- 10 Jul, 2020 2 commits
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Takashi Iwai authored
Recently syzkaller reported a UAF in LINE6 driver, and it's likely because we call cancel_delayed_work() at the disconnect callback instead of cancel_delayed_work_sync(). Let's use the correct one instead. Reported-by: syzbot+145012a46658ac00fc9e@syzkaller.appspotmail.com Suggested-by:
Alan Stern <stern@rowland.harvard.edu> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/s5hlfjr4gio.wl-tiwai@suse.deSigned-off-by:
Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
LINE6 drivers create stream URBs with a fixed pipe without checking its validity, and this may lead to a kernel WARNING at the submission when a malformed USB descriptor is passed. For avoiding the kernel warning, perform the similar sanity checks for each pipe type at creating a URB. Reported-by: syzbot+c190f6858a04ea7fbc52@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/s5hv9iv4hq8.wl-tiwai@suse.deSigned-off-by:
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- 07 Jul, 2020 10 commits
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Vinod Koul authored
On partial_drain completion we should be in SNDRV_PCM_STATE_RUNNING state, so set that for partially draining streams in snd_compr_drain_notify() and use a flag for partially draining streams While at it, add locks for stream state change in snd_compr_drain_notify() as well. Fixes: f44f2a54 ("ALSA: compress: fix drain calls blocking other compress functions (v6)") Reviewed-by:
Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Tested-by:
Charles Keepax <ckeepax@opensource.cirrus.com> Tested-by:
Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20200629134737.105993-4-vkoul@kernel.orgSigned-off-by:
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Pavel Hofman authored
USB Audio analyzer RTX6001 uses the same implicit feedback quirk as other XMOS-based devices. Signed-off-by:
Pavel Hofman <pavel.hofman@ivitera.com> Tested-by:
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Hector Martin authored
These devices claim to be 96kHz mono, but actually are 48kHz stereo with swapped channels and unaligned transfers. Cc: stable@vger.kernel.org Signed-off-by:
Hector Martin <marcan@marcan.st> Link: https://lore.kernel.org/r/20200702071433.237843-1-marcan@marcan.stSigned-off-by:
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Jian-Hong Pan authored
The Acer Veriton N4660G desktop's audio (1025:1248) with ALC269VC cannot detect the headset microphone until ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x18 as the headset mic pin. Signed-off-by:
Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200706071826.39726-3-jian-hong@endlessm.comSigned-off-by:
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Jian-Hong Pan authored
The Acer Aspire C20-820 AIO's audio (1025:1065) with ALC269VC can't detect the headset microphone until ALC269VC_FIXUP_ACER_HEADSET_MIC quirk maps the NID 0x18 as the headset mic pin. Signed-off-by:
Daniel Drake <drake@endlessm.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200706071826.39726-2-jian-hong@endlessm.comSigned-off-by:
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Jian-Hong Pan authored
The Acer desktop vCopperbox with ALC269VC cannot detect the MIC of headset, the line out and internal speaker until ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied. Signed-off-by:
Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200706071826.39726-1-jian-hong@endlessm.comSigned-off-by:
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Benjamin Poirier authored
1) In snd_hda_pick_fixup(), quirks are first matched by PCI SSID and then, if there is no match, by codec SSID. The Lenovo "ThinkPad X1 Carbon 7th" has an audio chip with PCI SSID 0x2292 and codec SSID 0x2293[1]. Therefore, fix the quirk meant for that device to match on .subdevice == 0x2292. 2) The "Thinkpad X1 Yoga 7th" does not exist. The companion product to the Carbon 7th is the Yoga 4th. That device has an audio chip with PCI SSID 0x2292 and codec SSID 0x2292[2]. Given the behavior of snd_hda_pick_fixup(), it is not possible to have a separate quirk for the Yoga based on SSID. Therefore, merge the quirks meant for the Carbon and Yoga. This preserves the current behavior for the Yoga. [1] This is the case on my own machine and can also be checked here https://github.com/linuxhw/LsPCI/tree/master/Notebook/Lenovo/ThinkPad https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3225701 [2] https://github.com/linuxhw/LsPCI/tree/master/Convertible/Lenovo/ThinkPad https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3176355 Fixes: d2cd795c ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Fixes: 54a6a7dc ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") Cc: Jaroslav Kysela <perex@perex.cz> Cc: Kailang Yang <kailang@realtek.com> Tested-by:
Vincent Bernat <vincent@bernat.ch> Tested-by:
Even Brenden <evenbrenden@gmail.com> Signed-off-by:
Benjamin Poirier <benjamin.poirier@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200703080005.8942-2-benjamin.poirier@gmail.comSigned-off-by:
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Kai Vehmanen authored
The HDMI codec driver has two debug traces printed from different functions but with identical message content: "HDMI: hinfo 000000006a6b84d9 not registered" Fix this duplication and also add a bit more context in addition to raw object pointer, to help analysis of kernel logs. Reviewed-by:
Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by:
Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-2-kai.vehmanen@linux.intel.comSigned-off-by:
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Kai Vehmanen authored
When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217Reviewed-by:
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xidongwang authored
The stack object “info” in snd_opl3_ioctl() has a leaking problem. It has 2 padding bytes which are not initialized and leaked via “copy_to_user”. Signed-off-by:
xidongwang <wangxidong_97@163.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1594006058-30362-1-git-send-email-wangxidong_97@163.comSigned-off-by:
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- 30 Jun, 2020 3 commits
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Takashi Iwai authored
Merge tag 'asoc-fix-v5.8-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.8 A few small driver specific fixes, nothing particularly dramatic.
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Alexander Tsoy authored
Replace several occurences of "frame" with a "packet" where appropriate. Signed-off-by:
Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200629025934.154288-2-alexander@tsoy.meSigned-off-by:
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Alexander Tsoy authored
Commit f0bd62b6 ("ALSA: usb-audio: Improve frames size computation") introduced a regression for devices which have playback endpoints with bInterval > 1. Fix this by taking ep->datainterval into account. Note that frame and fps are actually mean packet and packets per second in the code introduces by the mentioned commit. This will be fixed in a follow-up patch. Fixes: f0bd62b6 ("ALSA: usb-audio: Improve frames size computation") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208353Signed-off-by:
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- 25 Jun, 2020 2 commits
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Jaroslav Kysela authored
Signed-off-by:
Jaroslav Kysela <perex@perex.cz> Acked-by:
Alex Deucher <alexander.deucher@amd.com> Cc: Mark Brown <broonie@kernel.org> Cc: vijendar.mukunda@amd.com Cc: Alexander.Deucher@amd.com Link: https://lore.kernel.org/r/20200625115829.791750-1-perex@perex.czSigned-off-by:
Mark Brown <broonie@kernel.org>
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Hui Wang authored
We have a Dell AIO, there is neither internal speaker nor internal mic, only a multi-function audio jack on it. Users reported that after freshly installing the OS and plug a headset to the audio jack, the headset can't output sound. I reproduced this bug, at that moment, the Input Source is as below: Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Headphone Mic' 'Headset Mic' Item0: 'Headphone Mic' That is because the patch_realtek will set this audio jack as mic_in mode if Input Source's value is hp_mic. If it is not fresh installing, this issue will not happen since the systemd will run alsactl restore -f /var/lib/alsa/asound.state, this will set the 'Input Source' according to history value. If there is internal speaker or internal mic, this issue will not happen since there is valid sink/source in the pulseaudio, the PA will set the 'Input Source' according to active_port. To fix this issue, change the parser function to let the hs_mic be stored ahead of hp_mic. Cc: stable@vger.kernel.org Signed-off-by:
Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200625083833.11264-1-hui.wang@canonical.comSigned-off-by:
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- 24 Jun, 2020 2 commits
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Shuming Fan authored
To turn the headphone output switch off during jack type detection, it could avoid the pop noise when jack type switches to OMTP type. Signed-off-by:
Shuming Fan <shumingf@realtek.com> Link: https://lore.kernel.org/r/20200623125312.27896-1-shumingf@realtek.comSigned-off-by:
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Takashi Iwai authored
The USB-audio mixer code holds a linked list of usb_mixer_elem_list, and several operations are performed for each mixer element. A few of them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2()) assume each mixer element being a usb_mixer_elem_info object that is a subclass of usb_mixer_elem_list, cast via container_of() and access it members. This may result in an out-of-bound access when a non-standard list element has been added, as spotted by syzkaller recently. This patch adds a new field, is_std_info, in usb_mixer_elem_list to indicate that the element is the usb_mixer_elem_info type or not, and skip the access to such an element if needed. Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.deSigned-off-by:
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- 23 Jun, 2020 4 commits
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Macpaul Lin authored
We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051) need a tiny delay after each class compliant request. Otherwise the device might not be able to be recognized each times. Signed-off-by:
Chihhao Chen <chihhao.chen@mediatek.com> Signed-off-by:
Macpaul Lin <macpaul.lin@mediatek.com> Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/1592910203-24035-1-git-send-email-macpaul.lin@mediatek.comSigned-off-by:
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Shengjiu Wang authored
Fix unchecked return value for clk_prepare_enable, add error handler in fsl_mqs_runtime_resume. Fixes: 9e28f653 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by:
Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by:
Nicolin Chen <nicoleotsuka@gmail.com> Link: https://lore.kernel.org/r/5edd68d03def367d96268f1a9a00bd528ea5aaf2.1592888591.git.shengjiu.wang@nxp.comSigned-off-by:
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Shengjiu Wang authored
Because clk_prepare_enable and clk_disable_unprepare should check input clock parameter is NULL or not internally, then we don't need to check them before calling the function. Fixes: 9e28f653 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by:
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Christoffer Nielsen authored
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud Alpha S (0951:0x16ea) uses two interfaces, but only the second interface contains the capture stream. This patch delays the registration until the second interface appears. Signed-off-by:
Christoffer Nielsen <cn@obviux.dk> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/CAOtG2YHOM3zy+ed9KS-J4HkZo_QGzcUG9MigSp4e4_-13r6B=Q@mail.gmail.comSigned-off-by:
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- 22 Jun, 2020 1 commit
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Takashi Iwai authored
Merge tag 'asoc-fix-v5.8-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.8 This is a collection of mostly small fixes, mostly fixing fallout from some of the DPCM changes that went in last time around which shook out some issues on i.MX and Qualcomm platforms. The addition of a managed version of snd_soc_register_dai() is to fix resource leaks. There's also a few new device IDs for x86 systems.
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- 18 Jun, 2020 3 commits
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Qiushi Wu authored
Calling pm_runtime_get_sync increments the counter even in case of failure, causing incorrect ref count if pm_runtime_put is not called in error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails. Fixes: fc05a5b2 ("ASoC: rockchip: add support for pdm controller") Signed-off-by:
Qiushi Wu <wu000273@umn.edu> Reviewed-by:
Heiko Stuebner <heiko@sntech.de> Link: https://lore.kernel.org/r/20200613205158.27296-1-wu000273@umn.eduSigned-off-by:
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Ravulapati Vishnu vardhan rao authored
The steps to reproduce: Record from the internal mic : (arecord -D hw:1,2 -f dat /dev/null -V stereos) Record from the headphone mic: (arecord -D hw:1,0 -f dat /dev/null -V stereos) Kill the recording from internal mic. We can see the recording from the headphone mic is broken. This patch rectifies the issue reported. Signed-off-by:
Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com> Link: https://lore.kernel.org/r/20200618072653.27103-1-Vishnuvardhanrao.Ravulapati@amd.comSigned-off-by:
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Pierre-Louis Bossart authored
Mirror PCI ids used for SOF. Signed-off-by:
Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Reviewed-by:
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- 17 Jun, 2020 7 commits
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Mark Brown authored
Merge series "ASoC: SOF: Intel: update PCI IDs" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: Clean-up CometLake and add missing PCI IDs. Changes for the legacy driver are sent separately. Pierre-Louis Bossart (3): ASoC: Intel: SOF: merge COMETLAKE_LP and COMETLAKE_H ASoC: SOF: Intel: add PCI ID for CometLake-S ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H sound/hda/intel-dsp-config.c | 4 +--- sound/soc/intel/boards/Kconfig | 4 ++-- sound/soc/sof/intel/Kconfig | 29 ++++++++--------------------- sound/soc/sof/sof-pci-dev.c | 24 ++++++++++++++---------- 4 files changed, 25 insertions(+), 36 deletions(-) -- 2.20.1
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Pierre-Louis Bossart authored
kmemleak throws error reports on module load/unload tests, add snd_hdac_regmap_exit() in .remove(). While we are at it, also fix the error handling flow in .probe() to use snd_hdac_regmap_exit() if needed. Signed-off-by:
Daniel Baluta <daniel.baluta@gmail.com> Reviewed-by:
Rander Wang <rander.wang@linux.intel.com> Reviewed-by:
Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20200617164144.17859-1-pierre-louis.bossart@linux.intel.comSigned-off-by:
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Pierre-Louis Bossart authored
Usually the DSP is not traditionally enabled on H skews but this might be used moving forward. Signed-off-by:
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Pierre-Louis Bossart authored
Mirror ID added for legacy HDaudio Signed-off-by:
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Pierre-Louis Bossart authored
We already have two configurations for CometLake, and a third one coming. On other platforms, we used a single Kconfig option, so we should follow the same trend by merging the two cases in a backwards compatible way. The backwards compatibility is handled by overloading the COMETLAKE_LP kconfig as COMETLAKE. In practice we've never seen a case where COMETLAKE_H is not selected along with COMETLAKE_LP, so keeping one of the two is enough. Signed-off-by:
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Kai-Heng Feng authored
There are two more HP systems control mute LED from HDA codec and need to expose micmute led class so SoF can control micmute LED. Add quirks to support them. Signed-off-by:
Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200617102906.16156-2-kai.heng.feng@canonical.comSigned-off-by:
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Takashi Iwai authored
With the recent full-duplex support of implicit feedback streams, an endpoint can be still running after closing the capture stream as long as the playback stream with the sync-endpoint is running. In such a state, the URBs are still be handled and they may call retire_data_urb callback, which tries to transfer the data from the PCM buffer. Since the PCM stream gets closed, this may lead to use-after-free. This patch adds the proper clearance of the callback at stopping the capture stream for addressing the possible UAF above. Fixes: 10ce77e4 ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback") Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.deSigned-off-by:
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- 16 Jun, 2020 2 commits
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Takashi Iwai authored
MSI GE63 laptop with ALC1220 codec requires the very same quirk (ALC1220_FIXUP_CLEVO_P950) as other MSI devices for the proper sound output. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208057 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200616132150.8778-1-tiwai@suse.deSigned-off-by:
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Shengjiu Wang authored
For mono channel, SSI will switch to Normal mode. In Normal mode and Network mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S Master mode (the word length is fixed to 32bit), it should be the value of params_width(hw_params). The condition "slots == 2" is not good for I2S Master mode, because for Network mode and Normal mode, the slots can also be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) to check if it is I2S Master mode. So we refine the formula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by:
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- 15 Jun, 2020 4 commits
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Mark Brown authored
Merge series "ASoC: topology: fix use-after-free when removing components" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>: This patchset fixes a memory allocation issue and removes a 100% reproducible use-after-free report thrown by KASAN in automated module removal tests across multiple platforms. All the credit goes to Bard Liao for root-causing the issue. DAIs may be registered at the same time as a component, or when the topology is loaded. This two-step registration causes the memory for topology-based DAIs to allocated last, and conversely to be released first by devres, before the component is released and the DAIs removed from the component DAI list with snd_soc_unregister_dais(). When we remove a component, by the time we walk through its dai list to unregister all dais, the dais allocated by the topology have been freed already by devres and the list is corrupted with pointers that are no longer valid. The suggestion is to add an explicit devm_ based registration for topology-based dais, so that each dai is cleanly removed from the component dai list in the release operation before devres releases the allocated memory. Pierre-Louis Bossart (2): ASoC: soc-devres: add devm_snd_soc_register_dai() ASoC: soc-topology: use devm_snd_soc_register_dai() include/sound/soc.h | 4 ++++ sound/soc/soc-devres.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/soc-topology.c | 3 +-- 3 files changed, 42 insertions(+), 2 deletions(-) -- 2.20.1
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Brent Lu authored
Port commit 6d011d50 ("ALSA: hda: Clear RIRB status before reading WP") from legacy HDA driver to fix the get response timeout issue. Current SOF driver does not suffer from this issue because sync write is enabled in hda_init. The issue will come back if the sync write is disabled for some reason. Signed-off-by:
Brent Lu <brent.lu@intel.com> Reviewed-by:
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Jack Yu authored
Update rt1015 default register value according to spec modification. Signed-off-by:
Jack Yu <jack.yu@realtek.com> Link: https://lore.kernel.org/r/20200615032433.31061-1-jack.yu@realtek.comSigned-off-by:
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Srinivas Kandagatla authored
Currently both FE and BE dai-links are configured bi-directional, However the DSP BE dais are only single directional, so set the directions as supported by the BE dais. Fixes: c25e295c (ASoC: qcom: Add support to parse common audio device nodes) Reported-by:
John Stultz <john.stultz@linaro.org> Signed-off-by:
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