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Commit ee15ffdb authored by Jarkko Nikula's avatar Jarkko Nikula Committed by Jaroslav Kysela
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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: default avatarJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: default avatarLiam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
Signed-off-by: default avatarJaroslav Kysela <perex@perex.cz>
parent 81971a14

sound/soc/codecs/tlv320aic3x.c

+31 −0
Original line number Original line Diff line number Diff line
@@ -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,

	SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,

			 &aic3x_right_line2_mux_controls),

			 &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 */

	/* Mic Bias */

	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V",

	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V",

			 MICBIAS_CTRL, 6, 3, 1, 0),

			 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 PGA Mixer", "Mic3L Switch", "MIC3L"},





	{"Left ADC", NULL, "Left PGA Mixer"},

	{"Left ADC", NULL, "Left PGA Mixer"},

	{"Left ADC", NULL, "GPIO1 dmic modclk"},





	/* Right Input */

	/* Right Input */

	{"Right Line1R Mux", "single-ended", "LINE1R"},

	{"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 PGA Mixer", "Mic3R Switch", "MIC3R"},





	{"Right ADC", NULL, "Right PGA Mixer"},

	{"Right ADC", NULL, "Right PGA Mixer"},

	{"Right ADC", NULL, "GPIO1 dmic modclk"},





	/* Left PGA Bypass */

	/* Left PGA Bypass */

	{"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"},

	{"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"},

	{"Right Line Out", NULL, "Right Line2 Bypass Mixer"},

	{"Mono Out", NULL, "Right Line2 Bypass Mixer"},

	{"Mono Out", NULL, "Right Line2 Bypass Mixer"},

	{"Right HP 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)

static int aic3x_add_widgets(struct snd_soc_codec *codec)