From 3dbb4e3602d217d7139b95a36077a6b7252dc290 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 4 Jan 2024 22:57:16 +0800 Subject: ASoC: dt-bindings: move tas2563 from tas2562.yaml to tas2781.yaml Move tas2563 from tas2562.yaml to tas2781.yaml to unbind tas2563 from tas2562 driver code and bind it to tas2781 driver code, because tas2563 only work in bypass-DSP mode with tas2562 driver. In order to enable DSP mode for tas2563, it has been moved to tas2781 driver. As to the hardware part, such as register setting and DSP firmware, all these are stored in the binary firmware. What tas2781 drivder does is to parse the firmware and download it to the chip, then power on the chip. So, tas2781 driver can be resued as tas2563 driver. Only attention will be paid to downloading corresponding firmware. Signed-off-by: Shenghao Ding Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240104145721.1398-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tas2562.yaml | 2 - .../devicetree/bindings/sound/ti,tas2781.yaml | 78 +++++++++++++++++----- 2 files changed, 63 insertions(+), 17 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index f01c0dde0cf7..d28c102c0ce7 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -18,7 +18,6 @@ description: | Specifications about the audio amplifier can be found at: https://www.ti.com/lit/gpn/tas2562 - https://www.ti.com/lit/gpn/tas2563 https://www.ti.com/lit/gpn/tas2564 https://www.ti.com/lit/gpn/tas2110 @@ -29,7 +28,6 @@ properties: compatible: enum: - ti,tas2562 - - ti,tas2563 - ti,tas2564 - ti,tas2110 diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml index a69e6c223308..976238689249 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -5,36 +5,46 @@ $id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: Texas Instruments TAS2781 SmartAMP +title: Texas Instruments TAS2563/TAS2781 SmartAMP maintainers: - Shenghao Ding -description: - The TAS2781 is a mono, digital input Class-D audio amplifier - optimized for efficiently driving high peak power into small - loudspeakers. An integrated on-chip DSP supports Texas Instruments - Smart Amp speaker protection algorithm. The integrated speaker - voltage and current sense provides for real time +description: | + The TAS2563/TAS2781 is a mono, digital input Class-D audio + amplifier optimized for efficiently driving high peak power into + small loudspeakers. An integrated on-chip DSP supports Texas + Instruments Smart Amp speaker protection algorithm. The + integrated speaker voltage and current sense provides for real time monitoring of loudspeaker behavior. -allOf: - - $ref: dai-common.yaml# + Specifications about the audio amplifier can be found at: + https://www.ti.com/lit/gpn/tas2563 + https://www.ti.com/lit/gpn/tas2781 properties: compatible: - enum: - - ti,tas2781 + description: | + ti,tas2563: 6.1-W Boosted Class-D Audio Amplifier With Integrated + DSP and IV Sense, 16/20/24/32bit stereo I2S or multichannel TDM. + + ti,tas2781: 24-V Class-D Amplifier with Real Time Integrated Speaker + Protection and Audio Processing, 16/20/24/32bit stereo I2S or + multichannel TDM. + oneOf: + - items: + - enum: + - ti,tas2563 + - const: ti,tas2781 + - enum: + - ti,tas2781 reg: description: - I2C address, in multiple tas2781s case, all the i2c address + I2C address, in multiple-AMP case, all the i2c address aggregate as one Audio Device to support multiple audio slots. maxItems: 8 minItems: 1 - items: - minimum: 0x38 - maximum: 0x3f reset-gpios: maxItems: 1 @@ -49,6 +59,44 @@ required: - compatible - reg +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - ti,tas2563 + then: + properties: + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 4 + minItems: 1 + items: + minimum: 0x4c + maximum: 0x4f + + - if: + properties: + compatible: + contains: + enum: + - ti,tas2781 + then: + properties: + reg: + description: + I2C address, in multiple-AMP case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + maxItems: 8 + minItems: 1 + items: + minimum: 0x38 + maximum: 0x3f + additionalProperties: false examples: -- cgit v1.2.3 From 645994d21287a1ad2f637818d737f7a3d84e97d7 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 4 Jan 2024 22:57:17 +0800 Subject: ASoC: tas2562: move tas2563 from tas2562 driver to tas2781 driver Move tas2563 from tas2562 driver to tas2781 driver to unbind tas2563 from tas2562 driver code and bind it to tas2781 driver code, because tas2563 only work in bypass-DSP mode with tas2562 driver. In order to enable DSP mode for tas2563, it has been moved to tas2781 driver. As to the hardware part, such as register setting and DSP firmware, all these are stored in the binary firmware. What tas2781 drivder does is to parse the firmware and download it to the chip, then power on the chip. So, tas2781 driver can be resued as tas2563 driver. Only attention will be paid to downloading corresponding firmware. Signed-off-by: Shenghao Ding Link: https://msgid.link/r/20240104145721.1398-2-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2562.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 962c2cdfa017..54561ae598b8 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -59,7 +59,6 @@ struct tas2562_data { enum tas256x_model { TAS2562, - TAS2563, TAS2564, TAS2110, }; @@ -721,7 +720,6 @@ static int tas2562_parse_dt(struct tas2562_data *tas2562) static const struct i2c_device_id tas2562_id[] = { { "tas2562", TAS2562 }, - { "tas2563", TAS2563 }, { "tas2564", TAS2564 }, { "tas2110", TAS2110 }, { } @@ -770,7 +768,6 @@ static int tas2562_probe(struct i2c_client *client) #ifdef CONFIG_OF static const struct of_device_id tas2562_of_match[] = { { .compatible = "ti,tas2562", }, - { .compatible = "ti,tas2563", }, { .compatible = "ti,tas2564", }, { .compatible = "ti,tas2110", }, { }, -- cgit v1.2.3 From e9aa44736cb75e901d76ee59d80db1ae79d516f1 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 4 Jan 2024 22:57:18 +0800 Subject: ASoC: tas2781: Add tas2563 into header file for DSP mode Move tas2563 from tas2562 header file to tas2781 header file to unbind tas2563 from tas2562 driver code and bind it to tas2781 driver code, because tas2563 only work in bypass-DSP mode with tas2562 driver. In order to enable DSP mode for tas2563, it has been moved to tas2781 driver. As to the hardware part, such as register setting and DSP firmware, all these are stored in the binary firmware. What tas2781 drivder does is to parse the firmware and download it to the chip, then power on the chip. So, tas2781 driver can be resued as tas2563 driver. Only attention will be paid to downloading corresponding firmware. Signed-off-by: Shenghao Ding Link: https://msgid.link/r/20240104145721.1398-3-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781.h | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index a6c808b22318..813cf9446a58 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -1,13 +1,13 @@ /* SPDX-License-Identifier: GPL-2.0 */ // -// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // // Copyright (C) 2022 - 2023 Texas Instruments Incorporated // https://www.ti.com // -// The TAS2781 driver implements a flexible and configurable +// The TAS2563/TAS2781 driver implements a flexible and configurable // algo coefficient setting for one, two, or even multiple -// TAS2781 chips. +// TAS2563/TAS2781 chips. // // Author: Shenghao Ding // Author: Kevin Lu @@ -59,7 +59,8 @@ #define TASDEVICE_CMD_FIELD_W 0x4 enum audio_device { - TAS2781 = 0, + TAS2563, + TAS2781, }; enum device_catlog_id { -- cgit v1.2.3 From 9f1bcd16e2bd41d758438f1d74e5f2d35f1e8c8e Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 4 Jan 2024 22:57:19 +0800 Subject: ASoC: tas2781: Add tas2563 into driver Move tas2563 from tas2562 driver to tas2781 driver to unbind tas2563 from tas2562 driver code and bind it to tas2781 driver code, because tas2563 only work in bypass-DSP mode with tas2562 driver. In order to enable DSP mode for tas2563, it has been moved to tas2781 driver. As to the hardware part, such as register setting and DSP firmware, all these are stored in the binary firmware. What tas2781 drivder does is to parse the firmware and download it to the chip, then power on the chip. So, tas2781 driver can be resued as tas2563 driver. Only attention will be paid to downloading corresponding firmware. Signed-off-by: Shenghao Ding Link: https://msgid.link/r/20240104145721.1398-4-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-i2c.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 55cd5e3c23a5..bd5ef4ff96fe 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1,13 +1,13 @@ // SPDX-License-Identifier: GPL-2.0 // -// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // // Copyright (C) 2022 - 2023 Texas Instruments Incorporated // https://www.ti.com // -// The TAS2781 driver implements a flexible and configurable +// The TAS2563/TAS2781 driver implements a flexible and configurable // algo coefficient setting for one, two, or even multiple -// TAS2781 chips. +// TAS2563/TAS2781 chips. // // Author: Shenghao Ding // Author: Kevin Lu @@ -32,6 +32,7 @@ #include static const struct i2c_device_id tasdevice_id[] = { + { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, {} }; @@ -39,6 +40,7 @@ MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, {}, }; -- cgit v1.2.3 From 35040410372ca27a33cec8382d42c90b6b6c99f6 Mon Sep 17 00:00:00 2001 From: ChiYuan Huang Date: Fri, 29 Dec 2023 09:46:01 +0800 Subject: ASoC: codecs: rtq9128: Fix PM_RUNTIME usage If 'pm_runtime_resume_and_get' is used, must check the return value to prevent the active count not matched problem. Signed-off-by: ChiYuan Huang Link: https://msgid.link/r/bebd9e2bed9e0528a7fd9c528d785da02caf4f1a.1703813842.git.cy_huang@richtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rtq9128.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rtq9128.c b/sound/soc/codecs/rtq9128.c index c22b047115cc..bda64f9eeb62 100644 --- a/sound/soc/codecs/rtq9128.c +++ b/sound/soc/codecs/rtq9128.c @@ -391,7 +391,11 @@ static int rtq9128_component_probe(struct snd_soc_component *comp) unsigned int val; int i, ret; - pm_runtime_resume_and_get(comp->dev); + ret = pm_runtime_resume_and_get(comp->dev); + if (ret < 0) { + dev_err(comp->dev, "Failed to resume device (%d)\n", ret); + return ret; + } val = snd_soc_component_read(comp, RTQ9128_REG_EFUSE_DATA); -- cgit v1.2.3 From 415d10ccef712f3ec73cd880c1fef3eb48601c3a Mon Sep 17 00:00:00 2001 From: ChiYuan Huang Date: Fri, 29 Dec 2023 09:46:02 +0800 Subject: ASoC: codecs: rtq9128: Fix TDM enable and DAI format control flow To enable TDM mode, the current control flow limits the function calling order should be 'set_tdm_slot->set_dai_fmt'. But not all platform sound card like as simeple card to follow this design. To bypass this limit, adjust the DAI format setting in runtime 'hw_param' callback. Signed-off-by: ChiYuan Huang Link: https://msgid.link/r/c4c8df00d8d179b8b5b39a8521de3a85325c57e8.1703813842.git.cy_huang@richtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rtq9128.c | 67 +++++++++++++++++++++++++--------------------- 1 file changed, 36 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/rtq9128.c b/sound/soc/codecs/rtq9128.c index bda64f9eeb62..aa3eadecd974 100644 --- a/sound/soc/codecs/rtq9128.c +++ b/sound/soc/codecs/rtq9128.c @@ -59,6 +59,7 @@ struct rtq9128_data { struct gpio_desc *enable; + unsigned int daifmt; int tdm_slots; int tdm_slot_width; bool tdm_input_data2_select; @@ -441,10 +442,7 @@ static const struct snd_soc_component_driver rtq9128_comp_driver = { static int rtq9128_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct rtq9128_data *data = snd_soc_dai_get_drvdata(dai); - struct snd_soc_component *comp = dai->component; struct device *dev = dai->dev; - unsigned int audfmt, fmtval; - int ret; dev_dbg(dev, "%s: fmt 0x%8x\n", __func__, fmt); @@ -454,35 +452,10 @@ static int rtq9128_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - fmtval = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - if (data->tdm_slots && fmtval != SND_SOC_DAIFMT_DSP_A && fmtval != SND_SOC_DAIFMT_DSP_B) { - dev_err(dev, "TDM is used, format only support DSP_A or DSP_B\n"); - return -EINVAL; - } + /* Store here and will be used in runtime hw_params for DAI format setting */ + data->daifmt = fmt; - switch (fmtval) { - case SND_SOC_DAIFMT_I2S: - audfmt = 8; - break; - case SND_SOC_DAIFMT_LEFT_J: - audfmt = 9; - break; - case SND_SOC_DAIFMT_RIGHT_J: - audfmt = 10; - break; - case SND_SOC_DAIFMT_DSP_A: - audfmt = data->tdm_slots ? 12 : 11; - break; - case SND_SOC_DAIFMT_DSP_B: - audfmt = data->tdm_slots ? 4 : 3; - break; - default: - dev_err(dev, "Unsupported format 0x%8x\n", fmt); - return -EINVAL; - } - - ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDFMT_MASK, audfmt); - return ret < 0 ? ret : 0; + return 0; } static int rtq9128_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, @@ -558,10 +531,38 @@ static int rtq9128_dai_hw_params(struct snd_pcm_substream *stream, struct snd_pc unsigned int width, slot_width, bitrate, audbit, dolen; struct snd_soc_component *comp = dai->component; struct device *dev = dai->dev; + unsigned int fmtval, audfmt; int ret; dev_dbg(dev, "%s: width %d\n", __func__, params_width(param)); + fmtval = FIELD_GET(SND_SOC_DAIFMT_FORMAT_MASK, data->daifmt); + if (data->tdm_slots && fmtval != SND_SOC_DAIFMT_DSP_A && fmtval != SND_SOC_DAIFMT_DSP_B) { + dev_err(dev, "TDM is used, format only support DSP_A or DSP_B\n"); + return -EINVAL; + } + + switch (fmtval) { + case SND_SOC_DAIFMT_I2S: + audfmt = 8; + break; + case SND_SOC_DAIFMT_LEFT_J: + audfmt = 9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audfmt = 10; + break; + case SND_SOC_DAIFMT_DSP_A: + audfmt = data->tdm_slots ? 12 : 11; + break; + case SND_SOC_DAIFMT_DSP_B: + audfmt = data->tdm_slots ? 4 : 3; + break; + default: + dev_err(dev, "Unsupported format 0x%8x\n", fmtval); + return -EINVAL; + } + switch (width = params_width(param)) { case 16: audbit = 0; @@ -615,6 +616,10 @@ static int rtq9128_dai_hw_params(struct snd_pcm_substream *stream, struct snd_pc return -EINVAL; } + ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDFMT_MASK, audfmt); + if (ret < 0) + return ret; + ret = snd_soc_component_write_field(comp, RTQ9128_REG_I2S_OPT, RTQ9128_AUDBIT_MASK, audbit); if (ret < 0) return ret; -- cgit v1.2.3 From 59b946ea30806064c4ac78f0ac93642655dd4f2e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 8 Jan 2024 11:48:41 +0200 Subject: ASoC: Intel: bxt_da7219_max98357a: Fix kernel ops due to COMP_DUMMY change The change to avoid dummy components will leave the component name and dai_name NULL which will cause NULL dereference when trying to access to it in the machine driver when applying fixups. Link: https://github.com/thesofproject/linux/pull/4759#issuecomment-1878641868 Fixes: 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240108094842.28782-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 816fad8c1ff0..540f7a29310a 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -797,6 +797,9 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_audio_card.name = "glkda7219max"; /* Fixup the SSP entries for geminilake */ for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { + if (!broxton_dais[i].codecs->dai_name) + continue; + /* MAXIM_CODEC is connected to SSP1. */ if (!strcmp(broxton_dais[i].codecs->dai_name, BXT_MAXIM_CODEC_DAI)) { @@ -822,6 +825,9 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_audio_card.name = "cmlda7219max"; for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { + if (!broxton_dais[i].codecs->dai_name) + continue; + /* MAXIM_CODEC is connected to SSP1. */ if (!strcmp(broxton_dais[i].codecs->dai_name, BXT_MAXIM_CODEC_DAI)) { -- cgit v1.2.3 From 3ec71290db4de298b67659ef2bc2a8f84cf9537b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 8 Jan 2024 11:48:42 +0200 Subject: ASoC: Intel: bxt_rt298: Fix kernel ops due to COMP_DUMMY change The change to avoid dummy components will leave the component name and dai_name NULL which will cause NULL dereference when trying to access to it in the machine driver when applying fixups. Link: https://github.com/thesofproject/linux/pull/4759#issuecomment-1878641868 Fixes: 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240108094842.28782-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_rt298.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 4631106f2a28..c0eb65c14aa9 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -604,7 +604,8 @@ static int broxton_audio_probe(struct platform_device *pdev) int i; for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) { - if (!strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00", + if (card->dai_link[i].codecs->name && + !strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00", I2C_NAME_SIZE)) { if (!strncmp(card->name, "broxton-rt298", PLATFORM_NAME_SIZE)) { -- cgit v1.2.3 From 4d4e1b6319e5c4425ea3faeaf9a10b8b4c16c1e1 Mon Sep 17 00:00:00 2001 From: "Nícolas F. R. A. Prado" Date: Mon, 8 Jan 2024 17:44:58 -0300 Subject: ASoC: mediatek: mt8192: Check existence of dai_name before dereferencing MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Following commit 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()"), the dai_name field is only populated for dummy components after the card is registered. This causes a null pointer dereference in the mt8192-mt6359 sound card driver's probe function when searching for a dai_name among all the card's dai links. Verify that the dai_name is non-null before passing it to strcmp. While at it, also check that there's at least one codec. Reported-by: kernelci.org bot Closes: https://linux.kernelci.org/test/case/id/6582cd6d992645c680e13478/ Fixes: 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") Signed-off-by: Nícolas F. R. A. Prado Link: https://msgid.link/r/20240108204508.691739-1-nfraprado@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 5bd6addd1450..bfcb2c486c39 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1208,7 +1208,8 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) dai_link->ignore = 0; } - if (strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) + if (dai_link->num_codecs && dai_link->codecs[0].dai_name && + strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) dai_link->ops = &mt8192_rt1015_i2s_ops; if (!dai_link->platforms->name) -- cgit v1.2.3 From e70b8dd26711704b1ff1f1b4eb3d048ba69e29da Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Wed, 10 Jan 2024 11:57:57 +0100 Subject: ASoC: mediatek: mt8195: Remove afe-dai component and rework codec link Remove the extra 'mt8195-afe-pcm-dai' component, register the DAI drivers to the main AFE component, and rework the DAI linking between the headset codec (RT5682/RT5682S) and the TDM interface in the probe function to stop assigning name, relying on the of_node of the codec. Also replace the COMP_DUMMY codec entry with a COMP_EMPTY for the ETDM2_IN and remove it entirely from ETDM1_OUT to fix the registration flow for this sound card. While at it, since we also need to swap the codec init function from ETDM2_IN to ETDM1_OUT, remove the static assignment of both `ops` and `init` for both, as we now assign these dynamically during probe. Fixes: 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") Signed-off-by: AngeloGioacchino Del Regno Link: https://msgid.link/r/20240110105757.539089-1-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 33 +----------------------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 41 ++++++++++++++++++++---------- 2 files changed, 29 insertions(+), 45 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 1e33863c85ca..620d7ade1992 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -1795,10 +1795,6 @@ static const struct snd_kcontrol_new mt8195_memif_controls[] = { MT8195_AFE_IRQ_28), }; -static const struct snd_soc_component_driver mt8195_afe_pcm_dai_component = { - .name = "mt8195-afe-pcm-dai", -}; - static const struct mtk_base_memif_data memif_data[MT8195_AFE_MEMIF_NUM] = { [MT8195_AFE_MEMIF_DL2] = { .name = "DL2", @@ -3037,7 +3033,6 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct reset_control *rstc; int i, irq_id, ret; - struct snd_soc_component *component; ret = of_reserved_mem_device_init(dev); if (ret) @@ -3170,36 +3165,12 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) /* register component */ ret = devm_snd_soc_register_component(dev, &mt8195_afe_component, - NULL, 0); + afe->dai_drivers, afe->num_dai_drivers); if (ret) { dev_warn(dev, "err_platform\n"); goto err_pm_put; } - component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL); - if (!component) { - ret = -ENOMEM; - goto err_pm_put; - } - - ret = snd_soc_component_initialize(component, - &mt8195_afe_pcm_dai_component, - dev); - if (ret) - goto err_pm_put; - -#ifdef CONFIG_DEBUG_FS - component->debugfs_prefix = "pcm"; -#endif - - ret = snd_soc_add_component(component, - afe->dai_drivers, - afe->num_dai_drivers); - if (ret) { - dev_warn(dev, "err_dai_component\n"); - goto err_pm_put; - } - ret = regmap_multi_reg_write(afe->regmap, mt8195_afe_reg_defaults, ARRAY_SIZE(mt8195_afe_reg_defaults)); if (ret) @@ -3224,8 +3195,6 @@ err_pm_put: static void mt8195_afe_pcm_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt8195_afe_runtime_suspend(&pdev->dev); diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 4feb9fb76967..53fd8a897b9d 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -934,12 +934,11 @@ SND_SOC_DAILINK_DEFS(ETDM1_IN_BE, SND_SOC_DAILINK_DEFS(ETDM2_IN_BE, DAILINK_COMP_ARRAY(COMP_CPU("ETDM2_IN")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(ETDM1_OUT_BE, DAILINK_COMP_ARRAY(COMP_CPU("ETDM1_OUT")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(ETDM2_OUT_BE, @@ -1237,8 +1236,6 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_capture = 1, - .init = mt8195_rt5682_init, - .ops = &mt8195_rt5682_etdm_ops, .be_hw_params_fixup = mt8195_etdm_hw_params_fixup, SND_SOC_DAILINK_REG(ETDM2_IN_BE), }, @@ -1249,7 +1246,6 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_playback = 1, - .ops = &mt8195_rt5682_etdm_ops, .be_hw_params_fixup = mt8195_etdm_hw_params_fixup, SND_SOC_DAILINK_REG(ETDM1_OUT_BE), }, @@ -1381,7 +1377,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link; struct mtk_soc_card_data *soc_card_data; struct mt8195_mt6359_priv *mach_priv; - struct device_node *platform_node, *adsp_node, *dp_node, *hdmi_node; + struct device_node *platform_node, *adsp_node, *codec_node, *dp_node, *hdmi_node; struct mt8195_card_data *card_data; int is5682s = 0; int init6359 = 0; @@ -1401,8 +1397,12 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) if (!card->name) card->name = card_data->name; - if (strstr(card->name, "_5682s")) + if (strstr(card->name, "_5682s")) { + codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682s"); is5682s = 1; + } else + codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682i"); + soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL); if (!soc_card_data) return -ENOMEM; @@ -1488,12 +1488,27 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_hdmi_codec_init; } - } else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0 || - strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { - dai_link->codecs->name = - is5682s ? RT5682S_DEV0_NAME : RT5682_DEV0_NAME; - dai_link->codecs->dai_name = - is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI; + } else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0) { + if (!codec_node) { + dev_err(&pdev->dev, "Codec not found!\n"); + } else { + dai_link->codecs->of_node = codec_node; + dai_link->codecs->name = NULL; + dai_link->codecs->dai_name = + is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI; + dai_link->init = mt8195_rt5682_init; + dai_link->ops = &mt8195_rt5682_etdm_ops; + } + } else if (strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { + if (!codec_node) { + dev_err(&pdev->dev, "Codec not found!\n"); + } else { + dai_link->codecs->of_node = codec_node; + dai_link->codecs->name = NULL; + dai_link->codecs->dai_name = + is5682s ? RT5682S_CODEC_DAI : RT5682_CODEC_DAI; + dai_link->ops = &mt8195_rt5682_etdm_ops; + } } else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 || strcmp(dai_link->name, "UL_SRC1_BE") == 0 || strcmp(dai_link->name, "UL_SRC2_BE") == 0) { -- cgit v1.2.3 From e3b3ec967a7d93b9010a5af9a2394c8b5c8f31ed Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Thu, 11 Jan 2024 11:52:26 +0100 Subject: ASoC: mediatek: sof-common: Add NULL check for normal_link string It's not granted that all entries of struct sof_conn_stream declare a `normal_link` (a non-SOF, direct link) string, and this is the case for SoCs that support only SOF paths (hence do not support both direct and SOF usecases). For example, in the case of MT8188 there is no normal_link string in any of the sof_conn_stream entries and there will be more drivers doing that in the future. To avoid possible NULL pointer KPs, add a NULL check for `normal_link`. Fixes: 0caf1120c583 ("ASoC: mediatek: mt8195: extract SOF common code") Signed-off-by: AngeloGioacchino Del Regno Link: https://msgid.link/r/20240111105226.117603-1-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-dsp-sof-common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c index f3894010f656..7ec8965a70c0 100644 --- a/sound/soc/mediatek/common/mtk-dsp-sof-common.c +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c @@ -24,7 +24,7 @@ int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai_link *sof_dai_link = NULL; const struct sof_conn_stream *conn = &sof_priv->conn_streams[i]; - if (strcmp(rtd->dai_link->name, conn->normal_link)) + if (conn->normal_link && strcmp(rtd->dai_link->name, conn->normal_link)) continue; for_each_card_rtds(card, runtime) { -- cgit v1.2.3 From cdac6e1f716419ce307ad3e44a718557a5469c17 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Thu, 11 Jan 2024 11:52:19 +0900 Subject: ALSA: aloop: Introduce a function to get if access is interleaved mode There's a use case that playback stream of a loopback cable works on RW_INTERLEAVED mode while capture stream works on MMAP_INTERLEAVED mode: aplay -Dhw:Loopback,0,0 S32_48K_2ch.wav; arecord -Dplughw:Loopback,1,0 -fS32_LE -r16000 -c2 cap.wav; The plug plugin handles only slave PCM support MMAP mode. Not only plug plugin but also other plugins like direct plugins(dmix/dsnoop/dshare) work on MMAP access mode. In this case capture stream is the slave PCM works on MMAP_INTERLEAVED mode. However loopback_check_format() rejects this access setting and return: arecord: pcm_read:2240: read error: Input/output error To fix it a function called is_access_interleaved() is introduced to get if access is interleaved mode. If both access of capture stream and playback stream is interleaved mode loopback_check_format() will allow this kind of access setting. Fixes: 462494565c27 ("ALSA: aloop: Add support for the non-interleaved access mode") Signed-off-by: Chancel Liu Link: https://lore.kernel.org/r/20240111025219.2678764-1-chancel.liu@nxp.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index e87dc67f33c6..1c65e0a3b13c 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -322,6 +322,17 @@ static int loopback_snd_timer_close_cable(struct loopback_pcm *dpcm) return 0; } +static bool is_access_interleaved(snd_pcm_access_t access) +{ + switch (access) { + case SNDRV_PCM_ACCESS_MMAP_INTERLEAVED: + case SNDRV_PCM_ACCESS_RW_INTERLEAVED: + return true; + default: + return false; + } +}; + static int loopback_check_format(struct loopback_cable *cable, int stream) { struct snd_pcm_runtime *runtime, *cruntime; @@ -341,7 +352,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) check = runtime->format != cruntime->format || runtime->rate != cruntime->rate || runtime->channels != cruntime->channels || - runtime->access != cruntime->access; + is_access_interleaved(runtime->access) != + is_access_interleaved(cruntime->access); if (!check) return 0; if (stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -369,7 +381,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) &setup->channels_id); setup->channels = runtime->channels; } - if (setup->access != runtime->access) { + if (is_access_interleaved(setup->access) != + is_access_interleaved(runtime->access)) { snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &setup->access_id); setup->access = runtime->access; @@ -584,8 +597,7 @@ static void copy_play_buf(struct loopback_pcm *play, size = play->pcm_buffer_size - src_off; if (dst_off + size > capt->pcm_buffer_size) size = capt->pcm_buffer_size - dst_off; - if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || - runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) + if (!is_access_interleaved(runtime->access)) copy_play_buf_part_n(play, capt, size, src_off, dst_off); else memcpy(dst + dst_off, src + src_off, size); @@ -1544,8 +1556,7 @@ static int loopback_access_get(struct snd_kcontrol *kcontrol, mutex_lock(&loopback->cable_lock); access = loopback->setup[kcontrol->id.subdevice][kcontrol->id.device].access; - ucontrol->value.enumerated.item[0] = access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || - access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED; + ucontrol->value.enumerated.item[0] = !is_access_interleaved(access); mutex_unlock(&loopback->cable_lock); return 0; -- cgit v1.2.3 From a03cfad512ac24a35184d7d87ec0d5489e1cb763 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Jan 2024 12:10:23 +0100 Subject: ALSA: oxygen: Fix right channel of capture volume mixer There was a typo in oxygen mixer code that didn't update the right channel value properly for the capture volume. Let's fix it. This trivial fix was originally reported on Bugzilla. Fixes: a3601560496d ("[ALSA] oxygen: add front panel controls") Cc: Link: https://bugzilla.kernel.org/show_bug.cgi?id=156561 Link: https://lore.kernel.org/r/20240112111023.6208-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 46705ec77b48..eb3aca16359c 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -718,7 +718,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN); newreg = oldreg & ~0x0707; newreg = newreg | (value->value.integer.value[0] & 7); - newreg = newreg | ((value->value.integer.value[0] & 7) << 8); + newreg = newreg | ((value->value.integer.value[1] & 7) << 8); change = newreg != oldreg; if (change) oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg); -- cgit v1.2.3 From 813c2f2925ee9c10dc4acd5aa7410cd3357e8da8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 12 Jan 2024 15:27:49 +0200 Subject: ASoC: SOF: icp3-dtrace: Revert "Fix wrong kfree() usage" The offending patch introduces memory leak when there is no error, the memory allocated for the temporary storage is not freed up. As I have commented, the original code was correct and cleaner to follow but it was not obvious from the patch that it will introduce regression. Fixes: 8c91ca76f448 ("ASoC: SOF: icp3-dtrace: Fix wrong kfree() usage") Link: https://lore.kernel.org/all/aec61f67-6b4f-49e6-b458-c332983a0ad6@linux.intel.com/ Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240112132749.28970-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-dtrace.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 93b189c2d2ee..0dca139322f3 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -137,7 +137,6 @@ static int trace_filter_parse(struct snd_sof_dev *sdev, char *string, dev_err(sdev->dev, "Parsing filter entry '%s' failed with %d\n", entry, entry_len); - kfree(*out); return -EINVAL; } } @@ -209,13 +208,13 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user ret = ipc3_trace_update_filter(sdev, num_elems, elems); if (ret < 0) { dev_err(sdev->dev, "Filter update failed: %d\n", ret); - kfree(elems); goto error; } } ret = count; error: kfree(string); + kfree(elems); return ret; } -- cgit v1.2.3 From 301bda18ac735eaaad5823dbdd067b3b2728c780 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 12 Jan 2024 06:10:15 +0000 Subject: ASoC: audio-graph-card2: fix index check on graph_parse_node_multi_nm() commit d685aea5e0a8 ("ASoC: audio-graph-card2: fix off by one in graph_parse_node_multi_nm()") uses ">=" instead of ">" for index check, but it was wrong. The nm_idx will be increment at end of loop, thus, ">" is correct. while (1) { ... => if (*nm_idx > nm_max) break; ... (*nm_idx)++; } Without this patch, "Multi-Codec-1" sample on ${LINUX}/sound/soc/generic/audio-graph-card2-custom-sample.dtsi will be error. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87o7drdqux.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 9c94677f681a..62606e20be9a 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -556,7 +556,7 @@ static int graph_parse_node_multi_nm(struct snd_soc_dai_link *dai_link, struct device_node *mcodec_port; int codec_idx; - if (*nm_idx >= nm_max) + if (*nm_idx > nm_max) break; mcpu_ep_n = of_get_next_child(mcpu_port, mcpu_ep_n); -- cgit v1.2.3 From 454abb80e26ab85323a30e52aa7b0ee9aae1d38a Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 12 Jan 2024 12:33:49 +0100 Subject: ALSA: hda: Properly setup HDMI stream MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since commit 4005d1ba0a7e ("ASoC: soc-dai: don't call PCM audio ops if the stream is not supported") HDMI playback is broken with avs driver. This happens because for HDMI stream (unlike generic HDA one) channels_min for stream is not set when creating PCMs. Fix this by setting the value based on first available converter. Fixes: 4005d1ba0a7e ("ASoC: soc-dai: don't call PCM audio ops if the stream is not supported") Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240112113349.2905328-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 200779296a1b..495d63101186 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2301,6 +2301,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) codec_dbg(codec, "hdmi: pcm_num set to %d\n", pcm_num); for (idx = 0; idx < pcm_num; idx++) { + struct hdmi_spec_per_cvt *per_cvt; struct hda_pcm *info; struct hda_pcm_stream *pstr; @@ -2316,6 +2317,11 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; pstr->substreams = 1; pstr->ops = generic_ops; + + per_cvt = get_cvt(spec, 0); + pstr->channels_min = per_cvt->channels_min; + pstr->channels_max = per_cvt->channels_max; + /* pcm number is less than pcm_rec array size */ if (spec->pcm_used >= ARRAY_SIZE(spec->pcm_rec)) break; -- cgit v1.2.3 From 3787ffdd13de81ba406e5b42c6c24f823395ba5e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Jan 2024 18:10:00 +0100 Subject: ALSA: scarlett2: Fix yet more -Wformat-truncation warnings The recent code change introduced a few false-positive compile warnings with -Wformat-trucation again. Suppress them by replacing snprintf() with scnprintf(). Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202401062344.AzZCYlpa-lkp@intel.com/ Link: https://lore.kernel.org/r/20240112171000.31855-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 42 +++++++++++++++++++++--------------------- 1 file changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 1de3ddc50eb6..6de605a601e5 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -5361,9 +5361,9 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) if (private->vol_sw_hw_switch[index]) scarlett2_vol_ctl_set_writable(mixer, i, 0); - snprintf(s, sizeof(s), - "Line Out %02d Volume Control Playback Enum", - i + 1); + scnprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); err = scarlett2_add_new_ctl(mixer, &scarlett2_sw_hw_enum_ctl, i, 1, s, @@ -5406,8 +5406,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input level (line/inst) controls */ for (i = 0; i < info->level_input_count; i++) { - snprintf(s, sizeof(s), fmt, i + 1 + info->level_input_first, - "Level", "Enum"); + scnprintf(s, sizeof(s), fmt, i + 1 + info->level_input_first, + "Level", "Enum"); err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, i, 1, s, &private->level_ctls[i]); if (err < 0) @@ -5416,7 +5416,7 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input pad controls */ for (i = 0; i < info->pad_input_count; i++) { - snprintf(s, sizeof(s), fmt, i + 1, "Pad", "Switch"); + scnprintf(s, sizeof(s), fmt, i + 1, "Pad", "Switch"); err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, i, 1, s, &private->pad_ctls[i]); if (err < 0) @@ -5425,8 +5425,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input air controls */ for (i = 0; i < info->air_input_count; i++) { - snprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first, - "Air", info->air_option ? "Enum" : "Switch"); + scnprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first, + "Air", info->air_option ? "Enum" : "Switch"); err = scarlett2_add_new_ctl( mixer, &scarlett2_air_ctl[info->air_option], i, 1, s, &private->air_ctls[i]); @@ -5481,9 +5481,9 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) for (i = 0; i < info->gain_input_count; i++) { if (i % 2) { - snprintf(s, sizeof(s), - "Line In %d-%d Link Capture Switch", - i, i + 1); + scnprintf(s, sizeof(s), + "Line In %d-%d Link Capture Switch", + i, i + 1); err = scarlett2_add_new_ctl( mixer, &scarlett2_input_link_ctl, i / 2, 1, s, @@ -5492,30 +5492,30 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) return err; } - snprintf(s, sizeof(s), fmt, i + 1, - "Gain", "Volume"); + scnprintf(s, sizeof(s), fmt, i + 1, + "Gain", "Volume"); err = scarlett2_add_new_ctl( mixer, &scarlett2_input_gain_ctl, i, 1, s, &private->input_gain_ctls[i]); if (err < 0) return err; - snprintf(s, sizeof(s), fmt, i + 1, - "Autogain", "Switch"); + scnprintf(s, sizeof(s), fmt, i + 1, + "Autogain", "Switch"); err = scarlett2_add_new_ctl( mixer, &scarlett2_autogain_switch_ctl, i, 1, s, &private->autogain_ctls[i]); if (err < 0) return err; - snprintf(s, sizeof(s), fmt, i + 1, - "Autogain Status", "Enum"); + scnprintf(s, sizeof(s), fmt, i + 1, + "Autogain Status", "Enum"); err = scarlett2_add_new_ctl( mixer, &scarlett2_autogain_status_ctl, i, 1, s, &private->autogain_status_ctls[i]); - snprintf(s, sizeof(s), fmt, i + 1, - "Safe", "Switch"); + scnprintf(s, sizeof(s), fmt, i + 1, + "Safe", "Switch"); err = scarlett2_add_new_ctl( mixer, &scarlett2_safe_ctl, i, 1, s, &private->safe_ctls[i]); @@ -5902,8 +5902,8 @@ static int scarlett2_add_direct_monitor_ctls(struct usb_mixer_interface *mixer) for (k = 0; k < private->num_mix_in; k++, index++) { char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - snprintf(name, sizeof(name), format, - mix_type, 'A' + j, k + 1); + scnprintf(name, sizeof(name), format, + mix_type, 'A' + j, k + 1); err = scarlett2_add_new_ctl( mixer, &scarlett2_monitor_mix_ctl, -- cgit v1.2.3 From 19adbe96d3e3c2188ad5838b936550e073cba54d Mon Sep 17 00:00:00 2001 From: Heiner Kallweit Date: Sat, 13 Jan 2024 17:08:54 +0100 Subject: ALSA: hda: generic: Remove obsolete call to ledtrig_audio_get Since 64f67b5240db ("leds: trigger: audio: Add an activate callback to ensure the initial brightness is set") the audio triggers have an activate callback which sets the LED brightness as soon as the (default) trigger is bound to the LED device. So we can remove the call to ledtrig_audio_get. Positive side effect: We have no code dependency to ledtrig-audio any longer, therefore, if built as module, it's no longer loaded if not needed. Signed-off-by: Heiner Kallweit Link: https://lore.kernel.org/r/3dc9167d-fb33-43a6-baa6-dbef8b5da7b9@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index bf685d01259d..de2a3d08c73c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3946,7 +3946,6 @@ static int create_mute_led_cdev(struct hda_codec *codec, cdev->max_brightness = 1; cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute"; cdev->brightness_set_blocking = callback; - cdev->brightness = ledtrig_audio_get(idx); cdev->flags = LED_CORE_SUSPENDRESUME; err = led_classdev_register(&codec->core.dev, cdev); -- cgit v1.2.3 From 848c8f563dadfdf01358b001ef7c9afe2a6ece8f Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Mon, 15 Jan 2024 11:22:08 +0200 Subject: ASoC: SOF: ipc4-pcm: remove log message for LLP MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit LLP is supported by I2S and SDW platforms but not supported by HDA platforms. This log is used to notify developer current LLP status for current device. Since above case is known to developers, the log is unnecessary and should be removed. Fixes: 7cb19007baba ("ASoC: SOF: ipc4-pcm: add hw_params") Signed-off-by: Rander Wang Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240115092209.7184-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 39039a647cca..85d3f390e4b2 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -768,10 +768,8 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_evad_reading_slot) + sdev->fw_info_box.offset; sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot)); - if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id) { - dev_info(sdev->dev, "no llp found, fall back to default HDA path"); + if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id) info->llp_offset = 0; - } } static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, -- cgit v1.2.3 From ab09fb9c629ed3aaea6a82467f08595dbc549726 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Mon, 15 Jan 2024 11:22:09 +0200 Subject: ASoC: SOF: ipc4-loader: remove the CPC check warnings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Warnings related to missing data in firmware manifest have proven to be too verbose. This relates to description of DSP module cost expressed in cycles per chunk (CPC). If a matching value is not found in the manifest, kernel will pass a zero value and DSP firmware will use a conservative value in its place. Downgrade the warnings to dev_dbg(). Fixes: d8a2c9879349 ("ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest") Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Liam Girdwood Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240115092209.7184-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-loader.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index 3539b0a66e1b..c79479afa8d0 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -482,13 +482,10 @@ void sof_ipc4_update_cpc_from_manifest(struct snd_sof_dev *sdev, msg = "No CPC match in the firmware file's manifest"; no_cpc: - dev_warn(sdev->dev, "%s (UUID: %pUL): %s (ibs/obs: %u/%u)\n", - fw_module->man4_module_entry.name, - &fw_module->man4_module_entry.uuid, msg, basecfg->ibs, - basecfg->obs); - dev_warn_once(sdev->dev, "Please try to update the firmware.\n"); - dev_warn_once(sdev->dev, "If the issue persists, file a bug at\n"); - dev_warn_once(sdev->dev, "https://github.com/thesofproject/sof/issues/\n"); + dev_dbg(sdev->dev, "%s (UUID: %pUL): %s (ibs/obs: %u/%u)\n", + fw_module->man4_module_entry.name, + &fw_module->man4_module_entry.uuid, msg, basecfg->ibs, + basecfg->obs); } const struct sof_ipc_fw_loader_ops ipc4_loader_ops = { -- cgit v1.2.3 From bc7863d18677df66b2c7a0e172c91296ff380f11 Mon Sep 17 00:00:00 2001 From: Çağhan Demir Date: Mon, 15 Jan 2024 20:23:03 +0300 Subject: ALSA: hda/relatek: Enable Mute LED on HP Laptop 15s-fq2xxx MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This HP Laptop uses ALC236 codec with COEF 0x07 idx 1 controlling the mute LED. This patch enables the already existing quirk for this device. Signed-off-by: Çağhan Demir Cc: Link: https://lore.kernel.org/r/20240115172303.4718-1-caghandemir@marun.edu.tr Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b68c94757051..0f0a03e89015 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9861,6 +9861,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), -- cgit v1.2.3 From b018cee7369896c7a15bfdbe88f168f3dbd8ba27 Mon Sep 17 00:00:00 2001 From: Yo-Jung Lin Date: Tue, 16 Jan 2024 10:07:19 +0800 Subject: ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on HP ZBook On some HP ZBooks, the audio LEDs can be enabled by ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF. So use it accordingly. Signed-off-by: Yo-Jung Lin Cc: Link: https://lore.kernel.org/r/20240116020722.27236-1-leo.lin@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0f0a03e89015..dbf31fe901da 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9956,6 +9956,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8c97, "HP ZBook", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8ca4, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), -- cgit v1.2.3 From fb3c007fde80d9d3b4207943e74c150c9116cead Mon Sep 17 00:00:00 2001 From: Bin Li Date: Wed, 17 Jan 2024 23:41:23 +0800 Subject: ALSA: hda/realtek: Enable headset mic on Lenovo M70 Gen5 Lenovo M70 Gen5 is equipped with ALC623, and it needs ALC283_FIXUP_HEADSET_MIC quirk to make its headset mic work. Signed-off-by: Bin Li Cc: Link: https://lore.kernel.org/r/20240117154123.21578-1-bin.li@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dbf31fe901da..f6f16622f9cc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10233,6 +10233,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), + SND_PCI_QUIRK(0x17aa, 0x334b, "Lenovo ThinkCentre M70 Gen5", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), -- cgit v1.2.3