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Message-ID: <s5h1v2errkc.wl%tiwai@suse.de>
Date: Fri, 11 Mar 2011 09:26:27 +0100
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lrg@...mlogic.co.uk>,
Andrew Morton <akpm@...ux-foundation.org>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 2.6.38
Linus,
please pull sound (ASoC) fixes for v2.6.38 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git fix/asoc
containing the following small fixes. Most of them are stable-fixes.
Thanks!
Takashi
===
Abhilash K V (1):
ASoC: AM3517: Update codec name after multi-component update
Mark Brown (4):
ASoC: Fix broken bitfield definitions in WM8978
ASoC: Use the correct DAPM context when cleaning up final widget set
ASoC: Fix typo in late revision WM8994 DAC2R name
ASoC: Ensure WM8958 gets all WM8994 late revision widgets
---
sound/soc/codecs/wm8978.c | 14 ++++++++------
sound/soc/codecs/wm8994.c | 10 +++++++++-
sound/soc/omap/am3517evm.c | 2 +-
sound/soc/soc-dapm.c | 2 +-
4 files changed, 19 insertions(+), 9 deletions(-)
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 4bbc344..8dfb0a0 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -145,18 +145,18 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = {
SOC_SINGLE("DAC Playback Limiter Threshold",
WM8978_DAC_LIMITER_2, 4, 7, 0),
SOC_SINGLE("DAC Playback Limiter Boost",
- WM8978_DAC_LIMITER_2, 0, 15, 0),
+ WM8978_DAC_LIMITER_2, 0, 12, 0),
SOC_ENUM("ALC Enable Switch", alc1),
SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0),
SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0),
- SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0),
+ SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0),
SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0),
SOC_ENUM("ALC Capture Mode", alc3),
- SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0),
- SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0),
+ SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0),
SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0),
SOC_SINGLE("ALC Capture Noise Gate Threshold",
@@ -211,8 +211,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = {
WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1),
/* DAC / ADC oversampling */
- SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0),
- SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0),
+ SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL,
+ 5, 1, 0),
+ SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL,
+ 5, 1, 0),
};
/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 4afbe3b..c6c958e 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1418,7 +1418,7 @@ SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0,
static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = {
SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
-SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
+SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
};
@@ -3325,6 +3325,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
case WM8958:
snd_soc_add_controls(codec, wm8958_snd_controls,
ARRAY_SIZE(wm8958_snd_controls));
+ snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+ ARRAY_SIZE(wm8994_lateclk_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+ ARRAY_SIZE(wm8994_adc_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+ ARRAY_SIZE(wm8994_dac_widgets));
snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
ARRAY_SIZE(wm8958_dapm_widgets));
break;
@@ -3350,6 +3356,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
}
break;
case WM8958:
+ snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+ ARRAY_SIZE(wm8994_lateclk_intercon));
snd_soc_dapm_add_routes(dapm, wm8958_intercon,
ARRAY_SIZE(wm8958_intercon));
break;
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 1617504..73dde4a 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = {
.cpu_dai_name ="omap-mcbsp-dai.0",
.codec_dai_name = "tlv320aic23-hifi",
.platform_name = "omap-pcm-audio",
- .codec_name = "tlv320aic23-codec",
+ .codec_name = "tlv320aic23-codec.2-001a",
.init = am3517evm_aic23_init,
.ops = &am3517evm_ops,
};
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 25e5423..1790f83 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -941,7 +941,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
}
if (!list_empty(&pending))
- dapm_seq_run_coalesced(dapm, &pending);
+ dapm_seq_run_coalesced(cur_dapm, &pending);
}
static void dapm_widget_update(struct snd_soc_dapm_context *dapm)
--
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