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* ASoC: tegra: Fix reference count leaks.Qiushi Wu2020-09-032-2/+6
| | | | | | | | | | | | | | [ Upstream commit deca195383a6085be62cb453079e03e04d618d6e ] 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. Signed-off-by: Qiushi Wu <wu000273@umn.edu> Reviewed-by: Jon Hunter <jonathanh@nvidia.com> Link: https://lore.kernel.org/r/20200613204422.24484-1-wu000273@umn.edu Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: pci: delete repeated words in commentsRandy Dunlap2020-09-037-7/+7
| | | | | | | | | | | | [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap@infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: intel: Fix memleak in sst_media_openDinghao Liu2020-08-261-2/+3
| | | | | | | | | | | | | | | [ Upstream commit 062fa09f44f4fb3776a23184d5d296b0c8872eb9 ] When power_up_sst() fails, stream needs to be freed just like when try_module_get() fails. However, current code is returning directly and ends up leaking memory. Fixes: 0121327c1a68b ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200813084112.26205-1-dinghao.liu@zju.edu.cn Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: msm8916-wcd-analog: fix register Interrupt offsetSrinivas Kandagatla2020-08-261-2/+2
| | | | | | | | | | | | | | | | [ Upstream commit ff69c97ef84c9f7795adb49e9f07c9adcdd0c288 ] For some reason interrupt set and clear register offsets are not set correctly. This patch corrects them! Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Tested-by: Stephan Gerhold <stephan@gerhold.net> Reviewed-by: Stephan Gerhold <stephan@gerhold.net> Link: https://lore.kernel.org/r/20200811103452.20448-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: echoaudio: Fix potential Oops in snd_echo_resume()Dinghao Liu2020-08-211-2/+0
| | | | | | | | | | | | | | [ Upstream commit 5a25de6df789cc805a9b8ba7ab5deef5067af47e ] Freeing chip on error may lead to an Oops at the next time the system goes to resume. Fix this by removing all snd_echo_free() calls on error. Fixes: 47b5d028fdce8 ("ALSA: Echoaudio - Add suspend support #2") Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn> Link: https://lore.kernel.org/r/20200813074632.17022-1-dinghao.liu@zju.edu.cn Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109Hector Martin2020-08-214-0/+11
| | | | | | | | | | | | | | | | | | | | | | commit 1b7ecc241a67ad6b584e071bd791a54e0cd5f097 upstream. Further investigation of the L-R swap problem on the MS2109 reveals that the problem isn't that the channels are swapped, but rather that they are swapped and also out of phase by one sample. In other words, the issue is actually that the very first frame that comes from the hardware is a half-frame containing only the right channel, and after that everything becomes offset. So introduce a new quirk field to drop the very first 2 bytes that come in after the format is configured and a capture stream starts. This puts the channels in phase and in the correct order. Cc: stable@vger.kernel.org Signed-off-by: Hector Martin <marcan@marcan.st> Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: add quirk for Pioneer DDJ-RBHector Martin2020-08-211-0/+56
| | | | | | | | | | | | | | commit 6e8596172ee1cd46ec0bfd5adcf4ff86371478b6 upstream. This is just another Pioneer device with fixed endpoints. Input is dummy but used as feedback (it always returns silence). Cc: stable@vger.kernel.org Signed-off-by: Hector Martin <marcan@marcan.st> Link: https://lore.kernel.org/r/20200810082502.225979-1-marcan@marcan.st Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: fix overeager device match for MacroSilicon MS2109Hector Martin2020-08-211-1/+7
| | | | | | | | | | | | | | | commit 14a720dc1f5332f3bdf30a23a3bc549e81be974c upstream. Matching by device matches all interfaces, which breaks the video/HID portions of the device depending on module load order. Fixes: e337bf19f6af ("ALSA: usb-audio: add quirk for MacroSilicon MS2109") Cc: stable@vger.kernel.org Signed-off-by: Hector Martin <marcan@marcan.st> Link: https://lore.kernel.org/r/20200810045319.128745-1-marcan@marcan.st Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Creative USB X-Fi Pro SB1095 volume knob supportMirko Dietrich2020-08-211-0/+1
| | | | | | | | | | | | | | | | commit fec9008828cde0076aae595ac031bfcf49d335a4 upstream. Adds an entry for Creative USB X-Fi to the rc_config array in mixer_quirks.c to allow use of volume knob on the device. Adds support for newer X-Fi Pro card, known as "Model No. SB1095" with USB ID "041e:3263" Signed-off-by: Mirko Dietrich <buzz@l4m1.de> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: Intel: bxt_rt298: add missing .owner fieldPierre-Louis Bossart2020-08-211-0/+2
| | | | | | | | | | | | | | | | | | | [ Upstream commit 88cee34b776f80d2da04afb990c2a28c36799c43 ] This field is required for ASoC cards. Not setting it will result in a module->name pointer being NULL and generate problems such as cat /proc/asound/modules 0 (efault) Fixes: 76016322ec56 ('ASoC: Intel: Add Broxton-P machine driver') Reported-by: Jaroslav Kysela <perex@perex.cz> Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200625191308.3322-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: seq: oss: Serialize ioctlsTakashi Iwai2020-08-211-1/+7
| | | | | | | | | | | | | | | | | | | | | | | commit 80982c7e834e5d4e325b6ce33757012ecafdf0bb upstream. Some ioctls via OSS sequencer API may race and lead to UAF when the port create and delete are performed concurrently, as spotted by a couple of syzkaller cases. This patch is an attempt to address it by serializing the ioctls with the existing register_mutex. Basically OSS sequencer API is an obsoleted interface and was designed without much consideration of the concurrency. There are very few applications with it, and the concurrent performance isn't asked, hence this "big hammer" approach should be good enough. Reported-by: syzbot+1a54a94bd32716796edd@syzkaller.appspotmail.com Reported-by: syzbot+9d2abfef257f3e2d4713@syzkaller.appspotmail.com Suggested-by: Hillf Danton <hdanton@sina.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200804185815.2453-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: rt5670: Correct RT5670_LDO_SEL_MASKHans de Goede2020-07-291-1/+1
| | | | | | | | | | | | | | | | | | | | | | | commit 5cacc6f5764e94fa753b2c1f5f7f1f3f74286e82 upstream. The RT5670_PWR_ANLG1 register has 3 bits to select the LDO voltage, so the correct mask is 0x7 not 0x3. Because of this wrong mask we were programming the ldo bits to a setting of binary 001 (0x05 & 0x03) instead of binary 101 when moving to SND_SOC_BIAS_PREPARE. According to the datasheet 001 is a reserved value, so no idea what it did, since the driver was working fine before I guess we got lucky and it does something which is ok. Fixes: 5e8351de740d ("ASoC: add RT5670 CODEC driver") Signed-off-by: Hans de Goede <hdegoede@redhat.com> Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20200628155231.71089-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: info: Drop WARN_ON() from buffer NULL sanity checkTakashi Iwai2020-07-291-1/+3
| | | | | | | | | | | | | | | | | | | | | | commit 60379ba08532eca861e933b389526a4dc89e0c42 upstream. snd_info_get_line() has a sanity check of NULL buffer -- both buffer itself being NULL and buffer->buffer being NULL. Basically both checks are valid and necessary, but the problem is that it's with snd_BUG_ON() macro that triggers WARN_ON(). The latter condition (NULL buffer->buffer) can be met arbitrarily by user since the buffer is allocated at the first write, so it means that user can trigger WARN_ON() at will. This patch addresses it by simply moving buffer->buffer NULL check out of snd_BUG_ON() so that spurious WARNING is no longer triggered. Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix race against the error recovery URB submissionTakashi Iwai2020-07-221-5/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 9b7e5208a941e2e491a83eb5fa83d889e888fa2f upstream. USB MIDI driver has an error recovery mechanism to resubmit the URB in the delayed timer handler, and this may race with the standard start / stop operations. Although both start and stop operations themselves don't race with each other due to the umidi->mutex protection, but this isn't applied to the timer handler. For fixing this potential race, the following changes are applied: - Since the timer handler can't use the mutex, we apply the umidi->disc_lock protection at each input stream URB submission; this also needs to change the GFP flag to GFP_ATOMIC - Add a check of the URB refcount and skip if already submitted - Move the timer cancel call at disconnection to the beginning of the procedure; this assures the in-flight timer handler is gone properly before killing all pending URBs Reported-by: syzbot+0f4ecfe6a2c322c81728@syzkaller.appspotmail.com Reported-by: syzbot+5f1d24c49c1d2c427497@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200710160656.16819-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: line6: Perform sanity check for each URB creationTakashi Iwai2020-07-222-0/+4
| | | | | | | | | | | | | | | | | | commit 6e8a914ad619042c5f25a4feb663357c4170fd8d upstream. 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.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: add quirk for MacroSilicon MS2109Hector Martin2020-07-221-0/+52
| | | | | | | | | | | | | | commit e337bf19f6af38d5c3fa6d06cd594e0f890ca1ac upstream. 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.st Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - let hs_mic be picked ahead of hp_micHui Wang2020-07-221-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 6a6ca7881b1ab1c13fe0d70bae29211a65dd90de upstream. 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.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: opl3: fix infoleak in opl3xidongwang2020-07-221-0/+2
| | | | | | | | | | | | | | | commit ad155712bb1ea2151944cf06a0e08c315c70c1e3 upstream. 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.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: compress: fix partial_drain completion stateVinod Koul2020-07-221-0/+4
| | | | | | | | | | | | | | | | | | | | | [ Upstream commit f79a732a8325dfbd570d87f1435019d7e5501c6d ] 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: f44f2a5417b2 ("ALSA: compress: fix drain calls blocking other compress functions (v6)") Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com> Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20200629134737.105993-4-vkoul@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* Revert "ALSA: usb-audio: Improve frames size computation"Greg Kroah-Hartman2020-07-094-45/+5
| | | | | | | | | | | | | | This reverts commit 99703c921864a318e3e8aae74fde071b1ff35bea which is commit f0bd62b64016508938df9babe47f65c2c727d25c upstream. It causes a number of reported issues and a fix for it has not hit Linus's tree yet. Revert this to resolve those problems. Cc: Alexander Tsoy <alexander@tsoy.me> Cc: Takashi Iwai <tiwai@suse.de> Cc: Sasha Levin <sashal@kernel.org> Cc: Hans de Goede <jwrdegoede@fedoraproject.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda: Add NVIDIA codec IDs 9a & 9d through a0 to patch tableAaron Plattner2020-06-301-0/+5
| | | | | | | | | | | | | | commit adb36a8203831e40494a92095dacd566b2ad4a69 upstream. These IDs are for upcoming NVIDIA chips with audio functions that are largely similar to the existing ones. Signed-off-by: Aaron Plattner <aplattner@nvidia.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200611180845.39942-1-aplattner@nvidia.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: rockchip: Fix a reference count leak.Qiushi Wu2020-06-301-1/+3
| | | | | | | | | | | | | | | [ Upstream commit f141a422159a199f4c8dedb7e0df55b3b2cf16cd ] 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: fc05a5b22253 ("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.edu Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: Fix OOB access of mixer element listTakashi Iwai2020-06-303-7/+20
| | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 220345e98f1cdc768eeb6e3364a0fa7ab9647fe7 ] 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.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: Clean up mixer element list traverseTakashi Iwai2020-06-304-16/+18
| | | | | | | | | | | | | | [ Upstream commit 8c558076c740e8009a96c6fdc3d4245dde62be77 ] Introduce a new macro for iterating over mixer element list for avoiding the open codes in many places. Also the open-coded container_of() and the forced cast to struct usb_mixer_elem_info are replaced with another simple macro, too. No functional changes but just readability improvement. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: uac1: Invalidate ctl on interruptJulian Scheel2020-06-301-1/+6
| | | | | | | | | | | | | [ Upstream commit b2500b584cfd228d67e1e43daf27c8af865b499e ] When an interrupt occurs, the value of at least one of the belonging controls should have changed. To make sure they get re-read from device on the next read, invalidate the cache. This was correctly implemented for uac2 already, but missing for uac1. Signed-off-by: Julian Scheel <julian@jusst.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: add quirk for Denon DCD-1500REYick W. Tse2020-06-301-0/+1
| | | | | | | | | | | | | | | | | | | | | commit c9808bbfed3cfc911ecb60fe8e80c0c27876c657 upstream. fix error "clock source 41 is not valid, cannot use" [] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00 [] New USB device strings: Mfr=1, Product=2, SerialNumber=0 [] Product: DCD-1500RE [] Manufacturer: D & M Holdings Inc. [] [] clock source 41 is not valid, cannot use [] usbcore: registered new interface driver snd-usb-audio Signed-off-by: Yick W. Tse <y_w_tse@yahoo.com.hk> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1373857985.210365.1592048406997@mail.yahoo.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failedXiyu Yang2020-06-251-0/+1
| | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf@gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fudan.edu.cn Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: Improve frames size computationAlexander Tsoy2020-06-254-5/+45
| | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: isa/wavefront: prevent out of bounds write in ioctlDan Carpenter2020-06-251-1/+7
| | | | | | | | | | | | [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma typeXiyu Yang2020-06-251-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf@gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fudan.edu.cn Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* vga_switcheroo: Use device link for HDA controllerLukas Wunner2020-06-202-14/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 07f4f97d7b4bf325d9f558c5b58230387e4e57e0 ] Back in 2013, runtime PM for GPUs with integrated HDA controller was introduced with commits 0d69704ae348 ("gpu/vga_switcheroo: add driver control power feature. (v3)") and 246efa4a072f ("snd/hda: add runtime suspend/resume on optimus support (v4)"). Briefly, the idea was that the HDA controller is forced on and off in unison with the GPU. The original code is mostly still in place even though it was never a 100% perfect solution: E.g. on access to the HDA controller, the GPU is powered up via vga_switcheroo_runtime_resume_hdmi_audio() but there are no provisions to keep it resumed until access to the HDA controller has ceased: The GPU autosuspends after 5 seconds, rendering the HDA controller inaccessible. Additionally, a kludge is required when hda_intel.c probes: It has to check whether the GPU is powered down (check_hdmi_disabled()) and defer probing if so. However in the meantime (in v4.10) the driver core has gained a feature called device links which promises to solve such issues in a clean way: It allows us to declare a dependency from the HDA controller (consumer) to the GPU (supplier). The PM core then automagically ensures that the GPU is runtime resumed as long as the HDA controller's ->probe hook is executed and whenever the HDA controller is accessed. By default, the HDA controller has a dependency on its parent, a PCIe Root Port. Adding a device link creates another dependency on its sibling: PCIe Root Port ^ ^ | | | | HDA ===> GPU The device link is not only used for runtime PM, it also guarantees that on system sleep, the HDA controller suspends before the GPU and resumes after the GPU, and on system shutdown the HDA controller's ->shutdown hook is executed before the one of the GPU. It is a complete solution. Using this functionality is as simple as calling device_link_add(), which results in a dmesg entry like this: pci 0000:01:00.1: Linked as a consumer to 0000:01:00.0 The code for the GPU-governed audio power management can thus be removed (except where it's still needed for legacy manual power control). The device link is added in a PCI quirk rather than in hda_intel.c. It is therefore legal for the GPU to runtime suspend to D3cold even if the HDA controller is not bound to a driver or if CONFIG_SND_HDA_INTEL is not enabled, for accesses to the HDA controller will cause the GPU to wake up regardless if they're occurring outside of hda_intel.c (think config space readout via sysfs). Contrary to the previous implementation, the HDA controller's power state is now self-governed, rather than GPU-governed, whereas the GPU's power state is no longer fully self-governed. (The HDA controller needs to runtime suspend before the GPU can.) It is thus crucial that runtime PM is always activated on the HDA controller even if CONFIG_SND_HDA_POWER_SAVE_DEFAULT is set to 0 (which is the default), lest the GPU stays awake. This is achieved by setting the auto_runtime_pm flag on every codec and the AZX_DCAPS_PM_RUNTIME flag on the HDA controller. A side effect is that power consumption might be reduced if the GPU is in use but the HDA controller is not, because the HDA controller is now allowed to go to D3hot. Before, it was forced to stay in D0 as long as the GPU was in use. (There is no reduction in power consumption on my Nvidia GK107, but there might be on other chips.) The code paths for legacy manual power control are adjusted such that runtime PM is disabled during power off, thereby preventing the PM core from resuming the HDA controller. Note that the device link is not only added on vga_switcheroo capable systems, but for *any* GPU with integrated HDA controller. The idea is that the HDA controller streams audio via connectors located on the GPU, so the GPU needs to be on for the HDA controller to do anything useful. This commit implicitly fixes an unbalanced runtime PM ref upon unbind of hda_intel.c: On ->probe, a runtime PM ref was previously released under the condition "azx_has_pm_runtime(chip) || hda->use_vga_switcheroo", but on ->remove a runtime PM ref was only acquired under the first of those conditions. Thus, binding and unbinding the driver twice on a vga_switcheroo capable system caused the runtime PM refcount to drop below zero. The issue is resolved because the AZX_DCAPS_PM_RUNTIME flag is now always set if use_vga_switcheroo is true. For more information on device links please refer to: https://www.kernel.org/doc/html/latest/driver-api/device_link.html Documentation/driver-api/device_link.rst Cc: Dave Airlie <airlied@redhat.com> Cc: Ben Skeggs <bskeggs@redhat.com> Cc: Alex Deucher <alexander.deucher@amd.com> Cc: Rafael J. Wysocki <rafael.j.wysocki@intel.com> Acked-by: Bjorn Helgaas <bhelgaas@google.com> Reviewed-by: Takashi Iwai <tiwai@suse.de> Reviewed-by: Peter Wu <peter@lekensteyn.nl> Tested-by: Kai Heng Feng <kai.heng.feng@canonical.com> # AMD PowerXpress Tested-by: Mike Lothian <mike@fireburn.co.uk> # AMD PowerXpress Tested-by: Denis Lisov <dennis.lissov@gmail.com> # Nvidia Optimus Tested-by: Peter Wu <peter@lekensteyn.nl> # Nvidia Optimus Tested-by: Lukas Wunner <lukas@wunner.de> # MacBook Pro Signed-off-by: Lukas Wunner <lukas@wunner.de> Link: https://patchwork.freedesktop.org/patch/msgid/51bd38360ff502a8c42b1ebf4405ee1d3f27118d.1520068884.git.lukas@wunner.de Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: pcm: disallow linking stream to itselfMichał Mirosław2020-06-201-0/+5
| | | | | | | | | | | | | | | | | commit 951e2736f4b11b58dc44d41964fa17c3527d882a upstream. Prevent SNDRV_PCM_IOCTL_LINK linking stream to itself - the code can't handle it. Fixed commit is not where bug was introduced, but changes the context significantly. Cc: stable@vger.kernel.org Fixes: 0888c321de70 ("pcm_native: switch to fdget()/fdput()") Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl> Link: https://lore.kernel.org/r/89c4a2487609a0ed6af3ecf01cc972bdc59a7a2d.1591634956.git.mirq-linux@rere.qmqm.pl Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix inconsistent card PM state after resumeTakashi Iwai2020-06-202-8/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 862b2509d157c629dd26d7ac6c6cdbf043d332eb upstream. When a USB-audio interface gets runtime-suspended via auto-pm feature, the driver suspends all functionality and increment chip->num_suspended_intf. Later on, when the system gets suspended to S3, the driver increments chip->num_suspended_intf again, skips the device changes, and sets the card power state to SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from S3, the resume callback decrements chip->num_suspended_intf. Since this refcount is still not zero (it's been runtime-suspended), the whole resume is skipped. But there is a small pitfall here. The problem is that the driver doesn't restore the card power state after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So, even after the system resume finishes, the card instance still appears as if it were system-suspended, and this confuses many ioctl accesses that are blocked unexpectedly. In details, we have two issues behind the scene: one is that the card power state is changed only when the refcount becomes zero, and another is that the prior auto-suspend check is kept in a boolean flag. Although the latter problem is almost negligible since the auto-pm feature is imposed only on the primary interface, but this can be a potential problem on the devices with multiple interfaces. This patch addresses those issues by the following: - Replace chip->autosuspended boolean flag with chip->system_suspend counter - At the first system-suspend, chip->num_suspended_intf is recorded to chip->system_suspend - At system-resume, the card power state is restored when the chip->num_suspended_intf refcount reaches to chip->system_suspend, i.e. the state returns to the auto-suspended Also, the patch fixes yet another hidden problem by the code refactoring along with the fixes above: namely, when some resume procedure failed, the driver left chip->num_suspended_intf that was already decreased, and it might lead to the refcount unbalance. In the new code, the refcount decrement is done after the whole resume procedure, and the problem is avoided as well. Fixes: 0662292aec05 ("ALSA: usb-audio: Handle normal and auto-suspend equally") Reported-and-tested-by: Macpaul Lin <macpaul.lin@mediatek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200603153709.6293-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - add a pintbl quirk for several Lenovo machinesHui Wang2020-06-201-0/+6
| | | | | | | | | | | | | | | | commit 573fcbfd319ccef26caa3700320242accea7fd5c upstream. A couple of Lenovo ThinkCentre machines all have 2 front mics and they use the same codec alc623 and have the same pin config, so add a pintbl entry for those machines to apply the fixup ALC283_FIXUP_HEADSET_MIC. Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200608115541.9531-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: es1688: Add the missed snd_card_free()Chuhong Yuan2020-06-201-1/+3
| | | | | | | | | | | | | | | commit d9b8fbf15d05350b36081eddafcf7b15aa1add50 upstream. snd_es968_pnp_detect() misses a snd_card_free() in a failed path. Add the missed function call to fix it. Fixes: a20971b201ac ("ALSA: Merge es1688 and es968 drivers") Signed-off-by: Chuhong Yuan <hslester96@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200603092459.1424093-1-hslester96@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Add new codec supported for ALC287Kailang Yang2020-06-031-0/+3
| | | | | | | | | | | | [ Upstream commit 630e36126e420e1756378b3427b42711ce0b9ddd ] Enable new codec supported for ALC287. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/dcf5ce5507104d0589a917cbb71dc3c6@realtek.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: mixer: volume quirk for ESS Technology Asus USB DACChris Chiu2020-06-031-0/+8
| | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 4020d1ccbe55bdf67b31d718d2400506eaf4b43f ] The Asus USB DAC is a USB type-C audio dongle for connecting to the headset and headphone. The volume minimum value -23040 which is 0xa600 in hexadecimal with the resolution value 1 indicates this should be endianness issue caused by the firmware bug. Add a volume quirk to fix the volume control problem. Also fixes this warning: Warning! Unlikely big volume range (=23040), cval->res is probably wrong. [5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1 Warning! Unlikely big volume range (=23040), cval->res is probably wrong. [7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1 Signed-off-by: Chris Chiu <chiu@endlessm.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200526062613.55401-1-chiu@endlessm.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hwdep: fix a left shifting 1 by 31 UB bugChangming Liu2020-06-031-2/+2
| | | | | | | | | | | | | | | [ Upstream commit fb8cd6481ffd126f35e9e146a0dcf0c4e8899f2e ] The "info.index" variable can be 31 in "1 << info.index". This might trigger an undefined behavior since 1 is signed. Fix this by casting 1 to 1u just to be sure "1u << 31" is defined. Signed-off-by: Changming Liu <liu.changm@northeastern.edu> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/BL0PR06MB4548170B842CB055C9AF695DE5B00@BL0PR06MB4548.namprd06.prod.outlook.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: pcm: fix incorrect hw_base increaseBrent Lu2020-05-271-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit e7513c5786f8b33f0c107b3759e433bc6cbb2efa upstream. There is a corner case that ALSA keeps increasing the hw_ptr but DMA already stop working/updating the position for a long time. In following log we can see the position returned from DMA driver does not move at all but the hw_ptr got increased at some point of time so snd_pcm_avail() will return a large number which seems to be a buffer underrun event from user space program point of view. The program thinks there is space in the buffer and fill more data. [ 418.510086] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368 [ 418.510149] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6910 avail 9554 ... [ 418.681052] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15102 avail 1362 [ 418.681130] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0 [ 418.726515] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 16464 avail 16368 This is because the hw_base will be increased by runtime->buffer_size frames unconditionally if the hw_ptr is not updated for over half of buffer time. As the hw_base increases, so does the hw_ptr increased by the same number. The avail value returned from snd_pcm_avail() could exceed the limit (buffer_size) easily becase the hw_ptr itself got increased by same buffer_size samples when the corner case happens. In following log, the buffer_size is 16368 samples but the avail is 21810 samples so CRAS server complains about it. [ 418.851755] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 27390 avail 5442 [ 418.926491] sound pcmC0D5p: pos 96 hw_ptr 32832 appl_ptr 27390 avail 21810 cras_server[1907]: pcm_avail returned frames larger than buf_size: sof-glkda7219max: :0,5: 21810 > 16368 By updating runtime->hw_ptr_jiffies each time the HWSYNC is called, the hw_base will keep the same when buffer stall happens at long as the interval between each HWSYNC call is shorter than half of buffer time. Following is a log captured by a patched kernel. The hw_base/hw_ptr value is fixed in this corner case and user space program should be aware of the buffer stall and handle it. [ 293.525543] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368 [ 293.525606] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6880 avail 9584 [ 293.525975] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 10976 avail 5488 [ 293.611178] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15072 avail 1392 [ 293.696429] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0 ... [ 381.139517] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0 Signed-off-by: Brent Lu <brent.lu@intel.com> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1589776238-23877-1-git-send-email-brent.lu@intel.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio optionScott Bahling2020-05-271-1/+2
| | | | | | | | | | | | | | | | | | commit b0cb099062b0c18246c3a20caaab4c0afc303255 upstream. The ST Audio ADCIII is an STDSP24 card plus extension box. With commit e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") we enabled the ADCIII ports using the model=staudio option but forgot this part to ensure the STDSP24 card is initialized properly. Fixes: e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") Signed-off-by: Scott Bahling <sbahling@suse.com> Cc: <stable@vger.kernel.org> BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1048934 Link: https://lore.kernel.org/r/20200518175728.28766-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* Revert "ALSA: hda/realtek: Fix pop noise on ALC225"Kai-Heng Feng2020-05-201-2/+0
| | | | | | | | | | | | | | | | | commit f41224efcf8aafe80ea47ac870c5e32f3209ffc8 upstream. This reverts commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757. Enable power save node breaks some systems with ACL225. Revert the patch and use a platform specific quirk for the original issue isntead. Fixes: 3b36b13d5e69 ("ALSA: hda/realtek: Fix pop noise on ALC225") BugLink: https://bugs.launchpad.net/bugs/1875916 Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Add control message quirk delay for Kingston HyperX headsetJesus Ramos2020-05-201-4/+5
| | | | | | | | | | | | | | commit 073919e09ca445d4486968e3f851372ff44cf2b5 upstream. Kingston HyperX headset with 0951:16ad also needs the same quirk for delaying the frequency controls. Signed-off-by: Jesus Ramos <jesus-ramos@live.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/BY5PR19MB3634BA68C7CCA23D8DF428E796AF0@BY5PR19MB3634.namprd19.prod.outlook.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: rawmidi: Fix racy buffer resize under concurrent accessesTakashi Iwai2020-05-201-4/+27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit c1f6e3c818dd734c30f6a7eeebf232ba2cf3181d upstream. The rawmidi core allows user to resize the runtime buffer via ioctl, and this may lead to UAF when performed during concurrent reads or writes: the read/write functions unlock the runtime lock temporarily during copying form/to user-space, and that's the race window. This patch fixes the hole by introducing a reference counter for the runtime buffer read/write access and returns -EBUSY error when the resize is performed concurrently against read/write. Note that the ref count field is a simple integer instead of refcount_t here, since the all contexts accessing the buffer is basically protected with a spinlock, hence we need no expensive atomic ops. Also, note that this busy check is needed only against read / write functions, and not in receive/transmit callbacks; the race can happen only at the spinlock hole mentioned in the above, while the whole function is protected for receive / transmit callbacks. Reported-by: butt3rflyh4ck <butterflyhuangxx@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/CAFcO6XMWpUVK_yzzCpp8_XP7+=oUpQvuBeCbMffEDkpe8jWrfg@mail.gmail.com Link: https://lore.kernel.org/r/s5heerw3r5z.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: rawmidi: Initialize allocated buffersTakashi Iwai2020-05-201-2/+2
| | | | | | | | | | | | | | | commit 5a7b44a8df822e0667fc76ed7130252523993bda upstream. syzbot reported the uninitialized value exposure in certain situations using virmidi loop. It's likely a very small race at writing and reading, and the influence is almost negligible. But it's safer to paper over this just by replacing the existing kvmalloc() with kvzalloc(). Reported-by: syzbot+194dffdb8b22fc5d207a@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Limit int mic boost for Thinkpad T530Takashi Iwai2020-05-201-1/+9
| | | | | | | | | | | | | | | | | | commit b590b38ca305d6d7902ec7c4f7e273e0069f3bcc upstream. Lenovo Thinkpad T530 seems to have a sensitive internal mic capture that needs to limit the mic boost like a few other Thinkpad models. Although we may change the quirk for ALC269_FIXUP_LENOVO_DOCK, this hits way too many other laptop models, so let's add a new fixup model that limits the internal mic boost on top of the existing quirk and apply to only T530. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1171293 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200514160533.10337-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Fix S3 pop noise on Dell WyseKai-Heng Feng2020-05-201-0/+16
| | | | | | | | | | | | | | | | | | | | [ Upstream commit 52e4e36807aeac1cdd07b14e509c8a64101e1a09 ] Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. The previous fix enable power save node universally for ALC225, however it makes some ALC225 systems unable to produce any sound. So let's only enable power save node for the affected Dell Wyse platform. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-2-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda/hdmi: fix race in monitor detection during probeKai Vehmanen2020-05-201-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit ca76282b6faffc83601c25bd2a95f635c03503ef ] A race exists between build_pcms() and build_controls() phases of codec setup. Build_pcms() sets up notifier for jack events. If a monitor event is received before build_controls() is run, the initial jack state is lost and never reported via mixer controls. The problem can be hit at least with SOF as the controller driver. SOF calls snd_hda_codec_build_controls() in its workqueue-based probe and this can be delayed enough to hit the race condition. Fix the issue by invalidating the per-pin ELD information when build_controls() is called. The existing call to hdmi_present_sense() will update the ELD contents. This ensures initial monitor state is correctly reflected via mixer controls. BugLink: https://github.com/thesofproject/linux/issues/1687 Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda: Match both PCI ID and SSID for driver blacklistTakashi Iwai2020-05-101-4/+5
| | | | | | | | | | | | | | | | | | | | | commit 977dfef40c8996b69afe23a9094d184049efb7bb upstream. The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. Since the empty codec problem appear on the certain AMD platform (PCI ID 1022:1487), this patch changes the blacklist matching to both PCI ID and SSID using pci_match_id(). Also, the entry that was removed by the previous fix for ASUS ROG Zenigh II is re-added. Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entryAmadeusz Sławiński2020-05-101-3/+3
| | | | | | | | | | | | | [ Upstream commit 326b509238171d37402dbe308e154cc234ed1960 ] If we don't find any pcm, pcm will point at address at an offset from the the list head and not a meaningful structure. Fix this by returning correct pcm if found and NULL if not. Found with coccinelle. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: rsnd: Fix HDMI channel mapping for multi-SSI modeMatthias Blankertz2020-05-101-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit b94e164759b82d0c1c80d4b1c8f12c9bee83f11d ] The HDMI?_SEL register maps up to four stereo SSI data lanes onto the sdata[0..3] inputs of the HDMI output block. The upper half of the register contains four blocks of 4 bits, with the most significant controlling the sdata3 line and the least significant the sdata0 line. The shift calculation has an off-by-one error, causing the parent SSI to be mapped to sdata3, the first multi-SSI child to sdata0 and so forth. As the parent SSI transmits the stereo L/R channels, and the HDMI core expects it on the sdata0 line, this causes no audio to be output when playing stereo audio on a multichannel capable HDMI out, and multichannel audio has permutated channels. Fix the shift calculation to map the parent SSI to sdata0, the first child to sdata1 etc. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
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