Merge branch 'fix/asoc' into for-linus
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 @@
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 @@
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 @@
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 @@
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 @@
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-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 2758b90..e693229 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -277,7 +277,11 @@
if (!dai->active)
return 0;
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ ret = sport_set_multichannel(sport, 16, 0x3FF, 1);
+#else
ret = sport_set_multichannel(sport, 16, 0x1F, 1);
+#endif
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
@@ -334,7 +338,11 @@
goto sport_err;
}
/*SPORT works in TDM mode to simulate AC97 transfers*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+ ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1);
+#else
ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
+#endif
if (ret) {
pr_err("SPORT is busy!\n");
ret = -EBUSY;
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 3f2a911..a1f97dd 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -1,5 +1,5 @@
/*
- * linux/sound/arm/bf5xx-ac97.h
+ * sound/soc/blackfin/bf5xx-ac97.h
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 876abad..084b688 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -77,12 +77,12 @@
* 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,
@@ -227,7 +227,8 @@
return 0;
}
-static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
{
pr_debug("%s enter\n", __func__);
peripheral_free_list(&sport_req[sport_num][0]);
@@ -236,36 +237,31 @@
#ifdef CONFIG_PM
static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
{
- struct sport_device *sport =
- (struct sport_device *)dai->private_data;
pr_debug("%s : sport %d\n", __func__, dai->id);
- if (!dai->active)
- return 0;
+
if (dai->capture.active)
- sport_rx_stop(sport);
+ sport_rx_stop(sport_handle);
if (dai->playback.active)
- sport_tx_stop(sport);
+ sport_tx_stop(sport_handle);
return 0;
}
static int bf5xx_i2s_resume(struct snd_soc_dai *dai)
{
int ret;
- struct sport_device *sport =
- (struct sport_device *)dai->private_data;
pr_debug("%s : sport %d\n", __func__, dai->id);
- if (!dai->active)
- return 0;
- ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0);
+ ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
+ bf5xx_i2s.rcr2, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
- ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0);
+ ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
+ bf5xx_i2s.tcr2, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h
index 7107d1a..264ecdc 100644
--- a/sound/soc/blackfin/bf5xx-i2s.h
+++ b/sound/soc/blackfin/bf5xx-i2s.h
@@ -1,5 +1,5 @@
/*
- * linux/sound/arm/bf5xx-i2s.h
+ * sound/soc/blackfin/bf5xx-i2s.h
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index 469ce7f..99051ff 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -326,7 +326,7 @@
int sport_tx_start(struct sport_device *sport)
{
- unsigned flags;
+ unsigned long flags;
pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__,
sport->tx_run, sport->rx_run);
if (sport->tx_run)
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 @@
* 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/ad1836.c b/sound/soc/codecs/ad1836.c
index 01343dc..c48485f 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -251,8 +251,7 @@
static struct spi_driver ad1836_spi_driver = {
.driver = {
- .name = "ad1836-spi",
- .bus = &spi_bus_type,
+ .name = "ad1836",
.owner = THIS_MODULE,
},
.probe = ad1836_spi_probe,
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
index 9a049a1..34b30ef 100644
--- a/sound/soc/codecs/ad1938.c
+++ b/sound/soc/codecs/ad1938.c
@@ -456,7 +456,6 @@
static struct spi_driver ad1938_spi_driver = {
.driver = {
.name = "ad1938",
- .bus = &spi_bus_type,
.owner = THIS_MODULE,
},
.probe = ad1938_spi_probe,
@@ -515,6 +514,7 @@
codec->num_dai = 1;
codec->write = ad1938_write_reg;
codec->read = ad1938_read_reg_cache;
+ codec->set_bias_level = ad1938_set_bias_level;
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d80d414..5ad677c 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -595,6 +595,7 @@
/* Mono Capture mixer-mux */
{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
+ {"Capture Left Mixer", "Stereo", "Capture Left Mux"},
{"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
{"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
{"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
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 @@
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 @@
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 @@
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 @@
#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 @@
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 @@
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 @@
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 eca22d7..5d1f98a 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -332,14 +332,6 @@
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 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 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,
@@ -512,34 +504,49 @@
int channel_size)
{
u32 fmt = 0;
+ u32 mask, rotate;
switch (channel_size) {
case DAVINCI_AUDIO_WORD_8:
fmt = 0x03;
+ rotate = 6;
+ mask = 0x000000ff;
break;
case DAVINCI_AUDIO_WORD_12:
fmt = 0x05;
+ rotate = 5;
+ mask = 0x00000fff;
break;
case DAVINCI_AUDIO_WORD_16:
fmt = 0x07;
+ rotate = 4;
+ mask = 0x0000ffff;
break;
case DAVINCI_AUDIO_WORD_20:
fmt = 0x09;
+ rotate = 3;
+ mask = 0x000fffff;
break;
case DAVINCI_AUDIO_WORD_24:
fmt = 0x0B;
+ rotate = 2;
+ mask = 0x00ffffff;
break;
case DAVINCI_AUDIO_WORD_28:
fmt = 0x0D;
+ rotate = 1;
+ mask = 0x0fffffff;
break;
case DAVINCI_AUDIO_WORD_32:
fmt = 0x0F;
+ rotate = 0;
+ mask = 0xffffffff;
break;
default:
@@ -550,6 +557,13 @@
RXSSZ(fmt), RXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
TXSSZ(fmt), TXSSZ(0x0F));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate),
+ TXROT(7));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate),
+ RXROT(7));
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask);
+
return 0;
}
@@ -638,7 +652,6 @@
printk(KERN_ERR "playback tdm slot %d not supported\n",
dev->tdm_slots);
- mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
} else {
/* bit stream is MSB first with no delay */
@@ -655,7 +668,6 @@
printk(KERN_ERR "capture tdm slot %d not supported\n",
dev->tdm_slots);
- mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF);
mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
}
}
@@ -700,7 +712,7 @@
{
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;
@@ -778,7 +790,6 @@
}
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,
@@ -829,20 +840,12 @@
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");
@@ -877,11 +880,10 @@
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);
@@ -890,13 +892,12 @@
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) {
@@ -904,7 +905,7 @@
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]);
@@ -916,8 +917,6 @@
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;
@@ -926,7 +925,6 @@
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;
@@ -939,8 +937,6 @@
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 @@
};
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 091dacb..359e99e 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -126,16 +126,9 @@
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 @@
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 @@
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/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6375b4e..dcb3181 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -138,7 +138,7 @@
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