- 26 5月, 2020 2 次提交
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由 Takashi Iwai 提交于
We fixed the regression of the speaker volume for some Thinkpad models (e.g. T570) by the commit 54947cd6 ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570"). Essentially it fixes the DAC / pin pairing by a static table. It was confirmed and merged to stable kernel later. Now, interestingly, we got another regression report for the very same model (T570) about the similar problem, and the commit above was the culprit. That is, by some reason, there are devices that prefer the DAC1, and another device DAC2! Unfortunately those have the same ID and we have no idea what can differentiate, in this patch, a new fixup model "tpt470-dock-fix" is provided, so that users with such a machine can apply it manually. When model=tpt470-dock-fix option is passed to snd-hda-intel module, it avoids the fixed DAC pairing and the DAC1 is assigned to the speaker like the earlier versions. Fixes: 54947cd6 ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570") BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1172017 Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200526062406.9799-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Changming Liu 提交于
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: NChangming Liu <liu.changm@northeastern.edu> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/BL0PR06MB4548170B842CB055C9AF695DE5B00@BL0PR06MB4548.namprd06.prod.outlook.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 19 5月, 2020 2 次提交
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由 PeiSen Hou 提交于
A few known Clevo machines (PC50, PC70, X170) with ALC1220 codec need the existing quirk for pins for PB51 and co. Signed-off-by: NPeiSen Hou <pshou@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200519065012.13119-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Scott Bahling 提交于
The ST Audio ADCIII is an STDSP24 card plus extension box. With commit e8a91ae1 ("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: e8a91ae1 ("ALSA: ice1712: Add support for STAudio ADCIII") Signed-off-by: NScott 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.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 5月, 2020 2 次提交
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由 Christian Lachner 提交于
The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275Signed-off-by: NChristian Lachner <gladiac@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200518053844.42743-2-gladiac@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Brent Lu 提交于
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: NBrent Lu <brent.lu@intel.com> Reviewed-by: NJaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/1589776238-23877-1-git-send-email-brent.lu@intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 5月, 2020 1 次提交
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由 Takashi Iwai 提交于
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.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 5月, 2020 4 次提交
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由 Takashi Iwai 提交于
ASUS ZenBook UX431DA requires an additional COEF setup when booted from the recent Windows 10, otherwise it produces the noisy output. The quirk turns on COEF 0x1b bit 10 that has been cleared supposedly due to the pop noise reduction. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207553 Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200512073203.14091-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Jian-Hong Pan 提交于
The ASUS UX581LV laptop's audio (1043:19e1) with ALC295 can't detect the headset microphone until ALC295_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: NJian-Hong Pan <jian-hong@endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-3-jian-hong@endlessm.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Jian-Hong Pan 提交于
The ASUS laptop UX550GE with ALC295 can't detect the headset microphone until ALC295_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: NJian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: NDaniel Drake <drake@endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-2-jian-hong@endlessm.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Chris Chiu 提交于
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: NChris Chiu <chiu@endlessm.com> Signed-off-by: NDaniel Drake <drake@endlessm.com> Signed-off-by: NJian-Hong Pan <jian-hong@endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-1-jian-hong@endlessm.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 5月, 2020 1 次提交
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由 Mike Pozulp 提交于
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: NMike 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: NTakashi Iwai <tiwai@suse.de>
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- 08 5月, 2020 1 次提交
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由 Takashi Iwai 提交于
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: Nbutt3rflyh4ck <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: NTakashi Iwai <tiwai@suse.de>
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- 04 5月, 2020 3 次提交
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由 Andrew Oakley 提交于
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: NAndrew Oakley <andrew@adoakley.name> Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.nameSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Kai-Heng Feng 提交于
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: NKai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-2-kai.heng.feng@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Kai-Heng Feng 提交于
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: NKai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-1-kai.heng.feng@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 03 5月, 2020 1 次提交
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由 Takashi Sakamoto 提交于
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: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20200503045718.86337-1-o-takashi@sakamocchi.jpSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 5月, 2020 1 次提交
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由 Jesus Ramos 提交于
Kingston HyperX headset with 0951:16ad also needs the same quirk for delaying the frequency controls. Signed-off-by: NJesus Ramos <jesus-ramos@live.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/BY5PR19MB3634BA68C7CCA23D8DF428E796AF0@BY5PR19MB3634.namprd19.prod.outlook.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 30 4月, 2020 2 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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由 Arnd Bergmann 提交于
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: NArnd Bergmann <arnd@arndb.de> Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 29 4月, 2020 1 次提交
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由 Wu Bo 提交于
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: NWu 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: NTakashi Iwai <tiwai@suse.de>
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- 28 4月, 2020 1 次提交
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由 Kai Vehmanen 提交于
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: NKai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 4月, 2020 1 次提交
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由 Hui Wang 提交于
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: NHui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20200427030039.10121-1-hui.wang@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 4月, 2020 1 次提交
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由 Vasily Khoruzhick 提交于
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: NVasily Khoruzhick <anarsoul@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200425201115.3430-1-anarsoul@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 25 4月, 2020 3 次提交
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由 Takashi Iwai 提交于
An empty merge of PCM OSS fix for 5.6 code base. The fix for 5.7 was already applied. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
[ 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: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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- 24 4月, 2020 1 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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- 23 4月, 2020 3 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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由 Kailang Yang 提交于
Enable new codec supported for ALC245. Signed-off-by: NKailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/8c0804738b2c42439f59c39c8437817f@realtek.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Xiyu Yang 提交于
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: NXiyu Yang <xiyuyang19@fudan.edu.cn> Signed-off-by: NXin 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: NTakashi Iwai <tiwai@suse.de>
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- 22 4月, 2020 3 次提交
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
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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由 Alexander Tsoy 提交于
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: NAlexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200421190908.462860-1-alexander@tsoy.meSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 4月, 2020 3 次提交
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由 Shengjiu Wang 提交于
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: NShengjiu Wang <shengjiu.wang@nxp.com> Acked-by: NCharles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/1587468525-27514-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Takashi Iwai 提交于
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: NLin Yi <teroincn@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Gregor Pintar 提交于
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: NGregor Pintar <grpintar@gmail.com> Reviewed-by: NAlexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 4月, 2020 3 次提交
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由 YueHaibing 提交于
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: NHulk Robot <hulkci@huawei.com> Signed-off-by: NYueHaibing <yuehaibing@huawei.com> Acked-by: NCharles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200420125343.20920-1-yuehaibing@huawei.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Mark Brown 提交于
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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由 Mark Brown 提交于
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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