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

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

22 Feb '20

alsa-project/alsa-lib issue #27 was opened 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 :( . Issue URL : https://github.com/alsa-project/alsa-lib/issues/27 Repository URL: https://github.com/alsa-project/alsa-lib

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[asoc:for-5.7 134/139] sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon
by kbuild test robot 22 Feb '20

22 Feb '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.7 head: b38c4a8a0291c31a660cd77761202ebb18332fb7 commit: 3e25c44598aa44134207ad7b3c5ad6b586135777 [134/139] ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> coccinelle warnings: (new ones prefixed by >>) >> sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon Please review and possibly fold the followup patch. --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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amixer not setting Master volume at a given percentage value (check with 85% to 90%)
by GitHub issues - opened 22 Feb '20

22 Feb '20

alsa-project/alsa-utils issue #35 was opened from immortalaxay: I am using Ubuntu 18.04 (alsa-utils/amixer version 1.1.3) and trying to set volume using amixer command but amixer not setting Master volume at the given percentage value. $ amixer -M -c PCH sset Master **85**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 81 [**83**%] [-4.50dB] [on] $ amixer -M -c PCH sset Master **86**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 82 [**85**%] [-3.75dB] [on] $ amixer -M -c PCH sset Master **87**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 82 [**85**%] [-3.75dB] [on] $ amixer -M -c PCH sset Master **88**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 82 [**85**%] [-3.75dB] [on] $ amixer -M -c PCH sset Master **89**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 83 [**88**%] [-3.00dB] [on] $ amixer -M -c PCH sset Master **90**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 83 [**88**%] [-3.00dB] [on] Issue URL : https://github.com/alsa-project/alsa-utils/issues/35 Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH] ASoC: ti: Allocate dais dynamically for TDM and audio graph card
by Tony Lindgren 21 Feb '20

21 Feb '20

We can have multiple connections on a single McBSP instance configured with audio graph card when using TDM (Time Division Multiplexing). Let's allow that by configuring dais dynamically. See Documentation/devicetree/bindings/sound/audio-graph-card.txt and Documentation/devicetree/bindings/graph.txt for more details for multiple endpoints. I've tested this with droid4 where cpcap pmic and modem voice are both both wired to mcbsp3. I've also tested this on droid4 both with and without the pending modem audio codec driver that is waiting for n_gsm serdev dependencies to clear. Cc: Aaro Koskinen <aaro.koskinen(a)iki.fi> Cc: Arthur D. <spinal.by(a)gmail.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Merlijn Wajer <merlijn(a)wizzup.org> Cc: Pavel Machek <pavel(a)ucw.cz> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Sebastian Reichel <sre(a)kernel.org> Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- sound/soc/ti/omap-mcbsp-priv.h | 2 + sound/soc/ti/omap-mcbsp.c | 76 ++++++++++++++++++++++++---------- 2 files changed, 55 insertions(+), 23 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -262,6 +262,8 @@ struct omap_mcbsp { struct omap_mcbsp_platform_data *pdata; struct omap_mcbsp_st_data *st_data; struct omap_mcbsp_reg_cfg cfg_regs; + struct snd_soc_dai_driver *dais; + int dai_count; struct snd_dmaengine_dai_dma_data dma_data[2]; unsigned int dma_req[2]; int dma_op_mode; diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -14,6 +14,7 @@ #include <linux/pm_runtime.h> #include <linux/of.h> #include <linux/of_device.h> +#include <linux/of_graph.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1304,23 +1305,53 @@ static int omap_mcbsp_remove(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_driver omap_mcbsp_dai = { - .probe = omap_mcbsp_probe, - .remove = omap_mcbsp_remove, - .playback = { - .channels_min = 1, - .channels_max = 16, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 16, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, - .ops = &mcbsp_dai_ops, -}; +static int omap_mcbsp_init_dais(struct omap_mcbsp *mcbsp) +{ + struct device_node *np = mcbsp->dev->of_node; + int i; + + if (np) + mcbsp->dai_count = of_graph_get_endpoint_count(np); + + if (!mcbsp->dai_count) + mcbsp->dai_count = 1; + + mcbsp->dais = devm_kcalloc(mcbsp->dev, mcbsp->dai_count, + sizeof(*mcbsp->dais), GFP_KERNEL); + if (!mcbsp->dais) + return -ENOMEM; + + for (i = 0; i < mcbsp->dai_count; i++) { + struct snd_soc_dai_driver *dai = &mcbsp->dais[i]; + + dai->name = devm_kasprintf(mcbsp->dev, GFP_KERNEL, "%s-dai%i", + dev_name(mcbsp->dev), i); + + if (i == 0) { + dai->probe = omap_mcbsp_probe; + dai->remove = omap_mcbsp_remove; + dai->ops = &mcbsp_dai_ops; + } + dai->playback.channels_min = 1; + dai->playback.channels_max = 16; + dai->playback.rates = OMAP_MCBSP_RATES; + if (mcbsp->pdata->reg_size == 2) + dai->playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + else + dai->playback.formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE; + dai->capture.channels_min = 1; + dai->capture.channels_max = 16; + dai->capture.rates = OMAP_MCBSP_RATES; + if (mcbsp->pdata->reg_size == 2) + dai->capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + else + dai->capture.formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE; + } + + return 0; +} static const struct snd_soc_component_driver omap_mcbsp_component = { .name = "omap-mcbsp", @@ -1409,18 +1440,17 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) mcbsp->dev = &pdev->dev; platform_set_drvdata(pdev, mcbsp); - ret = omap_mcbsp_init(pdev); + ret = omap_mcbsp_init_dais(mcbsp); if (ret) return ret; - if (mcbsp->pdata->reg_size == 2) { - omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; - omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; - } + ret = omap_mcbsp_init(pdev); + if (ret) + return ret; ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, - &omap_mcbsp_dai, 1); + mcbsp->dais, mcbsp->dai_count); if (ret) return ret; -- 2.25.0

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Applied "ASoC: meson: g12a: add toacodec dt-binding documentation" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch ASoC: meson: g12a: add toacodec dt-binding documentation 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 bd56e593da19de22284951c33ce5c419258171bf Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Fri, 21 Feb 2020 16:36:05 +0100 Subject: [PATCH] ASoC: meson: g12a: add toacodec dt-binding documentation Add the DT bindings and documentation of the internal audio DAC glue found on Amlogic g12a and sm1 SoC families Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200221153607.1585499-2-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 +++++++++++++++++++ .../dt-bindings/sound/meson-g12a-toacodec.h | 10 ++++ 2 files changed, 61 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml new file mode 100644 index 000000000000..f778d3371fde --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-toacodec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a Internal DAC Control Glue + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 1 + + compatible: + oneOf: + - items: + - const: + amlogic,g12a-toacodec + - items: + - enum: + - amlogic,sm1-toacodec + - const: + amlogic,g12a-toacodec + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - resets + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + toacodec: audio-controller@740 { + compatible = "amlogic,g12a-toacodec"; + reg = <0x0 0x740 0x0 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOACODEC>; + }; diff --git a/include/dt-bindings/sound/meson-g12a-toacodec.h b/include/dt-bindings/sound/meson-g12a-toacodec.h new file mode 100644 index 000000000000..69d7a75592a2 --- /dev/null +++ b/include/dt-bindings/sound/meson-g12a-toacodec.h @@ -0,0 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_MESON_G12A_TOACODEC_H +#define __DT_MESON_G12A_TOACODEC_H + +#define TOACODEC_IN_A 0 +#define TOACODEC_IN_B 1 +#define TOACODEC_IN_C 2 +#define TOACODEC_OUT 3 + +#endif /* __DT_MESON_G12A_TOACODEC_H */ -- 2.20.1

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[PATCH v2 0/3] ASoC: meson: g12a: add internal audio DAC support
by Jerome Brunet 21 Feb '20

21 Feb '20

Like the gxl, the Amlogic g12a and sm1 SoC families have a t9015 internal audio DAC. On these more recent SoCs, any of the 3 TDM outputs can be routed to the internal DAC. This routing is done by a small glue device called 'toacodec'. This patchset adds support for it. This was tested on the amlogic reference design g12a-u200. Changes since v1 [0]: * Fixup patch 2 which was left in an intermediate state after rebasing, missing part of the changes. Thanks to Sergey Bolshakov for reporting it. [0]: https://lore.kernel.org/r/20200221122242.1500093-1-jbrunet@baylibre.com Jerome Brunet (3): ASoC: meson: g12a: add toacodec dt-binding documentation ASoC: meson: g12a: add internal DAC glue driver ASoC: meson: axg-card: add toacodec support .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 ++++ .../dt-bindings/sound/meson-g12a-toacodec.h | 10 + sound/soc/meson/Kconfig | 9 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/g12a-toacodec.c | 252 ++++++++++++++++++ 6 files changed, 326 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h create mode 100644 sound/soc/meson/g12a-toacodec.c -- 2.24.1

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