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-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/wm8994.c11
5 files changed, 30 insertions, 35 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570e3f1f..65f46047b1cb 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 e2a7608d3944..7859bdcc93db 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 c3d96fc8c267..789453d44ec5 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 2dc964b55e4f..76b4361e9b80 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/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360b..83014a7c2e14 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1190,7 +1190,6 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"),
SND_SOC_DAPM_INPUT("DMIC2DAT"),
SND_SOC_DAPM_INPUT("Clock"),
-SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0),
SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
SND_SOC_DAPM_PRE_PMU),
@@ -1509,8 +1508,10 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
{ "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
{ "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
- { "MICBIAS", NULL, "CLK_SYS" },
- { "MICBIAS", NULL, "MICBIAS Supply" },
+ { "MICBIAS1", NULL, "CLK_SYS" },
+ { "MICBIAS1", NULL, "MICBIAS Supply" },
+ { "MICBIAS2", NULL, "CLK_SYS" },
+ { "MICBIAS2", NULL, "MICBIAS Supply" },
};
static const struct snd_soc_dapm_route wm8994_intercon[] = {
@@ -1713,6 +1714,8 @@ 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;
@@ -2761,7 +2764,7 @@ static void wm8958_default_micdet(u16 status, void *data)
report = SND_JACK_MICROPHONE;
/* Everything else is buttons; just assign slots */
- if (status & 0x1c0)
+ if (status & 0x1c)
report |= SND_JACK_BTN_0;
done:
generated by cgit v1.2.3 (git 2.0.4) at 2025-01-23 14:45:47 +0000