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

[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

1 0

[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

[PATCH alsa-lib] ump_msg: Add missing definition for Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The Set Key Signature message was missing at the time we defined for Flex Data types. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/ump_msg.h | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/include/ump_msg.h b/include/ump_msg.h index a64cb317889e..5f0a45acd44c 100644 --- a/include/ump_msg.h +++ b/include/ump_msg.h @@ -606,6 +606,39 @@ typedef struct _snd_ump_msg_set_metronome { #endif } __attribute((packed)) snd_ump_msg_set_metronome_t; +/** Set Key Signature Message (128bit) */ +typedef struct _snd_ump_msg_set_key_sig { +#ifdef SNDRV_BIG_ENDIAN_BITFIELD + uint8_t type:4; /**< UMP packet type */ + uint8_t group:4; /**< UMP Group */ + uint8_t format:2; /**< Format */ + uint8_t addrs:2; /**< Address */ + uint8_t channel:4; /**< Channel */ + uint8_t status_bank; /**< Status Bank */ + uint8_t status; /**< Status */ + + uint8_t sharps_flats:4; /**< Sharps/Flats */ + uint8_t tonic_note:4; /**< Tonic Note 1 */ + uint8_t reserved1[3]; /**< Unused */ + + uint32_t reserved2[2]; /**< Unused */ +#else + uint8_t status; /**< Status */ + uint8_t status_bank; /**< Status Bank */ + uint8_t channel:4; /**< Channel */ + uint8_t addrs:2; /**< Address */ + uint8_t format:2; /**< Format */ + uint8_t group:4; /**< UMP Group */ + uint8_t type:4; /**< UMP packet type */ + + uint8_t reserved1[3]; /**< Unused */ + uint8_t tonic_note:4; /**< Tonic Note */ + uint8_t sharps_flats:4; /**< Sharps/Flats */ + + uint32_t reserved2[2]; /**< Unused */ +#endif +} __attribute((packed)) snd_ump_msg_set_key_sig_t; + /** Set Chord Name Message (128bit) */ typedef struct _snd_ump_msg_set_chord_name { #ifdef SNDRV_BIG_ENDIAN_BITFIELD @@ -678,6 +711,7 @@ typedef union _snd_ump_msg_flex_data { snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ snd_ump_msg_set_time_sig_t set_time_sig; /**< Set Time Signature */ snd_ump_msg_set_metronome_t set_metronome; /**< Set Metronome */ + snd_ump_msg_set_key_sig_t set_key_sig; /**< Set Key Signature */ snd_ump_msg_set_chord_name_t set_chord_name; /**< Set Chord Name */ snd_ump_msg_hdr_t hdr; /**< UMP header */ uint32_t raw[4]; /**< raw UMP packet */ -- 2.43.0

1 0

[PATCH 1/1] ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml
by Frank Li 19 Aug '24

19 Aug '24

Convert binding doc tlv320aic31xx.txt to yaml format. Additional change: - add i2c node in example. - replace MICBIAS_OFF with MICBIAS_2_0v in example because MICBIAS_OFF have been defined in header file. - add ref to dai-common.yaml. - add #sound-dai-cells. Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a30000/codec@18: failed to match any schema with compatible: ['ti,tlv320dac3100'] Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- .../bindings/sound/ti,tlv320dac3100.yaml | 127 ++++++++++++++++++ .../bindings/sound/tlv320aic31xx.txt | 77 ----------- 2 files changed, 127 insertions(+), 77 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml new file mode 100644 index 0000000000000..85e937e34962d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml @@ -0,0 +1,127 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320dac3100.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tlv320aic31xx Codec module + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: | + CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + + CODEC input pins: + * MIC1LP, devices with ADC + * MIC1RP, devices with ADC + * MIC1LM, devices with ADC + * AIN1, devices without ADC + * AIN2, devices without ADC + + The pins can be used in referring sound node's audio-routing property. + +properties: + compatible: + enum: + - ti,tlv320aic310x # - Generic TLV320AIC31xx with mono speaker amp + - ti,tlv320aic311x # - Generic TLV320AIC31xx with stereo speaker amp + - ti,tlv320aic3100 # - TLV320AIC3100 (mono speaker amp, no MiniDSP) + - ti,tlv320aic3110 # - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + - ti,tlv320aic3120 # - TLV320AIC3120 (mono speaker amp, MiniDSP) + - ti,tlv320aic3111 # - TLV320AIC3111 (stereo speaker amp, MiniDSP) + - ti,tlv320dac3100 # - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) + - ti,tlv320dac3101 # - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + HPVDD-supply: true + + SPRVDD-supply: true + + SPLVDD-supply: true + + AVDD-supply: true + + IOVDD-supply: true + + DVDD-supply: true + + reset-gpios: + description: GPIO specification for the active low RESET input. + + ai31xx-micbias-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: [1, 2, 3] + description: | + MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + + ai31xx-ocmv: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V + + gpio-reset: + description: gpio pin number used for codec reset + deprecated: true + + +required: + - compatible + - reg + - HPVDD-supply + - SPRVDD-supply + - SPLVDD-supply + - AVDD-supply + - IOVDD-supply + - DVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320aic31xx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + sound@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_2_0V>; + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt deleted file mode 100644 index bbad98d5b9862..0000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ /dev/null @@ -1,77 +0,0 @@ -Texas Instruments - tlv320aic31xx Codec module - -The tlv320aic31xx serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp - "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp - "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) - "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) - "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) - "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) - "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) - "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) - -- reg - <int> - I2C slave address -- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, - DVDD-supply : power supplies for the device as covered in - Documentation/devicetree/bindings/regulator/regulator.txt - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai31xx-micbias-vg - MicBias Voltage setting - 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V - 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V - 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD - If this node is not mentioned or if the value is unknown, then - micbias is set to 2.0V. -- ai31xx-ocmv - output common-mode voltage setting - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * HPL - * HPR - * SPL, devices with stereo speaker amp - * SPR, devices with stereo speaker amp - * SPK, devices with mono speaker amp - * MICBIAS - -CODEC input pins: - * MIC1LP, devices with ADC - * MIC1RP, devices with ADC - * MIC1LM, devices with ADC - * AIN1, devices without ADC - * AIN2, devices without ADC - -The pins can be used in referring sound node's audio-routing property. - -Example: -#include <dt-bindings/gpio/gpio.h> -#include <dt-bindings/sound/tlv320aic31xx.h> - -tlv320aic31xx: tlv320aic31xx@18 { - compatible = "ti,tlv320aic311x"; - reg = <0x18>; - - ai31xx-micbias-vg = <MICBIAS_OFF>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - HPVDD-supply = <&regulator>; - SPRVDD-supply = <&regulator>; - SPLVDD-supply = <&regulator>; - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; -- 2.34.1

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