The patch ASoC: ssm2602: Fix ADC powerup sequencing 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 37768e3917405e82611af8e693ccac1a20df38b5 Mon Sep 17 00:00:00 2001 From: Philipp Zabel <p.zabel@xxxxxxxxxxxxxx> Date: Wed, 6 Feb 2019 16:29:47 +0100 Subject: [PATCH] ASoC: ssm2602: Fix ADC powerup sequencing According to the ssm2603 data sheet (control register sequencing), the digital core should be activated only after all necessary bits in the power register are enabled, and a delay determined by the decoupling capacitor on the VMID pin has passed. If the digital core is activated too early, or even before the ADC is powered up, audible artifacts appear at the beginning of the recorded signal. The digital core is also needed for playback, so when recording starts it may already be enabled. This means we cannot get the power sequence correct when we want to be able to start recording after playback. As a workaround put the MIC mute switch into the DAPM routes. This way we can keep the recording disabled until the MIC Bias has settled and thus get rid of audible artifacts. Signed-off-by: Philipp Zabel <p.zabel@xxxxxxxxxxxxxx> m.felsch@xxxxxxxxxxxxxx: adapt commit message m.felsch@xxxxxxxxxxxxxx: drop of configuration as mentioned by Mark: https://patchwork.kernel.org/patch/10407449/ Signed-off-by: Marco Felsch <m.felsch@xxxxxxxxxxxxxx> Signed-off-by: Mark Brown <broonie@xxxxxxxxxx> --- sound/soc/codecs/ssm2602.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 501a4e73b185..6f79cb0fbc63 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -26,6 +26,7 @@ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA */ +#include <linux/delay.h> #include <linux/module.h> #include <linux/regmap.h> #include <linux/slab.h> @@ -111,7 +112,6 @@ SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1, SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0), -SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), }; /* Output Mixer */ @@ -121,10 +121,31 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), }; +static const struct snd_kcontrol_new mic_ctl = + SOC_DAPM_SINGLE("Switch", SSM2602_APANA, 1, 1, 1); + /* Input mux */ static const struct snd_kcontrol_new ssm2602_input_mux_controls = SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); +static int ssm2602_mic_switch_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* + * According to the ssm2603 data sheet (control register sequencing), + * the digital core should be activated only after all necessary bits + * in the power register are enabled, and a delay determined by the + * decoupling capacitor on the VMID pin has passed. If the digital core + * is activated too early, or even before the ADC is powered up, audible + * artifacts appear at the beginning and end of the recorded signal. + * + * In practice, audible artifacts disappear well over 500 ms. + */ + msleep(500); + + return 0; +} + static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), @@ -146,6 +167,9 @@ SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), +SND_SOC_DAPM_SWITCH_E("Mic Switch", SSM2602_APANA, 1, 1, &mic_ctl, + ssm2602_mic_switch_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_OUTPUT("LHPOUT"), SND_SOC_DAPM_OUTPUT("RHPOUT"), SND_SOC_DAPM_INPUT("MICIN"), @@ -178,9 +202,11 @@ static const struct snd_soc_dapm_route ssm2602_routes[] = { {"LHPOUT", NULL, "Output Mixer"}, {"Input Mux", "Line", "Line Input"}, - {"Input Mux", "Mic", "Mic Bias"}, + {"Input Mux", "Mic", "Mic Switch"}, {"ADC", NULL, "Input Mux"}, + {"Mic Switch", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "MICIN"}, }; -- 2.20.1 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
