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

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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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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