Merge tag 'asoc-v3.16-rc5' into asoc-linus

ASoC: Fixes for v3.16

A bigger batch of changes than I would like as I didn't send any for a
few weeks without noticing how many had built up.  They are almost all
driver specific though, larger changes are:

 - Fixes to the newly added Baytrail/MAX98090 which look like some QA
   was missed on the microphone detection.
 - Deletion of some erroniously listed audio formats for Haswell.
 - Fix debugfs creation in the core so that we don't try to generate
   multiple directories with the same name, relatively large textually
   but simple to inspect by eye and test.
 - A couple of bugfixes for the rcar driver one of which which involves
   a bit of code motion to move initailisation of some hardware out of
   common paths into device specific ones.
 - Ensure both channels are powered up for mono outputs on Arizona
   devices, involving some simple data tables listing the outputs and a
   loop over them.
 - A couple of fixes to save and restore information on suspended and
   idle Samsung I2S controllers.

# gpg: Signature made Tue 22 Jul 2014 00:52:53 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"

24 files changed, 185 insertions, 72 deletions

diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index a3881c4381c9..bcf591373a7a 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -290,19 +290,19 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream,
         unsigned int sample_size = runtime->sample_bits / 8;
         void *buf = runtime->dma_area;
         struct bf5xx_i2s_pcm_data *dma_data;
-        unsigned int offset, size;
+        unsigned int offset, samples;
 
         dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
         if (dma_data->tdm_mode) {
                 offset = pos * 8 * sample_size;
-                size = count * 8 * sample_size;
+                samples = count * 8;
         } else {
                 offset = frames_to_bytes(runtime, pos);
-                size = frames_to_bytes(runtime, count);
+                samples = count * runtime->channels;
         }
 
-        snd_pcm_format_set_silence(runtime->format, buf + offset, size);
+        snd_pcm_format_set_silence(runtime->format, buf + offset, samples);
 
         return 0;
 }
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index d71c59cf7bdd..370b742117ef 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg,
 
         *value = 0;
 
-        for (i = 0; i < size; i++)
-                *value |= recv_buf[i] << (i * 8);
+        for (i = 0; i < size; i++) {
+                *value <<= 8;
+                *value |= recv_buf[i];
+        }
 
         return 0;
 }
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 29e198f57d4c..747c71e59c04 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(arizona_init_spk);
 
+static const struct snd_soc_dapm_route arizona_mono_routes[] = {
+        { "OUT1R", NULL, "OUT1L" },
+        { "OUT2R", NULL, "OUT2L" },
+        { "OUT3R", NULL, "OUT3L" },
+        { "OUT4R", NULL, "OUT4L" },
+        { "OUT5R", NULL, "OUT5L" },
+        { "OUT6R", NULL, "OUT6L" },
+};
+
+int arizona_init_mono(struct snd_soc_codec *codec)
+{
+        struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+        struct arizona *arizona = priv->arizona;
+        int i;
+
+        for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) {
+                if (arizona->pdata.out_mono[i])
+                        snd_soc_dapm_add_routes(&codec->dapm,
+                                                &arizona_mono_routes[i], 1);
+        }
+
+        return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_mono);
+
 int arizona_init_gpio(struct snd_soc_codec *codec)
 {
         struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 05ae17f5bca3..942cfb197b6d 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source,
 
 extern int arizona_init_spk(struct snd_soc_codec *codec);
 extern int arizona_init_gpio(struct snd_soc_codec *codec);
+extern int arizona_init_mono(struct snd_soc_codec *codec);
 
 extern int arizona_init_dai(struct arizona_priv *priv, int dai);
 
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index fdc4bd27b0df..8e68ef5de849 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -445,9 +445,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
         SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
 
         SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
-                              CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+                              CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv),
         SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
-                              CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+                              CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv),
 
         SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
                         0, 0x00, 1, tone_tlv),
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f5fccc7a8e89..d97f1ce7ff7d 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2284,7 +2284,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
         /* Register for interrupts */
         dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
 
-        ret = request_threaded_irq(max98090->irq, NULL,
+        ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL,
                 max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
                 "max98090_interrupt", codec);
         if (ret < 0) {
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 3d39f0b5b4a8..8f4c73d17c87 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1277,7 +1277,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
                         return ret;
         }
 
-        ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+        ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
                                  sgtl5000->supplies);
         if (ret)
                 goto err_ldo_remove;
@@ -1285,13 +1285,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
         ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
                                         sgtl5000->supplies);
         if (ret)
-                goto err_ldo_remove;
+                goto err_regulator_free;
 
         /* wait for all power rails bring up */
         udelay(10);
 
         return 0;
 
+err_regulator_free:
+        regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+                                sgtl5000->supplies);
 err_ldo_remove:
         if (!external_vddd)
                 ldo_regulator_remove(codec);
@@ -1361,6 +1364,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 err:
         regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
                                                 sgtl5000->supplies);
+        regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+                                sgtl5000->supplies);
         ldo_regulator_remove(codec);
 
         return ret;
@@ -1374,6 +1379,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
 
         regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
                                                 sgtl5000->supplies);
+        regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+                                sgtl5000->supplies);
         ldo_regulator_remove(codec);
 
         return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index e12fafbb1e09..5360772bc1ad 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
         case SNDRV_PCM_FORMAT_S20_3LE:
                 data |= (0x01 << 4);
                 break;
-        case SNDRV_PCM_FORMAT_S24_LE:
+        case SNDRV_PCM_FORMAT_S24_3LE:
                 data |= (0x02 << 4);
                 break;
         case SNDRV_PCM_FORMAT_S32_LE:
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2e5fcb559e90..62ef54456499 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 
         arizona_init_spk(codec);
         arizona_init_gpio(codec);
+        arizona_init_mono(codec);
 
         ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8);
         if (ret != 0)
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 060027182dcb..2537725dd53f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs)
         return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_init);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 50a098749b9e..fdbb16fffd30 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -6,6 +6,7 @@ config SND_DAVINCI_SOC_I2S
         tristate
 
 config SND_DAVINCI_SOC_MCASP
+        depends on SND_DAVINCI_SOC || SND_OMAP_SOC
         tristate
 
 config SND_DAVINCI_SOC_VCIF
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 9afb14629a17..bfcc6c3dc2fd 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -720,6 +720,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 
         case SNDRV_PCM_FORMAT_U24_LE:
         case SNDRV_PCM_FORMAT_S24_LE:
+                dma_params->data_type = 4;
+                word_length = 24;
+                break;
+
         case SNDRV_PCM_FORMAT_U32_LE:
         case SNDRV_PCM_FORMAT_S32_LE:
                 dma_params->data_type = 4;
@@ -1223,14 +1227,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
                 goto err;
 
         switch (mcasp->version) {
+#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \
+        (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+         IS_MODULE(CONFIG_SND_DAVINCI_SOC))
         case MCASP_VERSION_1:
         case MCASP_VERSION_2:
         case MCASP_VERSION_3:
                 ret = davinci_soc_platform_register(&pdev->dev);
                 break;
+#endif
+#if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \
+        (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+         IS_MODULE(CONFIG_SND_OMAP_SOC))
         case MCASP_VERSION_4:
                 ret = omap_pcm_platform_register(&pdev->dev);
                 break;
+#endif
         default:
                 dev_err(&pdev->dev, "Invalid McASP version: %d\n",
                         mcasp->version);
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index c5a0e8af8226..1b6ee2ce849f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -106,7 +106,7 @@ irq_rx:
         xcsr &= ~FSL_SAI_CSR_xF_MASK;
 
         if (flags)
-                regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr);
+                regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr);
 
 out:
         if (irq_none)
@@ -371,10 +371,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 
                 /* Check if the opposite FRDE is also disabled */
                 if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) {
+                        /* Disable both directions and reset their FIFOs */
                         regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
-                                           FSL_SAI_CSR_TERE, 0);
+                                           FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR,
+                                           FSL_SAI_CSR_FR);
                         regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
-                                           FSL_SAI_CSR_TERE, 0);
+                                           FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR,
+                                           FSL_SAI_CSR_FR);
                 }
                 break;
         default:
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 03a7fdcdf114..159e517fa09a 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -116,6 +116,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 {
         struct device_node *node;
         struct clk *clk;
+        u32 val;
         int ret;
 
         /*
@@ -151,10 +152,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
                 }
 
                 dai->sysclk = clk_get_rate(clk);
-        } else if (of_property_read_bool(np, "system-clock-frequency")) {
-                of_property_read_u32(np,
-                                     "system-clock-frequency",
-                                     &dai->sysclk);
+        } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) {
+                dai->sysclk = val;
         } else {
                 clk = of_clk_get(node, 0);
                 if (!IS_ERR(clk))
@@ -303,6 +302,7 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 {
         struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
         struct simple_dai_props *dai_props = priv->dai_props;
+        u32 val;
         int ret;
 
         /* parsing the card name from DT */
@@ -325,8 +325,9 @@ static int asoc_simple_card_parse_of(struct device_node *node,
         }
 
         /* Factor to mclk, used in hw_params() */
-        of_property_read_u32(node, "simple-audio-card,mclk-fs",
-                             &priv->mclk_fs);
+        ret = of_property_read_u32(node, "simple-audio-card,mclk-fs", &val);
+        if (ret == 0)
+                priv->mclk_fs = val;
 
         dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ?
                 priv->snd_card.name : "");
diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c
index 5fc98c64a3f4..5cfb41ec3fab 100644
--- a/sound/soc/intel/byt-max98090.c
+++ b/sound/soc/intel/byt-max98090.c
@@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = {
 
 static const struct snd_soc_dapm_route byt_max98090_audio_map[] = {
         {"IN34", NULL, "Headset Mic"},
-        {"IN34", NULL, "MICBIAS"},
-        {"MICBIAS", NULL, "Headset Mic"},
+        {"Headset Mic", NULL, "MICBIAS"},
         {"DMICL", NULL, "Int Mic"},
         {"Headphone", NULL, "HPL"},
         {"Headphone", NULL, "HPR"},
@@ -84,7 +83,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = {
         {
                 .name           = "mic-gpio",
                 .idx            = 1,
-                .report         = SND_JACK_MICROPHONE | SND_JACK_LINEIN,
+                .invert         = 1,
+                .report         = SND_JACK_MICROPHONE,
                 .debounce_time  = 200,
         },
 };
@@ -108,7 +108,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
         }
 
         /* Enable jack detection */
-        ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack);
+        ret = snd_soc_jack_new(codec, "Headset",
+                               SND_JACK_LINEOUT | SND_JACK_HEADSET, jack);
         if (ret)
                 return ret;
 
@@ -117,13 +118,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
         if (ret)
                 return ret;
 
-        ret = snd_soc_jack_add_gpiods(card->dev->parent, jack,
-                                      ARRAY_SIZE(hs_jack_gpios),
-                                      hs_jack_gpios);
-        if (ret)
-                return ret;
-
-        return max98090_mic_detect(codec, jack);
+        return snd_soc_jack_add_gpiods(card->dev->parent, jack,
+                                       ARRAY_SIZE(hs_jack_gpios),
+                                       hs_jack_gpios);
 }
 
 static struct snd_soc_dai_link byt_max98090_dais[] = {
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 8eab97368ea7..599401c0c655 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = {
                                   SNDRV_PCM_INFO_PAUSE |
                                   SNDRV_PCM_INFO_RESUME,
         .formats                = SNDRV_PCM_FMTBIT_S16_LE |
-                                  SNDRV_PCM_FORMAT_S24_LE,
+                                  SNDRV_PCM_FMTBIT_S24_LE,
         .period_bytes_min       = 384,
         .period_bytes_max       = 48000,
         .periods_min            = 2,
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index 535f517629fd..a33b931181dc 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -359,6 +359,17 @@ static u32 hsw_block_get_bit(struct sst_mem_block *block)
         return bit;
 }
 
+/*dummy read a SRAM block.*/
+static void sst_mem_block_dummy_read(struct sst_mem_block *block)
+{
+        u32 size;
+        u8 tmp_buf[4];
+        struct sst_dsp *sst = block->dsp;
+
+        size = block->size > 4 ? 4 : block->size;
+        memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size);
+}
+
 /* enable 32kB memory block - locks held by caller */
 static int hsw_block_enable(struct sst_mem_block *block)
 {
@@ -378,6 +389,8 @@ static int hsw_block_enable(struct sst_mem_block *block)
         /* wait 18 DSP clock ticks */
         udelay(10);
 
+        /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/
+        sst_mem_block_dummy_read(block);
         return 0;
 }
 
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index 058efb17c568..61bf6da4bb02 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -80,7 +80,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = {
                                   SNDRV_PCM_INFO_PAUSE |
                                   SNDRV_PCM_INFO_RESUME |
                                   SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
-        .formats                = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE |
+        .formats                = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
                                   SNDRV_PCM_FMTBIT_S32_LE,
         .period_bytes_min       = PAGE_SIZE,
         .period_bytes_max       = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE,
@@ -400,7 +400,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
                 sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16);
                 break;
         case SNDRV_PCM_FORMAT_S24_LE:
-                bits = SST_HSW_DEPTH_24BIT;
+                bits = SST_HSW_DEPTH_32BIT;
+                sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24);
+                break;
+        case SNDRV_PCM_FORMAT_S8:
+                bits = SST_HSW_DEPTH_8BIT;
+                sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8);
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                bits = SST_HSW_DEPTH_32BIT;
                 sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32);
                 break;
         default:
@@ -685,8 +693,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
 }
 
 #define HSW_FORMATS \
-        (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
-         SNDRV_PCM_FMTBIT_S32_LE)
+        (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \
+        SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
+        SNDRV_PCM_FMTBIT_S8)
 
 static struct snd_soc_dai_driver hsw_dais[] = {
         {
@@ -696,7 +705,7 @@ static struct snd_soc_dai_driver hsw_dais[] = {
                         .channels_min = 2,
                         .channels_max = 2,
                         .rates = SNDRV_PCM_RATE_48000,
-                        .formats = SNDRV_PCM_FMTBIT_S16_LE,
+                        .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
                 },
         },
         {
@@ -727,8 +736,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
                         .stream_name = "Loopback Capture",
                         .channels_min = 2,
                         .channels_max = 2,
-                        .rates = SNDRV_PCM_RATE_8000_192000,
-                        .formats = HSW_FORMATS,
+                        .rates = SNDRV_PCM_RATE_48000,
+                        .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
                 },
         },
         {
@@ -737,8 +746,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
                         .stream_name = "Analog Capture",
                         .channels_min = 2,
                         .channels_max = 2,
-                        .rates = SNDRV_PCM_RATE_8000_192000,
-                        .formats = HSW_FORMATS,
+                        .rates = SNDRV_PCM_RATE_48000,
+                        .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
                 },
         },
 };
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 7eba7979b9af..1c8d01166e5b 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -570,7 +570,7 @@ err_release_none:
         return ret;
 }
 
-static void s6000_i2s_remove(struct platform_device *pdev)
+static int s6000_i2s_remove(struct platform_device *pdev)
 {
         struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
         struct resource *region;
@@ -597,6 +597,8 @@ static void s6000_i2s_remove(struct platform_device *pdev)
         iounmap(mmio);
         region = platform_get_resource(pdev, IORESOURCE_IO, 0);
         release_mem_region(region->start, resource_size(region));
+
+        return 0;
 }
 
 static struct platform_driver s6000_i2s_driver = {
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 2ac76fa3e742..d2533dbc8399 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -68,6 +68,8 @@ struct i2s_dai {
 #define DAI_OPENED      (1 << 0) /* Dai is opened */
 #define DAI_MANAGER     (1 << 1) /* Dai is the manager */
         unsigned mode;
+        /* CDCLK pin direction: 0  - input, 1 - output */
+        unsigned int cdclk_out:1;
         /* Driver for this DAI */
         struct snd_soc_dai_driver i2s_dai_drv;
         /* DMA parameters */
@@ -737,6 +739,9 @@ static int i2s_startup(struct snd_pcm_substream *substream,
 
         spin_unlock_irqrestore(&lock, flags);
 
+        if (!is_opened(other) && i2s->cdclk_out)
+                i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK,
+                                0, SND_SOC_CLOCK_OUT);
         return 0;
 }
 
@@ -752,9 +757,13 @@ static void i2s_shutdown(struct snd_pcm_substream *substream,
         i2s->mode &= ~DAI_OPENED;
         i2s->mode &= ~DAI_MANAGER;
 
-        if (is_opened(other))
+        if (is_opened(other)) {
                 other->mode |= DAI_MANAGER;
-
+        } else {
+                u32 mod = readl(i2s->addr + I2SMOD);
+                i2s->cdclk_out = !(mod & MOD_CDCLKCON);
+                other->cdclk_out = i2s->cdclk_out;
+        }
         /* Reset any constraint on RFS and BFS */
         i2s->rfs = 0;
         i2s->bfs = 0;
@@ -920,11 +929,9 @@ static int i2s_suspend(struct snd_soc_dai *dai)
 {
         struct i2s_dai *i2s = to_info(dai);
 
-        if (dai->active) {
-                i2s->suspend_i2smod = readl(i2s->addr + I2SMOD);
-                i2s->suspend_i2scon = readl(i2s->addr + I2SCON);
-                i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR);
-        }
+        i2s->suspend_i2smod = readl(i2s->addr + I2SMOD);
+        i2s->suspend_i2scon = readl(i2s->addr + I2SCON);
+        i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR);
 
         return 0;
 }
@@ -933,11 +940,9 @@ static int i2s_resume(struct snd_soc_dai *dai)
 {
         struct i2s_dai *i2s = to_info(dai);
 
-        if (dai->active) {
-                writel(i2s->suspend_i2scon, i2s->addr + I2SCON);
-                writel(i2s->suspend_i2smod, i2s->addr + I2SMOD);
-                writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR);
-        }
+        writel(i2s->suspend_i2scon, i2s->addr + I2SCON);
+        writel(i2s->suspend_i2smod, i2s->addr + I2SMOD);
+        writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR);
 
         return 0;
 }
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 4e86265f625c..ed76901f8202 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -297,7 +297,6 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma,
         for (i = 1; i < MOD_MAX; i++) {
                 if (!src) {
                         mod[i] = ssi;
-                        break;
                 } else if (!dvc) {
                         mod[i] = src;
                         src = NULL;
@@ -308,6 +307,9 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma,
 
                 if (mod[i] == this)
                         index = i;
+
+                if (mod[i] == ssi)
+                        break;
         }
 
         if (is_play) {
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 1dd2b7d38c2c..0280a11c0899 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -184,7 +184,7 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
 #define RDMA_CMD_O_N(addr, i)   (addr ##_reg - 0x004f8000 + (0x400 * i))
 #define RDMA_CMD_O_P(addr, i)   (addr ##_reg - 0x001f8000 + (0x400 * i))
 
-void rsnd_gen_dma_addr(struct rsnd_priv *priv,
+static void rsnd_gen2_dma_addr(struct rsnd_priv *priv,
                        struct rsnd_dma *dma,
                        struct dma_slave_config *cfg,
                        int is_play, int slave_id)
@@ -226,17 +226,6 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv,
                 }
         };
 
-        cfg->slave_id   = slave_id;
-        cfg->src_addr   = 0;
-        cfg->dst_addr   = 0;
-        cfg->direction  = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
-
-        /*
-         * gen1 uses default DMA addr
-         */
-        if (rsnd_is_gen1(priv))
-                return;
-
         /* it shouldn't happen */
         if (use_dvc & !use_src) {
                 dev_err(dev, "DVC is selected without SRC\n");
@@ -250,6 +239,26 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv,
                 id, cfg->src_addr, cfg->dst_addr);
 }
 
+void rsnd_gen_dma_addr(struct rsnd_priv *priv,
+                       struct rsnd_dma *dma,
+                       struct dma_slave_config *cfg,
+                       int is_play, int slave_id)
+{
+        cfg->slave_id   = slave_id;
+        cfg->src_addr   = 0;
+        cfg->dst_addr   = 0;
+        cfg->direction  = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
+
+        /*
+         * gen1 uses default DMA addr
+         */
+        if (rsnd_is_gen1(priv))
+                return;
+
+        rsnd_gen2_dma_addr(priv, dma, cfg, is_play, slave_id);
+}
+
+
 /*
  *              Gen2
  */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b87d7d882e6d..91120b8e283e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -270,12 +270,32 @@ static const struct file_operations codec_reg_fops = {
         .llseek = default_llseek,
 };
 
+static struct dentry *soc_debugfs_create_dir(struct dentry *parent,
+        const char *fmt, ...)
+{
+        struct dentry *de;
+        va_list ap;
+        char *s;
+
+        va_start(ap, fmt);
+        s = kvasprintf(GFP_KERNEL, fmt, ap);
+        va_end(ap);
+
+        if (!s)
+                return NULL;
+
+        de = debugfs_create_dir(s, parent);
+        kfree(s);
+
+        return de;
+}
+
 static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
 {
         struct dentry *debugfs_card_root = codec->card->debugfs_card_root;
 
-        codec->debugfs_codec_root = debugfs_create_dir(codec->name,
-                                                       debugfs_card_root);
+        codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root,
+                                                "codec:%s", codec->name);
         if (!codec->debugfs_codec_root) {
                 dev_warn(codec->dev,
                         "ASoC: Failed to create codec debugfs directory\n");
@@ -306,8 +326,8 @@ static void soc_init_platform_debugfs(struct snd_soc_platform *platform)
 {
         struct dentry *debugfs_card_root = platform->card->debugfs_card_root;
 
-        platform->debugfs_platform_root = debugfs_create_dir(platform->name,
-                                                       debugfs_card_root);
+        platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root,
+                                                "platform:%s", platform->name);
         if (!platform->debugfs_platform_root) {
                 dev_warn(platform->dev,
                         "ASoC: Failed to create platform debugfs directory\n");
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 54d18f22a33e..4ea656770d65 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2069,6 +2069,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
                         dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
                 }
 
+                dpcm_path_put(&list);
 capture:
                 /* skip if FE doesn't have capture capability */
                 if (!fe->cpu_dai->driver->capture.channels_min)