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-rw-r--r--sound/soc/codecs/ad1836.c14
-rw-r--r--sound/soc/codecs/ad1836.h6
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/tlv320aic26.c14
-rw-r--r--sound/soc/codecs/tlv320aic3x.c9
-rw-r--r--sound/soc/codecs/wm8731.c29
-rw-r--r--sound/soc/codecs/wm8804.c9
-rw-r--r--sound/soc/codecs/wm8915.c3
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/codecs/wm8991.c1
-rw-r--r--sound/soc/codecs/wm8994.c2
11 files changed, 49 insertions, 44 deletions
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index ab63d52..754c496 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- word_len = 3;
+ word_len = AD1836_WORD_LEN_16;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- word_len = 1;
+ word_len = AD1836_WORD_LEN_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
- word_len = 0;
+ word_len = AD1836_WORD_LEN_24;
break;
}
- snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
- AD1836_DAC_WORD_LEN_MASK, word_len);
+ snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK,
+ word_len << AD1836_DAC_WORD_LEN_OFFSET);
- snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
- AD1836_ADC_WORD_LEN_MASK, word_len);
+ snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK,
+ word_len << AD1836_ADC_WORD_OFFSET);
return 0;
}
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 8455967..9d6a3f8 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -25,6 +25,7 @@
#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5)
#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5)
#define AD1836_DAC_WORD_LEN_MASK 0x18
+#define AD1836_DAC_WORD_LEN_OFFSET 3
#define AD1836_DAC_CTRL2 1
#define AD1836_DACL1_MUTE 0
@@ -51,6 +52,7 @@
#define AD1836_ADCL2_MUTE 2
#define AD1836_ADCR2_MUTE 3
#define AD1836_ADC_WORD_LEN_MASK 0x30
+#define AD1836_ADC_WORD_OFFSET 5
#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
@@ -60,4 +62,8 @@
#define AD1836_NUM_REGS 16
+#define AD1836_WORD_LEN_24 0x0
+#define AD1836_WORD_LEN_20 0x1
+#define AD1836_WORD_LEN_16 0x2
+
#endif
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570..65f4604 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
default:
return -EINVAL;
}
- snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+ snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
/* format type */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e2a7608..7859bdc 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
}
- /* Configure PLL */
+ /**
+ * Configure PLL
+ * fsref = (mclk * PLLM) / 2048
+ * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal)
+ */
pval = 1;
- jval = (fsref == 44100) ? 7 : 8;
- dval = (fsref == 44100) ? 5264 : 1920;
+ /* compute J portion of multiplier */
+ jval = fsref / (aic26->mclk / 2048);
+ /* compute fractional DDDD component of multiplier */
+ dval = fsref - (jval * (aic26->mclk / 2048));
+ dval = (10000 * dval) / (aic26->mclk / 2048);
+ dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
qval = 0;
reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index c3d96fc..789453d 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1114,12 +1114,19 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
/* Sync reg_cache with the hardware */
codec->cache_only = 0;
- for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++)
+ for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
snd_soc_write(codec, i, cache[i]);
if (aic3x->model == AIC3X_MODEL_3007)
aic3x_init_3007(codec);
codec->cache_sync = 0;
} else {
+ /*
+ * Do soft reset to this codec instance in order to clear
+ * possible VDD leakage currents in case the supply regulators
+ * remain on
+ */
+ snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
+ codec->cache_sync = 1;
aic3x->power = 0;
/* HW writes are needless when bias is off */
codec->cache_only = 1;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2dc964b..76b4361 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
&wm8731_output_mixer_controls[0],
@@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
static const struct snd_soc_dapm_route wm8731_intercon[] = {
{"DAC", NULL, "OSC", wm8731_check_osc},
{"ADC", NULL, "OSC", wm8731_check_osc},
+ {"DAC", NULL, "ACTIVE"},
+ {"ADC", NULL, "ACTIVE"},
/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
@@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
-
- /* set active */
- snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
-
- return 0;
-}
-
-static void wm8731_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
-
- /* deactivate */
- if (!codec->active) {
- udelay(50);
- snd_soc_write(codec, WM8731_ACTIVE, 0x0);
- }
-}
-
static int wm8731_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
@@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
break;
case SND_SOC_BIAS_OFF:
- snd_soc_write(codec, WM8731_ACTIVE, 0x0);
snd_soc_write(codec, WM8731_PWR, 0xffff);
regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
wm8731->supplies);
@@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops wm8731_dai_ops = {
- .prepare = wm8731_pcm_prepare,
.hw_params = wm8731_hw_params,
- .shutdown = wm8731_shutdown,
.digital_mute = wm8731_mute,
.set_sysclk = wm8731_set_dai_sysclk,
.set_fmt = wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6785688..9a5e67c 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = {
#define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
+#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
static struct snd_soc_dai_driver wm8804_dai = {
.name = "wm8804-spdif",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = WM8804_RATES,
.formats = WM8804_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
+ .rates = WM8804_RATES,
.formats = WM8804_FORMATS,
},
.ops = &wm8804_dai_ops,
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index a0b1a72..e2ab4fa 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
int old;
/* Disable SYSCLK while we reconfigure */
- old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1);
+ old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
WM8915_SYSCLK_ENA, 0);
@@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
break;
case WM8915_FLL_MCLK2:
reg = 1;
+ break;
case WM8915_FLL_DACLRCLK1:
reg = 2;
break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index f90ae42..5e05eed 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
return 0;
/* If the left PGA is enabled hit that VU bit... */
- if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA)
+ if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
reg_cache[WM8962_HPOUTL_VOLUME]);
/* ...otherwise the right. The VU is stereo. */
- if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA)
+ if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
reg_cache[WM8962_HPOUTR_VOLUME]);
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1b..6af23d0 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
-#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c..c2fc035 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
reg);
+
+ msleep(5);
}
wm8994->fll[id].in = freq_in;
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