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@xxxxxx>
> ---
> 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,
>
_______________________________________________
Alsa-devel mailing list
Alsa-devel@xxxxxxxxxxxxxxxx
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
