Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound sound fixes from Takashi Iwai:
 "As good as nothing exciting here; just a few trivial fixes for various
  ASoC stuff."

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: omap-pcm: Free dma buffers in case of error.
  ASoC: s3c2412-i2s: Fix dai registration
  ASoC: wm8350: Don't use locally allocated codec struct
  ASoC: tlv312aic23: unbreak resume
  ASoC: bf5xx-ssm2602: Set DAI format
  ASoC: core: check of_property_count_strings failure
  ASoC: dt: sgtl5000.txt: Add description for 'reg' field
  ASoC: wm_hubs: Make sure we don't disable differential line outputs

7 files changed, 27 insertions, 17 deletions

diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index df3ac73f8778..b39ad356b92b 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
                 .platform_name = "bfin-i2s-pcm-audio",
                 .codec_name = "ssm2602.0-001b",
                 .ops = &bf5xx_ssm2602_ops,
+                .dai_fmt = BF5XX_SSM2602_DAIFMT,
         },
         {
                 .name = "ssm2602",
@@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
                 .platform_name = "bfin-i2s-pcm-audio",
                 .codec_name = "ssm2602.0-001b",
                 .ops = &bf5xx_ssm2602_ops,
+                .dai_fmt = BF5XX_SSM2602_DAIFMT,
         },
 };
 
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 16d55f91a653..df1e07ffac32 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
                                       enum snd_soc_bias_level level)
 {
-        u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f;
+        u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
 
         switch (level) {
         case SND_SOC_BIAS_ON:
@@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
         case SND_SOC_BIAS_OFF:
                 /* everything off, dac mute, inactive */
                 snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
-                snd_soc_write(codec, TLV320AIC23_PWR, 0xffff);
+                snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
                 break;
         }
         codec->dapm.bias_level = level;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 8c4c9591ec05..aa12c6b6beeb 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -60,7 +60,7 @@ struct wm8350_jack_data {
 };
 
 struct wm8350_data {
-        struct snd_soc_codec codec;
+        struct wm8350 *wm8350;
         struct wm8350_output out1;
         struct wm8350_output out2;
         struct wm8350_jack_data hpl;
@@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv,
                            struct wm8350_jack_data *jack,
                            u16 mask)
 {
-        struct wm8350 *wm8350 = priv->codec.control_data;
+        struct wm8350 *wm8350 = priv->wm8350;
         u16 reg;
         int report;
 
@@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work)
 static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
 {
         struct wm8350_data *priv = data;
-        struct wm8350 *wm8350 = priv->codec.control_data;
+        struct wm8350 *wm8350 = priv->wm8350;
         struct wm8350_jack_data *jack = NULL;
 
         switch (irq - wm8350->irq_base) {
@@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
 static irqreturn_t wm8350_mic_handler(int irq, void *data)
 {
         struct wm8350_data *priv = data;
-        struct wm8350 *wm8350 = priv->codec.control_data;
+        struct wm8350 *wm8350 = priv->wm8350;
         u16 reg;
         int report = 0;
 
@@ -1536,6 +1536,8 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
                 return -ENOMEM;
         snd_soc_codec_set_drvdata(codec, priv);
 
+        priv->wm8350 = wm8350;
+
         for (i = 0; i < ARRAY_SIZE(supply_names); i++)
                 priv->supplies[i].supply = supply_names[i];
 
@@ -1544,7 +1546,6 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
         if (ret != 0)
                 return ret;
 
-        wm8350->codec.codec = codec;
         codec->control_data = wm8350;
 
         /* Put the codec into reset if it wasn't already */
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index f13f2886339c..6c028c470601 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
                             enum snd_soc_bias_level level)
 {
         struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
-        int val;
+        int mask, val;
 
         switch (level) {
         case SND_SOC_BIAS_STANDBY:
@@ -1047,6 +1047,13 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
         case SND_SOC_BIAS_ON:
                 /* Turn off any unneded single ended outputs */
                 val = 0;
+                mask = 0;
+
+                if (hubs->lineout1_se)
+                        mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA;
+
+                if (hubs->lineout2_se)
+                        mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA;
 
                 if (hubs->lineout1_se && hubs->lineout1n_ena)
                         val |= WM8993_LINEOUT1N_ENA;
@@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
                         val |= WM8993_LINEOUT2P_ENA;
 
                 snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3,
-                                    WM8993_LINEOUT1N_ENA |
-                                    WM8993_LINEOUT1P_ENA |
-                                    WM8993_LINEOUT2N_ENA |
-                                    WM8993_LINEOUT2P_ENA,
-                                    val);
+                                    mask, val);
 
                 /* Remove the input clamps */
                 snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG,
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index a59bd352d342..5a649da9122a 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
         }
 
 out:
+        /* free preallocated buffers in case of error */
+        if (ret)
+                omap_pcm_free_dma_buffers(pcm);
+
         return ret;
 }
 
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 72185078ddf8..79fbeea99d46 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = {
 
 static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev)
 {
-        return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai);
+        return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai);
 }
 
 static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1d6a80c9f4c2..c88d9741b9e7 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3625,10 +3625,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
         int i, ret;
 
         num_routes = of_property_count_strings(np, propname);
-        if (num_routes & 1) {
+        if (num_routes < 0 || num_routes & 1) {
                 dev_err(card->dev,
-                        "Property '%s's length is not even\n",
-                        propname);
+                     "Property '%s' does not exist or its length is not even\n",
+                     propname);
                 return -EINVAL;
         }
         num_routes /= 2;