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

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

1 0

[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

1 1

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

1 0

[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

6 32

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

1 0

[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

2 4

Applied "ASoC: meson: axg-card: add toacodec support" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch ASoC: meson: axg-card: add toacodec support 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 b38c4a8a0291c31a660cd77761202ebb18332fb7 Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Fri, 21 Feb 2020 16:36:07 +0100 Subject: [PATCH] ASoC: meson: axg-card: add toacodec support Make sure the axg audio card driver recognise the dai_link as a codec-to-codec link if the cpu dai is the internal dac glue. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200221153607.1585499-4-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/axg-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 372dc696cc8e..48651631bdcf 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -303,7 +303,8 @@ static int axg_card_cpu_is_tdm_iface(struct device_node *np) static int axg_card_cpu_is_codec(struct device_node *np) { - return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx"); + return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx") || + of_device_is_compatible(np, DT_PREFIX "g12a-toacodec"); } static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, -- 2.20.1

1 0

Re: [PATCH 3/3] ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families
by Jerome Brunet 21 Feb '20

21 Feb '20

On Thu 20 Feb 2020 at 21:57, Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> wrote: > The AIU audio controller on the Meson8 and Meson8b SoC families is > compatible with the one found in the later GXBB family. Add compatible > strings for these two older SoC families so the driver can be loaded for > them. > > Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we > need to use the I2S divider from the AIU_CLK_CTRL register. This older > register is less flexible because it only supports four divider settings > (1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports > dividers in the range 0..64). > > Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> > --- > sound/soc/meson/Kconfig | 2 +- > sound/soc/meson/aiu-encoder-i2s.c | 92 +++++++++++++++++++++++-------- > sound/soc/meson/aiu.c | 9 +++ > sound/soc/meson/aiu.h | 1 + > 4 files changed, 81 insertions(+), 23 deletions(-) > > diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig > index 897a706dcda0..d27e9180b453 100644 > --- a/sound/soc/meson/Kconfig > +++ b/sound/soc/meson/Kconfig > @@ -10,7 +10,7 @@ config SND_MESON_AIU > imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI > help > Select Y or M to add support for the Audio output subsystem found > - in the Amlogic GX SoC family > + in the Amlogic Meson8, Meson8b and GX SoC families > > config SND_MESON_AXG_FIFO > tristate > diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c > index 4900e38e7e49..cc73b5d5c2b7 100644 > --- a/sound/soc/meson/aiu-encoder-i2s.c > +++ b/sound/soc/meson/aiu-encoder-i2s.c > @@ -111,34 +111,40 @@ static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component, > return 0; > } > > -static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > - struct snd_pcm_hw_params *params) > +static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params, > + unsigned int bs) > { > - struct aiu *aiu = snd_soc_component_get_drvdata(component); > - unsigned int srate = params_rate(params); > - unsigned int fs, bs; > - > - /* Get the oversampling factor */ > - fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); > + switch (bs) { > + case 1: > + case 2: > + case 4: > + case 8: > + /* These are the only valid legacy dividers */ > + break; > > - if (fs % 64) > + default: > + dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); > return -EINVAL; > + }; Not major but this semicolon should be removed (Thx coccinelle!) > > - /* Send data MSB first */ > - snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, > - AIU_I2S_DAC_CFG_MSB_FIRST, > - AIU_I2S_DAC_CFG_MSB_FIRST); > + snd_soc_component_update_bits(component, AIU_CLK_CTRL, > + AIU_CLK_CTRL_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, > + __ffs(bs))); > > - /* Set bclk to lrlck ratio */ > - snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, > - AIU_CODEC_DAC_LRCLK_CTRL_DIV, > - FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, > - 64 - 1)); > + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > + AIU_CLK_CTRL_MORE_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, > + 0)); > > - /* Use CLK_MORE for mclk to bclk divider */ > - snd_soc_component_update_bits(component, AIU_CLK_CTRL, > - AIU_CLK_CTRL_I2S_DIV, 0); > + return 0; > +} > > +static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params, > + unsigned int bs) > +{ > /* > * NOTE: this HW is odd. > * In most configuration, the i2s divider is 'mclk / blck'. > @@ -146,7 +152,6 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > * increased by 50% to get the correct output rate. > * No idea why ! > */ > - bs = fs / 64; > if (params_width(params) == 16 && params_channels(params) == 8) { > if (bs % 2) { > dev_err(component->dev, > @@ -156,11 +161,54 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > bs += bs / 2; > } > > + /* Use CLK_MORE for mclk to bclk divider */ > + snd_soc_component_update_bits(component, AIU_CLK_CTRL, > + AIU_CLK_CTRL_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0)); > + > snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > AIU_CLK_CTRL_MORE_I2S_DIV, > FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, > bs - 1)); > > + return 0; > +} > + > +static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params) > +{ > + struct aiu *aiu = snd_soc_component_get_drvdata(component); > + unsigned int srate = params_rate(params); > + unsigned int fs, bs; > + int ret; > + > + /* Get the oversampling factor */ > + fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); > + > + if (fs % 64) > + return -EINVAL; > + > + /* Send data MSB first */ > + snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, > + AIU_I2S_DAC_CFG_MSB_FIRST, > + AIU_I2S_DAC_CFG_MSB_FIRST); > + > + /* Set bclk to lrlck ratio */ > + snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, > + AIU_CODEC_DAC_LRCLK_CTRL_DIV, > + FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, > + 64 - 1)); > + > + bs = fs / 64; > + > + if (aiu->platform->has_clk_ctrl_more_i2s_div) > + ret = aiu_encoder_i2s_set_more_div(component, params, bs); > + else > + ret = aiu_encoder_i2s_set_legacy_div(component, params, bs); > + > + if (ret) > + return ret; > + > /* Make sure amclk is used for HDMI i2s as well */ > snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > AIU_CLK_CTRL_MORE_HDMI_AMCLK, > diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c > index 38209312a8c3..dc35ca79021c 100644 > --- a/sound/soc/meson/aiu.c > +++ b/sound/soc/meson/aiu.c > @@ -351,15 +351,24 @@ static int aiu_remove(struct platform_device *pdev) > > static const struct aiu_platform_data aiu_gxbb_pdata = { > .has_acodec = false, > + .has_clk_ctrl_more_i2s_div = true, > }; > > static const struct aiu_platform_data aiu_gxl_pdata = { > .has_acodec = true, > + .has_clk_ctrl_more_i2s_div = true, > +}; > + > +static const struct aiu_platform_data aiu_meson8_pdata = { > + .has_acodec = false, > + .has_clk_ctrl_more_i2s_div = false, > }; > > static const struct of_device_id aiu_of_match[] = { > { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, > { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, > + { .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata }, > + { .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata }, > {} > }; > MODULE_DEVICE_TABLE(of, aiu_of_match); > diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h > index ab003638d5e5..87aa19ac4af3 100644 > --- a/sound/soc/meson/aiu.h > +++ b/sound/soc/meson/aiu.h > @@ -29,6 +29,7 @@ struct aiu_interface { > > struct aiu_platform_data { > bool has_acodec; > + bool has_clk_ctrl_more_i2s_div; > }; > > struct aiu {

1 0

[PATCH 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. 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 | 240 ++++++++++++++++++ 6 files changed, 314 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

1 4

[PATCH for-next v3] ASoC: tas2562: Add support for digital volume control
by Dan Murphy 21 Feb '20

21 Feb '20

Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v3 - Removed volume init in codec_probe. sound/soc/codecs/tas2562.c | 78 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 82 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b2682c2360b6..d5e04030a0c1 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -414,6 +433,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -427,6 +490,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -517,6 +591,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -- 2.25.0

2 1