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

[PATCH v1] ASoc: tas2781: fixed the issue that the chip do not shutdown immediatly after aplay stopped
by Shenghao Ding 20 Aug '24

20 Aug '24

Issue reported from customer that the chip do not shutdown after aplay stopped until 6 mins later. Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops - Remove unnecessary line feed for tasdevice_codec_remove --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..145c45349399 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -557,30 +557,22 @@ static void tasdevice_fw_ready(const struct firmware *fmw, release_firmware(fmw); } -static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, int stream) { - struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int state = 0; - /* Codec Lock Hold */ - mutex_lock(&tas_priv->codec_lock); - if (event == SND_SOC_DAPM_PRE_PMD) - state = 1; - tasdevice_tuning_switch(tas_priv, state); - /* Codec Lock Release*/ - mutex_unlock(&tas_priv->codec_lock); + /* Codec Lock/UnLock */ + guard(mutex)(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, mute); return 0; } static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, - 0, 0, tasdevice_dapm_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SPK("SPK", NULL), SND_SOC_DAPM_OUTPUT("OUT"), SND_SOC_DAPM_INPUT("DMIC") }; @@ -667,6 +659,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { .startup = tasdevice_startup, .hw_params = tasdevice_hw_params, .set_sysclk = tasdevice_set_dai_sysclk, + .mute_stream = tasdevice_mute, }; static struct snd_soc_dai_driver tasdevice_dai_driver[] = { -- 2.34.1

3 3

[PATCH v7 2/2] ARM: dts: imx6: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi | 9 +++++++-- .../boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 9 +++++++-- 12 files changed, 104 insertions(+), 28 deletions(-) diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts index 95b49fc83f7b..299106fbe51c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts @@ -127,12 +127,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts index a7d5693c5ab7..8d2b608e0b90 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts @@ -111,12 +111,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts index 7c298d9aa21e..5353a0c24420 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts @@ -90,11 +90,16 @@ sound-sgtl5000 { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi index ea40623d12e5..edf55760a5c1 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi @@ -197,11 +197,20 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi index 3a46ade3b6bd..9e97ef5e43f2 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi @@ -121,11 +121,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi index d3a7a6eeb8e0..b01670cdd52c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi @@ -142,12 +142,21 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + /* Optional S/PDIF in on SODIMM 88 and out on SODIMM 90, 137 or 168 */ sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi index 761566ae3cf5..bd66430c1d78 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi @@ -100,12 +100,17 @@ v_usb1: regulator-v-usb1 { vin-supply = <&v_5v0>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "Integrated SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; gpio-keys { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi index a955c77cd499..d1ad65ab6b72 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi @@ -140,12 +140,17 @@ sound_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi index 0a3deaf92eea..55fb7b904220 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi @@ -143,12 +143,17 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-sabreauto-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; backlight { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi index 38abb6b50f6c..7130b9c3b3aa 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi @@ -26,11 +26,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; reg_1p5v: regulator-1p5v { diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts index b0c27b9b0244..dfbfb8119bf3 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts @@ -97,11 +97,16 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi index 7d4170c27732..277a6e039045 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi @@ -183,12 +183,17 @@ panel_in: endpoint { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx6sx-sdb-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; -- 2.34.1

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[PATCH v7 1/2] arm64: dts: imx8m: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 930e14fec423..5f8336217bb8 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..33d73f3dc187 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..a87d0692c3bb 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; + }; + + hdmi_arc_in: hdmi-arc-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; }; sound-hdmi-arc { compatible = "fsl,imx-audio-spdif"; model = "imx-hdmi-arc"; - spdif-controller = <&spdif2>; - spdif-in; + audio-cpu = <&spdif2>; + audio-codec = <&hdmi_arc_in>; }; }; -- 2.34.1

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[PATCH] ASoc: tas2781: replace devm_kzalloc and scnprintf with devm_kasprintf
by Shenghao Ding 20 Aug '24

20 Aug '24

Replace devm_kzalloc and scnprintf with devm_kasprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..1451ba0fa5b4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -346,13 +346,11 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) } /* Create a mixer item for selecting the active profile */ - name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); + name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Profile Id"); if (!name) { ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -441,18 +439,13 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) goto out; } - /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); - if (!prog_name || !conf_name) { + /* Create mixer items for selecting the active Program and Config */ + prog_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, + "Speaker Program Id"); + if (!prog_name) { ret = -ENOMEM; goto out; } - - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -460,8 +453,12 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + conf_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Config Id"); + if (!conf_name) { + ret = -ENOMEM; + goto out; + } dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; -- 2.34.1

3 2

[PATCH] soundwire: qcom: Use devm_clk_get_enabled() helpers
by Huan Yang 20 Aug '24

20 Aug '24

The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: Huan Yang <link(a)vivo.com> --- drivers/soundwire/qcom.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2a57ed9464b4 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1527,24 +1527,22 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (IS_ERR(ctrl->audio_cgcr)) { dev_err(dev, "Failed to get cgcr reset ctrl required for SW gating\n"); ret = PTR_ERR(ctrl->audio_cgcr); - goto err_init; + return ret; } } ctrl->irq = of_irq_get(dev->of_node, 0); if (ctrl->irq < 0) { ret = ctrl->irq; - goto err_init; + return ret; } - ctrl->hclk = devm_clk_get(dev, "iface"); + ctrl->hclk = devm_clk_get_enabled(dev, "iface"); if (IS_ERR(ctrl->hclk)) { ret = dev_err_probe(dev, PTR_ERR(ctrl->hclk), "unable to get iface clock\n"); - goto err_init; + return ret; } - clk_prepare_enable(ctrl->hclk); - ctrl->dev = dev; dev_set_drvdata(&pdev->dev, ctrl); mutex_init(&ctrl->port_lock); @@ -1558,7 +1556,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) ret = qcom_swrm_get_port_config(ctrl); if (ret) - goto err_clk; + return ret; params = &ctrl->bus.params; params->max_dr_freq = DEFAULT_CLK_FREQ; @@ -1586,7 +1584,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "soundwire", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire irq\n"); - goto err_clk; + return ret; } ctrl->wake_irq = of_irq_get(dev->of_node, 1); @@ -1597,7 +1595,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "swr_wake_irq", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire wake irq\n"); - goto err_init; + return ret; } } @@ -1612,7 +1610,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (ret) { dev_err(dev, "Failed to register Soundwire controller (%d)\n", ret); - goto err_clk; + return ret; } qcom_swrm_init(ctrl); @@ -1642,9 +1640,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) err_master_add: sdw_bus_master_delete(&ctrl->bus); -err_clk: - clk_disable_unprepare(ctrl->hclk); -err_init: return ret; } @@ -1653,7 +1648,6 @@ static void qcom_swrm_remove(struct platform_device *pdev) struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(&pdev->dev); sdw_bus_master_delete(&ctrl->bus); - clk_disable_unprepare(ctrl->hclk); } static int __maybe_unused swrm_runtime_resume(struct device *dev) -- 2.45.2

1 0

[PATCH v3 0/8] MSM8953/MSM8976 ASoC support
by Adam Skladowski 20 Aug '24

20 Aug '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. Changes since v2 =============== 1. Guarded ioremap on quinary mux with checking if resource is not NULL 2. Introduced compatible based reg overriding in wcd-analog driver 3. Separated PM8950/PM8953 codec patches 4. Moved reg/reg-names back into top-level and rewrote as in comments 5. Separated MSM8953/MSM8976 QDSP6 patches 6. Added Lineout patch [1] - https://lore.kernel.org/lkml/20240723083300.35605-1-krzysztof.kozlowski@lin… Changes since v1 ================ 1. Rebased dt-bindings documentation based on Krzysztof's split patch[1] 2. Resolved clang errors by guarding ret verification inside if 3. Switched quin-iomux to devm_ioremap_resource to not fail on msm8916 Signed-off-by: Adam Skladowski <a39.skl(a)gmail.com> --- Adam Skladowski (4): ASoC: dt-bindings: pm8916-wcd-analog-codec: Document pm8950/pm8953 ASoC: msm8916-wcd-analog: add pm8950 codec ASoC: dt-bindings: apq8016-sbc: Add msm8953/msm8976-qdsp6-sndcard ASoC: qcom: apq8016_sbc: Add support for msm8976 SoC Vladimir Lypak (4): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add pm8953 codec ASoC: qcom: apq8016_sbc: Add support for msm8953 SoC ASoC: msm8916-wcd-analog: add lineout output .../bindings/sound/qcom,apq8016-sbc-sndcard.yaml | 27 ++++ .../sound/qcom,pm8916-wcd-analog-codec.yaml | 6 +- sound/soc/codecs/msm8916-wcd-analog.c | 145 ++++++++++++++++++++- sound/soc/qcom/apq8016_sbc.c | 88 ++++++++++++- 4 files changed, 257 insertions(+), 9 deletions(-) --- base-commit: 31c2a5c4bf3673b0bf098f358ea931b9ada1e7de change-id: 20240729-msm8953-msm8976-asoc-65f973a13165 Best regards, -- Adam Skladowski <a39.skl(a)gmail.com>

6 21

[DISCUSSION] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

1 0

discussion: why is the mono profile in ALSA not autogenerated by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

1 0

[RFC PATCH v2 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 20 Aug '24

20 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: reserve space in snd_compr_metadata.key for private usage ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 791 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1297 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

4 13

[PATCH alsa-utils] aseqdump: Add missing dump of UMP Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The handling of Set Key Signature message of the Flex Data type was missing by some reason. Now the definition was added in alsa-lib ump_msg.h, so let's show the contents. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- seq/aseqdump/aseqdump.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/seq/aseqdump/aseqdump.c b/seq/aseqdump/aseqdump.c index 1d9c01e01dd3..e37c87b45fc1 100644 --- a/seq/aseqdump/aseqdump.c +++ b/seq/aseqdump/aseqdump.c @@ -964,6 +964,14 @@ static void dump_ump_flex_data_event(const unsigned int *ump) return; } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && + fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_KEY_SIGNATURE) { + printf("UMP Set Key Signature sharps/flats %d, tonic %d\n", + fh->set_key_sig.sharps_flats, + fh->set_key_sig.tonic_note); + return; + } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_CHORD_NAME) { printf("UMP Set Chord Name tonic %d %d %d, alt1 %d/%d, alt2 %d/%d, alt3 %d/%d, alt4 %d/%d, bass %d %d %d, alt1 %d/%d alt2 %d/%d\n", -- 2.43.0

1 0