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authorTakashi Iwai <tiwai@suse.de>2013-12-13 14:54:53 +0100
committerTakashi Iwai <tiwai@suse.de>2013-12-13 14:54:53 +0100
commitafdcd431cebe3498db9aa963c780fdd5099917ec (patch)
tree16d9155e136f0df56689eead95cf44935aab2a0c /sound
parentc29cb5eb8157a0049c882672a7f941261f23ea34 (diff)
parente20ab019e28dcf09c2727aa69e2a073ed66718b3 (diff)
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Merge tag 'asoc-v3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13 A few driver and error handling fixes plus a fix to ensure that we mute streams when we should. The Atmel trigger addition is a fix to ensure that we do the correct sequence of interactions with the hardware.
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c30
-rw-r--r--sound/soc/atmel/sam9x5_wm8731.c2
-rw-r--r--sound/soc/codecs/wm8962.c13
-rw-r--r--sound/soc/fsl/imx-wm8962.c2
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c38
-rw-r--r--sound/soc/soc-pcm.c5
-rw-r--r--sound/soc/tegra/tegra20_i2s.c6
-rw-r--r--sound/soc/tegra/tegra20_spdif.c10
-rw-r--r--sound/soc/tegra/tegra30_i2s.c6
9 files changed, 84 insertions, 28 deletions
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 8697cedccd21..1ead3c977a51 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
dma_params = ssc_p->dma_params[dir];
- ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error);
pr_debug("%s enabled SSC_SR=0x%08x\n",
@@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
return 0;
}
+static int atmel_ssc_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
+ struct atmel_pcm_dma_params *dma_params;
+ int dir;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = 0;
+ else
+ dir = 1;
+
+ dma_params = ssc_p->dma_params[dir];
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+ break;
+ default:
+ ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+ break;
+ }
+
+ return 0;
+}
#ifdef CONFIG_PM
static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
@@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = {
.startup = atmel_ssc_startup,
.shutdown = atmel_ssc_shutdown,
.prepare = atmel_ssc_prepare,
+ .trigger = atmel_ssc_trigger,
.hw_params = atmel_ssc_hw_params,
.set_fmt = atmel_ssc_set_dai_fmt,
.set_clkdiv = atmel_ssc_set_dai_clkdiv,
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 1b372283bd01..7d6a9055874b 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -109,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
dai->stream_name = "WM8731 PCM";
dai->codec_dai_name = "wm8731-hifi";
dai->init = sam9x5_wm8731_init;
- dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM;
ret = snd_soc_of_parse_card_name(card, "atmel,model");
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2631b6..0f17ed3e29f4 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8962_CLOCKING_4,
WM8962_SYSCLK_RATE_MASK, clocking4);
+ /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK.
+ * So we here provisionally enable it and then disable it afterward
+ * if current bias_level hasn't reached SND_SOC_BIAS_ON.
+ */
+ if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA);
+
dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
+
+ if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_SYSCLK_ENA_MASK, 0);
+
if (dspclk < 0) {
dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
return;
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 61e48852b9e8..3fd76bc391de 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
break;
}
- dapm->bias_level = level;
-
return 0;
}
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index cbc9c96ce1f4..41949af3baae 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
}
}
+static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
+{
+ unsigned int i;
+
+ for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
+ i++) {
+ if (!pcm->chan[i])
+ continue;
+ dma_release_channel(pcm->chan[i]);
+ if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+ break;
+ }
+}
+
/**
* snd_dmaengine_pcm_register - Register a dmaengine based PCM device
* @dev: The parent device for the PCM device
@@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev,
const struct snd_dmaengine_pcm_config *config, unsigned int flags)
{
struct dmaengine_pcm *pcm;
+ int ret;
pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
if (!pcm)
@@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev,
dmaengine_pcm_request_chan_of(pcm, dev);
if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
- return snd_soc_add_platform(dev, &pcm->platform,
+ ret = snd_soc_add_platform(dev, &pcm->platform,
&dmaengine_no_residue_pcm_platform);
else
- return snd_soc_add_platform(dev, &pcm->platform,
+ ret = snd_soc_add_platform(dev, &pcm->platform,
&dmaengine_pcm_platform);
+ if (ret)
+ goto err_free_dma;
+
+ return 0;
+
+err_free_dma:
+ dmaengine_pcm_release_chan(pcm);
+ kfree(pcm);
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register);
@@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
{
struct snd_soc_platform *platform;
struct dmaengine_pcm *pcm;
- unsigned int i;
platform = snd_soc_lookup_platform(dev);
if (!platform)
@@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
pcm = soc_platform_to_pcm(platform);
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
- if (pcm->chan[i]) {
- dma_release_channel(pcm->chan[i]);
- if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
- break;
- }
- }
-
snd_soc_remove_platform(platform);
+ dmaengine_pcm_release_chan(pcm);
kfree(pcm);
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 11a90cd027fa..891b9a9bcbf8 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -600,12 +600,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
/* apply codec digital mute */
- if (!codec->active)
+ if ((playback && codec_dai->playback_active == 1) ||
+ (!playback && codec_dai->capture_active == 1))
snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
/* free any machine hw params */
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 364bf6a907e1..8c819f811470 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
unsigned int fmt)
{
struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- unsigned int mask, val;
+ unsigned int mask = 0, val = 0;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
@@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+ mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+ val |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
break;
case SND_SOC_DAIFMT_CBM_CFM:
break;
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 08bc6931c7c7..8c7c1028e579 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
{
struct device *dev = dai->dev;
struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
- unsigned int mask, val;
+ unsigned int mask = 0, val = 0;
int ret, spdifclock;
- mask = TEGRA20_SPDIF_CTRL_PACK |
- TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+ mask |= TEGRA20_SPDIF_CTRL_PACK |
+ TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- val = TEGRA20_SPDIF_CTRL_PACK |
- TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+ val |= TEGRA20_SPDIF_CTRL_PACK |
+ TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
break;
default:
return -EINVAL;
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 231a785b3921..02247fee1cf7 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
unsigned int fmt)
{
struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
- unsigned int mask, val;
+ unsigned int mask = 0, val = 0;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
@@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+ mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+ val |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
break;
case SND_SOC_DAIFMT_CBM_CFM:
break;