February 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: codecs: make PCM5102A codec selectable
by David Douard 24 Feb '20

24 Feb '20

The PCM5102A codec entry in the Kconfig file lacks a title/description to make it selectable in menuconfig (since it has no dependent i2s/spi sub-modules like e.g. the pcm512x one). --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7e90f5d83097..676fa62e9b7f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -934,7 +934,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.20.1

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[PATCH 1/2] mfd: wm8994: Fix driver operation if loaded as modules
by Marek Szyprowski 24 Feb '20

24 Feb '20

WM8994 chip has built-in regulators, which might be used for chip operation. They are controlled by a separate wm8994-regulator driver, which should be loaded before this driver calls regulator_get(), because that driver also provides consumer-supply mapping for the them. If that driver is not yet loaded, regulator core substitute them with dummy regulator, what breaks chip operation, because the built-in regulators are never enabled. Fix this by adding a comment and a call to module_request() helper. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- drivers/mfd/wm8994-core.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 1e9fe7d92597..f15f12d8bccc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -321,6 +321,13 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) int ret, i, patch_regs = 0; int pulls = 0; + /* + * Request a driver for built-in regulators first, because they are + * default supplies for "AVDD1" and "DCVDD". Otherwise, getting them + * fails due to lack of consumer mapping defined by that driver. + */ + request_module("wm8994_regulator"); + if (dev_get_platdata(wm8994->dev)) { pdata = dev_get_platdata(wm8994->dev); wm8994->pdata = *pdata; -- 2.17.1

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[PATCH for-next 1/2] ASoC: tlv320adcx140: Add DRE and AGC support
by Dan Murphy 24 Feb '20

24 Feb '20

The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined in Section 8.3.2 of the data sheets. The DRE achieves a complete-channel dynamic range as high as 120 dB. At a system level, the DRE scheme enables far-field, high-fidelity recording of audio signals in very quiet environments and low-distortion recording in loud environments. There are 2 enables for DRE. The first is a global setting that enables the DRE engine in the device and the other enable is per channel. If the DRE is enabled globally then either DRE or AGC can be used per each configured channel. If global DRE is disabled then even setting the DRE enable bit in the channel config register will have no effect. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 55 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 1 + 2 files changed, 56 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8182c584de9c..105e51be6fe6 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -108,6 +108,7 @@ static const struct reg_default adcx140_reg_defaults[] = { { ADCX140_DSP_CFG0, 0x01 }, { ADCX140_DSP_CFG1, 0x40 }, { ADCX140_DRE_CFG0, 0x7b }, + { ADCX140_AGC_CFG0, 0xe7 }, { ADCX140_IN_CH_EN, 0xf0 }, { ADCX140_ASI_OUT_CH_EN, 0x00 }, { ADCX140_PWR_CFG, 0x00 }, @@ -158,6 +159,16 @@ static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); +/* DRE Level. From -12 dB to -66 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_thresh_tlv, -6600, 100, 0); +/* DRE Max Gain. From 2 dB to 26 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_gain_tlv, 200, 200, 0); + +/* AGC Level. From -6 dB to -36 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0); +/* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0); + static const char * const resistor_text[] = { "2.5 kOhm", "10 kOhm", "20 kOhm" }; @@ -281,6 +292,18 @@ static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch2_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH2_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch3_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH3_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0); + +static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0); + /* Output Mixer */ static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = { SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0), @@ -361,6 +384,18 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch4_en_switch), + SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_dre_en_switch), + + SND_SOC_DAPM_SWITCH("CH1_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch1_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH2_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch2_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH3_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch3_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH4_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch4_dre_en_switch), + SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, in1_resistor_controls), SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, @@ -383,6 +418,16 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"DRE_ENABLE", "Switch", "CH1_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH2_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH3_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH4_DRE_EN"}, + + {"CH1_DRE_EN", "Switch", "CH1_ADC"}, + {"CH2_DRE_EN", "Switch", "CH2_ADC"}, + {"CH3_DRE_EN", "Switch", "CH3_ADC"}, + {"CH4_DRE_EN", "Switch", "CH4_ADC"}, + /* Mic input */ {"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"}, {"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"}, @@ -455,6 +500,16 @@ static const struct snd_kcontrol_new adcx140_snd_controls[] = { SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0, adc_tlv), + SOC_SINGLE_TLV("DRE Threshold", ADCX140_DRE_CFG0, 4, 9, 0, + dre_thresh_tlv), + SOC_SINGLE_TLV("DRE Max Gain", ADCX140_DRE_CFG0, 0, 12, 0, + dre_gain_tlv), + + SOC_SINGLE_TLV("AGC Threshold", ADCX140_AGC_CFG0, 4, 15, 0, + agc_thresh_tlv), + SOC_SINGLE_TLV("AGC Max Gain", ADCX140_AGC_CFG0, 0, 13, 0, + agc_gain_tlv), + SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2, 0, 0xff, 0, dig_vol_tlv), SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2, diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 66b1c3b33f1e..6d055e55909e 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -84,6 +84,7 @@ #define ADCX140_DSP_CFG0 0x6b #define ADCX140_DSP_CFG1 0x6c #define ADCX140_DRE_CFG0 0x6d +#define ADCX140_AGC_CFG0 0x70 #define ADCX140_IN_CH_EN 0x73 #define ADCX140_ASI_OUT_CH_EN 0x74 #define ADCX140_PWR_CFG 0x75 -- 2.25.0

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Applied "ASoC: meson: aiu: fix semicolon.cocci warnings" to the asoc tree
by Mark Brown 24 Feb '20

24 Feb '20

The patch ASoC: meson: aiu: fix semicolon.cocci warnings 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 1640c8df0bbac9e5156132e24c0f0a932c2b2865 Mon Sep 17 00:00:00 2001 From: kbuild test robot <lkp(a)intel.com> Date: Sun, 23 Feb 2020 01:01:54 +0800 Subject: [PATCH] ASoC: meson: aiu: fix semicolon.cocci warnings sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Fixes: 3e25c44598aa ("ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families") Signed-off-by: kbuild test robot <lkp(a)intel.com> CC: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Link: https://lore.kernel.org/r/20200222170154.GA119396@e50d7db646c3 Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/aiu-encoder-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index cc73b5d5c2b7..832e22d275fe 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -126,7 +126,7 @@ static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, default: dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); return -EINVAL; - }; + } snd_soc_component_update_bits(component, AIU_CLK_CTRL, AIU_CLK_CTRL_I2S_DIV, -- 2.20.1

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Re: [PATCH] docs: dt: fix several broken doc references
by Mark Brown 24 Feb '20

24 Feb '20

On Sun, Feb 23, 2020 at 09:59:53AM +0100, Mauro Carvalho Chehab wrote: > There are several DT doc references that require manual fixes. > I found 3 cases fixed on this patch: > > - directory named "binding/" instead of "bindings/"; > - .txt to .yaml renames; > - file renames (still on txt format); This seems like it should've been split up a bit. :/ Acked-by: Mark Brown <broonie(a)kernel.org>

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[alsa-devel] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 24 Feb '20

24 Feb '20

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..f1f25704fe52 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -36,7 +36,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484ab3c2ad67..1a0b55beb282 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -20,13 +20,12 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S tristate diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1b41c0f2a8fb..80e4757176d8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -219,6 +212,9 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -231,10 +227,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -245,8 +245,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

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Re: [PATCH] Intel: Skylake: Fix inconsistent IS_ERR and PTR_ERR
by Joe Perches 24 Feb '20

24 Feb '20

On Fri, 2020-02-21 at 18:11 +0800, Xu Wang wrote: > PTR_ERR should access the value just tested by IS_ERR. > In skl_clk_dev_probe(),it is inconsistent. [] > diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c [] > @@ -384,7 +384,7 @@ static int skl_clk_dev_probe(struct platform_device *pdev) > &clks[i], clk_pdata, i); > > if (IS_ERR(data->clk[data->avail_clk_cnt])) { > - ret = PTR_ERR(data->clk[data->avail_clk_cnt++]); > + ret = PTR_ERR(data->clk[data->avail_clk_cnt]); NAK. This is not inconsistent and you are removing the ++ which is a post increment. Likely that is necessary. You could write the access and the increment as two separate statements if it confuses you.

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Re: [PATCH] docs: dt: fix several broken doc references
by Miquel Raynal 24 Feb '20

24 Feb '20

Hi Mauro, Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> wrote on Sun, 23 Feb 2020 09:59:53 +0100: > There are several DT doc references that require manual fixes. > I found 3 cases fixed on this patch: > > - directory named "binding/" instead of "bindings/"; > - .txt to .yaml renames; > - file renames (still on txt format); > > Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> > --- > .../devicetree/bindings/mtd/cadence-nand-controller.txt | 2 +- > .../devicetree/bindings/net/brcm,bcm7445-switch-v4.0.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-sai.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt | 2 +- > Documentation/devicetree/bindings/spi/st,stm32-spi.yaml | 2 +- > MAINTAINERS | 4 ++-- > .../devicetree/bindings/net/wireless/siliabs,wfx.txt | 2 +- > 7 files changed, 8 insertions(+), 8 deletions(-) For the Cadence file, Reviewed-by: Miquel Raynal <miquel.raynal(a)bootlin.com> Thanks, Miquèl

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[alsa-devel] [PATCH 0/8] ASoC: sun50i-codec-analog: Cleanup and power management
by Samuel Holland 23 Feb '20

23 Feb '20

This series performs some minor cleanup on the driver for the analog codec in the Allwinner A64, and hooks up the existing mute switches to DAPM widgets, in order to provide improved power management. Samuel Holland (8): ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend ASoC: sun50i-codec-analog: Group and sort mixer routes ASoC: sun50i-codec-analog: Make headphone routes stereo ASoC: sun50i-codec-analog: Enable DAPM for headphone switch ASoC: sun50i-codec-analog: Make line out routes stereo ASoC: sun50i-codec-analog: Enable DAPM for line out switch ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch sound/soc/sunxi/sun50i-codec-analog.c | 174 ++++++++++++++++---------- 1 file changed, 109 insertions(+), 65 deletions(-) -- 2.24.1

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alsa-lib 1.2.2 breaks Shovel Knight, 1.2.1.2 is still working.
by GitHub issues - edited 22 Feb '20

22 Feb '20

alsa-project/alsa-lib issue #27 was edited from DarkWav: So, today after upgrading to alsa-lib 1.2.2 I noticed the game "Shovel Knight" does no longer launch. Any Ideas why this could be happening? Sadly steam or the game executable won't provide any usable debug info when looking at terminal output :( . Can I maybe use GDB to gather more info or is that useless in this case? Issue URL : https://github.com/alsa-project/alsa-lib/issues/27 Repository URL: https://github.com/alsa-project/alsa-lib

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