From 58eafe1ff52ee1ce255759fc15729519af180cbb Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Sep 2021 13:44:36 -0500 Subject: [PATCH 01/25] 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 02/25] 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 03/25] 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 04/25] 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 05/25] 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 06/25] 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 07/25] 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 08/25] 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 09/25] 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 64794d6db49730d22f440aef0cf4da98a56a4ea3 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 13 Sep 2021 11:10:42 +0900 Subject: [PATCH 10/25] ALSA: oxfw: fix transmission method for Loud models based on OXFW971 Loud Technologies Mackie Onyx 1640i (former model) is identified as the model which uses OXFW971. The analysis of packet dump shows that it transfers events in blocking method of IEC 61883-6, however the default behaviour of ALSA oxfw driver is for non-blocking method. This commit adds code to detect it assuming that all of loud models based on OXFW971 have such quirk. It brings no functional change except for alignment rule of PCM buffer. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20210913021042.10085-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cb5b5e3a481b9..daf731364695b 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id model = val; } - /* - * Mackie Onyx Satellite with base station has a quirk to report a wrong - * value in 'dbs' field of CIP header against its format information. - */ - if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + if (vendor == VENDOR_LOUD) { + // Mackie Onyx Satellite with base station has a quirk to report a wrong + // value in 'dbs' field of CIP header against its format information. oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS; + // OXFW971-based models may transfer events by blocking method. + if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + } + return 0; } From 6f44578430d7888ade1e3bd919c1c2c0724409e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Sep 2021 14:43:30 +0200 Subject: [PATCH 11/25] Revert "ALSA: hda: Drop workaround for a hang at shutdown again" This reverts commit 8fc8e903156f42c66245838441d03607e9067381. It was expected that the fixes in HD-audio codec side would make the workaround redundant, but unfortunately it doesn't seem sufficing. Resurrect the workaround for now. Fixes: 8fc8e903156f ("ALSA: hda: Drop workaround for a hang at shutdown again") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214045 Link: https://lore.kernel.org/r/20210913124330.24530-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3aa432d814a24..47777439961c3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void azx_shutdown_chip(struct azx *chip) +static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) { azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!skip_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); +static void azx_shutdown_chip(struct azx *chip) +{ + __azx_shutdown_chip(chip, false); +} + static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -2357,7 +2363,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_shutdown_chip(chip); + __azx_shutdown_chip(chip, true); } /* PCI IDs */ From 7b9cf9036609428e845dc300aec13822ba2c4ab3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Sep 2021 12:51:55 +0200 Subject: [PATCH 12/25] ALSA: usb-audio: Unify mixer resume and reset_resume procedure USB-audio driver assumes that the normal resume would preserve the device configuration while reset_resume wouldn't, and tries to restore the mixer elements only at reset_resume callback. However, this seems too naive, and some devices do behave differently, resetting the volume at the normal resume; this resulted in the inconsistent volume that surprised users. This patch changes the mixer resume code to handle both the normal and reset resume in the same way, always restoring the original mixer element values. This allows us to unify the both callbacks as well as dropping the no longer used reset_resume field, which ends up with a good code reduction. A slight behavior change by this patch is that now we assign restore_mixer_value() as the default resume callback, and the function is no longer called at reset-resume when the resume callback is overridden by the quirk function. That is, if needed, the quirk resume function would have to handle similarly as restore_mixer_value() by itself. Reported-by: En-Shuo Hsu Cc: Yu-Hsuan Hsu Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsMew@mail.gmail.com Link: https://lore.kernel.org/r/20210910105155.12862-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 18 ++++-------------- sound/usb/mixer.c | 26 ++++---------------------- sound/usb/mixer.h | 3 +-- sound/usb/mixer_quirks.c | 2 +- 4 files changed, 10 insertions(+), 39 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index fd570a42f0431..1764b9302d467 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1054,7 +1054,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) +static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; @@ -1080,7 +1080,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_resume(mixer, reset_resume); + err = snd_usb_mixer_resume(mixer); if (err < 0) goto err_out; } @@ -1100,20 +1100,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } - -static int usb_audio_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, false); -} - -static int usb_audio_reset_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, true); -} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL -#define usb_audio_reset_resume NULL +#define usb_audio_resume NULL #endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { @@ -1135,7 +1125,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, - .reset_resume = usb_audio_reset_resume, + .reset_resume = usb_audio_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 43bc59575a6e3..a2ce535df14b7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3653,33 +3653,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) -{ - int err; - - if (list->resume) { - err = list->resume(list); - if (err < 0) - return err; - } - return restore_mixer_value(list); -} - -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) { struct usb_mixer_elem_list *list; - usb_mixer_elem_resume_func_t f; int id, err; /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { for_each_mixer_elem(list, mixer, id) { - if (reset_resume) - f = list->reset_resume; - else - f = list->resume; - if (f) { - err = f(list); + if (list->resume) { + err = list->resume(list); if (err < 0) return err; } @@ -3700,7 +3683,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = NULL; - list->reset_resume = default_mixer_reset_resume; + list->resume = restore_mixer_value; #endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 876bbc9a71ad7..98ea24d91d803 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -70,7 +70,6 @@ struct usb_mixer_elem_list { bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; - usb_mixer_elem_resume_func_t reset_resume; }; /* iterate over mixer element list of the given unit id */ @@ -121,7 +120,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, #ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); #endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a66ce0375fd97..46082dc57be09 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, *listp = list; list->mixer = mixer; list->id = id; - list->reset_resume = resume; + list->resume = resume; kctl = snd_ctl_new1(knew, list); if (!kctl) { kfree(list); From ad7cc2d41b7a8d0c5c5ecff37c3de7a4e137b3a6 Mon Sep 17 00:00:00 2001 From: Cameron Berkenpas Date: Mon, 13 Sep 2021 14:26:29 -0700 Subject: [PATCH 13/25] ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops. This patch initializes and enables speaker output on the Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 series of laptops using the HDA verb sequence specific to each model. Speaker automute is suppressed for the Lenovo Legion 7i 15IMHG05 to avoid breaking speaker output on resume and when devices are unplugged from its headphone jack. Thanks to: Andreas Holzer, Vincent Morel, sycxyc, Max Christian Pohle and all others that helped. [ minor coding style fixes by tiwai ] BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas Cc: Link: https://lore.kernel.org/r/20210913212627.339362-1-cam@neo-zeon.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 129 ++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b7a389b6aedb..4407f7da57c40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } +/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */ +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.suppress_auto_mute = 1; + break; + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6646,6 +6660,10 @@ enum { ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC287_FIXUP_13S_GEN2_SPEAKERS }; static const struct hda_fixup alc269_fixups[] = { @@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + //.v.verbs = legion_15imhg05_coefs, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Legion 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Legion 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Yoga 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Yoga 7i. + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), From ac4dfccb96571ca03af7cac64b7a0b2952c97f3a Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Wed, 15 Sep 2021 09:32:30 +0300 Subject: [PATCH 14/25] 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 15/25] 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 16/25] 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 be830389bd49d3f1f8737bd45513361628641c08 Mon Sep 17 00:00:00 2001 From: Alexey Dobriyan Date: Tue, 14 Sep 2021 14:08:47 +0300 Subject: [PATCH 17/25] ALSA: pcxhr: "fix" PCXHR_REG_TO_PORT definition The following preprocessor directive is non-compliant: #undef PCXHR_REG_TO_PORT(x) gcc warns about extra tokens but nobody sees them as they are under if branch which is never parsed. Make it an #error, it is not clear to me what the author meant. Signed-off-by: Alexey Dobriyan Link: https://lore.kernel.org/r/YUCCv47sm4zf9OVO@localhost.localdomain Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 87d24224c042e..23f253effb4fa 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -52,7 +52,7 @@ #define PCXHR_DSP 2 #if (PCXHR_DSP_OFFSET_MAX > PCXHR_PLX_OFFSET_MIN) -#undef PCXHR_REG_TO_PORT(x) +#error PCXHR_REG_TO_PORT(x) #else #define PCXHR_REG_TO_PORT(x) ((x)>PCXHR_DSP_OFFSET_MAX ? PCXHR_PLX : PCXHR_DSP) #endif From 94d508fa3186d0cbc63765aa94d5cf3bd847694c Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 16 Sep 2021 10:56:46 +0100 Subject: [PATCH 18/25] ALSA: hda/cs8409: Setup Dolphin Headset Mic as Phantom Jack Dell's requirement to have headset mic as phantom jack on this specific dolphin hardware platform. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20210916095646.7631-1-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 3c7ef55d016e9..31ff11ab868e1 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -1207,6 +1207,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac snd_hda_jack_add_kctl(codec, DOLPHIN_LO_PIN_NID, "Line Out", true, SND_JACK_HEADPHONE, NULL); + snd_hda_jack_add_kctl(codec, DOLPHIN_AMIC_PIN_NID, "Microphone", true, + SND_JACK_MICROPHONE, NULL); + cs8409_fix_caps(codec, DOLPHIN_HP_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID); From 8a8e1813ffc35111fc0b6db49968ceb0e1615ced Mon Sep 17 00:00:00 2001 From: Marc Herbert Date: Thu, 16 Sep 2021 11:50:08 +0300 Subject: [PATCH 19/25] 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 20/25] 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 21/25] 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 22/25] 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 cb1bcf5ed536747013fe2b3f9bd56ce3242c295a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 20 Sep 2021 20:07:34 +0900 Subject: [PATCH 23/25] ALSA: firewire-motu: fix truncated bytes in message tracepoints In MOTU protocol v2/v3, first two data chunks across 2nd and 3rd data channels includes message bytes from device. The total size of message is 48 bits per data block. The 'data_block_message' tracepoints event produced by ALSA firewire-motu driver exposes the sequence of messages to userspace in 64 bit storage, however lower 32 bits are actually available since current implementation truncates 16 bits in upper of the message as a result of bit shift operation within 32 bit storage. This commit fixes the bug by perform the bit shift in 64 bit storage. Fixes: c6b0b9e65f09 ("ALSA: firewire-motu: add tracepoints for messages for unique protocol") Cc: Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20210920110734.27161-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/amdtp-motu.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 5388b85fb60e5..a18c2c033e836 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer, /* This is just for v2/v3 protocol. */ for (i = 0; i < data_blocks; ++i) { - *frames = (be32_to_cpu(buffer[1]) << 16) | - (be32_to_cpu(buffer[2]) >> 16); + *frames = be32_to_cpu(buffer[1]); + *frames <<= 16; + *frames |= be32_to_cpu(buffer[2]) >> 16; + ++frames; buffer += data_block_quadlets; - frames++; } } From 09d23174402da0f10e98da2c61bb5ac8e7d79fdd Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 20 Sep 2021 19:18:50 +0200 Subject: [PATCH 24/25] ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION The new framing mode causes the user space regression, because the alsa-lib code does not initialize the reserved space in the params structure when the device is opened. This change adds SNDRV_RAWMIDI_IOCTL_USER_PVERSION like we do for the PCM interface for the protocol acknowledgment. Cc: David Henningsson Cc: Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20210920171850.154186-1-perex@perex.cz Signed-off-by: Takashi Iwai --- include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/rawmidi.c | 9 +++++++++ 3 files changed, 11 insertions(+) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 989e1517332d6..7a08ed2acd609 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -98,6 +98,7 @@ struct snd_rawmidi_file { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; + unsigned int user_pversion; /* supported protocol version */ }; struct snd_rawmidi_str { diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1d84ec9db93bd..5859ca0a1439b 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -784,6 +784,7 @@ struct snd_rawmidi_status { #define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) #define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) +#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('W', 0x02, int) #define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params) #define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status) #define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6c0a4a67ad2e3..6f30231bdb884 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long return -EINVAL; } } + case SNDRV_RAWMIDI_IOCTL_USER_PVERSION: + if (get_user(rfile->user_pversion, (unsigned int __user *)arg)) + return -EFAULT; + return 0; + case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; if (copy_from_user(¶ms, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; + if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) { + params.mode = 0; + memset(params.reserved, 0, sizeof(params.reserved)); + } switch (params.stream) { case SNDRV_RAWMIDI_STREAM_OUTPUT: if (rfile->output == NULL) From f2ff7147c6834f244b8ce636b12e71a3bd044629 Mon Sep 17 00:00:00 2001 From: Thomas Gleixner Date: Thu, 23 Sep 2021 18:04:25 +0200 Subject: [PATCH 25/25] ALSA: pcsp: Make hrtimer forwarding more robust The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner Cc: alsa-devel@alsa-project.org Cc: Takashi Iwai Cc: Jaroslav Kysela Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index ed40d0f7432c8..773db4bf08769 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -143,7 +143,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; }