[alsa-devel] [PATCH] ASoC: adau17x1: Workaround for noise bug in ADC

[Ricard Wanderlof]

The ADC in the ADAU1361 (and possibly other Analog Devices codecs) 
exhibits a cyclic variation in the noise floor (in our test setup between 
-87 and -93 dB), a new value being attained within this range whenever a 
new capture stream is started. The cycle repeats after about 10 or 11 
restarts.

The workaround recommended by the manufacturer is to toggle the ADOSR bit 
in the Converter Control 0 register each time a new capture stream is 
started.

I have verified that the patch fixes this problem on the ADAU1361, and 
according to the manufacturer toggling the bit in question in this manner 
will at least have no detrimental effect on other chips served by this 
driver.

Signed-off-by: Ricard Wanderlof <ricardw at axis.com>
---
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 2c1bd27..6758f78 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -90,6 +90,27 @@ static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int adau17x1_adc_fixup(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct adau *adau = snd_soc_codec_get_drvdata(codec);
+
+	/*
+	 * If we are capturing, toggle the ADOSR bit in Converter Control 0 to
+	 * avoid losing SNR (workaround from ADI). This must be done after
+	 * the ADC(s) have been enabled. According to the data sheet, it is
+	 * normally illegal to set this bit when the sampling rate is 96 kHz,
+	 * but according to ADI it is acceptable for this workaround.
+	 */
+	regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0,
+		ADAU17X1_CONVERTER0_ADOSR, ADAU17X1_CONVERTER0_ADOSR);
+	regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0,
+		ADAU17X1_CONVERTER0_ADOSR, 0);
+
+	return 0;
+}
+
 static const char * const adau17x1_mono_stereo_text[] = {
 	"Stereo",
 	"Mono Left Channel (L+R)",
@@ -121,7 +142,8 @@ static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("Right DAC Mode Mux", SND_SOC_NOPM, 0, 0,
 		&adau17x1_dac_mode_mux),
 
-	SND_SOC_DAPM_ADC("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0),
+	SND_SOC_DAPM_ADC_E("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0,
+			   adau17x1_adc_fixup, SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_ADC("Right Decimator", NULL, ADAU17X1_ADC_CONTROL, 1, 0),
 	SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0),
 	SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0),
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index bf04b7ef..db35003 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -129,5 +129,7 @@ bool adau17x1_has_dsp(struct adau *adau);
 
 #define ADAU17X1_CONVERTER0_CONVSR_MASK		0x7
 
+#define ADAU17X1_CONVERTER0_ADOSR		BIT(3)
+
 
 #endif

-- 
Ricard Wolf Wanderlöf                           ricardw(at)axis.com
Axis Communications AB, Lund, Sweden            www.axis.com
Phone +46 46 272 2016                           Fax +46 46 13 61 30