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-rw-r--r--sound/soc/Makefile5
-rw-r--r--sound/soc/codecs/sigmadsp.c2
-rw-r--r--sound/soc/codecs/wm2000.c6
-rw-r--r--sound/soc/codecs/wm2200.c18
-rw-r--r--sound/soc/codecs/wm5100.c6
-rw-r--r--sound/soc/codecs/wm8978.c2
-rw-r--r--sound/soc/codecs/wm9712.c21
-rw-r--r--sound/soc/omap/mcbsp.c2
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c2
-rw-r--r--sound/soc/samsung/dma.c8
-rw-r--r--sound/soc/sh/fsi.c15
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/soc/tegra/tegra_alc5632.c1
13 files changed, 43 insertions, 47 deletions
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 2feaf376e94b..0f370861bbae 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/sigmadsp.c b/sound/soc/codecs/sigmadsp.c
index 5be42bf56996..4068f2491232 100644
--- a/sound/soc/codecs/sigmadsp.c
+++ b/sound/soc/codecs/sigmadsp.c
@@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware);
static int sigma_action_write_regmap(void *control_data,
const struct sigma_action *sa, size_t len)
{
- return regmap_raw_write(control_data, le16_to_cpu(sa->addr),
+ return regmap_raw_write(control_data, be16_to_cpu(sa->addr),
sa->payload, len - 2);
}
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index a75c3766aede..bb9f07037485 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -190,9 +190,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
if (wm2000->speech_clarity)
- ret &= ~WM2000_SPEECH_CLARITY;
- else
ret |= WM2000_SPEECH_CLARITY;
+ else
+ ret &= ~WM2000_SPEECH_CLARITY;
wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
@@ -692,7 +692,7 @@ static int wm2000_resume(struct snd_soc_codec *codec)
#endif
static const struct regmap_config wm2000_regmap = {
- .reg_bits = 8,
+ .reg_bits = 16,
.val_bits = 8,
};
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 32682c1b7cde..9932aacaa5b8 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -897,8 +897,6 @@ static const char *wm2200_mixer_texts[] = {
"EQR",
"LHPF1",
"LHPF2",
- "LHPF3",
- "LHPF4",
"DSP1.1",
"DSP1.2",
"DSP1.3",
@@ -931,7 +929,6 @@ static int wm2200_mixer_values[] = {
0x25,
0x50, /* EQ */
0x51,
- 0x52,
0x60, /* LHPF1 */
0x61, /* LHPF2 */
0x68, /* DSP1 */
@@ -993,9 +990,9 @@ SOC_DOUBLE_R_TLV("IN3 Volume", WM2200_IN3L_CONTROL, WM2200_IN3R_CONTROL,
SOC_DOUBLE_R("IN1 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L,
WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L,
+SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_2L,
WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L,
+SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_3L,
WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_MUTE_SHIFT, 1, 1),
SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_1L,
@@ -1028,7 +1025,7 @@ 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);
@@ -1380,15 +1377,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
fmt_val = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- fmt_val = 1;
- break;
case SND_SOC_DAIFMT_I2S:
fmt_val = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- fmt_val = 3;
- break;
default:
dev_err(codec->dev, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -1440,7 +1431,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
lrclk);
snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
- WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+ WM2200_AIF1_FMT_MASK, fmt_val);
return 0;
}
@@ -2091,6 +2082,7 @@ 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/wm5100.c b/sound/soc/codecs/wm5100.c
index b9c185ce64e4..a8d03ab5ea2e 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1296,15 +1296,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
mask = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- mask = 1;
- break;
case SND_SOC_DAIFMT_I2S:
mask = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- mask = 3;
- break;
default:
dev_err(codec->dev, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 72d5fdcd3cc2..6c37c7c2327e 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -783,7 +783,7 @@ 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 b342ae50bcd6..757a52aed804 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, 1),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
@@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]);
/* Mic select */
static const struct snd_kcontrol_new wm9712_mic_src_controls =
-SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
/* diff select */
static const struct snd_kcontrol_new wm9712_diff_sel_controls =
@@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectl_controls),
SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectr_controls),
-SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
&wm9712_mic_src_controls),
SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
&wm9712_diff_sel_controls),
@@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
SND_SOC_DAPM_OUTPUT("MONOOUT"),
SND_SOC_DAPM_OUTPUT("HPOUTL"),
@@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = {
{"Mic PGA", NULL, "MIC1"},
{"Mic PGA", NULL, "MIC2"},
+ /* microphones */
+ {"Differential Mic", NULL, "MIC1"},
+ {"Differential Mic", NULL, "MIC2"},
+ {"Left Mic Select Source", "Mic 1", "MIC1"},
+ {"Left Mic Select Source", "Mic 2", "MIC2"},
+ {"Left Mic Select Source", "Stereo", "MIC1"},
+ {"Left Mic Select Source", "Differential", "Differential Mic"},
+ {"Right Mic Select Source", "Mic 1", "MIC1"},
+ {"Right Mic Select Source", "Mic 2", "MIC2"},
+ {"Right Mic Select Source", "Stereo", "MIC2"},
+ {"Right Mic Select Source", "Differential", "Differential Mic"},
+
/* left capture selector */
{"Left Capture Select", "Mic", "MIC1"},
{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index e5f44440d1b9..fb67772130b5 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -691,7 +691,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux)
{
const char *signal, *src;
- if (mcbsp->pdata->mux_signal)
+ if (!mcbsp->pdata->mux_signal)
return -EINVAL;
switch (mux) {
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 93bb8eee22b3..9c2f090167cc 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -181,7 +181,7 @@ 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/samsung/dma.c b/sound/soc/samsung/dma.c
index ddc6cde14e2a..2526ecada5f1 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME,
+ SNDRV_PCM_INFO_MMAP_VALID,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_LE |
SNDRV_PCM_FMTBIT_U8 |
@@ -246,15 +244,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->state |= ST_RUNNING;
prtd->params->ops->trigger(prtd->params->ch);
break;
case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
prtd->state &= ~ST_RUNNING;
prtd->params->ops->stop(prtd->params->ch);
break;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 74ed2dffbffd..91b728774dba 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>
@@ -199,7 +200,7 @@ 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;
};
@@ -968,9 +969,9 @@ static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
}
-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 dma_chan *chan;
struct snd_soc_dai *dai;
@@ -1023,7 +1024,7 @@ 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.
@@ -1047,7 +1048,7 @@ 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;
}
@@ -1087,14 +1088,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)
if (!io->chan)
return -EIO;
- 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-dapm.c b/sound/soc/soc-dapm.c
index f9aa9b50f82b..310ee4addc69 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3276,7 +3276,7 @@ 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,
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index e45ccd851f6a..76d759ec40a2 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -95,7 +95,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = {
.name = "Headset detection",
.report = SND_JACK_HEADSET,
.debounce_time = 150,
- .invert = 1,
};
static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = {
generated by cgit v1.2.3 (git 2.0.4) at 2024-09-18 11:44:31 +0000