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* Merge tag 'sound-5.7-rc6' of ↵Linus Torvalds2020-05-156-13/+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
| * ALSA: hda/realtek - Limit int mic boost for Thinkpad T530Takashi Iwai2020-05-141-1/+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>
| * ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DATakashi Iwai2020-05-121-0/+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>
| * ALSA: hda/realtek: Enable headset mic of ASUS UX581LV with ALC295Jian-Hong Pan2020-05-121-0/+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>
| * ALSA: hda/realtek - Enable headset mic of ASUS UX550GE with ALC295Jian-Hong Pan2020-05-121-0/+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>
| * ALSA: hda/realtek - Enable headset mic of ASUS GL503VM with ALC295Chris Chiu2020-05-121-0/+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>
| * ALSA: hda/realtek: Add quirk for Samsung NotebookMike Pozulp2020-05-101-0/+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>
| * ALSA: rawmidi: Fix racy buffer resize under concurrent accessesTakashi Iwai2020-05-071-4/+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>
| * ALSA: usb-audio: add mapping for ASRock TRX40 CreatorAndrew Oakley2020-05-042-0/+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>
| * ALSA: hda/realtek - Fix S3 pop noise on Dell WyseKai-Heng Feng2020-05-031-0/+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>
| * Revert "ALSA: hda/realtek: Fix pop noise on ALC225"Kai-Heng Feng2020-05-031-2/+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>
| * ALSA: firewire-lib: fix 'function sizeof not defined' error of tracepoints ↵Takashi Sakamoto2020-05-031-2/+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>
| * ALSA: usb-audio: Add control message quirk delay for Kingston HyperX headsetJesus Ramos2020-05-011-4/+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>
* | Merge tag 'sound-5.7-rc4' of ↵Linus Torvalds2020-05-018-33/+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
| * ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB IDTakashi Iwai2020-04-301-1/+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>
| * ALSA: opti9xx: shut up gcc-10 range warningArnd Bergmann2020-04-302-6/+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>
| * ALSA: hda/hdmi: fix without unlocked before returnWu Bo2020-04-291-1/+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>
| * ALSA: hda/hdmi: fix race in monitor detection during probeKai Vehmanen2020-04-281-0/+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>
| * ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenterHui Wang2020-04-271-0/+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>
| * ALSA: line6: Fix POD HD500 audio playbackVasily Khoruzhick2020-04-261-17/+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>
| * ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)Takashi Iwai2020-04-241-4/+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>
| * ALSA: hda: Match both PCI ID and SSID for driver blacklistTakashi Iwai2020-04-241-4/+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>
* | Merge tag 'sound-5.7-rc3' of ↵Linus Torvalds2020-04-2439-341/+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 ...
| * ALSA: hda: Always use jackpoll helper for jack update after resumeTakashi Iwai2020-04-232-26/+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>
| * ALSA: hda/realtek - Add new codec supported for ALC245Kailang Yang2020-04-231-0/+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>
| * ALSA: usb-audio: Fix usb audio refcnt leak when getting spdifXiyu Yang2020-04-231-4/+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>
| * ALSA: usb-audio: Add connector notifier delegationTakashi Iwai2020-04-223-0/+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>
| * Merge tag 'asoc-fix-v5.7-rc2' of ↵Takashi Iwai2020-04-2127-217/+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.
| | * ASoC: wm8960: Fix wrong clock after suspend & resumeShengjiu Wang2020-04-211-2/+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>
| | * ASoC: wm89xx: Add missing dependencyYueHaibing2020-04-201-0/+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>
| | * Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz ↵Mark Brown2020-04-201-5/+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
| | | * ASoC: rsnd: Fix "status check failed" spam for multi-SSIMatthias Blankertz2020-04-201-1/+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>
| | | * ASoC: rsnd: Don't treat master SSI in multi SSI setup as parentMatthias Blankertz2020-04-201-4/+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>
| | * | Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet ↵Mark Brown2020-04-202-2/+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
| | | * | ASoC: meson: gx-card: fix codec-to-codec link setupJerome Brunet2020-04-201-1/+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>
| | | * | ASoC: meson: axg-card: fix codec-to-codec link setupJerome Brunet2020-04-201-1/+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>
| | * | | ASoC: dapm: fixup dapm kcontrol widgetGyeongtaek Lee2020-04-201-3/+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>
| | * | ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/BroadwellPierre-Louis Bossart2020-04-172-0/+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>
| | * | ASoC: stm32: sai: fix sai probeOlivier Moysan2020-04-171-7/+5
| | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | pcm config must be set before snd_dmaengine_pcm_register() call. Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Olivier Moysan <olivier.moysan@st.com> Link: https://lore.kernel.org/r/20200417142122.10212-1-olivier.moysan@st.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * Merge series "ASoC: rsnd: Fixes for multichannel HDMI audio output" from ↵Mark Brown2020-04-162-5/+5
| | |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Matthias Blankertz <matthias.blankertz@cetitec.com>: This fixes two issues in the snd-soc-rcar driver blocking multichannel HDMI audio out: The parent SSI in a multi-SSI configuration is not correctly set up and started, and the SSI->HDMI channel mapping is wrong. With these patches, the following device tree snippet can be used on an r8a7795-based platform (Salvator-X) to enable multichannel HDMI audio on HDMI0: rsnd_port1: port@1 { rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; dai-format = "i2s"; bitclock-master = <&rsnd_endpoint1>; frame-master = <&rsnd_endpoint1>; playback = <&ssi0 &ssi1 &ssi2 &ssi9>; }; }; With a capable receiver attached, all of 2ch (stereo), 6ch (e.g. 5.1) and 8ch audio output should work. Matthias Blankertz (2): ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode sound/soc/sh/rcar/ssi.c | 8 ++++---- sound/soc/sh/rcar/ssiu.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) base-commit: 7111951b8d4973bda27ff663f2cf18b663d15b48 -- 2.26.0
| | | * ASoC: rsnd: Fix HDMI channel mapping for multi-SSI modeMatthias Blankertz2020-04-161-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The HDMI?_SEL register maps up to four stereo SSI data lanes onto the sdata[0..3] inputs of the HDMI output block. The upper half of the register contains four blocks of 4 bits, with the most significant controlling the sdata3 line and the least significant the sdata0 line. The shift calculation has an off-by-one error, causing the parent SSI to be mapped to sdata3, the first multi-SSI child to sdata0 and so forth. As the parent SSI transmits the stereo L/R channels, and the HDMI core expects it on the sdata0 line, this causes no audio to be output when playing stereo audio on a multichannel capable HDMI out, and multichannel audio has permutated channels. Fix the shift calculation to map the parent SSI to sdata0, the first child to sdata1 etc. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * ASoC: rsnd: Fix parent SSI start/stop in multi-SSI modeMatthias Blankertz2020-04-161-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The parent SSI of a multi-SSI setup must be fully setup, started and stopped since it is also part of the playback/capture setup. So only skip the SSI (as per commit 203cdf51f288 ("ASoC: rsnd: SSI parent cares SWSP bit") and commit 597b046f0d99 ("ASoC: rsnd: control SSICR::EN correctly")) if the SSI is parent outside of a multi-SSI setup. Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entryAmadeusz Sławiński2020-04-161-3/+3
| | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | If we don't find any pcm, pcm will point at address at an offset from the the list head and not a meaningful structure. Fix this by returning correct pcm if found and NULL if not. Found with coccinelle. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: soc-dai: revert all changes to DAI startup/shutdown sequencePierre-Louis Bossart2020-04-151-9/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On Baytrail/Cherrytrail, the Atom/SST driver fails miserably: [ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed [ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed Commit b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") was the initial problematic commit. Commit 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") was an attempt to fix things but it does not work on Baytrail, reverting all changes seems necessary for now. Fixes: 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Tested-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20200415030437.23803-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: topology: Fix endianness issueAmadeusz Sławiński2020-04-151-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | As done in already existing cases, we should use le32_to_cpu macro while accessing hdr->magic. Found with sparse. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Link: https://lore.kernel.org/r/20200415162435.31859-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIsStephan Gerhold2020-04-151-0/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For some reason, the MI2S DAIs do not have channels_min/max defined. This means that snd_soc_dai_stream_valid() returns false, i.e. the DAIs have neither valid playback nor capture stream. It's quite surprising that this ever worked correctly, but in 5.7-rc1 this is now failing badly: :) Commit 0e9cf4c452ad ("ASoC: pcm: check if cpu-dai supports a given stream") introduced a check for snd_soc_dai_stream_valid() before calling hw_params(), which means that the q6i2s_hw_params() function was never called, eventually resulting in: qcom-q6afe aprsvc:q6afe:4:4: no line is assigned ... even though "qcom,sd-lines" is set in the device tree. Commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") now even avoids creating PCM devices if the stream is not supported, which means that it is failing even earlier with e.g.: Primary MI2S: ASoC: no backend playback stream Avoid all that trouble by adding channels_min/max for the MI2S DAIs. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Link: https://lore.kernel.org/r/20200415150050.616392-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: soc-pcm: dpcm: Only allow playback/capture if supportedStephan Gerhold2020-04-151-2/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | At the moment, PCM devices for DPCM are only created based on the dpcm_playback/capture parameters of the DAI link, without considering if the CPU/FE DAI is actually capable of playback/capture. Normally the dpcm_playback/capture parameter should match the capabilities of the CPU DAI. However, there is no way to set that parameter from the device tree (e.g. with simple-audio-card or qcom sound cards). dpcm_playback/capture are always both set to 1. This causes problems when the CPU DAI does only support playback or capture. Attemting to open that PCM device with an unsupported stream type then results in a null pointer dereference: 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 ... ... because the DAI playback/capture_widget is not set in that case. We could add checks there to fix the problem (maybe we should anyway), but much easier is to not expose the device as playback/capture in the first place. Attemting to use that device would always fail later anyway. Add checks for snd_soc_dai_stream_valid() to the DPCM case to avoid exposing playback/capture if it is not supported. Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: sgtl5000: Fix VAG power-on handlingSebastian Reichel2020-04-152-0/+35
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As mentioned slightly out of patch context in the code, there is no reset routine for the chip. On boards where the chip is supplied by a fixed regulator, it might not even be resetted during (e.g. watchdog) reboot and can be in any state. If the device is probed with VAG enabled, the driver's probe routine will generate a loud pop sound when ANA_POWER is being programmed. Avoid this by properly disabling just the VAG bit and waiting the required power down time. Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com> Reviewed-by: Fabio Estevam <festivem@gmail.com> Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: madera: Remove a couple of stray blank linesCharles Keepax2020-04-141-4/+0
| | | | | | | | | | | | | | | | | | Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200409181311.30247-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: wsa881x: mark read_only_wordlength flagSrinivas Kandagatla2020-04-141-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown <broonie@kernel.org>
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