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| author | Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-28 14:25:01 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-28 14:25:01 -0700 |
| commit | 68d99b2c8efcb6ed3807a55569300c53b5f88be5 (patch) | |
| tree | f189c8f2132d3668a2f0e503f5c3f8695b26a1c8 /sound/soc/codecs/88pm860x-codec.c | |
| parent | 0e59e7e7feb5a12938fbf9135147eeda3238c6c4 (diff) | |
| parent | 8128c9f21509f9a8b6da94ac432d845dda458406 (diff) | |
| download | linux-stable-68d99b2c8efcb6ed3807a55569300c53b5f88be5.tar.gz linux-stable-68d99b2c8efcb6ed3807a55569300c53b5f88be5.tar.bz2 linux-stable-68d99b2c8efcb6ed3807a55569300c53b5f88be5.zip | |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
Diffstat (limited to 'sound/soc/codecs/88pm860x-codec.c')
| -rw-r--r-- | sound/soc/codecs/88pm860x-codec.c | 14 |
1 files changed, 12 insertions, 2 deletions
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 19241576b6b5..5ca122e51183 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -15,6 +15,7 @@ #include <linux/platform_device.h> #include <linux/mfd/88pm860x.h> #include <linux/slab.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -772,11 +773,12 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), @@ -868,6 +870,11 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left ADC", NULL, "Left ADC MOD"}, {"Right ADC", NULL, "Right ADC MOD"}, + /* I2S Clock */ + {"I2S DIN", NULL, "I2S CLK"}, + {"I2S DIN1", NULL, "I2S CLK"}, + {"I2S DOUT", NULL, "I2S CLK"}, + /* PCM/AIF1 Inputs */ {"PCM SDO", NULL, "ADC Left Mux"}, {"PCM SDO", NULL, "ADCR EC Mux"}, @@ -1173,6 +1180,9 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, REG_MISC2, data); + udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; pm860x_reg_write(codec->control_data, REG_MISC2, data); |