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

[alsa-devel] [PATCH 0/4 v2] ASoC: missing replace codec to component
by Kuninori Morimoto 19 Feb '18

19 Feb '18

Hi Mark These are v2 of replace patch for new added driver on mark/for-next branch. The diff v1 -> v2 is that compile warning issue was solved on ak driver Kuninori Morimoto (4): ASoC: intel: kbl_da7219_max98357: replace codec to component ASoC: amd: acp-da7219-max98357: replace codec to component ASoC: ak4458: replace codec to component ASoC: ak5558: replace codec to component sound/soc/amd/acp-da7219-max98357a.c | 4 +- sound/soc/codecs/ak4458.c | 107 +++++++++++++------------- sound/soc/codecs/ak5558.c | 67 ++++++++-------- sound/soc/intel/boards/kbl_da7219_max98357a.c | 12 +-- 4 files changed, 94 insertions(+), 96 deletions(-) -- 1.9.1

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[alsa-devel] Applied "ASoC: sgtl5000: add 'Audio Switch' block implementation" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: sgtl5000: add 'Audio Switch' block implementation 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 7f7d5239f56cfd25ba8470354d9705815c079157 Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Date: Mon, 19 Feb 2018 10:03:45 +0100 Subject: [PATCH] ASoC: sgtl5000: add 'Audio Switch' block implementation Add mux for choosing DAC source: ADC, I2S or DAP. DAP - digital audio processing block. Moreover create choosing mux for DAP main source (ADC or I2S) and DAP mixer source (ADC or I2S). Mixer block allows summing DAP main and mixer sources togheter with defined volumes. Signed-off-by: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 78 ++++++++++++++++++++++++++++++++++++++------- 1 file changed, 67 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 05592ddcf0e6..1724c4622052 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -216,17 +216,54 @@ static SOC_ENUM_SINGLE_DECL(adc_enum, static const struct snd_kcontrol_new adc_mux = SOC_DAPM_ENUM("Capture Mux", adc_enum); +/* input sources for headphone */ +static const char *hp_mux_text[] = { + "DAC", "LINE_IN" +}; + +static SOC_ENUM_SINGLE_DECL(hp_enum, + SGTL5000_CHIP_ANA_CTRL, 6, + hp_mux_text); + +static const struct snd_kcontrol_new hp_mux = +SOC_DAPM_ENUM("Headphone Mux", hp_enum); + /* input sources for DAC */ static const char *dac_mux_text[] = { - "DAC", "LINE_IN" + "ADC", "I2S", "Rsvrd", "DAP" }; static SOC_ENUM_SINGLE_DECL(dac_enum, - SGTL5000_CHIP_ANA_CTRL, 6, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAC_SEL_SHIFT, dac_mux_text); static const struct snd_kcontrol_new dac_mux = -SOC_DAPM_ENUM("Headphone Mux", dac_enum); +SOC_DAPM_ENUM("Digital Input Mux", dac_enum); + +/* input sources for DAP */ +static const char *dap_mux_text[] = { + "ADC", "I2S" +}; + +static SOC_ENUM_SINGLE_DECL(dap_enum, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAP_SEL_SHIFT, + dap_mux_text); + +static const struct snd_kcontrol_new dap_mux = +SOC_DAPM_ENUM("DAP Mux", dap_enum); + +/* input sources for DAP mix */ +static const char *dapmix_mux_text[] = { + "ADC", "I2S" +}; + +static SOC_ENUM_SINGLE_DECL(dapmix_enum, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAP_MIX_SEL_SHIFT, + dapmix_mux_text); + +static const struct snd_kcontrol_new dapmix_mux = +SOC_DAPM_ENUM("DAP MIX Mux", dapmix_enum); + static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LINE_IN"), @@ -243,7 +280,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), - SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &hp_mux), + SND_SOC_DAPM_MUX("Digital Input Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + SND_SOC_DAPM_MUX("DAP Mux", SGTL5000_DAP_CTRL, 0, 0, &dap_mux), + SND_SOC_DAPM_MUX("DAP MIX Mux", SGTL5000_DAP_CTRL, 4, 0, &dapmix_mux), + SND_SOC_DAPM_MIXER("DAP", SGTL5000_CHIP_DIG_POWER, 4, 0, NULL, 0), + /* aif for i2s input */ SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", @@ -270,7 +312,19 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"DAP Mux", "ADC", "ADC"}, /* adc --> DAP mux */ + {"DAP Mux", NULL, "AIFIN"}, /* i2s --> DAP mux */ + {"DAP", NULL, "DAP Mux"}, /* DAP mux --> dap */ + + {"DAP MIX Mux", "ADC", "ADC"}, /* adc --> DAP MIX mux */ + {"DAP MIX Mux", NULL, "AIFIN"}, /* i2s --> DAP MIX mux */ + {"DAP", NULL, "DAP MIX Mux"}, /* DAP MIX mux --> dap */ + + {"Digital Input Mux", "ADC", "ADC"}, /* adc --> audio mux */ + {"Digital Input Mux", NULL, "AIFIN"}, /* i2s --> audio mux */ + {"Digital Input Mux", NULL, "DAP"}, /* dap --> audio mux */ + {"DAC", NULL, "Digital Input Mux"}, /* audio mux --> dac */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ @@ -463,6 +517,9 @@ static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0) ); +/* tlv for DAP channels, 0% - 100% - 200% */ +static const DECLARE_TLV_DB_SCALE(dap_volume, 0, 1, 0); + /* tlv for hp volume, -51.5db to 12.0db, step .5db */ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); @@ -514,6 +571,11 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + SOC_SINGLE_TLV("DAP Main channel", SGTL5000_DAP_MAIN_CHAN, + 0, 0xffff, 0, dap_volume), + + SOC_SINGLE_TLV("DAP Mix channel", SGTL5000_DAP_MIX_CHAN, + 0, 0xffff, 0, dap_volume), /* Automatic Volume Control (DAP AVC) */ SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), @@ -1217,12 +1279,6 @@ static int sgtl5000_probe(struct snd_soc_component *component) /* disable short cut detector */ snd_soc_component_write(component, SGTL5000_CHIP_SHORT_CTRL, 0); - /* - * set i2s as default input of sound switch - * TODO: add sound switch to control and dapm widge. - */ - snd_soc_component_write(component, SGTL5000_CHIP_SSS_CTRL, - SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); snd_soc_component_write(component, SGTL5000_CHIP_DIG_POWER, SGTL5000_ADC_EN | SGTL5000_DAC_EN); -- 2.16.1

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[alsa-devel] Applied "ASoC: sgtl5000: add 5 band graphic equalizer" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: sgtl5000: add 5 band graphic equalizer 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 92a2742f9813ef65da19145a77e8067d6abbc822 Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Date: Mon, 19 Feb 2018 10:03:46 +0100 Subject: [PATCH] ASoC: sgtl5000: add 5 band graphic equalizer Enable 5-band graphic equalizer which allows manipulation of gain in range -11.75dB to 12dB for 5 specific sound bands: 115Hz, 330Hz, 990Hz, 3000Hz, 9900Hz. Signed-off-by: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 24 +++++++++++++++++++++--- sound/soc/codecs/sgtl5000.h | 7 +++++++ 2 files changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1724c4622052..7c1d65830c05 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -520,6 +520,9 @@ static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, /* tlv for DAP channels, 0% - 100% - 200% */ static const DECLARE_TLV_DB_SCALE(dap_volume, 0, 1, 0); +/* tlv for bass bands, -11.75db to 12.0db, step .25db */ +static const DECLARE_TLV_DB_SCALE(bass_band, -1175, 25, 0); + /* tlv for hp volume, -51.5db to 12.0db, step .5db */ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); @@ -585,6 +588,21 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, 0, 96, 0, avc_get_threshold, avc_put_threshold, avc_threshold), + + SOC_SINGLE_TLV("BASS 0", SGTL5000_DAP_EQ_BASS_BAND0, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 1", SGTL5000_DAP_EQ_BASS_BAND1, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 2", SGTL5000_DAP_EQ_BASS_BAND2, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 3", SGTL5000_DAP_EQ_BASS_BAND3, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 4", SGTL5000_DAP_EQ_BASS_BAND4, + 0, 0x5F, 0, bass_band), }; /* mute the codec used by alsa core */ @@ -1303,11 +1321,11 @@ static int sgtl5000_probe(struct snd_soc_component *component) SGTL5000_BIAS_VOLT_MASK, sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* - * disable DAP + * enable DAP Graphic EQ * TODO: - * Enable DAP in kcontrol and dapm. + * Add control for changing between PEQ/Tone Control/GEQ */ - snd_soc_component_write(component, SGTL5000_DAP_CTRL, 0); + snd_soc_component_write(component, SGTL5000_DAP_AUDIO_EQ, SGTL5000_DAP_SEL_GEQ); /* Unmute DAC after start */ snd_soc_component_update_bits(component, SGTL5000_CHIP_ADCDAC_CTRL, diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 22f3442af982..28cf637155bb 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -399,4 +399,11 @@ #define SGTL5000_SYSCLK 0x00 #define SGTL5000_LRCLK 0x01 +/* + * SGTL5000_DAP_AUDIO_EQ + */ +#define SGTL5000_DAP_SEL_PEQ 1 +#define SGTL5000_DAP_SEL_TONE_CTRL 2 +#define SGTL5000_DAP_SEL_GEQ 3 + #endif -- 2.16.1

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[alsa-devel] Applied "ASoC: rt5651: Fix regcache sync errors on resume" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: rt5651: Fix regcache sync errors on resume 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 2d30e9494f1ea320aaaad0cff9ddd92c87eac355 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:44 +0100 Subject: [PATCH] ASoC: rt5651: Fix regcache sync errors on resume The ALC5651 does not like multi-write accesses, avoid them. This fixes: rt5651 i2c-10EC5651:00: Unable to sync registers 0x27-0x28. -121 Errors on resume (and all registers after the registers in the error not being synced). Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/rt5651.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 831b297978a4..45a73049cf64 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1722,6 +1722,7 @@ static const struct regmap_config rt5651_regmap = { .num_reg_defaults = ARRAY_SIZE(rt5651_reg), .ranges = rt5651_ranges, .num_ranges = ARRAY_SIZE(rt5651_ranges), + .use_single_rw = true, }; #if defined(CONFIG_OF) -- 2.16.1

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[alsa-devel] [PATCH v3] ASoC: dmic: Add optional wakeup delay
by Matthias Kaehlcke 19 Feb '18

19 Feb '18

On some systems a delay is needed after switching on the clocks, to allow the output to stabilize and avoid a popping noise at the beginning of the recording. Add the optional device tree property 'wakeup-delay-ms' and apply the specified delay after enabling the mic. A blocking delay can't be applied in dmic_daiops_trigger() since the function is called in atomic context. Instead use a DAPM event handler to set the enable GPIO and apply the delay in the handler. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v3: - rebased on latest dmic topic branch Changes in v2: - use DAPM event handler instead of _prepare() and get rid of _trigger() - skip error check for optional 'wakeup-delay-ms' property - updated commit message .../devicetree/bindings/sound/dmic.txt | 2 + sound/soc/codecs/dmic.c | 63 ++++++++++--------- 2 files changed, 37 insertions(+), 28 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt index f7bf65611453..e957b4136716 100644 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -8,6 +8,7 @@ Required properties: Optional properties: - dmicen-gpios: GPIO specifier for dmic to control start and stop - num-channels: Number of microphones on this DAI + - wakeup-delay-ms: Delay (in ms) after enabling the DMIC Example node: @@ -15,4 +16,5 @@ Example node: compatible = "dmic-codec"; dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; num-channels = <1>; + wakeup-delay-ms <50>; }; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 32a8f71d51f1..bac5bbb2dd59 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,7 @@ * */ +#include <linux/delay.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> #include <linux/platform_device.h> @@ -29,34 +30,33 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -static int dmic_daiops_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); +struct dmic { + struct gpio_desc *gpio_en; + int wakeup_delay; +}; - if (!dmic_en) - return 0; +static int dmic_aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct dmic *dmic = snd_soc_codec_get_drvdata(codec); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(dmic_en, 1); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 1); + + if (dmic->wakeup_delay) + msleep(dmic->wakeup_delay); break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(dmic_en, 0); + case SND_SOC_DAPM_POST_PMD: + if (dmic->gpio_en) + gpiod_set_value(dmic->gpio_en, 0); break; } return 0; } -static const struct snd_soc_dai_ops dmic_dai_ops = { - .trigger = dmic_daiops_trigger, -}; - static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -68,26 +68,33 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &dmic_dai_ops, }; static int dmic_component_probe(struct snd_soc_component *component) { - struct gpio_desc *dmic_en; + struct dmic *dmic; + + dmic = devm_kzalloc(component->dev, sizeof(*dmic), GFP_KERNEL); + if (!dmic) + return -ENOMEM; + + dmic->gpio_en = devm_gpiod_get_optional(component->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic->gpio_en)) + return PTR_ERR(dmic->gpio_en); - dmic_en = devm_gpiod_get_optional(component->dev, - "dmicen", GPIOD_OUT_LOW); - if (IS_ERR(dmic_en)) - return PTR_ERR(dmic_en); + device_property_read_u32(component->dev, "wakeup-delay-ms", + &dmic->wakeup_delay); - snd_soc_component_set_drvdata(component, dmic_en); + snd_soc_component_set_drvdata(component, dmic); return 0; } static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { - SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("DMIC AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0, dmic_aif_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_INPUT("DMic"), }; -- 2.16.1.291.g4437f3f132-goog

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix compiler warning -Wmaybe-uninitialized
by Kirill Marinushkin 19 Feb '18

19 Feb '18

Configuration: SND_SOC_INTEL_SKYLAKE=y PM_SLEEP=y Warning message: sound/soc/intel/skylake/skl.c: In function 'skl_resume': sound/soc/intel/skylake/skl.c:326:6: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] int ret; ^~~ Fixes: 4557c305d4fc ("ASoC: Intel: Skylake: Add support for active suspend") Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Guneshwor Singh <guneshwor.o.singh(a)intel.com> Cc: Naveen Manohar <naveen.m(a)intel.com> Cc: Harsha Priya N <harshapriya.n(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/intel/skylake/skl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 31d8634e8aa1..273a9ab75489 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -323,7 +323,7 @@ static int skl_resume(struct device *dev) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); struct hdac_ext_link *hlink = NULL; - int ret; + int ret = 0; /* Turned OFF in HDMI codec driver after codec reconfiguration */ if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { -- 2.13.6

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[alsa-devel] [PATCH 0/1] Use of shared resource in designware
by Akshu Agrawal 19 Feb '18

19 Feb '18

This patch will allow the driver to map same registers to multiple instances of driver. It is useful when playback and capture have same regsiters and are required to be present on separate cpu dais. This patch is dependant on "Add devm_ioremap_shared_resource()" patch. https://patchwork.kernel.org/patch/10177127/ Akshu Agrawal (1): ASoC: dwc: Use devm_ioremap_shared_resource include/sound/designware_i2s.h | 1 + sound/soc/dwc/dwc-i2s.c | 18 +++++++++++++++--- 2 files changed, 16 insertions(+), 3 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] Makefile.am: Add missing code for enhanced configure option of alsa plugins
by Lars Wendler 18 Feb '18

18 Feb '18

Commit 1201e158fb00bf15c353d088c94c1cb3eff9653c added the pieces for configure but did not do the same for Makefile. As result all plugins were still built regardless of corresponding --disable-* option being called with configure. --- Makefile.am | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/Makefile.am b/Makefile.am index 69cfe0d..76cefef 100644 --- a/Makefile.am +++ b/Makefile.am @@ -1,4 +1,4 @@ -SUBDIRS = oss mix usb_stream arcam-av doc +SUBDIRS = doc if HAVE_JACK SUBDIRS += jack endif @@ -20,6 +20,18 @@ endif if HAVE_SPEEXDSP SUBDIRS += speex endif +if HAVE_OSS +SUBDIRS += oss +endif +if HAVE_MIX +SUBDIRS += mix +endif +if HAVE_USBSTREAM +SUBDIRS += usb_stream +endif +if HAVE_ARCAMAV +SUBDIRS += arcam-av +endif EXTRA_DIST = gitcompile version COPYING.GPL m4/attributes.m4 AUTOMAKE_OPTIONS = foreign -- 2.16.2

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[alsa-devel] [asoc:for-next 19/53] sound/soc//amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type
by kbuild test robot 16 Feb '18

16 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 5372fcd763f83d0e4d38ac008a2076f36664cbae commit: b2eeca7e9dc8f413d446f397a0220e21d6eb01d1 [19/53] Merge remote-tracking branches 'asoc/topic/ak5386', 'asoc/topic/ak5558', 'asoc/topic/alc5623', 'asoc/topic/alc5632' and 'asoc/topic/amd' into asoc-next config: x86_64-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout b2eeca7e9dc8f413d446f397a0220e21d6eb01d1 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc//amd/acp-da7219-max98357a.c: In function 'cz_da7219_init': >> sound/soc//amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type [-Werror=incompatible-pointer-types] da7219_aad_jack_det(codec, &cz_jack); ^~~~~ In file included from sound/soc//amd/acp-da7219-max98357a.c:38:0: sound/soc//amd/../codecs/da7219-aad.h:209:6: note: expected 'struct snd_soc_component *' but argument is of type 'struct snd_soc_codec *' void da7219_aad_jack_det(struct snd_soc_component *component, struct snd_soc_jack *jack); ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors sparse warnings: (new ones prefixed by >>) >> sound/soc/amd/acp-da7219-max98357a.c:79:29: sparse: incorrect type in argument 1 (different base types) @@ expected struct snd_soc_component @@ got ct snd_soc_component @@ sound/soc/amd/acp-da7219-max98357a.c:79:29: expected struct snd_soc_component sound/soc/amd/acp-da7219-max98357a.c:79:29: got struct snd_soc_codec sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_da7219_init': sound/soc/amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type da7219_aad_jack_det(codec, &cz_jack); ^~~~~ In file included from sound/soc/amd/acp-da7219-max98357a.c:38:0: sound/soc/amd/../codecs/da7219-aad.h:209:6: note: expected 'struct snd_soc_component but argument is of type 'struct snd_soc_codec void da7219_aad_jack_det(struct snd_soc_component struct snd_soc_jack ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/da7219_aad_jack_det +79 sound/soc//amd/acp-da7219-max98357a.c 608a300f Akshu Agrawal 2018-02-16 45 608a300f Akshu Agrawal 2018-02-16 46 static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) 608a300f Akshu Agrawal 2018-02-16 47 { 608a300f Akshu Agrawal 2018-02-16 48 int ret; 608a300f Akshu Agrawal 2018-02-16 49 struct snd_soc_card *card = rtd->card; 608a300f Akshu Agrawal 2018-02-16 50 struct snd_soc_codec *codec = rtd->codec; 608a300f Akshu Agrawal 2018-02-16 51 struct snd_soc_dai *codec_dai = rtd->codec_dai; 608a300f Akshu Agrawal 2018-02-16 52 608a300f Akshu Agrawal 2018-02-16 53 dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); 608a300f Akshu Agrawal 2018-02-16 54 608a300f Akshu Agrawal 2018-02-16 55 ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 608a300f Akshu Agrawal 2018-02-16 56 CZ_PLAT_CLK, SND_SOC_CLOCK_IN); 608a300f Akshu Agrawal 2018-02-16 57 if (ret < 0) { 608a300f Akshu Agrawal 2018-02-16 58 dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); 608a300f Akshu Agrawal 2018-02-16 59 return ret; 608a300f Akshu Agrawal 2018-02-16 60 } 608a300f Akshu Agrawal 2018-02-16 61 608a300f Akshu Agrawal 2018-02-16 62 ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL, 608a300f Akshu Agrawal 2018-02-16 63 CZ_PLAT_CLK, MCLK_RATE); 608a300f Akshu Agrawal 2018-02-16 64 if (ret < 0) { 608a300f Akshu Agrawal 2018-02-16 65 dev_err(rtd->dev, "can't set codec pll: %d\n", ret); 608a300f Akshu Agrawal 2018-02-16 66 return ret; 608a300f Akshu Agrawal 2018-02-16 67 } 608a300f Akshu Agrawal 2018-02-16 68 608a300f Akshu Agrawal 2018-02-16 69 ret = snd_soc_card_jack_new(card, "Headset Jack", 608a300f Akshu Agrawal 2018-02-16 70 SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | 608a300f Akshu Agrawal 2018-02-16 71 SND_JACK_BTN_0 | SND_JACK_BTN_1 | 608a300f Akshu Agrawal 2018-02-16 72 SND_JACK_BTN_2 | SND_JACK_BTN_3, 608a300f Akshu Agrawal 2018-02-16 73 &cz_jack, NULL, 0); 608a300f Akshu Agrawal 2018-02-16 74 if (ret) { 608a300f Akshu Agrawal 2018-02-16 75 dev_err(card->dev, "HP jack creation failed %d\n", ret); 608a300f Akshu Agrawal 2018-02-16 76 return ret; 608a300f Akshu Agrawal 2018-02-16 77 } 608a300f Akshu Agrawal 2018-02-16 78 608a300f Akshu Agrawal 2018-02-16 @79 da7219_aad_jack_det(codec, &cz_jack); 608a300f Akshu Agrawal 2018-02-16 80 608a300f Akshu Agrawal 2018-02-16 81 return 0; 608a300f Akshu Agrawal 2018-02-16 82 } 608a300f Akshu Agrawal 2018-02-16 83 :::::: The code at line 79 was first introduced by commit :::::: 608a300fc1f0e088446cbef8e13ded686c5172b6 ASoC: AMD: Add machine driver for ST DA7219 MAX98357 :::::: TO: Akshu Agrawal <akshu.agrawal(a)amd.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: wm_adsp: Fix some signedness errors in register access
by Richard Fitzgerald 16 Feb '18

16 Feb '18

base and sysclk_reg should be unsigned int, as that is what regmap takes as a register address. sysclk_mask is used to mask unsigned register values so should be unsigned. sysclk_shift is a shift value so is not allowed to be negative. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41cc11c19b83..d4042c95315e 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -64,10 +64,10 @@ struct wm_adsp { struct regmap *regmap; struct snd_soc_codec *codec; - int base; - int sysclk_reg; - int sysclk_mask; - int sysclk_shift; + unsigned int base; + unsigned int sysclk_reg; + unsigned int sysclk_mask; + unsigned int sysclk_shift; struct list_head alg_regions; -- 2.11.0

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