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2021-05-14ALSA: hda/realtek: Re-order ALC882 Clevo quirk table entriesTakashi Iwai1-11/+11
commit 13e1a4cd490b959a4c72c9f4fb502ef56b190062 upstream. Just re-order the alc882_fixup_tbl[] entries for Clevo devices for avoiding the oversight of the duplicated or unapplied item in future. No functional changes. Also, user lower hex letters in the entry. Also Cc-to-stable for the further patch applications. Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210428112704.23967-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-14ALSA: hda/realtek: Re-order ALC882 Sony quirk table entriesTakashi Iwai1-2/+2
commit b7529c18feecb1af92f9db08c8e7fe446a82d96d upstream. Just re-order the alc882_fixup_tbl[] entries for Sony devices for avoiding the oversight of the duplicated or unapplied item in future. No functional changes. Also Cc-to-stable for the further patch applications. Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210428112704.23967-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-14ALSA: hda/realtek: Re-order ALC882 Acer quirk table entriesTakashi Iwai1-3/+3
commit b265047ac56bad8c4f3d0c8bf9cb4e828ee0d28e upstream. Just re-order the alc882_fixup_tbl[] entries for Acer devices for avoiding the oversight of the duplicated or unapplied item in future. No functional changes. Also Cc-to-stable for the further patch applications. Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210428112704.23967-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-14ASoC: tlv320aic32x4: Increase maximum register in regmapAnnaliese McDermond1-2/+2
commit 29654ed8384e9dbaf4cfba689dbcb664a6ab4bb7 upstream. AIC32X4_REFPOWERUP was added as a register, but the maximum register value in the regmap and regmap range was not correspondingly increased. This caused an error when this register was attempted to be written. Fixes: ec96690de82c ("ASoC: tlv320aic32x4: Enable fast charge") Cc: stable@vger.kernel.org Signed-off-by: Annaliese McDermond <nh6z@nh6z.net> Link: https://lore.kernel.org/r/0101017889851cab-ce60cfdb-d88c-43d8-bbd2-7fbf34a0c912-000000@us-west-2.amazonses.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-14ASoC: tlv320aic32x4: Register clocks before registering componentAnnaliese McDermond1-4/+4
commit 1ca1156cfd69530e6b7cb99943baf90c8bd871a5 upstream. Clock registration must be performed before the component is registered. aic32x4_component_probe attempts to get all the clocks right off the bat. If the component is registered before the clocks there is a race condition where the clocks may not be registered by the time aic32x4_componet_probe actually runs. Fixes: d1c859d314d8 ("ASoC: codec: tlv3204: Increased maximum supported channels") Cc: stable@vger.kernel.org Signed-off-by: Annaliese McDermond <nh6z@nh6z.net> Link: https://lore.kernel.org/r/0101017889850206-dcac4cce-8cc8-4a21-80e9-4e4bef44b981-000000@us-west-2.amazonses.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-14ASoC: Intel: kbl_da7219_max98927: Fix kabylake_ssp_fixup functionLukasz Majczak1-8/+30
commit a523ef731ac6674dc07574f31bf44cc5bfa14e4d upstream. kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Tested for all use cases of the driver. Based on similar fix in kbl_rt5663_rt5514_max98927.c from Harsha Priya <harshapriya.n@intel.com> and Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com> Cc: <stable@vger.kernel.org> # 5.4+ Signed-off-by: Lukasz Majczak <lma@semihalf.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210415124347.475432-1-lma@semihalf.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-14ASoC: samsung: tm2_wm5110: check of of_parse return valuePierre-Louis Bossart1-1/+1
commit d58970da324732686529655c21791cef0ee547c4 upstream. cppcheck warning: sound/soc/samsung/tm2_wm5110.c:605:6: style: Variable 'ret' is reassigned a value before the old one has been used. [redundantAssignment] ret = devm_snd_soc_register_component(dev, &tm2_component, ^ sound/soc/samsung/tm2_wm5110.c:554:7: note: ret is assigned ret = of_parse_phandle_with_args(dev->of_node, "i2s-controller", ^ sound/soc/samsung/tm2_wm5110.c:605:6: note: ret is overwritten ret = devm_snd_soc_register_component(dev, &tm2_component, ^ The args is a stack variable, so it could have junk (uninitialized) therefore args.np could have a non-NULL and random value even though property was missing. Later could trigger invalid pointer dereference. There's no need to check for args.np because args.np won't be initialized on errors. Fixes: 8d1513cef51a ("ASoC: samsung: Add support for HDMI audio on TM2 board") Cc: <stable@vger.kernel.org> Suggested-by: Krzysztof Kozlowski <krzk@kernel.org> Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com> Reviewed-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210312180231.2741-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: hda/realtek: Add quirk for Intel Clevo PCx0DxEckhart Mohr1-0/+2
commit 970e3012c04c96351c413f193a9c909e6d871ce2 upstream. This applies a SND_PCI_QUIRK(...) to the Clevo PCx0Dx barebones. This fix enables audio output over the headset jack and ensures that a microphone connected via the headset combo jack is correctly recognized when pluged in. [ Rearranged the list entries in a sorted order -- tiwai ] Signed-off-by: Eckhart Mohr <e.mohr@tuxedocomputers.com> Co-developed-by: Werner Sembach <wse@tuxedocomputers.com> Signed-off-by: Werner Sembach <wse@tuxedocomputers.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210427153025.451118-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: hda/realtek: fix static noise on ALC285 Lenovo laptopsSami Loone1-5/+4
commit 9bbb94e57df135ef61bef075d9c99b8d9e89e246 upstream. Remove a duplicate vendor+subvendor pin fixup entry as one is masking the other and making it unreachable. Consider the more specific newcomer as a second chance instead. The generic entry is made less strict to also match for laptops with slightly different 0x12 pin configuration. Tested on Lenovo Yoga 6 (AMD) where 0x12 is 0x40000000. Fixes: 607184cb1635 ("ALSA: hda/realtek - Add supported for more Lenovo ALC285 Headset Button") Signed-off-by: Sami Loone <sami@loone.fi> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/YIXS+GT/dGI/LtK6@yoga Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: hda/realtek - Headset Mic issue on HP platformKailang Yang1-0/+2
commit 1c9d9dfd2d254211cb37b1513b1da3e6835b8f00 upstream. Boot with plugged headset, the Headset Mic will be gone. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/207eecfc3189466a820720bc0c409ea9@realtek.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: hda/realtek: fix mic boost on Intel NUC 8Phil Calvin1-0/+13
commit d1ee66c5d3c5a0498dd5e3f2af5b8c219a98bba5 upstream. Fix two bugs with the Intel HDA Realtek ALC233 sound codec present in Intel NUC NUC8i7BEH and probably a few other similar NUC models. These codecs advertise a 4-level microphone input boost amplifier on pin 0x19, but the highest two boost settings do not work correctly, and produce only low analog noise that does not seem to contain any discernible signal. There is an existing fixup for this exact problem but for a different PCI subsystem ID, so we re-use that logic. Changing the boost level also triggers a DC spike in the input signal that bleeds off over about a second and overwhelms any input during that time. Thankfully, the existing fixup has the side effect of making the boost control show up in userspace as a mute/unmute switch, and this keeps (e.g.) PulseAudio from fiddling with it during normal input volume adjustments. Finally, the NUC hardware has built-in inverted stereo mics. This patch also enables the usual fixup for this so the two channels cancel noise instead of the actual signal. [ Re-ordered the quirk entry point by tiwai ] Signed-off-by: Phil Calvin <phil@philcalvin.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/80dc5663-7734-e7e5-25ef-15b5df24511a@philcalvin.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: hda/realtek: GA503 use same quirks as GA401Luke D Jones1-0/+1
commit 76fae6185f5456865ff1bcb647709d44fd987eb6 upstream. The GA503 has almost exactly the same default setup as the GA401 model with the same issues. The GA401 quirks solve all the issues so we will use the full quirk chain. Signed-off-by: Luke D Jones <luke@ljones.dev> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210419030411.28304-1-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G7Jonas Witschel1-0/+25
commit 75b62ab65d2715ce6ff0794033d61ab9dc4a2dfc upstream. The HP ProBook 445 G7 (17T32ES) uses ALC236. Like ALC236_FIXUP_HP_GPIO_LED, COEF index 0x34 bit 5 is used to control the playback mute LED, but the microphone mute LED is controlled using pin VREF instead of a COEF index. AlsaInfo: https://alsa-project.org/db/?f=0d3f4d1af39cc359f9fea9b550727ee87e5cf45a Signed-off-by: Jonas Witschel <diabonas@archlinux.org> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210416105852.52588-1-diabonas@archlinux.org Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: usb-audio: Add dB range mapping for Sennheiser Communications Headset PC 8Timo Gurr1-0/+12
commit ab2165e2e6ed17345ffa8ee88ca764e8788ebcd7 upstream. The decibel volume range contains a negative maximum value resulting in pipewire complaining about the device and effectivly having no sound output. The wrong values also resulted in the headset sounding muted already at a mixer level of about ~25%. PipeWire BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1049 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212897 Signed-off-by: Timo Gurr <timo.gurr@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210503110822.10222-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: usb-audio: Explicitly set up the clock selectorTakashi Iwai1-4/+14
commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964 upstream. In the current code, we have some assumption that the audio clock selector has been set up implicitly and don't want to touch it unless it's really needed for the fallback autoclock setup. This works for most devices but some seem having a problem. Partially this was covered for the devices with a single connector at the initialization phase (commit 086b957cc17f "ALSA: usb-audio: Skip the clock selector inquiry for single connections"), but also there are cases where the wrong clock set up is kept silently. The latter seems to be the cause of the noises on Behringer devices. In this patch, we explicitly set up the audio clock selector whenever the appropriate node is found. Reported-by: Geraldo Nascimento <geraldogabriel@gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199327 Link: https://lore.kernel.org/r/CAEsQvcvF7LnO8PxyyCxuRCx=7jNeSCvFAd-+dE0g_rd1rOxxdw@mail.gmail.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210413084152.32325-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: sb: Fix two use after free in snd_sb_qsound_buildLv Yunlong1-2/+6
commit 4fb44dd2c1dda18606348acdfdb97e8759dde9df upstream. In snd_sb_qsound_build, snd_ctl_add(..,p->qsound_switch...) and snd_ctl_add(..,p->qsound_space..) are called. But the second arguments of snd_ctl_add() could be freed via snd_ctl_add_replace() ->snd_ctl_free_one(). After the error code is returned, snd_sb_qsound_destroy(p) is called in __error branch. But in snd_sb_qsound_destroy(), the freed p->qsound_switch and p->qsound_space are still used by snd_ctl_remove(). My patch set p->qsound_switch and p->qsound_space to NULL if snd_ctl_add() failed to avoid the uaf bugs. But these codes need to further be improved with the code style. Signed-off-by: Lv Yunlong <lyl2019@mail.ustc.edu.cn> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210426145541.8070-1-lyl2019@mail.ustc.edu.cn Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: hda/conexant: Re-order CX5066 quirk table entriesTakashi Iwai1-7/+7
commit 2e6a731296be9d356fdccee9fb6ae345dad96438 upstream. Just re-order the cx5066_fixups[] entries for HP devices for avoiding the oversight of the duplicated or unapplied item in future. No functional changes. Also Cc-to-stable for the further patch applications. Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210428112704.23967-14-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-11ALSA: emu8000: Fix a use after free in snd_emu8000_create_mixerLv Yunlong1-1/+3
commit 1c98f574403dbcf2eb832d5535a10d967333ef2d upstream. Our code analyzer reported a uaf. In snd_emu8000_create_mixer, the callee snd_ctl_add(..,emu->controls[i]) calls snd_ctl_add_replace(.., kcontrol,..). Inside snd_ctl_add_replace(), if error happens, kcontrol will be freed by snd_ctl_free_one(kcontrol). Then emu->controls[i] points to a freed memory, and the execution comes to __error branch of snd_emu8000_create_mixer. The freed emu->controls[i] is used in snd_ctl_remove(card, emu->controls[i]). My patch set emu->controls[i] to NULL if snd_ctl_add() failed to avoid the uaf. Signed-off-by: Lv Yunlong <lyl2019@mail.ustc.edu.cn> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210426131129.4796-1-lyl2019@mail.ustc.edu.cn Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-05-07ALSA: usb-audio: Add MIDI quirk for Vox ToneLab EXTakashi Iwai1-0/+10
commit 64f40f9be14106e7df0098c427cb60be645bddb7 upstream. ToneLab EX guitar pedal device requires the same quirk like ToneLab ST for supporting the MIDI. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212593 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210407144549.1530-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-21ASoC: fsl_esai: Fix TDM slot setup for I2S modeAlexander Shiyan1-3/+5
[ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] 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 <shc_work@mail.ru> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-21ASoC: max98373: Added 30ms turn on/off time delayRyan Lee1-0/+2
[ Upstream commit 3a27875e91fb9c29de436199d20b33f9413aea77 ] Amp requires 10 ~ 30ms for the power ON and OFF. Added 30ms delay for stability. Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com> Link: https://lore.kernel.org/r/20210325033555.29377-2-ryans.lee@maximintegrated.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-21ASoC: max98373: Changed amp shutdown register as volatileRyan Lee2-0/+2
[ Upstream commit a23f9099ff1541f15704e96b784d3846d2a4483d ] 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 <ryans.lee@maximintegrated.com> Link: https://lore.kernel.org/r/20210325033555.29377-1-ryans.lee@maximintegrated.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-14ASoC: sunxi: sun4i-codec: fill ASoC card ownerBastian Germann1-0/+5
[ Upstream commit 7c0d6e482062eb5c06ecccfab340abc523bdca00 ] 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 <bage@linutronix.de> Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-14ASoC: wm8960: Fix wrong bclk and lrclk with pll enabled for some chipsShengjiu Wang1-1/+7
[ Upstream commit 16b82e75c15a7dbd564ea3654f3feb61df9e1e6f ] 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 <shengjiu.wang@nxp.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/1616150926-22892-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-14ASoC: SOF: Intel: HDA: fix core status verificationGuennadi Liakhovetski1-4/+11
[ Upstream commit 927280909fa7d8e61596800d82f18047c6cfbbe4 ] 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 <kai.vehmanen@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210322163728.16616-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-14ASoC: intel: atom: Stop advertising non working S24LE supportHans de Goede1-3/+3
commit aa65bacdb70e549a81de03ec72338e1047842883 upstream. 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 <pierre-louis.bossart@linux.intel.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20210324132711.216152-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-14ALSA: hda/conexant: Apply quirk for another HP ZBook G5 modelTakashi Iwai1-0/+1
commit c6423ed2da6214a68527446b5f8e09cf7162b2ce upstream. 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: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210401171314.667-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-14ALSA: hda/realtek: Fix speaker amp setup on Acer Aspire E1Takashi Iwai1-0/+16
commit c8426b2700b57d2760ff335840a02f66a64b6044 upstream. 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: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210407095730.12560-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-14ALSA: aloop: Fix initialization of controlsJonas Holmberg1-3/+8
commit 168632a495f49f33a18c2d502fc249d7610375e9 upstream. 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 <jonashg@axis.com> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210407075428.2666787-1-jonashg@axis.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-07ALSA: hda/realtek: fix mute/micmute LEDs for HP 640 G8Jeremy Szu1-0/+1
commit 417eadfdd9e25188465280edf3668ed163fda2d0 upstream. 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 <jeremy.szu@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210330114428.40490-1-jeremy.szu@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-07ALSA: hda/realtek: call alc_update_headset_mode() in hp_automute_hookHui Wang1-0/+1
commit e54f30befa7990b897189b44a56c1138c6bfdbb5 upstream. 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: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20210320091542.6748-2-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-07ALSA: hda/realtek: fix a determine_headset_type issue for a Dell AIOHui Wang1-1/+1
commit febf22565549ea7111e7d45e8f2d64373cc66b11 upstream. 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: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20210320091542.6748-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-07ALSA: hda: Add missing sanity checks in PM prepare/complete callbacksTakashi Iwai1-0/+6
commit 66affb7bb0dc0905155a1b2475261aa704d1ddb5 upstream. 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: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210329113059.25035-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-07ALSA: hda: Re-add dropped snd_poewr_change_state() callsTakashi Iwai1-0/+2
commit c8f79808cd8eb5bc8d14de129bd6d586d3fce0aa upstream. 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: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210329113059.25035-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-07ALSA: usb-audio: Apply sample rate quirk to Logitech ConnectIkjoon Jang1-0/+1
commit 625bd5a616ceda4840cd28f82e957c8ced394b6a upstream. 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 <ikjn@chromium.org> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210324105153.2322881-1-ikjn@chromium.org Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2021-04-07ASoC: rt711: add snd_soc_component remove callbackBard Liao1-0/+8
[ Upstream commit 899b12542b0897f92de9ba30944937c39ebb246d ] We do some IO operations in the snd_soc_component_set_jack callback function and snd_soc_component_set_jack() will be called when soc component is removed. However, we should not access SoundWire registers when the bus is suspended. So set regcache_cache_only(regmap, true) to avoid accessing in the soc component removal process. Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: rt5659: Update MCLK rate in set_sysclk()Sameer Pujar1-0/+5
[ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie@kernel.org> Suggested-by: Michael Walle <michael@walle.cc> Signed-off-by: Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: cs42l42: Always wait at least 3ms after resetLucas Tanure2-1/+3
[ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ] This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: cs42l42: Fix mixer volume controlLucas Tanure1-2/+2
[ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ] The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: cs42l42: Fix channel width supportLucas Tanure2-26/+22
[ Upstream commit 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 ] Remove the hard coded 32 bits width and replace with the correct width calculated by params_width. Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: cs42l42: Fix Bitclock polarity inversionLucas Tanure2-17/+14
[ Upstream commit e793c965519b8b7f2fea51a48398405e2a501729 ] The driver was setting bit clock polarity opposite to intended polarity. Also simplify the code by grouping ADC and DAC clock configurations into a single field. Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: soc-core: Prevent warning if no DMI table is presentJon Hunter1-0/+4
[ Upstream commit 7de14d581dbed57c2b3c6afffa2c3fdc6955a3cd ] Many systems do not use ACPI and hence do not provide a DMI table. On non-ACPI systems a warning, such as the following, is printed on boot. WARNING KERN tegra-audio-graph-card sound: ASoC: no DMI vendor name! The variable 'dmi_available' is not exported and so currently cannot be used by kernel modules without adding an accessor. However, it is possible to use the function is_acpi_device_node() to determine if the sound card is an ACPI device and hence indicate if we expect a DMI table to be present. Therefore, call is_acpi_device_node() to see if we are using ACPI and only parse the DMI table if we are booting with ACPI. Signed-off-by: Jon Hunter <jonathanh@nvidia.com> Link: https://lore.kernel.org/r/20210303115526.419458-1-jonathanh@nvidia.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: es8316: Simplify adc_pga_gain_tlv tableHans de Goede1-7/+2
[ Upstream commit bb18c678754ce1514100fb4c0bf6113b5af36c48 ] Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20210228160441.241110-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probeBenjamin Rood1-1/+1
[ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald@curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood@gmail.com> Reviewed-by: Fabio Estevam <festevam@gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10Hans de Goede1-2/+2
[ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10Hans de Goede1-2/+2
[ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-04-07ASoC: rt1015: fix i2c communication errorJack Yu1-0/+1
[ Upstream commit 9e0bdaa9fcb8c64efc1487a7fba07722e7bc515e ] Remove 0x100 cache re-sync to solve i2c communication error. Signed-off-by: Jack Yu <jack.yu@realtek.com> Link: https://lore.kernel.org/r/20210222090057.29532-1-jack.yu@realtek.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-03-30ALSA: hda: ignore invalid NHLT tableMark Pearson1-0/+5
[ Upstream commit a14a6219996ee6f6e858d83b11affc7907633687 ] On some Lenovo systems if the microphone is disabled in the BIOS only the NHLT table header is created, with no data. This means the endpoints field is not correctly set to zero - leading to an unintialised variable and hence invalid descriptors are parsed leading to page faults. The Lenovo firmware team is addressing this, but adding a check preventing invalid tables being parsed is worthwhile. Tested on a Lenovo T14. Tested-by: Philipp Leskovitz <philipp.leskovitz@secunet.com> Reported-by: Philipp Leskovitz <philipp.leskovitz@secunet.com> Signed-off-by: Mark Pearson <markpearson@lenovo.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210302141003.7342-1-markpearson@lenovo.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
2021-03-25ALSA: usb-audio: Fix unintentional sign extension issueColin Ian King1-2/+2
commit 50b1affc891cbc103a2334ce909a026e25f4c84d upstream. The shifting of the u8 in