December 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 1/2] ASoC: dt-bindings: rt5682s: add AMIC delay time property
by shumingf＠realtek.com 08 Dec '21

08 Dec '21

From: Shuming Fan <shumingf(a)realtek.com> Add the AMIC delay time to control how much delay time (ms) to unmute the stereo1 ADC. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml index 2b8b7b51fe55..d65c0ed5060c 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -61,6 +61,10 @@ properties: description: | Set the delay time (ms) for the requirement of the particular DMIC. + realtek,amic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular platform or AMIC. + realtek,dmic-clk-driving-high: type: boolean description: | -- 2.34.1

2 1

[PATCH] ASoC: Intel: boards: add 'static' qualifiers for max98390 routes
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

Sparse warnings: sound/soc/intel/boards/sof_maxim_common.c:140:33: error: symbol 'max_98390_dapm_routes' was not declared. Should it be static? sound/soc/intel/boards/sof_maxim_common.c:156:33: error: symbol 'max_98390_tt_dapm_routes' was not declared. Should it be static? Fixes: f316c9d9ba8ea ('ASoC: Intel: boards: add max98390 2/4 speakers support') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index 9171d9cd179e..112e89951da0 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -137,7 +137,7 @@ EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); /* * Maxim MAX98390 */ -const struct snd_soc_dapm_route max_98390_dapm_routes[] = { +static const struct snd_soc_dapm_route max_98390_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "Left BE_OUT" }, { "Right Spk", NULL, "Right BE_OUT" }, @@ -153,7 +153,7 @@ static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { SND_SOC_DAPM_SPK("TR Spk", NULL), }; -const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { +static const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { /* Tweeter speaker */ { "TL Spk", NULL, "Tweeter Left BE_OUT" }, { "TR Spk", NULL, "Tweeter Right BE_OUT" }, -- 2.25.1

2 1

[PATCH] ASoC: Intel: sof_rt5682: Move rt1015 speaker amp to common file
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

From: Yong Zhi <yong.zhi(a)intel.com> Move rt1015 driver code to common file to be consistent with rt1011 and rt1015p. No functional change. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_realtek_common.c | 119 +++++++++++++++++++- sound/soc/intel/boards/sof_realtek_common.h | 7 ++ sound/soc/intel/boards/sof_rt5682.c | 107 +----------------- 3 files changed, 127 insertions(+), 106 deletions(-) diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 2ec34f8df9e1..4cf131310ad3 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -12,12 +12,13 @@ #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> #include "../../codecs/rt1011.h" +#include "../../codecs/rt1015.h" #include "sof_realtek_common.h" /* * Current only 2-amp configuration is supported for rt1011 */ -static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { +static const struct snd_soc_dapm_route speaker_map_lr[] = { /* speaker */ { "Left Spk", NULL, "Left SPO" }, { "Right Spk", NULL, "Right SPO" }, @@ -117,8 +118,8 @@ static int rt1011_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; int ret; - ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_dapm_routes, - ARRAY_SIZE(rt1011_dapm_routes)); + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); if (ret) dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); return ret; @@ -241,3 +242,115 @@ void sof_rt1015p_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1015p_codec_confs; card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); } + +/* + * RT1015 audio amplifier + */ + +static int rt1015_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i, fs = 64, ret; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * fs, + params_rate(params) * 256); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + + return 0; +} + +static int rt1015_hw_params_pll_and_tdm(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i, fs = 100, ret; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * fs, + params_rate(params) * 256); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + /* rx slot 1 for RT1015_DEV0_NAME */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), + 0x0, 0x1, 4, 24); + if (ret) + return ret; + + /* rx slot 2 for RT1015_DEV1_NAME */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 1), + 0x0, 0x2, 4, 24); + if (ret) + return ret; + + return 0; +} + +static struct snd_soc_ops rt1015_ops = { + .hw_params = rt1015_hw_params, +}; + +static struct snd_soc_codec_conf rt1015_amp_conf[] = { + { + .dlc = COMP_CODEC_CONF(RT1015_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015_components[] = { + { + .name = RT1015_DEV0_NAME, + .dai_name = RT1015_CODEC_DAI, + }, + { + .name = RT1015_DEV1_NAME, + .dai_name = RT1015_CODEC_DAI, + }, +}; + +static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) +{ + return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); +} + +void sof_rt1015_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1015_amp_conf; + card->num_configs = ARRAY_SIZE(rt1015_amp_conf); +} + +void sof_rt1015_dai_link(struct snd_soc_dai_link *link, unsigned int fs) +{ + link->codecs = rt1015_components; + link->num_codecs = ARRAY_SIZE(rt1015_components); + link->init = speaker_codec_init_lr; + link->ops = &rt1015_ops; + + if (fs == 100) + rt1015_ops.hw_params = rt1015_hw_params_pll_and_tdm; +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index cb0b49b2855c..228ac9c08430 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -28,4 +28,11 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card); void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); void sof_rt1015p_codec_conf(struct snd_soc_card *card); +#define RT1015_CODEC_DAI "rt1015-aif" +#define RT1015_DEV0_NAME "i2c-10EC1015:00" +#define RT1015_DEV1_NAME "i2c-10EC1015:01" + +void sof_rt1015_dai_link(struct snd_soc_dai_link *link, unsigned int fs); +void sof_rt1015_codec_conf(struct snd_soc_card *card); + #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 6cadb5fb72e0..bd6d2e7dea53 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -20,7 +20,6 @@ #include <sound/rt5682.h> #include <sound/rt5682s.h> #include <sound/soc-acpi.h> -#include "../../codecs/rt1015.h" #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" #include "../../codecs/hdac_hdmi.h" @@ -400,67 +399,6 @@ static struct snd_soc_ops sof_rt5682_ops = { .hw_params = sof_rt5682_hw_params, }; -static int sof_rt1015_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai; - int i, fs, ret; - - if (!snd_soc_card_get_codec_dai(card, "rt1015-aif")) - return 0; - - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_100FS) - fs = 100; - else - fs = 64; - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, - params_rate(params) * fs, - params_rate(params) * 256); - if (ret < 0) { - dev_err(card->dev, "failed to set pll\n"); - return ret; - } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, - params_rate(params) * 256, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "failed to set sysclk\n"); - return ret; - } - - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_100FS) { - if (!strcmp(codec_dai->component->name, "i2c-10EC1015:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x0, 0x1, 4, 24); - if (ret < 0) { - dev_err(card->dev, "failed to set tdm slot\n"); - return ret; - } - } - - if (!strcmp(codec_dai->component->name, "i2c-10EC1015:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x0, 0x2, 4, 24); - if (ret < 0) { - dev_err(card->dev, "failed to set tdm slot\n"); - return ret; - } - } - } - } - - return 0; -} - -static struct snd_soc_ops sof_rt1015_ops = { - .hw_params = sof_rt1015_hw_params, -}; - static struct snd_soc_dai_link_component platform_component[] = { { /* name might be overridden during probe */ @@ -551,22 +489,11 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; -static const struct snd_soc_dapm_route speaker_map_lr[] = { - { "Left Spk", NULL, "Left SPO" }, - { "Right Spk", NULL, "Right SPO" }, -}; - static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; -static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) -{ - return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, - ARRAY_SIZE(speaker_map_lr)); -} - static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -589,17 +516,6 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static struct snd_soc_codec_conf rt1015_amp_conf[] = { - { - .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"), - .name_prefix = "Left", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"), - .name_prefix = "Right", - }, -}; - /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { .name = "rt5682", /* the sof- prefix is added by the core */ @@ -635,17 +551,6 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; -static struct snd_soc_dai_link_component rt1015_components[] = { - { - .name = "i2c-10EC1015:00", - .dai_name = "rt1015-aif", - }, - { - .name = "i2c-10EC1015:01", - .dai_name = "rt1015-aif", - }, -}; - static struct snd_soc_dai_link_component dummy_component[] = { { .name = "snd-soc-dummy", @@ -798,10 +703,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].id = id; if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { - links[id].codecs = rt1015_components; - links[id].num_codecs = ARRAY_SIZE(rt1015_components); - links[id].init = speaker_codec_init_lr; - links[id].ops = &sof_rt1015_ops; + sof_rt1015_dai_link(&links[id], (sof_rt5682_quirk & + SOF_RT1015_SPEAKER_AMP_100FS) ? 100 : 64); } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & @@ -995,10 +898,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_rt5682.dai_link = dai_links; - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { - sof_audio_card_rt5682.codec_conf = rt1015_amp_conf; - sof_audio_card_rt5682.num_configs = ARRAY_SIZE(rt1015_amp_conf); - } + if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) + sof_rt1015_codec_conf(&sof_audio_card_rt5682); INIT_LIST_HEAD(&ctx->hdmi_pcm_list); -- 2.25.1

2 1

[PATCH] sound: fsl: add missing put_device() call in imx_hdmi_probe()
by Qing Wang 08 Dec '21

08 Dec '21

From: Wang Qing <wangqing(a)vivo.com> of_find_device_by_node() takes a reference to the embedded struct device which needs to be dropped when error return. Signed-off-by: Wang Qing <wangqing(a)vivo.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 929f69b..89fd8ad --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -125,6 +125,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { + put_device(&cpu_pdev->dev); ret = -ENOMEM; goto fail; } -- 2.7.4

2 1

Re: [Linux-stm32] [PATCH v2 1/4] ASoC: dt-bindings: stm32: i2s: add audio-graph-card port
by Alexandre TORGUE 08 Dec '21

08 Dec '21

Hi Ahmad On 12/7/21 2:59 PM, Ahmad Fatoum wrote: > Hello Alex, > > On 07.12.21 14:52, Alexandre TORGUE wrote: >> Hi Rob >> >> On 12/1/21 11:34 PM, Rob Herring wrote: >>> On Fri, Nov 26, 2021 at 11:25:27AM +0100, Olivier MOYSAN wrote: >>>> Hi Rob, >>>> >>>> On 11/25/21 10:26 PM, Rob Herring wrote: >>>>> On Thu, 25 Nov 2021 15:40:50 +0100, Olivier Moysan wrote: >>>>>> The STM2 I2S DAI can be connected via the audio-graph-card. >>>>>> Add port entry into the bindings. >>>>>> >>>>>> Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> >>>>>> --- >>>>>> Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 5 +++++ >>>>>> 1 file changed, 5 insertions(+) >>>>>> >>>>> >>>>> Running 'make dtbs_check' with the schema in this patch gives the >>>>> following warnings. Consider if they are expected or the schema is >>>>> incorrect. These may not be new warnings. >>>>> >>>>> Note that it is not yet a requirement to have 0 warnings for dtbs_check. >>>>> This will change in the future. >>>>> >>>>> Full log is available here: https://patchwork.ozlabs.org/patch/1559750 >>>>> >>>>> >>>>> audio-controller@4000b000: 'port' does not match any of the regexes: '^port@[0-9]', 'pinctrl-[0-9]+' >>>>> arch/arm/boot/dts/stm32mp157a-dk1.dt.yaml >>>>> arch/arm/boot/dts/stm32mp157c-dk2.dt.yaml >>>>> >>>> >>>> This warning is not a new one. >>>> >>>> The i2s2 node in stm32mp15xx-dkx.dtsi would require the following binding: >>>> port: >>>> $ref: audio-graph-port.yaml# >>>> unevaluatedProperties: false >>>> >>>> However the spi binding requires to introduce a unit address: >>>> patternProperties: >>>> '^port@[0-9]': >>>> $ref: audio-graph-port.yaml# >>>> unevaluatedProperties: false >>>> >>>> The warning can be removed by re-ordering the bindings patches in the serie, >>>> as "additionalProperties: true" makes the check more tolerant on extra >>>> properties. >>> >>> That's never right. >>> >>>> The patch "ASoC: dt-bindings: stm32: i2s: add audio-graph-card port" can >>>> even be merely dropped. >>>> So, I suggest to resend the serie without audio-graph-card patch. >>> >>> Only if you aren't using audio-graph-card. >>> >>>> >>>> Does it sound too permissive to you ? >>> >>> I think perhaps you need to combine the schemas into 1. Or you need to >>> restructure your dtsi files such that you only add spi specific >>> properties when spi mode is enabled and only add i2s specific properties >>> when i2s mode is enabled. Or use the /delete-property/ directive. >> >> Initially the aim of this series was to fix a "make W=1" warnings seen on spi and i2s nodes (duplicate unit-address). Moving both nodes in a common node + using a different compatible depending on SPI or I2S usage sounded good) but it is not enough. In this series the common node is named as following: "spi2s2: spi@4000b000". It is fine for a spi usage but if we want to use this "common node" with I2S compatible and specific bindings, the node name remains spi@... and then specific spi checks are done. For this with this series applied we got this issue reported by spi-controller.yaml: >> >> spi@4000b000: port@0: 'compatible' is a required property >> >> So, if we use two separates nodes we got W=1 warning and if we use a common node we got yaml check issue. One possibility would be to use a common node with a new node name (for example i2spi@...) but I think it is not acceptable. >> >> How to progress ? > > Atmel Flexcom can be configured to be either UART, SPI or i2c. Functions > are child nodes of the flexcom node and the MFD driver matching against it, > just configure the operating mode and then calls of_platform_populate. > > Would something along these lines fit here as well? Yes it could but in my mind it was not a MFD as both feature cannot be used at the same time: it is either SPI or I2S and choice is done "statically" in device tree depending board usage. Lee, what it is your feeling about that ? Will you accept to add a MFD driver for this SPI/I2S peripheral whose prurpose is only to populate child node (either SPI or I2S) ? Cheers Alex > > Cheers, > Ahmad > >> >> Thanks >> Alex >> >> >>> Rob >>> >> >> _______________________________________________ >> Linux-stm32 mailing list >> Linux-stm32(a)st-md-mailman.stormreply.com >> https://st-md-mailman.stormreply.com/mailman/listinfo/linux-stm32 >> > >

2 2

[PATCH 1/1] ASoC: dt-bindings: audio-graph-port: enable both flag/phandle for bitclock/frame-master
by Kuninori Morimoto 08 Dec '21

08 Dec '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_daifmt_parse_clock_provider_raw() is handling both bitclock/frame-master, and is supporting both flag/phandle. Current DT is assuming it is flag style. This patch allows both case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/audio-graph-port.yaml | 23 +++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 43e7f86e3b23..6c3645a0d36b 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -14,6 +14,24 @@ select: false allOf: - $ref: /schemas/graph.yaml#/$defs/port-base +definitions: + flag_or_phandle: + oneOf: + - type: boolean + const: true + - type: 'null' + - type: array + minItems: 1 + maxItems: 1 + items: + type: array + items: + phandle: true + type: integer + maximum: 0xffffffff + minItems: 1 + maxItems: 1 + properties: prefix: description: "device name prefix" @@ -42,10 +60,11 @@ patternProperties: $ref: /schemas/types.yaml#/definitions/flag frame-master: description: Indicates dai-link frame master. - $ref: /schemas/types.yaml#/definitions/phandle + $ref: "#/definitions/flag_or_phandle" bitclock-master: description: Indicates dai-link bit clock master - $ref: /schemas/types.yaml#/definitions/phandle + $ref: "#/definitions/flag_or_phandle" + dai-format: description: audio format. items: -- 2.25.1

1 1

topology: Add option to pass include path for conditional includes
by GitHub pull_request - opened 08 Dec '21

08 Dec '21

alsa-project/alsa-utils pull request #125 was opened from ranj063: The include path passed with -I option will override the relative include path based on the source file. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/125 Patch URL : https://github.com/alsa-project/alsa-utils/pull/125.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Feature request: ACPI NHLT generation for topology2
by GitHub issues - opened 07 Dec '21

07 Dec '21

alsa-project/alsa-utils issue #124 was opened from juimonen: I'd like to add ACPI NHLT generation into topology2 parsing. Why: In certain future sof firmware versions this is needed for DAI configuration Why to alsa-utils: adding it to kernel driver was internally rejected, also I can't link to topology2 parsing outside alsa-utils Good thing: if you ever wondered how acpi nhlt blobs are created and what they contain in intel platforms -> this code is doing it. Bad things: I can think many, but I'm trying to do this with least harm to every party (kernel driver, firmware) Example of the code can be found here: https://github.com/thesofproject/alsa-utils/pull/4 So mainly asking for comments/suggestions from @perexg @lrgirdwo @ranj063 FYI. Issue URL : https://github.com/alsa-project/alsa-utils/issues/124 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RESEND, PATCH v6 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 07 Dec '21

07 Dec '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..3b3fa43fa961 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH] ALSA: seq: Set upper limit of processed events
by Takashi Iwai 07 Dec '21

07 Dec '21

Currently ALSA sequencer core tries to process the queued events as much as possible when they become dispatchable. If applications try to queue too massive events to be processed at the very same timing, the sequencer core would still try to process such all events, either in the interrupt context or via some notifier; in either away, it might be a cause of RCU stall or such problems. As a potential workaround for those problems, this patch adds the upper limit of the amount of events to be processed. The remaining events are processed in the next batch, so they won't be lost. For the time being, it's limited up to 1000 events per queue, which should be high enough for any normal usages. Reported-by: Zqiang <qiang.zhang1211(a)gmail.com> Reported-by: syzbot+bb950e68b400ab4f65f8(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20211102033222.3849-1-qiang.zhang1211@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index d6c02dea976c..bc933104c3ee 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -235,12 +235,15 @@ struct snd_seq_queue *snd_seq_queue_find_name(char *name) /* -------------------------------------------------------- */ +#define MAX_CELL_PROCESSES_IN_QUEUE 1000 + void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) { unsigned long flags; struct snd_seq_event_cell *cell; snd_seq_tick_time_t cur_tick; snd_seq_real_time_t cur_time; + int processed = 0; if (q == NULL) return; @@ -263,6 +266,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } /* Process time queue... */ @@ -272,14 +277,19 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } + out: /* free lock */ spin_lock_irqsave(&q->check_lock, flags); if (q->check_again) { q->check_again = 0; - spin_unlock_irqrestore(&q->check_lock, flags); - goto __again; + if (processed < MAX_CELL_PROCESSES_IN_QUEUE) { + spin_unlock_irqrestore(&q->check_lock, flags); + goto __again; + } } q->check_blocked = 0; spin_unlock_irqrestore(&q->check_lock, flags); -- 2.31.1

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