Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/arizona', 'asoc/fix/atmel...

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 8697cedccd21240f76b4b5076ba6d0968a394d75..1ead3c977a51743619b0bb2084ec174b7cf867a8 100644 (file)
--- 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 1b372283bd01a947c26dc4da302f9b077adfe74a..7d6a9055874b822bb440eab72a66dc5c034127ed 100644 (file)
--- 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/wm5110.c b/sound/soc/codecs/wm5110.c
index 99b359e19d35f46a3c95946d976b294300c6ea70..0ab2dc296474373e1e407abf0762afeeffead4c6 100644 (file)
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1012,7 +1012,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
        { "AEC Loopback", "HPOUT3L", "OUT3L" },
        { "AEC Loopback", "HPOUT3R", "OUT3R" },
        { "HPOUT3L", NULL, "OUT3L" },
-       { "HPOUT3R", NULL, "OUT3L" },
+       { "HPOUT3R", NULL, "OUT3R" },
 
        { "AEC Loopback", "SPKOUTL", "OUT4L" },
        { "SPKOUTLN", NULL, "OUT4L" },

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3938fb1c203ed86b5452ff130071d349a6821a87..53bbfac6a83ad14ed43d2aafca2514ea3a319468 100644 (file)
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
        case SND_SOC_DAIFMT_DSP_B:
-               aif1 |= WM8904_AIF_LRCLK_INV;
+               aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
        case SND_SOC_DAIFMT_DSP_A:
                aif1 |= 0x3;
                break;

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2631b61827bcebca8af1926ea02b5655e9..0f17ed3e29f41dc9d7ddebb505b7fad8fc545b02 100644 (file)
--- 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/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9744d4b88b50cc922cfdc65a0b4247e567..4fbcab63e61f1c5d8b4843fdec5c6d3dbfa498ed 100644 (file)
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
                return ret;
 
        /* Wait for the RAM to start, should be near instantaneous */
-       count = 0;
-       do {
+       for (count = 0; count < 10; ++count) {
                ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
                                  &val);
                if (ret != 0)
                        return ret;
-       } while (!(val & ADSP2_RAM_RDY) && ++count < 10);
+
+               if (val & ADSP2_RAM_RDY)
+                       break;
+
+               msleep(1);
+       }
 
        if (!(val & ADSP2_RAM_RDY)) {
                adsp_err(dsp, "Failed to start DSP RAM\n");

diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 61e48852b9e8bd21b26eec98cca4fc9ba0eea145..3fd76bc391de19a2431dd320d0abfbdcd9616cf5 100644 (file)
--- 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/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 0b18f654b41340908d022003485d8a88c5e0c94e..3920a5e8125f886e15caa6607fed9e84a32950cd 100644 (file)
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -473,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
        .playback = {
                .channels_min = 1,
                .channels_max = 2,
-               .rates = SNDRV_PCM_RATE_8000_192000 |
-                        SNDRV_PCM_RATE_CONTINUOUS |
-                        SNDRV_PCM_RATE_KNOT,
+               .rates = SNDRV_PCM_RATE_CONTINUOUS,
+               .rate_min = 5512,
+               .rate_max = 192000,
                .formats = KIRKWOOD_I2S_FORMATS,
        },
        .capture = {
                .channels_min = 1,
                .channels_max = 2,
-               .rates = SNDRV_PCM_RATE_8000_192000 |
-                        SNDRV_PCM_RATE_CONTINUOUS |
-                        SNDRV_PCM_RATE_KNOT,
+               .rates = SNDRV_PCM_RATE_CONTINUOUS,
+               .rate_min = 5512,
+               .rate_max = 192000,
                .formats = KIRKWOOD_I2S_FORMATS,
        },
        .ops = &kirkwood_i2s_dai_ops,
@@ -494,17 +494,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
        .playback = {
                .channels_min = 1,
                .channels_max = 2,
-               .rates = SNDRV_PCM_RATE_8000_192000 |
-                        SNDRV_PCM_RATE_CONTINUOUS |
-                        SNDRV_PCM_RATE_KNOT,
+               .rates = SNDRV_PCM_RATE_CONTINUOUS,
+               .rate_min = 5512,
+               .rate_max = 192000,
                .formats = KIRKWOOD_SPDIF_FORMATS,
        },
        .capture = {
                .channels_min = 1,
                .channels_max = 2,
-               .rates = SNDRV_PCM_RATE_8000_192000 |
-                        SNDRV_PCM_RATE_CONTINUOUS |
-                        SNDRV_PCM_RATE_KNOT,
+               .rates = SNDRV_PCM_RATE_CONTINUOUS,
+               .rate_min = 5512,
+               .rate_max = 192000,
                .formats = KIRKWOOD_SPDIF_FORMATS,
        },
        .ops = &kirkwood_i2s_dai_ops,

diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 364bf6a907e1c39a36df89ded1d6b2681172940f..8c819f8114708f7a94e33929a04ebecbdf3e2ffc 100644 (file)
--- 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 08bc6931c7c7fc0477703037098a8e7185d7c69d..8c7c1028e5797dbc9b23ec88b58a4cd840db1c5c 100644 (file)
--- 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 231a785b3921a5bd95d87914a92e7f5f2d266a63..02247fee1cf7e1450fb4bbcdf2e58cf5ee76c7e8 100644 (file)
--- 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;