summaryrefslogtreecommitdiffstats
path: root/sound/soc
diff options
context:
space:
mode:
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c5
-rw-r--r--sound/soc/blackfin/bf5xx-ad1836.c4
-rw-r--r--sound/soc/blackfin/bf5xx-i2s-pcm.c13
-rw-r--r--sound/soc/codecs/ad1836.c14
-rw-r--r--sound/soc/codecs/ad1836.h6
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/tlv320aic26.c14
-rw-r--r--sound/soc/codecs/tlv320aic3x.c9
-rw-r--r--sound/soc/codecs/wm8731.c29
-rw-r--r--sound/soc/codecs/wm8804.c9
-rw-r--r--sound/soc/codecs/wm8915.c3
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/codecs/wm8991.c1
-rw-r--r--sound/soc/codecs/wm8994.c2
-rw-r--r--sound/soc/fsl/fsl_dma.c9
-rw-r--r--sound/soc/imx/Kconfig7
-rw-r--r--sound/soc/imx/imx-pcm-dma-mx2.c2
-rw-r--r--sound/soc/imx/imx-ssi.c2
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c4
-rw-r--r--sound/soc/samsung/i2s.c4
-rw-r--r--sound/soc/soc-cache.c6
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dapm.c17
-rw-r--r--sound/soc/tegra/tegra_i2s.c6
24 files changed, 97 insertions, 80 deletions
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 7fbfa051f6e1..eda955b15834 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id)
if (IS_ERR(ssc))
pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
PTR_ERR(ssc));
- else
+ else {
ssc_pdev->dev.parent = &(ssc->pdev->dev);
- ssc_free(ssc);
+ ssc_free(ssc);
+ }
ret = platform_device_add(ssc_pdev);
if (ret < 0)
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index ea4951cf5526..f79d1655e035 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
.cpu_dai_name = "bfin-tdm.0",
.codec_dai_name = "ad1836-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad1836.0",
+ .codec_name = "spi0.4",
.ops = &bf5xx_ad1836_ops,
},
{
@@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
.cpu_dai_name = "bfin-tdm.1",
.codec_dai_name = "ad1836-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad1836.0",
+ .codec_name = "spi0.4",
.ops = &bf5xx_ad1836_ops,
},
};
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index b5101efd1c87..f1fd95bb6416 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
pr_debug("%s enter\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
diff = sport_curr_offset_tx(sport);
- frames = bytes_to_frames(substream->runtime, diff);
} else {
diff = sport_curr_offset_rx(sport);
- frames = bytes_to_frames(substream->runtime, diff);
}
+
+ /*
+ * TX at least can report one frame beyond the end of the
+ * buffer if we hit the wraparound case - clamp to within the
+ * buffer as the ALSA APIs require.
+ */
+ if (diff == snd_pcm_lib_buffer_bytes(substream))
+ diff = 0;
+
+ frames = bytes_to_frames(substream->runtime, diff);
+
return frames;
}
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index ab63d52e36e1..754c496412bd 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- word_len = 3;
+ word_len = AD1836_WORD_LEN_16;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- word_len = 1;
+ word_len = AD1836_WORD_LEN_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
- word_len = 0;
+ word_len = AD1836_WORD_LEN_24;
break;
}
- snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
- AD1836_DAC_WORD_LEN_MASK, word_len);
+ snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK,
+ word_len << AD1836_DAC_WORD_LEN_OFFSET);
- snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
- AD1836_ADC_WORD_LEN_MASK, word_len);
+ snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK,
+ word_len << AD1836_ADC_WORD_OFFSET);
return 0;
}
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 845596717fdf..9d6a3f8f8aaf 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -25,6 +25,7 @@
#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5)
#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5)
#define AD1836_DAC_WORD_LEN_MASK 0x18
+#define AD1836_DAC_WORD_LEN_OFFSET 3
#define AD1836_DAC_CTRL2 1
#define AD1836_DACL1_MUTE 0
@@ -51,6 +52,7 @@
#define AD1836_ADCL2_MUTE 2
#define AD1836_ADCR2_MUTE 3
#define AD1836_ADC_WORD_LEN_MASK 0x30
+#define AD1836_ADC_WORD_OFFSET 5
#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
@@ -60,4 +62,8 @@
#define AD1836_NUM_REGS 16
+#define AD1836_WORD_LEN_24 0x0
+#define AD1836_WORD_LEN_20 0x1
+#define AD1836_WORD_LEN_16 0x2
+
#endif
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570e3f1f..65f46047b1cb 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
default:
return -EINVAL;
}
- snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+ snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
/* format type */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e2a7608d3944..7859bdcc93db 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
}
- /* Configure PLL */
+ /**
+ * Configure PLL
+ * fsref = (mclk * PLLM) / 2048
+ * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal)
+ */
pval = 1;
- jval = (fsref == 44100) ? 7 : 8;
- dval = (fsref == 44100) ? 5264 : 1920;
+ /* compute J portion of multiplier */
+ jval = fsref / (aic26->mclk / 2048);
+ /* compute fractional DDDD component of multiplier */
+ dval = fsref - (jval * (aic26->mclk / 2048));
+ dval = (10000 * dval) / (aic26->mclk / 2048);
+ dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
qval = 0;
reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index c3d96fc8c267..789453d44ec5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1114,12 +1114,19 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
/* Sync reg_cache with the hardware */
codec->cache_only = 0;
- for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++)
+ for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
snd_soc_write(codec, i, cache[i]);
if (aic3x->model == AIC3X_MODEL_3007)
aic3x_init_3007(codec);
codec->cache_sync = 0;
} else {
+ /*
+ * Do soft reset to this codec instance in order to clear
+ * possible VDD leakage currents in case the supply regulators
+ * remain on
+ */
+ snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
+ codec->cache_sync = 1;
aic3x->power = 0;
/* HW writes are needless when bias is off */
codec->cache_only = 1;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2dc964b55e4f..76b4361e9b80 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
&wm8731_output_mixer_controls[0],
@@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
static const struct snd_soc_dapm_route wm8731_intercon[] = {
{"DAC", NULL, "OSC", wm8731_check_osc},
{"ADC", NULL, "OSC", wm8731_check_osc},
+ {"DAC", NULL, "ACTIVE"},
+ {"ADC", NULL, "ACTIVE"},
/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
@@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
-
- /* set active */
- snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
-
- return 0;
-}
-
-static void wm8731_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
-
- /* deactivate */
- if (!codec->active) {
- udelay(50);
- snd_soc_write(codec, WM8731_ACTIVE, 0x0);
- }
-}
-
static int wm8731_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
@@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
break;
case SND_SOC_BIAS_OFF:
- snd_soc_write(codec, WM8731_ACTIVE, 0x0);
snd_soc_write(codec, WM8731_PWR, 0xffff);
regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
wm8731->supplies);
@@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops wm8731_dai_ops = {
- .prepare = wm8731_pcm_prepare,
.hw_params = wm8731_hw_params,
- .shutdown = wm8731_shutdown,
.digital_mute = wm8731_mute,
.set_sysclk = wm8731_set_dai_sysclk,
.set_fmt = wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6785688f8806..9a5e67c5a6bd 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = {
#define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
+#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
static struct snd_soc_dai_driver wm8804_dai = {
.name = "wm8804-spdif",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = WM8804_RATES,
.formats = WM8804_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = WM8804_RATES,
.formats = WM8804_FORMATS,
},
.ops = &wm8804_dai_ops,
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index a0b1a7278284..e2ab4fac2819 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
int old;
/* Disable SYSCLK while we reconfigure */
- old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1);
+ old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
WM8915_SYSCLK_ENA, 0);
@@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
break;
case WM8915_FLL_MCLK2:
reg = 1;
+ break;
case WM8915_FLL_DACLRCLK1:
reg = 2;
break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index f90ae427242b..5e05eed96c38 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
return 0;
/* If the left PGA is enabled hit that VU bit... */
- if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA)
+ if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
reg_cache[WM8962_HPOUTL_VOLUME]);
/* ...otherwise the right. The VU is stereo. */
- if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA)
+ if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
reg_cache[WM8962_HPOUTR_VOLUME]);
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1bb73cd..6af23d06870f 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360b..c2fc0356c2a4 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
reg);
+
+ msleep(5);
}
wm8994->fll[id].in = freq_in;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 15dac0f20cd8..6680c0b4d203 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
* should allocate a DMA buffer only for the streams that are valid.
*/
- if (dai->driver->playback.channels_min) {
+ if (pcm->streams[0].substream) {
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
fsl_dma_hardware.buffer_bytes_max,
&pcm->streams[0].substream->dma_buffer);
@@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
}
}
- if (dai->driver->capture.channels_min) {
+ if (pcm->streams[1].substream) {
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) {
- snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
dev_err(card->dev, "can't alloc capture dma buffer\n");
+ snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
return ret;
}
}
@@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
dma_private->ld_buf_phys = ld_buf_phys;
dma_private->dma_buf_phys = substream->dma_buffer.addr;
- ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private);
+ ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio",
+ dma_private);
if (ret) {
dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n",
dma_private->irq, ret);
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index d8f130d39dd9..bb699bb55a50 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC
if SND_IMX_SOC
-config SND_MXC_SOC_SSI
- tristate
-
config SND_MXC_SOC_FIQ
tristate
@@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1
tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
select SND_SOC_WM8350
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_FIQ
help
Enable support for audio on the i.MX31ADS with the WM1133-EV1
@@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4
tristate "SoC audio support for Visstrim M10 boards"
depends on MACH_IMX27_VISSTRIM_M10
select SND_SOC_TVL320AIC32X4
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_MX2
help
Say Y if you want to add support for SoC audio on Visstrim SM10
@@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97
tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
depends on MACH_PCM043 || MACH_PCA100
select SND_SOC_WM9712
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_FIQ
help
Say Y if you want to add support for SoC audio on Phytec phyCORE
@@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320
|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
|| MACH_EUKREA_MBIMXSD51_BASEBOARD
select SND_SOC_TLV320AIC23
- select SND_MXC_SOC_SSI
select SND_MXC_SOC_FIQ
help
Enable I2S based access to the TLV320AIC23B codec attached
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index aab7765f401a..4173b3d87f97 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void)
platform_driver_unregister(&imx_pcm_driver);
}
module_exit(snd_imx_pcm_exit);
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 5b13feca7537..61fceb09cdb5 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -774,4 +774,4 @@ module_exit(imx_ssi_exit);
MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2ce0b2d891d5..fab20a54e863 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
- if (dai->driver->playback.channels_min) {
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
goto out;
}
- if (dai->driver->capture.channels_min) {
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index ffa09b3b2caa..992a732b5211 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s)
if (!i2s)
return false;
- active = readl(i2s->addr + I2SMOD);
+ active = readl(i2s->addr + I2SCON);
if (is_secondary(i2s))
active &= CON_TXSDMA_ACTIVE;
@@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s)
if (!i2s)
return false;
- active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE;
+ active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE;
return active ? true : false;
}
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 06b7b81a1601..039b9532b270 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
switch (control) {
- case SND_SOC_CUSTOM:
- break;
-
case SND_SOC_I2C:
#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
codec->hw_write = (hw_write_t)i2c_master_send;
@@ -466,6 +463,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx,
static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx,
unsigned int word_size)
{
+ if (!base)
+ return -1;
+
switch (word_size) {
case 1: {
const u8 *cache = base;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043ed7fc0..b194be09e74d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->long_name ? card->long_name : card->name);
- snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
- "%s", card->driver_name ? card->driver_name : card->name);
+ if (card->driver_name)
+ strlcpy(card->snd_card->driver, card->driver_name,
+ sizeof(card->snd_card->driver));
if (card->late_probe) {
ret = card->late_probe(card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 776e6f418306..32ab7fc4579a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
}
/* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dapm_widget *w)
+static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
{
+ struct snd_soc_dapm_context *dapm = w->dapm;
int i, ret = 0;
size_t name_len, prefix_len;
struct snd_soc_dapm_path *path;
@@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
}
/* create new dapm mux control */
-static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dapm_widget *w)
+static int dapm_new_mux(struct snd_soc_dapm_widget *w)
{
+ struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_soc_dapm_path *path = NULL;
struct snd_kcontrol *kcontrol;
struct snd_card *card = dapm->card->snd_card;
@@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
}
/* create new dapm volume control */
-static int dapm_new_pga(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dapm_widget *w)
+static int dapm_new_pga(struct snd_soc_dapm_widget *w)
{
if (w->num_kcontrols)
dev_err(w->dapm->dev,
@@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
w->power_check = dapm_generic_check_power;
- dapm_new_mixer(dapm, w);
+ dapm_new_mixer(w);
break;
case snd_soc_dapm_mux:
case snd_soc_dapm_virt_mux:
case snd_soc_dapm_value_mux:
w->power_check = dapm_generic_check_power;
- dapm_new_mux(dapm, w);
+ dapm_new_mux(w);
break;
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
@@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
case snd_soc_dapm_pga:
case snd_soc_dapm_out_drv:
w->power_check = dapm_generic_check_power;
- dapm_new_pga(dapm, w);
+ dapm_new_pga(w);
break;
case snd_soc_dapm_input:
case snd_soc_dapm_output:
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e20548c..95f03c10b4f7 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
if (i2sclock % (2 * srate))
reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
+ if (!i2s->clk_refs)
+ clk_enable(i2s->clk_i2s);
+
tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+ if (!i2s->clk_refs)
+ clk_disable(i2s->clk_i2s);
+
return 0;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-29 00:03:21 +0100