* [GIT PULL] sound fixes for 3.0
@ 2011-07-18 14:11 Takashi Iwai
0 siblings, 0 replies; 2+ messages in thread
From: Takashi Iwai @ 2011-07-18 14:11 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
Linus,
please pull the last sound fixes for v3.0 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/asoc
containing only a few trivial ASoC fixes as below.
Thanks!
Takashi
===
Kuninori Morimoto (3):
ASoC: sh: fsi-ak4642: fixup snd_soc_card name
ASoC: sh: fsi-da7210: fixup snd_soc_card name
ASoC: sh: fsi-hdmi: fixup snd_soc_card name
Mark Brown (2):
ASoC: Fix shift in WM8958 accessory detection default implementation
ASoC: Correct WM8994 MICBIAS supply widget hookup
---
sound/soc/codecs/wm8994.c | 9 +++++----
sound/soc/sh/fsi-ak4642.c | 16 ++++++++--------
sound/soc/sh/fsi-da7210.c | 2 +-
sound/soc/sh/fsi-hdmi.c | 4 ++--
4 files changed, 16 insertions(+), 15 deletions(-)
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index c2fc035..83014a7 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[] = {
@@ -2763,7 +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:
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index d6f4703..770a71a 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -97,7 +97,7 @@ static int fsi_ak4642_remove(struct platform_device *pdev)
static struct fsi_ak4642_data fsi_a_ak4642 = {
.name = "AK4642",
- .card = "FSIA (AK4642)",
+ .card = "FSIA-AK4642",
.cpu_dai = "fsia-dai",
.codec = "ak4642-codec.0-0012",
.platform = "sh_fsi.0",
@@ -106,7 +106,7 @@ static struct fsi_ak4642_data fsi_a_ak4642 = {
static struct fsi_ak4642_data fsi_b_ak4642 = {
.name = "AK4642",
- .card = "FSIB (AK4642)",
+ .card = "FSIB-AK4642",
.cpu_dai = "fsib-dai",
.codec = "ak4642-codec.0-0012",
.platform = "sh_fsi.0",
@@ -115,7 +115,7 @@ static struct fsi_ak4642_data fsi_b_ak4642 = {
static struct fsi_ak4642_data fsi_a_ak4643 = {
.name = "AK4643",
- .card = "FSIA (AK4643)",
+ .card = "FSIA-AK4643",
.cpu_dai = "fsia-dai",
.codec = "ak4642-codec.0-0013",
.platform = "sh_fsi.0",
@@ -124,7 +124,7 @@ static struct fsi_ak4642_data fsi_a_ak4643 = {
static struct fsi_ak4642_data fsi_b_ak4643 = {
.name = "AK4643",
- .card = "FSIB (AK4643)",
+ .card = "FSIB-AK4643",
.cpu_dai = "fsib-dai",
.codec = "ak4642-codec.0-0013",
.platform = "sh_fsi.0",
@@ -133,7 +133,7 @@ static struct fsi_ak4642_data fsi_b_ak4643 = {
static struct fsi_ak4642_data fsi2_a_ak4642 = {
.name = "AK4642",
- .card = "FSI2A (AK4642)",
+ .card = "FSI2A-AK4642",
.cpu_dai = "fsia-dai",
.codec = "ak4642-codec.0-0012",
.platform = "sh_fsi2",
@@ -142,7 +142,7 @@ static struct fsi_ak4642_data fsi2_a_ak4642 = {
static struct fsi_ak4642_data fsi2_b_ak4642 = {
.name = "AK4642",
- .card = "FSI2B (AK4642)",
+ .card = "FSI2B-AK4642",
.cpu_dai = "fsib-dai",
.codec = "ak4642-codec.0-0012",
.platform = "sh_fsi2",
@@ -151,7 +151,7 @@ static struct fsi_ak4642_data fsi2_b_ak4642 = {
static struct fsi_ak4642_data fsi2_a_ak4643 = {
.name = "AK4643",
- .card = "FSI2A (AK4643)",
+ .card = "FSI2A-AK4643",
.cpu_dai = "fsia-dai",
.codec = "ak4642-codec.0-0013",
.platform = "sh_fsi2",
@@ -160,7 +160,7 @@ static struct fsi_ak4642_data fsi2_a_ak4643 = {
static struct fsi_ak4642_data fsi2_b_ak4643 = {
.name = "AK4643",
- .card = "FSI2B (AK4643)",
+ .card = "FSI2B-AK4643",
.cpu_dai = "fsib-dai",
.codec = "ak4642-codec.0-0013",
.platform = "sh_fsi2",
diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c
index dbafd7a..59553fd 100644
--- a/sound/soc/sh/fsi-da7210.c
+++ b/sound/soc/sh/fsi-da7210.c
@@ -42,7 +42,7 @@ static struct snd_soc_dai_link fsi_da7210_dai = {
};
static struct snd_soc_card fsi_soc_card = {
- .name = "FSI (DA7210)",
+ .name = "FSI-DA7210",
.dai_link = &fsi_da7210_dai,
.num_links = 1,
};
diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c
index 9719985..d3d9fd8 100644
--- a/sound/soc/sh/fsi-hdmi.c
+++ b/sound/soc/sh/fsi-hdmi.c
@@ -83,13 +83,13 @@ static int fsi_hdmi_remove(struct platform_device *pdev)
static struct fsi_hdmi_data fsi2_a_hdmi = {
.cpu_dai = "fsia-dai",
- .card = "FSI2A (SH MOBILE HDMI)",
+ .card = "FSI2A-HDMI",
.id = FSI_PORT_A,
};
static struct fsi_hdmi_data fsi2_b_hdmi = {
.cpu_dai = "fsib-dai",
- .card = "FSI2B (SH MOBILE HDMI)",
+ .card = "FSI2B-HDMI",
.id = FSI_PORT_B,
};
^ permalink raw reply [flat|nested] 2+ messages in thread* [GIT PULL] sound fixes for 3.0
@ 2011-07-10 7:43 Takashi Iwai
0 siblings, 0 replies; 2+ messages in thread
From: Takashi Iwai @ 2011-07-10 7:43 UTC (permalink / raw)
To: Linus Torvalds; +Cc: Mark Brown, Liam Girdwood, linux-kernel
Linus,
please pull sound fixes for v3.0 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
All fixes are fairly small and trivial.
Thanks!
Takashi
===
Jarkko Nikula (2):
ASoC: tlv320aic3x: Don't sync first two registers from register cache
ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state
Kuninori Morimoto (1):
ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
Mark Brown (3):
ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values
ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting
ASoC: Manage WM8731 ACTIVE bit as a supply widget
Michael Williamson (1):
audio: tlv320aic26: fix PLL register configuration
Stephen Warren (1):
ASoC: Tegra: I2S: Ensure clock is enabled when writing regs
Takashi Iwai (3):
ASoC: Don't set invalid name string to snd_card->driver field
ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek
ALSA: hda - Fix a copmile warning
---
sound/pci/hda/patch_realtek.c | 33 +++++++++++++++++++++++----------
sound/soc/blackfin/bf5xx-i2s-pcm.c | 13 +++++++++++--
sound/soc/codecs/ak4642.c | 2 +-
sound/soc/codecs/tlv320aic26.c | 14 +++++++++++---
sound/soc/codecs/tlv320aic3x.c | 9 ++++++++-
sound/soc/codecs/wm8731.c | 29 +++--------------------------
sound/soc/codecs/wm8994.c | 2 ++
sound/soc/soc-core.c | 5 +++--
sound/soc/tegra/tegra_i2s.c | 6 ++++++
9 files changed, 68 insertions(+), 45 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d21191d..b48fb43 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol,
- getput_call_t func)
+ getput_call_t func, bool check_adc_switch)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- int err;
+ int i, err = 0;
mutex_lock(&codec->control_mutex);
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
- 3, 0, HDA_INPUT);
- err = func(kcontrol, ucontrol);
+ if (check_adc_switch && spec->dual_adc_switch) {
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+ 3, 0, HDA_INPUT);
+ err = func(kcontrol, ucontrol);
+ if (err < 0)
+ goto error;
+ }
+ } else {
+ i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ kcontrol->private_value =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+ 3, 0, HDA_INPUT);
+ err = func(kcontrol, ucontrol);
+ }
+ error:
mutex_unlock(&codec->control_mutex);
return err;
}
@@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_volume_get);
+ snd_hda_mixer_amp_volume_get, false);
}
static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_volume_put);
+ snd_hda_mixer_amp_volume_put, true);
}
/* capture mixer elements */
@@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_switch_get);
+ snd_hda_mixer_amp_switch_get, false);
}
static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return alc_cap_getput_caller(kcontrol, ucontrol,
- snd_hda_mixer_amp_switch_put);
+ snd_hda_mixer_amp_switch_put, true);
}
#define _DEFINE_CAPMIX(num) \
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index b5101ef..f1fd95b 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
pr_debug("%s enter\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
diff = sport_curr_offset_tx(sport);
- frames = bytes_to_frames(substream->runtime, diff);
} else {
diff = sport_curr_offset_rx(sport);
- frames = bytes_to_frames(substream->runtime, diff);
}
+
+ /*
+ * TX at least can report one frame beyond the end of the
+ * buffer if we hit the wraparound case - clamp to within the
+ * buffer as the ALSA APIs require.
+ */
+ if (diff == snd_pcm_lib_buffer_bytes(substream))
+ diff = 0;
+
+ frames = bytes_to_frames(substream->runtime, diff);
+
return frames;
}
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/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;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043e..b194be0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->long_name ? card->long_name : card->name);
- snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
- "%s", card->driver_name ? card->driver_name : card->name);
+ if (card->driver_name)
+ strlcpy(card->snd_card->driver, card->driver_name,
+ sizeof(card->snd_card->driver));
if (card->late_probe) {
ret = card->late_probe(card);
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e2..95f03c1 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
if (i2sclock % (2 * srate))
reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
+ if (!i2s->clk_refs)
+ clk_enable(i2s->clk_i2s);
+
tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+ if (!i2s->clk_refs)
+ clk_disable(i2s->clk_i2s);
+
return 0;
}
^ permalink raw reply [flat|nested] 2+ messages in thread
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