From febf22565549ea7111e7d45e8f2d64373cc66b11 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Sat, 20 Mar 2021 17:15:41 +0800 Subject: [PATCH 01/27] ALSA: hda/realtek: fix a determine_headset_type issue for a Dell AIO We found a recording issue on a Dell AIO, users plug a headset-mic and select headset-mic from UI, but can't record any sound from headset-mic. The root cause is the determine_headset_type() returns a wrong type, e.g. users plug a ctia type headset, but that function returns omtp type. On this machine, the internal mic is not connected to the codec, the "Input Source" is headset mic by default. And when users plug a headset, the determine_headset_type() will be called immediately, the codec on this AIO is alc274, the delay time for this codec in the determine_headset_type() is only 80ms, the delay is too short to correctly determine the headset type, the fail rate is nearly 99% when users plug the headset with the normal speed. Other codecs set several hundred ms delay time, so here I change the delay time to 850ms for alc2x4 series, after this change, the fail rate is zero unless users plug the headset slowly on purpose. Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20210320091542.6748-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 316b9b4ccb32..8935bbe411a2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5256,7 +5256,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0274: case 0x10ec0294: alc_process_coef_fw(codec, coef0274); - msleep(80); + msleep(850); val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x00f0) == 0x00f0; break; From e54f30befa7990b897189b44a56c1138c6bfdbb5 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Sat, 20 Mar 2021 17:15:42 +0800 Subject: [PATCH 02/27] ALSA: hda/realtek: call alc_update_headset_mode() in hp_automute_hook We found the alc_update_headset_mode() is not called on some machines when unplugging the headset, as a result, the mode of the ALC_HEADSET_MODE_UNPLUGGED can't be set, then the current_headset_type is not cleared, if users plug a differnt type of headset next time, the determine_headset_type() will not be called and the audio jack is set to the headset type of previous time. On the Dell machines which connect the dmic to the PCH, if we open the gnome-sound-setting and unplug the headset, this issue will happen. Those machines disable the auto-mute by ucm and has no internal mic in the input source, so the update_headset_mode() will not be called by cap_sync_hook or automute_hook when unplugging, and because the gnome-sound-setting is opened, the codec will not enter the runtime_suspend state, so the update_headset_mode() will not be called by alc_resume when unplugging. In this case the hp_automute_hook is called when unplugging, so add update_headset_mode() calling to this function. Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20210320091542.6748-2-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8935bbe411a2..fc2f60c58ad8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5440,6 +5440,7 @@ static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_callback *jack) { snd_hda_gen_hp_automute(codec, jack); + alc_update_headset_mode(codec); } static void alc_probe_headset_mode(struct hda_codec *codec) From 927280909fa7d8e61596800d82f18047c6cfbbe4 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Mon, 22 Mar 2021 11:37:21 -0500 Subject: [PATCH 03/27] ASoC: SOF: Intel: HDA: fix core status verification When checking for enabled cores it isn't enough to check that some of the requested cores are running, we have to check that all of them are. Fixes: 747503b1813a ("ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations") Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210322163728.16616-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c3b757cf01a0..2543ef1b5098 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -226,10 +226,17 @@ bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS); - is_enable = (val & HDA_DSP_ADSPCS_CPA_MASK(core_mask)) && - (val & HDA_DSP_ADSPCS_SPA_MASK(core_mask)) && - !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) && - !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask)); +#define MASK_IS_EQUAL(v, m, field) ({ \ + u32 _m = field(m); \ + ((v) & _m) == _m; \ +}) + + is_enable = MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_CPA_MASK) && + MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_SPA_MASK) && + !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) && + !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask)); + +#undef MASK_IS_EQUAL dev_dbg(sdev->dev, "DSP core(s) enabled? %d : core_mask %x\n", is_enable, core_mask); From 91ec48f540f83022377723a774a0a37a630801af Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 22 Mar 2021 11:37:22 -0500 Subject: [PATCH 04/27] ASoC: SOF: core: harden shutdown helper When the probe is handled in a workqueue, we must use cancel_work_sync() in the shutdown helper to avoid possible race conditions. We must also take care of possible errors happening in a probe workqueue or during pm_runtime resume (called e.g. before shutdown for PCI devices). We should really only try to access hardware registers and initiate IPCs if the DSP is fully booted. Fixes: daff7f1478e12 ("ASoC: SOF: add snd_sof_device_shutdown() helper for shutdown") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Reviewed-by: Libin Yang Link: https://lore.kernel.org/r/20210322163728.16616-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 6d8f7d9fd192..4a3d522f612b 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -399,7 +399,13 @@ int snd_sof_device_shutdown(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); - return snd_sof_shutdown(sdev); + if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) + cancel_work_sync(&sdev->probe_work); + + if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) + return snd_sof_shutdown(sdev); + + return 0; } EXPORT_SYMBOL(snd_sof_device_shutdown); From 3c429f861ed483517a0a352281a16503bcc60b55 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Mon, 22 Mar 2021 11:37:23 -0500 Subject: [PATCH 05/27] ASoC: SOF: Intel: TGL: fix EHL ops EHL is derived from TGL, not CNL, so we shall use the TGL ops. Fixes: 8d4ba1be3d22 ("ASoC: SOF: pci: split PCI into different drivers") Reviewed-by: Ranjani Sridharan Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210322163728.16616-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-tgl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index 485607471181..38bc353f7313 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -65,7 +65,7 @@ static const struct sof_dev_desc ehl_desc = { .default_tplg_path = "intel/sof-tplg", .default_fw_filename = "sof-ehl.ri", .nocodec_tplg_filename = "sof-ehl-nocodec.tplg", - .ops = &sof_cnl_ops, + .ops = &sof_tgl_ops, }; static const struct sof_dev_desc adls_desc = { From 22aa9e021ad1ee7ce640270e75f4bdccff65d287 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Mon, 22 Mar 2021 11:37:24 -0500 Subject: [PATCH 06/27] ASoC: SOF: Intel: TGL: set shutdown callback to hda_dsp_shutdown According to hardware spec and PMC FW requirement, the DSP must be in D3 state before entering S5. Define the shutdown function to use snd_sof_suspend as shutdown callback to make sure DSP is in D3 state. Fixes: 44a4cfad8d78 ("ASoC: SOF: Intel: tgl: do thorough remove at .shutdown() callback") Reviewed-by: Ranjani Sridharan Signed-off-by: Pan Xiuli Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210322163728.16616-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 6 ++++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/tgl.c | 2 +- 3 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 2543ef1b5098..736a54beca23 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -892,6 +892,12 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); } +int hda_dsp_shutdown(struct snd_sof_dev *sdev) +{ + sdev->system_suspend_target = SOF_SUSPEND_S3; + return snd_sof_suspend(sdev->dev); +} + int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) { #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 7c7579daee7f..ae80725b0e33 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -517,6 +517,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev); int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev); int hda_dsp_runtime_resume(struct snd_sof_dev *sdev); int hda_dsp_runtime_idle(struct snd_sof_dev *sdev); +int hda_dsp_shutdown(struct snd_sof_dev *sdev); int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev); void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags); void hda_ipc_dump(struct snd_sof_dev *sdev); diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 419f05ba1920..3e46fac53f78 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -25,7 +25,7 @@ const struct snd_sof_dsp_ops sof_tgl_ops = { /* probe/remove/shutdown */ .probe = hda_dsp_probe, .remove = hda_dsp_remove, - .shutdown = hda_dsp_remove, + .shutdown = hda_dsp_shutdown, /* Register IO */ .write = sof_io_write, From 4939e49ea5804f89941df86d35f1a1e1cd8b435b Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Mon, 22 Mar 2021 11:37:25 -0500 Subject: [PATCH 07/27] ASoC: SOF: Intel: ICL: set shutdown callback to hda_dsp_shutdown According to hardware spec and PMC FW requirement, the DSP must be in D3 state before entering S5. Set shutdown call to hda_dsp_shutdown. Reviewed-by: Ranjani Sridharan Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210322163728.16616-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/icl.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c index e9d5a0a58504..88a74be8a0c1 100644 --- a/sound/soc/sof/intel/icl.c +++ b/sound/soc/sof/intel/icl.c @@ -26,9 +26,10 @@ static const struct snd_sof_debugfs_map icl_dsp_debugfs[] = { /* Icelake ops */ const struct snd_sof_dsp_ops sof_icl_ops = { - /* probe and remove */ + /* probe/remove/shutdown */ .probe = hda_dsp_probe, .remove = hda_dsp_remove, + .shutdown = hda_dsp_shutdown, /* Register IO */ .write = sof_io_write, From b0503e8410e5ee43da116772576dbdeb2a414e0b Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Mon, 22 Mar 2021 11:37:26 -0500 Subject: [PATCH 08/27] ASoC: SOF: Intel: CNL: set shutdown callback to hda_dsp_shutdown According to hardware spec and PMC FW requirement, the DSP must be in D3 state before entering S5. Set shutdown call to hda_dsp_shutdown. Reviewed-by: Ranjani Sridharan Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210322163728.16616-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index e38db519f38d..094cde17a1b7 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -232,9 +232,10 @@ void cnl_ipc_dump(struct snd_sof_dev *sdev) /* cannonlake ops */ const struct snd_sof_dsp_ops sof_cnl_ops = { - /* probe and remove */ + /* probe/remove/shutdown */ .probe = hda_dsp_probe, .remove = hda_dsp_remove, + .shutdown = hda_dsp_shutdown, /* Register IO */ .write = sof_io_write, From d3aa96bf349882763b9903e5800d2e83fc086886 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Mon, 22 Mar 2021 11:37:27 -0500 Subject: [PATCH 09/27] ASoC: SOF: Intel: APL: set shutdown callback to hda_dsp_shutdown According to hardware spec and PMC FW requirement, the DSP must be in D3 state before entering S5. Set shutdown call to hda_dsp_shutdown. Reviewed-by: Ranjani Sridharan Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210322163728.16616-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/apl.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index fc29b91b8932..c7ed2b3d6abc 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -27,9 +27,10 @@ static const struct snd_sof_debugfs_map apl_dsp_debugfs[] = { /* apollolake ops */ const struct snd_sof_dsp_ops sof_apl_ops = { - /* probe and remove */ + /* probe/remove/shutdown */ .probe = hda_dsp_probe, .remove = hda_dsp_remove, + .shutdown = hda_dsp_shutdown, /* Register IO */ .write = sof_io_write, From 8bb84ca873d2222ca220e58a097090775b1fd8df Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 22 Mar 2021 11:37:28 -0500 Subject: [PATCH 10/27] ASoC: SOF: Intel: move ELH chip info ELH is a derivative of TGL, so it should be exposed in tgl.c for consistency. No functional change. Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210322163728.16616-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 16 ---------------- sound/soc/sof/intel/tgl.c | 16 ++++++++++++++++ 2 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 094cde17a1b7..821f25fbcf08 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -350,22 +350,6 @@ const struct sof_intel_dsp_desc cnl_chip_info = { }; EXPORT_SYMBOL_NS(cnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); -const struct sof_intel_dsp_desc ehl_chip_info = { - /* Elkhartlake */ - .cores_num = 4, - .init_core_mask = 1, - .host_managed_cores_mask = BIT(0), - .ipc_req = CNL_DSP_REG_HIPCIDR, - .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY, - .ipc_ack = CNL_DSP_REG_HIPCIDA, - .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE, - .ipc_ctl = CNL_DSP_REG_HIPCCTL, - .rom_init_timeout = 300, - .ssp_count = ICL_SSP_COUNT, - .ssp_base_offset = CNL_SSP_BASE_OFFSET, -}; -EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); - const struct sof_intel_dsp_desc jsl_chip_info = { /* Jasperlake */ .cores_num = 2, diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 3e46fac53f78..54ba1b88ba86 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -156,6 +156,22 @@ const struct sof_intel_dsp_desc tglh_chip_info = { }; EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); +const struct sof_intel_dsp_desc ehl_chip_info = { + /* Elkhartlake */ + .cores_num = 4, + .init_core_mask = 1, + .host_managed_cores_mask = BIT(0), + .ipc_req = CNL_DSP_REG_HIPCIDR, + .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY, + .ipc_ack = CNL_DSP_REG_HIPCIDA, + .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE, + .ipc_ctl = CNL_DSP_REG_HIPCCTL, + .rom_init_timeout = 300, + .ssp_count = ICL_SSP_COUNT, + .ssp_base_offset = CNL_SSP_BASE_OFFSET, +}; +EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); + const struct sof_intel_dsp_desc adls_chip_info = { /* Alderlake-S */ .cores_num = 2, From 16b82e75c15a7dbd564ea3654f3feb61df9e1e6f Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 19 Mar 2021 18:48:46 +0800 Subject: [PATCH 11/27] ASoC: wm8960: Fix wrong bclk and lrclk with pll enabled for some chips The input MCLK is 12.288MHz, the desired output sysclk is 11.2896MHz and sample rate is 44100Hz, with the configuration pllprescale=2, postscale=sysclkdiv=1, some chip may have wrong bclk and lrclk output with pll enabled in master mode, but with the configuration pllprescale=1, postscale=2, the output clock is correct. >From Datasheet, the PLL performs best when f2 is between 90MHz and 100MHz when the desired sysclk output is 11.2896MHz or 12.288MHz, so sysclkdiv = 2 (f2/8) is the best choice. So search available sysclk_divs from 2 to 1 other than from 1 to 2. Fixes: 84fdc00d519f ("ASoC: codec: wm9860: Refactor PLL out freq search") Signed-off-by: Shengjiu Wang Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1616150926-22892-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..cda9cd935d4f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -707,7 +707,13 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, best_freq_out = -EINVAL; *sysclk_idx = *dac_idx = *bclk_idx = -1; - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + /* + * From Datasheet, the PLL performs best when f2 is between + * 90MHz and 100MHz, the desired sysclk output is 11.2896MHz + * or 12.288MHz, then sysclkdiv = 2 is the best choice. + * So search sysclk_divs from 2 to 1 other than from 1 to 2. + */ + for (i = ARRAY_SIZE(sysclk_divs) - 1; i >= 0; --i) { if (sysclk_divs[i] == -1) continue; for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { From aa65bacdb70e549a81de03ec72338e1047842883 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 24 Mar 2021 14:27:10 +0100 Subject: [PATCH 12/27] ASoC: intel: atom: Stop advertising non working S24LE support The SST firmware's media and deep-buffer inputs are hardcoded to S16LE, the corresponding DAIs don't have a hw_params callback and their prepare callback also does not take the format into account. So far the advertising of non working S24LE support has not caused issues because pulseaudio defaults to S16LE, but changing pulse-audio's config to use S24LE will result in broken sound. Pipewire is replacing pulse now and pipewire prefers S24LE over S16LE when available, causing the problem of the broken S24LE support to come to the surface now. Cc: stable@vger.kernel.org BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/866 Fixes: 098c2cd281409 ("ASoC: Intel: Atom: add 24-bit support for media playback and capture") Acked-by: Pierre-Louis Bossart Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210324132711.216152-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 9e9b05883557..aa5dd590ddd5 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -488,14 +488,14 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Headset Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -506,7 +506,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { From 632aeebe1b7a3a8b193d71942a10e66919bebfb8 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 24 Mar 2021 14:27:11 +0100 Subject: [PATCH 13/27] ASoC: intel: atom: Remove 44100 sample-rate from the media and deep-buffer DAI descriptions The media and deep-buffer DAIs only support 48000 Hz samplerate, remove the 44100 sample-rate from their descriptions. Acked-by: Pierre-Louis Bossart Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210324132711.216152-3-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index aa5dd590ddd5..4124aa2fc247 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -487,14 +487,14 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .stream_name = "Headset Playback", .channels_min = SST_STEREO, .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Headset Capture", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, @@ -505,7 +505,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .stream_name = "Deepbuffer Playback", .channels_min = SST_STEREO, .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, From a23f9099ff1541f15704e96b784d3846d2a4483d Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Wed, 24 Mar 2021 20:35:53 -0700 Subject: [PATCH 14/27] ASoC: max98373: Changed amp shutdown register as volatile 0x20FF(amp global enable) register was defined as non-volatile, but it is not. Overheating, overcurrent can cause amp shutdown in hardware. 'regmap_write' compare register readback value before writing to avoid same value writing. 'regmap_read' just read cache not actual hardware value for the non-volatile register. When amp is internally shutdown by some reason, next 'AMP ON' command can be ignored because regmap think amp is already ON. Signed-off-by: Ryan Lee Link: https://lore.kernel.org/r/20210325033555.29377-1-ryans.lee@maximintegrated.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373-i2c.c | 1 + sound/soc/codecs/max98373-sdw.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 85f6865019d4..ddb6436835d7 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -446,6 +446,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: return true; default: diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index d8c47667a9ea..f3a12205cd48 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -220,6 +220,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: /* SoundWire Control Port Registers */ case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR: From 3a27875e91fb9c29de436199d20b33f9413aea77 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Wed, 24 Mar 2021 20:35:54 -0700 Subject: [PATCH 15/27] ASoC: max98373: Added 30ms turn on/off time delay Amp requires 10 ~ 30ms for the power ON and OFF. Added 30ms delay for stability. Signed-off-by: Ryan Lee Link: https://lore.kernel.org/r/20210325033555.29377-2-ryans.lee@maximintegrated.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 746c829312b8..1346a98ce8a1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -28,11 +28,13 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w, regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 1); + usleep_range(30000, 31000); break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 0); + usleep_range(30000, 31000); max98373->tdm_mode = false; break; default: From 625bd5a616ceda4840cd28f82e957c8ced394b6a Mon Sep 17 00:00:00 2001 From: Ikjoon Jang Date: Wed, 24 Mar 2021 18:51:52 +0800 Subject: [PATCH 16/27] ALSA: usb-audio: Apply sample rate quirk to Logitech Connect Logitech ConferenceCam Connect is a compound USB device with UVC and UAC. Not 100% reproducible but sometimes it keeps responding STALL to every control transfer once it receives get_freq request. This patch adds 046d:0x084c to a snd_usb_get_sample_rate_quirk list. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=203419 Signed-off-by: Ikjoon Jang Cc: Link: https://lore.kernel.org/r/20210324105153.2322881-1-ikjn@chromium.org Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d3001fb18141..176437a441e6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1521,6 +1521,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */ + case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */ return true; } From aa320c7cd45647b75af2233430d36a8d154703d4 Mon Sep 17 00:00:00 2001 From: kernel test robot Date: Sun, 28 Mar 2021 16:54:45 +0200 Subject: [PATCH 17/27] ASoC: cygnus: fix for_each_child.cocci warnings Function "for_each_available_child_of_node" should have of_node_put() before return around line 1352. Generated by: scripts/coccinelle/iterators/for_each_child.cocci CC: Sumera Priyadarsini Reported-by: kernel test robot Signed-off-by: kernel test robot Signed-off-by: Julia Lawall Link: https://lore.kernel.org/r/alpine.DEB.2.22.394.2103281651320.2854@hadrien Signed-off-by: Mark Brown --- sound/soc/bcm/cygnus-ssp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 6e634b448293..aa16a2375134 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1348,8 +1348,10 @@ static int cygnus_ssp_probe(struct platform_device *pdev) &cygnus_ssp_dai[active_port_count]); /* negative is err, 0 is active and good, 1 is disabled */ - if (err < 0) + if (err < 0) { + of_node_put(child_node); return err; + } else if (!err) { dev_dbg(dev, "Activating DAI: %s\n", cygnus_ssp_dai[active_port_count].name); From c8f79808cd8eb5bc8d14de129bd6d586d3fce0aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2021 13:30:58 +0200 Subject: [PATCH 18/27] ALSA: hda: Re-add dropped snd_poewr_change_state() calls The card power state change via snd_power_change_state() at the system suspend/resume seems dropped mistakenly during the PM code rewrite. The card power state doesn't play much role nowadays but it's still referred in a few places such as the HDMI codec driver. This patch restores them, but in a more appropriate place now in the prepare and complete callbacks. Fixes: f5dac54d9d93 ("ALSA: hda: Separate runtime and system suspend") Cc: Link: https://lore.kernel.org/r/20210329113059.25035-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5eea130dcf0a..c4146e8617de 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1025,6 +1025,7 @@ static int azx_prepare(struct device *dev) chip = card->private_data; chip->pm_prepared = 1; + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); flush_work(&azx_bus(chip)->unsol_work); @@ -1040,6 +1041,7 @@ static void azx_complete(struct device *dev) struct azx *chip; chip = card->private_data; + snd_power_change_state(card, SNDRV_CTL_POWER_D0); chip->pm_prepared = 0; } From 66affb7bb0dc0905155a1b2475261aa704d1ddb5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2021 13:30:59 +0200 Subject: [PATCH 19/27] ALSA: hda: Add missing sanity checks in PM prepare/complete callbacks The recently added PM prepare and complete callbacks don't have the sanity check whether the card instance has been properly initialized, which may potentially lead to Oops. This patch adds the azx_is_pm_ready() call in each place appropriately like other PM callbacks. Fixes: f5dac54d9d93 ("ALSA: hda: Separate runtime and system suspend") Cc: Link: https://lore.kernel.org/r/20210329113059.25035-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c4146e8617de..65551eea6752 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1023,6 +1023,9 @@ static int azx_prepare(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + if (!azx_is_pm_ready(card)) + return 0; + chip = card->private_data; chip->pm_prepared = 1; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1040,6 +1043,9 @@ static void azx_complete(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + if (!azx_is_pm_ready(card)) + return; + chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D0); chip->pm_prepared = 0; From 417eadfdd9e25188465280edf3668ed163fda2d0 Mon Sep 17 00:00:00 2001 From: Jeremy Szu Date: Tue, 30 Mar 2021 19:44:27 +0800 Subject: [PATCH 20/27] ALSA: hda/realtek: fix mute/micmute LEDs for HP 640 G8 The HP EliteBook 640 G8 Notebook PC is using ALC236 codec which is using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu Cc: Link: https://lore.kernel.org/r/20210330114428.40490-1-jeremy.szu@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc2f60c58ad8..58946d069ee5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8058,6 +8058,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), From 7c0d6e482062eb5c06ecccfab340abc523bdca00 Mon Sep 17 00:00:00 2001 From: Bastian Germann Date: Wed, 31 Mar 2021 17:18:43 +0200 Subject: [PATCH 21/27] ASoC: sunxi: sun4i-codec: fill ASoC card owner card->owner is a required property and since commit 81033c6b584b ("ALSA: core: Warn on empty module") a warning is issued if it is empty. Add it. This fixes following warning observed on Lamobo R1: WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 Hardware name: Allwinner sun7i (A20) Family Call trace: (snd_card_new [snd]) (snd_soc_bind_card [snd_soc_core]) (snd_soc_register_card [snd_soc_core]) (sun4i_codec_probe [sun4i_codec]) Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Related: commit 3c27ea23ffb4 ("ASoC: qcom: Set card->owner to avoid warnings") Related: commit ec653df2a0cb ("drm/vc4/vc4_hdmi: fill ASoC card owner") Cc: linux-arm-kernel@lists.infradead.org Cc: alsa-devel@alsa-project.org Signed-off-by: Bastian Germann Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 6c13cc84b3fb..2173991c13db 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1364,6 +1364,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1396,6 +1397,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1449,6 +1451,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1487,6 +1490,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1525,6 +1529,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); From b861106f3cd693f944ba46d9ea8744a3fbfd14db Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 31 Mar 2021 18:12:34 +0100 Subject: [PATCH 22/27] ASoC: codecs: lpass-tx-macro: set npl clock rate correctly NPL clock rate is twice the MCLK rate, so set this correctly to avoid soundwire timeouts. Fixes: c39667ddcfc5 ("ASoC: codecs: lpass-tx-macro: add support for lpass tx macro") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210331171235.24824-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-tx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 36d7a6442cdb..e8c6c738bbaa 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1811,7 +1811,7 @@ static int tx_macro_probe(struct platform_device *pdev) /* set MCLK and NPL rates */ clk_set_rate(tx->clks[2].clk, MCLK_FREQ); - clk_set_rate(tx->clks[3].clk, MCLK_FREQ); + clk_set_rate(tx->clks[3].clk, 2 * MCLK_FREQ); ret = clk_bulk_prepare_enable(TX_NUM_CLKS_MAX, tx->clks); if (ret) From adfc3ed7dcb98f7411d3632e3bdf81690294fe7d Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 31 Mar 2021 18:12:35 +0100 Subject: [PATCH 23/27] ASoC: codecs: lpass-rx-macro: set npl clock rate correctly NPL clock rate is twice the MCLK rate, so set this correctly to avoid soundwire timeouts. Fixes: af3d54b99764 ("ASoC: codecs: lpass-rx-macro: add support for lpass rx macro") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210331171235.24824-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 8c04b3b2c907..7878da89d8e0 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3551,7 +3551,7 @@ static int rx_macro_probe(struct platform_device *pdev) /* set MCLK and NPL rates */ clk_set_rate(rx->clks[2].clk, MCLK_FREQ); - clk_set_rate(rx->clks[3].clk, MCLK_FREQ); + clk_set_rate(rx->clks[3].clk, 2 * MCLK_FREQ); ret = clk_bulk_prepare_enable(RX_NUM_CLKS_MAX, rx->clks); if (ret) From e7a48c710defa0e0fef54d42b7d9e4ab596e2761 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Fri, 2 Apr 2021 11:14:05 +0300 Subject: [PATCH 24/27] ASoC: fsl_esai: Fix TDM slot setup for I2S mode When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan Acked-by: Nicolin Chen Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 08056fa0a0fa..a857a624864f 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -519,11 +519,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; From c6423ed2da6214a68527446b5f8e09cf7162b2ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Apr 2021 19:13:14 +0200 Subject: [PATCH 25/27] ALSA: hda/conexant: Apply quirk for another HP ZBook G5 model There is another HP ZBook G5 model with the PCI SSID 103c:844f that requires the same quirk for controlling the mute LED. Add the corresponding entry to the quirk table. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212407 Cc: Link: https://lore.kernel.org/r/20210401171314.667-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c20dad46a7c9..dfef9c17e140 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -944,6 +944,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8427, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x844f, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), From 168632a495f49f33a18c2d502fc249d7610375e9 Mon Sep 17 00:00:00 2001 From: Jonas Holmberg Date: Wed, 7 Apr 2021 09:54:28 +0200 Subject: [PATCH 26/27] ALSA: aloop: Fix initialization of controls Add a control to the card before copying the id so that the numid field is initialized in the copy. Otherwise the numid field of active_id, format_id, rate_id and channels_id will be the same (0) and snd_ctl_notify() will not queue the events properly. Signed-off-by: Jonas Holmberg Reviewed-by: Jaroslav Kysela Cc: Link: https://lore.kernel.org/r/20210407075428.2666787-1-jonashg@axis.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 8a24e5ae7cef..ef0cdfddfd3f 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1572,6 +1572,14 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) return -ENOMEM; kctl->id.device = dev; kctl->id.subdevice = substr; + + /* Add the control before copying the id so that + * the numid field of the id is set in the copy. + */ + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + switch (idx) { case ACTIVE_IDX: setup->active_id = kctl->id; @@ -1588,9 +1596,6 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) default: break; } - err = snd_ctl_add(card, kctl); - if (err < 0) - return err; } } } From c8426b2700b57d2760ff335840a02f66a64b6044 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Apr 2021 11:57:30 +0200 Subject: [PATCH 27/27] ALSA: hda/realtek: Fix speaker amp setup on Acer Aspire E1 We've got a report about Acer Aspire E1 (PCI SSID 1025:0840) that loses the speaker output after resume. With the comparison of COEF dumps, it was identified that the COEF 0x0d bits 0x6000 corresponds to the speaker amp. This patch adds the specific quirk for the device to restore the COEF bits at the codec (re-)initialization. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1183869 Cc: Link: https://lore.kernel.org/r/20210407095730.12560-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 58946d069ee5..a7544b77d3f7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3927,6 +3927,15 @@ static void alc271_fixup_dmic(struct hda_codec *codec, snd_hda_sequence_write(codec, verbs); } +/* Fix the speaker amp after resume, etc */ +static void alc269vb_fixup_aspire_e1_coef(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action == HDA_FIXUP_ACT_INIT) + alc_update_coef_idx(codec, 0x0d, 0x6000, 0x6000); +} + static void alc269_fixup_pcm_44k(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6301,6 +6310,7 @@ enum { ALC283_FIXUP_HEADSET_MIC, ALC255_FIXUP_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, + ALC269VB_FIXUP_ASPIRE_E1_COEF, ALC280_FIXUP_HP_GPIO4, ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, @@ -6979,6 +6989,10 @@ static const struct hda_fixup alc269_fixups[] = { { }, }, }, + [ALC269VB_FIXUP_ASPIRE_E1_COEF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269vb_fixup_aspire_e1_coef, + }, [ALC280_FIXUP_HP_GPIO4] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_gpio4, @@ -7901,6 +7915,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF), SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), @@ -8395,6 +8410,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"}, {.id = ALC255_FIXUP_MIC_MUTE_LED, .name = "alc255-dell-mute"}, {.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"}, + {.id = ALC269VB_FIXUP_ASPIRE_E1_COEF, .name = "aspire-e1-coef"}, {.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"}, {.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"},