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-rw-r--r--sound/soc/Makefile5
-rw-r--r--sound/soc/codecs/cs42l52.c3
-rw-r--r--sound/soc/codecs/wm2200.c3
-rw-r--r--sound/soc/codecs/wm5110.c4
-rw-r--r--sound/soc/codecs/wm8978.c2
-rw-r--r--sound/soc/codecs/wm9712.c2
-rw-r--r--sound/soc/codecs/wm_hubs.c5
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c2
-rw-r--r--sound/soc/sh/fsi.c15
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dapm.c2
11 files changed, 30 insertions, 18 deletions
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 00a555a743b..824f66fb556 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,8 +1,9 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
snd-soc-core-objs += soc-pcm.o soc-io.o
-snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o
-obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
+ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
+snd-soc-core-objs += soc-dmaengine-pcm.o
+endif
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 628daf6a1d9..d8cfcc7c2f6 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -774,7 +774,6 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
- int ret = 0;
u8 iface = 0;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -823,7 +822,7 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
case SND_SOC_DAIFMT_NB_IF:
break;
default:
- ret = -EINVAL;
+ return -EINVAL;
}
cs42l52->config.format = iface;
snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 32682c1b7cd..c8bff6da532 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0,
digital_tlv),
SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
- WM2200_SPK1R_MUTE_SHIFT, 1, 0),
+ WM2200_SPK1R_MUTE_SHIFT, 1, 1),
};
WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
@@ -2091,6 +2091,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c,
switch (wm2200->rev) {
case 0:
+ case 1:
ret = regmap_register_patch(wm2200->regmap, wm2200_reva_patch,
ARRAY_SIZE(wm2200_reva_patch));
if (ret != 0) {
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 01ebbcc5c6a..11dfc339048 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -869,6 +869,8 @@ static unsigned int wm5110_digital_vu[] = {
ARIZONA_ADC_DIGITAL_VOLUME_2R,
ARIZONA_ADC_DIGITAL_VOLUME_3L,
ARIZONA_ADC_DIGITAL_VOLUME_3R,
+ ARIZONA_ADC_DIGITAL_VOLUME_4L,
+ ARIZONA_ADC_DIGITAL_VOLUME_4R,
ARIZONA_DAC_DIGITAL_VOLUME_1L,
ARIZONA_DAC_DIGITAL_VOLUME_1R,
@@ -880,6 +882,8 @@ static unsigned int wm5110_digital_vu[] = {
ARIZONA_DAC_DIGITAL_VOLUME_4R,
ARIZONA_DAC_DIGITAL_VOLUME_5L,
ARIZONA_DAC_DIGITAL_VOLUME_5R,
+ ARIZONA_DAC_DIGITAL_VOLUME_6L,
+ ARIZONA_DAC_DIGITAL_VOLUME_6R,
};
static struct snd_soc_codec_driver soc_codec_dev_wm5110 = {
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index a5be3adecf7..2f46d66fb0c 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -782,7 +782,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream,
wm8978->mclk_idx = -1;
f_sel = wm8978->f_mclk;
} else {
- if (!wm8978->f_pllout) {
+ if (!wm8978->f_opclk) {
/* We only enter here, if OPCLK is not used */
int ret = wm8978_configure_pll(codec);
if (ret < 0)
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index c6d2076a796..c9516f0c7a9 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -146,7 +146,7 @@ SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1),
SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
-SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 61baa48823c..57b27efb549 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -634,6 +634,11 @@ void wm_hubs_update_class_w(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8993_CLASS_W_0,
WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ, enable);
+
+ snd_soc_write(codec, WM8993_LEFT_OUTPUT_VOLUME,
+ snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME));
+ snd_soc_write(codec, WM8993_RIGHT_OUTPUT_VOLUME,
+ snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME));
}
EXPORT_SYMBOL_GPL(wm_hubs_update_class_w);
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 9d93793d307..f8fba57f54f 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -190,7 +190,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
- twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
+ twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator");
twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 0540408a9fa..1bb0d58c8c2 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -20,6 +20,7 @@
#include <linux/sh_dma.h>
#include <linux/slab.h>
#include <linux/module.h>
+#include <linux/workqueue.h>
#include <sound/soc.h>
#include <sound/sh_fsi.h>
@@ -223,7 +224,7 @@ struct fsi_stream {
*/
struct dma_chan *chan;
struct sh_dmae_slave slave; /* see fsi_handler_init() */
- struct tasklet_struct tasklet;
+ struct work_struct work;
dma_addr_t dma;
};
@@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data)
snd_pcm_period_elapsed(io->substream);
}
-static void fsi_dma_do_tasklet(unsigned long data)
+static void fsi_dma_do_work(struct work_struct *work)
{
- struct fsi_stream *io = (struct fsi_stream *)data;
+ struct fsi_stream *io = container_of(work, struct fsi_stream, work);
struct fsi_priv *fsi = fsi_stream_to_priv(io);
struct snd_soc_dai *dai;
struct dma_async_tx_descriptor *desc;
@@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data)
* FIXME
*
* In DMAEngine case, codec and FSI cannot be started simultaneously
- * since FSI is using tasklet.
+ * since FSI is using the scheduler work queue.
* Therefore, in capture case, probably FSI FIFO will have got
* overflow error in this point.
* in that case, DMA cannot start transfer until error was cleared.
@@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param)
static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
{
- tasklet_schedule(&io->tasklet);
+ schedule_work(&io->work);
return 0;
}
@@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev
return fsi_stream_probe(fsi, dev);
}
- tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io);
+ INIT_WORK(&io->work, fsi_dma_do_work);
return 0;
}
static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io)
{
- tasklet_kill(&io->tasklet);
+ cancel_work_sync(&io->work);
fsi_stream_stop(fsi, io);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c501af6d8db..8bf05d7a86c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2776,8 +2776,9 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
val = (ucontrol->value.integer.value[0] + min) & mask;
val = val << shift;
- if (snd_soc_update_bits_locked(codec, reg, val_mask, val))
- return err;
+ err = snd_soc_update_bits_locked(codec, reg, val_mask, val);
+ if (err < 0)
+ return err;
if (snd_soc_volsw_is_stereo(mc)) {
val_mask = mask << rshift;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f90139b5f50..c4a08a2b961 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3710,7 +3710,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
{
struct snd_soc_codec *codec;
- list_for_each_entry(codec, &card->codec_dev_list, list) {
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
soc_dapm_shutdown_codec(&codec->dapm);
if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
snd_soc_dapm_set_bias_level(&codec->dapm,
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-16 03:31:34 +0000