October 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx
by Nikita Yushchenko 25 Oct '16

25 Oct '16

tlv320dac31xx is a subset of tlv320aic31xx: - it does not have MIC inputs and ADC, thus capture is not supported, - it has analog inputs AIN1/AIN2 that can be mixed into output. Although tlv320dac31xx does work with tlv320aic31xx driver, this setup does register non-existent widgets and non-existent capture stream. Thus userspace lists non-existent objects in user interfaces, an can access these, causing operations with device registers that are declared as "reserved" in tlv320dac31xx datasheet. This patch fixes this situation by separating controls/widgets/routes into common, aic31xx-specific, and dac31xx-specific parts. Only parts that match actual hardware (as declared in "compatible" device tree property) are registered. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/codecs/tlv320aic31xx.c | 212 ++++++++++++++++++++++++++++----------- sound/soc/codecs/tlv320aic31xx.h | 2 + 2 files changed, 158 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 3c5e1df..fb7d648 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -273,10 +273,20 @@ static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); /* * controls to be exported to the user space */ -static const struct snd_kcontrol_new aic31xx_snd_controls[] = { +static const struct snd_kcontrol_new common31xx_snd_controls[] = { SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, adc_fgain_tlv), @@ -286,14 +296,6 @@ static const struct snd_kcontrol_new aic31xx_snd_controls[] = { SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, 119, 0, mic_pga_tlv), - - SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, - AIC31XX_HPRGAIN, 2, 1, 0), - SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, - AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), - - SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, - AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), }; static const struct snd_kcontrol_new aic311x_snd_controls[] = { @@ -397,17 +399,28 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_kcontrol_new left_output_switches[] = { +static const struct snd_kcontrol_new aic31xx_left_output_switches[] = { SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), }; -static const struct snd_kcontrol_new right_output_switches[] = { +static const struct snd_kcontrol_new aic31xx_right_output_switches[] = { SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), }; +static const struct snd_kcontrol_new dac31xx_left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From AIN1", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new dac31xx_right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + static const struct snd_kcontrol_new p_term_mic1lp = SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); @@ -457,7 +470,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { +static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("DAC Left Input", @@ -473,14 +486,7 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - /* Output Mixers */ - SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, - left_output_switches, - ARRAY_SIZE(left_output_switches)), - SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, - right_output_switches, - ARRAY_SIZE(right_output_switches)), - + /* HP */ SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, &aic31xx_dapm_hpl_switch), SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, @@ -494,10 +500,34 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { NULL, 0, aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), - /* ADC */ - SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, - aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_POST_PMD), + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_widget dac31xx_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + dac31xx_left_output_switches, + ARRAY_SIZE(dac31xx_left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + dac31xx_right_output_switches, + ARRAY_SIZE(dac31xx_right_output_switches)), +}; + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), /* Input Selection to MIC_PGA */ SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, @@ -507,24 +537,25 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, &p_term_mic1lm), + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, 7, 1, NULL, 0), - /* Mic Bias */ - SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - - /* Outputs */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* Inputs */ - SND_SOC_DAPM_INPUT("MIC1LP"), - SND_SOC_DAPM_INPUT("MIC1RP"), - SND_SOC_DAPM_INPUT("MIC1LM"), + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + aic31xx_left_output_switches, + ARRAY_SIZE(aic31xx_left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + aic31xx_right_output_switches, + ARRAY_SIZE(aic31xx_right_output_switches)), }; static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { @@ -554,7 +585,7 @@ static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { }; static const struct snd_soc_dapm_route -aic31xx_audio_map[] = { +common31xx_audio_map[] = { /* DAC Input Routing */ {"DAC Left Input", "Left Data", "DAC IN"}, {"DAC Left Input", "Right Data", "DAC IN"}, @@ -565,6 +596,31 @@ aic31xx_audio_map[] = { {"DAC Left", NULL, "DAC Left Input"}, {"DAC Right", NULL, "DAC Right Input"}, + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +dac31xx_audio_map[] = { + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From AIN1", "AIN1"}, + {"Output Left", "From AIN2", "AIN2"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From AIN2", "AIN2"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { /* Mic input */ {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, @@ -595,16 +651,6 @@ aic31xx_audio_map[] = { /* Right Output */ {"Output Right", "From Right DAC", "DAC Right"}, {"Output Right", "From MIC1RP", "MIC1RP"}, - - /* HPL path */ - {"HP Left", "Switch", "Output Left"}, - {"HPL Driver", NULL, "HP Left"}, - {"HPL", NULL, "HPL Driver"}, - - /* HPR path */ - {"HP Right", "Switch", "Output Right"}, - {"HPR Driver", NULL, "HP Right"}, - {"HPR", NULL, "HPR Driver"}, }; static const struct snd_soc_dapm_route @@ -633,6 +679,13 @@ static int aic31xx_add_controls(struct snd_soc_codec *codec) int ret = 0; struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + if (!(aic31xx->pdata.codec_type & DAC31XX_BIT)) + ret = snd_soc_add_codec_controls( + codec, aic31xx_snd_controls, + ARRAY_SIZE(aic31xx_snd_controls)); + if (ret) + return ret; + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) ret = snd_soc_add_codec_controls( codec, aic311x_snd_controls, @@ -651,6 +704,30 @@ static int aic31xx_add_widgets(struct snd_soc_codec *codec) struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); int ret = 0; + if (aic31xx->pdata.codec_type & DAC31XX_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, dac31xx_dapm_widgets, + ARRAY_SIZE(dac31xx_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, dac31xx_audio_map, + ARRAY_SIZE(dac31xx_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic31xx_dapm_widgets, + ARRAY_SIZE(aic31xx_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic31xx_audio_map, + ARRAY_SIZE(aic31xx_audio_map)); + if (ret) + return ret; + } + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { ret = snd_soc_dapm_new_controls( dapm, aic311x_dapm_widgets, @@ -1114,12 +1191,12 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .set_bias_level = aic31xx_set_bias_level, .suspend_bias_off = true, - .controls = aic31xx_snd_controls, - .num_controls = ARRAY_SIZE(aic31xx_snd_controls), - .dapm_widgets = aic31xx_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), - .dapm_routes = aic31xx_audio_map, - .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), + .controls = common31xx_snd_controls, + .num_controls = ARRAY_SIZE(common31xx_snd_controls), + .dapm_widgets = common31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(common31xx_dapm_widgets), + .dapm_routes = common31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(common31xx_audio_map), }; static const struct snd_soc_dai_ops aic31xx_dai_ops = { @@ -1129,6 +1206,21 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { .digital_mute = aic31xx_dac_mute, }; +static struct snd_soc_dai_driver dac31xx_dai_driver[] = { + { + .name = "tlv32dac31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + static struct snd_soc_dai_driver aic31xx_dai_driver[] = { { .name = "tlv320aic31xx-hifi", @@ -1259,9 +1351,16 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, if (ret) return ret; - return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, - aic31xx_dai_driver, - ARRAY_SIZE(aic31xx_dai_driver)); + if (aic31xx->pdata.codec_type & DAC31XX_BIT) + return snd_soc_register_codec(&i2c->dev, + &soc_codec_driver_aic31xx, + dac31xx_dai_driver, + ARRAY_SIZE(dac31xx_dai_driver)); + else + return snd_soc_register_codec(&i2c->dev, + &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); } static int aic31xx_i2c_remove(struct i2c_client *i2c) @@ -1277,6 +1376,7 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { { "tlv320aic3110", AIC3110 }, { "tlv320aic3120", AIC3120 }, { "tlv320aic3111", AIC3111 }, + { "tlv320dac3100", DAC3100 }, { } }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index ac9b146..5acd5b6 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -24,12 +24,14 @@ #define AIC31XX_STEREO_CLASS_D_BIT 0x1 #define AIC31XX_MINIDSP_BIT 0x2 +#define DAC31XX_BIT 0x4 enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, AIC3120 = AIC31XX_MINIDSP_BIT, AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), + DAC3100 = DAC31XX_BIT, }; struct aic31xx_pdata { -- 2.1.4

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[alsa-devel] [PATCH 0/8] ASoC: s3c24xx_uda134x card fixes and cleanups
by Sylwester Nawrocki 25 Oct '16

25 Oct '16

This series might have a bit of an archaeological aspect, nevertheless my goal was to have sound working on one of the older S3C24XX SoC based boards for testing any patches touching the older Samsung SoC IP block related code. Together with patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card ARM: S3C24XX: Add dma_mask assignments for DMA devices ARM: S3C24XX: Add missing DMA device for Mini2440 board this series fixes sound on Micro2440SDK/mini2440 board broken since the 3.12 kernel release. I'm not sure how it worked before but currently the uda134x codec always fails to probe due to uninitialized platform_data, which is supposed to be set by the ASoC machine driver. This series addresses the issue by moving all required codec platform_data to the board file and exposing optional GPIO bitbang L3 bus ops which are selected by the codec. ASoC dependency patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card The patch "ARM: S3C24XX: Specify audio codec platform_data for mini2440 board" is for arm-soc but it would be good to merge whole series through one tree to avoid compilation breaks due to unmet dependencies. Sylwester Nawrocki (8): ASoC: L3 bus: Add default gpio ops ARM: S3C24XX: Specify audio codec platform_data for mini2440 board ASoC: uda134x: Optionally initialize L3 ops to default GPIO ops ASoC: s3c24xx_uda134x: Remove unused power() callback ASoC: s3c24xx_uda134x: Drop initialization of codec's platform data ASoC: samsung: Convert s3c24xx_uda134x to use devm_snd_soc_register_card() ASoC: samsung: s3c24xx_uda134x: debug/error trace cleanup ASoC: s3c244_uda134x: Allocate private data dynamically arch/arm/mach-s3c24xx/mach-mini2440.c | 19 ++++ include/sound/l3.h | 15 ++- include/sound/s3c24xx_uda134x.h | 1 - sound/soc/codecs/l3.c | 71 ++++++++++-- sound/soc/codecs/uda134x.c | 7 ++ sound/soc/samsung/s3c24xx_uda134x.c | 205 ++++++++++------------------------ 6 files changed, 158 insertions(+), 160 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v4] ASoC: rt5514: Add the DMIC initial delay to wait it ready.
by Oder Chiou 25 Oct '16

25 Oct '16

Due to the DMIC that needs time to initial after the MCLK is provided, the field of delay time is implemented by the platform data. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5514.txt | 3 +++ include/sound/rt5514.h | 20 ++++++++++++++++++++ sound/soc/codecs/rt5514.c | 17 +++++++++++++++++ sound/soc/codecs/rt5514.h | 2 ++ 4 files changed, 42 insertions(+) create mode 100644 include/sound/rt5514.h diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 9cabfc1..929ca67 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -13,6 +13,9 @@ Optional properties: - clocks: The phandle of the master clock to the CODEC - clock-names: Should be "mclk" +- realtek,dmic-init-delay-ms + Set the DMIC initial delay (ms) to wait it ready. + Pins on the device (for linking into audio routes) for RT5514: * DMIC1L diff --git a/include/sound/rt5514.h b/include/sound/rt5514.h new file mode 100644 index 0000000..ef18494 --- /dev/null +++ b/include/sound/rt5514.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt5514.h -- Platform data for RT5514 + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5514_H +#define __LINUX_SND_RT5514_H + +struct rt5514_platform_data { + unsigned int dmic_init_delay; +}; + +#endif + diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f24b7cf..b281a46 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -452,6 +452,9 @@ static int rt5514_set_dmic_clk(struct snd_soc_dapm_widget *w, RT5514_CLK_DMIC_OUT_SEL_MASK, idx << RT5514_CLK_DMIC_OUT_SEL_SFT); + if (rt5514->pdata.dmic_init_delay) + msleep(rt5514->pdata.dmic_init_delay); + return idx; } @@ -1073,9 +1076,18 @@ static const struct of_device_id rt5514_of_match[] = { MODULE_DEVICE_TABLE(of, rt5514_of_match); #endif +static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dmic-init-delay-ms", + &rt5514->pdata.dmic_init_delay); + + return 0; +} + static int rt5514_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt5514_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5514_priv *rt5514; int ret; unsigned int val; @@ -1087,6 +1099,11 @@ static int rt5514_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5514); + if (pdata) + rt5514->pdata = *pdata; + else if (i2c->dev.of_node) + rt5514_parse_dt(rt5514, &i2c->dev); + rt5514->i2c_regmap = devm_regmap_init_i2c(i2c, &rt5514_i2c_regmap); if (IS_ERR(rt5514->i2c_regmap)) { ret = PTR_ERR(rt5514->i2c_regmap); diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 229de0e..5d343fb 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -13,6 +13,7 @@ #define __RT5514_H__ #include <linux/clk.h> +#include <sound/rt5514.h> #define RT5514_DEVICE_ID 0x10ec5514 @@ -243,6 +244,7 @@ enum { }; struct rt5514_priv { + struct rt5514_platform_data pdata; struct snd_soc_codec *codec; struct regmap *i2c_regmap, *regmap; struct clk *mclk; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2 1/3] ASoC: cs42xx8: Mark chip ID as volatile and remove cache bypass
by Charles Keepax 25 Oct '16

25 Oct '16

Rather than manually enabling cache bypass when reading the ID registers simply remove the default which will cause the first read to go to the hardware. The old code worked this is simply the more standard way to implement this. There is a comment included in the code that claims the chip ID register also contains the right input volume, however this is clearly not the case from the rest of the driver. Further investigation reveals exactly the same comment in the wm8962 driver, where this is the case, so this is almost certainly a copy and paste error from when the driver was created. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- Changes since v1: - Don't mark the ID register as volatile. Thanks, Charles sound/soc/codecs/cs42xx8.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index b4d8737..c1785bd 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -321,7 +321,6 @@ static struct snd_soc_dai_driver cs42xx8_dai = { }; static const struct reg_default cs42xx8_reg[] = { - { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ { 0x02, 0x00 }, /* Power Control */ { 0x03, 0xF0 }, /* Functional Mode */ { 0x04, 0x46 }, /* Interface Formats */ @@ -498,13 +497,6 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) /* Make sure hardware reset done */ msleep(5); - /* - * We haven't marked the chip revision as volatile due to - * sharing a register with the right input volume; explicitly - * bypass the cache to read it. - */ - regcache_cache_bypass(cs42xx8->regmap, true); - /* Validate the chip ID */ ret = regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); if (ret < 0) { @@ -523,8 +515,6 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) dev_info(dev, "found device, revision %X\n", val & CS42XX8_CHIPID_REV_ID_MASK); - regcache_cache_bypass(cs42xx8->regmap, false); - cs42xx8_dai.name = cs42xx8->drvdata->name; /* Each adc supports stereo input */ -- 2.1.4

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[alsa-devel] [PATCH 1/6] ASoC: cs42xx8: Mark chip ID as volatile and remove cache bypass
by Charles Keepax 25 Oct '16

25 Oct '16

Rather than manually enabling cache bypass when reading the ID registers simply mark them as volatile. The old code worked this is simply the more standard way to implement this. There is a comment included in the code that claims the chip ID register also contains the right input volume, however this is clearly not the case from the rest of the driver. Further investigation reveals exactly the same comment in the wm8962 driver, where this is the case, so this is almost certainly a copy and paste error from when the driver was created. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs42xx8.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index b4d8737..1d50748 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -321,7 +321,6 @@ static struct snd_soc_dai_driver cs42xx8_dai = { }; static const struct reg_default cs42xx8_reg[] = { - { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ { 0x02, 0x00 }, /* Power Control */ { 0x03, 0xF0 }, /* Functional Mode */ { 0x04, 0x46 }, /* Interface Formats */ @@ -352,6 +351,7 @@ static const struct reg_default cs42xx8_reg[] = { static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { + case CS42XX8_CHIPID: case CS42XX8_STATUS: return true; default: @@ -498,13 +498,6 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) /* Make sure hardware reset done */ msleep(5); - /* - * We haven't marked the chip revision as volatile due to - * sharing a register with the right input volume; explicitly - * bypass the cache to read it. - */ - regcache_cache_bypass(cs42xx8->regmap, true); - /* Validate the chip ID */ ret = regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); if (ret < 0) { @@ -523,8 +516,6 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) dev_info(dev, "found device, revision %X\n", val & CS42XX8_CHIPID_REV_ID_MASK); - regcache_cache_bypass(cs42xx8->regmap, false); - cs42xx8_dai.name = cs42xx8->drvdata->name; /* Each adc supports stereo input */ -- 2.1.4

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[alsa-devel] [PATCH v3 0/3] topology: Support widgets' stream name and file inclusion for text conf file
by mengdong.lin＠linux.intel.com 25 Oct '16

25 Oct '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series doesn't require or cause any ABI updates. It doesn't need the kernel patches that trying to udpate ABI from v4 or v5. Patch 1 fixes a missing topology feature for text conf file to configure stream name of a widget. Patch 2 allows alsaconf to search included files under global configuration directories. This will benefit topology and UCM. Patch 3 adds document about how to include files for topology configuration. There is no c code change but actually it depends on patch 2. History: v2: No longer use topology-specific code to support searching included files, but extend alsaconf to support this and user can just use alsaconf syntax <xxx> and <confdir:xxx>. v3: Not use <confdir:xxx> but define new syntax <searchdir:xxx> to define directories for searching included file. Mengdong Lin (3): topology: Fix missing stream name of widgets in text conf file alsaconf: Search included files under user specified configuration directories topology: Add doc for including other files in the text conf file include/topology.h | 31 ++++++++++ src/conf.c | 170 +++++++++++++++++++++++++++++++++++++++++++++++++++- src/topology/dapm.c | 10 ++++ 3 files changed, 210 insertions(+), 1 deletion(-) -- 2.5.0

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Re: [alsa-devel] [PATCH 5/7] ASoC: constify snd_soc_ops structures
by Julia Lawall 25 Oct '16

25 Oct '16

On Mon, 24 Oct 2016, Mark Brown wrote: > On Sat, Oct 15, 2016 at 04:55:49PM +0200, Julia Lawall wrote: > > > --- > > sound/soc/generic/simple-card.c | 2 +- > > sound/soc/generic/simple-scu-card.c | 2 +- > > sound/soc/intel/boards/bdw-rt5677.c | 2 +- > > sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 ++-- > > sound/soc/intel/boards/bxt_rt298.c | 4 ++-- > > sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- > > sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- > > sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- > > sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- > > sound/soc/intel/boards/cht_bsw_rt5672.c | 4 ++-- > > sound/soc/intel/boards/haswell.c | 2 +- > > sound/soc/intel/boards/skl_nau88l25_max98357a.c | 6 +++--- > > sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 6 +++--- > > sound/soc/intel/boards/skl_rt286.c | 4 ++-- > > sound/soc/kirkwood/armada-370-db.c | 2 +- > > sound/soc/mxs/mxs-sgtl5000.c | 2 +- > > 16 files changed, 28 insertions(+), 28 deletions(-) > > You split out a few drivers as separate patches but not these ones - > what's special about these ones? If it's preferred, I could send them all one file at a time, or when the maintainers are very similar just send one patch to all the maintainers. julia

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[alsa-devel] [PATCH] ALSA: seq: Fix time account regression
by Takashi Iwai 25 Oct '16

25 Oct '16

The recent rewrite of the sequencer time accounting using timespec64 in the commit [3915bf294652: ALSA: seq_timer: use monotonic times internally] introduced a bad regression. Namely, the time reported back doesn't increase but goes back and forth. The culprit was obvious: the delta is stored to the result (cur_time = delta), instead of adding the delta (cur_time += delta)! Let's fix it. Fixes: 3915bf294652 ('ALSA: seq_timer: use monotonic times internally') Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=177571 Reported-by: Yves Guillemot <yc.guillemot(a)wanadoo.fr> Cc: <stable(a)vger.kernel.org> # v4.8+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_timer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index dcc102813aef..37d9cfbc29f9 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -448,8 +448,8 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr) ktime_get_ts64(&tm); tm = timespec64_sub(tm, tmr->last_update); - cur_time.tv_nsec = tm.tv_nsec; - cur_time.tv_sec = tm.tv_sec; + cur_time.tv_nsec += tm.tv_nsec; + cur_time.tv_sec += tm.tv_sec; snd_seq_sanity_real_time(&cur_time); } spin_unlock_irqrestore(&tmr->lock, flags); -- 2.10.1

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[alsa-devel] [PATCH] ASoC: s3c24xx_uda134x: Move global variables to driver's data structure
by Sylwester Nawrocki 25 Oct '16

25 Oct '16

Gather all driver's private variables in common data structure and allocate the data structure dynamically. Also unused ENFORCE_RATES symbol and local variable (leftovers from an erroneous rebase) are removed. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/s3c24xx_uda134x.c | 79 ++++++++++++++++++++----------------- 1 file changed, 43 insertions(+), 36 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 7853fbe..81a7894 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -19,9 +19,15 @@ #include <sound/s3c24xx_uda134x.h> #include "regs-iis.h" - #include "s3c24xx-i2s.h" +struct s3c24xx_uda134x { + struct clk *xtal; + struct clk *pclk; + struct mutex clk_lock; + int clk_users; +}; + /* #define ENFORCE_RATES 1 */ /* Unfortunately the S3C24XX in master mode has a limited capacity of @@ -36,15 +42,6 @@ possible an error will be returned. */ -static struct clk *xtal; -static struct clk *pclk; -/* this is need because we don't have a place where to keep the - * pointers to the clocks in each substream. We get the clocks only - * when we are actually using them so we don't block stuff like - * frequency change or oscillator power-off */ -static int clk_users; -static DEFINE_MUTEX(clk_lock); - static unsigned int rates[33 * 2]; #ifdef ENFORCE_RATES static struct snd_pcm_hw_constraint_list hw_constraints_rates = { @@ -57,26 +54,24 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *cpu_dai = rtd->cpu_dai; -#ifdef ENFORCE_RATES - struct snd_pcm_runtime *runtime = substream->runtime; -#endif int ret = 0; - mutex_lock(&clk_lock); + mutex_lock(&priv->clk_lock); - if (clk_users == 0) { - xtal = clk_get(rtd->dev, "xtal"); - if (IS_ERR(xtal)) { + if (priv->clk_users == 0) { + priv->xtal = clk_get(rtd->dev, "xtal"); + if (IS_ERR(priv->xtal)) { dev_err(rtd->dev, "%s cannot get xtal\n", __func__); - ret = PTR_ERR(xtal); + ret = PTR_ERR(priv->xtal); } else { - pclk = clk_get(cpu_dai->dev, "iis"); - if (IS_ERR(pclk)) { + priv->pclk = clk_get(cpu_dai->dev, "iis"); + if (IS_ERR(priv->pclk)) { dev_err(rtd->dev, "%s cannot get pclk\n", __func__); - clk_put(xtal); - ret = PTR_ERR(pclk); + clk_put(priv->xtal); + ret = PTR_ERR(priv->pclk); } } if (!ret) { @@ -85,18 +80,19 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) for (i = 0; i < 2; i++) { int fs = i ? 256 : 384; - rates[i*33] = clk_get_rate(xtal) / fs; + rates[i*33] = clk_get_rate(priv->xtal) / fs; for (j = 1; j < 33; j++) - rates[i*33 + j] = clk_get_rate(pclk) / + rates[i*33 + j] = clk_get_rate(priv->pclk) / (j * fs); } } } - clk_users += 1; - mutex_unlock(&clk_lock); + priv->clk_users += 1; + mutex_unlock(&priv->clk_lock); + if (!ret) { #ifdef ENFORCE_RATES - ret = snd_pcm_hw_constraint_list(runtime, 0, + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (ret < 0) @@ -109,15 +105,18 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) { - mutex_lock(&clk_lock); - clk_users -= 1; - if (clk_users == 0) { - clk_put(xtal); - xtal = NULL; - clk_put(pclk); - pclk = NULL; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card); + + mutex_lock(&priv->clk_lock); + priv->clk_users -= 1; + if (priv->clk_users == 0) { + clk_put(priv->xtal); + priv->xtal = NULL; + clk_put(priv->pclk); + priv->pclk = NULL; } - mutex_unlock(&clk_lock); + mutex_unlock(&priv->clk_lock); } static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, @@ -228,10 +227,18 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, static int s3c24xx_uda134x_probe(struct platform_device *pdev) { struct snd_soc_card *card = &snd_soc_s3c24xx_uda134x; + struct s3c24xx_uda134x *priv; int ret; - platform_set_drvdata(pdev, card); + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + mutex_init(&priv->clk_lock); + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) -- 1.9.1

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[alsa-devel] [PATCH 0/2 v2] ASoC: Component has suspend/resume support
by Kuninori Morimoto 25 Oct '16

25 Oct '16

Hi Mark, Lars-Peter These patches are v2 of Component level suspend/resume support. Current Card has has aux_comp_list and codec_dev_list. Now, this patch will add component_dev_list. Here, we can simply replace codec_dev_list -> component_dev_list. and we can replace aux_comp_list by component_dev_list and new flags. V1 patchset fixuped Intel's broadwell and cht_bsw_rt5672 driver which is using codec_dev_list directly to uses component pointer instead of codec, but v2 keeps it as-is, just only replaced codec_dev_list to component_dev_list. These drivers are better to update in accordance with Lars's idea Kuninori Morimoto (2): 1) ASoC: core: add component_dev_list on Card 2) ASoC: add Component level suspend/resume include/sound/soc.h | 16 +++++++++------- sound/soc/intel/boards/broadwell.c | 10 ++++++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++++++---- sound/soc/soc-core.c | 90 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++---------------------------------- 4 files changed, 77 insertions(+), 49 deletions(-) Best regards --- Kuninori Morimoto

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