Commit df1efe6f authored by Linus Torvalds's avatar Linus Torvalds
Browse files

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
parents 9a5467fd 11589418
......@@ -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);
......
......@@ -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,
......
......@@ -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.
......
......@@ -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,
......
......@@ -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"
......
......@@ -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.
......
Supports Markdown
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment