summaryrefslogtreecommitdiff
path: root/sound
diff options
context:
space:
mode:
authorLinus Torvalds <torvalds@linux-foundation.org>2012-05-05 10:07:06 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2012-05-05 10:07:06 -0700
commit1c2f95480648ed7326ab2288ca0e2d35551db4be (patch)
treefa69d267423242eaad195e60c74570152e6c3d84 /sound
parent59068e369b6a2a0a15b93624887525d9ec0f36e5 (diff)
parente9e7183fd2677aca24e90ca1556d4afe7436d42d (diff)
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
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/blackfin/bf5xx-ssm2602.c2
-rw-r--r--sound/soc/codecs/tlv320aic23.c4
-rw-r--r--sound/soc/codecs/wm8350.c11
-rw-r--r--sound/soc/codecs/wm_hubs.c15
-rw-r--r--sound/soc/omap/omap-pcm.c4
-rw-r--r--sound/soc/samsung/s3c2412-i2s.c2
-rw-r--r--sound/soc/soc-core.c6
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;