chip;
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 1f63d387a2f..dd338020276 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -659,7 +659,8 @@ static int ak4535_remove(struct platform_device *pdev)
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_unregister_device(codec->control_data);
+ if (codec->control_data)
+ i2c_unregister_device(codec->control_data);
i2c_del_driver(&ak4535_i2c_driver);
#endif
kfree(codec->private_data);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 97738e2ece0..bfda7a88e82 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -122,6 +122,9 @@ struct twl4030_priv {
unsigned int bypass_state;
unsigned int codec_powered;
unsigned int codec_muted;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
};
/*
@@ -1217,6 +1220,50 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
+static int twl4030_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
+
+ /* If we already have a playback or capture going then constrain
+ * this substream to match it.
+ */
+ if (twl4030->master_substream) {
+ struct snd_pcm_runtime *master_runtime;
+ master_runtime = twl4030->master_substream->runtime;
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ twl4030->slave_substream = substream;
+ } else
+ twl4030->master_substream = substream;
+
+ return 0;
+}
+
+static void twl4030_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
+
+ if (twl4030->master_substream == substream)
+ twl4030->master_substream = twl4030->slave_substream;
+
+ twl4030->slave_substream = NULL;
+}
+
static int twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1224,8 +1271,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
u8 mode, old_mode, format, old_format;
+ if (substream == twl4030->slave_substream)
+ /* Ignoring hw_params for slave substream */
+ return 0;
+
/* bit rate */
old_mode = twl4030_read_reg_cache(codec,
TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1259,6 +1311,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
case 48000:
mode |= TWL4030_APLL_RATE_48000;
break;
+ case 96000:
+ mode |= TWL4030_APLL_RATE_96000;
+ break;
default:
printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
params_rate(params));
@@ -1384,6 +1439,8 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
static struct snd_soc_dai_ops twl4030_dai_ops = {
+ .startup = twl4030_startup,
+ .shutdown = twl4030_shutdown,
.hw_params = twl4030_hw_params,
.set_sysclk = twl4030_set_dai_sysclk,
.set_fmt = twl4030_set_dai_fmt,
@@ -1395,7 +1452,7 @@ struct snd_soc_dai twl4030_dai = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = TWL4030_RATES,
+ .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
.formats = TWL4030_FORMATS,},
.capture = {
.stream_name = "Capture",
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 33dbb144dad..cb63765db1d 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -109,6 +109,7 @@
#define TWL4030_APLL_RATE_32000 0x80
#define TWL4030_APLL_RATE_44100 0x90
#define TWL4030_APLL_RATE_48000 0xA0
+#define TWL4030_APLL_RATE_96000 0xE0
#define TWL4030_SEL_16K 0x04
#define TWL4030_CODECPDZ 0x02
#define TWL4030_OPT_MODE 0x01
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 3265817c5c2..6e23a81dba7 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec)
return -EIO;
}
+#ifdef CONFIG_PM
+static int wm9705_soc_suspend(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
+
+ return 0;
+}
+
+static int wm9705_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9705_reset(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+
+ return 0;
+}
+#else
+#define wm9705_soc_suspend NULL
+#define wm9705_soc_resume NULL
+#endif
+
static int wm9705_soc_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev)
struct snd_soc_codec_device soc_codec_dev_wm9705 = {
.probe = wm9705_soc_probe,
.remove = wm9705_soc_remove,
+ .suspend = wm9705_soc_suspend,
+ .resume = wm9705_soc_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index b3eb8570cd7..b1a3a278819 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -300,7 +300,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = fsl_dma_dmamask;
- ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev,
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[0].substream->dma_buffer);
if (ret) {
@@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
return -ENOMEM;
}
- ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev,
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[1].substream->dma_buffer);
if (ret) {
@@ -418,7 +418,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
return -EBUSY;
}
- dma_private = dma_alloc_coherent(substream->pcm->dev,
+ dma_private = dma_alloc_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL);
if (!dma_private) {
dev_err(substream->pcm->card->dev,
@@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
dev_err(substream->pcm->card->dev,
"can't register ISR for IRQ %u (ret=%i)\n",
dma_private->irq, ret);
- dma_free_coherent(substream->pcm->dev,
+ dma_free_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private),
dma_private, dma_private->ld_buf_phys);
return ret;
@@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
else
position = in_be32(&dma_channel->dar);
+ /*
+ * When capture is started, the SSI immediately starts to fill its FIFO.
+ * This means that the DMA controller is not started until the FIFO is
+ * full. However, ALSA calls this function before that happens, when
+ * MR.DAR is still zero. In this case, just return zero to indicate
+ * that nothing has been received yet.
+ */
+ if (!position)
+ return 0;
+
+ if ((position < dma_private->dma_buf_phys) ||
+ (position > dma_private->dma_buf_end)) {
+ dev_err(substream->pcm->card->dev,
+ "dma pointer is out of range, halting stream\n");
+ return SNDRV_PCM_POS_XRUN;
+ }
+
frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
/*
@@ -761,13 +778,13 @@ static int fsl_dma_close(struct snd_pcm_substream *substream)
free_irq(dma_private->irq, dma_private);
if (dma_private->ld_buf_phys) {
- dma_unmap_single(substream->pcm->dev,
+ dma_unmap_single(substream->pcm->card->dev,
dma_private->ld_buf_phys,
sizeof(dma_private->link), DMA_TO_DEVICE);
}
/* Deallocate the fsl_dma_private structure */
- dma_free_coherent(substream->pcm->dev,
+ dma_free_coherent(substream->pcm->card->dev,
sizeof(struct fsl_dma_private),
dma_private, dma_private->ld_buf_phys);
substream->runtime->private_data = NULL;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 169bca295b7..3711d8454d9 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -60,6 +60,13 @@
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
#endif
+/* SIER bitflag of interrupts to enable */
+#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \
+ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \
+ CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \
+ CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
+ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
+
/**
* fsl_ssi_private: per-SSI private data
*
@@ -140,7 +147,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
were interrupted for. We mask it with the Interrupt Enable register
so that we only check for events that we're interested in.
*/
- sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier);
+ sisr = in_be32(&ssi->sisr) & SIER_FLAGS;
if (sisr & CCSR_SSI_SISR_RFRC) {
ssi_private->stats.rfrc++;
@@ -324,12 +331,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
*/
/* 4. Enable the interrupts and DMA requests */
- out_be32(&ssi->sier,
- CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE |
- CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN |
- CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN |
- CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE |
- CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN);
+ out_be32(&ssi->sier, SIER_FLAGS);
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
@@ -466,28 +468,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_START:
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
- } else {
- long timeout = jiffies + 10;
-
+ else
setbits32(&ssi->scr,
CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
-
- /* Wait until the SSI has filled its FIFO. Without this
- * delay, ALSA complains about overruns. When the FIFO
- * is full, the DMA controller initiates its first
- * transfer. Until then, however, the DMA's DAR
- * register is zero, which translates to an
- * out-of-bounds pointer. This makes ALSA think an
- * overrun has occurred.
- */
- while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
- (jiffies < timeout));
- if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
- return -EIO;
- }
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -606,39 +592,52 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
.ops = &fsl_ssi_dai_ops,
};
+/* Show the statistics of a flag only if its interrupt is enabled. The
+ * compiler will optimze this code to a no-op if the interrupt is not
+ * enabled.
+ */
+#define SIER_SHOW(flag, name) \
+ do { \
+ if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \
+ length += sprintf(buf + length, #name "=%u\n", \
+ ssi_private->stats.name); \
+ } while (0)
+
+
/**
* fsl_sysfs_ssi_show: display SSI statistics
*
- * Display the statistics for the current SSI device.
+ * Display the statistics for the current SSI device. To avoid confusion,
+ * we only show those counts that are enabled.
*/
static ssize_t fsl_sysfs_ssi_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct fsl_ssi_private *ssi_private =
- container_of(attr, struct fsl_ssi_private, dev_attr);
- ssize_t length;
-
- length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc);
- length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc);
- length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau);
- length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu);
- length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt);
- length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1);
- length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0);
- length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1);
- length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0);
- length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1);
- length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0);
- length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1);
- length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0);
- length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs);
- length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs);
- length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls);
- length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls);
- length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1);
- length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0);
- length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1);
- length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0);
+ container_of(attr, struct fsl_ssi_private, dev_attr);
+ ssize_t length = 0;
+
+ SIER_SHOW(RFRC_EN, rfrc);
+ SIER_SHOW(TFRC_EN, tfrc);
+ SIER_SHOW(CMDAU_EN, cmdau);
+ SIER_SHOW(CMDDU_EN, cmddu);
+ SIER_SHOW(RXT_EN, rxt);
+ SIER_SHOW(RDR1_EN, rdr1);
+ SIER_SHOW(RDR0_EN, rdr0);
+ SIER_SHOW(TDE1_EN, tde1);
+ SIER_SHOW(TDE0_EN, tde0);
+ SIER_SHOW(ROE1_EN, roe1);
+ SIER_SHOW(ROE0_EN, roe0);
+ SIER_SHOW(TUE1_EN, tue1);
+ SIER_SHOW(TUE0_EN, tue0);
+ SIER_SHOW(TFS_EN, tfs);
+ SIER_SHOW(RFS_EN, rfs);
+ SIER_SHOW(TLS_EN, tls);
+ SIER_SHOW(RLS_EN, rls);
+ SIER_SHOW(RFF1_EN, rff1);
+ SIER_SHOW(RFF0_EN, rff0);
+ SIER_SHOW(TFE1_EN, tfe1);
+ SIER_SHOW(TFE0_EN, tfe0);
return length;
}
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d6882be3345..9c09b94f0cf 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -146,6 +146,17 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
int err = 0;
+ if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
+ /*
+ * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
+ * Set constraint for minimum buffer size to the same than FIFO
+ * size in order to avoid underruns in playback startup because
+ * HW is keeping the DMA request active until FIFO is filled.
+ */
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
+ }
+
if (!cpu_dai->active)
err = omap_mcbsp_request(mcbsp_data->bus_id);
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 5998ab366e8..ad8a10fe629 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE
Say Y if you want to add support for SoC audio on the
Marvell Zylonite reference platform.
+config SND_PXA2XX_SOC_MAGICIAN
+ tristate "SoC Audio support for HTC Magician"
+ depends on SND_PXA2XX_SOC && MACH_MAGICIAN
+ select SND_PXA2XX_SOC_I2S
+ select SND_PXA_SOC_SSP
+ select SND_SOC_UDA1380
+ help
+ Say Y if you want to add support for SoC audio on the
+ HTC Magician.
+
config SND_PXA2XX_SOC_MIOA701
tristate "SoC Audio support for MIO A701"
depends on SND_PXA2XX_SOC && MACH_MIOA701
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 8ed881c5e5c..4b90c3ccae4 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
snd-soc-palm27x-objs := palm27x.o
snd-soc-zylonite-objs := zylonite.o
+snd-soc-magician-objs := magician.o
snd-soc-mioa701-objs := mioa701_wm9713.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
@@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
new file mode 100644
index 00000000000..f7c4544f785
--- /dev/null
+++ b/sound/soc/pxa/magician.c
@@ -0,0 +1,560 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC 0
+#define MAGICIAN_MIC_EXT 1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_codec *codec)
+{
+ if (magician_spk_switch)
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ if (magician_hp_switch)
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Call Mic");
+ break;
+ case MAGICIAN_MIC_EXT:
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ break;
+ }
+
+ snd_soc_dapm_sync(codec);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->card->codec;
+
+ /* check the jack status at stream startup */
+ magician_ext_control(codec);
+
+ return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int acps, acds, width, rate;
+ unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+ int ret = 0;
+
+ rate = params_rate(params);
+ width = snd_pcm_format_physical_width(params_format(params));
+
+ /*
+ * rate = SSPSCLK / (2 * width(16 or 32))
+ * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+ */
+ switch (params_rate(params)) {
+ case 8000:
+ /* off by a factor of 2: bug in the PXA27x audio clock? */
+ acps = 32842000;
+ switch (width) {
+ case 16:
+ /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_16;
+ break;
+ case 32:
+ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_8;
+ }
+ break;
+ case 11025:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_4;
+ break;
+ case 32:
+ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ }
+ break;
+ case 22050:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 44100:
+ acps = 5622000;
+ switch (width) {
+ case 16:
+ /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 48000:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ break;
+ case 32:
+ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ }
+ break;
+ case 96000:
+ acps = 12235000;
+ switch (width) {
+ case 16:
+ /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_1;
+ break;
+ case 32:
+ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+ acds = PXA_SSP_CLK_AUDIO_DIV_2;
+ div4 = PXA_SSP_CLK_SCDB_1;
+ break;
+ }
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+ if (ret < 0)
+ return ret;
+
+ /* set audio clock as clock source */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock ACDS divider */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_ACDS, acds);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock SCDB divider4 */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai,
+ PXA_SSP_AUDIO_DIV_SCDB, div4);
+ if (ret < 0)
+ return ret;
+
+ /* set SSP audio pll clock */
+ ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as output */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_hp_switch;
+ return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_hp_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_hp_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(codec);
+ return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_spk_switch;
+ return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_spk_switch == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_spk_switch = ucontrol->value.integer.value[0];
+ magician_ext_control(codec);
+ return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_in_sel;
+ return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (magician_in_sel == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_in_sel = ucontrol->value.integer.value[0];
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+ break;
+ case MAGICIAN_MIC_EXT:
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+ }
+
+ return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+ SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+ SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+ SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* Headphone connected to VOUTL, VOUTR */
+ {"Headphone Jack", NULL, "VOUTL"},
+ {"Headphone Jack", NULL, "VOUTR"},
+
+ /* Speaker connected to VOUTL, VOUTR */
+ {"Speaker", NULL, "VOUTL"},
+ {"Speaker", NULL, "VOUTR"},
+
+ /* Mics are connected to VINM */
+ {"VINM", NULL, "Headset Mic"},
+ {"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+ SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+ SOC_SINGLE_BOOL_EXT("Headphone Switch",
+ (unsigned long)&magician_hp_switch,
+ magician_get_hp, magician_set_hp),
+ SOC_SINGLE_BOOL_EXT("Speaker Switch",
+ (unsigned long)&magician_spk_switch,
+ magician_get_spk, magician_set_spk),
+ SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+ magician_get_input, magician_set_input),
+};
+
+/*
+ * Logic for a uda1380 as connected on a HTC Magician
+ */
+static int magician_uda1380_init(struct snd_soc_codec *codec)
+{
+ int err;
+
+ /* NC codec pins */
+ snd_soc_dapm_nc_pin(codec, "VOUTLHP");
+ snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+
+ /* FIXME: is anything connected here? */
+ snd_soc_dapm_nc_pin(codec, "VINL");
+ snd_soc_dapm_nc_pin(codec, "VINR");
+
+ /* Add magician specific controls */
+ err = snd_soc_add_controls(codec, uda1380_magician_controls,
+ ARRAY_SIZE(uda1380_magician_controls));
+ if (err < 0)
+ return err;
+
+ /* Add magician specific widgets */
+ snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+
+ /* Set up magician specific audio path interconnects */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Playback",
+ .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
+ .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
+ .init = magician_uda1380_init,
+ .ops = &magician_playback_ops,
+},
+{
+ .name = "uda1380",
+ .stream_name = "UDA1380 Capture",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
+ .ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+ .name = "Magician",
+ .dai_link = magician_dai,
+ .num_links = ARRAY_SIZE(magician_dai),
+ .platform = &pxa2xx_soc_platform,
+};
+
+/* magician audio private data */
+static struct uda1380_setup_data magician_uda1380_setup = {
+ .i2c_address = 0x18,
+ .dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+/* magician audio subsystem */
+static struct snd_soc_device magician_snd_devdata = {
+ .card = &snd_soc_card_magician,
+ .codec_dev = &soc_codec_dev_uda1380,
+ .codec_data = &magician_uda1380_setup,
+};
+
+static struct platform_device *magician_snd_device;
+
+static int __init magician_init(void)
+{
+ int ret;
+
+ if (!machine_is_magician())
+ return -ENODEV;
+
+ ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
+ if (ret)
+ goto err_request_power;
+ ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
+ if (ret)
+ goto err_request_reset;
+ ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+ if (ret)
+ goto err_request_spk;
+ ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+ if (ret)
+ goto err_request_ep;
+ ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+ if (ret)
+ goto err_request_mic;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+ if (ret)
+ goto err_request_in_sel0;
+ ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+ if (ret)
+ goto err_request_in_sel1;
+
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
+ gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+ /* we may need to have the clock running here - pH5 */
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
+ udelay(5);
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
+
+ magician_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!magician_snd_device) {
+ ret = -ENOMEM;
+ goto err_pdev;
+ }
+
+ platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
+ magician_snd_devdata.dev = &magician_snd_device->dev;
+ ret = platform_device_add(magician_snd_device);
+ if (ret) {
+ platform_device_put(magician_snd_device);
+ goto err_pdev;
+ }
+
+ return 0;
+
+err_pdev:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+ gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+err_request_reset:
+ gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+err_request_power:
+ return ret;
+}
+
+static void __exit magician_exit(void)
+{
+ platform_device_unregister(magician_snd_device);
+
+ gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+ gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
+
+ gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+ gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+ gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+ gpio_free(EGPIO_MAGICIAN_EP_POWER);
+ gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+ gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+ gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 7acd3febf8b..308a657928d 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -627,12 +627,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
u32 sscr0;
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
+ int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
/* select correct DMA params */
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
dma = 1; /* capture DMA offset is 1,3 */
- if (chn == 2)
- dma += 2; /* stereo DMA offset is 2, mono is 0 */
+ /* Network mode with one active slot (ttsa == 1) can be used
+ * to force 16-bit frame width on the wire (for S16_LE), even
+ * with two channels. Use 16-bit DMA transfers for this case.
+ */
+ if (((chn == 2) && (ttsa != 1)) || (width == 32))
+ dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
+
cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
@@ -712,7 +718,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
/* When we use a network mode, we always require TDM slots
* - complain loudly and fail if they've not been set up yet.
*/
- if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+ if ((sscr0 & SSCR0_MOD) && !ttsa) {
dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
return -EINVAL;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6e710f705a7..99712f652d0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
int err;
codec->ac97->dev.bus = &ac97_bus_type;
- codec->ac97->dev.parent = NULL;
+ codec->ac97->dev.parent = codec->card->dev;
codec->ac97->dev.release = soc_ac97_device_release;
dev_set_name(&codec->ac97->dev, "%d-%d:%s",
@@ -767,11 +767,21 @@ static int soc_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
+ struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
- dev_dbg(socdev->dev, "scheduling resume work\n");
-
- if (!schedule_work(&card->deferred_resume_work))
- dev_err(socdev->dev, "resume work item may be lost\n");
+ /* AC97 devices might have other drivers hanging off them so
+ * need to resume immediately. Other drivers don't have that
+ * problem and may take a substantial amount of time to resume
+ * due to I/O costs and anti-pop so handle them out of line.
+ */
+ if (cpu_dai->ac97_control) {
+ dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
+ soc_resume_deferred(&card->deferred_resume_work);
+ } else {
+ dev_dbg(socdev->dev, "Scheduling resume work\n");
+ if (!schedule_work(&card->deferred_resume_work))
+ dev_err(socdev->dev, "resume work item may be lost\n");
+ }
return 0;
}
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c2db0f95968..823296d7d57 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -121,6 +121,7 @@ struct audioformat {
unsigned char attributes; /* corresponding attributes of cs endpoint */
unsigned char endpoint; /* endpoint */
unsigned char ep_attr; /* endpoint attributes */
+ unsigned char datainterval; /* log_2 of data packet interval */
unsigned int maxpacksize; /* max. packet size */
unsigned int rates; /* rate bitmasks */
unsigned int rate_min, rate_max; /* min/max rates */
@@ -170,7 +171,6 @@ struct snd_usb_substream {
unsigned int curframesize; /* current packet size in frames (for capture) */
unsigned int fill_max: 1; /* fill max packet size always */
unsigned int fmt_type; /* USB audio format type (1-3) */
- unsigned int packs_per_ms; /* packets per millisecond (for playback) */
unsigned int running: 1; /* running status */
@@ -607,9 +607,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs,
break;
}
}
- /* finish at the frame boundary at/after the period boundary */
- if (period_elapsed &&
- (i & (subs->packs_per_ms - 1)) == subs->packs_per_ms - 1)
+ if (period_elapsed) /* finish at the period boundary */
break;
}
if (subs->hwptr_done + offs > runtime->buffer_size) {
@@ -1067,7 +1065,6 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
packs_per_ms = 8 >> subs->datainterval;
else
packs_per_ms = 1;
- subs->packs_per_ms = packs_per_ms;
if (is_playback) {
urb_packs = max(nrpacks, 1);
@@ -1087,18 +1084,17 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
minsize -= minsize >> 3;
minsize = max(minsize, 1u);
total_packs = (period_bytes + minsize - 1) / minsize;
- /* round up to multiple of packs_per_ms */
- total_packs = (total_packs + packs_per_ms - 1)
- & ~(packs_per_ms - 1);
/* we need at least two URBs for queueing */
- if (total_packs < 2 * packs_per_ms) {
- total_packs = 2 * packs_per_ms;
+ if (total_packs < 2) {
+ total_packs = 2;
} else {
/* and we don't want too long a queue either */
maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
total_packs = min(total_packs, maxpacks);
}
} else {
+ while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+ urb_packs >>= 1;
total_packs = MAX_URBS * urb_packs;
}
subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
@@ -1350,12 +1346,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
subs->datapipe = usb_sndisocpipe(dev, ep);
else
subs->datapipe = usb_rcvisocpipe(dev, ep);
- if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH &&
- get_endpoint(alts, 0)->bInterval >= 1 &&
- get_endpoint(alts, 0)->bInterval <= 4)
- subs->datainterval = get_endpoint(alts, 0)->bInterval - 1;
- else
- subs->datainterval = 0;
+ subs->datainterval = fmt->datainterval;
subs->syncpipe = subs->syncinterval = 0;
subs->maxpacksize = fmt->maxpacksize;
subs->fill_max = 0;
@@ -1568,11 +1559,15 @@ static struct snd_pcm_hardware snd_usb_hardware =
#define hwc_debug(fmt, args...) /**/
#endif
-static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audioformat *fp)
+static int hw_check_valid_format(struct snd_usb_substream *subs,
+ struct snd_pcm_hw_params *params,
+ struct audioformat *fp)
{
struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME);
+ unsigned int ptime;
/* check the format */
if (!snd_mask_test(fmts, fp->format)) {
@@ -1593,6 +1588,14 @@ static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audiof
hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min);
return 0;
}
+ /* check whether the period time is >= the data packet interval */
+ if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) {
+ ptime = 125 * (1 << fp->datainterval);
+ if (ptime > pt->max || (ptime == pt->max && pt->openmax)) {
+ hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max);
+ return 0;
+ }
+ }
return 1;
}
@@ -1611,7 +1614,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params,
list_for_each(p, &subs->fmt_list) {
struct audioformat *fp;
fp = list_entry(p, struct audioformat, list);
- if (!hw_check_valid_format(params, fp))
+ if (!hw_check_valid_format(subs, params, fp))
continue;
if (changed++) {
if (rmin > fp->rate_min)
@@ -1665,7 +1668,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params,
list_for_each(p, &subs->fmt_list) {
struct audioformat *fp;
fp = list_entry(p, struct audioformat, list);
- if (!hw_check_valid_format(params, fp))
+ if (!hw_check_valid_format(subs, params, fp))
continue;
if (changed++) {
if (rmin > fp->channels)
@@ -1718,7 +1721,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params,
list_for_each(p, &subs->fmt_list) {
struct audioformat *fp;
fp = list_entry(p, struct audioformat, list);
- if (!hw_check_valid_format(params, fp))
+ if (!hw_check_valid_format(subs, params, fp))
continue;
fbits |= (1ULL << fp->format);
}
@@ -1736,95 +1739,42 @@ static int hw_rule_format(struct snd_pcm_hw_params *params,
return changed;
}
-#define MAX_MASK 64
-
-/*
- * check whether the registered audio formats need special hw-constraints
- */
-static int check_hw_params_convention(struct snd_usb_substream *subs)
+static int hw_rule_period_time(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
{
- int i;
- u32 *channels;
- u32 *rates;
- u32 cmaster, rmaster;
- u32 rate_min = 0, rate_max = 0;
- struct list_head *p;
- int err = 1;
-
- channels = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL);
- rates = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL);
- if (!channels || !rates) {
- err = -ENOMEM;
- goto __out;
- }
+ struct snd_usb_substream *subs = rule->private;
+ struct audioformat *fp;
+ struct snd_interval *it;
+ unsigned char min_datainterval;
+ unsigned int pmin;
+ int changed;
- list_for_each(p, &subs->fmt_list) {
- struct audioformat *f;
- f = list_entry(p, struct audioformat, list);
- /* unconventional channels? */
- if (f->channels > 32)
- goto __out;
- /* continuous rate min/max matches? */
- if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) {
- if (rate_min && f->rate_min != rate_min)
- goto __out;
- if (rate_max && f->rate_max != rate_max)
- goto __out;
- rate_min = f->rate_min;
- rate_max = f->rate_max;
- }
- /* combination of continuous rates and fixed rates? */
- if (rates[f->format] & SNDRV_PCM_RATE_CONTINUOUS) {
- if (f->rates != rates[f->format])
- goto __out;
- }
- if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) {
- if (rates[f->format] && rates[f->format] != f->rates)
- goto __out;
- }
- channels[f->format] |= 1 << (f->channels - 1);
- rates[f->format] |= f->rates;
- /* needs knot? */
- if (f->rates & SNDRV_PCM_RATE_KNOT)
- goto __out;
- }
- /* check whether channels and rates match for all formats */
- cmaster = rmaster = 0;
- for (i = 0; i < MAX_MASK; i++) {
- if (cmaster != channels[i] && cmaster && channels[i])
- goto __out;
- if (rmaster != rates[i] && rmaster && rates[i])
- goto __out;
- if (channels[i])
- cmaster = channels[i];
- if (rates[i])
- rmaster = rates[i];
- }
- /* check whether channels match for all distinct rates */
- memset(channels, 0, MAX_MASK * sizeof(u32));
- list_for_each(p, &subs->fmt_list) {
- struct audioformat *f;
- f = list_entry(p, struct audioformat, list);
- if (f->rates & SNDRV_PCM_RATE_CONTINUOUS)
+ it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME);
+ hwc_debug("hw_rule_period_time: (%u,%u)\n", it->min, it->max);
+ min_datainterval = 0xff;
+ list_for_each_entry(fp, &subs->fmt_list, list) {
+ if (!hw_check_valid_format(subs, params, fp))
continue;
- for (i = 0; i < 32; i++) {
- if (f->rates & (1 << i))
- channels[i] |= 1 << (f->channels - 1);
- }
+ min_datainterval = min(min_datainterval, fp->datainterval);
}
- cmaster = 0;
- for (i = 0; i < 32; i++) {
- if (cmaster != channels[i] && cmaster && channels[i])
- goto __out;
- if (channels[i])
- cmaster = channels[i];
+ if (min_datainterval == 0xff) {
+ hwc_debug(" --> get emtpy\n");
+ it->empty = 1;
+ return -EINVAL;
}
- err = 0;
-
- __out:
- kfree(channels);
- kfree(rates);
- return err;
+ pmin = 125 * (1 << min_datainterval);
+ changed = 0;
+ if (it->min < pmin) {
+ it->min = pmin;
+ it->openmin = 0;
+ changed = 1;
+ }
+ if (snd_interval_checkempty(it)) {
+ it->empty = 1;
+ return -EINVAL;
+ }
+ hwc_debug(" --> (%u,%u) (changed = %d)\n", it->min, it->max, changed);
+ return changed;
}
/*
@@ -1872,6 +1822,8 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime,
static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs)
{
struct list_head *p;
+ unsigned int pt, ptmin;
+ int param_period_time_if_needed;
int err;
runtime->hw.formats = subs->formats;
@@ -1881,6 +1833,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
runtime->hw.channels_min = 256;
runtime->hw.channels_max = 0;
runtime->hw.rates = 0;
+ ptmin = UINT_MAX;
/* check min/max rates and channels */
list_for_each(p, &subs->fmt_list) {
struct audioformat *fp;
@@ -1899,42 +1852,54 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
runtime->hw.period_bytes_min = runtime->hw.period_bytes_max =
fp->frame_size;
}
+ pt = 125 * (1 << fp->datainterval);
+ ptmin = min(ptmin, pt);
}
- /* set the period time minimum 1ms */
- /* FIXME: high-speed mode allows 125us minimum period, but many parts
- * in the current code assume the 1ms period.
- */
+ param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME;
+ if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH)
+ /* full speed devices have fixed data packet interval */
+ ptmin = 1000;
+ if (ptmin == 1000)
+ /* if period time doesn't go below 1 ms, no rules needed */
+ param_period_time_if_needed = -1;
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME,
- 1000,
- /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX);
-
- err = check_hw_params_convention(subs);
- if (err < 0)
+ ptmin, UINT_MAX);
+
+ if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_rate, subs,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ param_period_time_if_needed,
+ -1)) < 0)
return err;
- else if (err) {
- hwc_debug("setting extra hw constraints...\n");
- if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
- hw_rule_rate, subs,
- SNDRV_PCM_HW_PARAM_FORMAT,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- -1)) < 0)
- return err;
- if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
- hw_rule_channels, subs,
- SNDRV_PCM_HW_PARAM_FORMAT,
- SNDRV_PCM_HW_PARAM_RATE,
- -1)) < 0)
- return err;
- if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
- hw_rule_format, subs,
- SNDRV_PCM_HW_PARAM_RATE,
- SNDRV_PCM_HW_PARAM_CHANNELS,
- -1)) < 0)
- return err;
- if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0)
+ if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_channels, subs,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ SNDRV_PCM_HW_PARAM_RATE,
+ param_period_time_if_needed,
+ -1)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_format, subs,
+ SNDRV_PCM_HW_PARAM_RATE,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ param_period_time_if_needed,
+ -1)) < 0)
+ return err;
+ if (param_period_time_if_needed >= 0) {
+ err = snd_pcm_hw_rule_add(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+ hw_rule_period_time, subs,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ SNDRV_PCM_HW_PARAM_RATE,
+ -1);
+ if (err < 0)
return err;
}
+ if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0)
+ return err;
return 0;
}
@@ -2147,7 +2112,8 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
fp = list_entry(p, struct audioformat, list);
snd_iprintf(buffer, " Interface %d\n", fp->iface);
snd_iprintf(buffer, " Altset %d\n", fp->altsetting);
- snd_iprintf(buffer, " Format: %#x\n", fp->format);
+ snd_iprintf(buffer, " Format: %#x (%d bits)\n",
+ fp->format, snd_pcm_format_width(fp->format));
snd_iprintf(buffer, " Channels: %d\n", fp->channels);
snd_iprintf(buffer, " Endpoint: %d %s (%s)\n",
fp->endpoint & USB_ENDPOINT_NUMBER_MASK,
@@ -2166,6 +2132,9 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
}
snd_iprintf(buffer, "\n");
}
+ if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH)
+ snd_iprintf(buffer, " Data packet interval: %d us\n",
+ 125 * (1 << fp->datainterval));
// snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize);
// snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes);
}
@@ -2659,6 +2628,17 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp
return 0;
}
+static unsigned char parse_datainterval(struct snd_usb_audio *chip,
+ struct usb_host_interface *alts)
+{
+ if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH &&
+ get_endpoint(alts, 0)->bInterval >= 1 &&
+ get_endpoint(alts, 0)->bInterval <= 4)
+ return get_endpoint(alts, 0)->bInterval - 1;
+ else
+ return 0;
+}
+
static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
int iface, int altno);
static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
@@ -2764,6 +2744,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
fp->altset_idx = i;
fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->datainterval = parse_datainterval(chip, alts);
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
@@ -2955,6 +2936,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
return -EINVAL;
}
alts = &iface->altsetting[fp->altset_idx];
+ fp->datainterval = parse_datainterval(chip, alts);
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
usb_set_interface(chip->dev, fp->iface, 0);
init_usb_pitch(chip->dev, fp->iface, alts, fp);
@@ -3049,6 +3031,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
fp->iface = altsd->bInterfaceNumber;
fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->datainterval = 0;
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
switch (fp->maxpacksize) {
@@ -3116,6 +3099,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip,
fp->iface = altsd->bInterfaceNumber;
fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->datainterval = parse_datainterval(chip, alts);
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
fp->rate_max = fp->rate_min = combine_triple(&alts->extra[8]);
@@ -3168,6 +3152,7 @@ static int create_ua101_quirk(struct snd_usb_audio *chip,
fp->iface = altsd->bInterfaceNumber;
fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->datainterval = parse_datainterval(chip, alts);
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]);
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-26 01:27:16 +0000