sound fixes for 3.17-final

Just a few pending bits of random fixes in ASoC.  Nothing
 exciting, but would be nice to be merged in 3.17, as most of
 them are also for stable kernels.
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Merge tag 'sound-3.17' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Just a few pending bits of random fixes in ASoC.  Nothing exciting,
  but would be nice to be merged in 3.17, as most of them are also for
  stable kernels"

* tag 'sound-3.17' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: ssm2602: do not hardcode type to SSM2602
  ASoC: core: fix possible ZERO_SIZE_PTR pointer dereferencing error.
  MAINTAINERS: add atmel audio alsa driver maintainer entry
  ASoC: rt286: Fix sync function
  ASoC: rt286: Correct default value
  ASoC: soc-compress: fix double unlock of fe card mutex
  ASoC: fsl_ssi: fix kernel panic in probe function

MAINTAINERS

@@ -1665,6 +1665,12 @@ M:	Nicolas Ferre <nicolas.ferre@atmel.com>
 S:	Supported
 F:	drivers/tty/serial/atmel_serial.c
 
+ATMEL Audio ALSA driver
+M:	Bo Shen <voice.shen@atmel.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Supported
+F:	sound/soc/atmel
+
 ATMEL DMA DRIVER
 M:	Nicolas Ferre <nicolas.ferre@atmel.com>
 L:	linux-arm-kernel@lists.infradead.org (moderated for non-subscribers)

sound/soc/codecs/rt286.c

@@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = {
 	{ 0x04, 0xaf01 },
 	{ 0x08, 0x000d },
 	{ 0x09, 0xd810 },
-	{ 0x0a, 0x0060 },
+	{ 0x0a, 0x0120 },
 	{ 0x0b, 0x0000 },
 	{ 0x0d, 0x2800 },
 	{ 0x0f, 0x0000 },
@@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = {
 	{ 0x33, 0x0208 },
 	{ 0x49, 0x0004 },
 	{ 0x4f, 0x50e9 },
-	{ 0x50, 0x2c00 },
+	{ 0x50, 0x2000 },
 	{ 0x63, 0x2902 },
 	{ 0x67, 0x1111 },
 	{ 0x68, 0x1016 },
@@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = {
 	{ 0x02170700, 0x00000000 },
 	{ 0x02270100, 0x00000000 },
 	{ 0x02370100, 0x00000000 },
-	{ 0x02040000, 0x00004002 },
 	{ 0x01870700, 0x00000020 },
 	{ 0x00830000, 0x000000c3 },
 	{ 0x00930000, 0x000000c3 },
@@ -192,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
 	/*handle index registers*/
 	if (reg <= 0xff) {
 		rt286_hw_write(client, RT286_COEF_INDEX, reg);
-		reg = RT286_PROC_COEF;
 		for (i = 0; i < INDEX_CACHE_SIZE; i++) {
 			if (reg == rt286->index_cache[i].reg) {
 				rt286->index_cache[i].def = value;
@@ -200,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
 			}
 
 		}
+		reg = RT286_PROC_COEF;
 	}
 
 	data[0] = (reg >> 24) & 0xff;

sound/soc/codecs/ssm2602.c

@@ -647,7 +647,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type,
 		return -ENOMEM;
 
 	dev_set_drvdata(dev, ssm2602);
-	ssm2602->type = SSM2602;
+	ssm2602->type = type;
 	ssm2602->regmap = regmap;
 
 	return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,

sound/soc/fsl/fsl_ssi.c

@@ -748,7 +748,8 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
+static int _fsl_ssi_set_dai_fmt(struct device *dev,
+				struct fsl_ssi_private *ssi_private,
 				unsigned int fmt)
 {
 	struct regmap *regs = ssi_private->regs;
@@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
 	ssi_private->dai_fmt = fmt;
 
 	if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
-		dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
+		dev_err(dev, "baudclk is missing which is necessary for master mode\n");
 		return -EINVAL;
 	}
 
@@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 {
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
 
-	return _fsl_ssi_set_dai_fmt(ssi_private, fmt);
+	return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt);
 }
 
 /**
@@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 
 done:
 	if (ssi_private->dai_fmt)
-		_fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
+		_fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private,
+				     ssi_private->dai_fmt);
 
 	return 0;
 

sound/soc/soc-compress.c

@@ -102,13 +102,11 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 	fe->dpcm[stream].runtime = fe_substream->runtime;
 
 	ret = dpcm_path_get(fe, stream, &list);
-	if (ret < 0) {
+	if (ret < 0)
-		mutex_unlock(&fe->card->mutex);
 		goto fe_err;
-	} else if (ret == 0) {
+	else if (ret == 0)
 		dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
 			fe->dai_link->name, stream ? "capture" : "playback");
-	}
 
 	/* calculate valid and active FE <-> BE dpcms */
 	dpcm_process_paths(fe, stream, &list, 1);

sound/soc/soc-core.c

@@ -3203,7 +3203,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 	unsigned int val, mask;
 	void *data;
 
-	if (!component->regmap)
+	if (!component->regmap || !params->num_regs)
 		return -EINVAL;
 
 	len = params->num_regs * component->val_bytes;