[GIT PULL] sound fixes for 3.17
From: Takashi Iwai
Date: Thu Oct 02 2014 - 04:14:37 EST
Linus,
please pull sound fixes for v3.17 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-3.17
The topmost commit is b7c2dc8e684a37d8f9544b9e528c050f2bb7ef47
----------------------------------------------------------------
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.
----------------------------------------------------------------
Bard Liao (2):
ASoC: rt286: Correct default value
ASoC: rt286: Fix sync function
Bo Shen (1):
MAINTAINERS: add atmel audio alsa driver maintainer entry
Michael Trimarchi (1):
ASoC: fsl_ssi: fix kernel panic in probe function
Qiao Zhou (1):
ASoC: soc-compress: fix double unlock of fe card mutex
Stefan Kristiansson (1):
ASoC: ssm2602: do not hardcode type to SSM2602
Xiubo Li (1):
ASoC: core: fix possible ZERO_SIZE_PTR pointer dereferencing error.
---
MAINTAINERS | 6 ++++++
sound/soc/codecs/rt286.c | 7 +++----
sound/soc/codecs/ssm2602.c | 2 +-
sound/soc/fsl/fsl_ssi.c | 12 +++++++-----
sound/soc/soc-compress.c | 6 ++----
sound/soc/soc-core.c | 2 +-
6 files changed, 20 insertions(+), 15 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index 8fd05d4e3b0a..c3cfa1be846e 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -1665,6 +1665,12 @@ M: Nicolas Ferre <nicolas.ferre@xxxxxxxxx>
S: Supported
F: drivers/tty/serial/atmel_serial.c
+ATMEL Audio ALSA driver
+M: Bo Shen <voice.shen@xxxxxxxxx>
+L: alsa-devel@xxxxxxxxxxxxxxxx (moderated for non-subscribers)
+S: Supported
+F: sound/soc/atmel
+
ATMEL DMA DRIVER
M: Nicolas Ferre <nicolas.ferre@xxxxxxxxx>
L: linux-arm-kernel@xxxxxxxxxxxxxxxxxxx (moderated for non-subscribers)
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index e4f6102efc1a..b86b426f159d 100644
--- a/sound/soc/codecs/rt286.c
+++ b/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;
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 484b3bbe8624..4021cd435740 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/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,
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 87eb5776a39b..de6ab06f58a5 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -748,8 +748,9 @@ 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,
- unsigned int fmt)
+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;
u32 strcr = 0, stcr, srcr, scr, mask;
@@ -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;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3092b58fede6..cecfab3cc948 100644
--- a/sound/soc/soc-compress.c
+++ b/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) {
- mutex_unlock(&fe->card->mutex);
+ if (ret < 0)
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);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 889f4e3d35dc..d074aa91b023 100644
--- a/sound/soc/soc-core.c
+++ b/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;
--
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