[alsa-devel] [PATCH v4] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks

Stefan Müller-Klieser s.mueller-klieser at phytec.de
Wed Feb 14 13:42:13 CET 2018


Hi Peter,

On 14.02.2018 09:13, Peter Ujfalusi wrote:
> 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.
> 

I ran into this issue with an aic3254. I don't know if I am missing something
from your case, but I think the aic's have a special register for that
use case, e.g. for 3254: P0-R29-D2:

Primary BCLK and Primary WCLK Power control
0: Priamry BCLK and Primary WCLK buffers are powered down when the codec is powered down
1: Primary BCLK and Primary WCLK buffers are powered up when they are used in clock
generation even when the codec is powered down

Might this fix your issue?

Regards, Stefan

> 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>
> ---
> Hi,
> 
> 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 858cb8be445f..bd659c803f14 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_codec *codec)
>  {
>  	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_codec *codec,
> +				       unsigned int fmt)
> +{
> +	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
> +	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
> +	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(codec, fmt);
>  }
>  
>  static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai,
> 


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