[alsa-devel] [PATCH v5] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks

[Peter Ujfalusi]

In the reset state of the codec we do not have complete playback or capture
routes.

The audio playback/capture will not work due to missing clock signals on
the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down.

To make sure that even if all output/input is disconnected the codec is
generating clocks, we need to have valid DAPM route in every case to power
up the must needed parts of the codec.

I have verified that switching DAC (during playback) or ADC (during
capture) will stop the I2S clocks, so the only solution is to connect the
'Off' routes as well to output/input.

The routes will be only added if the codec is clock master. In case the
role changes runtime, the applied routes are removed.

Tested on am43x-epos-evm with aic3111 codec in master mode.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi at ti.com>
Reviewed-by: Jyri Sarha <jsarha at ti.com>
---
Hi,

CHanges since v4:
- Rebased on top of CODEC to component conversion
- Added Reviewed-by from Jyri

Changes since v3:
- install or remove the master mode DAPM routes if needed
- move the clock master DAPM route 'management' to a separate function

Changes since v2:
- Leftover debug prints removed.

Changes since v1:
- Only apply the master mode DAPM routes when the codec is clock master
- comments added to explain the need.

Regards,
Peter

 sound/soc/codecs/tlv320aic31xx.c | 73 +++++++++++++++++++++++++++++++++++++++-
 1 file changed, 72 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index d3cd924dc300..7090342e8285 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -166,6 +166,7 @@ struct aic31xx_priv {
 	unsigned int sysclk;
 	u8 p_div;
 	int rate_div_line;
+	bool master_dapm_route_applied;
 };
 
 struct aic31xx_rate_divs {
@@ -670,6 +671,29 @@ aic310x_audio_map[] = {
 	{"SPK", NULL, "SPK ClassD"},
 };
 
+/*
+ * Always connected DAPM routes for codec clock master modes.
+ * If the codec is the master on the I2S bus, we need to power on components
+ * to have valid DAC_CLK and also the DACs and ADC for playback/capture.
+ * Otherwise the codec will not generate clocks on the bus.
+ */
+static const struct snd_soc_dapm_route
+common31xx_cm_audio_map[] = {
+	{"DAC Left Input", "Off", "DAC IN"},
+	{"DAC Right Input", "Off", "DAC IN"},
+
+	{"HPL", NULL, "DAC Left"},
+	{"HPR", NULL, "DAC Right"},
+};
+
+static const struct snd_soc_dapm_route
+aic31xx_cm_audio_map[] = {
+	{"MIC1LP P-Terminal", "Off", "MIC1LP"},
+	{"MIC1RP P-Terminal", "Off", "MIC1RP"},
+	{"MIC1LM P-Terminal", "Off", "MIC1LM"},
+	{"MIC1LM M-Terminal", "Off", "MIC1LM"},
+};
+
 static int aic31xx_add_controls(struct snd_soc_component *component)
 {
 	int ret = 0;
@@ -912,6 +936,53 @@ static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute)
 	return 0;
 }
 
+static int aic31xx_clock_master_routes(struct snd_soc_component *component,
+				       unsigned int fmt)
+{
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component);
+	int ret;
+
+	fmt &= SND_SOC_DAIFMT_MASTER_MASK;
+	if (fmt == SND_SOC_DAIFMT_CBS_CFS &&
+	    aic31xx->master_dapm_route_applied) {
+		/*
+		 * Remove the DAPM route(s) for codec clock master modes,
+		 * if applied
+		 */
+		ret = snd_soc_dapm_del_routes(dapm, common31xx_cm_audio_map,
+					ARRAY_SIZE(common31xx_cm_audio_map));
+		if (!ret && !(aic31xx->codec_type & DAC31XX_BIT))
+			ret = snd_soc_dapm_del_routes(dapm,
+					aic31xx_cm_audio_map,
+					ARRAY_SIZE(aic31xx_cm_audio_map));
+
+		if (ret)
+			return ret;
+
+		aic31xx->master_dapm_route_applied = false;
+	} else if (fmt != SND_SOC_DAIFMT_CBS_CFS &&
+		   !aic31xx->master_dapm_route_applied) {
+		/*
+		 * Add the needed DAPM route(s) for codec clock master modes,
+		 * if it is not done already
+		 */
+		ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map,
+					ARRAY_SIZE(common31xx_cm_audio_map));
+		if (!ret && !(aic31xx->codec_type & DAC31XX_BIT))
+			ret = snd_soc_dapm_add_routes(dapm,
+					aic31xx_cm_audio_map,
+					ARRAY_SIZE(aic31xx_cm_audio_map));
+
+		if (ret)
+			return ret;
+
+		aic31xx->master_dapm_route_applied = true;
+	}
+
+	return 0;
+}
+
 static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			       unsigned int fmt)
 {
@@ -992,7 +1063,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			    AIC31XX_BCLKINV_MASK,
 			    iface_reg2);
 
-	return 0;
+	return aic31xx_clock_master_routes(component, fmt);
 }
 
 static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai,
-- 
Peter

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