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* 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>
* 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 Goede2020-07-311-1/+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 Iwai2020-07-311-1/+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 02c56650f3c118d3752122996d96173d26bb13aa 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: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()Takashi Iwai2020-06-291-4/+4
| | | | | | | | | | | | | | | | | | | | | commit 6de3c9e3f6b3eaf66859e1379b3f35dda781416b upstream. The quirk function snd_emuusb_set_samplerate() has a NULL check for the mixer element, but this is useless in the current code. It used to be a check against mixer->id_elems[unitid] but it was changed later to the value after mixer_eleme_list_to_info() which is always non-NULL due to the container_of() usage. This patch fixes the check before the conversion. While we're at it, correct a typo in the comment in the function, too. Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix OOB access of mixer element listTakashi Iwai2020-06-293-7/+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 Iwai2020-06-294-16/+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: allow clock source validity interruptsDaniel Mack2020-06-291-1/+8
| | | | | | | | | | | | | | | | | [ Upstream commit 191227d99a281333b50aaf82ab4152fbfa249c19 ] miniDSP USBStreamer UAC2 devices send clock validity changes with the control field set to zero. The current interrupt handler ignores all packets if the control field does not match the mixer element's, but it really should only do that in case that field is needed to distinguish multiple elements with the same ID. This patch implements a logic that lets notifications packets pass if the element ID is unique for a given device. Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: uac1: Invalidate ctl on interruptJulian Scheel2020-06-291-1/+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. Tse2020-06-291-0/+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 Yang2020-06-291-0/+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 Tsoy2020-06-294-5/+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 Carpenter2020-06-291-1/+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>
* 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: 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 - No loopback on ALC299 codecTakashi Iwai2020-06-111-0/+3
| | | | | | | | | | | | | | | | commit fa16b69f1299004b60b625f181143500a246e5cb upstream. ALC299 has no loopback mixer, but the driver still tries to add a beep control over the mixer NID which leads to the error at accessing it. This patch fixes it by properly declaring mixer_nid=0 for this codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195775 Fixes: 28f1f9b26cee ("ALSA: hda/realtek - Add new codec ID ALC299") Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de> Cc: Guenter Roeck <linux@roeck-us.net> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.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>
* 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: 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: 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: 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: 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>
* ALSA: fm801: Initialize chip after IRQ handler is registeredAndy Shevchenko2020-05-101-2/+2
| | | | | | | | | | | | | | | | | | | | | commit 610e1ae9b533be82b3aa118b907e0a703256913d upstream. The commit b56fa687e02b ("ALSA: fm801: detect FM-only card earlier") rearranged initialization calls, i.e. it makes snd_fm801_chip_init() to be called before we register interrupt handler and set PCI bus mastering. Somehow it prevents FM801-AU to work properly. Thus, partially revert initialization order changed by commit mentioned above. Fixes: b56fa687e02b ("ALSA: fm801: detect FM-only card earlier") Reported-by: Émeric MASCHINO <emeric.maschino@gmail.com> Tested-by: Émeric MASCHINO <emeric.maschino@gmail.com> Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Cc: <stable@vger.kernel.org> # v4.5+ Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: fsl_ssi: mark SACNT register volatileMaciej S. Szmigiero2020-05-101-1/+7
| | | | | | | | | | | | | | | | | commit 3f1c241f0f5f90046258e6b8d4aeb6463ffdc08e upstream. SACNT register should be marked volatile since its WR and RD bits are cleared by SSI after completing the relevant operation. This unbreaks AC'97 register access. Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast") Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name> Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: tegra_alc5632: check return valueSudip Mukherjee2020-05-101-4/+8
| | | | | | | | | | | | | commit 319c32597fc22a58b946a6146f2be1fd208582e0 upstream. We have been returning success even if snd_soc_card_jack_new() fails. Lets check the return value and return error if it fails. Fixes: 12cc6d1dca4d ("ASoC: tegra_alc5632: Register jacks at the card level") Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: Intel: pass correct parameter in sst_alloc_stream_mrfld()Dan Carpenter2020-05-101-1/+1
| | | | | | | | | | | | | | | commit d16a2b9f2465b5486f830178fbfb7d203e0a17ae upstream. "data" is always NULL in this function. I think we should be passing "&data" to sst_prepare_and_post_msg() instead of "data". Fixes: 3d9ff34622ba ('ASoC: Intel: sst: add stream operations') Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Tested-by: Dinesh Mirche <dinesh.mirche@intel.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: fm801: detect FM-only card earlierAndy Shevchenko2020-05-101-30/+39
| | | | | | | | | | | | | | | | | | | | commit b56fa687e02b27f8bd9d282950a88c2ed23d766b upstream. If user does not supply tea575x_tuner parameter the driver tries to detect the tuner type. The failed codec initialization is considered as FM-only card present, however the driver still registers an IRQ handler for it. Move codec detection earlier to set tea575x_tuner parameter before check. Here the following functions are introduced reset_coded() resets AC97 codec snd_fm801_chip_multichannel_init() initializes cards with multichannel support Fixes: 5618955c4269 (ALSA: fm801: move to pcim_* and devm_* functions) Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: fm801: propagate TUNER_ONLY bit when autodetectedAndy Shevchenko2020-05-101-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit dbec6719ac036f68568d8488805d41346c021eff upstream. The commit d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection) brings autodetection to the driver. However the autodetection algorithm misses the TUNER_ONLY bit if it is supplied by the user. Thus, user gets weird messages and no card registered. snd_fm801 0000:0d:01.0: detected TEA575x radio type SF64-PCR snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) ... snd_fm801 0000:0d:01.0: AC'97 0 does not respond - RESET snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 0 access is not valid [0x0], removing mixer. snd_fm801: probe of 0000:0d:01.0 failed with error -5 Do a copy of TUNER_ONLY bit to be applied after autodetection is done. Fixes: d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection) Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Cc: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: fm801: explicitly free IRQ lineAndy Shevchenko2020-05-101-0/+2
| | | | | | | | | | | | | | | commit e97e98c63b43040732ad5d1f0b38ad4a8371c73a upstream. Otherwise we will have a warning on ->remove() since device is a PCI one. WARNING: CPU: 4 PID: 1411 at /home/andy/prj/linux/fs/proc/generic.c:575 remove_proc_entry+0x137/0x160() remove_proc_entry: removing non-empty directory 'irq/21', leaking at least 'snd_fm801' Fixes: 5618955c4269 (ALSA: fm801: move to pcim_* and devm_* functions) Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: imx-spdif: Fix crash on suspendLars-Peter Clausen2020-05-051-2/+0
| | | | | | | | | | | | | | | | | | | | commit 9954859185c6e8359e71121037b627f1e294057d upstream. When registering a ASoC card the driver data of the parent device is set to point to the card. This driver data is used in the snd_soc_suspend()/resume() callbacks. The imx-spdif driver overwrites the driver data with custom data which causes snd_soc_suspend() to crash. Since the custom driver is not used anywhere simply deleting the line which sets the custom driver data fixes the issue. Fixes: 43ac946922b3 ("ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver") Tested-by: Fabio Estevam <fabio.estevam@nxp.com> Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: wm8960: Fix WM8960_SYSCLK_PLL modeStuart Henderson2020-05-051-15/+17
| | | | | | | | | | | | | | commit 6bb7451429084cefcb3a18fff809f7992595d2af upstream. With the introduction of WM8960_SYSCLK_AUTO mode, WM8960_SYSCLK_PLL mode was made unusable. Ensure we're not PLL mode before trying to use MCLK. Fixes: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function") Signed-off-by: Stuart Henderson <stuart.henderson@cirrus.com> Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: opti9xx: shut up gcc-10 range warningArnd Bergmann2020-05-052-6/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 5ce00760a84848d008554c693ceb6286f4d9c509 upstream. gcc-10 points out a few instances of suspicious integer arithmetic leading to value truncation: sound/isa/opti9xx/opti92x-ad1848.c: In function 'snd_opti9xx_configure': sound/isa/opti9xx/opti92x-ad1848.c:322:43: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_opti9xx_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 322 | (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/opti92x-ad1848.c:351:3: note: in expansion of macro 'snd_opti9xx_write_mask' 351 | snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c: In function 'snd_miro_configure': sound/isa/opti9xx/miro.c:873:40: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_miro_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow] 873 | (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~ sound/isa/opti9xx/miro.c:1010:3: note: in expansion of macro 'snd_miro_write_mask' 1010 | snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); | ^~~~~~~~~~~~~~~~~~~ These are all harmless here as only the low 8 bit are passed down anyway. Change the macros to inline functions to make the code more readable and also avoid the warning. Strictly speaking those functions also need locking to make the read/write pair atomic, but it seems unlikely that anyone would still run into that issue. Fixes: 1841f613fd2e ("[ALSA] Add snd-miro driver") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: pcm: oss: Place the plugin buffer overflow checks correctlyTakashi Iwai2020-05-051-8/+12
| | | | | | | | | | | | | | | | | | | | | commit 4285de0725b1bf73608abbcd35ad7fd3ddc0b61e upstream. The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Fixes: f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200424193350.19678-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: dapm: fixup dapm kcontrol widgetGyeongtaek Lee2020-05-021-3/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | commit ebf1474745b4373fdde0fcf32d9d1f369b50b212 upstream. snd_soc_dapm_kcontrol widget which is created by autodisable control should contain correct on_val, mask and shift because it is set when the widget is powered and changed value is applied on registers by following code in dapm_seq_run_coalesced(). mask |= w->mask << w->shift; if (w->power) value |= w->on_val << w->shift; else value |= w->off_val << w->shift; Shift on the mask in dapm_kcontrol_data_alloc() is removed to prevent double shift. And, on_val in dapm_kcontrol_set_value() is modified to get correct value in the dapm_seq_run_coalesced(). Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com> Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/000001d61537$b212f620$1638e260$@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devicesAlexander Tsoy2020-05-021-0/+52
| | | | | | | | | | | | | | | | | | | | | | commit 1c826792586f526a5a5cd21d55aad388f5bb0b23 upstream. Many Focusrite devices supports a limited set of sample rates per altsetting. These includes audio interfaces with ADAT ports: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - Scarlett 18i8 3rd gen, 18i20 3rd gen; - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Maximum rate is exposed in the last 4 bytes of Format Type descriptor which has a non-standard bLength = 10. Tested-by: Alexey Skobkin <skobkin-ru@ya.ru> Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix usb audio refcnt leak when getting spdifXiyu Yang2020-05-021-4/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 59e1947ca09ebd1cae147c08c7c41f3141233c84 upstream. snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which increases the refcount of the snd_usb_audio object "chip". When snd_microii_spdif_default_get() returns, local variable "chip" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in several exception handling paths of snd_microii_spdif_default_get(). When those error scenarios occur such as usb_ifnum_to_if() returns NULL, the function forgets to decrease the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak. Fix this issue by jumping to "end" label when those error scenarios occur. Fixes: 447d6275f0c2 ("ALSA: usb-audio: Add sanity checks for endpoint accesses") Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usx2y: Fix potential NULL dereferenceTakashi Iwai2020-05-021-0/+2
| | | | | | | | | | | | | | | commit 7686e3485253635c529cdd5f416fc640abaf076f upstream. The error handling code in usX2Y_rate_set() may hit a potential NULL dereference when an error occurs before allocating all us->urb[]. Add a proper NULL check for fixing the corner case. Reported-by: Lin Yi <teroincn@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda: Remove ASUS ROG Zenith from the blacklistTakashi Iwai2020-05-021-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit a8cf44f085ac12c0b5b8750ebb3b436c7f455419 ] 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. This patch reverts the corresponding entry as a temporary solution. Although Zenith II and co will see get the empty HD-audio bus again, it'd be merely resource wastes and won't affect the functionality, so it's no end of the world. We'll need to address this later, e.g. by either switching to DMI string matching or using PCI ID & SSID pairs. Fixes: 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") Reported-by: Johnathan Smithinovic <johnathan.smithinovic@gmx.at> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
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