From 58eafe1ff52ee1ce255759fc15729519af180cbb Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Sep 2021 13:44:36 -0500 Subject: [PATCH 01/45] ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic The SoundWire BEs make use of 'stream' functions for .prepare and .trigger. These functions will in turn force a Bank Switch, which implies a wait operation. Mark SoundWire BEs as nonatomic for consistency, but keep all other types of BEs as is. The initialization of .nonatomic is done outside of the create_sdw_dailink helper to avoid adding more parameters to deal with a single exception to the rule that BEs are atomic. Suggested-by: Takashi Iwai Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20210907184436.33152-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6602eda89e8ef..6b06248a9327a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -929,6 +929,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, cpus + *cpu_id, cpu_dai_num, codecs, codec_num, NULL, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + dai_links[*be_index].nonatomic = true; ret = set_codec_init_func(card, link, dai_links + (*be_index)++, playback, group_id); From 5a80dea93191d55840f42252ed3e4565a125a514 Mon Sep 17 00:00:00 2001 From: Trevor Wu Date: Thu, 9 Sep 2021 14:55:33 +0800 Subject: [PATCH 02/45] ASoC: mediatek: add required config dependency Because SND_SOC_MT8195 depends on COMPILE_TEST, it's possible to build MT8195 driver in many different config combinations. Add all dependent config for SND_SOC_MT8195 in case some errors happen when COMPILE_TEST is enabled. Signed-off-by: Trevor Wu Reported-by: Randy Dunlap Link: https://lore.kernel.org/r/20210909065533.2114-1-trevor.wu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 5a2f4667d50b3..81ad2dcee9ebc 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -1,6 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_SOC_MEDIATEK tristate + select REGMAP_MMIO config SND_SOC_MT2701 tristate "ASoC support for Mediatek MT2701 chip" @@ -188,7 +189,9 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 config SND_SOC_MT8195 tristate "ASoC support for Mediatek MT8195 chip" depends on ARCH_MEDIATEK || COMPILE_TEST + depends on COMMON_CLK select SND_SOC_MEDIATEK + select MFD_SYSCON if SND_SOC_MT6359 help This adds ASoC platform driver support for Mediatek MT8195 chip that can be used with other codecs. From 26be23af1866eead5a29f8501f9d774ac277d0bd Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen Date: Thu, 9 Sep 2021 16:54:49 +0200 Subject: [PATCH 03/45] MAINTAINERS: fix update references to stm32 audio bindings The 00d38fd8d2524 ("MAINTAINERS: update references to stm32 audio bindings") commit update the bindings reference, by removing bindings/sound/st,stm32-adfsdm.txt, to set the new reference to bindings/iio/adc/st,stm32-*.yaml. This leads to "get_maintainer finds" the match for the Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml, but also to the IIO bindings Documentation/devicetree/bindings/iio/adc/st,stm32-adc.yaml And The commit fixes only a part of the problem: Documentation/devicetree/bindings/sound/st,stm32-*.txt file have been also moved to yaml. Update references to include all stm32 audio bindings file and exclude the st,stm32-adc.yaml bindings file. cc: Mauro Carvalho Chehab Fixes: 0d38fd8d2524 ("MAINTAINERS: update references to stm32 audio bindings") Signed-off-by: Arnaud Pouliquen Link: https://lore.kernel.org/r/20210909145449.24388-1-arnaud.pouliquen@foss.st.com Signed-off-by: Mark Brown --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index c6b8a720c0bcc..33d99e9cf3e18 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -17718,7 +17718,8 @@ M: Olivier Moysan M: Arnaud Pouliquen L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/iio/adc/st,stm32-*.yaml +F: Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +F: Documentation/devicetree/bindings/sound/st,stm32-*.yaml F: sound/soc/stm/ STM32 TIMER/LPTIMER DRIVERS From 9c3ad33b5a412d8bc0a377e7cd9baa53ed52f22d Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 3 Sep 2021 18:30:02 +0800 Subject: [PATCH 04/45] ASoC: fsl_sai: register platform component before registering cpu dai There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. Fixes: 435508214942 ("ASoC: Add SAI SoC Digital Audio Interface driver") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1630665006-31437-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 223fcd15bfccc..38f6362099d58 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1152,11 +1152,10 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret < 0) goto err_pm_get_sync; - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, - &sai->cpu_dai_drv, 1); - if (ret) - goto err_pm_get_sync; - + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ if (sai->soc_data->use_imx_pcm) { ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE); if (ret) @@ -1167,6 +1166,11 @@ static int fsl_sai_probe(struct platform_device *pdev) goto err_pm_get_sync; } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + &sai->cpu_dai_drv, 1); + if (ret) + goto err_pm_get_sync; + return ret; err_pm_get_sync: From f12ce92e98b21c1fc669cd74e12c54a0fe3bc2eb Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 3 Sep 2021 18:30:03 +0800 Subject: [PATCH 05/45] ASoC: fsl_esai: register platform component before registering cpu dai There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1630665006-31437-3-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a961f837cd094..bda66b30e063c 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1073,6 +1073,16 @@ static int fsl_esai_probe(struct platform_device *pdev) if (ret < 0) goto err_pm_get_sync; + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); + if (ret) { + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); + goto err_pm_get_sync; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { @@ -1082,12 +1092,6 @@ static int fsl_esai_probe(struct platform_device *pdev) INIT_WORK(&esai_priv->work, fsl_esai_hw_reset); - ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); - if (ret) { - dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); - goto err_pm_get_sync; - } - return ret; err_pm_get_sync: From 0adf292069dcca8bab76a603251fcaabf77468ca Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 3 Sep 2021 18:30:04 +0800 Subject: [PATCH 06/45] ASoC: fsl_micfil: register platform component before registering cpu dai There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. Fixes: 47a70e6fc9a8 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1630665006-31437-4-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 8c0c75ce9490f..9f90989ac59a6 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -737,18 +737,23 @@ static int fsl_micfil_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); regcache_cache_only(micfil->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_micfil_component, &fsl_micfil_dai, 1); if (ret) { dev_err(&pdev->dev, "failed to register component %s\n", fsl_micfil_component.name); - return ret; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); - if (ret) - dev_err(&pdev->dev, "failed to pcm register\n"); - return ret; } From ee8ccc2eb5840e34fce088bdb174fd5329153ef0 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 3 Sep 2021 18:30:05 +0800 Subject: [PATCH 07/45] ASoC: fsl_spdif: register platform component before registering cpu dai There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. Fixes: a2388a498ad2 ("ASoC: fsl: Add S/PDIF CPU DAI driver") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1630665006-31437-5-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8ffb1a6048d63..1c53719bb61e2 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1434,16 +1434,20 @@ static int fsl_spdif_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); regcache_cache_only(spdif_priv->regmap, true); - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, - &spdif_priv->cpu_dai_drv, 1); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); if (ret) { - dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); goto err_pm_disable; } - ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); if (ret) { - dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); goto err_pm_disable; } From c590fa80b39287a91abeb487829f3190e7ae775f Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 3 Sep 2021 18:30:06 +0800 Subject: [PATCH 08/45] ASoC: fsl_xcvr: register platform component before registering cpu dai There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1630665006-31437-6-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 31c5ee641fe76..7ba2fd15132d9 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1215,18 +1215,23 @@ static int fsl_xcvr_probe(struct platform_device *pdev) pm_runtime_enable(dev); regcache_cache_only(xcvr->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); + if (ret) { + dev_err(dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); - return ret; } - ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); - if (ret) - dev_err(dev, "failed to pcm register\n"); - return ret; } From 1dd038522615b70f5f8945c5631e9e2fa5bd58b1 Mon Sep 17 00:00:00 2001 From: Trevor Wu Date: Fri, 10 Sep 2021 17:26:13 +0800 Subject: [PATCH 09/45] ASoC: mediatek: common: handle NULL case in suspend/resume function When memory allocation for afe->reg_back_up fails, reg_back_up can't be used. Keep the suspend/resume flow but skip register backup when afe->reg_back_up is NULL, in case illegal memory access happens. Fixes: 283b612429a2 ("ASoC: mediatek: implement mediatek common structure") Signed-off-by: Trevor Wu Reported-by: Dan Carpenter Link: https://lore.kernel.org/r/20210910092613.30188-1-trevor.wu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index baaa5881b1d48..e95c7c018e7d4 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component) devm_kcalloc(dev, afe->reg_back_up_list_num, sizeof(unsigned int), GFP_KERNEL); - for (i = 0; i < afe->reg_back_up_list_num; i++) - regmap_read(regmap, afe->reg_back_up_list[i], - &afe->reg_back_up[i]); + if (afe->reg_back_up) { + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_read(regmap, afe->reg_back_up_list[i], + &afe->reg_back_up[i]); + } afe->suspended = true; afe->runtime_suspend(dev); @@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component) afe->runtime_resume(dev); - if (!afe->reg_back_up) + if (!afe->reg_back_up) { dev_dbg(dev, "%s no reg_backup\n", __func__); - - for (i = 0; i < afe->reg_back_up_list_num; i++) - mtk_regmap_write(regmap, afe->reg_back_up_list[i], - afe->reg_back_up[i]); + } else { + for (i = 0; i < afe->reg_back_up_list_num; i++) + mtk_regmap_write(regmap, afe->reg_back_up_list[i], + afe->reg_back_up[i]); + } afe->suspended = false; return 0; From ac4dfccb96571ca03af7cac64b7a0b2952c97f3a Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Wed, 15 Sep 2021 09:32:30 +0300 Subject: [PATCH 10/45] ASoC: SOF: Fix DSP oops stack dump output contents Fix @buf arg given to hex_dump_to_buffer() and stack address used in dump error output. Fixes: e657c18a01c8 ('ASoC: SOF: Add xtensa support') Signed-off-by: Yong Zhi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Daniel Baluta Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20210915063230.29711-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/xtensa/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c index bbb9a2282ed9e..f6e3411b33cf1 100644 --- a/sound/soc/sof/xtensa/core.c +++ b/sound/soc/sof/xtensa/core.c @@ -122,9 +122,9 @@ static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, * 0x0049fbb0: 8000f2d0 0049fc00 6f6c6c61 00632e63 */ for (i = 0; i < stack_words; i += 4) { - hex_dump_to_buffer(stack + i * 4, 16, 16, 4, + hex_dump_to_buffer(stack + i, 16, 16, 4, buf, sizeof(buf), false); - dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i, buf); + dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i * 4, buf); } } From 10d93a98190aec2c3ff98d9472ab1bf0543aa02c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 15 Sep 2021 15:21:08 +0300 Subject: [PATCH 11/45] ASoC: SOF: imx: imx8: Bar index is only valid for IRAM and SRAM types i.MX8 only uses SOF_FW_BLK_TYPE_IRAM (1) and SOF_FW_BLK_TYPE_SRAM (3) bars, everything else is left as 0 in sdev->bar[] array. If a broken or purposefully crafted firmware image is loaded with other types of FW_BLK_TYPE then a kernel crash can be triggered. Make sure that only IRAM/SRAM type is converted to bar index. Fixes: 202acc565a1f0 ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Peter Ujfalusi Reviewed-by: Daniel Baluta Reviewed-by: Ranjani Sridharan Reviewed-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20210915122116.18317-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 12fedf0984bd9..7e9723a10d02e 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -365,7 +365,14 @@ static int imx8_remove(struct snd_sof_dev *sdev) /* on i.MX8 there is 1 to 1 match between type and BAR idx */ static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type) { - return type; + /* Only IRAM and SRAM bars are valid */ + switch (type) { + case SOF_FW_BLK_TYPE_IRAM: + case SOF_FW_BLK_TYPE_SRAM: + return type; + default: + return -EINVAL; + } } static void imx8_ipc_msg_data(struct snd_sof_dev *sdev, From d9be4a88c3627c270bbe032b623dc43f3b764565 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 15 Sep 2021 15:21:09 +0300 Subject: [PATCH 12/45] ASoC: SOF: imx: imx8m: Bar index is only valid for IRAM and SRAM types i.MX8 only uses SOF_FW_BLK_TYPE_IRAM (1) and SOF_FW_BLK_TYPE_SRAM (3) bars, everything else is left as 0 in sdev->bar[] array. If a broken or purposefully crafted firmware image is loaded with other types of FW_BLK_TYPE then a kernel crash can be triggered. Make sure that only IRAM/SRAM type is converted to bar index. Fixes: afb93d716533d ("ASoC: SOF: imx: Add i.MX8M HW support") Signed-off-by: Peter Ujfalusi Reviewed-by: Daniel Baluta Reviewed-by: Ranjani Sridharan Reviewed-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20210915122116.18317-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8m.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index cb822d9537678..892e1482f97fa 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -228,7 +228,14 @@ static int imx8m_remove(struct snd_sof_dev *sdev) /* on i.MX8 there is 1 to 1 match between type and BAR idx */ static int imx8m_get_bar_index(struct snd_sof_dev *sdev, u32 type) { - return type; + /* Only IRAM and SRAM bars are valid */ + switch (type) { + case SOF_FW_BLK_TYPE_IRAM: + case SOF_FW_BLK_TYPE_SRAM: + return type; + default: + return -EINVAL; + } } static void imx8m_ipc_msg_data(struct snd_sof_dev *sdev, From 8a8e1813ffc35111fc0b6db49968ceb0e1615ced Mon Sep 17 00:00:00 2001 From: Marc Herbert Date: Thu, 16 Sep 2021 11:50:08 +0300 Subject: [PATCH 13/45] ASoC: SOF: loader: release_firmware() on load failure to avoid batching Invoke release_firmware() when the firmware fails to boot in sof_probe_continue(). The request_firmware() framework must be informed of failures in sof_probe_continue() otherwise its internal "batching" feature (different from caching) cached the firmware image forever. Attempts to correct the file in /lib/firmware/ were then silently and confusingly ignored until the next reboot. Unloading the drivers did not help because from their disconnected perspective the firmware had failed so there was nothing to release. Also leverage the new snd_sof_fw_unload() function to simplify the snd_sof_device_remove() function. Signed-off-by: Marc Herbert Reviewed-by: Pierre-Louis Bossart Reviewed-by: Guennadi Liakhovetski Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20210916085008.28929-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 4 +--- sound/soc/sof/loader.c | 2 ++ 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 3e4dd4a86363b..59d0d7b2b55c8 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -371,7 +371,6 @@ int snd_sof_device_remove(struct device *dev) dev_warn(dev, "error: %d failed to prepare DSP for device removal", ret); - snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_free_trace(sdev); @@ -394,8 +393,7 @@ int snd_sof_device_remove(struct device *dev) snd_sof_remove(sdev); /* release firmware */ - release_firmware(pdata->fw); - pdata->fw = NULL; + snd_sof_fw_unload(sdev); return 0; } diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594f..9c3f251a0dd05 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware); void snd_sof_fw_unload(struct snd_sof_dev *sdev) { /* TODO: support module unloading at runtime */ + release_firmware(sdev->pdata->fw); + sdev->pdata->fw = NULL; } EXPORT_SYMBOL(snd_sof_fw_unload); From 25766ee44ff8db4cdf8471b587dffb28b7b9d17f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Sep 2021 11:53:42 +0300 Subject: [PATCH 14/45] ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication In case the firmware is missing we will have the following in the kernel log: 1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2 2 | error: request firmware intel/sof/sof-tgl-h.ri failed err: -2 3 | you may need to download the firmware from https://github.com/thesofproject/sof-bin/ 4 | error: failed to load DSP firmware -2 5 | error: sof_probe_work failed err: -2 The first line is the standard, request_firmware() warning. The second and third line is printed in snd_sof_load_firmware_raw() Note that the first and second line is mostly identical. With this patch the log will be changed to: 1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2 2 | error: sof firmware file is missing, you might need to 3 | download it from https://github.com/thesofproject/sof-bin/ 4 | error: failed to load DSP firmware -2 5 | error: sof_probe_work failed err: -2 Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210916085342.29993-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/loader.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 9c3f251a0dd05..bb79c77775b3d 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -729,10 +729,10 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev); if (ret < 0) { - dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", - fw_filename, ret); dev_err(sdev->dev, - "you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n"); + "error: sof firmware file is missing, you might need to\n"); + dev_err(sdev->dev, + " download it from https://github.com/thesofproject/sof-bin/\n"); goto err; } else { dev_dbg(sdev->dev, "request_firmware %s successful\n", From 3abe2eec87059260bf31033a8863c67c5d45b9d0 Mon Sep 17 00:00:00 2001 From: Trevor Wu Date: Fri, 17 Sep 2021 16:28:05 +0800 Subject: [PATCH 15/45] ASoC: mediatek: mt8195: remove wrong fixup assignment on HDMITX S24_LE params fixup is only required for DPTX. Remove fixup ops assignment for HDMITX. Fixes: 40d605df0a7b ("ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682") Signed-off-by: Trevor Wu Link: https://lore.kernel.org/r/20210917082805.30898-1-trevor.wu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index c97ace7387b4c..de09f67c04502 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -424,8 +424,8 @@ static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL); } -static int mt8195_hdmitx_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { /* fix BE i2s format to 32bit, clean param mask first */ @@ -902,7 +902,7 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ops = &mt8195_dptx_ops, - .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup, + .be_hw_params_fixup = mt8195_dptx_hw_params_fixup, SND_SOC_DAILINK_REG(DPTX_BE), }, [DAI_LINK_ETDM1_IN_BE] = { @@ -953,7 +953,6 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_playback = 1, - .be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup, SND_SOC_DAILINK_REG(ETDM3_OUT_BE), }, [DAI_LINK_PCM1_BE] = { From cfacfefd382af3b42905108b54f02820dca225c4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 17 Sep 2021 11:51:08 +0300 Subject: [PATCH 16/45] ASoC: SOF: trace: Omit error print when waking up trace sleepers Do not print error message from snd_sof_trace_notify_for_error() when possible sleeping trace work is woken up to flush the remaining debug information. This action by itself is not an error, it is just an action we take when an error occurs to make sure that all information have been fed to the userspace (if we have trace in use). Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210917085108.25532-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/trace.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index f72a6e83e6af2..58f6ca5cf491a 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -530,7 +530,6 @@ void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev) return; if (sdev->dtrace_is_enabled) { - dev_err(sdev->dev, "error: waking up any trace sleepers\n"); sdev->dtrace_error = true; wake_up(&sdev->trace_sleep); } From 3f4b57ad07d9237acf1b8cff3f8bf530cacef87a Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 20 Sep 2021 16:49:39 +0200 Subject: [PATCH 17/45] ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers Commit 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") breaks the TSE-850 device, which is using a pcm5142 in I2S and CBM_CFS mode (maybe not relevant). Without this fix, the result is: pcm512x 0-004c: Failed to set data format: -16 And after that, no sound. This fix is not 100% correct. The datasheet of at least the pcm5142 states that four bits (0xcc) in the I2S_1 register are "RSV" ("Reserved. Do not access.") and no hint is given as to what the initial values are supposed to be. So, specifying defaults for these bits is wrong. But perhaps better than a broken driver? Fixes: 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Kirill Marinushkin Cc: Peter Ujfalusi Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Peter Rosin Signed-off-by: Peter Ujfalusi Reviewed-by: Peter Ujfalusi Link: https://lore.kernel.org/r/2d221984-7a2e-7006-0f8a-ffb5f64ee885@axentia.se Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4dc844f3c1fc0..60dee41816dc2 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -116,6 +116,8 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_FS_SPEED_MODE, 0x00 }, { PCM512x_IDAC_1, 0x01 }, { PCM512x_IDAC_2, 0x00 }, + { PCM512x_I2S_1, 0x02 }, + { PCM512x_I2S_2, 0x00 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) From 74b7ee0e7b61838a0a161a84d105aeff0d042646 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 10 Sep 2021 17:18:30 +0800 Subject: [PATCH 18/45] ASoC: fsl_xcvr: Fix channel swap issue with ARC With pause and resume test for ARC, there is occasionally channel swap issue. The reason is that currently driver set the DPATH out of reset first, then start the DMA, the first data got from FIFO may not be the Left channel. Moving DPATH out of reset operation after the dma enablement to fix this issue. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1631265510-27384-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 7ba2fd15132d9..d0556c79fdb15 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -487,8 +487,9 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, return ret; } - /* clear DPATH RESET */ + /* set DPATH RESET */ m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); + v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -590,10 +591,6 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, val |= FSL_XCVR_EXT_CTRL_CMDC_RESET(tx); } - /* set DPATH RESET */ - mask |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - val |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, mask, val); if (ret < 0) { dev_err(dai->dev, "Err setting DPATH RESET: %d\n", ret); @@ -643,6 +640,16 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, dev_err(dai->dev, "Failed to enable DMA: %d\n", ret); return ret; } + + /* clear DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear DPATH RESET: %d\n", ret); + return ret; + } + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: From ceef3240f9b7e592dd8d10d619c312c7336117fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Sep 2021 20:49:56 +0100 Subject: [PATCH 19/45] ASoC: pcm179x: Add missing entries SPI to device ID table Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20210924194956.46079-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/pcm179x-spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/pcm179x-spi.c b/sound/soc/codecs/pcm179x-spi.c index 0a542924ec5f9..ebf63ea90a1c4 100644 --- a/sound/soc/codecs/pcm179x-spi.c +++ b/sound/soc/codecs/pcm179x-spi.c @@ -36,6 +36,7 @@ static const struct of_device_id pcm179x_of_match[] = { MODULE_DEVICE_TABLE(of, pcm179x_of_match); static const struct spi_device_id pcm179x_spi_ids[] = { + { "pcm1792a", 0 }, { "pcm179x", 0 }, { }, }; From 0cc3687eadd0971d5d38ff90d14819d88f854960 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Sep 2021 20:48:44 +0100 Subject: [PATCH 20/45] ASoC: cs4341: Add SPI device ID table Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown Cc: patches@opensource.cirrus.com Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20210924194844.45974-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs4341.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 7d3e54d8eef36..29d05e32d3417 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -305,12 +305,19 @@ static int cs4341_spi_probe(struct spi_device *spi) return cs4341_probe(&spi->dev); } +static const struct spi_device_id cs4341_spi_ids[] = { + { "cs4341a" }, + { } +}; +MODULE_DEVICE_TABLE(spi, cs4341_spi_ids); + static struct spi_driver cs4341_spi_driver = { .driver = { .name = "cs4341-spi", .of_match_table = of_match_ptr(cs4341_dt_ids), }, .probe = cs4341_spi_probe, + .id_table = cs4341_spi_ids, }; #endif From 42871e95a3afea8956d8cc567ea725b33a837775 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 29 Sep 2021 22:15:12 +0200 Subject: [PATCH 21/45] ASoC: nau8824: Fix headphone vs headset, button-press detection no longer working Commit 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") replaced the nau8824_dapm_enable_pin() helper with direct calls to snd_soc_dapm_enable_pin(), but the helper was using snd_soc_dapm_force_enable_pin() and not forcing the MICBIAS + SAR supplies on breaks headphone vs headset and button-press detection. Replace the snd_soc_dapm_enable_pin() calls with snd_soc_dapm_force_enable_pin() to fix this. Cc: stable@vger.kernel.org Fixes: 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210929201512.460360-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index db88be48c9980..f946ef65a4c19 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -867,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_enable_pin(dapm, "SAR"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); From db0767b8a6e620b99459d2e688c1983c2e5add0d Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 7 Oct 2021 19:20:19 +0530 Subject: [PATCH 22/45] ASoC: wcd938x: Fix jack detection issue This patch is to fix audio 3.5mm jack detection failure on wcd938x codec based target. Fixes: bcee7ed09b8e (ASoC: codecs: wcd938x: add Multi Button Headset Control support) Signed-off-by: Venkata Prasad Potturu Signed-off-by: Srinivasa Rao Mandadapu Reviewed-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/1633614619-27026-1-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index f0daf8defcf1e..52de7d14b1398 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -4144,10 +4144,10 @@ static int wcd938x_codec_set_jack(struct snd_soc_component *comp, { struct wcd938x_priv *wcd = dev_get_drvdata(comp->dev); - if (!jack) + if (jack) return wcd_mbhc_start(wcd->wcd_mbhc, &wcd->mbhc_cfg, jack); - - wcd_mbhc_stop(wcd->wcd_mbhc); + else + wcd_mbhc_stop(wcd->wcd_mbhc); return 0; } From 2577b868a48ef3601116908738efbe570451e605 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:25 +0300 Subject: [PATCH 23/45] ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() Access to platform data via dev_get_platdata() getter to make code cleaner. Signed-off-by: Andy Shevchenko Acked-by: Pierre-Louis Bossart Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211006150428.16434-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 055248f104b24..b1f9c9cb33555 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -456,12 +456,12 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; static const char * const mic_name[] = { "in1", "in2" }; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; const struct dmi_system_id *dmi_id; - struct device *dev = &pdev->dev; - struct snd_soc_acpi_mach *mach; struct fwnode_handle *fwnode; const char *platform_name; struct acpi_device *adev; @@ -476,7 +476,6 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - mach = dev->platform_data; /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { if (!strcmp(byt_cht_es8316_dais[i].codecs->name, From 6f32c521061b704c0198be3ba9834f5a64ea5605 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:26 +0300 Subject: [PATCH 24/45] ASoC: Intel: bytcht_es8316: Use temporary variable for struct device Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20211006150428.16434-2-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index b1f9c9cb33555..efd71e6e42b33 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -493,7 +493,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) put_device(&adev->dev); byt_cht_es8316_dais[dai_index].codecs->name = codec_name; } else { - dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + dev_err(dev, "Error cannot find '%s' dev\n", mach->id); return -ENXIO; } @@ -596,7 +596,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_card.long_name = long_name; #endif - sof_parent = snd_soc_acpi_sof_parent(&pdev->dev); + sof_parent = snd_soc_acpi_sof_parent(dev); /* set card and driver name */ if (sof_parent) { From 10f4a96543b744c8cc7ef8b0799af21d911dd37d Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:27 +0300 Subject: [PATCH 25/45] ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() First of all, replace indexed API by plain one since we have index 0. Second, switch to optional variant and drop duplicated code. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20211006150428.16434-3-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index efd71e6e42b33..421a04d96d841 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -566,16 +566,12 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); priv->speaker_en_gpio = - gpiod_get_index(codec_dev, "speaker-enable", 0, - /* see comment in byt_cht_es8316_resume */ - GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); - + gpiod_get_optional(codec_dev, "speaker-enable", + /* see comment in byt_cht_es8316_resume() */ + GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); if (IS_ERR(priv->speaker_en_gpio)) { ret = PTR_ERR(priv->speaker_en_gpio); switch (ret) { - case -ENOENT: - priv->speaker_en_gpio = NULL; - break; default: dev_err(dev, "get speaker GPIO failed: %d\n", ret); fallthrough; From c25d4546ca452b2e8c03bc735e4c65bc6dd751dd Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:28 +0300 Subject: [PATCH 26/45] ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation dev_err_probe() avoids printing into log when the deferred probe is invoked. This is possible when clock provider is pending to appear. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20211006150428.16434-4-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 421a04d96d841..4d313d0d0f23e 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -532,11 +532,8 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* get the clock */ priv->mclk = devm_clk_get(dev, "pmc_plt_clk_3"); - if (IS_ERR(priv->mclk)) { - ret = PTR_ERR(priv->mclk); - dev_err(dev, "clk_get pmc_plt_clk_3 failed: %d\n", ret); - return ret; - } + if (IS_ERR(priv->mclk)) + return dev_err_probe(dev, PTR_ERR(priv->mclk), "clk_get pmc_plt_clk_3 failed\n"); /* get speaker enable GPIO */ codec_dev = acpi_get_first_physical_node(adev); @@ -570,14 +567,9 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* see comment in byt_cht_es8316_resume() */ GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); if (IS_ERR(priv->speaker_en_gpio)) { - ret = PTR_ERR(priv->speaker_en_gpio); - switch (ret) { - default: - dev_err(dev, "get speaker GPIO failed: %d\n", ret); - fallthrough; - case -EPROBE_DEFER: - goto err_put_codec; - } + ret = dev_err_probe(dev, PTR_ERR(priv->speaker_en_gpio), + "get speaker GPIO failed\n"); + goto err_put_codec; } snprintf(components_string, sizeof(components_string), From 5af82c81b2c49cfb1cad84d9eb6eab0e3d1c4842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Oct 2021 16:17:12 +0200 Subject: [PATCH 27/45] ASoC: DAPM: Fix missing kctl change notifications The put callback of a kcontrol is supposed to return 1 when the value is changed, and this will be notified to user-space. However, some DAPM kcontrols always return 0 (except for errors), hence the user-space misses the update of a control value. This patch corrects the behavior by properly returning 1 when the value gets updated. Reported-and-tested-by: Hans de Goede Cc: Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20211006141712.2439-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7b67f1e19ae95..59d07648a7e7f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2561,6 +2561,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + int ret = 0; dapm_assert_locked(dapm); @@ -2573,13 +2574,14 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(w, "pin configuration"); dapm_widget_invalidate_input_paths(w); dapm_widget_invalidate_output_paths(w); + ret = 1; } w->connected = status; if (status == 0) w->force = 0; - return 0; + return ret; } /** @@ -3583,14 +3585,15 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; + int ret; if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&card->dapm, pin); + ret = snd_soc_dapm_enable_pin(&card->dapm, pin); else - snd_soc_dapm_disable_pin(&card->dapm, pin); + ret = snd_soc_dapm_disable_pin(&card->dapm, pin); snd_soc_dapm_sync(&card->dapm); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -4023,7 +4026,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, rtd->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } static void From 214174d9f56c7f81f4860a26b6b8b961a6b92654 Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 7 Oct 2021 19:21:15 +0530 Subject: [PATCH 28/45] ASoC: codec: wcd938x: Add irq config support This patch fixes compilation error in wcd98x codec driver. Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Signed-off-by: Venkata Prasad Potturu Signed-off-by: Srinivasa Rao Mandadapu Reviewed-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/1633614675-27122-1-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d0..216cea04ad704 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1583,6 +1583,7 @@ config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" select SND_SOC_WCD938X select SND_SOC_WCD_MBHC + select REGMAP_IRQ depends on SOUNDWIRE select REGMAP_SOUNDWIRE help From c448b7aa3e66042fc0f849d9a0fb90d1af82e948 Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Sat, 9 Oct 2021 14:58:40 +0800 Subject: [PATCH 29/45] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked() 'component' is allocated in snd_soc_register_component(), but component->list is not initalized, this may cause snd_soc_del_component_unlocked() deref null ptr in the error handing case. KASAN: null-ptr-deref in range [0x0000000000000000-0x0000000000000007] RIP: 0010:__list_del_entry_valid+0x81/0xf0 Call Trace: snd_soc_del_component_unlocked+0x69/0x1b0 [snd_soc_core] snd_soc_add_component.cold+0x54/0x6c [snd_soc_core] snd_soc_register_component+0x70/0x90 [snd_soc_core] devm_snd_soc_register_component+0x5e/0xd0 [snd_soc_core] tas2552_probe+0x265/0x320 [snd_soc_tas2552] ? tas2552_component_probe+0x1e0/0x1e0 [snd_soc_tas2552] i2c_device_probe+0xa31/0xbe0 Fix by adding INIT_LIST_HEAD() to snd_soc_component_initialize(). Reported-by: Hulk Robot Signed-off-by: Yang Yingliang Link: https://lore.kernel.org/r/20211009065840.3196239-1-yangyingliang@huawei.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c830e96afba24..80ca260595fda 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,6 +2599,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->dai_list); INIT_LIST_HEAD(&component->dobj_list); INIT_LIST_HEAD(&component->card_list); + INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); component->name = fmt_single_name(dev, &component->id); From aa18457c4af7a9dad1f2b150b11beae1d8ab57aa Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 11 Oct 2021 15:49:03 +0100 Subject: [PATCH 30/45] ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets Ensure the default 0dB playback path is always used. The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was spurious, and resulted in a -6dB attenuation being accidentally inserted into the playback path. Signed-off-by: Stefan Binding Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211011144903.28915-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index fb1e4c33e27d3..9a463ab54bddc 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -922,7 +922,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +992,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } From 6b9b546dc00797c74bef491668ce5431ff54e1e2 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 13 Oct 2021 13:17:04 +0800 Subject: [PATCH 31/45] ASoC: wm8960: Fix clock configuration on slave mode There is a noise issue for 8kHz sample rate on slave mode. Compared with master mode, the difference is the DACDIV setting, after correcting the DACDIV, the noise is gone. There is no noise issue for 48kHz sample rate, because the default value of DACDIV is correct for 48kHz. So wm8960_configure_clocking() should be functional for ADC and DAC function even if it is slave mode. In order to be compatible for old use case, just add condition for checking that sysclk is zero with slave mode. Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock") Signed-off-by: Shengjiu Wang Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1634102224-3922-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392c..499604f1e1789 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,9 +742,16 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { - dev_dbg(component->dev, - "Codec is slave mode, no need to configure clock\n"); + /* + * For Slave mode clocking should still be configured, + * so this if statement should be removed, but some platform + * may not work if the sysclk is not configured, to avoid such + * compatible issue, just add '!wm8960->sysclk' condition in + * this if statement. + */ + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { + dev_warn(component->dev, + "slave mode, but proceeding with no clock configuration\n"); return 0; } From 06441c82f0cd836402ca5fa4162d28ed07cfb0ed Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:04 +0100 Subject: [PATCH 32/45] ASoC: cs42l42: Don't reconfigure the PLL while it is running When capture and playback substreams are both running at the same time, cs42l42_pcm_hw_params() would be called for each direction. The first call will configure the PLL. The second call must not write the PLL configuration registers again if the first substream is already running, as this could destabilize the PLL. The DAI is marked symmetric sample bits and sample rate, so the two directions will always have the same SCLK (I2S always has 2 channel slots so the DAI does not need to require symmetric channels to guarantee the same SCLK). However, since cs42l42_pll_config() is checking for an active stream it may as well test that the requested SCLK is the same as the currently active configuration. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 9a463ab54bddc..cdcb6d81d9002 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -618,6 +618,14 @@ static int cs42l42_pll_config(struct snd_soc_component *component) else clk = cs42l42->sclk; + /* Don't reconfigure if there is an audio stream running */ + if (cs42l42->stream_use) { + if (pll_ratio_table[cs42l42->pll_config].sclk == clk) + return 0; + else + return -EBUSY; + } + for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { if (pll_ratio_table[i].sclk == clk) { cs42l42->pll_config = i; From 6e6825801ab926360f7f4f2dbcfd107d5ab8f025 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:05 +0100 Subject: [PATCH 33/45] ASoC: cs42l42: Always configure both ASP TX channels An I2S frame always has two slots (left and right) even when sending mono. The right channel (channel 2) of ASP TX will always have the same bit width as the left channel and will always be on the high phase of LRCLK. The previous implementation always passed the field masks for both channels to snd_soc_component_update_bits() but for mono the written value only contained the settings for channel 1. The result was that for mono channel 2 was set to 8-bit (which is an invalid configuration) with both channels on the low phase of LRCLK. Signed-off-by: Richard Fitzgerald Fixes: 585e7079de0e ("ASoC: cs42l42: Add Capture Support") Link: https://lore.kernel.org/r/20211015133619.4698-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index cdcb6d81d9002..914257180c016 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -861,11 +861,10 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, switch(substream->stream) { case SNDRV_PCM_STREAM_CAPTURE: - if (channels == 2) { - val |= CS42L42_ASP_TX_CH2_AP_MASK; - val |= width << CS42L42_ASP_TX_CH2_RES_SHIFT; - } - val |= width << CS42L42_ASP_TX_CH1_RES_SHIFT; + /* channel 2 on high LRCLK */ + val = CS42L42_ASP_TX_CH2_AP_MASK | + (width << CS42L42_ASP_TX_CH2_RES_SHIFT) | + (width << CS42L42_ASP_TX_CH1_RES_SHIFT); snd_soc_component_update_bits(component, CS42L42_ASP_TX_CH_AP_RES, CS42L42_ASP_TX_CH1_AP_MASK | CS42L42_ASP_TX_CH2_AP_MASK | From d591d4b32aa9552af14a0c7c586a2d3fe9ecc6e0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:06 +0100 Subject: [PATCH 34/45] ASoC: cs42l42: Correct some register default values Some registers had wrong default values in cs42l42_reg_defaults[]. Signed-off-by: Richard Fitzgerald Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Link: https://lore.kernel.org/r/20211015133619.4698-4-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 914257180c016..a45026809aa48 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -93,7 +93,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_ASP_RX_INT_MASK, 0x1F }, { CS42L42_ASP_TX_INT_MASK, 0x0F }, { CS42L42_CODEC_INT_MASK, 0x03 }, - { CS42L42_SRCPL_INT_MASK, 0xFF }, + { CS42L42_SRCPL_INT_MASK, 0x7F }, { CS42L42_VPMON_INT_MASK, 0x01 }, { CS42L42_PLL_LOCK_INT_MASK, 0x01 }, { CS42L42_TSRS_PLUG_INT_MASK, 0x0F }, @@ -130,7 +130,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_MIXER_CHA_VOL, 0x3F }, { CS42L42_MIXER_ADC_VOL, 0x3F }, { CS42L42_MIXER_CHB_VOL, 0x3F }, - { CS42L42_EQ_COEF_IN0, 0x22 }, + { CS42L42_EQ_COEF_IN0, 0x00 }, { CS42L42_EQ_COEF_IN1, 0x00 }, { CS42L42_EQ_COEF_IN2, 0x00 }, { CS42L42_EQ_COEF_IN3, 0x00 }, From 917d5758014b37cf97b946dd130aad9353c354dc Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:07 +0100 Subject: [PATCH 35/45] ASoC: cs42l42: Don't set defaults for volatile registers Volatile registers don't need a default value. Signed-off-by: Richard Fitzgerald Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Link: https://lore.kernel.org/r/20211015133619.4698-5-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 20 -------------------- 1 file changed, 20 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a45026809aa48..e5a97e904d40c 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -41,7 +41,6 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_FRZ_CTL, 0x00 }, { CS42L42_SRC_CTL, 0x10 }, - { CS42L42_MCLK_STATUS, 0x02 }, { CS42L42_MCLK_CTL, 0x02 }, { CS42L42_SFTRAMP_RATE, 0xA4 }, { CS42L42_I2C_DEBOUNCE, 0x88 }, @@ -57,11 +56,9 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_RSENSE_CTL3, 0x1B }, { CS42L42_TSENSE_CTL, 0x1B }, { CS42L42_TSRS_INT_DISABLE, 0x00 }, - { CS42L42_TRSENSE_STATUS, 0x00 }, { CS42L42_HSDET_CTL1, 0x77 }, { CS42L42_HSDET_CTL2, 0x00 }, { CS42L42_HS_SWITCH_CTL, 0xF3 }, - { CS42L42_HS_DET_STATUS, 0x00 }, { CS42L42_HS_CLAMP_DISABLE, 0x00 }, { CS42L42_MCLK_SRC_SEL, 0x00 }, { CS42L42_SPDIF_CLK_CFG, 0x00 }, @@ -75,18 +72,6 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_IN_ASRC_CLK, 0x00 }, { CS42L42_OUT_ASRC_CLK, 0x00 }, { CS42L42_PLL_DIV_CFG1, 0x00 }, - { CS42L42_ADC_OVFL_STATUS, 0x00 }, - { CS42L42_MIXER_STATUS, 0x00 }, - { CS42L42_SRC_STATUS, 0x00 }, - { CS42L42_ASP_RX_STATUS, 0x00 }, - { CS42L42_ASP_TX_STATUS, 0x00 }, - { CS42L42_CODEC_STATUS, 0x00 }, - { CS42L42_DET_INT_STATUS1, 0x00 }, - { CS42L42_DET_INT_STATUS2, 0x00 }, - { CS42L42_SRCPL_INT_STATUS, 0x00 }, - { CS42L42_VPMON_STATUS, 0x00 }, - { CS42L42_PLL_LOCK_STATUS, 0x00 }, - { CS42L42_TSRS_PLUG_STATUS, 0x00 }, { CS42L42_ADC_OVFL_INT_MASK, 0x01 }, { CS42L42_MIXER_INT_MASK, 0x0F }, { CS42L42_SRC_INT_MASK, 0x0F }, @@ -105,8 +90,6 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_PLL_CTL3, 0x10 }, { CS42L42_PLL_CAL_RATIO, 0x80 }, { CS42L42_PLL_CTL4, 0x03 }, - { CS42L42_LOAD_DET_RCSTAT, 0x00 }, - { CS42L42_LOAD_DET_DONE, 0x00 }, { CS42L42_LOAD_DET_EN, 0x00 }, { CS42L42_HSBIAS_SC_AUTOCTL, 0x03 }, { CS42L42_WAKE_CTL, 0xC0 }, @@ -115,8 +98,6 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_MISC_DET_CTL, 0x03 }, { CS42L42_MIC_DET_CTL1, 0x1F }, { CS42L42_MIC_DET_CTL2, 0x2F }, - { CS42L42_DET_STATUS1, 0x00 }, - { CS42L42_DET_STATUS2, 0x00 }, { CS42L42_DET_INT1_MASK, 0xE0 }, { CS42L42_DET_INT2_MASK, 0xFF }, { CS42L42_HS_BIAS_CTL, 0xC2 }, @@ -182,7 +163,6 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_ASP_RX_DAI1_CH2_AP_RES, 0x03 }, { CS42L42_ASP_RX_DAI1_CH2_BIT_MSB, 0x00 }, { CS42L42_ASP_RX_DAI1_CH2_BIT_LSB, 0x00 }, - { CS42L42_SUB_REVID, 0x03 }, }; static bool cs42l42_readable_register(struct device *dev, unsigned int reg) From 0306988789d9d91a18ff70bd2bf165d3ae0ef1dd Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:08 +0100 Subject: [PATCH 36/45] ASoC: cs42l42: Defer probe if request_threaded_irq() returns EPROBE_DEFER The driver can run without an interrupt so if devm_request_threaded_irq() failed, the probe() just carried on. But if this was EPROBE_DEFER the driver would continue without an interrupt instead of deferring to wait for the interrupt to become available. Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-6-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index e5a97e904d40c..dc84b3de3da31 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1935,8 +1935,9 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, NULL, cs42l42_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs42l42", cs42l42); - - if (ret != 0) + if (ret == -EPROBE_DEFER) + goto err_disable; + else if (ret != 0) dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); From 2a031a99428bafba089437e9044b8fd5dc6e7551 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:13 +0100 Subject: [PATCH 37/45] ASoC: cs42l42: Don't claim to support 192k The driver currently only supports configuring for sample rates <= 96k and it isn't possible to setup a configuration that will support all sample rates up to 192k. For sample rates up to 96k MCLK is in the 12MHz group. However, although 192k only requires an I2S clock in the 12MHz group, the cs42l42 audio path is not natively 192k so the audio must be resampled. But for 192k the SRC requires a 24MHz MCLK. It is not possible to switch MCLK between 12MHz and 24MHz groups on-the-fly. The 12MHz group supports all sample rates up to 96k. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-11-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index dc84b3de3da31..71390c20e7a1e 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -812,7 +812,7 @@ static int cs42l42_dai_startup(struct snd_pcm_substream *substream, struct snd_s /* Machine driver has not set a SCLK, limit bottom end to 44.1 kHz */ return snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - 44100, 192000); + 44100, 96000); } static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, @@ -1008,14 +1008,14 @@ static struct snd_soc_dai_driver cs42l42_dai = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_8000_96000, .formats = CS42L42_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_8000_96000, .formats = CS42L42_FORMATS, }, .symmetric_rate = 1, From 3c211cb7db2905221f9f006aa66b8af17bfcd480 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:14 +0100 Subject: [PATCH 38/45] ASoC: cs42l42: Use PLL for SCLK > 12.288MHz It isn't possible to switch MCLK between 12MHz and 24MHz rate groups on-the-fly - this can only be done when cs42l42 is powered-down. All "normal" SCLK rates use an MCLK in the 12MHz group, so change the configs for SCLK > 12.288 MHz to use the PLL to generate an MCLK in the 12MHz group. As this means MCLK_DIV is always 0 it can be removed from the pll configuration setup. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-12-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 41 +++++++++++++++++--------------------- 1 file changed, 18 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 71390c20e7a1e..4ec7fdcedca7d 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -549,7 +549,6 @@ static const struct reg_sequence cs42l42_to_osc_seq[] = { struct cs42l42_pll_params { u32 sclk; - u8 mclk_div; u8 mclk_src_sel; u8 sclk_prediv; u8 pll_div_int; @@ -566,24 +565,24 @@ struct cs42l42_pll_params { * Table 4-5 from the Datasheet */ static const struct cs42l42_pll_params pll_ratio_table[] = { - { 1411200, 0, 1, 0x00, 0x80, 0x000000, 0x03, 0x10, 11289600, 128, 2}, - { 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125, 2}, - { 2304000, 0, 1, 0x00, 0x55, 0xC00000, 0x02, 0x10, 12288000, 85, 2}, - { 2400000, 0, 1, 0x00, 0x50, 0x000000, 0x03, 0x10, 12000000, 80, 2}, - { 2822400, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, - { 3000000, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, - { 3072000, 0, 1, 0x00, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, - { 4000000, 0, 1, 0x00, 0x30, 0x800000, 0x03, 0x10, 12000000, 96, 1}, - { 4096000, 0, 1, 0x00, 0x2E, 0xE00000, 0x03, 0x10, 12000000, 94, 1}, - { 5644800, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, - { 6000000, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, - { 6144000, 0, 1, 0x01, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, - { 11289600, 0, 0, 0, 0, 0, 0, 0, 11289600, 0, 1}, - { 12000000, 0, 0, 0, 0, 0, 0, 0, 12000000, 0, 1}, - { 12288000, 0, 0, 0, 0, 0, 0, 0, 12288000, 0, 1}, - { 22579200, 1, 0, 0, 0, 0, 0, 0, 22579200, 0, 1}, - { 24000000, 1, 0, 0, 0, 0, 0, 0, 24000000, 0, 1}, - { 24576000, 1, 0, 0, 0, 0, 0, 0, 24576000, 0, 1} + { 1411200, 1, 0x00, 0x80, 0x000000, 0x03, 0x10, 11289600, 128, 2}, + { 1536000, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125, 2}, + { 2304000, 1, 0x00, 0x55, 0xC00000, 0x02, 0x10, 12288000, 85, 2}, + { 2400000, 1, 0x00, 0x50, 0x000000, 0x03, 0x10, 12000000, 80, 2}, + { 2822400, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, + { 3000000, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, + { 3072000, 1, 0x00, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, + { 4000000, 1, 0x00, 0x30, 0x800000, 0x03, 0x10, 12000000, 96, 1}, + { 4096000, 1, 0x00, 0x2E, 0xE00000, 0x03, 0x10, 12000000, 94, 1}, + { 5644800, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, + { 6000000, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, + { 6144000, 1, 0x01, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, + { 11289600, 0, 0, 0, 0, 0, 0, 11289600, 0, 1}, + { 12000000, 0, 0, 0, 0, 0, 0, 12000000, 0, 1}, + { 12288000, 0, 0, 0, 0, 0, 0, 12288000, 0, 1}, + { 22579200, 1, 0x03, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, + { 24000000, 1, 0x03, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, + { 24576000, 1, 0x03, 0x40, 0x000000, 0x03, 0x10, 12288000, 128, 1} }; static int cs42l42_pll_config(struct snd_soc_component *component) @@ -619,10 +618,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) 24000000)) << CS42L42_INTERNAL_FS_SHIFT); - snd_soc_component_update_bits(component, CS42L42_MCLK_SRC_SEL, - CS42L42_MCLKDIV_MASK, - (pll_ratio_table[i].mclk_div << - CS42L42_MCLKDIV_SHIFT)); /* Set up the LRCLK */ fsync = clk / cs42l42->srate; if (((fsync * cs42l42->srate) != clk) From 4ae1d8f911d6fc20baefd5eb061bf6964fa22a32 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:15 +0100 Subject: [PATCH 39/45] ASoC: cs42l42: Allow time for HP/ADC to power-up after enable After enabling the HP or ADC by writing the corresponding PDN=0, it takes around 20 milliseconds for it to power up and the midrail supply to be stable. Add this wait into a DAPM widget callback. If HP and ADC are both powering up in a DAPM sequence, there's no need to do the wait twice. The widget will perform one wait in the POST_PMU if there was a PRE_PMU for one or both. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-13-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 31 +++++++++++++++++++++++++++++-- sound/soc/codecs/cs42l42.h | 2 ++ 2 files changed, 31 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 4ec7fdcedca7d..d6d74a7bbde9a 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -435,10 +435,36 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { 0x3f, 1, mixer_tlv) }; +static int cs42l42_hp_adc_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + cs42l42->hp_adc_up_pending = true; + break; + case SND_SOC_DAPM_POST_PMU: + /* Only need one delay if HP and ADC are both powering-up */ + if (cs42l42->hp_adc_up_pending) { + usleep_range(CS42L42_HP_ADC_EN_TIME_US, + CS42L42_HP_ADC_EN_TIME_US + 1000); + cs42l42->hp_adc_up_pending = false; + } + break; + default: + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = { /* Playback Path */ SND_SOC_DAPM_OUTPUT("HP"), - SND_SOC_DAPM_DAC("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1), + SND_SOC_DAPM_DAC_E("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1, + cs42l42_hp_adc_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("MIXER", CS42L42_PWR_CTL1, CS42L42_MIXER_PDN_SHIFT, 1, NULL, 0), SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, SND_SOC_NOPM, 0, 0), @@ -448,7 +474,8 @@ static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = { /* Capture Path */ SND_SOC_DAPM_INPUT("HS"), - SND_SOC_DAPM_ADC("ADC", NULL, CS42L42_PWR_CTL1, CS42L42_ADC_PDN_SHIFT, 1), + SND_SOC_DAPM_ADC_E("ADC", NULL, CS42L42_PWR_CTL1, CS42L42_ADC_PDN_SHIFT, 1, + cs42l42_hp_adc_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_AIF_OUT("SDOUT1", NULL, 0, CS42L42_ASP_TX_CH_EN, CS42L42_ASP_TX0_CH1_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("SDOUT2", NULL, 1, CS42L42_ASP_TX_CH_EN, CS42L42_ASP_TX0_CH2_SHIFT, 0), diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 8734f6828f3ed..ded61af6ea8b6 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -761,6 +761,7 @@ #define CS42L42_CLOCK_SWITCH_DELAY_US 150 #define CS42L42_PLL_LOCK_POLL_US 250 #define CS42L42_PLL_LOCK_TIMEOUT_US 1250 +#define CS42L42_HP_ADC_EN_TIME_US 20000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", @@ -794,6 +795,7 @@ struct cs42l42_private { u8 hs_bias_ramp_time; u8 hs_bias_sense_en; u8 stream_use; + bool hp_adc_up_pending; }; #endif /* __CS42L42_H__ */ From fdbd256175a1e11c1ba827112d56b9a3952e1219 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:16 +0100 Subject: [PATCH 40/45] ASoC: cs42l42: Set correct SRC MCLK According to the datasheet the SRC MCLK must be as near as possible to (125 * sample rate). This means it should be ~6MHz for rates up to 48k and ~12MHz for rates above that. As per datasheet table 4-21. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-14-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 58 +++++++++++++++++++++++++++----------- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 42 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d6d74a7bbde9a..d62272d0ab8c0 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -678,22 +678,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_FSYNC_PULSE_WIDTH_MASK, CS42L42_FRAC1_VAL(fsync - 1) << CS42L42_FSYNC_PULSE_WIDTH_SHIFT); - /* Set the sample rates (96k or lower) */ - snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN, - CS42L42_FS_EN_MASK, - (CS42L42_FS_EN_IASRC_96K | - CS42L42_FS_EN_OASRC_96K) << - CS42L42_FS_EN_SHIFT); - /* Set the input/output internal MCLK clock ~12 MHz */ - snd_soc_component_update_bits(component, CS42L42_IN_ASRC_CLK, - CS42L42_CLK_IASRC_SEL_MASK, - CS42L42_CLK_IASRC_SEL_12 << - CS42L42_CLK_IASRC_SEL_SHIFT); - snd_soc_component_update_bits(component, - CS42L42_OUT_ASRC_CLK, - CS42L42_CLK_OASRC_SEL_MASK, - CS42L42_CLK_OASRC_SEL_12 << - CS42L42_CLK_OASRC_SEL_SHIFT); if (pll_ratio_table[i].mclk_src_sel == 0) { /* Pass the clock straight through */ snd_soc_component_update_bits(component, @@ -756,6 +740,39 @@ static int cs42l42_pll_config(struct snd_soc_component *component) return -EINVAL; } +static void cs42l42_src_config(struct snd_soc_component *component, unsigned int sample_rate) +{ + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + unsigned int fs; + + /* Don't reconfigure if there is an audio stream running */ + if (cs42l42->stream_use) + return; + + /* SRC MCLK must be as close as possible to 125 * sample rate */ + if (sample_rate <= 48000) + fs = CS42L42_CLK_IASRC_SEL_6; + else + fs = CS42L42_CLK_IASRC_SEL_12; + + /* Set the sample rates (96k or lower) */ + snd_soc_component_update_bits(component, + CS42L42_FS_RATE_EN, + CS42L42_FS_EN_MASK, + (CS42L42_FS_EN_IASRC_96K | + CS42L42_FS_EN_OASRC_96K) << + CS42L42_FS_EN_SHIFT); + + snd_soc_component_update_bits(component, + CS42L42_IN_ASRC_CLK, + CS42L42_CLK_IASRC_SEL_MASK, + fs << CS42L42_CLK_IASRC_SEL_SHIFT); + snd_soc_component_update_bits(component, + CS42L42_OUT_ASRC_CLK, + CS42L42_CLK_OASRC_SEL_MASK, + fs << CS42L42_CLK_OASRC_SEL_SHIFT); +} + static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_component *component = codec_dai->component; @@ -846,6 +863,7 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); unsigned int width = (params_width(params) / 8) - 1; unsigned int val = 0; + int ret; cs42l42->srate = params_rate(params); cs42l42->bclk = snd_soc_params_to_bclk(params); @@ -899,7 +917,13 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, break; } - return cs42l42_pll_config(component); + ret = cs42l42_pll_config(component); + if (ret) + return ret; + + cs42l42_src_config(component, params_rate(params)); + + return 0; } static int cs42l42_set_sysclk(struct snd_soc_dai *dai, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index ded61af6ea8b6..d13749e9d8c57 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -288,6 +288,7 @@ #define CS42L42_IN_ASRC_CLK (CS42L42_PAGE_12 + 0x0A) #define CS42L42_CLK_IASRC_SEL_SHIFT 0 #define CS42L42_CLK_IASRC_SEL_MASK (1 << CS42L42_CLK_IASRC_SEL_SHIFT) +#define CS42L42_CLK_IASRC_SEL_6 0 #define CS42L42_CLK_IASRC_SEL_12 1 #define CS42L42_OUT_ASRC_CLK (CS42L42_PAGE_12 + 0x0B) From 0c3d6c6ff75aa6b21cd4ac872dd3050b6525c75c Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:17 +0100 Subject: [PATCH 41/45] ASoC: cs42l42: Mark OSC_SWITCH_STATUS register volatile OSC_SWITCH_STATUS is a volatile register indicating the current state of the clock switch logic. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-15-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d62272d0ab8c0..390cd7ea3a7ff 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -52,7 +52,6 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_RSENSE_CTL1, 0x40 }, { CS42L42_RSENSE_CTL2, 0x00 }, { CS42L42_OSC_SWITCH, 0x00 }, - { CS42L42_OSC_SWITCH_STATUS, 0x05 }, { CS42L42_RSENSE_CTL3, 0x1B }, { CS42L42_TSENSE_CTL, 0x1B }, { CS42L42_TSRS_INT_DISABLE, 0x00 }, @@ -331,6 +330,7 @@ static bool cs42l42_volatile_register(struct device *dev, unsigned int reg) case CS42L42_DEVID_CD: case CS42L42_DEVID_E: case CS42L42_MCLK_STATUS: + case CS42L42_OSC_SWITCH_STATUS: case CS42L42_TRSENSE_STATUS: case CS42L42_HS_DET_STATUS: case CS42L42_ADC_OVFL_STATUS: From 4c8d49bc476c7cf1fb7377b469ced43ced470027 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:18 +0100 Subject: [PATCH 42/45] ASoC: cs42l42: Fix WARN in remove() if running without an interrupt The driver must free the IRQ in remove() to prevent the potential race where an IRQ starts to be handled while the driver is being removed but devres has not yet called free_irq(). However, the driver can run without an interrupt but devm_free_irq() will hit a WARN() if no devres-managed interrupt was ever created. Fix this by only attempting to create the interrupt handler if the hardware config specified an interrupt, and failing probe() if the interrupt could not be created. This means that in cs42l42_remove() an interrupt must have been registered if the irq number is valid and therefore it is safe to call devm_free_irq(). Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-16-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 390cd7ea3a7ff..c6d91ad996e0b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1975,17 +1975,21 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, } usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); - /* Request IRQ */ - ret = devm_request_threaded_irq(&i2c_client->dev, - i2c_client->irq, - NULL, cs42l42_irq_thread, - IRQF_ONESHOT | IRQF_TRIGGER_LOW, - "cs42l42", cs42l42); - if (ret == -EPROBE_DEFER) - goto err_disable; - else if (ret != 0) - dev_err(&i2c_client->dev, - "Failed to request IRQ: %d\n", ret); + /* Request IRQ if one was specified */ + if (i2c_client->irq) { + ret = devm_request_threaded_irq(&i2c_client->dev, + i2c_client->irq, + NULL, cs42l42_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs42l42", cs42l42); + if (ret == -EPROBE_DEFER) { + goto err_disable; + } else if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request IRQ: %d\n", ret); + goto err_disable; + } + } /* initialize codec */ devid = cirrus_read_device_id(cs42l42->regmap, CS42L42_DEVID_AB); @@ -2056,7 +2060,9 @@ static int cs42l42_i2c_remove(struct i2c_client *i2c_client) { struct cs42l42_private *cs42l42 = i2c_get_clientdata(i2c_client); - devm_free_irq(&i2c_client->dev, i2c_client->irq, cs42l42); + if (i2c_client->irq) + devm_free_irq(&i2c_client->dev, i2c_client->irq, cs42l42); + pm_runtime_suspend(&i2c_client->dev); pm_runtime_disable(&i2c_client->dev); From 4ca239f33737198827c7f4ac68a1f6fc8a9d79ba Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 15 Oct 2021 14:36:19 +0100 Subject: [PATCH 43/45] ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts The headset type detection must run to set the analogue switches correctly for the attached headset type. Without this only headsets with wiring matching the chip default will have a functioning mic. commit c26a5289e865 ("ASoC: cs42l42: Add support for set_jack calls") moved the interrupt unmasking to the component set_jack() callback. But it's not mandatory for a machine driver to register a struct snd_soc_jack handler. Without a registered handler the type detection would not have run and so the mic would not work on some types of headset. This patch restores the unmasking of TS_PLUG and TS_UNPLUG interrupts during probe. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211015133619.4698-17-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index c6d91ad996e0b..ac145915445a9 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -524,12 +524,6 @@ static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_ cs42l42->jack = jk; - regmap_update_bits(cs42l42->regmap, CS42L42_TSRS_PLUG_INT_MASK, - CS42L42_RS_PLUG_MASK | CS42L42_RS_UNPLUG_MASK | - CS42L42_TS_PLUG_MASK | CS42L42_TS_UNPLUG_MASK, - (1 << CS42L42_RS_PLUG_SHIFT) | (1 << CS42L42_RS_UNPLUG_SHIFT) | - (0 << CS42L42_TS_PLUG_SHIFT) | (0 << CS42L42_TS_UNPLUG_SHIFT)); - return 0; } @@ -1691,8 +1685,8 @@ static void cs42l42_set_interrupt_masks(struct cs42l42_private *cs42l42) CS42L42_TS_UNPLUG_MASK, (1 << CS42L42_RS_PLUG_SHIFT) | (1 << CS42L42_RS_UNPLUG_SHIFT) | - (1 << CS42L42_TS_PLUG_SHIFT) | - (1 << CS42L42_TS_UNPLUG_SHIFT)); + (0 << CS42L42_TS_PLUG_SHIFT) | + (0 << CS42L42_TS_UNPLUG_SHIFT)); } static void cs42l42_setup_hs_type_detect(struct cs42l42_private *cs42l42) From 5ba8ecf2272d34de9cd2271a0ac12f5f615ef7aa Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Tue, 28 Sep 2021 09:35:57 +0800 Subject: [PATCH 44/45] ASoC: rockchip: Use generic dmaengine code This reverts commit 75b31192fe6ad20b42276b20ee3bdf1493216d63. The original purpose of customized pcm was to config prealloc buffer size flexibly. but, we can do the same thing by soc-generic-dmaengine-pcm. And the generic one can generated the better config by querying DMA capabilities from dmaengine driver rather than the Hard-Coded one. e.g. the customized one: static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, ... the generic one: ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause && dma_caps.cmd_resume) hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw.info |= SNDRV_PCM_INFO_BATCH; ... So, let's revert back to use the generic dmaengine pcm. Signed-off-by: Sugar Zhang Reviewed-by: John Keeping Link: https://lore.kernel.org/r/1632792957-80428-1-git-send-email-sugar.zhang@rock-chips.com Signed-off-by: Mark Brown --- sound/soc/rockchip/Makefile | 3 +-- sound/soc/rockchip/rockchip_i2s.c | 3 +-- sound/soc/rockchip/rockchip_pcm.c | 44 ------------------------------- sound/soc/rockchip/rockchip_pcm.h | 11 -------- 4 files changed, 2 insertions(+), 59 deletions(-) delete mode 100644 sound/soc/rockchip/rockchip_pcm.c delete mode 100644 sound/soc/rockchip/rockchip_pcm.h diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 65e814d460064..05b078e7b87f9 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,11 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o -snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 7e89f5b0c237f..a6d7656c206e5 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -20,7 +20,6 @@ #include #include "rockchip_i2s.h" -#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -756,7 +755,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = rockchip_pcm_platform_register(&pdev->dev); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); goto err_suspend; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c deleted file mode 100644 index 02254e42135e5..0000000000000 --- a/sound/soc/rockchip/rockchip_pcm.c +++ /dev/null @@ -1,44 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#include -#include -#include - -#include -#include -#include -#include - -#include "rockchip_pcm.h" - -static const struct snd_pcm_hardware snd_rockchip_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_INTERLEAVED, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 52, - .buffer_bytes_max = 64 * 1024, - .fifo_size = 32, -}; - -static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { - .pcm_hardware = &snd_rockchip_hardware, - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .prealloc_buffer_size = 32 * 1024, -}; - -int rockchip_pcm_platform_register(struct device *dev) -{ - return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); -} -EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); - -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h deleted file mode 100644 index 7f00e2ce36030..0000000000000 --- a/sound/soc/rockchip/rockchip_pcm.h +++ /dev/null @@ -1,11 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#ifndef _ROCKCHIP_PCM_H -#define _ROCKCHIP_PCM_H - -int rockchip_pcm_platform_register(struct device *dev); - -#endif From 5d03907bbf9ccf10e0d2cfb4f4d312b7cc4274f4 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Tue, 26 Oct 2021 20:27:54 +0200 Subject: [PATCH 45/45] ASoC: meson: t9015: Add missing AVDD-supply property Fixes the schema check warning "audio-controller@32000: 'AVDD-supply' do not match any of the regexes: 'pinctrl-[0-9]+'" Fixes: 5c36abcd2621 ("ASoC: meson: add t9015 internal codec binding documentation") Reviewed-by: Jerome Brunet Signed-off-by: Alexander Stein Link: https://lore.kernel.org/r/20211026182754.900688-1-alexander.stein@mailbox.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/amlogic,t9015.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml index c7613ea728d4c..db7b04da0b394 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -34,6 +34,10 @@ properties: resets: maxItems: 1 + AVDD-supply: + description: + Analogue power supply. + required: - "#sound-dai-cells" - compatible @@ -41,6 +45,7 @@ required: - clocks - clock-names - resets + - AVDD-supply additionalProperties: false @@ -56,4 +61,5 @@ examples: clocks = <&clkc CLKID_AUDIO_CODEC>; clock-names = "pclk"; resets = <&reset RESET_AUDIO_CODEC>; + AVDD-supply = <&vddao_1v8>; };