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* ALSA: firewire-lib: fix to check cycle continuityTakashi Sakamoto2024-03-061-1/+1
| | | | | | | | | | | | | | | | | | | | commit 77ce96543b03f437c6b45f286d8110db2b6622a3 upstream. The local helper function to compare the given pair of cycle count evaluates them. If the left value is less than the right value, the function returns negative value. If the safe cycle is less than the current cycle, it is the case of cycle lost. However, it is not currently handled properly. This commit fixes the bug. Cc: <stable@vger.kernel.org> Fixes: 705794c53b00 ("ALSA: firewire-lib: check cycle continuity") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20240218033026.72577-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()Richard Fitzgerald2024-03-064-5/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit eba2eb2495f47690400331c722868902784e59de ] snd_soc_card_get_kcontrol() must be holding a read lock on card->controls_rwsem while walking the controls list. Compare with snd_ctl_find_numid(). The existing function is renamed snd_soc_card_get_kcontrol_locked() so that it can be called from contexts that are already holding card->controls_rwsem (for example, control get/put functions). There are few direct or indirect callers of snd_soc_card_get_kcontrol(), and most are safe. Three require changes, which have been included in this patch: codecs/cs35l45.c: cs35l45_activate_ctl() is called from a control put() function so is changed to call snd_soc_card_get_kcontrol_locked(). codecs/cs35l56.c: cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from control get()/put() functions so is changed to call snd_soc_card_get_kcontrol_locked(). fsl/fsl_xcvr.c: fsl_xcvr_activate_ctl() is called from three places, one of which already holds card->controls_rwsem: 1. fsl_xcvr_mode_put(), a control put function, which will already be holding card->controls_rwsem. 2. fsl_xcvr_startup(), a DAI startup function. 3. fsl_xcvr_shutdown(), a DAI shutdown function. To fix this, fsl_xcvr_activate_ctl() has been changed to call snd_soc_card_get_kcontrol_locked() so that it is safe to call directly from fsl_xcvr_mode_put(). The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been changed to take a read lock on card->controls_rsem() around calls to fsl_xcvr_activate_ctl(). While this is not very elegant, it keeps the change small, to avoid this patch creating a large collateral churn in fsl/fsl_xcvr.c. Analysis of other callers of snd_soc_card_get_kcontrol() is that they do not need any changes, they are not holding card->controls_rwsem when they call snd_soc_card_get_kcontrol(). Direct callers of snd_soc_card_get_kcontrol(): fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler Indirect callers via soc_component_notify_control(): codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler codecs/ak4118.c: ak4118_irq_handler() - IRQ handler codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used Indirect callers via snd_soc_limit_volume(): qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function ti/rx51.c: rx51_aic34_init() - DAI init function I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c and ak4118.c changes. Backport note: The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added since the Fixes commit so won't all be present on older kernels. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: 209c6cdfd283 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card") Link: https://lore.kernel.org/r/20240221123710.690224-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: cs35l56: Fix deadlock in ASP1 mixer register initializationRichard Fitzgerald2024-03-062-82/+73
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit c14f09f010cc569ae7e2f6ef02374f6bfef9917e ] Rewrite the handling of ASP1 TX mixer mux initialization to prevent a deadlock during component_remove(). The firmware can overwrite the ASP1 TX mixer registers with system-specific settings. This is mainly for hardware that uses the ASP as a chip-to-chip link controlled by the firmware. Because of this the driver cannot know the starting state of the ASP1 mixer muxes until the firmware has been downloaded and rebooted. The original workaround for this was to queue a work function from the dsp_work() job. This work then read the register values (populating the regmap cache the first time around) and then called snd_soc_dapm_mux_update_power(). The problem with this is that it was ultimately triggered by cs35l56_component_probe() queueing dsp_work, which meant that it would be running in parallel with the rest of the ASoC component and card initialization. To prevent accessing DAPM before it was fully initialized the work function took the card mutex. But this would deadlock if cs35l56_component_remove() was called before the work job had completed, because ASoC calls component_remove() with the card mutex held. This new version removes the work function. Instead the regmap cache and DAPM mux widgets are initialized the first time any of the associated ALSA controls is read or written. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") Link: https://lore.kernel.org/r/20240208123742.1278104-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Stable-dep-of: eba2eb2495f4 ("ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()") Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: cs35l56: Fix misuse of wm_adsp 'part' string for silicon revisionRichard Fitzgerald2024-03-061-7/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit f6c967941c5d6fa526fdd64733a8d86bf2bfab31 ] Put the silicon revision and secured flag in the wm_adsp fwf_name string instead of including them in the part string. This changes the format of the firmware name string from cs35l56[s]-rev-misc[-system_name] to cs35l56-rev[-s]-misc[-system_name] No firmware files have been published, so this doesn't cause a compatibility break. Silicon revision and secured flag are included in the firmware filename to pick a firmware compatible with the part. These strings were being added to the part string, but that is a misuse of the string. The correct place for these is the fwf_name string, which is specifically intended to select between multiple firmware files for the same part. Backport note: This won't apply to kernels older than v6.6. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: 608f1b0dbdde ("ASoC: cs35l56: Move DSP part string generation so that it is done only once") Link: https://msgid.link/r/20240129162737.497-12-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Stable-dep-of: eba2eb2495f4 ("ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()") Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: cs35l56: Fix for initializing ASP1 mixer registersRichard Fitzgerald2024-03-063-17/+163
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 07f7d6e7a124d3e4de36771e2a4926d0e31c2258 ] Defer initializing the state of the ASP1 mixer registers until the firmware has been downloaded and rebooted. On a SoundWire system the ASP is free for use as a chip-to-chip interconnect. This can be either for the firmware on multiple CS35L56 to share reference audio; or as a bridge to another device. If it is a firmware interconnect it is owned by the firmware and the Linux driver should avoid writing the registers. However, if it is a bridge then Linux may take over and handle it as a normal codec-to-codec link. Even if the ASP is used as a firmware-firmware interconnect it is useful to have ALSA controls for the ASP mixer. They are at least useful for debugging. CS35L56 is designed for SDCA and a generic SDCA driver would know nothing about these chip-specific registers. So if the ASP is being used on a SoundWire system the firmware sets up the ASP mixer registers. This means that we can't assume the default state of these registers. But we don't know the initial state that the firmware set them to until after the firmware has been downloaded and booted, which can take several seconds when downloading multiple amps. DAPM normally reads the initial state of mux registers during probe() but this would mean blocking probe() for several seconds until the firmware has initialized them. To avoid this, the mixer muxes are set SND_SOC_NOPM to prevent DAPM trying to read the register state. Custom get/set callbacks are implemented for ALSA control access, and these can safely block waiting for the firmware download. After the firmware download has completed, the state of the mux registers is known so a work job is queued to call snd_soc_dapm_mux_update_power() on each of the mux widgets. Backport note: This won't apply cleanly to kernels older than v6.6. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Link: https://msgid.link/r/20240129162737.497-11-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Stable-dep-of: eba2eb2495f4 ("ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()") Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: cs35l56: Don't add the same register patch multiple timesRichard Fitzgerald2024-03-061-4/+4
| | | | | | | | | | | | | | | | | | [ Upstream commit 07687cd0539f8185b6ba0c0afba8473517116d6a ] Move the call to cs35l56_set_patch() earlier in cs35l56_init() so that it only adds the register patch on first-time initialization. The call was after the post_soft_reset label, so every time this function was run to re-initialize the hardware after a reset it would call regmap_register_patch() and add the same reg_sequence again. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: 898673b905b9 ("ASoC: cs35l56: Move shared data into a common data structure") Link: https://msgid.link/r/20240129162737.497-6-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Stable-dep-of: eba2eb2495f4 ("ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()") Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: cs35l56: cs35l56_component_remove() must clean up wm_adspRichard Fitzgerald2024-03-061-0/+5
| | | | | | | | | | | | | | [ Upstream commit cd38ccbecdace1469b4e0cfb3ddeec72a3fad226 ] cs35l56_component_remove() must call wm_adsp_power_down() and wm_adsp2_component_remove(). Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Link: https://msgid.link/r/20240129162737.497-5-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Stable-dep-of: eba2eb2495f4 ("ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()") Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: cs35l56: cs35l56_component_remove() must clear cs35l56->componentRichard Fitzgerald2024-03-061-0/+2
| | | | | | | | | | | | | | | | | [ Upstream commit ae861c466ee57e15a29d97629e1c564e3f714a4f ] The cs35l56->component pointer is used by the suspend-resume handling to know whether the driver is fully instantiated. This is to prevent it queuing dsp_work which would result in calling wm_adsp when the driver is not an instantiated ASoC component. So this pointer must be cleared by cs35l56_component_remove(). Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Link: https://msgid.link/r/20240129162737.497-4-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Stable-dep-of: eba2eb2495f4 ("ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()") Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: qcom: Fix uninitialized pointer dmactlColin Ian King2024-03-061-1/+1
| | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 1382d8b55129875b2e07c4d2a7ebc790183769ee ] In the case where __lpass_get_dmactl_handle is called and the driver id dai_id is invalid the pointer dmactl is not being assigned a value, and dmactl contains a garbage value since it has not been initialized and so the null check may not work. Fix this to initialize dmactl to NULL. One could argue that modern compilers will set this to zero, but it is useful to keep this initialized as per the same way in functions __lpass_platform_codec_intf_init and lpass_cdc_dma_daiops_hw_params. Cleans up clang scan build warning: sound/soc/qcom/lpass-cdc-dma.c:275:7: warning: Branch condition evaluates to a garbage value [core.uninitialized.Branch] Fixes: b81af585ea54 ("ASoC: qcom: Add lpass CPU driver for codec dma control") Signed-off-by: Colin Ian King <colin.i.king@gmail.com> Link: https://msgid.link/r/20240221134804.3475989-1-colin.i.king@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: Drop leftover snd-rtctimer stuff from MakefileTakashi Iwai2024-03-061-1/+0
| | | | | | | | | | | | [ Upstream commit 4df49712eb54141be00a9312547436d55677f092 ] We forgot to remove the line for snd-rtctimer from Makefile while dropping the functionality. Get rid of the stale line. Fixes: 34ce71a96dcb ("ALSA: timer: remove legacy rtctimer") Link: https://lore.kernel.org/r/20240221092156.28695-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: cs35l56: Must clear HALO_STATE before issuing SYSTEM_RESETRichard Fitzgerald2024-03-061-0/+1
| | | | | | | | | | | | | | | | | | | | | [ Upstream commit e33625c84b75e4f078d7f9bf58f01fe71ab99642 ] The driver must write 0 to HALO_STATE before sending the SYSTEM_RESET command to the firmware. HALO_STATE is in DSP memory, which is preserved across a soft reset. The SYSTEM_RESET command does not change the value of HALO_STATE. There is period of time while the CS35L56 is resetting, before the firmware has started to boot, where a read of HALO_STATE will return the value it had before the SYSTEM_RESET. If the driver does not clear HALO_STATE, this would return BOOT_DONE status even though the firmware has not booted. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file") Link: https://msgid.link/r/20240216140535.1434933-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: Ignore clock selector errors for single connectionAlexander Tsoy2024-03-011-1/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit eaa1b01fe709d6a236a9cec74813e0400601fd23 ] For devices with multiple clock sources connected to a selector, we need to check what a clock selector control request has returned. This is needed to ensure that a requested clock source is indeed selected and for autoclock feature to work. For devices with single clock source connected, if we get an error there is nothing else we can do about it. We can't skip clock selector setup as it is required by some devices. So lets just ignore error in this case. This should fix various buggy Mackie devices: [ 649.109785] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32) [ 649.111946] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32) [ 649.113822] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32) There is also interesting info from the Windows documentation [1] (this is probably why manufacturers dont't even test this feature): "The USB Audio 2.0 driver doesn't support clock selection. The driver uses the Clock Source Entity, which is selected by default and never issues a Clock Selector Control SET CUR request." Link: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/usb-2-0-audio-drivers [1] Link: https://bugzilla.kernel.org/show_bug.cgi?id=217314 Link: https://bugzilla.kernel.org/show_bug.cgi?id=218175 Link: https://bugzilla.kernel.org/show_bug.cgi?id=218342 Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20240201115308.17838-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: wm_adsp: Don't overwrite fwf_name with the defaultRichard Fitzgerald2024-03-011-17/+12
| | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit daf3f0f99cde93a066240462b7a87cdfeedc04c0 ] There's no need to overwrite fwf_name with a kstrdup() of the cs_dsp part name. It is trivial to select either fwf_name or cs_dsp.part as the string to use when building the filename in wm_adsp_request_firmware_file(). This leaves fwf_name entirely owned by the codec driver. It also avoids problems with freeing the pointer. With the original code fwf_name was either a pointer owned by the codec driver, or a kstrdup() created by wm_adsp. This meant wm_adsp must free it if it set it, but not if the codec driver set it. The code was handling this by using devm_kstrdup(). But there is no absolute requirement that wm_adsp_common_init() must be called from probe(), so this was a pseudo-memory leak - each new call to wm_adsp_common_init() would allocate another block of memory but these would only be freed if the owning codec driver was removed. Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://msgid.link/r/20240129162737.497-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda: cs35l41: Support ASUS Zenbook UM3402YARChhayly Leang2024-03-011-0/+2
| | | | | | | | | | | [ Upstream commit be220d2e5544ff094142d263db5cf94d034b5e39 ] Adds sound support for ASUS Zenbook UM3402YAR with missing DSD Signed-off-by: Chhayly Leang <clw.leang@gmail.com> Link: https://lore.kernel.org/r/20240126080912.87422-1-clw.leang@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda: cs35l41: Support additional ASUS Zenbook UX3402VAKenzo Gomez2024-03-011-0/+2
| | | | | | | | | | | [ Upstream commit c16dfab33f99fc3ff43d48253bc2784ccb84c1de ] Add new model entry into configuration table. Signed-off-by: Kenzo Gomez <kenzo.sgomez@gmail.com> Link: https://lore.kernel.org/r/20240127164621.26431-1-kenzo.sgomez@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: sunxi: sun4i-spdif: Add support for Allwinner H616Chen-Yu Tsai2024-03-011-0/+5
| | | | | | | | | | | | | | | | | | [ Upstream commit 0adf963b8463faa44653e22e56ce55f747e68868 ] The SPDIF hardware block found in the H616 SoC has the same layout as the one found in the H6 SoC, except that it is missing the receiver side. Since the driver currently only supports the transmit function, support for the H616 is identical to what is currently done for the H6. Signed-off-by: Chen-Yu Tsai <wens@csie.org> Reviewed-by: Andre Przywara <andre.przywara@arm.com> Reviewed-by: Jernej Skrabec <jernej.skrabec@gmail.com> Link: https://msgid.link/r/20240127163247.384439-4-wens@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: Check presence of valid altsetting controlAlexander Tsoy2024-03-011-0/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 346f59d1e8ed0eed41c80e1acb657e484c308e6a ] Many devices with a single alternate setting do not have a Valid Alternate Setting Control and validation performed by validate_sample_rate_table_v2v3() doesn't work on them and is not really needed. So check the presense of control before sending altsetting validation requests. MOTU Microbook IIc is suffering the most without this check. It takes up to 40 seconds to bootup due to how slow it switches sampling rates: [ 2659.164824] usb 3-2: New USB device found, idVendor=07fd, idProduct=0004, bcdDevice= 0.60 [ 2659.164827] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 2659.164829] usb 3-2: Product: MicroBook IIc [ 2659.164830] usb 3-2: Manufacturer: MOTU [ 2659.166204] usb 3-2: Found last interface = 3 [ 2679.322298] usb 3-2: No valid sample rate available for 1:1, assuming a firmware bug [ 2679.322306] usb 3-2: 1:1: add audio endpoint 0x3 [ 2679.322321] usb 3-2: Creating new data endpoint #3 [ 2679.322552] usb 3-2: 1:1 Set sample rate 96000, clock 1 [ 2684.362250] usb 3-2: 2:1: cannot get freq (v2/v3): err -110 [ 2694.444700] usb 3-2: No valid sample rate available for 2:1, assuming a firmware bug [ 2694.444707] usb 3-2: 2:1: add audio endpoint 0x84 [ 2694.444721] usb 3-2: Creating new data endpoint #84 [ 2699.482103] usb 3-2: 2:1 Set sample rate 96000, clock 1 Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20240129121254.3454481-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda: Increase default bdl_pos_adj for Apollo LakeRui Salvaterra2024-03-011-0/+2
| | | | | | | | | | | | | | | | | | | | [ Upstream commit 56beedc88405fd8022edfd1c2e63d1bc6c95efcb ] Apollo Lake seems to also suffer from IRQ timing issues. After being up for ~4 minutes, a Pentium N4200 system ends up falling back to workqueue-based IRQ handling: [ 208.019906] snd_hda_intel 0000:00:0e.0: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. Unfortunately, the Baytrail and Braswell workaround value of 32 samples isn't enough to fix the issue here. Default to 64 samples. Signed-off-by: Rui Salvaterra <rsalvaterra@gmail.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20240122114512.55808-3-rsalvaterra@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda: Replace numeric device IDs with constant valuesRui Salvaterra2024-03-011-2/+2
| | | | | | | | | | | | | [ Upstream commit 3526860f26febbe46960f9b37f5dbd5ccc109ea8 ] We have self-explanatory constants for Intel HDA devices, let's use them instead of magic numbers and code comments. Signed-off-by: Rui Salvaterra <rsalvaterra@gmail.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20240122114512.55808-2-rsalvaterra@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: amd: acp: Add check for cpu dai link initializationVenkata Prasad Potturu2024-03-011-2/+7
| | | | | | | | | | | | | [ Upstream commit 6cc2aa9a75f2397d42b78d4c159bc06722183c78 ] Add condition check for cpu dai link initialization for amplifier codec path, as same pcm id uses for both headset and speaker path for RENOIR platforms. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com> Link: https://msgid.link/r/20240118143023.1903984-3-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: amd: yc: Add DMI quirk for Lenovo Ideapad Pro 5 16ARP8Mario Limonciello2024-02-231-0/+7
| | | | | | | | | | | | | | | commit 610010737f74482a61896596a0116876ecf9e65c upstream. The laptop requires a quirk ID to enable its internal microphone. Add it to the DMI quirk table. Reported-by: Stanislav Petrov <stanislav.i.petrov@gmail.com> Closes: https://bugzilla.kernel.org/show_bug.cgi?id=216925 Cc: stable@vger.kernel.org Signed-off-by: Mario Limonciello <mario.limonciello@amd.com> Link: https://lore.kernel.org/r/20240205214853.2689-1-mario.limonciello@amd.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: tas2781: add module parameter to tascodec_init()Gergo Koteles2024-02-233-3/+4
| | | | | | | | | | | | | | | | | | | | | | commit 34a1066981a967eab619938e7b35a9be6b4c34e1 upstream. The tascodec_init() of the snd-soc-tas2781-comlib module is called from snd-soc-tas2781-i2c and snd-hda-scodec-tas2781-i2c modules. It calls request_firmware_nowait() with parameter THIS_MODULE and a cont/callback from the latter modules. The latter modules can be removed while their callbacks are running, resulting in a general protection failure. Add module parameter to tascodec_init() so request_firmware_nowait() can be called with the module of the callback. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles <soyer@irl.hu> Link: https://lore.kernel.org/r/118dad922cef50525e5aab09badef2fa0eb796e5.1707076603.git.soyer@irl.hu Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: SOF: IPC3: fix message bounds on ipc opsCurtis Malainey2024-02-231-1/+1
| | | | | | | | | | | | | | | | | | | | | commit fcbe4873089c84da641df75cda9cac2e9addbb4b upstream. commit 74ad8ed65121 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops") introduced a new allocation before the upper bounds check in do_rx_work. As a result A DSP can cause bad allocations if spewing garbage. Fixes: 74ad8ed65121 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops") Reported-by: Tim Van Patten <timvp@google.com> Cc: stable@vger.kernel.org Signed-off-by: Curtis Malainey <cujomalainey@chromium.org> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://msgid.link/r/20240213123834.4827-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: add IDs for Dell dual spk platformShuming Fan2024-02-231-0/+2
| | | | | | | | | | | | commit fddab35fd064414c677e9488c4fb3a1f67725d37 upstream. This patch adds another two IDs for the Dell dual speaker platform. Signed-off-by: Shuming Fan <shumingf@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240205072252.3791500-1-shumingf@realtek.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/conexant: Add quirk for SWS JS201Dbo liu2024-02-231-0/+18
| | | | | | | | | | | | | commit 4639c5021029d49fd2f97fa8d74731f167f98919 upstream. The SWS JS201D need a different pinconfig from windows driver. Add a quirk to use a specific pinconfig to SWS JS201D. Signed-off-by: bo liu <bo.liu@senarytech.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240205013802.51907-1-bo.liu@senarytech.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: fix mute/micmute LED For HP mt645Eniac Zhang2024-02-231-0/+2
| | | | | | | | | | | | | | | | | | commit 32f03f4002c5df837fb920eb23fcd2f4af9b0b23 upstream. The HP mt645 G7 Thin Client uses an ALC236 codec and needs the ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make the mute and micmute LEDs work. There are two variants of the USB-C PD chip on this device. Each uses a different BIOS and board ID, hence the two entries. Signed-off-by: Eniac Zhang <eniac-xw.zhang@hp.com> Signed-off-by: Alexandru Gagniuc <alexandru.gagniuc@hp.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240215154922.778394-1-alexandru.gagniuc@hp.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook PowerAndy Chi2024-02-231-0/+2
| | | | | | | | | | | | | | commit 1513664f340289cf10402753110f3cff12a738aa upstream. The HP ZBook Power using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240122074826.1020964-1-andy.chi@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/cs8409: Suppress vmaster control for Dolphin modelsVitaly Rodionov2024-02-231-0/+1
| | | | | | | | | | | | | | | | | commit a2ed0a44d637ef9deca595054c206da7d6cbdcbc upstream. Customer has reported an issue with specific desktop platform where two CS42L42 codecs are connected to CS8409 HDA bridge. If "Master Volume Control" is created then on Ubuntu OS UCM left/right balance slider in UI audio settings has no effect. This patch will fix this issue for a target paltform. Fixes: 20e507724113 ("ALSA: hda/cs8409: Add support for dolphin") Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240122184710.5802-1-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: wcd938x: handle deferred probeKrzysztof Kozlowski2024-02-231-1/+1
| | | | | | | | | | | | | | | | | | commit 086df711d9b886194481b4fbe525eb43e9ae7403 upstream. WCD938x sound codec driver ignores return status of getting regulators and returns EINVAL instead of EPROBE_DEFER. If regulator provider probes after the codec, system is left without probed audio: wcd938x_codec audio-codec: wcd938x_probe: Fail to obtain platform data wcd938x_codec: probe of audio-codec failed with error -22 Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver") Cc: <stable@vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Link: https://msgid.link/r/20240117151208.1219755-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Add speaker pin verbtable for Dell dual speaker platformKailang Yang2024-02-231-2/+5
| | | | | | | | | | | | | | commit fcfc9f711d1e2fc7876ac12b1b16c509404b9625 upstream. SSID 0x0c0d platform. It can't mute speaker when HP plugged. This patch add quirk to fill speaker pin verbtable. And disable speaker passthrough. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/38b82976a875451d833d514cee34ff6a@realtek.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: Enable headset mic on Vaio VJFE-ADLEdson Juliano Drosdeck2024-02-231-0/+1
| | | | | | | | | | | | | commit c7de2d9bb68a5fc71c25ff96705a80a76c8436eb upstream. Vaio VJFE-ADL is equipped with ALC269VC, and it needs ALC298_FIXUP_SPK_VOLUME quirk to make its headset mic work. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240201122114.30080-1-edson.drosdeck@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: Apply headset jack quirk for non-bass alc287 thinkpadsJosé Relvas2024-02-231-1/+3
| | | | | | | | | | | | | | | | | | | | | commit 2468e8922d2f6da81a6192b73023eff67e3fefdd upstream. There currently exists two thinkpad headset jack fixups: ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK ALC285_FIXUP_THINKPAD_HEADSET_JACK The latter is applied to alc285 and alc287 thinkpads which contain bass speakers. However, the former was only being applied to alc285 thinkpads, leaving non-bass alc287 thinkpads with no headset button controls. This patch fixes that by adding ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK to the alc287 chains, allowing the detection of headset buttons. Signed-off-by: José Relvas <josemonsantorelvas@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240131113407.34698-3-josemonsantorelvas@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: Enable Mute LED on HP Laptop 14-fq0xxxLuka Guzenko2024-02-231-0/+1
| | | | | | | | | | | | | commit f0d78972f27dc1d1d51fbace2713ad3cdc60a877 upstream. This HP Laptop uses ALC236 codec with COEF 0x07 controlling the mute LED. Enable existing quirk for this device. Signed-off-by: Luka Guzenko <l.guzenko@web.de> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240128155704.2333812-1-l.guzenko@web.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: Fix the external mic not being recognised for Acer Swift ↵David Senoner2024-02-231-0/+1
| | | | | | | | | | | | | | | | | 1 SF114-32 commit efb56d84dd9c3de3c99fc396abb57c6d330038b5 upstream. If you connect an external headset/microphone to the 3.5mm jack on the Acer Swift 1 SF114-32 it does not recognize the microphone. This fixes that and gives the user the ability to choose between internal and headset mic. Signed-off-by: David Senoner <seda18@rolmail.net> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240126155626.2304465-1-seda18@rolmail.net Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: amd: yc: Add DMI quirk for MSI Bravo 15 C7VFTechno Mooney2024-02-231-0/+7
| | | | | | | | | | | | | | | | commit c6dce23ec993f7da7790a9eadb36864ceb60e942 upstream. The laptop requires a quirk ID to enable its internal microphone. Add it to the DMI quirk table. Reported-by: Techno Mooney <techno.mooney@gmail.com> Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218402 Cc: stable@vger.kernel.org Signed-off-by: Techno Mooney <techno.mooney@gmail.com> Signed-off-by: Bagas Sanjaya <bagasdotme@gmail.com> Link: https://msgid.link/r/20240129081148.1044891-1-bagasdotme@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: rt5645: Fix deadlock in rt5645_jack_detect_work()Alexey Khoroshilov2024-02-231-0/+1
| | | | | | | | | | | | | | | | [ Upstream commit 6ef5d5b92f7117b324efaac72b3db27ae8bb3082 ] There is a path in rt5645_jack_detect_work(), where rt5645->jd_mutex is left locked forever. That may lead to deadlock when rt5645_jack_detect_work() is called for the second time. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: cdba4301adda ("ASoC: rt5650: add mutex to avoid the jack detection failure") Signed-off-by: Alexey Khoroshilov <khoroshilov@ispras.ru> Link: https://lore.kernel.org/r/1707645514-21196-1-git-send-email-khoroshilov@ispras.ru Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda/cs35l56: select intended config FW_CS_DSPLukas Bulwahn2024-02-231-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit e5aa6d51a2ef8c7ef7e3fe76bebe530fb68e7f08 ] Commit 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") adds configs SND_HDA_SCODEC_CS35L56_{I2C,SPI}, which selects the non-existing config CS_DSP. Note the renaming in commit d7cfdf17cb9d ("firmware: cs_dsp: Rename KConfig symbol CS_DSP -> FW_CS_DSP"), though. Select the intended config FW_CS_DSP. This broken select command probably was not noticed as the configs also select SND_HDA_CS_DSP_CONTROLS and this then selects FW_CS_DSP. So, the select FW_CS_DSP could actually be dropped, but we will keep this redundancy in place as the author originally also intended to have this redundancy of selects in place. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Lukas Bulwahn <lukas.bulwahn@gmail.com> Reviewed-by: Simon Trimmer <simont@opensource.cirrus.com> Link: https://lore.kernel.org/r/20240209082044.3981-1-lukas.bulwahn@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: SOF: ipc3-topology: Fix pipeline tear down logicRanjani Sridharan2024-02-231-14/+41
| | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit d7332c4a4f1a7d16f054c6357fb65c597b6a86a7 ] With the change in the widget free logic to power down the cores only when the scheduler widgets are freed, we need to ensure that the scheduler widget is freed only after all the widgets associated with the scheduler are freed. This is to ensure that the secondary core that the scheduler is scheduled to run on is kept powered on until all widgets that need them are in use. While this works well for dynamic pipelines, in the case of static pipelines the current logic does not take this into account and frees all widgets in the order they occur in the widget_list. So, modify this to ensure that the scheduler widgets are freed only after all other types of widgets in the widget_list are freed. Link: https://github.com/thesofproject/linux/issues/4807 Fixes: 31ed8da1c8e5 ("ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores") Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Link: https://lore.kernel.org/r/20240208133432.1688-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: Intel: avs: Fix dynamic port assignment when TDM is setAmadeusz Sławiński2024-02-231-1/+1
| | | | | | | | | | | | | | | [ Upstream commit 44d3b8a19b91cd2af11f918b2fd05628383172de ] In case TDM is set in topology on SSP0, parser will overwrite vindex value, because it only checks if port is set. Fix this by checking whole field value. Fixes: e6d50e474e45 ("ASoC: Intel: avs: Improve topology parsing of dynamic strings") Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20240207112624.2132821-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: Intel: avs: Fix pci_probe() error pathCezary Rojewski2024-02-231-0/+3
| | | | | | | | | | | | | | | [ Upstream commit b5fbde22684af5456d1de60758950944d69d69ad ] Recent changes modified operation-order in the probe() function without updating its error path accordingly. If snd_hdac_i915_init() exists with status EPROBE_DEFER the error path must cleanup allocated IRQs before leaving the scope. Fixes: 2dddc514b6e4 ("ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work.") Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20240202114901.1002127-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: Sort quirk table entriesTakashi Iwai2024-02-161-18/+18
| | | | | | | | | | | | | | | | | | commit 668abe6dc7b61941fa5c724c06797efb0b87f070 upstream. The quirk table entries should be put in the USB ID order, but some entries have been put in random places. Re-sort them. Fixes: bf990c102319 ("ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue") Fixes: fd28941cff1c ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless") Fixes: dfd5fe19db7d ("ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless") Fixes: 4a63e68a2951 ("ALSA: usb-audio: Fix microphone sound on Nexigo webcam.") Fixes: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+") Fixes: 4fb7c24f69c4 ("ALSA: usb-audio: Add quirk for Fiero SC-01") Fixes: 2307a0e1ca0b ("ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0)") Link: https://lore.kernel.org/r/20240124155307.16996-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* Revert "ASoC: amd: Add new dmi entries for acp5x platform"Greg Kroah-Hartman2024-02-161-14/+1
| | | | | | | | | | | | | This reverts commit 4b6986b170f2f23e390bbd2d50784caa9cb67093 which is commit c3ab23a10771bbe06300e5374efa809789c65455 upstream. Link: https://lore.kernel.org/r/CAD_nV8BG0t7US=+C28kQOR==712MPfZ9m-fuKksgoZCgrEByCw@mail.gmail.com Reported-by: Ted Chang <tedchang2010@gmail.com> Cc: Takashi Iwai <tiwai@suse.de> Cc: Venkata Prasad Potturu <venkataprasad.potturu@amd.com> Cc: Mark Brown <broonie@kernel.org> Cc: Sasha Levin <sashal@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: add quirk for RODE NT-USB+Sean Young2024-02-161-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 7822baa844a87cbb93308c1032c3d47d4079bb8a upstream. The RODE NT-USB+ is marketed as a professional usb microphone, however the usb audio interface is a mess: [ 1.130977] usb 1-5: new full-speed USB device number 2 using xhci_hcd [ 1.503906] usb 1-5: config 1 has an invalid interface number: 5 but max is 4 [ 1.503912] usb 1-5: config 1 has no interface number 4 [ 1.519689] usb 1-5: New USB device found, idVendor=19f7, idProduct=0035, bcdDevice= 1.09 [ 1.519695] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 1.519697] usb 1-5: Product: RØDE NT-USB+ [ 1.519699] usb 1-5: Manufacturer: RØDE [ 1.519700] usb 1-5: SerialNumber: 1D773A1A [ 8.327495] usb 1-5: 1:1: cannot get freq at ep 0x82 [ 8.344500] usb 1-5: 1:2: cannot get freq at ep 0x82 [ 8.365499] usb 1-5: 2:1: cannot get freq at ep 0x2 Add QUIRK_FLAG_GET_SAMPLE_RATE to work around the broken sample rate get. I have asked Rode support to fix it, but they show no interest. Signed-off-by: Sean Young <sean@mess.org> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240124151524.23314-1-sean@mess.org Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Add a quirk for Yamaha YIT-W12TX transmitterJulian Sikorski2024-02-161-0/+2
| | | | | | | | | | | | | commit a969210066054ea109d8b7aff29a9b1c98776841 upstream. The device fails to initialize otherwise, giving the following error: [ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1 Signed-off-by: Julian Sikorski <belegdol+github@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20240123084935.2745-1-belegdol+github@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Add delay quirk for MOTU M Series 2nd revisionAlexander Tsoy2024-02-161-0/+2
| | | | | | | | | | | | | | commit d915a6850e27efb383cd4400caadfe47792623df upstream. Audio control requests that sets sampling frequency sometimes fail on this card. Adding delay between control messages eliminates that problem. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601 Cc: <stable@vger.kernel.org> Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20240124130239.358298-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: wsa883x: fix PA volume controlJohan Hovold2024-02-051-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | commit b53cc6144a3f6c8b56afcdec89d81195c9b0dc69 upstream. The PA gain can be set in steps of 1.5 dB from -3 dB to 18 dB, that is, in 15 levels. Fix the dB values for the PA volume control as experiments using wsa8835 show that the first 16 levels all map to the same lowest gain while the last three map to the highest gain. These values specifically need to be correct for the sound server to provide proper volume control. Note that level 0 (-3 dB) does not mute the PA so the mute flag should also not be set. Fixes: cdb09e623143 ("ASoC: codecs: wsa883x: add control, dapm widgets and map") Cc: stable@vger.kernel.org # 6.0 Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Johan Hovold <johan+linaro@kernel.org> Link: https://msgid.link/r/20240119112420.7446-2-johan+linaro@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: lpass-wsa-macro: fix compander volume hackJohan Hovold2024-02-051-7/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 46188db080bd1df7d2d28031b89e56f2fdbabd67 upstream. The LPASS WSA macro codec driver is updating the digital gain settings behind the back of user space on DAPM events if companding has been enabled. As compander control is exported to user space, this can result in the digital gain setting being incremented (or decremented) every time the sound server is started and the codec suspended depending on what the UCM configuration looks like. Soon enough playback will become distorted (or too quiet). This is specifically a problem on the Lenovo ThinkPad X13s as this bypasses the limit for the digital gain setting that has been set by the machine driver. Fix this by simply dropping the compander gain offset hack. If someone cares about modelling the impact of the compander setting this can possibly be done by exporting it as a volume control later. Note that the volume registers still need to be written after enabling clocks in order for any prior updates to take effect. Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route") Cc: stable@vger.kernel.org # 5.11 Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Johan Hovold <johan+linaro@kernel.org> Link: https://msgid.link/r/20240119112420.7446-4-johan+linaro@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: wcd938x: fix headphones volume controlsJohan Hovold2024-02-051-1/+1
| | | | | | | | | | | | | | | | | | commit 4d0e8bdfa4a57099dc7230952a460903f2e2f8de upstream. The lowest headphones volume setting does not mute so the leave the TLV mute flag unset. This is specifically needed to let the sound server use the lowest gain setting. Fixes: c03226ba15fe ("ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR") Cc: <stable@vger.kernel.org> # 6.5 Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Johan Hovold <johan+linaro@kernel.org> Link: https://msgid.link/r/20240122091130.27463-1-johan+linaro@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: qcom: sc8280xp: limit speaker volumesJohan Hovold2024-02-051-5/+7
| | | | | | | | | | | | | | | | | | | | | | | | | commit c481016bb4f8a9c059c39ac06e7b65e233a61f6a upstream. The UCM configuration for the Lenovo ThinkPad X13s has up until now been setting the speaker PA volume to the minimum -3 dB when enabling the speakers, but this does not prevent the user from increasing the volume further. Limit the digital gain and PA volumes to a combined -3 dB in the machine driver to reduce the risk of speaker damage until we have active speaker protection in place (or higher safe levels have been established). Note that the PA volume limit cannot be set lower than 0 dB or PulseAudio gets confused when the first 16 levels all map to -3 dB. Also note that this will probably need to be generalised using machine-specific limits, but a common limit should do for now. Cc: <stable@vger.kernel.org> # 6.5 Signed-off-by: Johan Hovold <johan+linaro@kernel.org> Link: https://msgid.link/r/20240122181819.4038-3-johan+linaro@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: rtq9128: Fix TDM enable and DAI format control flowChiYuan Huang2024-02-051-31/+36
| | | | | | | | | | | | | | | [ Upstream commit 415d10ccef712f3ec73cd880c1fef3eb48601c3a ] To enable TDM mode, the current control flow limits the function calling order should be 'set_tdm_slot->set_dai_fmt'. But not all platform sound card like as simeple card to follow this design. To bypass this limit, adjust the DAI format setting in runtime 'hw_param' callback. Signed-off-by: ChiYuan Huang <cy_huang@richtek.com> Link: https://msgid.link/r/c4c8df00d8d179b8b5b39a8521de3a85325c57e8.1703813842.git.cy_huang@richtek.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
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