From 75e5fab7db0cecb6e16b22c34608f0b40a4c7cd1 Mon Sep 17 00:00:00 2001 From: Ravulapati Vishnu Vardhan Rao Date: Thu, 11 May 2023 16:55:32 +0530 Subject: ASoC: lpass: Fix for KASAN use_after_free out of bounds When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- cgit v1.2.3 From 8b271370e963370703819bd9795a54d658071bed Mon Sep 17 00:00:00 2001 From: Matthias Kaehlcke Date: Tue, 16 May 2023 16:46:30 +0000 Subject: ASoC: rt5682: Disable jack detection interrupt during suspend The rt5682 driver switches its regmap to cache-only when the device suspends and back to regular mode on resume. When the jack detect interrupt fires rt5682_irq() schedules the jack detect work. This can result in invalid reads from the regmap in cache-only mode if the work runs before the device has resumed: [ 56.245502] rt5682 9-001a: ASoC: error at soc_component_read_no_lock on rt5682.9-001a for register: [0x000000f0] -16 Disable the jack detection interrupt during suspend and re-enable it on resume. The driver already schedules the jack detection work on resume, so any state change during suspend is still handled. This is essentially the same as commit f7d00a9be147 ("SoC: rt5682s: Disable jack detection interrupt during suspend") for the rt5682s. Cc: stable@kernel.org Signed-off-by: Matthias Kaehlcke dev, i2c->irq, NULL, rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5682", rt5682); - if (ret) + if (!ret) + rt5682->irq = i2c->irq; + else dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index f6c798b65c08..5d992543b791 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2959,6 +2959,9 @@ static int rt5682_suspend(struct snd_soc_component *component) if (rt5682->is_sdw) return 0; + if (rt5682->irq) + disable_irq(rt5682->irq); + cancel_delayed_work_sync(&rt5682->jack_detect_work); cancel_delayed_work_sync(&rt5682->jd_check_work); if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { @@ -3027,6 +3030,9 @@ static int rt5682_resume(struct snd_soc_component *component) mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(0)); + if (rt5682->irq) + enable_irq(rt5682->irq); + return 0; } #else diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d568c6993c33..e8efd8a84a6c 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1462,6 +1462,7 @@ struct rt5682_priv { int pll_out[RT5682_PLLS]; int jack_type; + int irq; int irq_work_delay_time; }; -- cgit v1.2.3 From e123036be377ddf628226a7c6d4f9af5efd113d3 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 17 May 2023 13:57:31 -0500 Subject: ASoC: soc-pcm: test if a BE can be prepared In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3042,3 +3045,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- cgit v1.2.3 From 7843380d07bbeffd3ce6504e73cf61f840ae76ca Mon Sep 17 00:00:00 2001 From: Adam Stylinski Date: Sun, 21 May 2023 10:52:23 -0400 Subject: ALSA: hda/ca0132: add quirk for EVGA X299 DARK This quirk is necessary for surround and other DSP effects to work with the onboard ca0132 based audio chipset for the EVGA X299 dark mainboard. Signed-off-by: Adam Stylinski Cc: Link: https://bugzilla.kernel.org/show_bug.cgi?id=67071 Link: https://lore.kernel.org/r/ZGopOe19T1QOwizS@eggsbenedict.adamsnet Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 099722ebaed8..748a3c40966e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1306,6 +1306,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI), + SND_PCI_QUIRK(0x3842, 0x104b, "EVGA X299 Dark", QUIRK_R3DI), SND_PCI_QUIRK(0x3842, 0x1055, "EVGA Z390 DARK", QUIRK_R3DI), SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D), -- cgit v1.2.3 From ff04437f6dcd138b50483afc7b313f016020ce8f Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 19 May 2023 22:17:05 +0200 Subject: ASoC: Intel: avs: Fix module lookup MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When changing value of kcontrol, FW module to which data should be send needs to be found. Currently it is done in improper way, fix it. Change function name to indicate that it looks only for volume module. This allows to change volume during runtime, instead of only changing init value. Fixes: be2b81b519d7 ("ASoC: Intel: avs: Parse control tuples") Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230519201711.4073845-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/control.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/avs/control.c b/sound/soc/intel/avs/control.c index a8b14b784f8a..3dfa2e9816db 100644 --- a/sound/soc/intel/avs/control.c +++ b/sound/soc/intel/avs/control.c @@ -21,17 +21,25 @@ static struct avs_dev *avs_get_kcontrol_adev(struct snd_kcontrol *kcontrol) return to_avs_dev(w->dapm->component->dev); } -static struct avs_path_module *avs_get_kcontrol_module(struct avs_dev *adev, u32 id) +static struct avs_path_module *avs_get_volume_module(struct avs_dev *adev, u32 id) { struct avs_path *path; struct avs_path_pipeline *ppl; struct avs_path_module *mod; - list_for_each_entry(path, &adev->path_list, node) - list_for_each_entry(ppl, &path->ppl_list, node) - list_for_each_entry(mod, &ppl->mod_list, node) - if (mod->template->ctl_id && mod->template->ctl_id == id) + spin_lock(&adev->path_list_lock); + list_for_each_entry(path, &adev->path_list, node) { + list_for_each_entry(ppl, &path->ppl_list, node) { + list_for_each_entry(mod, &ppl->mod_list, node) { + if (guid_equal(&mod->template->cfg_ext->type, &AVS_PEAKVOL_MOD_UUID) + && mod->template->ctl_id == id) { + spin_unlock(&adev->path_list_lock); return mod; + } + } + } + } + spin_unlock(&adev->path_list_lock); return NULL; } @@ -49,7 +57,7 @@ int avs_control_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va /* prevent access to modules while path is being constructed */ mutex_lock(&adev->path_mutex); - active_module = avs_get_kcontrol_module(adev, ctl_data->id); + active_module = avs_get_volume_module(adev, ctl_data->id); if (active_module) { ret = avs_ipc_peakvol_get_volume(adev, active_module->module_id, active_module->instance_id, &dspvols, @@ -89,7 +97,7 @@ int avs_control_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va changed = 1; } - active_module = avs_get_kcontrol_module(adev, ctl_data->id); + active_module = avs_get_volume_module(adev, ctl_data->id); if (active_module) { dspvol.channel_id = AVS_ALL_CHANNELS_MASK; dspvol.target_volume = *volume; -- cgit v1.2.3 From d849996f7458042af803b7d15a181922834c5249 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 19 May 2023 22:17:06 +0200 Subject: ASoC: Intel: avs: Access path components under lock MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Path and its components should be accessed under lock to prevent problems with one thread modifying them while other tries to read. Fixes: c8c960c10971 ("ASoC: Intel: avs: APL-based platforms support") Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230519201711.4073845-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/apl.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c index 02683dce277a..1860099c782a 100644 --- a/sound/soc/intel/avs/apl.c +++ b/sound/soc/intel/avs/apl.c @@ -169,6 +169,7 @@ static bool apl_lp_streaming(struct avs_dev *adev) { struct avs_path *path; + spin_lock(&adev->path_list_lock); /* Any gateway without buffer allocated in LP area disqualifies D0IX. */ list_for_each_entry(path, &adev->path_list, node) { struct avs_path_pipeline *ppl; @@ -188,11 +189,14 @@ static bool apl_lp_streaming(struct avs_dev *adev) if (cfg->copier.dma_type == INVALID_OBJECT_ID) continue; - if (!mod->gtw_attrs.lp_buffer_alloc) + if (!mod->gtw_attrs.lp_buffer_alloc) { + spin_unlock(&adev->path_list_lock); return false; + } } } } + spin_unlock(&adev->path_list_lock); return true; } -- cgit v1.2.3 From 95109657471311601b98e71f03d0244f48dc61bb Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 19 May 2023 22:17:07 +0200 Subject: ASoC: Intel: Skylake: Fix declaration of enum skl_ch_cfg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Constant 'C4_CHANNEL' does not exist on the firmware side. Value 0xC is reserved for 'C7_1' instead. Fixes: 04afbbbb1cba ("ASoC: Intel: Skylake: Update the topology interface structure") Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230519201711.4073845-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- include/uapi/sound/skl-tplg-interface.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index f29899b179a6..4bf9c4f9add8 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -66,7 +66,8 @@ enum skl_ch_cfg { SKL_CH_CFG_DUAL_MONO = 9, SKL_CH_CFG_I2S_DUAL_STEREO_0 = 10, SKL_CH_CFG_I2S_DUAL_STEREO_1 = 11, - SKL_CH_CFG_4_CHANNEL = 12, + SKL_CH_CFG_7_1 = 12, + SKL_CH_CFG_4_CHANNEL = SKL_CH_CFG_7_1, SKL_CH_CFG_INVALID }; -- cgit v1.2.3 From 1cf036deebcdec46d6348842bd2f8931202fd4cd Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 19 May 2023 22:17:08 +0200 Subject: ASoC: Intel: avs: Fix declaration of enum avs_channel_config MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Constant 'C4_CHANNEL' does not exist on the firmware side. Value 0xC is reserved for 'C7_1' instead. Fixes: 580a5912d1fe ("ASoC: Intel: avs: Declare module configuration types") Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230519201711.4073845-5-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/messages.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h index d3b60ae7d743..7f23a304b4a9 100644 --- a/sound/soc/intel/avs/messages.h +++ b/sound/soc/intel/avs/messages.h @@ -619,7 +619,7 @@ enum avs_channel_config { AVS_CHANNEL_CONFIG_DUAL_MONO = 9, AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_0 = 10, AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_1 = 11, - AVS_CHANNEL_CONFIG_4_CHANNEL = 12, + AVS_CHANNEL_CONFIG_7_1 = 12, AVS_CHANNEL_CONFIG_INVALID }; -- cgit v1.2.3 From 836855100b87b4dd7a82546131779dc255c18b67 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 19 May 2023 22:17:09 +0200 Subject: ASoC: Intel: avs: Account for UID of ACPI device MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Configurations with multiple codecs attached to the platform are supported but only if each from the set is different. Add new field representing the 'Unique ID' so that codecs that share Vendor and Part IDs can be differentiated and thus enabling support for such configurations. Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 1 + sound/soc/intel/avs/board_selection.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index b38fd25c5729..528279056b3a 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -170,6 +170,7 @@ struct snd_soc_acpi_link_adr { /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { u8 id[ACPI_ID_LEN]; + const char *uid; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index b2823c2107f7..60f8fb0bff95 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -443,7 +443,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev) } for (mach = boards->machs; mach->id[0]; mach++) { - if (!acpi_dev_present(mach->id, NULL, -1)) + if (!acpi_dev_present(mach->id, mach->uid, -1)) continue; if (mach->machine_quirk) -- cgit v1.2.3 From 320f4d868b83a804e3a4bd61a5b7d0f1db66380e Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 19 May 2023 22:17:10 +0200 Subject: ASoC: Intel: avs: Fix avs_path_module::instance_id size MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit All IPCs using instance_id use 8 bit value. Original commit used 16 bit value because FW reports possible max value in 16 bit field, but in practice FW limits the value to 8 bits. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230519201711.4073845-7-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/avs.h | 4 ++-- sound/soc/intel/avs/dsp.c | 4 ++-- sound/soc/intel/avs/path.h | 2 +- sound/soc/intel/avs/probes.c | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index d7fccdcb9c16..0cf38c9e768e 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -283,8 +283,8 @@ void avs_release_firmwares(struct avs_dev *adev); int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, u8 core_id, u8 domain, void *param, u32 param_size, - u16 *instance_id); -void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id, + u8 *instance_id); +void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id, u8 ppl_instance_id, u8 core_id); int avs_dsp_create_pipeline(struct avs_dev *adev, u16 req_size, u8 priority, bool lp, u16 attributes, u8 *instance_id); diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c index b881100d3e02..aa03af4473e9 100644 --- a/sound/soc/intel/avs/dsp.c +++ b/sound/soc/intel/avs/dsp.c @@ -225,7 +225,7 @@ err: int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, u8 core_id, u8 domain, void *param, u32 param_size, - u16 *instance_id) + u8 *instance_id) { struct avs_module_entry mentry; bool was_loaded = false; @@ -272,7 +272,7 @@ err_mod_entry: return ret; } -void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id, +void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id, u8 ppl_instance_id, u8 core_id) { struct avs_module_entry mentry; diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h index 197222c5e008..657f7b093e80 100644 --- a/sound/soc/intel/avs/path.h +++ b/sound/soc/intel/avs/path.h @@ -37,7 +37,7 @@ struct avs_path_pipeline { struct avs_path_module { u16 module_id; - u16 instance_id; + u8 instance_id; union avs_gtw_attributes gtw_attrs; struct avs_tplg_module *template; diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index 70a94201d6a5..275928281c6c 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -18,7 +18,7 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id { struct avs_probe_cfg cfg = {{0}}; struct avs_module_entry mentry; - u16 dummy; + u8 dummy; avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); -- cgit v1.2.3 From 25148f57a2a6d157779bae494852e172952ba980 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 19 May 2023 22:17:11 +0200 Subject: ASoC: Intel: avs: Add missing checks on FE startup MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Constraint functions have return values, they should be checked for potential errors. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230519201711.4073845-8-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 31c032a0f7e4..1fbb2c2fadb5 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -468,21 +468,34 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST); if (!host_stream) { - kfree(data); - return -EBUSY; + ret = -EBUSY; + goto err; } data->host_stream = host_stream; - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto err; + /* avoid wrap-around with wall-clock */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000); + if (ret < 0) + goto err; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); + if (ret < 0) + goto err; + snd_pcm_set_sync(substream); dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p", __func__, hdac_stream(host_stream)->stream_tag, substream); return 0; + +err: + kfree(data); + return ret; } static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) -- cgit v1.2.3 From 3a2e3fa795052b42da013931bc2e451bcecf4f0c Mon Sep 17 00:00:00 2001 From: David Epping Date: Fri, 19 May 2023 13:05:53 +0200 Subject: ASoC: dt-bindings: tlv320aic32x4: Fix supply names The term "-supply" is a suffix to regulator names. Signed-off-by: David Epping Link: https://lore.kernel.org/r/20230519110545.GA18663@nucnuc.mle Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tlv320aic32x4.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt index f59125bc79d1..0b4e21bde5bc 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -8,7 +8,7 @@ Required properties: "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 "ti,tas2505" TAS2505, TAS2521 - reg: I2C slave address - - supply-*: Required supply regulators are: + - *-supply: Required supply regulators are: "iov" - digital IO power supply "ldoin" - LDO power supply "dv" - Digital core power supply -- cgit v1.2.3 From 81302b1c7c997e8a56c1c2fc63a296ebeb0cd2d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 18 May 2023 13:35:20 +0200 Subject: ALSA: hda: Fix unhandled register update during auto-suspend period MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit It's reported that the recording started right after the driver probe doesn't work properly, and it turned out that this is related with the codec auto-suspend. Namely, after the probe phase, the usage count goes zero, and the auto-suspend is programmed, but the codec is kept still active until the auto-suspend expiration. When an application (e.g. alsactl) updates the mixer values at this moment, the values are cached but not actually written. Then, starting arecord thereafter also results in the silence because of the missing unmute. The root cause is the handling of "lazy update" mode; when a mixer value is updated *after* the suspend, it should update only the cache and exits. At the resume, the cached value is written to the device, in turn. The problem is that the current code misinterprets the state of auto-suspend as if it were already suspended. Although we can add the check of the actual device state after pm_runtime_get_if_in_use() for catching the missing state, this won't suffice; the second call of regmap_update_bits_check() will skip writing the register because the cache has been already updated by the first call. So we'd need fixes in two different places. OTOH, a simpler fix is to replace pm_runtime_get_if_in_use() with pm_runtime_get_if_active() (with ign_usage_count=true). This change implies that the driver takes the pm refcount if the device is still in ACTIVE state and continues the processing. A small caveat is that this will leave the auto-suspend timer. But, since the timer callback itself checks the device state and aborts gracefully when it's active, this won't be any substantial problem. Long story short: we address the missing register-write problem just by replacing the pm_runtime_*() call in snd_hda_keep_power_up(). Fixes: fc4f000bf8c0 ("ALSA: hda - Fix unexpected resume through regmap code path") Reported-by: Amadeusz Sławiński Closes: https://lore.kernel.org/r/a7478636-af11-92ab-731c-9b13c582a70d@linux.intel.com Suggested-by: Cezary Rojewski Cc: Link: https://lore.kernel.org/r/20230518113520.15213-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index accc9d279ce5..6c043fbd606f 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -611,7 +611,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); int snd_hdac_keep_power_up(struct hdac_device *codec) { if (!atomic_inc_not_zero(&codec->in_pm)) { - int ret = pm_runtime_get_if_in_use(&codec->dev); + int ret = pm_runtime_get_if_active(&codec->dev, true); if (!ret) return -1; if (ret < 0) -- cgit v1.2.3 From e2d035f5a7d597bbabc268e236ec6c0408c4af0e Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Fri, 14 Apr 2023 16:25:51 +0100 Subject: ASoC: cs35l41: Fix default regmap values for some registers Several values do not match the defaults of CS35L41, fix them. Signed-off-by: Stefan Binding Acked-by: Mark Brown Link: https://lore.kernel.org/r/20230414152552.574502-4-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-lib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index 8538e2871c5f..1e4205295a0d 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -46,7 +46,7 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_DSP1_RX5_SRC, 0x00000020 }, { CS35L41_DSP1_RX6_SRC, 0x00000021 }, { CS35L41_DSP1_RX7_SRC, 0x0000003A }, - { CS35L41_DSP1_RX8_SRC, 0x00000001 }, + { CS35L41_DSP1_RX8_SRC, 0x0000003B }, { CS35L41_NGATE1_SRC, 0x00000008 }, { CS35L41_NGATE2_SRC, 0x00000009 }, { CS35L41_AMP_DIG_VOL_CTRL, 0x00008000 }, @@ -58,8 +58,8 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_IRQ1_MASK2, 0xFFFFFFFF }, { CS35L41_IRQ1_MASK3, 0xFFFF87FF }, { CS35L41_IRQ1_MASK4, 0xFEFFFFFF }, - { CS35L41_GPIO1_CTRL1, 0xE1000001 }, - { CS35L41_GPIO2_CTRL1, 0xE1000001 }, + { CS35L41_GPIO1_CTRL1, 0x81000001 }, + { CS35L41_GPIO2_CTRL1, 0x81000001 }, { CS35L41_MIXER_NGATE_CFG, 0x00000000 }, { CS35L41_MIXER_NGATE_CH1_CFG, 0x00000303 }, { CS35L41_MIXER_NGATE_CH2_CFG, 0x00000303 }, -- cgit v1.2.3 From 011a8719d6105dcb48077ea7a6a88ac019d4aa50 Mon Sep 17 00:00:00 2001 From: Maxim Kochetkov Date: Fri, 12 May 2023 14:03:42 +0300 Subject: ASoC: dwc: move DMA init to snd_soc_dai_driver probe() When using DMA mode we are facing with Oops: [ 396.458157] Unable to handle kernel access to user memory without uaccess routines at virtual address 000000000000000c [ 396.469374] Oops [#1] [ 396.471839] Modules linked in: [ 396.475144] CPU: 0 PID: 114 Comm: arecord Not tainted 6.0.0-00164-g9a8eccdaf2be-dirty #68 [ 396.483619] Hardware name: YMP ELCT FPGA (DT) [ 396.488156] epc : dmaengine_pcm_open+0x1d2/0x342 [ 396.493227] ra : dmaengine_pcm_open+0x1d2/0x342 [ 396.498140] epc : ffffffff807fe346 ra : ffffffff807fe346 sp : ffffffc804e138f0 [ 396.505602] gp : ffffffff817bf730 tp : ffffffd8042c8ac0 t0 : 6500000000000000 [ 396.513045] t1 : 0000000000000064 t2 : 656e69676e65616d s0 : ffffffc804e13990 [ 396.520477] s1 : ffffffd801b86a18 a0 : 0000000000000026 a1 : ffffffff816920f8 [ 396.527897] a2 : 0000000000000010 a3 : fffffffffffffffe a4 : 0000000000000000 [ 396.535319] a5 : 0000000000000000 a6 : ffffffd801b87040 a7 : 0000000000000038 [ 396.542740] s2 : ffffffd801b94a00 s3 : 0000000000000000 s4 : ffffffd80427f5e8 [ 396.550153] s5 : ffffffd80427f5e8 s6 : ffffffd801b44410 s7 : fffffffffffffff5 [ 396.557569] s8 : 0000000000000800 s9 : 0000000000000001 s10: ffffffff8066d254 [ 396.564978] s11: ffffffd8059cf768 t3 : ffffffff817d5577 t4 : ffffffff817d5577 [ 396.572391] t5 : ffffffff817d5578 t6 : ffffffc804e136e8 [ 396.577876] status: 0000000200000120 badaddr: 000000000000000c cause: 000000000000000d [ 396.586007] [] snd_soc_component_open+0x1a/0x68 [ 396.592439] [] __soc_pcm_open+0xf0/0x502 [ 396.598217] [] soc_pcm_open+0x2e/0x4e [ 396.603741] [] snd_pcm_open_substream+0x442/0x68e [ 396.610313] [] snd_pcm_open+0xfa/0x212 [ 396.615868] [] snd_pcm_capture_open+0x3a/0x60 [ 396.622048] [] snd_open+0xa8/0x17a [ 396.627421] [] chrdev_open+0xa0/0x218 [ 396.632893] [] do_dentry_open+0x17c/0x2a6 [ 396.638713] [] vfs_open+0x1e/0x26 [ 396.643850] [] path_openat+0x96e/0xc96 [ 396.649518] [] do_filp_open+0x7c/0xf6 [ 396.655034] [] do_sys_openat2+0x8a/0x11e [ 396.660765] [] sys_openat+0x50/0x7c [ 396.666068] [] ret_from_syscall+0x0/0x2 [ 396.674964] ---[ end trace 0000000000000000 ]--- It happens because of play_dma_data/capture_dma_data pointers are NULL. Current implementation assigns these pointers at snd_soc_dai_driver startup() callback and reset them back to NULL at shutdown(). But soc_pcm_open() sequence uses DMA pointers in dmaengine_pcm_open() before snd_soc_dai_driver startup(). Most generic DMA capable I2S drivers use snd_soc_dai_driver probe() callback to init DMA pointers only once at probe. So move DMA init to dw_i2s_dai_probe and drop shutdown() and startup() callbacks. Signed-off-by: Maxim Kochetkov Link: https://lore.kernel.org/r/20230512110343.66664-1-fido_max@inbox.ru Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-i2s.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index ca20cade6840..399a489f24f2 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -183,30 +183,6 @@ static void i2s_stop(struct dw_i2s_dev *dev, } } -static int dw_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - union dw_i2s_snd_dma_data *dma_data = NULL; - - if (!(dev->capability & DWC_I2S_RECORD) && - (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) - return -EINVAL; - - if (!(dev->capability & DWC_I2S_PLAY) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) - return -EINVAL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &dev->play_dma_data; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - dma_data = &dev->capture_dma_data; - - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); - - return 0; -} - static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) { u32 ch_reg; @@ -305,12 +281,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void dw_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_dai_set_dma_data(dai, substream, NULL); -} - static int dw_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -382,8 +352,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } static const struct snd_soc_dai_ops dw_i2s_dai_ops = { - .startup = dw_i2s_startup, - .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, @@ -625,6 +593,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, } +static int dw_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data); + return 0; +} + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -643,6 +619,7 @@ static int dw_i2s_probe(struct platform_device *pdev) return -ENOMEM; dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->probe = dw_i2s_dai_probe; dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(dev->i2s_base)) -- cgit v1.2.3 From 4ca110cab46561cd74a2acd9b447435acb4bec5f Mon Sep 17 00:00:00 2001 From: Bin Li Date: Wed, 24 May 2023 19:37:55 +0800 Subject: ALSA: hda/realtek: Enable headset onLenovo M70/M90 Lenovo M70/M90 Gen4 are equipped with ALC897, and they need ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work. The previous quirk for M70/M90 is for Gen3. Signed-off-by: Bin Li Cc: Link: https://lore.kernel.org/r/20230524113755.1346928-1-bin.li@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7e4765eff80..7b5f194513c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11719,6 +11719,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x3321, "Lenovo ThinkCentre M70 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x331b, "Lenovo ThinkCentre M90 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), -- cgit v1.2.3