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>

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 29dc0ec..4f0bf26 100644 (file)
--- 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)