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| author | Jarkko Nikula <jarkko.nikula@nokia.com> | 2008-06-25 14:58:46 +0300 |
|---|---|---|
| committer | Jaroslav Kysela <perex@perex.cz> | 2008-06-26 09:02:21 +0200 |
| commit | ee15ffdb140b1a4fda5f6a3a8b9994031de5edb4 (patch) | |
| tree | 2ef5be4fa70998bf9eb215a36f6dffa9aafe4116 /sound/soc/codecs/tlv320aic3x.c | |
| parent | 81971a14947b344ee7f6e3781f7ef36d8996f57a (diff) | |
| download | linux-ee15ffdb140b1a4fda5f6a3a8b9994031de5edb4.tar.gz linux-ee15ffdb140b1a4fda5f6a3a8b9994031de5edb4.tar.bz2 linux-ee15ffdb140b1a4fda5f6a3a8b9994031de5edb4.zip | |
ALSA: ASoC: TLV320AIC3X: Add support for digital microphone input
AIC33 and AIC34 codecs in TLV320AIC3x family support digital microphone
input. When enabled, the codec ADC takes bitstream input to low-pass
filter from GPIO2 instead of its own delta-sigma modulator while providing
oversampling clock through GPIO1.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Diffstat (limited to 'sound/soc/codecs/tlv320aic3x.c')
| -rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 31 |
1 files changed, 31 insertions, 0 deletions
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 29dc0ec3dd23..4f0bf26f170b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -455,6 +455,27 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), + /* + * Not a real mic bias widget but similar function. This is for dynamic + * control of GPIO1 digital mic modulator clock output function when + * using digital mic. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk", + AIC3X_GPIO1_REG, 4, 0xf, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK, + AIC3X_GPIO1_FUNC_DISABLED), + + /* + * Also similar function like mic bias. Selects digital mic with + * configurable oversampling rate instead of ADC converter. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128", + AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64", + AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", + AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), + /* Mic Bias */ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", MICBIAS_CTRL, 6, 3, 1, 0), @@ -570,6 +591,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, {"Left ADC", NULL, "Left PGA Mixer"}, + {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -583,6 +605,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, + {"Right ADC", NULL, "GPIO1 dmic modclk"}, /* Left PGA Bypass */ {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, @@ -643,6 +666,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line Out", NULL, "Right Line2 Bypass Mixer"}, {"Mono Out", NULL, "Right Line2 Bypass Mixer"}, {"Right HP Out", NULL, "Right Line2 Bypass Mixer"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, }; static int aic3x_add_widgets(struct snd_soc_codec *codec) |