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-rw-r--r--sound/soc/blackfin/Kconfig98
-rw-r--r--sound/soc/blackfin/bf5xx-i2s.c8
-rw-r--r--sound/soc/blackfin/bf5xx-tdm.c8
-rw-r--r--sound/soc/codecs/tlv320aic23.c5
-rw-r--r--sound/soc/codecs/wm8350.c4
-rw-r--r--sound/soc/codecs/wm8940.c2
-rw-r--r--sound/soc/davinci/davinci-i2s.c37
-rw-r--r--sound/soc/davinci/davinci-mcasp.c80
-rw-r--r--sound/soc/davinci/davinci-mcasp.h7
-rw-r--r--sound/soc/davinci/davinci-pcm.c13
-rw-r--r--sound/soc/davinci/davinci-pcm.h1
-rw-r--r--sound/soc/imx/mxc-ssi.c8
-rw-r--r--sound/soc/omap/Kconfig13
-rw-r--r--sound/soc/omap/omap-pcm.c8
-rw-r--r--sound/soc/omap/omap3evm.c2
-rw-r--r--sound/soc/omap/omap3pandora.c3
-rw-r--r--sound/soc/pxa/Kconfig2
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c17
-rw-r--r--sound/soc/s3c24xx/s3c64xx-i2s.c2
-rw-r--r--sound/soc/soc-core.c11
-rw-r--r--sound/soc/soc-dapm.c27
21 files changed, 182 insertions, 174 deletions
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ac927ff..97f1a25 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -7,15 +7,6 @@ config SND_BF5XX_I2S
mode (supports single stereo In/Out).
You will also need to select the audio interfaces to support below.
-config SND_BF5XX_TDM
- tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
- depends on (BLACKFIN && SND_SOC)
- help
- Say Y or M if you want to add support for codecs attached to
- the Blackfin SPORT (synchronous serial ports) interface in TDM
- mode.
- You will also need to select the audio interfaces to support below.
-
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio support for BF52x ezkit"
depends on SND_BF5XX_I2S
@@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE
Enter the GPIO used to control AD73311's SE pin. Acceptable
values are 0 to 7
+config SND_BF5XX_TDM
+ tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
+ depends on (BLACKFIN && SND_SOC)
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the Blackfin SPORT (synchronous serial ports) interface in TDM
+ mode.
+ You will also need to select the audio interfaces to support below.
+
+config SND_BF5XX_SOC_AD1836
+ tristate "SoC AD1836 Audio support for BF5xx"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1836
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
+config SND_BF5XX_SOC_AD1938
+ tristate "SoC AD1938 Audio support for Blackfin"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1938
+ help
+ Say Y if you want to add support for AD1938 codec on Blackfin.
+
config SND_BF5XX_AC97
tristate "SoC AC97 Audio for the ADI BF5xx chip"
depends on BLACKFIN
@@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT
Say y if you want AC97 driver to support up to 5.1 channel audio.
this mode will consume much more memory for DMA.
+config SND_BF5XX_HAVE_COLD_RESET
+ bool "BOARD has COLD Reset GPIO"
+ depends on SND_BF5XX_AC97
+ default y if BFIN548_EZKIT
+ default n if !BFIN548_EZKIT
+
+config SND_BF5XX_RESET_GPIO_NUM
+ int "Set a GPIO for cold reset"
+ depends on SND_BF5XX_HAVE_COLD_RESET
+ range 0 159
+ default 19 if BFIN548_EZKIT
+ default 5 if BFIN537_STAMP
+ default 0
+ help
+ Set the correct GPIO for RESET the sound chip.
+
+config SND_BF5XX_SOC_AD1980
+ tristate "SoC AD1980/1 Audio support for BF5xx"
+ depends on SND_BF5XX_AC97
+ select SND_BF5XX_SOC_AC97
+ select SND_SOC_AD1980
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
config SND_BF5XX_SOC_SPORT
tristate
@@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97
select SND_SOC_AC97_BUS
select SND_BF5XX_SOC_SPORT
-config SND_BF5XX_SOC_AD1836
- tristate "SoC AD1836 Audio support for BF5xx"
- depends on SND_BF5XX_TDM
- select SND_BF5XX_SOC_TDM
- select SND_SOC_AD1836
- help
- Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1980
- tristate "SoC AD1980/1 Audio support for BF5xx"
- depends on SND_BF5XX_AC97
- select SND_BF5XX_SOC_AC97
- select SND_SOC_AD1980
- help
- Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1938
- tristate "SoC AD1938 Audio support for Blackfin"
- depends on SND_BF5XX_TDM
- select SND_BF5XX_SOC_TDM
- select SND_SOC_AD1938
- help
- Say Y if you want to add support for AD1938 codec on Blackfin.
-
config SND_BF5XX_SPORT_NUM
int "Set a SPORT for Sound chip"
depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
@@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM
default 0
help
Set the correct SPORT for sound chip.
-
-config SND_BF5XX_HAVE_COLD_RESET
- bool "BOARD has COLD Reset GPIO"
- depends on SND_BF5XX_AC97
- default y if BFIN548_EZKIT
- default n if !BFIN548_EZKIT
-
-config SND_BF5XX_RESET_GPIO_NUM
- int "Set a GPIO for cold reset"
- depends on SND_BF5XX_HAVE_COLD_RESET
- range 0 159
- default 19 if BFIN548_EZKIT
- default 5 if BFIN537_STAMP
- default 0
- help
- Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 1e9d161..084b688 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = {
* TFS. When Port G is selected and EMAC then there is a conflict between
* the PHY interrupt line and TFS. Current settings prevent the conflict
* by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
*/
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
#define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
#endif
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 3096bad..ff546e9 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = {
* TFS. When Port G is selected and EMAC then there is a conflict between
* the PHY interrupt line and TFS. Current settings prevent the conflict
* by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
*/
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
#define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
#endif
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 0b8dcb5..90a0264 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = {
#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
static const unsigned short sr_valid_mask[] = {
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
- LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
LOWER_GROUP, /* Usb, bosr - 0*/
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
UPPER_GROUP, /* Usb, bosr - 1*/
};
/*
@@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
- int i;
u16 reg;
/* Sync reg_cache with the hardware */
- for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+ for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3ff0373..593d5b9 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
- WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+ WM8350_LEFT_INPUT_VOLUME, 14, 1, 1),
};
/* Right Input Mixer */
@@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
- WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+ WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1),
};
/* Left Mic Mixer */
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index da97aae..1ef2454 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940,
codec->reg_cache = &wm8940->reg_cache;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
- if (ret == 0) {
+ if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 12a6c54..4ae7070 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -97,22 +97,19 @@ enum {
DAVINCI_MCBSP_WORD_32,
};
-static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
- .name = "I2S PCM Stereo out",
-};
-
-static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
- .name = "I2S PCM Stereo in",
-};
-
struct davinci_mcbsp_dev {
+ /*
+ * dma_params must be first because rtd->dai->cpu_dai->private_data
+ * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+ * davinci_pcm_open.
+ */
+ struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
#define MOD_DSP_A 0
#define MOD_DSP_B 1
int mode;
u32 pcr;
struct clk *clk;
- struct davinci_pcm_dma_params *dma_params[2];
};
static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
toggle_clock(dev, playback);
}
-static int davinci_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
- cpu_dai->dma_data = dev->dma_params[substream->stream];
- return 0;
-}
-
#define DEFAULT_BITPERSAMPLE 16
static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct davinci_pcm_dma_params *dma_params = dai->dma_data;
struct davinci_mcbsp_dev *dev = dai->private_data;
+ struct davinci_pcm_dma_params *dma_params =
+ &dev->dma_params[substream->stream];
struct snd_interval *i = NULL;
int mcbsp_word_length;
unsigned int rcr, xcr, srgr;
@@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
- .startup = davinci_i2s_startup,
.shutdown = davinci_i2s_shutdown,
.prepare = davinci_i2s_prepare,
.trigger = davinci_i2s_trigger,
@@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev)
dev->base = (void __iomem *)IO_ADDRESS(mem->start);
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
/* first TX, then RX */
@@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = -ENXIO;
goto err_free_mem;
}
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
@@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = -ENXIO;
goto err_free_mem;
}
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
davinci_i2s_dai.private_data = dev;
ret = snd_soc_register_dai(&davinci_i2s_dai);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7a06c0a..5d1f98a 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
printk(KERN_ERR "GBLCTL write error\n");
}
-static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct davinci_audio_dev *dev = cpu_dai->private_data;
- cpu_dai->dma_data = dev->dma_params[substream->stream];
- return 0;
-}
-
static void mcasp_start_rx(struct davinci_audio_dev *dev)
{
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
@@ -386,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (dev->txnumevt) /* enable FIFO */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_start_tx(dev);
- else
+ } else {
+ if (dev->rxnumevt) /* enable FIFO */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_start_rx(dev);
-
- /* enable FIFO */
- if (dev->txnumevt)
- mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
- if (dev->rxnumevt)
- mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+ }
}
static void mcasp_stop_rx(struct davinci_audio_dev *dev)
@@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev)
static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (dev->txnumevt) /* disable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_stop_tx(dev);
- else
+ } else {
+ if (dev->rxnumevt) /* disable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_stop_rx(dev);
-
- /* disable FIFO */
- if (dev->txnumevt)
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
- if (dev->rxnumevt)
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+ }
}
static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -720,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
{
struct davinci_audio_dev *dev = cpu_dai->private_data;
struct davinci_pcm_dma_params *dma_params =
- dev->dma_params[substream->stream];
+ &dev->dma_params[substream->stream];
int word_length;
u8 numevt;
@@ -798,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
}
static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
- .startup = davinci_mcasp_startup,
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
@@ -849,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
struct resource *mem, *ioarea, *res;
struct snd_platform_data *pdata;
struct davinci_audio_dev *dev;
- int count = 0;
int ret = 0;
dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
if (!dev)
return -ENOMEM;
- dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
- GFP_KERNEL);
- if (!dma_data) {
- ret = -ENOMEM;
- goto err_release_dev;
- }
-
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!mem) {
dev_err(&pdev->dev, "no mem resource?\n");
@@ -897,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dev->txnumevt = pdata->txnumevt;
dev->rxnumevt = pdata->rxnumevt;
- dma_data[count].name = "I2S PCM Stereo out";
- dma_data[count].eventq_no = pdata->eventq_no;
- dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
+ dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_data->eventq_no = pdata->eventq_no;
+ dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
io_v2p(dev->base));
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
/* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -910,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err_release_region;
}
- dma_data[count].channel = res->start;
- count++;
- dma_data[count].name = "I2S PCM Stereo in";
- dma_data[count].eventq_no = pdata->eventq_no;
- dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
+ dma_data->channel = res->start;
+
+ dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+ dma_data->eventq_no = pdata->eventq_no;
+ dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
io_v2p(dev->base));
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
@@ -924,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err_release_region;
}
- dma_data[count].channel = res->start;
+ dma_data->channel = res->start;
davinci_mcasp_dai[pdata->op_mode].private_data = dev;
davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
@@ -936,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
err_release_region:
release_mem_region(mem->start, (mem->end - mem->start) + 1);
err_release_data:
- kfree(dma_data);
-err_release_dev:
kfree(dev);
return ret;
@@ -946,7 +925,6 @@ err_release_dev:
static int davinci_mcasp_remove(struct platform_device *pdev)
{
struct snd_platform_data *pdata = pdev->dev.platform_data;
- struct davinci_pcm_dma_params *dma_data;
struct davinci_audio_dev *dev;
struct resource *mem;
@@ -959,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
release_mem_region(mem->start, (mem->end - mem->start) + 1);
- dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
- kfree(dma_data);
kfree(dev);
return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 554354c..9d179cc 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -39,10 +39,15 @@ enum {
};
struct davinci_audio_dev {
+ /*
+ * dma_params must be first because rtd->dai->cpu_dai->private_data
+ * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+ * davinci_pcm_open.
+ */
+ struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
int sample_rate;
struct clk *clk;
- struct davinci_pcm_dma_params *dma_params[2];
unsigned int codec_fmt;
/* McASP specific data */
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 2f7da49..c73a915 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
struct edmacc_param p_ram;
int ret;
- if (!dma_data)
- return -ENODEV;
-
- prtd->params = dma_data;
-
/* Request master DMA channel */
ret = edma_alloc_channel(prtd->params->channel,
davinci_pcm_dma_irq, substream,
@@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct davinci_runtime_data *prtd;
int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
+ struct davinci_pcm_dma_params *params = &pa[substream->stream];
+ if (!params)
+ return -ENODEV;
snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
/* ensure that buffer size is a multiple of period size */
@@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
return -ENOMEM;
spin_lock_init(&prtd->lock);
+ prtd->params = params;
runtime->private_data = prtd;
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 63d9625..8746606 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -17,7 +17,6 @@
struct davinci_pcm_dma_params {
- char *name; /* stream identifier */
int channel; /* sync dma channel ID */
unsigned short acnt;
dma_addr_t dma_addr; /* device physical address for DMA */
diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
index 3806ff2..ccdefe6 100644
--- a/sound/soc/imx/mxc-ssi.c
+++ b/sound/soc/imx/mxc-ssi.c
@@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai,
break;
}
- /* sync */
- if (!(fmt & SND_SOC_DAIFMT_ASYNC))
- scr |= SSI_SCR_SYN;
-
- /* tdm - only for stereo atm */
- if (fmt & SND_SOC_DAIFMT_TDM)
- scr |= SSI_SCR_NET;
-
if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
SSI1_STCR = stcr;
SSI1_SRCR = srcr;
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 2dee983..653a362 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA
select SND_OMAP_SOC_MCBSP
select SND_SOC_CX20442
help
- Say Y if you want to add support for SoC audio on Amstrad Delta.
+ Say Y if you want to add support for SoC audio device connected to
+ a handset and a speakerphone found on Amstrad E3 (Delta) videophone.
+
+ Note that in order to get those devices fully supported, you have to
+ build the kernel with standard serial port driver included and
+ configured for at least 4 ports. Then, from userspace, you must load
+ a line discipline #19 on the modem (ttyS3) serial line. The simplest
+ way to achieve this is to install util-linux-ng and use the included
+ ldattach utility. This can be started automatically from udev,
+ a simple rule like this one should do the trick (it does for me):
+ ACTION=="add", KERNEL=="controlC0", \
+ RUN+="/usr/sbin/ldattach 19 /dev/ttyS3"
config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 5735945..6a829ee 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
else
omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
- omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
- omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+ if (!(cpu_class_is_omap1())) {
+ omap_set_dma_src_burst_mode(prtd->dma_ch,
+ OMAP_DMA_DATA_BURST_16);
+ omap_set_dma_dest_burst_mode(prtd->dma_ch,
+ OMAP_DMA_DATA_BURST_16);
+ }
return 0;
}
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 9114c26..13aa380 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit);
MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index ad219aa..0cd06f5 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
* |P| <--- TWL4030 <--------- Line In and MICs
*/
static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
- SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
0, 0, NULL, 0, omap3pandora_hp_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
snd_soc_dapm_nc_pin(codec, "CARKITR");
snd_soc_dapm_nc_pin(codec, "HFL");
snd_soc_dapm_nc_pin(codec, "HFR");
+ snd_soc_dapm_nc_pin(codec, "VIBRA");
ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
ARRAY_SIZE(omap3pandora_out_dapm_widgets));
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6375b4e..dcb3181b 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701
config SND_PXA2XX_SOC_IMOTE2
tristate "SoC Audio support for IMote 2"
- depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+ depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_WM8940
help
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 5cbbdc8..1f35c6f 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -75,11 +75,19 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
{
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
dma_addr_t pos = prtd->dma_pos;
+ unsigned int limit;
int ret;
pr_debug("Entered %s\n", __func__);
- while (prtd->dma_loaded < prtd->dma_limit) {
+ if (s3c_dma_has_circular()) {
+ limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period;
+ } else
+ limit = prtd->dma_limit;
+
+ pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit);
+
+ while (prtd->dma_loaded < limit) {
unsigned long len = prtd->dma_period;
pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
@@ -123,7 +131,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
snd_pcm_period_elapsed(substream);
spin_lock(&prtd->lock);
- if (prtd->state & ST_RUNNING) {
+ if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) {
prtd->dma_loaded--;
s3c24xx_pcm_enqueue(substream);
}
@@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
printk(KERN_ERR "failed to get dma channel\n");
return ret;
}
+
+ /* use the circular buffering if we have it available. */
+ if (s3c_dma_has_circular())
+ s3c2410_dma_setflags(prtd->params->channel,
+ S3C2410_DMAF_CIRCULAR);
}
s3c2410_dma_set_buffdone_fn(prtd->params->channel,
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 3c06c40..105a77e 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -220,6 +220,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
goto err;
}
+ clk_enable(i2s->iis_cclk);
+
ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
if (ret)
goto err_clk;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7ff04ad..0a1b2f6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -834,6 +834,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume_device);
#define soc_resume NULL
#endif
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct platform_device *pdev = container_of(card->dev,
@@ -877,6 +880,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
ac97 = 1;
}
+ for (i = 0; i < card->num_links; i++) {
+ if (!card->dai_link[i].codec_dai->ops)
+ card->dai_link[i].codec_dai->ops = &null_dai_ops;
+ }
+
/* If we have AC97 in the system then don't wait for the
* codec. This will need revisiting if we have to handle
* systems with mixed AC97 and non-AC97 parts. Only check for
@@ -2329,9 +2337,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
return 0;
}
-static struct snd_soc_dai_ops null_dai_ops = {
-};
-
/**
* snd_soc_register_dai - Register a DAI with the ASoC core
*
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f79711b..66d4c16 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
/* connected jack or spk ? */
if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
- widget->id == snd_soc_dapm_line)
+ (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources)))
return 1;
}
@@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
return 1;
/* connected jack ? */
- if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+ if (widget->id == snd_soc_dapm_mic ||
+ (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks)))
return 1;
}
@@ -972,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
if (!w->power_check)
continue;
- power = w->power_check(w);
- if (power)
- sys_power = 1;
+ /* If we're suspending then pull down all the
+ * power. */
+ switch (event) {
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ power = 0;
+ break;
+
+ default:
+ power = w->power_check(w);
+ if (power)
+ sys_power = 1;
+ break;
+ }
if (w->power == power)
continue;
@@ -998,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
case SND_SOC_DAPM_STREAM_RESUME:
sys_power = 1;
break;
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ sys_power = 0;
+ break;
case SND_SOC_DAPM_STREAM_NOP:
sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY;
+ break;
default:
break;
}
@@ -2071,9 +2086,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
}
}
}
- mutex_unlock(&codec->mutex);
dapm_power_widgets(codec, event);
+ mutex_unlock(&codec->mutex);
dump_dapm(codec, __func__);
return 0;
}
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