June 2024 - Alsa-devel - mailman.alsa-project.org

[modern hardware] interleaved or plannar?
by GitHub issues - opened 08 Jun '24

08 Jun '24

alsa-project/alsa-lib issue #400 was opened from sylware: To have the less "technical distance" with modern audio hardware (for instance usb audio 2), is it better to have interleaved audio frames each with a pitch, or brutally planar with a pitch per plane? Issue URL : https://github.com/alsa-project/alsa-lib/issues/400 Repository URL: https://github.com/alsa-project/alsa-lib

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[bug report] ASoC: fsl-asoc-card: Add MQS support
by Dan Carpenter 08 Jun '24

08 Jun '24

Hello Shengjiu Wang, Commit 039652a5b965 ("ASoC: fsl-asoc-card: Add MQS support") from Jun 17, 2020 (linux-next), leads to the following Smatch static checker warning: sound/soc/fsl/fsl-asoc-card.c:796 fsl_asoc_card_probe() error: uninitialized symbol 'codec_dev_name'. sound/soc/fsl/fsl-asoc-card.c 542 static int fsl_asoc_card_probe(struct platform_device *pdev) 543 { 544 struct device_node *cpu_np, *codec_np, *asrc_np; 545 struct device_node *np = pdev->dev.of_node; 546 struct platform_device *asrc_pdev = NULL; 547 struct device_node *bitclkprovider = NULL; 548 struct device_node *frameprovider = NULL; 549 struct platform_device *cpu_pdev; 550 struct fsl_asoc_card_priv *priv; 551 struct device *codec_dev = NULL; 552 const char *codec_dai_name; 553 const char *codec_dev_name; 554 u32 asrc_fmt = 0; 555 u32 width; 556 int ret; 557 558 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); 559 if (!priv) 560 return -ENOMEM; 561 562 cpu_np = of_parse_phandle(np, "audio-cpu", 0); 563 /* Give a chance to old DT binding */ 564 if (!cpu_np) 565 cpu_np = of_parse_phandle(np, "ssi-controller", 0); 566 if (!cpu_np) { 567 dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); 568 ret = -EINVAL; 569 goto fail; 570 } 571 572 cpu_pdev = of_find_device_by_node(cpu_np); 573 if (!cpu_pdev) { 574 dev_err(&pdev->dev, "failed to find CPU DAI device\n"); 575 ret = -EINVAL; 576 goto fail; 577 } 578 579 codec_np = of_parse_phandle(np, "audio-codec", 0); 580 if (codec_np) { 581 struct platform_device *codec_pdev; 582 struct i2c_client *codec_i2c; 583 584 codec_i2c = of_find_i2c_device_by_node(codec_np); 585 if (codec_i2c) { 586 codec_dev = &codec_i2c->dev; 587 codec_dev_name = codec_i2c->name; 588 } 589 if (!codec_dev) { 590 codec_pdev = of_find_device_by_node(codec_np); 591 if (codec_pdev) { 592 codec_dev = &codec_pdev->dev; 593 codec_dev_name = codec_pdev->name; 594 } 595 } There are a bunch of ways that codec_dev_name can end up uninitialized. 596 } 597 598 asrc_np = of_parse_phandle(np, "audio-asrc", 0); 599 if (asrc_np) 600 asrc_pdev = of_find_device_by_node(asrc_np); 601 602 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ 603 if (codec_dev) { 604 struct clk *codec_clk = clk_get(codec_dev, NULL); 605 [ Snip ] 789 /* Initialize sound card */ 790 priv->pdev = pdev; 791 priv->card.dev = &pdev->dev; 792 priv->card.owner = THIS_MODULE; 793 ret = snd_soc_of_parse_card_name(&priv->card, "model"); 794 if (ret) { 795 snprintf(priv->name, sizeof(priv->name), "%s-audio", --> 796 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name); Used in error handling only. 797 priv->card.name = priv->name; 798 } 799 priv->card.dai_link = priv->dai_link; 800 priv->card.late_probe = fsl_asoc_card_late_probe; regards, dan carpenter

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[PATCH v2 0/4] ASoC: qcom: display port changes
by srinivas.kandagatla＠linaro.org 08 Jun '24

08 Jun '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support for. 1. parse Display Port module tokens from ASoC topology 2. add support to DP/HDMI Jack events. 3. fixes a typo in function name in sm8250 Verified these patches on X13s along with changes to tplg in https://git.codelinaro.org/linaro/qcomlt/audioreach-topology/-/tree/topic/x… and ucm changes from https://github.com/Srinivas-Kandagatla/alsa-ucm-conf/tree/topic/x13s-dp Thanks, Srini Changes since v1: - Fixed unused variable warning. - fixed warning 'break;' to avoid fall-through Srinivas Kandagatla (4): ASoC: qcom: q6dsp: parse Display port tokens ASoC: qcom: common: add Display port Jack function ASoC: qcom: sc8280xp: add Display port Jack ASoC: qcom: sm8250: fix a typo in function name sound/soc/qcom/common.c | 29 +++++++++++++++++++++++++++++ sound/soc/qcom/common.h | 3 +++ sound/soc/qcom/qdsp6/topology.c | 26 ++++++++++++++++++++++++++ sound/soc/qcom/sc8280xp.c | 14 ++++++++++++++ sound/soc/qcom/sm8250.c | 4 ++-- 5 files changed, 74 insertions(+), 2 deletions(-) -- 2.25.1

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[PATCH] ASoC: ti: omap-hdmi: Fix too long driver name
by Primoz Fiser 07 Jun '24

07 Jun '24

Set driver name to DRV_NAME. This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 639bc83f4263..3f35eedeea41 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = DRV_NAME; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.25.1

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[PATCH v2 0/6] ASoC: Drop or replace of_gpio.h
by Andy Shevchenko 07 Jun '24

07 Jun '24

Replace or drop the legacy header that is subject to remove. Not all of them were compile-tested, the series might have hidden compilation errors. In v3: - moved aw88399 from the "Remove ..." patch to the "Replace ..." (LKP) In v2: - added tags (Kuninori, Charles) - ripped out TAS2781 (it's a mess from GPIO handling perspective) Andy Shevchenko (6): ASoC: codecs: Remove unused of_gpio.h ASoC: fsl: Remove unused of_gpio.h ASoC: rockchip: Remove unused of_gpio.h ASoC: codecs: Replace of_gpio.h by proper one ASoC: generic: Replace of_gpio.h by proper one ASoC: samsung: Replace of_gpio.h by proper one sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4458.c | 1 - sound/soc/codecs/aw88395/aw88395.c | 2 +- sound/soc/codecs/aw88399.c | 2 +- sound/soc/codecs/cs53l30.c | 1 - sound/soc/codecs/max98390.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tlv320adc3xxx.c | 1 - sound/soc/codecs/tlv320adcx140.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/ts3a227e.c | 1 - sound/soc/codecs/wsa883x.c | 1 - sound/soc/fsl/imx-es8328.c | 1 - sound/soc/fsl/imx-rpmsg.c | 2 -- sound/soc/generic/audio-graph-card2-custom-sample.c | 3 ++- sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/rockchip/rockchip_spdif.c | 1 - sound/soc/samsung/aries_wm8994.c | 2 +- 23 files changed, 5 insertions(+), 24 deletions(-) -- 2.43.0.rc1.1336.g36b5255a03ac

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[PATCH v2 0/2] tlv320adc3xxx: Allow MICBIAS pins to be used as GPOs
by Ricard Wanderlof 07 Jun '24

07 Jun '24

In some cases, depending on system design, the MICBIAS pins on the chip are not needed as such, but a couple of extra GPIO pins would be useful. This patch allows the MICBIAS pins to be configured in the device tree as general purpose output pins, controlled via the GPIO framework. Owing to their originally intended purpose there are some limitations: when the MICBIAS pins are deactivated, they will float, so will likely need a pulldown in many applications. When activated, they will assume the voltage specified by the micbias1-vg and micbias2-vg properties, respectively, meaning that the resulting output voltage will be 2.0 V, 2.5 V or AVDD . Signed-off-by: Ricard Wanderlof <ricard.wanderlof(a)axis.com> --- Changes in v2: - Corrected typo in yaml document for ti,micbias2-vg property reference - Added dependecies to yaml document for ti,micbias1-gio and ti,micbias2-gpo - Link to v1: https://lore.kernel.org/r/20240529-tlv320adc3xxx-micbias-gpo-v1-0-300d39cec… --- Ricard Wanderlof (2): ASoC: dt-bindings: tlv320adc3xxx: Add MICBIAS-as-GPO properties tlv320adc3xxx: Add support for using MICBIAS pins as GPO .../bindings/sound/ti,tlv320adc3xxx.yaml | 24 +++++ sound/soc/codecs/tlv320adc3xxx.c | 105 ++++++++++++++++----- 2 files changed, 108 insertions(+), 21 deletions(-) --- base-commit: 47d09270d7776e46858a161f94b735640b2fb056 change-id: 20240528-tlv320adc3xxx-micbias-gpo-b8b4d1c846d3 Best regards, -- -- 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 0/2] Fixes for McASP and dmaengine_pcm
by Jai Luthra 06 Jun '24

06 Jun '24

This series fixes two patches: 1. Fix the dmaengine API usage by calling dmaengine_synchronize() after dmaengine_terminate_async() when xrun events occur in application 2. Use the McASP AFIFO property from DT to refine the period size, instead of hardcoding minimum to 64 samples Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- Jai Luthra (2): ALSA: dmaengine: Synchronize dma channel in prepare() ASoC: ti: davinci-mcasp: Set min period size using FIFO config include/sound/dmaengine_pcm.h | 1 + sound/core/pcm_dmaengine.c | 10 ++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 8 ++++++++ sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 4 files changed, 26 insertions(+), 2 deletions(-) --- base-commit: d97496ca23a2d4ee80b7302849404859d9058bcd change-id: 20240604-asoc_next-c063fcc190c6 Best regards, -- Jai Luthra <j-luthra(a)ti.com>

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[PATCHv4 9/9] ASoC: dt-bindings: fsl-asoc-card: add compatible for generic codec
by Elinor Montmasson 06 Jun '24

06 Jun '24

Add documentation about new dts bindings following new support for compatible "fsl,imx-audio-generic". Some CPU DAI don't require a real audio codec. The new compatible "fsl,imx-audio-generic" allows using the driver with codec drivers SPDIF DIT and SPDIF DIR as dummy codecs. It also allows using not pre-configured audio codecs which don't require specific control through a codec driver. The new dts properties give the possibility to set some parameters about the CPU DAI usually set through the codec configuration. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl-asoc-card.yaml | 96 ++++++++++++++++++- 1 file changed, 92 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..332d8bf96e06 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -23,6 +23,16 @@ description: and PCM DAI formats. However, it'll be also possible to support those non AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long as the driver has been properly upgraded. + To use CPU DAIs that do not require a codec such as an S/PDIF controller, + or to use a DAI to output or capture raw I2S/TDM data, you can + use the compatible "fsl,imx-audio-generic". + +definitions: + imx-audio-generic-dependency: + properties: + compatible: + contains: + const: fsl,imx-audio-generic maintainers: - Shengjiu Wang <shengjiu.wang(a)nxp.com> @@ -81,6 +91,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-generic model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +104,14 @@ properties: need to add ASRC support via DPCM. audio-codec: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of an audio codec + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + If using the "fsl,imx-audio-generic" compatible, give instead a pair of + phandles with the spdif_transmitter first (driver SPDIF DIT) and the + spdif_receiver second (driver SPDIF DIR). + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +167,8 @@ properties: description: dai-link uses bit clock inversion. mclk-id: - $ref: /schemas/types.yaml#/definitions/uint32 - description: main clock id, specific for each card configuration. + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -167,10 +184,68 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller + # Properties relevant only with "fsl,imx-audio-generic" compatible + dai-tdm-slot-width: + description: See tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-num: + description: See tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + clocks: + description: | + The CPU DAI system clock, used to retrieve + the CPU DAI system clock frequency with the generic codec. + maxItems: 1 + + clock-names: + items: + - const: cpu_sysclk + + cpu-system-clock-direction-out: + description: | + Specifies cpu system clock direction as 'out' on initialization. + If not set, direction is 'in'. + $ref: /schemas/types.yaml#/definitions/flag + +dependencies: + dai-tdm-slot-width: + $ref: "#/definitions/imx-audio-generic-dependency" + dai-tdm-slot-num: + $ref: "#/definitions/imx-audio-generic-dependency" + clocks: + $ref: "#/definitions/imx-audio-generic-dependency" + cpu-system-clock-direction-out: + $ref: "#/definitions/imx-audio-generic-dependency" + required: - compatible - model +allOf: + - if: + $ref: "#/definitions/imx-audio-generic-dependency" + then: + properties: + audio-codec: + items: + - description: SPDIF DIT phandle + - description: SPDIF DIR phandle + mclk-id: + maxItems: 1 + items: + minItems: 1 + maxItems: 2 + else: + properties: + audio-codec: + maxItems: 1 + mclk-id: + maxItems: 1 + items: + maxItems: 1 + unevaluatedProperties: false examples: @@ -195,3 +270,16 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + #include <dt-bindings/clock/imx8mn-clock.h> + sound-spdif-asrc { + compatible = "fsl,imx-audio-generic"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + clocks = <&clk IMX8MN_CLK_SAI5_ROOT>; + clock-names = "cpu_sysclk"; + cpu-system-clock-direction-out; + }; -- 2.34.1

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[PATCH v2] ASoC: dt-bindings: convert amlogic,g12a-tohdmitx to dt-schema
by Neil Armstrong 06 Jun '24

06 Jun '24

Convert text bindings to dt-schema format for the Amlogic TX HDMI control glue. Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Changes in v2: - re-order properties and required - switch to unevaluatedProperties and drop sound-name-prefix - Add review tag - Link to v1: https://lore.kernel.org/r/20240605-topic-amlogic-upstream-bindings-convert-… --- .../bindings/sound/amlogic,g12a-tohdmitx.txt | 58 ---------------------- .../bindings/sound/amlogic,g12a-tohdmitx.yaml | 54 ++++++++++++++++++++ 2 files changed, 54 insertions(+), 58 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt deleted file mode 100644 index 4e8cd7eb7cec..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt +++ /dev/null @@ -1,58 +0,0 @@ -* Amlogic HDMI Tx control glue - -Required properties: -- compatible: "amlogic,g12a-tohdmitx" or - "amlogic,sm1-tohdmitx" -- reg: physical base address of the controller and length of memory - mapped region. -- #sound-dai-cells: should be 1. -- resets: phandle to the dedicated reset line of the hdmitx glue. - -Example on the S905X2 SoC: - -tohdmitx: audio-controller@744 { - compatible = "amlogic,g12a-tohdmitx"; - reg = <0x0 0x744 0x0 0x4>; - #sound-dai-cells = <1>; - resets = <&clkc_audio AUD_RESET_TOHDMITX>; -}; - -Example of an 'amlogic,axg-sound-card': - -sound { - compatible = "amlogic,axg-sound-card"; - -[...] - - dai-link-x { - sound-dai = <&tdmif_a>; - dai-format = "i2s"; - dai-tdm-slot-tx-mask-0 = <1 1>; - - codec-0 { - sound-dai = <&tohdmitx TOHDMITX_I2S_IN_A>; - }; - - codec-1 { - sound-dai = <&external_dac>; - }; - }; - - dai-link-y { - sound-dai = <&tdmif_c>; - dai-format = "i2s"; - dai-tdm-slot-tx-mask-0 = <1 1>; - - codec { - sound-dai = <&tohdmitx TOHDMITX_I2S_IN_C>; - }; - }; - - dai-link-z { - sound-dai = <&tohdmitx TOHDMITX_I2S_OUT>; - - codec { - sound-dai = <&hdmi_tx>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml new file mode 100644 index 000000000000..b4b78475c5b8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-tohdmitx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a HDMI TX Control Glue + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + compatible: + oneOf: + - items: + - const: amlogic,g12a-tohdmitx + - items: + - enum: + - amlogic,sm1-tohdmitx + - const: amlogic,g12a-tohdmitx + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - resets + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + tohdmitx: audio-controller@744 { + compatible = "amlogic,g12a-tohdmitx"; + reg = <0x744 0x4>; + resets = <&clkc_audio AUD_RESET_TOHDMITX>; + #sound-dai-cells = <1>; + }; --- base-commit: c3f38fa61af77b49866b006939479069cd451173 change-id: 20240605-topic-amlogic-upstream-bindings-convert-g12a-tohdmitx-d85095861d88 Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

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Onboard Speaker does not work and Internal Mic (DMIC) isn't detected on Dell Latitude 7350 Detachable
by GitHub issues - opened 06 Jun '24

06 Jun '24

alsa-project/alsa-ucm-conf issue #424 was opened from DeepHarsora-Dell: Installed Fedora Rawhide on Dell Latitude 7350 Detachable resulting in no-audio output from onboard speaker and in-built mic (DMIC) can't be detected. Updated the SUT with the following 1. Topology built rightly. 2. Noticed proper SOF Machine driver in upstream as on v.6.10-rc1, able to load the topology. 3. Updated the UCM2 config files as per [[ALSA_UCM_README]](https://github.com/alsa-project/alsa-ucm-conf/blob/master… Below is the error found while alsaucm reload [linuxengg@fedora ~]$ sudo alsaucm reload ALSA lib ucm_subs.c:807:(uc_mgr_get_substituted_value) variable '${var:__RegEx}' is not defined in this context! ALSA lib parser.c:2024:(parse_verb_file) error: /sof-soundwire/HiFi.conf failed to parse device ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 alsaucm: error failed to open sound card hw:0: Invalid argument Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/424 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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