Merge branch 'fix/asoc' into for-linus

22 files changed, 95 insertions, 45 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index 46e3e6b99220..4f555d8e5346 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5532,6 +5532,7 @@ F:	include/media/*7146*
 
 SAMSUNG AUDIO (ASoC) DRIVERS
 M:      Jassi Brar <jassisinghbrar@gmail.com>
+M:      Sangbeom Kim <sbkim73@samsung.com>
 L:      alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:      Supported
 F:      sound/soc/samsung
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c033cb7..a118a0fb9d81 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
 
         switch (params_rate(params)) {
         case 48000:
-                clk = 12288000;
+                clk = 24576000;
                 break;
         }
 
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5ffe68..eedb6f5e5823 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -27,11 +27,6 @@ struct ad193x_priv {
         int sysclk;
 };
 
-/* ad193x register cache & default register settings */
-static const u8 ad193x_reg[AD193X_NUM_REGS] = {
-        0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
 /*
  * AD193X volume/mute/de-emphasis etc. controls
  */
@@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
         snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
 
         reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
-        reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+        reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+                | (word_len << AD193X_DAC_WORD_LEN_SHFT);
         snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
 
         reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
@@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
         .probe =        ad193x_probe,
-        .reg_cache_default = ad193x_reg,
-        .reg_cache_size = AD193X_NUM_REGS,
-        .reg_word_size = sizeof(u16),
 };
 
 #if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b5497877..cccc2e8e5fbd 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
 #define AD193X_DAC_LEFT_HIGH    (1 << 3)
 #define AD193X_DAC_BCLK_INV     (1 << 7)
 #define AD193X_DAC_CTRL2        0x804
-#define AD193X_DAC_WORD_LEN_MASK        0xC
+#define AD193X_DAC_WORD_LEN_SHFT        3
+#define AD193X_DAC_WORD_LEN_MASK        0x18
 #define AD193X_DAC_MASTER_MUTE  1
 #define AD193X_DAC_CHNL_MUTE    0x805
 #define AD193X_DACL1_MUTE       0
@@ -63,7 +64,7 @@
 #define AD193X_ADC_CTRL1        0x80f
 #define AD193X_ADC_SERFMT_MASK          0x60
 #define AD193X_ADC_SERFMT_STEREO        (0 << 5)
-#define AD193X_ADC_SERFMT_TDM           (1 << 2)
+#define AD193X_ADC_SERFMT_TDM           (1 << 5)
 #define AD193X_ADC_SERFMT_AUX           (2 << 5)
 #define AD193X_ADC_WORD_LEN_MASK        0x3
 #define AD193X_ADC_CTRL2        0x810
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 409d89d1f34c..fbd7eb9e61ce 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
         ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
         if (ret != 0) {
                 dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+                kfree(sta32x);
                 return ret;
         }
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 60d740ebeb5b..1725550c293e 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
         switch (event) {
         case SND_SOC_DAPM_PRE_PMU:
                 if (fll) {
+                        try_wait_for_completion(&wm8962->fll_lock);
+
                         snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
                                             WM8962_FLL_ENA, WM8962_FLL_ENA);
                         if (wm8962->irq) {
@@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
                                             WM8962_BIAS_ENA | 0x180);
 
                         msleep(5);
-
-                        snd_soc_update_bits(codec, WM8962_CLOCKING2,
-                                            WM8962_CLKREG_OVD,
-                                            WM8962_CLKREG_OVD);
                 }
 
                 /* VMID 2*250k */
@@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
         snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
         snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
 
+        try_wait_for_completion(&wm8962->fll_lock);
+
         snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
                             WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
                             WM8962_FLL_ENA, fll1);
@@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec)
          */
         snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
 
+        /* Ensure we have soft control over all registers */
+        snd_soc_update_bits(codec, WM8962_CLOCKING2,
+                            WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
         regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
 
         if (pdata) {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index ab8e9d1aaff0..0cdb9d105671 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = {
 };
 
 static const struct soc_enum sidetone_hpf =
-        SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text);
+        SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
 
 static const char *hpf_mode_text[] = {
         "HiFi", "Custom", "Voice"
@@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
 SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
 SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
 
-SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-
-SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux),
+SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux),
 
 SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0),
@@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
         { "AIF2RX0", NULL, "AIFCLK" },
         { "AIF2RX1", NULL, "AIFCLK" },
 
+        { "AIF1TX0", NULL, "AIFCLK" },
+        { "AIF1TX1", NULL, "AIFCLK" },
+        { "AIF1TX2", NULL, "AIFCLK" },
+        { "AIF1TX3", NULL, "AIFCLK" },
+        { "AIF1TX4", NULL, "AIFCLK" },
+        { "AIF1TX5", NULL, "AIFCLK" },
+
+        { "AIF2TX0", NULL, "AIFCLK" },
+        { "AIF2TX1", NULL, "AIFCLK" },
+
         { "DSP1RXL", NULL, "SYSDSPCLK" },
         { "DSP1RXR", NULL, "SYSDSPCLK" },
         { "DSP2RXL", NULL, "SYSDSPCLK" },
@@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 
         snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
 
+        /* Clear any pending completions (eg, from failed startups) */
+        try_wait_for_completion(&wm8996->fll_lock);
+
         snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
                             WM8996_FLL_ENA, WM8996_FLL_ENA);
 
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 56efa0c1c9a9..099614e16651 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
         res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
         if (!res) {
                 err = -ENODEV;
-                goto fail;
+                goto fail_free_info;
         }
 
         info->mem = request_mem_region(res->start, resource_size(res),
                                        pdev->name);
         if (!info->mem) {
                 err = -EBUSY;
-                goto fail;
+                goto fail_free_info;
         }
 
         info->regs = ioremap(info->mem->start, resource_size(info->mem));
@@ -435,6 +435,7 @@ fail_unmap_mem:
         iounmap(info->regs);
 fail_release_mem:
         release_mem_region(info->mem->start, resource_size(info->mem));
+fail_free_info:
         kfree(info);
 fail:
         return err;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 732208c8c0b4..cb50598338e9 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
                  * assume that device_node pointers are a valid comparison.
                  */
                 np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+                of_node_put(np);
                 if (np == dma_channel_np)
                         return ssi_np;
 
                 np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+                of_node_put(np);
                 if (np == dma_channel_np)
                         return ssi_np;
         }
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a19297959587..358f0baaf71b 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
         }
 
         machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
-        if (!machine_data)
-                return -ENOMEM;
+        if (!machine_data) {
+                ret = -ENOMEM;
+                goto error_alloc;
+        }
 
         machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
         machine_data->dai[0].ops = &mpc8610_hpcd_ops;
@@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
         ret = platform_device_add(sound_device);
         if (ret) {
                 dev_err(&pdev->dev, "platform device add failed\n");
-                goto error;
+                goto error_sound;
         }
         dev_set_drvdata(&pdev->dev, sound_device);
 
@@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 
         return 0;
 
+error_sound:
+        platform_device_unregister(sound_device);
 error:
-        of_node_put(codec_np);
-
-        if (sound_device)
-                platform_device_unregister(sound_device);
-
         kfree(machine_data);
-
+error_alloc:
+        of_node_put(codec_np);
         return ret;
 }
 
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 8fa4d5f8eda1..fcb862eb0c73 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np,
          * dai->platform name should already point to an allocated buffer.
          */
         ret = of_address_to_resource(dma_channel_np, 0, &res);
-        if (ret)
+        if (ret) {
+                of_node_put(dma_channel_np);
                 return ret;
+        }
         snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
                  (unsigned long long) res.start, dma_channel_np->name);
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index a33fc51f363b..8f16cd37c2af 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
         if (!priv->mem) {
                 dev_err(&pdev->dev, "request_mem_region failed\n");
                 err = -EBUSY;
-                goto error;
+                goto error_alloc;
         }
 
         priv->io = ioremap(priv->mem->start, SZ_16K);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 30fe0d0efe1c..0aa475f92efa 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
         }
 
         /* Set codec bias level */
-        ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+        ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
 
         /* Add hook switch - can be used to control the codec from userspace
          * even if line discipline fails */
@@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void)
                         ams_delta_hook_switch_gpios);
 
         /* Keep modem power on */
-        ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+        ams_delta_set_bias_level(&ams_delta_audio_card,
+                                 &ams_delta_audio_card.rtd[0].codec->dapm,
+                                 SND_SOC_BIAS_STANDBY);
 
         platform_device_unregister(cx20442_platform_device);
         platform_device_unregister(ams_delta_audio_platform_device);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index b99091fc34eb..65f980ef2870 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE
         select SND_SAMSUNG_I2S
         select SND_SOC_WM8996
         select SND_SOC_WM9081
+        select SND_SOC_WM1250_EV1
 
 config SND_SOC_SPEYSIDE_WM8962
         tristate "Audio support for Wolfson Speyside with WM8962"
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 241f55d00660..c6c65892294e 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,6 +13,7 @@
  *
  */
 
+#include <linux/types.h>
 #include <linux/gpio.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 1e574a5d440d..bc8c1676459f 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,6 +17,7 @@
  *
  */
 
+#include <linux/types.h>
 #include <linux/gpio.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 0b9eb5f7ec4c..72535f2daaf2 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
         struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
         int ret;
 
+        if (dapm->dev != codec_dai->dev)
+                return 0;
+
         switch (level) {
         case SND_SOC_BIAS_PREPARE:
                 if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
@@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
         struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
         int ret;
 
+        if (dapm->dev != codec_dai->dev)
+                return 0;
+
         switch (level) {
         case SND_SOC_BIAS_STANDBY:
                 ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca27490..b085d8e87574 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 
         if (prefix) {
                 name_len = strlen(long_name) + strlen(prefix) + 2;
-                name = kmalloc(name_len, GFP_ATOMIC);
+                name = kmalloc(name_len, GFP_KERNEL);
                 if (!name)
                         return NULL;
 
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c80589..a62f7dd4ba96 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
 #define snd_soc_16_8_read_i2c NULL
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
+                                          unsigned int r)
+{
+        struct spi_device *spi = codec->control_data;
+
+        const u16 reg = cpu_to_be16(r | 0x100);
+        u8 data;
+        int ret;
+
+        ret = spi_write_then_read(spi, &reg, 2, &data, 1);
+        if (ret < 0)
+                return 0;
+        return data;
+}
+#else
+#define snd_soc_16_8_read_spi NULL
+#endif
+
 static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
                               unsigned int value)
 {
@@ -295,6 +314,7 @@ static struct {
         int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
         unsigned int (*read)(struct snd_soc_codec *, unsigned int);
         unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+        unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
 } io_types[] = {
         {
                 .addr_bits = 4, .data_bits = 12,
@@ -318,6 +338,7 @@ static struct {
                 .addr_bits = 16, .data_bits = 8,
                 .write = snd_soc_16_8_write,
                 .i2c_read = snd_soc_16_8_read_i2c,
+                .spi_read = snd_soc_16_8_read_spi,
         },
         {
                 .addr_bits = 16, .data_bits = 16,
@@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 #ifdef CONFIG_SPI_MASTER
                 codec->hw_write = do_spi_write;
 #endif
+                if (io_types[i].spi_read)
+                        codec->hw_read = io_types[i].spi_read;
 
                 codec->control_data = container_of(codec->dev,
                                                    struct spi_device,
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98d5846..38b00131b2fe 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
                                               IRQF_TRIGGER_FALLING,
                                               gpios[i].name,
                                               &gpios[i]);
-                if (ret)
+                if (ret < 0)
                         goto err;
 
                 if (gpios[i].wake) {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b5759397afa3..2879c883eebc 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
         codec_dai->active--;
         codec->active--;
 
+        if (!cpu_dai->active && !codec_dai->active)
+                rtd->rate = 0;
+
         /* Muting the DAC suppresses artifacts caused during digital
          * shutdown, for example from stopping clocks.
          */
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 661373c2352a..be27f1d229af 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
         snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
 
         /* FIXME: Calculate automatically based on DAPM routes? */
-        if (!machine_is_harmony() && !machine_is_ventana())
+        if (!machine_is_harmony())
                 snd_soc_dapm_nc_pin(dapm, "IN1L");
         if (!machine_is_seaboard() && !machine_is_aebl())
                 snd_soc_dapm_nc_pin(dapm, "IN1R");
@@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
         platform_set_drvdata(pdev, card);
         snd_soc_card_set_drvdata(card, machine);
 
-        if (machine_is_harmony() || machine_is_ventana()) {
+        if (machine_is_harmony()) {
                 card->dapm_routes = harmony_audio_map;
                 card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
         } else if (machine_is_seaboard()) {