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2020-05-22Merge tag 'sound-5.7-rc7' of ↵Linus Torvalds3-1/+7
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Just a few small fixes: the only significant one is a slight improvement for PCM running position update with no-period-elapsed case while the rest are HD-audio fixups and ice1712 model quirk" * tag 'sound-5.7-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda/realtek - Add more fixup entries for Clevo machines ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio option ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Xtreme ALSA: pcm: fix incorrect hw_base increase
2020-05-19ALSA: hda/realtek - Add more fixup entries for Clevo machinesPeiSen Hou1-0/+3
A few known Clevo machines (PC50, PC70, X170) with ALC1220 codec need the existing quirk for pins for PB51 and co. Signed-off-by: PeiSen Hou <pshou@realtek.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200519065012.13119-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-18ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio optionScott Bahling1-1/+2
The ST Audio ADCIII is an STDSP24 card plus extension box. With commit e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") we enabled the ADCIII ports using the model=staudio option but forgot this part to ensure the STDSP24 card is initialized properly. Fixes: e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") Signed-off-by: Scott Bahling <sbahling@suse.com> Cc: <stable@vger.kernel.org> BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1048934 Link: https://lore.kernel.org/r/20200518175728.28766-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-18ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus XtremeChristian Lachner1-0/+1
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=205275 Signed-off-by: Christian Lachner <gladiac@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200518053844.42743-2-gladiac@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-18ALSA: pcm: fix incorrect hw_base increaseBrent Lu1-0/+1
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>
2020-05-15Merge tag 'sound-5.7-rc6' of ↵Linus Torvalds6-13/+110
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Things look good and calming down; the only change to ALSA core is the fix for racy rawmidi buffer accesses spotted by syzkaller, and the rest are all small device-specific quirks for HD-audio and USB-audio devices" * tag 'sound-5.7-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda/realtek - Limit int mic boost for Thinkpad T530 ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DA ALSA: hda/realtek: Enable headset mic of ASUS UX581LV with ALC295 ALSA: hda/realtek - Enable headset mic of ASUS UX550GE with ALC295 ALSA: hda/realtek - Enable headset mic of ASUS GL503VM with ALC295 ALSA: hda/realtek: Add quirk for Samsung Notebook ALSA: rawmidi: Fix racy buffer resize under concurrent accesses ALSA: usb-audio: add mapping for ASRock TRX40 Creator ALSA: hda/realtek - Fix S3 pop noise on Dell Wyse Revert "ALSA: hda/realtek: Fix pop noise on ALC225" ALSA: firewire-lib: fix 'function sizeof not defined' error of tracepoints format ALSA: usb-audio: Add control message quirk delay for Kingston HyperX headset
2020-05-14ALSA: hda/realtek - Limit int mic boost for Thinkpad T530Takashi Iwai1-1/+9
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>
2020-05-12ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DATakashi Iwai1-0/+13
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: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200512073203.14091-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-12ALSA: hda/realtek: Enable headset mic of ASUS UX581LV with ALC295Jian-Hong Pan1-0/+1
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: Jian-Hong Pan <jian-hong@endlessm.com> Link: https://lore.kernel.org/r/20200512061525.133985-3-jian-hong@endlessm.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-12ALSA: hda/realtek - Enable headset mic of ASUS UX550GE with ALC295Jian-Hong Pan1-0/+4
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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-12ALSA: hda/realtek - Enable headset mic of ASUS GL503VM with ALC295Chris Chiu1-0/+18
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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-10ALSA: hda/realtek: Add quirk for Samsung NotebookMike Pozulp1-0/+10
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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-07ALSA: rawmidi: Fix racy buffer resize under concurrent accessesTakashi Iwai1-4/+27
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>
2020-05-04ALSA: usb-audio: add mapping for ASRock TRX40 CreatorAndrew Oakley2-0/+6
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=0a742f89066527497b77ce16bca486daccf8a70c Signed-off-by: Andrew Oakley <andrew@adoakley.name> Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-03ALSA: hda/realtek - Fix S3 pop noise on Dell WyseKai-Heng Feng1-0/+16
Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. The previous fix enable power save node universally for ALC225, however it makes some ALC225 systems unable to produce any sound. So let's only enable power save node for the affected Dell Wyse platform. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Link: https://lore.kernel.org/r/20200503152449.22761-2-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-03Revert "ALSA: hda/realtek: Fix pop noise on ALC225"Kai-Heng Feng1-2/+0
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>
2020-05-03ALSA: firewire-lib: fix 'function sizeof not defined' error of tracepoints ↵Takashi Sakamoto1-2/+1
format 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: bde2bbdb307a ("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.jp Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-01Merge tag 'sound-5.7-rc4' of ↵Linus Torvalds8-33/+35
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Just a collection of small fixes around this time: - One more try for fixing PCM OSS regression - HD-audio: a new quirk for Lenovo, the improved driver blacklisting, a lock fix in the minor error path, and a fix for the possible race at monitor notifiaction - USB-audio: a quirk ID fix, a fix for POD HD500 workaround" * tag 'sound-5.7-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB ID ALSA: opti9xx: shut up gcc-10 range warning ALSA: hda/hdmi: fix without unlocked before return ALSA: hda/hdmi: fix race in monitor detection during probe ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter ALSA: line6: Fix POD HD500 audio playback ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7) ALSA: pcm: oss: Place the plugin buffer overflow checks correctly ALSA: hda: Match both PCI ID and SSID for driver blacklist
2020-05-01ALSA: usb-audio: Add control message quirk delay for Kingston HyperX headsetJesus Ramos1-4/+5
Kingston HyperX headset with 0951:16ad also needs the same quirk for delaying the frequency controls. Signed-off-by: Jesus Ramos <jesus-ramos@live.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/BY5PR19MB3634BA68C7CCA23D8DF428E796AF0@BY5PR19MB3634.namprd19.prod.outlook.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-30ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB IDTakashi Iwai1-1/+1
The USB vendor ID of NuPrime DAC-10 is not 16b0 but 16d0. Fixes: f656891c6619 ("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.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-30ALSA: opti9xx: shut up gcc-10 range warningArnd Bergmann2-6/+12
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>
2020-04-29ALSA: hda/hdmi: fix without unlocked before returnWu Bo1-1/+3
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: 960a581e22d9 ("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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-28ALSA: hda/hdmi: fix race in monitor detection during probeKai Vehmanen1-0/+2
A race exists between build_pcms() and build_controls() phases of codec setup. Build_pcms() sets up notifier for jack events. If a monitor event is received before build_controls() is run, the initial jack state is lost and never reported via mixer controls. The problem can be hit at least with SOF as the controller driver. SOF calls snd_hda_codec_build_controls() in its workqueue-based probe and this can be delayed enough to hit the race condition. Fix the issue by invalidating the per-pin ELD information when build_controls() is called. The existing call to hdmi_present_sense() will update the ELD contents. This ensures initial monitor state is correctly reflected via mixer controls. BugLink: https://github.com/thesofproject/linux/issues/1687 Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-27ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenterHui Wang1-0/+1
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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-26ALSA: line6: Fix POD HD500 audio playbackVasily Khoruzhick1-17/+5
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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-24ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)Takashi Iwai1-4/+6
[ This is again a forward-port of the fix applied for 5.6-base code (commit 4285de0725b1) 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 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. Link: https://lore.kernel.org/r/20200424193843.20397-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-24Merge tag 'sound-5.7-rc3' of ↵Linus Torvalds39-341/+605
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "This became a slightly big pull request, as the accumulated ASoC fixes are included here. Some highlights: - Revert of ASoC DAI startup changes that caused regression on some x86 platforms - Regression fix in HD-audio power management and driver blacklist - A collection of ASoC DAPM and topology fixes - Continued USB-audio fixes and quirks - Lots of small device-specific fixes - Rockchip S/PDIF DT stuff update for validation issues" * tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (51 commits) ALSA: hda: Always use jackpoll helper for jack update after resume ALSA: hda/realtek - Add new codec supported for ALC245 ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif ALSA: usb-audio: Add connector notifier delegation ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen ASoC: wm8960: Fix wrong clock after suspend & resume ALSA: usx2y: Fix potential NULL dereference ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2 ASoC: wm89xx: Add missing dependency ASoC: dapm: fixup dapm kcontrol widget ASoC: rsnd: Fix "status check failed" spam for multi-SSI ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: meson: gx-card: fix codec-to-codec link setup ASoC: meson: axg-card: fix codec-to-codec link setup ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos ALSA: hda: Remove ASUS ROG Zenith from the blacklist ALSA: hda/realtek - Fix unexpected init_amp override ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell ASoC: stm32: sai: fix sai probe ...
2020-04-24ALSA: hda: Match both PCI ID and SSID for driver blacklistTakashi Iwai1-4/+5
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>
2020-04-23ALSA: hda: Always use jackpoll helper for jack update after resumeTakashi Iwai2-26/+19
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: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-23ALSA: hda/realtek - Add new codec supported for ALC245Kailang Yang1-0/+3
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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-23ALSA: usb-audio: Fix usb audio refcnt leak when getting spdifXiyu Yang1-4/+8
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>
2020-04-22ALSA: usb-audio: Add connector notifier delegationTakashi Iwai3-0/+48
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 7dc3c5a0172e ("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: 7dc3c5a0172e ("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.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-21Merge tag 'asoc-fix-v5.7-rc2' of ↵Takashi Iwai27-217/+402
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.
2020-04-21ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd genAlexander Tsoy1-0/+1
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.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-21ASoC: wm8960: Fix wrong clock after suspend & resumeShengjiu Wang1-2/+1
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: 3176bf2d7ccd ("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.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-21ALSA: usx2y: Fix potential NULL dereferenceTakashi Iwai1-0/+2
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>
2020-04-21ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2Gregor Pintar2-84/+13
Force it to use asynchronous playback. Same quirk has already been added for Focusrite Scarlett Solo (2nd gen) with a commit 46f5710f0b88 ("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.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-20ASoC: wm89xx: Add missing dependencyYueHaibing1-0/+3
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: ea00d95200d02ece ("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.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz ↵Mark Brown1-5/+14
<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: 15a5760cb8b6d5c1ebbf1d2e1f0b77380ab68a82 -- 2.26.1
2020-04-20Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet ↵Mark Brown2-2/+6
<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
2020-04-20ASoC: dapm: fixup dapm kcontrol widgetGyeongtaek Lee1-3/+17
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>
2020-04-20ASoC: rsnd: Fix "status check failed" spam for multi-SSIMatthias Blankertz1-1/+7
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.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ASoC: rsnd: Don't treat master SSI in multi SSI setup as parentMatthias Blankertz1-4/+7
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 a09fb3f28a60 ("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.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ASoC: meson: gx-card: fix codec-to-codec link setupJerome Brunet1-1/+3
Since the addition of commit 9b5db059366a ("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: e37a0c313a0f ("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.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ASoC: meson: axg-card: fix codec-to-codec link setupJerome Brunet1-1/+3
Since the addition of commit 9b5db059366a ("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: 0a8f1117a680 ("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.com Signed-off-by: Mark Brown <broonie@kernel.org>
2020-04-20ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobosTakashi Iwai3-6/+44
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.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-20ALSA: hda: Remove ASUS ROG Zenith from the blacklistTakashi Iwai1-1/+0
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>
2020-04-18ALSA: hda/realtek - Fix unexpected init_amp overrideTakashi Iwai1-3/+5
The commit 1c76aa5fb48d ("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: 1c76aa5fb48d ("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.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-18ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devicesAlexander Tsoy1-0/+51
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>
2020-04-17ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/BroadwellPierre-Louis Bossart2-0/+64
Major regressions were detected by SOF CI on CherryTrail and Broadwell: [ 25.705750] SSP2-Codec: ASoC: no backend playback stream [ 27.923378] SSP2-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. For some reason the SSP structures used for SKL+ did provide this information but legacy platforms didn't. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200417172014.11760-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
generated by cgit v1.2.3 (git 2.39.0) at 2024-11-26 17:16:47 +0300