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.

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;