[alsa-devel] [PATCH v5] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
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@ti.com Reviewed-by: Jyri Sarha jsarha@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,
The patch
ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
From d460b3f861e18b9c826abe178b2db57c6dc6b3e4 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi peter.ujfalusi@ti.com Date: Wed, 14 Feb 2018 14:20:56 +0200 Subject: [PATCH] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
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@ti.com Reviewed-by: Jyri Sarha jsarha@ti.com Signed-off-by: Mark Brown broonie@kernel.org --- 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,
participants (2)
-
Mark Brown -
Peter Ujfalusi