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

Author: torvalds

Documentation/devicetree/bindings/sound/sgtl5000.txt

@@ -3,6 +3,8 @@
 Required properties:
 - compatible : "fsl,sgtl5000".
 
+- reg : the I2C address of the device
+
 Example:
 
 codec: sgtl5000@0a {

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,
 	},
 };
 

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;

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 */

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,

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;
 }
 

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)

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;