- 12 May, 2020 2 commits
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Jian-Hong Pan authored
The ASUS laptop UX550GE with ALC295 can't detect the headset microphone until ALC295_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-2-jian-hong@endlessm.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Chris Chiu authored
The ASUS laptop GL503VM with ALC295 can't detect the headset microphone. The headset microphone does not work until pin 0x19 is enabled for it. Signed-off-by: Chris Chiu <chiu@endlessm.com> Signed-off-by: Daniel Drake <drake@endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-1-jian-hong@endlessm.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 10 May, 2020 1 commit
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Mike Pozulp authored
Some models of the Samsung Notebook 9 have very quiet and distorted headphone output. This quirk changes the VREF value of the ALC298 codec NID 0x1a from default HIZ to new 100. [ adjusted to 5.7-base and rearranged in SSID order -- tiwai ] Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Link: https://lore.kernel.org/r/20200510032838.1989130-1-pozulp.kernel@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 07 May, 2020 1 commit
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Takashi Iwai authored
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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 04 May, 2020 1 commit
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Andrew Oakley authored
This is another TRX40 based motherboard with ALC1220-VB USB-audio that requires a static mapping table. This motherboard also has a PCI device which advertises no codecs. The PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD vendor ID, don't blacklist for now in case other boards have a working audio device with the same ssid. alsa-info.sh report for this board: http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70cSigned-off-by: Andrew Oakley <andrew@adoakley.name> Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.nameSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 03 May, 2020 3 commits
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Kai-Heng Feng authored
Commit 317d9313 ("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: 317d9313 ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-2-kai.heng.feng@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Kai-Heng Feng authored
This reverts commit 3b36b13d. Enable power save node breaks some systems with ACL225. Revert the patch and use a platform specific quirk for the original issue isntead. Fixes: 3b36b13d ("ALSA: hda/realtek: Fix pop noise on ALC225") BugLink: https://bugs.launchpad.net/bugs/1875916Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-1-kai.heng.feng@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Sakamoto authored
The snd-firewire-lib.ko has 'amdtp-packet' event of tracepoints. Current printk format for the event includes 'sizeof(u8)' macro expected to be extended in compilation time. However, this is not done. As a result, perf tools cannot parse the event for printing: $ mount -l -t debugfs debugfs on /sys/kernel/debug type debugfs (rw,nosuid,nodev,noexec,relatime) $ cat /sys/kernel/debug/tracing/events/snd_firewire_lib/amdtp_packet/format ... print fmt: "%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u %s", REC->second, REC->cycle, REC->src, REC->dest, REC->channel, REC->payload_quadlets, REC->data_blocks, REC->data_block_counter, REC->packet_index, REC->irq, REC->index, __print_array(__get_dynamic_array(cip_header), __get_dynamic_array_len(cip_header), sizeof(u8)) $ sudo perf record -e snd_firewire_lib:amdtp_packet [snd_firewire_lib:amdtp_packet] function sizeof not defined Error: expected type 5 but read 0 This commit fixes it by obsoleting the macro with actual size. Cc: <stable@vger.kernel.org> Fixes: bde2bbdb ("ALSA: firewire-lib: use dynamic array for CIP header of tracing events") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20200503045718.86337-1-o-takashi@sakamocchi.jpSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 01 May, 2020 1 commit
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Jesus Ramos authored
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.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 30 Apr, 2020 2 commits
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Takashi Iwai authored
The USB vendor ID of NuPrime DAC-10 is not 16b0 but 16d0. Fixes: f656891c ("ALSA: usb-audio: add more quirks for DSD interfaces") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200430124755.15940-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Arnd Bergmann authored
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: 1841f613 ("[ALSA] Add snd-miro driver") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 29 Apr, 2020 1 commit
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Wu Bo authored
Fix the following coccicheck warning: sound/pci/hda/patch_hdmi.c:1852:2-8: preceding lock on line 1846 After add sanity check to pass klockwork check, The spdif_mutex should be unlock before return true in check_non_pcm_per_cvt(). Fixes: 960a581e ("ALSA: hda: fix some klockwork scan warnings") Signed-off-by: Wu Bo <wubo40@huawei.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1587907042-694161-1-git-send-email-wubo40@huawei.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 28 Apr, 2020 1 commit
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Kai Vehmanen authored
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/1687Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 27 Apr, 2020 1 commit
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Hui Wang authored
This new Lenovo ThinkCenter has two front mics which can't be handled by PA so far, so apply the fixup ALC283_FIXUP_HEADSET_MIC to change the location for one of the mics. Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200427030039.10121-1-hui.wang@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 26 Apr, 2020 1 commit
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Vasily Khoruzhick authored
Apparently interface 1 is control interface akin to HD500X, setting LINE6_CAP_CONTROL and choosing it as ctrl_if fixes audio playback on POD HD500. Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200425201115.3430-1-anarsoul@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 24 Apr, 2020 4 commits
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Takashi Iwai authored
An empty merge of PCM OSS fix for 5.6 code base. The fix for 5.7 was already applied. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
[ This is again a forward-port of the fix applied for 5.6-base code (commit 4285de07) to 5.7-base, hence neither Fixes nor Cc-to-stable tags are included here -- tiwai ] The checks of the plugin buffer overflow in the previous fix by commit f2ecf903 ("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. Link: https://lore.kernel.org/r/20200424193843.20397-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The checks of the plugin buffer overflow in the previous fix by commit f2ecf903 ("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: f2ecf903 ("ALSA: pcm: oss: Avoid plugin buffer overflow") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200424193350.19678-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The commit 3c6fd1f0 ("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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 23 Apr, 2020 3 commits
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Takashi Iwai authored
HD-audio codec driver applies a tricky procedure to forcibly perform the runtime resume by mimicking the usage count even if the device has been runtime-suspended beforehand. This was needed to assure to trigger the jack detection update after the system resume. And recently we also applied the similar logic to the HD-audio controller side. However this seems leading to some inconsistency, and eventually PCI controller gets screwed up. This patch is an attempt to fix and clean up those behavior: instead of the tricky runtime resume procedure, the existing jackpoll work is scheduled when such a forced codec resume is required. The jackpoll work will power up the codec, and this alone should suffice for the jack status update in usual cases. If the extra polling is requested (by checking codec->jackpoll_interval), the manual update is invoked after that, and the codec is powered down again. Also, we filter the spurious wake up of the codec from the controller runtime resume by checking codec->relaxed_resume flag. If this flag is set, basically we don't need to wake up explicitly, but it's supposed to be done via the audio component notifier. Fixes: c4c8dd6e ("ALSA: hda: Skip controller resume if not needed") Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Kailang Yang authored
Enable new codec supported for ALC245. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/8c0804738b2c42439f59c39c8437817f@realtek.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Xiyu Yang authored
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: 447d6275 ("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.cnSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 22 Apr, 2020 1 commit
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Takashi Iwai authored
It turned out that ALC1220-VB USB-audio device gives the interrupt event to some PCM terminals while those don't allow the connector state request but only the actual I/O terminals return the request. The recent commit 7dc3c5a0 ("ALSA: usb-audio: Don't create jack controls for PCM terminals") excluded those phantom terminals, so those events are ignored, too. My first thought was that this could be easily deduced from the associated terminals, but some of them have even no associate terminal ID, hence it's not too trivial to figure out. Since the number of such terminals are small and limited, this patch implements another quirk table for the simple mapping of the connectors. It's not really scalable, but let's hope that there will be not many such funky devices in future. Fixes: 7dc3c5a0 ("ALSA: usb-audio: Don't create jack controls for PCM terminals") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 21 Apr, 2020 5 commits
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Takashi Iwai authored
Merge tag 'asoc-fix-v5.7-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.7 Quite a lot of fixes here, a lot of driver specific ones but the biggest one is the revert of changes to the startup and shutdown sequence for DAIs that went in during the merge window - they broke some older x86 platforms and attempts to fix them didn't succeed so it's safer to just roll them back and try to make sure those platforms are handled properly in any future attempt. The rockchip S/PDIF DT stuff was IIRC for validation issues.
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Alexander Tsoy authored
Due to rounding error driver sometimes incorrectly calculate next packet size, which results in audible clicks on devices with synchronous playback endpoints. For example on a high speed bus and a sample rate 44.1 kHz it loses one sample every ~40.9 seconds. Fortunately playback interface on Scarlett 2i4 2nd gen has a working explicit feedback endpoint, so we can switch playback data endpoint to asynchronous mode as a workaround. Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200421190908.462860-1-alexander@tsoy.meSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Shengjiu Wang authored
After suspend & resume, wm8960_hw_params may be called when bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking is not called. But if sample rate is changed at that time, then the output clock rate will be not correct. So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params is not necessary and it causes above issue. Fixes: 3176bf2d ("ASoC: wm8960: update pll and clock setting function") Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/1587468525-27514-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Takashi Iwai authored
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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Gregor Pintar authored
Force it to use asynchronous playback. Same quirk has already been added for Focusrite Scarlett Solo (2nd gen) with a commit 46f5710f ("ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo"). This also seems to prevent regular clicks when playing at 44100Hz on Scarlett 2i2 (2nd gen). I did not notice any side effects. Moved both quirks to snd_usb_audioformat_attributes_quirk() as suggested. Signed-off-by: Gregor Pintar <grpintar@gmail.com> Reviewed-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 20 Apr, 2020 9 commits
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YueHaibing authored
sound/soc/codecs/wm8900.o: In function `wm8900_i2c_probe': wm8900.c:(.text+0xa36): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8900.o: In function `wm8900_modinit': wm8900.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8900.o: In function `wm8900_exit': wm8900.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8988.o: In function `wm8988_i2c_probe': wm8988.c:(.text+0x857): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8988.o: In function `wm8988_modinit': wm8988.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8988.o: In function `wm8988_exit': wm8988.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8995.o: In function `wm8995_i2c_probe': wm8995.c:(.text+0x1c4f): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8995.o: In function `wm8995_modinit': wm8995.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8995.o: In function `wm8995_exit': wm8995.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' Add SND_SOC_I2C_AND_SPI dependency to fix this. Fixes: ea00d952 ("ASoC: Use imply for SND_SOC_ALL_CODECS") Reported-by: Hulk Robot <hulkci@huawei.com> Signed-off-by: YueHaibing <yuehaibing@huawei.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200420125343.20920-1-yuehaibing@huawei.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Mark Brown authored
Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz <matthias.blankertz@cetitec.com>: Fix rsnd_dai_call() operations being performed twice for the master SSI in multi-SSI setups, and fix the rsnd_ssi_stop operation for multi-SSI setups. The only visible effect of these issues was some "status check failed" spam when the rsnd_ssi_stop was called, but overall the code is cleaner now, and some questionable writes to the SSICR register which did not lead to any observable misbehaviour but were contrary to the datasheet are fixed. Mark: The first patch kind of reverts my "ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode" from a few days ago and achieves the same effect in a simpler fashion, if you would prefer a clean patch series based on v5.6 drop me a note. Greetings, Matthias Matthias Blankertz (2): ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: rsnd: Fix "status check failed" spam for multi-SSI sound/soc/sh/rcar/ssi.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) base-commit: 15a5760c -- 2.26.1
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Mark Brown authored
Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet <jbrunet@baylibre.com>: This patchset fixes the problem reported by Marc in this thread [0] The problem was due to an error in the meson card drivers which had the "no_pcm" dai_link property set on codec-to-codec links [0]: https://lore.kernel.org/r/20200417122732.GC5315@sirena.org.uk Jerome Brunet (2): ASoC: meson: axg-card: fix codec-to-codec link setup ASoC: meson: gx-card: fix codec-to-codec link setup sound/soc/meson/axg-card.c | 4 +++- sound/soc/meson/gx-card.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.25.2
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Gyeongtaek Lee authored
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.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Matthias Blankertz authored
Fix the rsnd_ssi_stop function to skip disabling the individual SSIs of a multi-SSI setup, as the actual stop is performed by rsnd_ssiu_stop_gen2 - the same logic as in rsnd_ssi_start. The attempt to disable these SSIs was harmless, but caused a "status check failed" message to be printed for every SSI in the multi-SSI setup. The disabling of interrupts is still performed, as they are enabled for all SSIs in rsnd_ssi_init, but care is taken to not accidentally set the EN bit for an SSI where it was not set by rsnd_ssi_start. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Matthias Blankertz authored
The master SSI of a multi-SSI setup was attached both to the RSND_MOD_SSI slot and the RSND_MOD_SSIP slot of the rsnd_dai_stream. This is not correct wrt. the meaning of being "parent" in the rest of the SSI code, where it seems to indicate an SSI that provides clock and word sync but is not transmitting/receiving audio data. Not treating the multi-SSI master as parent allows removal of various special cases to the rsnd_ssi_is_parent conditions introduced in commit a09fb3f2 ("ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode"). It also fixes the issue that operations performed via rsnd_dai_call() were performed twice for the master SSI. This caused some "status check failed" spam when stopping a multi-SSI stream as the driver attempted to stop the master SSI twice. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Jerome Brunet authored
Since the addition of commit 9b5db059 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported"), meson-axg cards which have codec-to-codec links fail to init and Oops. Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... While this error was initially reported the axg-card type, it also applies to the gx-card type. While initiliazing the links, ASoC treats the codec-to-codec links of this card type as a DPCM backend. This error eventually leads to the Oops. Most of the card driver code is shared between DPCM backends and codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on codec-to-codec links, leading to this problem. This commit fixes that. Fixes: e37a0c31 ("ASoC: meson: gx: add sound card support") Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200420114511.450560-3-jbrunet@baylibre.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Jerome Brunet authored
Since the addition of commit 9b5db059 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported"), meson-axg cards which have codec-to-codec links fail to init and Oops: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... While initiliazing the links, ASoC treats the codec-to-codec links of this card type as a DPCM backend. This error eventually leads to the Oops. Most of the card driver code is shared between DPCM backends and codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on codec-to-codec links, leading to this problem. This commit fixes that. Fixes: 0a8f1117 ("ASoC: meson: axg-card: add basic codec-to-codec link support") Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.comSigned-off-by: Mark Brown <broonie@kernel.org>
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Takashi Iwai authored
TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need yet more quirks for the proper control names. This patch provides the mapping table for those boards, correcting the FU names for volume and mute controls as well as the terminal names for jack controls. It also improves build_connector_control() not to add the directional suffix blindly if the string is given from the mapping table. With this patch applied, the new UCM profiles will be effective. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 19 Apr, 2020 1 commit
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Takashi Iwai authored
The commit 3c6fd1f0 ("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: 3c6fd1f0 ("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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 18 Apr, 2020 2 commits
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Takashi Iwai authored
The commit 1c76aa5f ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") changed the way to assign spec->init_amp field that specifies the way to initialize the amp. Along with the change, the commit also replaced a few fixups that set spec->init_amp in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather aligning to the other fixups, and not supposed to change the actual behavior. However, this change turned out to cause a regression on FSC S7020, which hit exactly the above. The reason was that there is still one place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE call, namely in alc_ssid_check(). This patch fixes the regression by adding the proper spec->init_amp override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED. Fixes: 1c76aa5f ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") Cc: <stable@vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329 Link: https://lore.kernel.org/r/20200418190639.10082-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Alexander Tsoy authored
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.meSigned-off-by: Takashi Iwai <tiwai@suse.de>
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