Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: ASoC: Export dapm_reg_event() fully
  ALSA: ASoC: Update Poodle to current ASoC API
  ALSA: asoc: restrict sample rate and size in Freescale MPC8610 sound drivers
  ALSA: sound/soc/pxa/tosa.c: removed duplicated include

include/sound/soc-dapm.h

@@ -202,6 +202,9 @@ struct snd_soc_dapm_path;
 struct snd_soc_dapm_pin;
 struct snd_soc_dapm_route;
 
+int dapm_reg_event(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol, int event);
+
 /* dapm controls */
 int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);

sound/soc/fsl/fsl_dma.c

@@ -132,12 +132,17 @@ struct fsl_dma_private {
  * Since each link descriptor has a 32-bit byte count field, we set
  * period_bytes_max to the largest 32-bit number.  We also have no maximum
  * number of periods.
+ *
+ * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
+ * limitation in the SSI driver requires the sample rates for playback and
+ * capture to be the same.
  */
 static const struct snd_pcm_hardware fsl_dma_hardware = {
 
 	.info   		= SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_MMAP |
-				  SNDRV_PCM_INFO_MMAP_VALID,
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_JOINT_DUPLEX,
 	.formats		= FSLDMA_PCM_FORMATS,
 	.rates  		= FSLDMA_PCM_RATES,
 	.rate_min       	= 5512,

sound/soc/fsl/fsl_ssi.c

@@ -67,6 +67,8 @@
  * @ssi: pointer to the SSI's registers
  * @ssi_phys: physical address of the SSI registers
  * @irq: IRQ of this SSI
+ * @first_stream: pointer to the stream that was opened first
+ * @second_stream: pointer to second stream
  * @dev: struct device pointer
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
@@ -79,6 +81,8 @@ struct fsl_ssi_private {
 	struct ccsr_ssi __iomem *ssi;
 	dma_addr_t ssi_phys;
 	unsigned int irq;
+	struct snd_pcm_substream *first_stream;
+	struct snd_pcm_substream *second_stream;
 	struct device *dev;
 	unsigned int playback;
 	unsigned int capture;
@@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
 		 */
 	}
 
+	if (!ssi_private->first_stream)
+		ssi_private->first_stream = substream;
+	else {
+		/* This is the second stream open, so we need to impose sample
+		 * rate and maybe sample size constraints.  Note that this can
+		 * cause a race condition if the second stream is opened before
+		 * the first stream is fully initialized.
+		 *
+		 * We provide some protection by checking to make sure the first
+		 * stream is initialized, but it's not perfect.  ALSA sometimes
+		 * re-initializes the driver with a different sample rate or
+		 * size.  If the second stream is opened before the first stream
+		 * has received its final parameters, then the second stream may
+		 * be constrained to the wrong sample rate or size.
+		 *
+		 * FIXME: This code does not handle opening and closing streams
+		 * repeatedly.  If you open two streams and then close the first
+		 * one, you may not be able to open another stream until you
+		 * close the second one as well.
+		 */
+		struct snd_pcm_runtime *first_runtime =
+			ssi_private->first_stream->runtime;
+
+		if (!first_runtime->rate || !first_runtime->sample_bits) {
+			dev_err(substream->pcm->card->dev,
+				"set sample rate and size in %s stream first\n",
+				substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+				? "capture" : "playback");
+			return -EAGAIN;
+		}
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE,
+			first_runtime->rate, first_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+			first_runtime->sample_bits,
+			first_runtime->sample_bits);
+
+		ssi_private->second_stream = substream;
+	}
+
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		ssi_private->playback++;
 
@@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
 
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-	u32 wl;
 
-	wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
+	if (substream == ssi_private->first_stream) {
+		u32 wl;
 
-	clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+		/* The SSI should always be disabled at this points (SSIEN=0) */
+		wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		/* In synchronous mode, the SSI uses STCCR for capture */
 		clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-	else
+	}
-		clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-
-	setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
 
 	return 0;
 }
@@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			setbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			setbits32(&ssi->scr,
+				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
 		} else {
-			setbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			setbits32(&ssi->scr,
+				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
 
 			/*
 			 * I think we need this delay to allow time for the SSI
@@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
 		ssi_private->capture--;
 
+	if (ssi_private->first_stream == substream)
+		ssi_private->first_stream = ssi_private->second_stream;
+
+	ssi_private->second_stream = NULL;
+
 	/*
 	 * If this is the last active substream, disable the SSI and release
 	 * the IRQ.

sound/soc/pxa/poodle.c

@@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on poodle */
-static int poodle_shutdown(struct snd_pcm_substream *substream)
+static void poodle_shutdown(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
-
 	/* set = unmute headphone */
 	locomo_gpio_write(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_MUTE_L, 1);
 	locomo_gpio_write(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_MUTE_R, 1);
-	return 0;
 }
 
 static int poodle_hw_params(struct snd_pcm_substream *substream,
@@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = {
 	SOC_ENUM_SINGLE_EXT(2, spk_function),
 };
 
-static const snd_kcontrol_new_t wm8731_poodle_controls[] = {
+static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
 	SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
 		poodle_set_jack),
 	SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,

sound/soc/pxa/tosa.c

@@ -33,7 +33,6 @@
 #include <asm/arch/pxa-regs.h>
 #include <asm/arch/hardware.h>
 #include <asm/arch/audio.h>
-#include <asm/arch/tosa.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"

sound/soc/soc-dapm.c

@@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(dapm_reg_event);
 
 /*
  * Scan each dapm widget for complete audio path.