From 6f95eec6fb89e195dbdf30de65553c7fc57d9372 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 20 Dec 2022 14:56:27 +0200 Subject: ASoC: SOF: pm: Set target state earlier MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If the DSP crashes before the system suspends, the setting of target state will be skipped because the firmware state will no longer be SOF_FW_BOOT_COMPLETE. This leads to the incorrect assumption that the DSP should suspend to D0I3 instead of suspending to D3. To fix this, set the target_state before we skip to DSP suspend even when the DSP has crashed. Signed-off-by: Ranjani Sridharan Reviewed-by: Curtis Malainey Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20221220125629.8469-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index df740be645e8..5f88c4a01fa3 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -182,7 +182,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; const struct sof_ipc_tplg_ops *tplg_ops = sdev->ipc->ops->tplg; pm_message_t pm_state; - u32 target_state = 0; + u32 target_state = snd_sof_dsp_power_target(sdev); int ret; /* do nothing if dsp suspend callback is not set */ @@ -206,7 +206,6 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) } } - target_state = snd_sof_dsp_power_target(sdev); pm_state.event = target_state; /* Skip to platform-specific suspend if DSP is entering D0 */ -- cgit v1.2.3-58-ga151 From d185e0689abc98ef55fb7a7d75aa0c48a0ed5838 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 20 Dec 2022 14:56:28 +0200 Subject: ASoC: SOF: pm: Always tear down pipelines before DSP suspend MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When the DSP is suspended while the firmware is in the crashed state, we skip tearing down the pipelines. This means that the widget reference counts will not get to reset to 0 before suspend. This will lead to errors with resuming audio after system resume. To fix this, invoke the tear_down_all_pipelines op before skipping to DSP suspend. Signed-off-by: Ranjani Sridharan Reviewed-by: Curtis Malainey Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20221220125629.8469-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 5f88c4a01fa3..8722bbd7fd3d 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -192,6 +192,9 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) if (runtime_suspend && !sof_ops(sdev)->runtime_suspend) return 0; + if (tplg_ops && tplg_ops->tear_down_all_pipelines) + tplg_ops->tear_down_all_pipelines(sdev, false); + if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) goto suspend; @@ -216,9 +219,6 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) goto suspend; } - if (tplg_ops->tear_down_all_pipelines) - tplg_ops->tear_down_all_pipelines(sdev, false); - /* suspend DMA trace */ sof_fw_trace_suspend(sdev, pm_state); -- cgit v1.2.3-58-ga151 From 9a9134fd56f6ba614ff7b2b3b0bac0bf1d0dc0c9 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Tue, 20 Dec 2022 14:56:29 +0200 Subject: ASoC: SOF: Add FW state to debugfs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Allow system health detection mechanisms to check the FW state, this will allow them to check if the FW is in its "crashed" state going forward to help automatically diagnose driver state. Signed-off-by: Curtis Malainey Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20221220125629.8469-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index d9a3ce7b69e1..ade0507328af 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -353,7 +353,9 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev) return err; } - return 0; + return snd_sof_debugfs_buf_item(sdev, &sdev->fw_state, + sizeof(sdev->fw_state), + "fw_state", 0444); } EXPORT_SYMBOL_GPL(snd_sof_dbg_init); -- cgit v1.2.3-58-ga151 From 896c3dc21f1e84cb2f60d54572fc3377eb57e004 Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Mon, 26 Dec 2022 09:09:16 +0800 Subject: ASoC: Intel: soc-acpi: add configuration for variant of 0C40 product Support configuration with SoundWire RT1316 amplifiers on link0 and link1, and RT711 on link2 for headphone/headset. This product does not support local microphones. Signed-off-by: Gongjun Song Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20221226010917.2632973-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 25 +++++++++++++++++++++++ 1 file changed, 25 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 31b43116e3d8..c70d85bfedbf 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -203,6 +203,25 @@ static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link2_rt1316_link01_rt71 {} }; +static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link2_rt1316_link01[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt711_sdca_2_adr), + .adr_d = rt711_sdca_2_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt1316_0_group2_adr), + .adr_d = rt1316_0_group2_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1316_1_group2_adr), + .adr_d = rt1316_1_group2_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12_rt714_link3[] = { { .mask = BIT(0), @@ -277,6 +296,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-rpl-rt1316-l12-rt714-l0.tplg", }, + { + .link_mask = 0x7, /* rt711 on link2 & two rt1316s on link0 and link1 */ + .links = rpl_sdw_rt711_link2_rt1316_link01, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-rpl-rt711-l2-rt1316-l01.tplg", + }, { .link_mask = 0x1, /* link0 required */ .links = rpl_rvp, -- cgit v1.2.3-58-ga151 From b25a31b463391cc47a654594eb154ebf5dd0d60a Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Mon, 26 Dec 2022 09:09:17 +0800 Subject: ASoC: Intel: soc-acpi: add configuration for variant of 0C11 product Support configuration with SoundWire RT1318 amplifiers on link1 and link2, and RT711 on link0 for headphone/headset. This product does not support local microphones. Signed-off-by: Gongjun Song Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20221226010917.2632973-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 25 +++++++++++++++++++++++ 1 file changed, 25 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index c70d85bfedbf..07f96a11ea2f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -246,6 +246,25 @@ static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12_rt71 {} }; +static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1318_1_group1_adr), + .adr_d = rt1318_1_group1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1318_2_group1_adr), + .adr_d = rt1318_2_group1_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr rpl_sdw_rt1316_link12_rt714_link0[] = { { .mask = BIT(1), @@ -290,6 +309,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-rpl-rt711-l0-rt1318-l12-rt714-l3.tplg", }, + { + .link_mask = 0x7, /* rt711 on link0 & two rt1318s on link1 and link2 */ + .links = rpl_sdw_rt711_link0_rt1318_link12, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-rpl-rt711-l0-rt1318-l12.tplg", + }, { .link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */ .links = rpl_sdw_rt1316_link12_rt714_link0, -- cgit v1.2.3-58-ga151 From 68506a173dd700c2bd794dcc3489edcdb8ee35c6 Mon Sep 17 00:00:00 2001 From: Wim Van Boven Date: Fri, 16 Dec 2022 09:18:27 +0100 Subject: ASoC: amd: yc: Add Razer Blade 14 2022 into DMI table Razer Blade 14 (2022) - RZ09-0427 needs the quirk to enable the built in microphone Signed-off-by: Wim Van Boven Reviewed-by: Mario Limonciello Link: https://lore.kernel.org/r/20221216081828.12382-1-wimvanboven@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 1f0b5527c594..469c5e79e0ea 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -220,6 +220,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Redmi Book Pro 14 2022"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Razer"), + DMI_MATCH(DMI_PRODUCT_NAME, "Blade 14 (2022) - RZ09-0427"), + } + }, {} }; -- cgit v1.2.3-58-ga151 From ba7523bb0f494fc440d3a9bb0b665cfcaa192d0c Mon Sep 17 00:00:00 2001 From: Ajye Huang Date: Thu, 22 Dec 2022 12:26:24 +0800 Subject: ASoC: Intel: sof_nau8825: add variant with nau8318 amplifier. This patch adds the driver data for two nau8318 speaker amplifiers on SSP1 and nau8825 on SSP0 for ADL platform. The nau8315 and nau8318 are both Nuvoton Amp chips. They use the same Amp driver nau8315.c. The acpi_device_id for nau8315 is "NVTN2010", for nau8318 is "NVTN2012". The nau8825 is one of Nuvoton headset codec, and its acpi_device_id is "10508825". Signed-off-by: Ajye Huang Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20221222042624.557869-1-ajye_huang@compal.corp-partner.google.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_nau8825.c | 23 +++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 12 ++++++++++++ 3 files changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index a472de1909f4..3f68e9edd853 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -554,6 +554,7 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH select SND_SOC_RT1015P select SND_SOC_MAX98373_I2C select SND_SOC_MAX98357A + select SND_SOC_NAU8315 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 27880224359d..78d84527081a 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -48,6 +48,7 @@ #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_NAU8318_SPEAKER_AMP_PRESENT BIT(18) static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); @@ -338,6 +339,13 @@ static struct snd_soc_dai_link_component rt1019p_component[] = { } }; +static struct snd_soc_dai_link_component nau8318_components[] = { + { + .name = "NVTN2012:00", + .dai_name = "nau8315-hifi", + } +}; + static struct snd_soc_dai_link_component dummy_component[] = { { .name = "snd-soc-dummy", @@ -486,6 +494,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, max_98360a_dai_link(&links[id]); } else if (sof_nau8825_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { sof_rt1015p_dai_link(&links[id]); + } else if (sof_nau8825_quirk & + SOF_NAU8318_SPEAKER_AMP_PRESENT) { + links[id].codecs = nau8318_components; + links[id].num_codecs = ARRAY_SIZE(nau8318_components); + links[id].init = speaker_codec_init; } else { goto devm_err; } @@ -657,6 +670,16 @@ static const struct platform_device_id board_ids[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .name = "adl_nau8318_8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_NAU8318_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 60aee56f94bd..b1c0a89a8787 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -450,6 +450,11 @@ static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = { .codecs = {"INTC10B0"} }; +static const struct snd_soc_acpi_codecs adl_nau8318_amp = { + .num_codecs = 1, + .codecs = {"NVTN2012"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &adl_rt5682_rt5682s_hp, @@ -507,6 +512,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_rt1015p_amp, .sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg", }, + { + .id = "10508825", + .drv_name = "adl_nau8318_8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_nau8318_amp, + .sof_tplg_filename = "sof-adl-nau8318-nau8825.tplg", + }, { .id = "10508825", .drv_name = "sof_nau8825", -- cgit v1.2.3-58-ga151 From 63f3d99b7efe4c5404a9388c05780917099cecf4 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 21 Dec 2022 14:25:48 +0100 Subject: ASoC: Intel: fix sof-nau8825 link failure The snd-soc-sof_nau8825.ko module fails to link unless the sof_realtek_common support is also enabled: ERROR: modpost: "sof_rt1015p_codec_conf" [sound/soc/intel/boards/snd-soc-sof_nau8825.ko] undefined! ERROR: modpost: "sof_rt1015p_dai_link" [sound/soc/intel/boards/snd-soc-sof_nau8825.ko] undefined! Fixes: 8d0872f6239f ("ASoC: Intel: add sof-nau8825 machine driver") Signed-off-by: Arnd Bergmann Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20221221132559.2402341-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3f68e9edd853..99308ed85277 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -559,6 +559,7 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_MAXIM_COMMON + select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms with nau8825 codec. -- cgit v1.2.3-58-ga151 From 3e78986a840d59dd27e636eae3f52dc11125c835 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 21 Dec 2022 14:24:56 +0100 Subject: ASoC: Intel: sof-nau8825: fix module alias overflow The maximum name length for a platform_device_id entry is 20 characters including the trailing NUL byte. The sof_nau8825.c file exceeds that, which causes an obscure error message: sound/soc/intel/boards/snd-soc-sof_nau8825.mod.c:35:45: error: illegal character encoding in string literal [-Werror,-Winvalid-source-encoding] MODULE_ALIAS("platform:adl_max98373_nau8825"); ^~~~ include/linux/module.h:168:49: note: expanded from macro 'MODULE_ALIAS' ^~~~~~ include/linux/module.h:165:56: note: expanded from macro 'MODULE_INFO' ^~~~ include/linux/moduleparam.h:26:47: note: expanded from macro '__MODULE_INFO' = __MODULE_INFO_PREFIX __stringify(tag) "=" info I could not figure out how to make the module handling robust enough to handle this better, but as a quick fix, using slightly shorter names that are still unique avoids the build issue. Fixes: 8d0872f6239f ("ASoC: Intel: add sof-nau8825 machine driver") Signed-off-by: Arnd Bergmann Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20221221132515.2363276-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 8 ++++---- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 8 ++++---- 2 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 78d84527081a..a800854c2831 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -631,7 +631,7 @@ static const struct platform_device_id board_ids[] = { }, { - .name = "adl_rt1019p_nau8825", + .name = "adl_rt1019p_8825", .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | SOF_RT1019P_SPEAKER_AMP_PRESENT | @@ -639,7 +639,7 @@ static const struct platform_device_id board_ids[] = { SOF_NAU8825_NUM_HDMIDEV(4)), }, { - .name = "adl_max98373_nau8825", + .name = "adl_max98373_8825", .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | SOF_MAX98373_SPEAKER_AMP_PRESENT | @@ -650,7 +650,7 @@ static const struct platform_device_id board_ids[] = { }, { /* The limitation of length of char array, shorten the name */ - .name = "adl_mx98360a_nau8825", + .name = "adl_mx98360a_8825", .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | SOF_MAX98360A_SPEAKER_AMP_PRESENT | @@ -661,7 +661,7 @@ static const struct platform_device_id board_ids[] = { }, { - .name = "adl_rt1015p_nau8825", + .name = "adl_rt1015p_8825", .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | SOF_RT1015P_SPEAKER_AMP_PRESENT | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index b1c0a89a8787..56ee5fef66a8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -479,21 +479,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10508825", - .drv_name = "adl_rt1019p_nau8825", + .drv_name = "adl_rt1019p_8825", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_rt1019p_amp, .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", }, { .id = "10508825", - .drv_name = "adl_max98373_nau8825", + .drv_name = "adl_max98373_8825", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98373_amp, .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", }, { .id = "10508825", - .drv_name = "adl_mx98360a_nau8825", + .drv_name = "adl_mx98360a_8825", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, .sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg", @@ -507,7 +507,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10508825", - .drv_name = "adl_rt1015p_nau8825", + .drv_name = "adl_rt1015p_8825", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_rt1015p_amp, .sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg", -- cgit v1.2.3-58-ga151 From 6e1dbf694d7cd1737ee14866e9e05016ccc9ac40 Mon Sep 17 00:00:00 2001 From: tongjian Date: Wed, 28 Dec 2022 20:22:29 +0800 Subject: ASoC: mediatek: mt8186: support rt5682s_max98360 Add support for using the rt5682s codec together with max98360a on MT8186-MT6366-RT1019-RT5682S machines. Signed-off-by: tongjian Link: https://lore.kernel.org/r/20221228122230.3818533-2-tongjian@huaqin.corp-partner.google.com Signed-off-by: Mark Brown --- .../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c index 8f77a0bc1dc8..af44e331dae8 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c @@ -1083,6 +1083,21 @@ static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = { .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf), }; +static struct snd_soc_card mt8186_mt6366_rt5682s_max98360_soc_card = { + .name = "mt8186_rt5682s_max98360", + .owner = THIS_MODULE, + .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links, + .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links), + .controls = mt8186_mt6366_rt1019_rt5682s_controls, + .num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls), + .dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets), + .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes, + .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes), + .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf, + .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf), +}; + static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card; @@ -1232,9 +1247,14 @@ err_adsp_node: #if IS_ENABLED(CONFIG_OF) static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = { - { .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound", + { + .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound", .data = &mt8186_mt6366_rt1019_rt5682s_soc_card, }, + { + .compatible = "mediatek,mt8186-mt6366-rt5682s-max98360-sound", + .data = &mt8186_mt6366_rt5682s_max98360_soc_card, + }, {} }; MODULE_DEVICE_TABLE(of, mt8186_mt6366_rt1019_rt5682s_dt_match); -- cgit v1.2.3-58-ga151 From 8a54f666db581bbf07494cca44a0124acbced581 Mon Sep 17 00:00:00 2001 From: Allen-KH Cheng Date: Wed, 28 Dec 2022 19:57:56 +0800 Subject: ASoC: mediatek: mt8186: Add machine support for max98357a Add support for mt8186 with mt6366 and max98357a. Signed-off-by: Allen-KH Cheng Link: https://lore.kernel.org/r/20221228115756.28014-1-allen-kh.cheng@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 363fa4d47680..7bdb0ded831c 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -182,9 +182,10 @@ config SND_SOC_MT8186_MT6366_DA7219_MAX98357 If unsure select "N". config SND_SOC_MT8186_MT6366_RT1019_RT5682S - tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S codec" + tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S MAX98357A/MAX98360 codec" depends on I2C && GPIOLIB depends on SND_SOC_MT8186 && MTK_PMIC_WRAP + select SND_SOC_MAX98357A select SND_SOC_MT6358 select SND_SOC_RT1015P select SND_SOC_RT5682S -- cgit v1.2.3-58-ga151 From 7161bd540eebebae2bbe8c79de25d8caf12dbf78 Mon Sep 17 00:00:00 2001 From: ChiYuan Huang Date: Thu, 29 Dec 2022 16:03:53 +0800 Subject: ASoC: rt9120: Make dev PM runtime bind AsoC component PM RT9120 uses PM runtime autosuspend to decrease the frequently on/off spent time. This exists one case, when pcm is closed and dev PM is waiting for autosuspend time expired to enter runtime suspend state. At the mean time, system is going to enter suspend, dev PM runtime suspend won't be called. It makes the rt9120 suspend consumption current not as expected. This patch can fix the rt9120 dev PM issue during runtime autosuspend and system suspend by binding dev PM runtime and ASoC component PM. Fixes: 80b949f332e3 ("ASoC: rt9120: Use pm_runtime and regcache to optimize 'pwdnn' logic") Signed-off-by: ChiYuan Huang Link: https://lore.kernel.org/r/1672301033-3675-1-git-send-email-u0084500@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt9120.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt9120.c b/sound/soc/codecs/rt9120.c index 644300e88b4c..fcf4fbaed3c7 100644 --- a/sound/soc/codecs/rt9120.c +++ b/sound/soc/codecs/rt9120.c @@ -177,8 +177,20 @@ static int rt9120_codec_probe(struct snd_soc_component *comp) return 0; } +static int rt9120_codec_suspend(struct snd_soc_component *comp) +{ + return pm_runtime_force_suspend(comp->dev); +} + +static int rt9120_codec_resume(struct snd_soc_component *comp) +{ + return pm_runtime_force_resume(comp->dev); +} + static const struct snd_soc_component_driver rt9120_component_driver = { .probe = rt9120_codec_probe, + .suspend = rt9120_codec_suspend, + .resume = rt9120_codec_resume, .controls = rt9120_snd_controls, .num_controls = ARRAY_SIZE(rt9120_snd_controls), .dapm_widgets = rt9120_dapm_widgets, -- cgit v1.2.3-58-ga151 From a0dd7fcab5cd221fa960f594c586e1f9f16c02c0 Mon Sep 17 00:00:00 2001 From: Aniol Martí Date: Tue, 27 Dec 2022 23:49:32 +0100 Subject: ASoC: amd: yc: Add ASUS M5402RA into DMI table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ASUS VivoBook 13 OLED (M5402RA) needs this quirk to get the built-in microphone working properly. Signed-off-by: Aniol Martí Link: https://lore.kernel.org/r/20221227224932.9771-1-aniol@aniolmarti.cat Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 469c5e79e0ea..0d283e41f66d 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -206,6 +206,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"), + } + }, { .driver_data = &acp6x_card, .matches = { -- cgit v1.2.3-58-ga151 From 810948f45d99c46b60852ef2a5a2777c12d6bb3e Mon Sep 17 00:00:00 2001 From: Mars Chen Date: Wed, 28 Dec 2022 18:38:12 +0800 Subject: ASoC: support machine driver with max98360 Signed-off-by: Mars Chen Link: https://lore.kernel.org/r/20221228103812.450956-1-chenxiangrui@huaqin.corp-partner.google.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 7bdb0ded831c..b027fba8233d 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -187,6 +187,7 @@ config SND_SOC_MT8186_MT6366_RT1019_RT5682S depends on SND_SOC_MT8186 && MTK_PMIC_WRAP select SND_SOC_MAX98357A select SND_SOC_MT6358 + select SND_SOC_MAX98357A select SND_SOC_RT1015P select SND_SOC_RT5682S select SND_SOC_BT_SCO -- cgit v1.2.3-58-ga151 From 472a6309c6467af89dbf660a8310369cc9cb041f Mon Sep 17 00:00:00 2001 From: Emanuele Ghidoli Date: Fri, 23 Dec 2022 09:02:47 +0100 Subject: ASoC: wm8904: fix wrong outputs volume after power reactivation Restore volume after charge pump and PGA activation to ensure that volume settings are correctly applied when re-enabling codec from SND_SOC_BIAS_OFF state. CLASS_W, CHARGE_PUMP and POWER_MANAGEMENT_2 register configuration affect how the volume register are applied and must be configured first. Fixes: a91eb199e4dc ("ASoC: Initial WM8904 CODEC driver") Link: https://lore.kernel.org/all/c7864c35-738c-a867-a6a6-ddf9f98df7e7@gmail.com/ Signed-off-by: Emanuele Ghidoli Signed-off-by: Francesco Dolcini Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221223080247.7258-1-francesco@dolcini.it Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index ca6a01a230af..791d8738d1c0 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -697,6 +697,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, int dcs_mask; int dcs_l, dcs_r; int dcs_l_reg, dcs_r_reg; + int an_out_reg; int timeout; int pwr_reg; @@ -712,6 +713,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1; dcs_r_reg = WM8904_DC_SERVO_8; dcs_l_reg = WM8904_DC_SERVO_9; + an_out_reg = WM8904_ANALOGUE_OUT1_LEFT; dcs_l = 0; dcs_r = 1; break; @@ -720,6 +722,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3; dcs_r_reg = WM8904_DC_SERVO_6; dcs_l_reg = WM8904_DC_SERVO_7; + an_out_reg = WM8904_ANALOGUE_OUT2_LEFT; dcs_l = 2; dcs_r = 3; break; @@ -792,6 +795,10 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, reg, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP); + + /* Update volume, requires PGA to be powered */ + val = snd_soc_component_read(component, an_out_reg); + snd_soc_component_write(component, an_out_reg, val); break; case SND_SOC_DAPM_POST_PMU: -- cgit v1.2.3-58-ga151 From 000bca8d706d1bf7cca01af75787247c5a2fdedf Mon Sep 17 00:00:00 2001 From: Brian Norris Date: Fri, 30 Dec 2022 22:15:45 -0800 Subject: ASoC: qcom: lpass-cpu: Fix fallback SD line index handling These indices should reference the ID placed within the dai_driver array, not the indices of the array itself. This fixes commit 4ff028f6c108 ("ASoC: qcom: lpass-cpu: Make I2S SD lines configurable"), which among others, broke IPQ8064 audio (sound/soc/qcom/lpass-ipq806x.c) because it uses ID 4 but we'd stop initializing the mi2s_playback_sd_mode and mi2s_capture_sd_mode arrays at ID 0. Fixes: 4ff028f6c108 ("ASoC: qcom: lpass-cpu: Make I2S SD lines configurable") Cc: Signed-off-by: Brian Norris Reviewed-by: Stephan Gerhold Link: https://lore.kernel.org/r/20221231061545.2110253-1-computersforpeace@gmail.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 54353842dc07..dbdaaa85ce48 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1037,10 +1037,11 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev, struct lpass_data *data) { struct device_node *node; - int ret, id; + int ret, i, id; /* Allow all channels by default for backwards compatibility */ - for (id = 0; id < data->variant->num_dai; id++) { + for (i = 0; i < data->variant->num_dai; i++) { + id = data->variant->dai_driver[i].id; data->mi2s_playback_sd_mode[id] = LPAIF_I2SCTL_MODE_8CH; data->mi2s_capture_sd_mode[id] = LPAIF_I2SCTL_MODE_8CH; } -- cgit v1.2.3-58-ga151 From 1f680609bf1beac20e2a31ddcb1b88874123c39f Mon Sep 17 00:00:00 2001 From: Yuchi Yang Date: Fri, 30 Dec 2022 15:22:25 +0800 Subject: ALSA: hda/realtek - Turn on power early Turn on power early to avoid wrong state for power relation register. This can earlier update JD state when resume back. Signed-off-by: Yuchi Yang Cc: Link: https://lore.kernel.org/r/e35d8f4fa18f4448a2315cc7d4a3715f@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++-------------- 1 file changed, 16 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3794b522c222..937b227e17c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3564,6 +3564,15 @@ static void alc256_init(struct hda_codec *codec) hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; + if (spec->ultra_low_power) { + alc_update_coef_idx(codec, 0x03, 1<<1, 1<<1); + alc_update_coef_idx(codec, 0x08, 3<<2, 3<<2); + alc_update_coef_idx(codec, 0x08, 7<<4, 0); + alc_update_coef_idx(codec, 0x3b, 1<<15, 0); + alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6); + msleep(30); + } + if (!hp_pin) hp_pin = 0x21; @@ -3575,14 +3584,6 @@ static void alc256_init(struct hda_codec *codec) msleep(2); alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */ - if (spec->ultra_low_power) { - alc_update_coef_idx(codec, 0x03, 1<<1, 1<<1); - alc_update_coef_idx(codec, 0x08, 3<<2, 3<<2); - alc_update_coef_idx(codec, 0x08, 7<<4, 0); - alc_update_coef_idx(codec, 0x3b, 1<<15, 0); - alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6); - msleep(30); - } snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -3713,6 +3714,13 @@ static void alc225_init(struct hda_codec *codec) hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; + if (spec->ultra_low_power) { + alc_update_coef_idx(codec, 0x08, 0x0f << 2, 3<<2); + alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6); + alc_update_coef_idx(codec, 0x33, 1<<11, 0); + msleep(30); + } + if (spec->codec_variant != ALC269_TYPE_ALC287 && spec->codec_variant != ALC269_TYPE_ALC245) /* required only at boot or S3 and S4 resume time */ @@ -3734,12 +3742,6 @@ static void alc225_init(struct hda_codec *codec) msleep(2); alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */ - if (spec->ultra_low_power) { - alc_update_coef_idx(codec, 0x08, 0x0f << 2, 3<<2); - alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6); - alc_update_coef_idx(codec, 0x33, 1<<11, 0); - msleep(30); - } if (hp1_pin_sense || spec->ultra_low_power) snd_hda_codec_write(codec, hp_pin, 0, -- cgit v1.2.3-58-ga151 From a5751933a7f6abbdad90d98f25a25bb4b133a9e6 Mon Sep 17 00:00:00 2001 From: Chris Chiu Date: Tue, 3 Jan 2023 17:53:32 +0800 Subject: ALSA: hda - Enable headset mic on another Dell laptop with ALC3254 There is another Dell Latitude laptop (1028:0c03) with Realtek codec ALC3254 which needs the ALC269_FIXUP_DELL4_MIC_NO_PRESENCE instead of the default matched ALC269_FIXUP_DELL1_MIC_NO_PRESENCE. Apply correct fixup for this particular model to enable headset mic. Signed-off-by: Chris Chiu Cc: Link: https://lore.kernel.org/r/20230103095332.730677-1-chris.chiu@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 937b227e17c5..eda4914b8aeb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9241,6 +9241,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), -- cgit v1.2.3-58-ga151 From 0599313e26666e79f6e7fe1450588431b8cb25d5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jan 2023 18:07:57 +0100 Subject: ALSA: usb-audio: Make sure to stop endpoints before closing EPs At the PCM hw params, we may re-configure the endpoints and it's done by a temporary EP close followed by re-open. A potential problem there is that the EP might be already running internally at the PCM prepare stage; it's seen typically in the playback stream with the implicit feedback sync. As this stream start isn't tracked by the core PCM layer, we'd need to stop it explicitly, and that's the missing piece. This patch adds the stop_endpoints() call at snd_usb_hw_params() to assure the stream stop before closing the EPs. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/4e509aea-e563-e592-e652-ba44af6733fe@veniogames.com Link: https://lore.kernel.org/r/20230102170759.29610-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 99a66d0ef5b2..7fc95ae9b2f0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -525,6 +525,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (snd_usb_endpoint_compatible(chip, subs->data_endpoint, fmt, hw_params)) goto unlock; + if (stop_endpoints(subs, false)) + sync_pending_stops(subs); close_endpoints(chip, subs); } -- cgit v1.2.3-58-ga151 From d463ac1acb454fafed58f695cb3067fbf489f3a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jan 2023 18:07:58 +0100 Subject: ALSA: usb-audio: Relax hw constraints for implicit fb sync The fix commit the commit e4ea77f8e53f ("ALSA: usb-audio: Always apply the hw constraints for implicit fb sync") tried to address the bug where an incorrect PCM parameter is chosen when two (implicit fb) streams are set up at the same time. This change had, however, some side effect: once when the sync endpoint is chosen and set up, this restriction is applied at the next hw params unless it's freed via hw free explicitly. This patch is a workaround for the problem by relaxing the hw constraints a bit for the implicit fb sync. We still keep applying the hw constraints for implicit fb sync, but only when the matching sync EP is being used by other streams. Fixes: e4ea77f8e53f ("ALSA: usb-audio: Always apply the hw constraints for implicit fb sync") Reported-by: Ruud van Asseldonk Link: https://lore.kernel.org/r/4e509aea-e563-e592-e652-ba44af6733fe@veniogames.com Link: https://lore.kernel.org/r/20230102170759.29610-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 7fc95ae9b2f0..2fd4ecc1b25a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -937,8 +937,13 @@ get_sync_ep_from_substream(struct snd_usb_substream *subs) continue; /* for the implicit fb, check the sync ep as well */ ep = snd_usb_get_endpoint(chip, fp->sync_ep); - if (ep && ep->cur_audiofmt) - return ep; + if (ep && ep->cur_audiofmt) { + /* ditto, if the sync (data) ep is used by others, + * this stream is restricted by the sync ep + */ + if (ep != subs->sync_endpoint || ep->opened > 1) + return ep; + } } return NULL; } -- cgit v1.2.3-58-ga151 From 37b3e56d8911f9ec1e1aaa5cccdff33cb0a7a832 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jan 2023 18:07:59 +0100 Subject: ALSA: usb-audio: More refactoring of hw constraint rules Although we applied a workaround for the hw constraints code with the implicit feedback sync, it still has a potential problem. Namely, as the code treats only the first matching (sync) endpoint, it might be too restrictive when multiple endpoints are listed in the substream's format list. This patch is another attempt to improve the hw constraint handling for the implicit feedback sync. The code is rewritten and the sync EP handling for the rate and the format is put inside the fmt_list loop in each hw_rule_*() function instead of the additional rules. The rules for the period size and periods are extended to loop over the fmt_list like others, and they apply the constraints only if needed. Link: https://lore.kernel.org/r/4e509aea-e563-e592-e652-ba44af6733fe@veniogames.com Link: https://lore.kernel.org/r/20230102170759.29610-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 218 ++++++++++++++++++++++++++++++++++---------------------- 1 file changed, 131 insertions(+), 87 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 2fd4ecc1b25a..fbd4798834e5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -789,11 +789,27 @@ static int apply_hw_params_minmax(struct snd_interval *it, unsigned int rmin, return changed; } +/* get the specified endpoint object that is being used by other streams + * (i.e. the parameter is locked) + */ +static const struct snd_usb_endpoint * +get_endpoint_in_use(struct snd_usb_audio *chip, int endpoint, + const struct snd_usb_endpoint *ref_ep) +{ + const struct snd_usb_endpoint *ep; + + ep = snd_usb_get_endpoint(chip, endpoint); + if (ep && ep->cur_audiofmt && (ep != ref_ep || ep->opened > 1)) + return ep; + return NULL; +} + static int hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_usb_substream *subs = rule->private; struct snd_usb_audio *chip = subs->stream->chip; + const struct snd_usb_endpoint *ep; const struct audioformat *fp; struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); unsigned int rmin, rmax, r; @@ -805,6 +821,29 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; + + ep = get_endpoint_in_use(chip, fp->endpoint, + subs->data_endpoint); + if (ep) { + hwc_debug("rate limit %d for ep#%x\n", + ep->cur_rate, fp->endpoint); + rmin = min(rmin, ep->cur_rate); + rmax = max(rmax, ep->cur_rate); + continue; + } + + if (fp->implicit_fb) { + ep = get_endpoint_in_use(chip, fp->sync_ep, + subs->sync_endpoint); + if (ep) { + hwc_debug("rate limit %d for sync_ep#%x\n", + ep->cur_rate, fp->sync_ep); + rmin = min(rmin, ep->cur_rate); + rmax = max(rmax, ep->cur_rate); + continue; + } + } + r = snd_usb_endpoint_get_clock_rate(chip, fp->clock); if (r > 0) { if (!snd_interval_test(it, r)) @@ -874,6 +913,8 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_usb_substream *subs = rule->private; + struct snd_usb_audio *chip = subs->stream->chip; + const struct snd_usb_endpoint *ep; const struct audioformat *fp; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); u64 fbits; @@ -883,6 +924,27 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; + + ep = get_endpoint_in_use(chip, fp->endpoint, + subs->data_endpoint); + if (ep) { + hwc_debug("format limit %d for ep#%x\n", + ep->cur_format, fp->endpoint); + fbits |= pcm_format_to_bits(ep->cur_format); + continue; + } + + if (fp->implicit_fb) { + ep = get_endpoint_in_use(chip, fp->sync_ep, + subs->sync_endpoint); + if (ep) { + hwc_debug("format limit %d for sync_ep#%x\n", + ep->cur_format, fp->sync_ep); + fbits |= pcm_format_to_bits(ep->cur_format); + continue; + } + } + fbits |= fp->formats; } return apply_hw_params_format_bits(fmt, fbits); @@ -915,103 +977,95 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, pmin, UINT_MAX); } -/* get the EP or the sync EP for implicit fb when it's already set up */ -static const struct snd_usb_endpoint * -get_sync_ep_from_substream(struct snd_usb_substream *subs) +/* additional hw constraints for implicit feedback mode */ +static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { + struct snd_usb_substream *subs = rule->private; struct snd_usb_audio *chip = subs->stream->chip; const struct audioformat *fp; const struct snd_usb_endpoint *ep; + struct snd_interval *it; + unsigned int rmin, rmax; + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + hwc_debug("hw_rule_period_size: (%u,%u)\n", it->min, it->max); + rmin = UINT_MAX; + rmax = 0; list_for_each_entry(fp, &subs->fmt_list, list) { - ep = snd_usb_get_endpoint(chip, fp->endpoint); - if (ep && ep->cur_audiofmt) { - /* if EP is already opened solely for this substream, - * we still allow us to change the parameter; otherwise - * this substream has to follow the existing parameter - */ - if (ep->cur_audiofmt != subs->cur_audiofmt || ep->opened > 1) - return ep; - } - if (!fp->implicit_fb) + if (!hw_check_valid_format(subs, params, fp)) + continue; + ep = get_endpoint_in_use(chip, fp->endpoint, + subs->data_endpoint); + if (ep) { + hwc_debug("period size limit %d for ep#%x\n", + ep->cur_period_frames, fp->endpoint); + rmin = min(rmin, ep->cur_period_frames); + rmax = max(rmax, ep->cur_period_frames); continue; - /* for the implicit fb, check the sync ep as well */ - ep = snd_usb_get_endpoint(chip, fp->sync_ep); - if (ep && ep->cur_audiofmt) { - /* ditto, if the sync (data) ep is used by others, - * this stream is restricted by the sync ep - */ - if (ep != subs->sync_endpoint || ep->opened > 1) - return ep; } - } - return NULL; -} -/* additional hw constraints for implicit feedback mode */ -static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - const struct snd_usb_endpoint *ep; - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - ep = get_sync_ep_from_substream(subs); - if (!ep) - return 0; - - hwc_debug("applying %s\n", __func__); - return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format)); -} - -static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - const struct snd_usb_endpoint *ep; - struct snd_interval *it; - - ep = get_sync_ep_from_substream(subs); - if (!ep) - return 0; - - hwc_debug("applying %s\n", __func__); - it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate); -} - -static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - const struct snd_usb_endpoint *ep; - struct snd_interval *it; - - ep = get_sync_ep_from_substream(subs); - if (!ep) - return 0; + if (fp->implicit_fb) { + ep = get_endpoint_in_use(chip, fp->sync_ep, + subs->sync_endpoint); + if (ep) { + hwc_debug("period size limit %d for sync_ep#%x\n", + ep->cur_period_frames, fp->sync_ep); + rmin = min(rmin, ep->cur_period_frames); + rmax = max(rmax, ep->cur_period_frames); + continue; + } + } + } - hwc_debug("applying %s\n", __func__); - it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); - return apply_hw_params_minmax(it, ep->cur_period_frames, - ep->cur_period_frames); + if (!rmax) + return 0; /* no limit by implicit fb */ + return apply_hw_params_minmax(it, rmin, rmax); } static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_usb_substream *subs = rule->private; + struct snd_usb_audio *chip = subs->stream->chip; + const struct audioformat *fp; const struct snd_usb_endpoint *ep; struct snd_interval *it; + unsigned int rmin, rmax; - ep = get_sync_ep_from_substream(subs); - if (!ep) - return 0; - - hwc_debug("applying %s\n", __func__); it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS); - return apply_hw_params_minmax(it, ep->cur_buffer_periods, - ep->cur_buffer_periods); + hwc_debug("hw_rule_periods: (%u,%u)\n", it->min, it->max); + rmin = UINT_MAX; + rmax = 0; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) + continue; + ep = get_endpoint_in_use(chip, fp->endpoint, + subs->data_endpoint); + if (ep) { + hwc_debug("periods limit %d for ep#%x\n", + ep->cur_buffer_periods, fp->endpoint); + rmin = min(rmin, ep->cur_buffer_periods); + rmax = max(rmax, ep->cur_buffer_periods); + continue; + } + + if (fp->implicit_fb) { + ep = get_endpoint_in_use(chip, fp->sync_ep, + subs->sync_endpoint); + if (ep) { + hwc_debug("periods limit %d for sync_ep#%x\n", + ep->cur_buffer_periods, fp->sync_ep); + rmin = min(rmin, ep->cur_buffer_periods); + rmax = max(rmax, ep->cur_buffer_periods); + continue; + } + } + } + + if (!rmax) + return 0; /* no limit by implicit fb */ + return apply_hw_params_minmax(it, rmin, rmax); } /* @@ -1120,16 +1174,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; /* additional hw constraints for implicit fb */ - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format_implicit_fb, subs, - SNDRV_PCM_HW_PARAM_FORMAT, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - hw_rule_rate_implicit_fb, subs, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, hw_rule_period_size_implicit_fb, subs, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); -- cgit v1.2.3-58-ga151 From 16f1f838442dc6430d32d51ddda347b8421ec34b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jan 2023 16:09:44 +0100 Subject: Revert "ALSA: usb-audio: Drop superfluous interface setup at parsing" This reverts commit ac5e2fb425e1121ceef2b9d1b3ffccc195d55707. The commit caused a regression on Behringer UMC404HD (and likely others). As the change was meant only as a minor optimization, it's better to revert it to address the regression. Reported-and-tested-by: Michael Ralston Cc: Link: https://lore.kernel.org/r/CAC2975JXkS1A5Tj9b02G_sy25ZWN-ys+tc9wmkoS=qPgKCogSg@mail.gmail.com Link: https://lore.kernel.org/r/20230104150944.24918-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/usb/stream.c b/sound/usb/stream.c index f75601ca2d52..f10f4e6d3fb8 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1222,6 +1222,12 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, if (err < 0) return err; } + + /* try to set the interface... */ + usb_set_interface(chip->dev, iface_no, 0); + snd_usb_init_pitch(chip, fp); + snd_usb_init_sample_rate(chip, fp, fp->rate_max); + usb_set_interface(chip->dev, iface_no, altno); } return 0; } -- cgit v1.2.3-58-ga151 From 9c694fbfe6f36017b060ad74c7565cb379852e40 Mon Sep 17 00:00:00 2001 From: Jeremy Szu Date: Thu, 5 Jan 2023 12:41:53 +0800 Subject: ALSA: hda/realtek: fix mute/micmute LEDs don't work for a HP platform There is a HP platform uses ALC236 codec which using GPIO2 to control mute LED and GPIO1 to control micmute LED. Thus, add a quirk to make them work. Signed-off-by: Jeremy Szu Cc: Link: https://lore.kernel.org/r/20230105044154.8242-1-jeremy.szu@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eda4914b8aeb..8362eb4642d8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9409,6 +9409,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v1.2.3-58-ga151 From cdfa92eb90f5770b26a79824ef213ebdbbd988b1 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Wed, 4 Jan 2023 10:57:54 +0800 Subject: ASoC: fsl_micfil: Correct the number of steps on SX controls The parameter "max" of SOC_SINGLE_SX_TLV() means the number of steps rather than maximum value. This patch corrects the minimum value to -8 and the number of steps to 15. Signed-off-by: Chancel Liu Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20230104025754.3019235-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 7b17f152bbf3..94341e4352b3 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -315,21 +315,21 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(1), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(2), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(3), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(4), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(5), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, - MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0x7, gain_tlv), + MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv), SOC_ENUM_EXT("MICFIL Quality Select", fsl_micfil_quality_enum, micfil_quality_get, micfil_quality_set), -- cgit v1.2.3-58-ga151 From 15a59cb0a3d6ddf2cb79f8dc3081b3130aad3767 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Jan 2023 10:35:30 +0100 Subject: ALSA: hda: cs35l41: Don't return -EINVAL from system suspend/resume The recent commit to support the system suspend for CS35L41 caused a regression on the models with CS35L41_EXT_BOOST_NO_VSPK_SWITC boost type, as the suspend/resume callbacks just return -EINVAL. This is eventually handled as a fatal error and blocks the whole system suspend/resume. For avoiding the problem, this patch corrects the return code from cs35l41_system_suspend() and _resume() to 0, and replace dev_err() with dev_err_once() for stop spamming too much. Fixes: 88672826e2a4 ("ALSA: hda: cs35l41: Support System Suspend") Cc: Link: https://lore.kernel.org/all/e6751ac2-34f3-d13f-13db-8174fade8308@pm.me Link: https://lore.kernel.org/r/20230105093531.16960-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 91842c0c8c74..0a5cee730268 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -598,8 +598,8 @@ static int cs35l41_system_suspend(struct device *dev) dev_dbg(cs35l41->dev, "System Suspend\n"); if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) { - dev_err(cs35l41->dev, "System Suspend not supported\n"); - return -EINVAL; + dev_err_once(cs35l41->dev, "System Suspend not supported\n"); + return 0; /* don't block the whole system suspend */ } ret = pm_runtime_force_suspend(dev); @@ -624,8 +624,8 @@ static int cs35l41_system_resume(struct device *dev) dev_dbg(cs35l41->dev, "System Resume\n"); if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) { - dev_err(cs35l41->dev, "System Resume not supported\n"); - return -EINVAL; + dev_err_once(cs35l41->dev, "System Resume not supported\n"); + return 0; /* don't block the whole system resume */ } if (cs35l41->reset_gpio) { -- cgit v1.2.3-58-ga151 From ae50e2ab122cef68f46b7799fb9deffe3334f5e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Jan 2023 10:35:31 +0100 Subject: ALSA: hda: cs35l41: Check runtime suspend capability at runtime_idle The runtime PM core checks with runtime_idle callback whether it can goes to the runtime suspend or not, and we can put the boost type check there instead of runtime_suspend and _resume calls. This will reduce the unnecessary runtime_suspend() calls. Fixes: 1873ebd30cc8 ("ALSA: hda: cs35l41: Support Hibernation during Suspend") Cc: Link: https://lore.kernel.org/r/20230105093531.16960-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 0a5cee730268..f7815ee24f83 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -647,6 +647,15 @@ static int cs35l41_system_resume(struct device *dev) return ret; } +static int cs35l41_runtime_idle(struct device *dev) +{ + struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + + if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) + return -EBUSY; /* suspend not supported yet on this model */ + return 0; +} + static int cs35l41_runtime_suspend(struct device *dev) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); @@ -1536,7 +1545,8 @@ void cs35l41_hda_remove(struct device *dev) EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); const struct dev_pm_ops cs35l41_hda_pm_ops = { - RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) + RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, + cs35l41_runtime_idle) SYSTEM_SLEEP_PM_OPS(cs35l41_system_suspend, cs35l41_system_resume) }; EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41); -- cgit v1.2.3-58-ga151 From 0cbf1ecd8c4801ec7566231491f7ad9cec31098b Mon Sep 17 00:00:00 2001 From: Stephan Gerhold Date: Sat, 31 Dec 2022 12:55:06 +0100 Subject: ASoC: qcom: Fix building APQ8016 machine driver without SOUNDWIRE Older Qualcomm platforms like APQ8016 do not have hardware support for SoundWire, so kernel configurations made specifically for those platforms will usually not have CONFIG_SOUNDWIRE enabled. Unfortunately commit 8d89cf6ff229 ("ASoC: qcom: cleanup and fix dependency of QCOM_COMMON") breaks those kernel configurations, because SOUNDWIRE is now a required dependency for SND_SOC_QCOM_COMMON (and in turn also SND_SOC_APQ8016_SBC). Trying to migrate such a kernel config silently disables SND_SOC_APQ8016_SBC and breaks audio functionality. The soundwire helpers in common.c are only used by two of the Qualcomm audio machine drivers, so building and requiring CONFIG_SOUNDWIRE for all platforms is unnecessary. There is no need to stuff all common code into a single module. Fix the issue by moving the soundwire helpers to a separate SND_SOC_QCOM_SDW module/option that is selected only by the machine drivers that make use of them. This also allows reverting the imply/depends changes from the previous fix because both SM8250 and SC8280XP already depend on SOUNDWIRE, so the soundwire helpers will be only built if SOUNDWIRE is really enabled. Cc: Srinivas Kandagatla Fixes: 8d89cf6ff229 ("ASoC: qcom: cleanup and fix dependency of QCOM_COMMON") Signed-off-by: Stephan Gerhold Link: https://lore.kernel.org/r/20221231115506.82991-1-stephan@gerhold.net Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 21 ++++---- sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 114 ------------------------------------------ sound/soc/qcom/common.h | 10 ---- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sdw.c | 123 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/sdw.h | 18 +++++++ sound/soc/qcom/sm8250.c | 1 + 8 files changed, 157 insertions(+), 133 deletions(-) create mode 100644 sound/soc/qcom/sdw.c create mode 100644 sound/soc/qcom/sdw.h (limited to 'sound') diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 96a6d4731e6f..e7b00d1d9e99 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -2,7 +2,6 @@ menuconfig SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST - imply SND_SOC_QCOM_COMMON help Say Y or M if you want to add support to use audio devices in Qualcomm Technologies SOC-based platforms. @@ -60,14 +59,16 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" select SND_SOC_LPASS_APQ8016 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8016 SOC-based systems. Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON - depends on SOUNDWIRE + tristate + +config SND_SOC_QCOM_SDW tristate config SND_SOC_QDSP6_COMMON @@ -144,7 +145,7 @@ config SND_SOC_MSM8996 depends on QCOM_APR depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. @@ -155,7 +156,7 @@ config SND_SOC_SDM845 depends on QCOM_APR && I2C && SOUNDWIRE depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 imply SND_SOC_CROS_EC_CODEC @@ -169,7 +170,8 @@ config SND_SOC_SM8250 depends on QCOM_APR && SOUNDWIRE depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW help To add support for audio on Qualcomm Technologies Inc. SM8250 SoC-based systems. @@ -180,7 +182,8 @@ config SND_SOC_SC8280XP depends on QCOM_APR && SOUNDWIRE depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW help To add support for audio on Qualcomm Technologies Inc. SC8280XP SoC-based systems. @@ -190,7 +193,7 @@ config SND_SOC_SC7180 tristate "SoC Machine driver for SC7180 boards" depends on I2C && GPIOLIB depends on SOUNDWIRE || SOUNDWIRE=n - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A select SND_SOC_RT5682_I2C @@ -204,7 +207,7 @@ config SND_SOC_SC7180 config SND_SOC_SC7280 tristate "SoC Machine driver for SC7280 boards" depends on I2C && SOUNDWIRE - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X_SDW diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 8b97172cf990..254350d9dc06 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -28,6 +28,7 @@ snd-soc-sdm845-objs := sdm845.o snd-soc-sm8250-objs := sm8250.o snd-soc-sc8280xp-objs := sc8280xp.o snd-soc-qcom-common-objs := common.o +snd-soc-qcom-sdw-objs := sdw.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_SOC_SC8280XP) += snd-soc-sc8280xp.o obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o obj-$(CONFIG_SND_SOC_SM8250) += snd-soc-sm8250.o obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o +obj-$(CONFIG_SND_SOC_QCOM_SDW) += snd-soc-qcom-sdw.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 49c74c1662a3..96fe80241fb4 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -180,120 +180,6 @@ err_put_np: } EXPORT_SYMBOL_GPL(qcom_snd_parse_of); -int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *sruntime, - bool *stream_prepared) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int ret; - - if (!sruntime) - return 0; - - switch (cpu_dai->id) { - case WSA_CODEC_DMA_RX_0: - case WSA_CODEC_DMA_RX_1: - case RX_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_1: - case TX_CODEC_DMA_TX_0: - case TX_CODEC_DMA_TX_1: - case TX_CODEC_DMA_TX_2: - case TX_CODEC_DMA_TX_3: - break; - default: - return 0; - } - - if (*stream_prepared) { - sdw_disable_stream(sruntime); - sdw_deprepare_stream(sruntime); - *stream_prepared = false; - } - - ret = sdw_prepare_stream(sruntime); - if (ret) - return ret; - - /** - * NOTE: there is a strict hw requirement about the ordering of port - * enables and actual WSA881x PA enable. PA enable should only happen - * after soundwire ports are enabled if not DC on the line is - * accumulated resulting in Click/Pop Noise - * PA enable/mute are handled as part of codec DAPM and digital mute. - */ - - ret = sdw_enable_stream(sruntime); - if (ret) { - sdw_deprepare_stream(sruntime); - return ret; - } - *stream_prepared = true; - - return ret; -} -EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare); - -int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct sdw_stream_runtime **psruntime) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - struct sdw_stream_runtime *sruntime; - int i; - - switch (cpu_dai->id) { - case WSA_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_1: - case TX_CODEC_DMA_TX_0: - case TX_CODEC_DMA_TX_1: - case TX_CODEC_DMA_TX_2: - case TX_CODEC_DMA_TX_3: - for_each_rtd_codec_dais(rtd, i, codec_dai) { - sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream); - if (sruntime != ERR_PTR(-ENOTSUPP)) - *psruntime = sruntime; - } - break; - } - - return 0; - -} -EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params); - -int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *sruntime, bool *stream_prepared) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - - switch (cpu_dai->id) { - case WSA_CODEC_DMA_RX_0: - case WSA_CODEC_DMA_RX_1: - case RX_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_1: - case TX_CODEC_DMA_TX_0: - case TX_CODEC_DMA_TX_1: - case TX_CODEC_DMA_TX_2: - case TX_CODEC_DMA_TX_3: - if (sruntime && *stream_prepared) { - sdw_disable_stream(sruntime); - sdw_deprepare_stream(sruntime); - *stream_prepared = false; - } - break; - default: - break; - } - - return 0; -} -EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free); - int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_jack *jack, bool *jack_setup) { diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h index 3ef5bb6d12df..d7f80ee5ae26 100644 --- a/sound/soc/qcom/common.h +++ b/sound/soc/qcom/common.h @@ -5,19 +5,9 @@ #define __QCOM_SND_COMMON_H__ #include -#include int qcom_snd_parse_of(struct snd_soc_card *card); int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_jack *jack, bool *jack_setup); -int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *runtime, - bool *stream_prepared); -int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct sdw_stream_runtime **psruntime); -int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *sruntime, - bool *stream_prepared); #endif diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index ade44ad7c585..14d9fea33d16 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -12,6 +12,7 @@ #include #include "qdsp6/q6afe.h" #include "common.h" +#include "sdw.h" #define DRIVER_NAME "sc8280xp" diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c new file mode 100644 index 000000000000..10249519a39e --- /dev/null +++ b/sound/soc/qcom/sdw.c @@ -0,0 +1,123 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#include +#include +#include "qdsp6/q6afe.h" +#include "sdw.h" + +int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *sruntime, + bool *stream_prepared) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int ret; + + if (!sruntime) + return 0; + + switch (cpu_dai->id) { + case WSA_CODEC_DMA_RX_0: + case WSA_CODEC_DMA_RX_1: + case RX_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_1: + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + break; + default: + return 0; + } + + if (*stream_prepared) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + *stream_prepared = false; + } + + ret = sdw_prepare_stream(sruntime); + if (ret) + return ret; + + /** + * NOTE: there is a strict hw requirement about the ordering of port + * enables and actual WSA881x PA enable. PA enable should only happen + * after soundwire ports are enabled if not DC on the line is + * accumulated resulting in Click/Pop Noise + * PA enable/mute are handled as part of codec DAPM and digital mute. + */ + + ret = sdw_enable_stream(sruntime); + if (ret) { + sdw_deprepare_stream(sruntime); + return ret; + } + *stream_prepared = true; + + return ret; +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare); + +int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sdw_stream_runtime **psruntime) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime; + int i; + + switch (cpu_dai->id) { + case WSA_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_1: + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + for_each_rtd_codec_dais(rtd, i, codec_dai) { + sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream); + if (sruntime != ERR_PTR(-ENOTSUPP)) + *psruntime = sruntime; + } + break; + } + + return 0; + +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params); + +int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *sruntime, bool *stream_prepared) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + + switch (cpu_dai->id) { + case WSA_CODEC_DMA_RX_0: + case WSA_CODEC_DMA_RX_1: + case RX_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_1: + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + if (sruntime && *stream_prepared) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + *stream_prepared = false; + } + break; + default: + break; + } + + return 0; +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/sdw.h b/sound/soc/qcom/sdw.h new file mode 100644 index 000000000000..d74cbb84da13 --- /dev/null +++ b/sound/soc/qcom/sdw.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#ifndef __QCOM_SND_SDW_H__ +#define __QCOM_SND_SDW_H__ + +#include + +int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *runtime, + bool *stream_prepared); +int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sdw_stream_runtime **psruntime); +int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *sruntime, + bool *stream_prepared); +#endif diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 8dbe9ef41b1c..9626a9ef78c2 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -12,6 +12,7 @@ #include #include "qdsp6/q6afe.h" #include "common.h" +#include "sdw.h" #define DRIVER_NAME "sm8250" #define MI2S_BCLK_RATE 1536000 -- cgit v1.2.3-58-ga151 From 291e9da91403e0e628d7692b5ed505100e7b7706 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 9 Jan 2023 15:11:33 +0100 Subject: ALSA: usb-audio: Always initialize fixed_rate in snd_usb_find_implicit_fb_sync_format() Handle the fallback code path, too. Fixes: fd28941cff1c ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless") BugLink: https://lore.kernel.org/alsa-devel/Y7frf3N%2FxzvESEsN@kili/ Reported-by: Dan Carpenter Cc: Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20230109141133.335543-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 3 ++- sound/usb/pcm.c | 2 ++ 2 files changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 41ac7185b42b..4727043fd745 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -471,7 +471,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, subs = find_matching_substream(chip, stream, target->sync_ep, target->fmt_type); if (!subs) - return sync_fmt; + goto end; high_score = 0; list_for_each_entry(fp, &subs->fmt_list, list) { @@ -485,6 +485,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, } } + end: if (fixed_rate) *fixed_rate = snd_usb_pcm_has_fixed_rate(subs); return sync_fmt; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index fbd4798834e5..1f72960d0d53 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -163,6 +163,8 @@ bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs) struct snd_usb_audio *chip = subs->stream->chip; int rate = -1; + if (!subs) + return false; if (!(chip->quirk_flags & QUIRK_FLAG_FIXED_RATE)) return false; list_for_each_entry(fp, &subs->fmt_list, list) { -- cgit v1.2.3-58-ga151 From 70051cffb31b5ee09096351c3b41fcae6f89de31 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 9 Jan 2023 16:12:49 +0100 Subject: ALSA: control-led: use strscpy in set_led_id() The use of strncpy() in the set_led_id() was incorrect. The len variable should use 'min(sizeof(buf2) - 1, count)' expression. Use strscpy() function to simplify things and handle the error gracefully. Fixes: a135dfb5de15 ("ALSA: led control - add sysfs kcontrol LED marking layer") Reported-by: yang.yang29@zte.com.cn Link: https://lore.kernel.org/alsa-devel/202301091945513559977@zte.com.cn/ Cc: Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/control_led.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/control_led.c b/sound/core/control_led.c index f975cc85772b..3cadd40100f3 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -530,12 +530,11 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si bool attach) { char buf2[256], *s, *os; - size_t len = max(sizeof(s) - 1, count); struct snd_ctl_elem_id id; int err; - strncpy(buf2, buf, len); - buf2[len] = '\0'; + if (strscpy(buf2, buf, sizeof(buf2)) < 0) + return -E2BIG; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; s = buf2; -- cgit v1.2.3-58-ga151 From de1ccb9e61728dd941fe0e955a7a129418657267 Mon Sep 17 00:00:00 2001 From: Adrian Chan Date: Mon, 9 Jan 2023 16:05:20 -0500 Subject: ALSA: hda/hdmi: Add a HP device 0x8715 to force connect list Add the 'HP Engage Flex Mini' device to the force connect list to enable audio through HDMI. Signed-off-by: Adrian Chan Cc: Link: https://lore.kernel.org/r/20230109210520.16060-1-adchan@google.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 386dd9d9143f..9ea633fe9339 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1981,6 +1981,7 @@ static const struct snd_pci_quirk force_connect_list[] = { SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1), SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), SND_PCI_QUIRK(0x103c, 0x8711, "HP", 1), + SND_PCI_QUIRK(0x103c, 0x8715, "HP", 1), SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1), {} -- cgit v1.2.3-58-ga151 From 8c6a42b5b0ed6f96624f56954e93eeae107440a6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Jan 2023 23:15:06 +0000 Subject: ASoC: fsl_ssi: Rename AC'97 streams to avoid collisions with AC'97 CODEC The SSI driver calls the AC'97 playback and transmit streams "AC97 Playback" and "AC97 Capture" respectively. This is the same name used by the generic AC'97 CODEC driver in ASoC, creating confusion for the Freescale ASoC card when it attempts to use these widgets in routing. Add a "CPU" in the name like the regular DAIs registered by the driver to disambiguate. Acked-by: Shengjiu Wang Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20230106-asoc-udoo-probe-v1-1-a5d7469d4f67@kernel.org Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++---- sound/soc/fsl/fsl_ssi.c | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c836848ef0a6..1dfd0341e487 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -121,11 +121,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static const struct snd_soc_dapm_route audio_map_ac97[] = { /* 1st half -- Normal DAPM routes */ - {"Playback", NULL, "AC97 Playback"}, - {"AC97 Capture", NULL, "Capture"}, + {"Playback", NULL, "CPU AC97 Playback"}, + {"CPU AC97 Capture", NULL, "Capture"}, /* 2nd half -- ASRC DAPM routes */ - {"AC97 Playback", NULL, "ASRC-Playback"}, - {"ASRC-Capture", NULL, "AC97 Capture"}, + {"CPU AC97 Playback", NULL, "ASRC-Playback"}, + {"ASRC-Capture", NULL, "CPU AC97 Capture"}, }; static const struct snd_soc_dapm_route audio_map_tx[] = { diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c9e0e31d5b34..46a53551b955 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1189,14 +1189,14 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .symmetric_channels = 1, .probe = fsl_ssi_dai_probe, .playback = { - .stream_name = "AC97 Playback", + .stream_name = "CPU AC97 Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20, }, .capture = { - .stream_name = "AC97 Capture", + .stream_name = "CPU AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, -- cgit v1.2.3-58-ga151 From 242fc66ae6e1e2b8519daacc7590a73cd0e8a6e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Jan 2023 23:15:07 +0000 Subject: ASoC: fsl-asoc-card: Fix naming of AC'97 CODEC widgets The fsl-asoc-card AC'97 support currently tries to route to Playback and Capture widgets provided by the AC'97 CODEC. This doesn't work since the generic AC'97 driver registers with an "AC97" at the front of the stream and hence widget names, update to reflect reality. It's not clear to me if or how this ever worked. Acked-by: Shengjiu Wang Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20230106-asoc-udoo-probe-v1-2-a5d7469d4f67@kernel.org Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 1dfd0341e487..8d14b5593658 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -121,8 +121,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static const struct snd_soc_dapm_route audio_map_ac97[] = { /* 1st half -- Normal DAPM routes */ - {"Playback", NULL, "CPU AC97 Playback"}, - {"CPU AC97 Capture", NULL, "Capture"}, + {"AC97 Playback", NULL, "CPU AC97 Playback"}, + {"CPU AC97 Capture", NULL, "AC97 Capture"}, /* 2nd half -- ASRC DAPM routes */ {"CPU AC97 Playback", NULL, "ASRC-Playback"}, {"ASRC-Capture", NULL, "CPU AC97 Capture"}, -- cgit v1.2.3-58-ga151 From ca88eeb308a221c2dcd4a64031d2e5fcd3db9eaa Mon Sep 17 00:00:00 2001 From: Luka Guzenko Date: Tue, 10 Jan 2023 21:25:14 +0100 Subject: ALSA: hda/realtek: Enable mute/micmute LEDs on HP Spectre x360 13-aw0xxx The HP Spectre x360 13-aw0xxx devices use the ALC285 codec with GPIO 0x04 controlling the micmute LED and COEF 0x0b index 8 controlling the mute LED. A quirk was added to make these work as well as a fixup. Signed-off-by: Luka Guzenko Cc: Link: https://lore.kernel.org/r/20230110202514.2792-1-l.guzenko@web.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8362eb4642d8..6fab7c8fc19a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4646,6 +4646,16 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, } } +static void alc285_fixup_hp_gpio_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->micmute_led_polarity = 1; + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); +} + static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4667,6 +4677,13 @@ static void alc285_fixup_hp_mute_led(struct hda_codec *codec, alc285_fixup_hp_coef_micmute_led(codec, fix, action); } +static void alc285_fixup_hp_spectre_x360_mute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc285_fixup_hp_mute_led_coefbit(codec, fix, action); + alc285_fixup_hp_gpio_micmute_led(codec, fix, action); +} + static void alc236_fixup_hp_mute_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -7108,6 +7125,7 @@ enum { ALC285_FIXUP_ASUS_G533Z_PINS, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, + ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED, ALC236_FIXUP_HP_GPIO_LED, ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, @@ -8488,6 +8506,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_mute_led, }, + [ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360_mute_led, + }, [ALC236_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_gpio_led, @@ -9330,6 +9352,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), -- cgit v1.2.3-58-ga151 From 92a9c0ad86d47ff4cce899012e355c400f02cfb8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 13 Jan 2023 09:53:11 +0100 Subject: ALSA: usb-audio: Fix possible NULL pointer dereference in snd_usb_pcm_has_fixed_rate() The subs function argument may be NULL, so do not use it before the NULL check. Fixes: 291e9da91403 ("ALSA: usb-audio: Always initialize fixed_rate in snd_usb_find_implicit_fb_sync_format()") Reported-by: coverity-bot Link: https://lore.kernel.org/alsa-devel/202301121424.4A79A485@keescook/ Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20230113085311.623325-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 1f72960d0d53..d959da7a1afb 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -160,11 +160,12 @@ find_substream_format(struct snd_usb_substream *subs, bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs) { const struct audioformat *fp; - struct snd_usb_audio *chip = subs->stream->chip; + struct snd_usb_audio *chip; int rate = -1; if (!subs) return false; + chip = subs->stream->chip; if (!(chip->quirk_flags & QUIRK_FLAG_FIXED_RATE)) return false; list_for_each_entry(fp, &subs->fmt_list, list) { -- cgit v1.2.3-58-ga151 From 56b88b50565cd8b946a2d00b0c83927b7ebb055e Mon Sep 17 00:00:00 2001 From: Clement Lecigne Date: Fri, 13 Jan 2023 13:07:45 +0100 Subject: ALSA: pcm: Move rwsem lock inside snd_ctl_elem_read to prevent UAF Takes rwsem lock inside snd_ctl_elem_read instead of snd_ctl_elem_read_user like it was done for write in commit 1fa4445f9adf1 ("ALSA: control - introduce snd_ctl_notify_one() helper"). Doing this way we are also fixing the following locking issue happening in the compat path which can be easily triggered and turned into an use-after-free. 64-bits: snd_ctl_ioctl snd_ctl_elem_read_user [takes controls_rwsem] snd_ctl_elem_read [lock properly held, all good] [drops controls_rwsem] 32-bits: snd_ctl_ioctl_compat snd_ctl_elem_write_read_compat ctl_elem_write_read snd_ctl_elem_read [missing lock, not good] CVE-2023-0266 was assigned for this issue. Cc: stable@kernel.org # 5.13+ Signed-off-by: Clement Lecigne Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20230113120745.25464-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 50e7ba66f187..82aa1af1d1d8 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1203,14 +1203,19 @@ static int snd_ctl_elem_read(struct snd_card *card, const u32 pattern = 0xdeadbeef; int ret; + down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &control->id); - if (kctl == NULL) - return -ENOENT; + if (kctl == NULL) { + ret = -ENOENT; + goto unlock; + } index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL) - return -EPERM; + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL) { + ret = -EPERM; + goto unlock; + } snd_ctl_build_ioff(&control->id, kctl, index_offset); @@ -1220,7 +1225,7 @@ static int snd_ctl_elem_read(struct snd_card *card, info.id = control->id; ret = __snd_ctl_elem_info(card, kctl, &info, NULL); if (ret < 0) - return ret; + goto unlock; #endif if (!snd_ctl_skip_validation(&info)) @@ -1230,7 +1235,7 @@ static int snd_ctl_elem_read(struct snd_card *card, ret = kctl->get(kctl, control); snd_power_unref(card); if (ret < 0) - return ret; + goto unlock; if (!snd_ctl_skip_validation(&info) && sanity_check_elem_value(card, control, &info, pattern) < 0) { dev_err(card->dev, @@ -1238,8 +1243,11 @@ static int snd_ctl_elem_read(struct snd_card *card, control->id.iface, control->id.device, control->id.subdevice, control->id.name, control->id.index); - return -EINVAL; + ret = -EINVAL; + goto unlock; } +unlock: + up_read(&card->controls_rwsem); return ret; } @@ -1253,9 +1261,7 @@ static int snd_ctl_elem_read_user(struct snd_card *card, if (IS_ERR(control)) return PTR_ERR(control); - down_read(&card->controls_rwsem); result = snd_ctl_elem_read(card, control); - up_read(&card->controls_rwsem); if (result < 0) goto error; -- cgit v1.2.3-58-ga151