December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: max9867: Fix signal paths" to the asoc tree
by Mark Brown 06 Dec '18

06 Dec '18

The patch ASoC: max9867: Fix signal paths 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 2f388effac683ee34016f915f7ff1d50b3f9df81 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:33 +0100 Subject: [PATCH] ASoC: max9867: Fix signal paths Sound capture and line bypass currently do not work as well as some mixer controls. Fix that by building proper audio paths and adjusting volume controls to match datasheet. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 188 ++++++++++++++++++++----------------- sound/soc/codecs/max9867.h | 8 +- 2 files changed, 102 insertions(+), 94 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index faec094161c1..7a08e2c70e17 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -23,107 +23,124 @@ static const char *const max9867_spmode[] = { "Stereo Single", "Mono Single", "Stereo Single Fast", "Mono Single Fast" }; -static const char *const max9867_sidetone_text[] = { - "None", "Left", "Right", "LeftRight", "LeftRightDiv2", -}; static const char *const max9867_filter_text[] = {"IIR", "FIR"}; static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, max9867_spmode); -static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, - max9867_sidetone_text); -static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); -static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, - 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_master_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-8600, 200, 1), + 3, 17, TLV_DB_SCALE_ITEM(-7800, 400, 0), + 18, 25, TLV_DB_SCALE_ITEM(-2000, 200, 0), + 26, 34, TLV_DB_SCALE_ITEM( -500, 100, 0), + 35, 40, TLV_DB_SCALE_ITEM( 350, 50, 0), +); +static DECLARE_TLV_DB_SCALE(max9867_mic_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_line_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9867_adc_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dac_tlv, -1500, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dacboost_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), ); static const struct snd_kcontrol_new max9867_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 63, 1), - SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, - 0, 15, 1, max9860_capture_tlv), - SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), - SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), - SOC_ENUM("Digital Sidetone Src", max9867_sidetone), - SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), - SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), - SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), - SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, - 4, 15, 1, max9860_adc_left_tlv), - SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, - 0, 15, 1, max9860_adc_right_tlv), + SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), + SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, + max9867_dac_tlv), + SOC_SINGLE_TLV("Digital Boost Playback Volume", MAX9867_DACLEVEL, 4, 3, 0, + max9867_dacboost_tlv), + SOC_DOUBLE_TLV("Digital Capture Volume", MAX9867_ADCLEVEL, 0, 4, 15, 1, + max9867_adc_tlv), SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), - SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), SOC_ENUM("DSP Filter", max9867_filter), }; -static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; +/* Input mixer */ +static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { + SOC_DAPM_DOUBLE("Line Capture Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), + SOC_DAPM_DOUBLE("Mic Capture Switch", MAX9867_INPUTCONFIG, 6, 4, 1, 0), +}; + +/* Output mixer */ +static const struct snd_kcontrol_new max9867_output_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Line Bypass Switch", + MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 6, 1, 1), +}; -static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, - MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, - max9867_mux); +/* Sidetone mixer */ +static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { + SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), +}; -static const struct snd_kcontrol_new max9867_dapm_mux_controls = - SOC_DAPM_ENUM("Route", max9867_mux_enum); +/* Line out switch */ +static const struct snd_kcontrol_new max9867_line_out_control = + SOC_DAPM_DOUBLE_R("Switch", + MAX9867_LEFTVOL, MAX9867_RIGHTVOL, 6, 1, 1); -static const struct snd_kcontrol_new max9867_left_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); -static const struct snd_kcontrol_new max9867_right_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); -static const struct snd_kcontrol_new max9867_line_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), - SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), - SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("HPOUT"), - - SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), - SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), - SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, - &max9867_dapm_mux_controls), - - SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, - &max9867_left_dapm_control), - SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, - &max9867_right_dapm_control), - SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, - &max9867_line_dapm_control), - SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + + SND_SOC_DAPM_PGA("Left Line Input", MAX9867_PWRMAN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Line Input", MAX9867_PWRMAN, 5, 0, NULL, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + max9867_input_mixer_controls, + ARRAY_SIZE(max9867_input_mixer_controls)), + SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + + SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, + max9867_sidetone_mixer_controls, + ARRAY_SIZE(max9867_sidetone_mixer_controls)), + SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", SND_SOC_NOPM, 0, 0, + max9867_output_mixer_controls, + ARRAY_SIZE(max9867_output_mixer_controls)), + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, + &max9867_line_out_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), }; static const struct snd_soc_dapm_route max9867_audio_map[] = { - {"Left DAC", NULL, "DAI_OUT"}, - {"Right DAC", NULL, "DAI_OUT"}, - {"Output Mixer", NULL, "Left DAC"}, - {"Output Mixer", NULL, "Right DAC"}, - {"HPOUT", NULL, "Output Mixer"}, - - {"Left ADC", NULL, "DAI_IN"}, - {"Right ADC", NULL, "DAI_IN"}, - {"Input Mixer", NULL, "Left ADC"}, - {"Input Mixer", NULL, "Right ADC"}, - {"Input Mux", "Line", "Input Mixer"}, - {"Input Mux", "Mic", "Input Mixer"}, - {"Input Mux", "Mic_Line", "Input Mixer"}, - {"Right Line", "Switch", "Input Mux"}, - {"Left Line", "Switch", "Input Mux"}, - {"LINE_IN", NULL, "Left Line"}, - {"LINE_IN", NULL, "Right Line"}, + {"Left Line Input", NULL, "LINL"}, + {"Right Line Input", NULL, "LINR"}, + {"Input Mixer", "Mic Capture Switch", "MICL"}, + {"Input Mixer", "Mic Capture Switch", "MICR"}, + {"Input Mixer", "Line Capture Switch", "Left Line Input"}, + {"Input Mixer", "Line Capture Switch", "Right Line Input"}, + {"ADCL", NULL, "Input Mixer"}, + {"ADCR", NULL, "Input Mixer"}, + + {"Digital", "Sidetone Switch", "ADCL"}, + {"Digital", "Sidetone Switch", "ADCR"}, + {"DACL", NULL, "Digital"}, + {"DACR", NULL, "Digital"}, + + {"Output Mixer", "Line Bypass Switch", "Left Line Input"}, + {"Output Mixer", "Line Bypass Switch", "Right Line Input"}, + {"Output Mixer", NULL, "DACL"}, + {"Output Mixer", NULL, "DACR"}, + {"Master Playback", "Switch", "Output Mixer"}, + {"LOUT", NULL, "Master Playback"}, + {"ROUT", NULL, "Master Playback"}, }; static const unsigned int max9867_rates_44k1[] = { @@ -235,13 +252,8 @@ static int max9867_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - if (mute) - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); - else - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, 0); - return 0; + return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + 1 << 6, !!mute << 6); } static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -464,8 +476,8 @@ static const struct reg_default max9867_reg[] = { { 0x0B, 0x00 }, { 0x0C, 0x00 }, { 0x0D, 0x00 }, - { 0x0E, 0x00 }, - { 0x0F, 0x00 }, + { 0x0E, 0x40 }, + { 0x0F, 0x40 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 7f037ab701a5..2277798291a1 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -48,20 +48,16 @@ #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a -#define MAX9867_DACGAIN 0x0b +#define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c -#define MAX9867_DAC_MUTE_SHIFT 0x6 -#define MAX9867_DAC_MUTE_WIDTH 0x1 -#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) #define MAX9867_ADCLEVEL 0x0d #define MAX9867_LEFTLINELVL 0x0e -#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_RIGHTLINELVL 0x0f #define MAX9867_LEFTVOL 0x10 #define MAX9867_RIGHTVOL 0x11 #define MAX9867_LEFTMICGAIN 0x12 #define MAX9867_RIGHTMICGAIN 0x13 #define MAX9867_INPUTCONFIG 0x14 -#define MAX9867_INPUT_SHIFT 0x6 #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 -- 2.19.0.rc2

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[alsa-devel] [PATCH 1/2] ASoC: Fix function return type
by Agrawal, Akshu 06 Dec '18

06 Dec '18

Function returns -1 on error and the return type should be int. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- include/sound/soc.h | 2 +- sound/soc/soc-io.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de8..591d743 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1341,7 +1341,7 @@ static inline int snd_soc_component_cache_sync( /* component IO */ int snd_soc_component_read(struct snd_soc_component *component, unsigned int reg, unsigned int *val); -unsigned int snd_soc_component_read32(struct snd_soc_component *component, +int snd_soc_component_read32(struct snd_soc_component *component, unsigned int reg); int snd_soc_component_write(struct snd_soc_component *component, unsigned int reg, unsigned int val); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 1ff9175..d08f5b1 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -38,7 +38,7 @@ int snd_soc_component_read(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_component_read); -unsigned int snd_soc_component_read32(struct snd_soc_component *component, +int snd_soc_component_read32(struct snd_soc_component *component, unsigned int reg) { unsigned int val; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: adau1761: Ensure DSP_RUN and DSP_ENABLE are disabled
by Niklas Carlsson 06 Dec '18

06 Dec '18

DSP_RUN and DSP_ENABLE needs to be disabled during FW load. This is not a problem after power-cycling but for soft reboots. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..3075c13bcbbc 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -460,8 +460,35 @@ static int adau1761_set_bias_level(struct snd_soc_component *component, regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* + * When going OFF -> STANDBY, the device can be in two + * states: + * 1) Power-cycled and reset + * 2) Soft reboot + * In the case of soft reboot, we need to sync the HW + * registers even if our regmap reports default values. + * Marking the cache as dirty ensures that both cases + * are handled. + */ + regcache_mark_dirty(adau->regmap); regcache_sync(adau->regmap); + /* + * In order for FW to load correctly, the device needs + * DSP_RUN and DSP_ENABLE to be 0. This might not be the + * case for soft reboots. Ensure that DSP_RUN and + * DSP_ENABLE is 0 by bypassing the cache and write + * directly to HW when going OFF -> STANDBY. + */ + if (adau17x1_has_dsp(adau)) { + regcache_cache_bypass(adau->regmap, true); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); + regmap_write(adau->regmap, + ADAU17X1_DSP_ENABLE, 0); + regcache_cache_bypass(adau->regmap, false); + } + } break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, -- 2.11.0

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[alsa-devel] [PATCH] MAINTAINERS: add ASoC maintainers for sound dt-bindings
by Clément Péron 06 Dec '18

06 Dec '18

Sound dt-bindings are applied by ASoC maintainers and should be submit to ASoC list in addition to the devicetree list. Hence, add this information into the MAINTAINERS file. Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 0abecc528dac..62701e0135af 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13930,6 +13930,7 @@ S: Supported F: Documentation/devicetree/bindings/sound/ F: Documentation/sound/soc/ F: sound/soc/ +F: include/dt-bindings/sound/ F: include/sound/soc* SOUNDWIRE SUBSYSTEM -- 2.19.1

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[alsa-devel] [PATCH v2] ASoC: imx-audmux: complete dt-bindings for i.MX6
by Clément Péron 06 Dec '18

06 Dec '18

From: Colin Didier <colin.didier(a)devialet.com> The i.MX6 audmux differs from i.MX51. This patch adds the audmux for i.MX6 family. Signed-off-by: Colin Didier <colin.didier(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v2: Repost with ASoC maintainer and list include/dt-bindings/sound/fsl-imx-audmux.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 15f138bebe16..a1d0741d9ed1 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -25,6 +25,14 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +#define MX6_AUDMUX_PORT1_SSI1 0 +#define MX6_AUDMUX_PORT2_SSI2 1 +#define MX6_AUDMUX_PORT3 2 +#define MX6_AUDMUX_PORT4 3 +#define MX6_AUDMUX_PORT5 4 +#define MX6_AUDMUX_PORT6 5 +#define MX6_AUDMUX_PORT7_SSI3 6 + /* * TFCSEL/RFCSEL (i.MX27) or TFSEL/TCSEL/RFSEL/RCSEL (i.MX31/51/53/6Q) * can be sourced from Rx/Tx. -- 2.19.1

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[alsa-devel] [PATCH] ALSA: Use of_node_name_eq for node name comparisons
by Rob Herring 06 Dec '18

06 Dec '18

Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. A couple of open coded iterating thru the child node names are converted to use for_each_child_of_node() instead. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Cc: Paul Mackerras <paulus(a)samba.org> Cc: Michael Ellerman <mpe(a)ellerman.id.au> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/fabrics/layout.c | 2 +- sound/aoa/soundbus/i2sbus/core.c | 6 +++--- sound/ppc/pmac.c | 4 ++-- sound/ppc/tumbler.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3a23fd032611..8797d42e2b76 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -776,7 +776,7 @@ static int check_codec(struct aoa_codec *codec, struct codec_connection *cc; /* if the codec has a 'codec' node, we require a reference */ - if (codec->node && (strcmp(codec->node->name, "codec") == 0)) { + if (of_node_name_eq(codec->node, "codec")) { snprintf(propname, sizeof(propname), "platform-%s-codec-ref", codec->name); ref = of_get_property(ldev->sound, propname, NULL); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index bd7c5029fc59..c3f57a3fb1a5 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -154,7 +154,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *np) { struct i2sbus_dev *dev; - struct device_node *child = NULL, *sound = NULL; + struct device_node *child, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; char node_name[6]; @@ -177,8 +177,8 @@ static int i2sbus_add_dev(struct macio_dev *macio, return 0; i = 0; - while ((child = of_get_next_child(np, child))) { - if (strcmp(child->name, "sound") == 0) { + for_each_child_of_node(np, child) { + if (of_node_name_eq(child, "sound")) { i++; sound = child; } diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 48dd44f8e914..d692e4070167 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -908,7 +908,7 @@ static void detect_byte_swap(struct snd_pmac *chip) /* if seems that Keylargo can't byte-swap */ for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { if (of_device_is_compatible(mio, "Keylargo")) chip->can_byte_swap = 0; break; @@ -1313,7 +1313,7 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) } else if (chip->is_pbook_G3) { struct device_node* mio; for (mio = chip->node->parent; mio; mio = mio->parent) { - if (strcmp(mio->name, "mac-io") == 0) { + if (of_node_name_eq(mio, "mac-io")) { struct resource r; if (of_address_to_resource(mio, 0, &r) == 0) chip->macio_base = diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 0779a2912237..6d7ffffcce95 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1365,8 +1365,8 @@ int snd_pmac_tumbler_init(struct snd_pmac *chip) mix->anded_reset = 0; mix->reset_on_sleep = 1; - for (np = chip->node->child; np; np = np->sibling) { - if (!strcmp(np->name, "sound")) { + for_each_child_of_node(chip->node, np) { + if (of_node_name_eq(np, "sound")) { if (of_get_property(np, "has-anded-reset", NULL)) mix->anded_reset = 1; if (of_get_property(np, "layout-id", NULL)) -- 2.19.1

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[alsa-devel] [PATCH] ALSA: soundbus: Remove direct OF name and type accesses
by Rob Herring 06 Dec '18

06 Dec '18

Convert soundbus uevent and sysfs OF node name and device type usage to use printf specifier and helper functions instead of directly accessing the name and type pointers. This will allow the eventual removal of the pointers. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/soundbus/core.c | 2 +- sound/aoa/soundbus/sysfs.c | 34 +++++++++++++++++++++------------- 2 files changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index f3076a3f7f2f..065d3a55725e 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -74,7 +74,7 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env) of = &soundbus_dev->ofdev; /* stuff we want to pass to /sbin/hotplug */ - retval = add_uevent_var(env, "OF_NAME=%s", of->dev.of_node->name); + retval = add_uevent_var(env, "OF_NAME=%pOFn", of->dev.of_node); if (retval) return retval; diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c index 81da020bddef..a2d55e15afbb 100644 --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -1,18 +1,10 @@ // SPDX-License-Identifier: GPL-2.0 #include <linux/kernel.h> +#include <linux/of.h> #include <linux/stat.h> /* FIX UP */ #include "soundbus.h" -#define soundbus_config_of_attr(field, format_string) \ -static ssize_t \ -field##_show (struct device *dev, struct device_attribute *attr, \ - char *buf) \ -{ \ - struct soundbus_dev *mdev = to_soundbus_device (dev); \ - return sprintf (buf, format_string, mdev->ofdev.dev.of_node->field); \ -} - static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, char *buf) { @@ -25,17 +17,33 @@ static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, strcat(buf, "\n"); length = strlen(buf); } else { - length = sprintf(buf, "of:N%sT%s\n", - of->dev.of_node->name, of->dev.of_node->type); + length = sprintf(buf, "of:N%pOFn%c%s\n", + of->dev.of_node, 'T', + of_node_get_device_type(of->dev.of_node)); } return length; } static DEVICE_ATTR_RO(modalias); -soundbus_config_of_attr (name, "%s\n"); +static ssize_t name_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%pOFn\n", of->dev.of_node); +} static DEVICE_ATTR_RO(name); -soundbus_config_of_attr (type, "%s\n"); + +static ssize_t type_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + + return sprintf(buf, "%s\n", of_node_get_device_type(of->dev.of_node)); +} static DEVICE_ATTR_RO(type); struct attribute *soundbus_dev_attrs[] = { -- 2.19.1

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[alsa-devel] Invalid TLS certificate for mailman.alsa-project.org
by Antonio Ospite 06 Dec '18

06 Dec '18

Hi, when logging in to mailman.alsa-project.org[1] to change my subscription options I noticed that the TLS certificate used for HTTPS connections does not cover said domain, firefox says: ----------------------------------------------------------------------- mailman.alsa-project.org uses an invalid security certificate. The certificate is only valid for the following names: alsa-project.org, alsa0.perex.cz, bugtrack.alsa-project.org, git.alsa-project.org, mail.alsa-project.org, www.alsa-project.org Error code: SSL_ERROR_BAD_CERT_DOMAIN ----------------------------------------------------------------------- Basically, mailman.alsa-project.org is not in Subject Alternative Names for the certificate. This is not a huge deal as: 1. mail.alsa-project.org works fine with HTTPS; 2. the password is sent by email in the monthly reminder anyways. However it would still be handy to have this fixed as mailman.alsa-project.org URLs are used in the Wiki[2] pages and in the footer of messages coming from the mailing list, and it can be surprising for users that HTTPS does not work. Thank you, Antonio [1] https://mailman.alsa-project.org/mailman/options/alsa-devel [2] https://www.alsa-project.org/main/index.php/Mailing-lists -- Antonio Ospite https://ao2.it https://twitter.com/ao2it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

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[alsa-devel] [PATCH linux-next v3 6/7] ASoC: rsnd: add avb clocks
by Jiada Wang 06 Dec '18

06 Dec '18

There are AVB Counter Clocks in ADG, each clock has 12bits integral and 8 bits fractional dividers which operates with S0D1ϕ clock. This patch registers 8 AVB Counter Clocks when clock-cells of rcar_sound node is 2, Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/adg.c | 316 +++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/gen.c | 9 ++ sound/soc/sh/rcar/rsnd.h | 9 ++ 3 files changed, 325 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 6768a66588eb..0708be1f38d9 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -5,6 +5,7 @@ // Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/clk-provider.h> +#include <linux/of_address.h> #include "rsnd.h" #define CLKA 0 @@ -21,13 +22,30 @@ #define BRRx_MASK(x) (0x3FF & x) +#define AVB_CLK_NUM 8 +#define AVB_MAX_RATE 25000000 +#define AVB_DIV_EN_COM BIT(31) +#define AVB_MAX_DIV 0x3ffc0 + static struct rsnd_mod_ops adg_ops = { .name = "adg", }; +struct clk_avb { + struct clk_hw hw; + unsigned int idx; + struct rsnd_adg *adg; + /* lock reg access */ + spinlock_t *lock; +}; + +#define hw_to_avb(_hw) container_of(_hw, struct clk_avb, hw) + struct rsnd_adg { struct clk *clk[CLKMAX]; struct clk *clkout[CLKOUTMAX]; + struct clk *clkavb[AVB_CLK_NUM]; + struct clk *clkadg; struct clk_onecell_data onecell; struct rsnd_mod mod; u32 flags; @@ -37,6 +55,7 @@ struct rsnd_adg { int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ + spinlock_t avb_lock; }; #define LRCLK_ASYNC (1 << 0) @@ -408,6 +427,248 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) } } +static struct clk *rsnd_adg_clk_src_twocell_get(struct of_phandle_args *clkspec, + void *data) +{ + unsigned int clktype = clkspec->args[0]; + unsigned int clkidx = clkspec->args[1]; + struct rsnd_adg *adg = data; + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + struct rsnd_priv *priv = rsnd_mod_to_priv(adg_mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + + switch (clktype) { + case 0: + if (clkidx >= CLKOUTMAX) { + dev_err(dev, "Invalid fixed clock index %u\n", clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkout[clkidx]; + break; + case 1: + if (clkidx >= AVB_CLK_NUM) { + dev_err(dev, "Invalid avb clock index %u\n", clkidx); + return ERR_PTR(-EINVAL); + } + clk = adg->clkavb[clkidx]; + break; + default: + dev_err(dev, "Invalid ADG clock type %u\n", clktype); + return ERR_PTR(-EINVAL); + } + + return clk; +} + +static void clk_avb_div_write(struct rsnd_adg *adg, u32 data, unsigned int idx) +{ + struct rsnd_mod *mod = rsnd_mod_get(adg); + + switch (idx) { + case 0: + rsnd_mod_write(mod, AVB_CLK_DIV0, data); + break; + case 1: + rsnd_mod_write(mod, AVB_CLK_DIV1, data); + break; + case 2: + rsnd_mod_write(mod, AVB_CLK_DIV2, data); + break; + case 3: + rsnd_mod_write(mod, AVB_CLK_DIV3, data); + break; + case 4: + rsnd_mod_write(mod, AVB_CLK_DIV4, data); + break; + case 5: + rsnd_mod_write(mod, AVB_CLK_DIV5, data); + break; + case 6: + rsnd_mod_write(mod, AVB_CLK_DIV6, data); + break; + case 7: + rsnd_mod_write(mod, AVB_CLK_DIV7, data); + break; + } +} + +static u32 clk_avb_div_read(struct rsnd_adg *adg, unsigned int idx) +{ + struct rsnd_mod *mod = rsnd_mod_get(adg); + u32 val = 0; + + switch (idx) { + case 0: + val = rsnd_mod_read(mod, AVB_CLK_DIV0); + break; + case 1: + val = rsnd_mod_read(mod, AVB_CLK_DIV1); + break; + case 2: + val = rsnd_mod_read(mod, AVB_CLK_DIV2); + break; + case 3: + val = rsnd_mod_read(mod, AVB_CLK_DIV3); + break; + case 4: + val = rsnd_mod_read(mod, AVB_CLK_DIV4); + break; + case 5: + val = rsnd_mod_read(mod, AVB_CLK_DIV5); + break; + case 6: + val = rsnd_mod_read(mod, AVB_CLK_DIV6); + break; + case 7: + val = rsnd_mod_read(mod, AVB_CLK_DIV7); + break; + } + + return val; +} + +static int clk_avb_is_enabled(struct clk_hw *hw) +{ + struct clk_avb *avb = hw_to_avb(hw); + struct rsnd_mod *mod = rsnd_mod_get(avb->adg); + + return rsnd_mod_read(mod, AVB_CLK_CONFIG) & BIT(avb->idx); +} + +static int clk_avb_enabledisable(struct clk_hw *hw, int enable) +{ + struct clk_avb *avb = hw_to_avb(hw); + u32 val; + struct rsnd_mod *mod = rsnd_mod_get(avb->adg); + + spin_lock(avb->lock); + + val = rsnd_mod_read(mod, AVB_CLK_CONFIG); + if (enable) + val |= BIT(avb->idx); + else + val &= ~BIT(avb->idx); + rsnd_mod_write(mod, AVB_CLK_CONFIG, val); + + spin_unlock(avb->lock); + + return 0; +} + +static int clk_avb_enable(struct clk_hw *hw) +{ + return clk_avb_enabledisable(hw, 1); +} + +static void clk_avb_disable(struct clk_hw *hw) +{ + clk_avb_enabledisable(hw, 0); +} + +static unsigned long clk_avb_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_avb *avb = hw_to_avb(hw); + u32 div; + + div = clk_avb_div_read(avb->adg, avb->idx); + if (!div) + return parent_rate; + + return parent_rate * 32 / div; +} + +static unsigned int clk_avb_calc_div(unsigned long rate, + unsigned long parent_rate) +{ + unsigned int div; + + if (!rate) + rate = 1; + + if (rate > AVB_MAX_RATE) + rate = AVB_MAX_RATE; + + div = DIV_ROUND_CLOSEST(parent_rate * 32, rate); + + if (div > AVB_MAX_DIV) + div = AVB_MAX_DIV; + + return div; +} + +static long clk_avb_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + unsigned int div = clk_avb_calc_div(rate, *parent_rate); + + return (*parent_rate * 32) / div; +} + +static int clk_avb_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct clk_avb *avb = hw_to_avb(hw); + unsigned int div = clk_avb_calc_div(rate, parent_rate); + u32 val; + + val = clk_avb_div_read(avb->adg, avb->idx); + clk_avb_div_write(avb->adg, val | div, avb->idx); + + return 0; +} + +static const struct clk_ops clk_avb_ops = { + .enable = clk_avb_enable, + .disable = clk_avb_disable, + .is_enabled = clk_avb_is_enabled, + .recalc_rate = clk_avb_recalc_rate, + .round_rate = clk_avb_round_rate, + .set_rate = clk_avb_set_rate, +}; + +static struct clk *clk_register_avb(struct device *dev, struct rsnd_adg *adg, + unsigned int id, spinlock_t *lock) +{ + struct clk_init_data init; + struct clk_avb *avb; + struct clk *clk; + char name[8]; + const char *parent_name; + + if (IS_ERR(adg->clkadg)) + return ERR_PTR(-ENODEV); + + avb = devm_kzalloc(dev, sizeof(*avb), GFP_KERNEL); + if (!avb) + return ERR_PTR(-ENOMEM); + + parent_name = __clk_get_name(adg->clkadg); + + snprintf(name, sizeof(name), "%s%u", "avb", id); + + avb->idx = id; + avb->lock = lock; + avb->adg = adg; + + /* Register the clock. */ + init.name = name; + init.ops = &clk_avb_ops; + init.flags = CLK_IS_BASIC; + init.parent_names = &parent_name; + init.num_parents = 1; + + avb->hw.init = &init; + + /* init DIV to a valid state */ + clk_avb_div_write(avb->adg, AVB_MAX_DIV, avb->idx); + + clk = devm_clk_register(dev, &avb->hw); + + return clk; +} + static void rsnd_adg_get_clkin(struct rsnd_priv *priv, struct rsnd_adg *adg) { @@ -436,6 +697,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, unsigned long req_48kHz_rate, req_441kHz_rate; int i, req_size; const char *parent_clk_name = NULL; + struct rsnd_mod *mod = rsnd_mod_get(adg); static const char * const clkout_name[] = { [CLKOUT] = "audio_clkout", [CLKOUT1] = "audio_clkout1", @@ -540,21 +802,23 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ of_property_read_u32(np, "#clock-cells", &count); - /* - * for clkout - */ - if (!count) { + + switch (count) { + case 0: + /* + * for clkout + */ clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, 0, req_rate[0]); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); } - } - /* - * for clkout0/1/2/3 - */ - else { + break; + case 1: + /* + * for clkout0/1/2/3 + */ for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, @@ -566,6 +830,33 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->onecell.clk_num = CLKOUTMAX; of_clk_add_provider(np, of_clk_src_onecell_get, &adg->onecell); + break; + case 2: + /* + * for clkout0/1/2/3 and avb clocks + */ + for (i = 0; i < CLKOUTMAX; i++) { + clk = clk_register_fixed_rate(dev, clkout_name[i], + parent_clk_name, 0, + req_rate[0]); + if (!IS_ERR(clk)) + adg->clkout[i] = clk; + } + + for (i = 0; i < AVB_CLK_NUM; i++) { + clk = clk_register_avb(dev, adg, i, &adg->avb_lock); + if (!IS_ERR(clk)) + adg->clkavb[i] = clk; + } + + of_clk_add_provider(np, rsnd_adg_clk_src_twocell_get, adg); + + rsnd_mod_write(mod, AVB_CLK_CONFIG, AVB_DIV_EN_COM); + + break; + default: + dev_err(dev, "Invalid clock-cell %d\n", count); + break; } rsnd_adg_get_clkout_end: @@ -606,17 +897,24 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) return -ENOMEM; + spin_lock_init(&adg->avb_lock); + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, 0, 0); if (ret) return ret; + clk = devm_clk_get(dev, "adg"); + if (!IS_ERR(clk)) + adg->clkadg = clk; + rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); rsnd_adg_clk_dbg_info(priv, adg); diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index ca639404f2cd..1b000d03b76e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -355,6 +355,15 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + RSND_GEN_S_REG(AVB_CLK_DIV0, 0x11c), + RSND_GEN_S_REG(AVB_CLK_DIV1, 0x120), + RSND_GEN_S_REG(AVB_CLK_DIV2, 0x124), + RSND_GEN_S_REG(AVB_CLK_DIV3, 0x128), + RSND_GEN_S_REG(AVB_CLK_DIV4, 0x12c), + RSND_GEN_S_REG(AVB_CLK_DIV5, 0x130), + RSND_GEN_S_REG(AVB_CLK_DIV6, 0x134), + RSND_GEN_S_REG(AVB_CLK_DIV7, 0x138), + RSND_GEN_S_REG(AVB_CLK_CONFIG, 0x13c), }; static const struct rsnd_regmap_field_conf conf_ssi[] = { RSND_GEN_M_REG(SSICR, 0x00, 0x40), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 3c57129af6d1..d31b8a65985f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -148,6 +148,15 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL0, RSND_REG_AUDIO_CLK_SEL1, RSND_REG_AUDIO_CLK_SEL2, + RSND_REG_AVB_CLK_DIV0, + RSND_REG_AVB_CLK_DIV1, + RSND_REG_AVB_CLK_DIV2, + RSND_REG_AVB_CLK_DIV3, + RSND_REG_AVB_CLK_DIV4, + RSND_REG_AVB_CLK_DIV5, + RSND_REG_AVB_CLK_DIV6, + RSND_REG_AVB_CLK_DIV7, + RSND_REG_AVB_CLK_CONFIG, /* SSIU */ RSND_REG_SSI_MODE, -- 2.19.2

2 6

[alsa-devel] alsa-utils: axfer: add manuals
by GitHub pull_request - opened 05 Dec '18

05 Dec '18

alsa-project/alsa-utils pull request #6 was opened from takaswie: alsa-utils: axfer: add manuals This patchset adds manuals. As a result, users can refer to documents by man(1): ``` $ man 1 axfer $ man 1 axfer-list $ man 1 axfer-transfer ``` The manuals are written according to groff_man(7) and man-pages(7), however in some points they don't follow the guideline; e.g. usage of hyphen-minus ('-') without backslash '\' in pre-formatted text. I'll welcome any fixes. In this time, description of each backend in transfer subcommand is not yet added. It will be added in later work. ``` Takashi Sakamoto (5): axfer: add a manual for entry point axfer: add a manual for list subcommand axfer: add a manual for transfer subcommand axfer: add a section to describe compatibility to aplay(1) axfer: add a section about design of transfer subcommand axfer/Makefile.am | 10 + axfer/axfer-list.1 | 86 +++++++ axfer/axfer-transfer.1 | 500 +++++++++++++++++++++++++++++++++++++++++ axfer/axfer.1 | 132 +++++++++++ 4 files changed, 728 insertions(+) create mode 100644 axfer/axfer-list.1 create mode 100644 axfer/axfer-transfer.1 create mode 100644 axfer/axfer.1 ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/6 Patch URL : https://github.com/alsa-project/alsa-utils/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-utils

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