Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices,
a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC
(USB ID 0414:a000). The latter serves solely for the headphone output
on the front panel while the former serves for the rest I/Os (mostly
for the I/Os in the rear panel but also including the front mic).
Both chips do work more or less with the unmodified USB-audio driver,
but there are a few glitches. The ALC1220-VB returns an error for an
inquiry to some jacks, as already seen on other TRX40-based mobos.
However this machine has a slightly incompatible configuration, hence
the existing mapping cannot be used as is.
Meanwhile the ESS chip seems working without any quirk. But since
both audio devices don't provide any specific names, both cards appear
as "USB-Audio", and it's quite confusing for users.
This patch is an attempt to overcome those issues:
- The specific mapping table for ALC1220-VB is provided, reducing the
non-working nodes and renaming the badly chosen controls.
The connector map isn't needed here unlike other TRX40 quirks.
- For both USB IDs (0414:a000 and 0414:a001), provide specific card
name strings, so that user-space can identify more easily; and more
importantly, UCM profile can be applied to each.
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526082810.29506-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Asus USB DAC is a USB type-C audio dongle for connecting to
the headset and headphone. The volume minimum value -23040 which
is 0xa600 in hexadecimal with the resolution value 1 indicates
this should be endianness issue caused by the firmware bug. Add
a volume quirk to fix the volume control problem.
Also fixes this warning:
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062613.55401-1-chiu@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We fixed the regression of the speaker volume for some Thinkpad models
(e.g. T570) by the commit 54947cd64c ("ALSA: hda/realtek - Fix
speaker output regression on Thinkpad T570"). Essentially it fixes
the DAC / pin pairing by a static table. It was confirmed and merged
to stable kernel later.
Now, interestingly, we got another regression report for the very same
model (T570) about the similar problem, and the commit above was the
culprit. That is, by some reason, there are devices that prefer the
DAC1, and another device DAC2!
Unfortunately those have the same ID and we have no idea what can
differentiate, in this patch, a new fixup model "tpt470-dock-fix" is
provided, so that users with such a machine can apply it manually.
When model=tpt470-dock-fix option is passed to snd-hda-intel module,
it avoids the fixed DAC pairing and the DAC1 is assigned to the
speaker like the earlier versions.
Fixes: 54947cd64c ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570")
BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1172017
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062406.9799-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "info.index" variable can be 31 in "1 << info.index".
This might trigger an undefined behavior since 1 is signed.
Fix this by casting 1 to 1u just to be sure "1u << 31" is defined.
Signed-off-by: Changming Liu <liu.changm@northeastern.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/BL0PR06MB4548170B842CB055C9AF695DE5B00@BL0PR06MB4548.namprd06.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The ST Audio ADCIII is an STDSP24 card plus extension box. With commit
e8a91ae18b ("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: e8a91ae18b ("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>
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>
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>
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>
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>
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>
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>
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>
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>
Commit 317d931392 ("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: 317d931392 ("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>
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: bde2bbdb30 ("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>
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: 1841f613fd ("[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>
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: 960a581e22 ("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>
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>
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>
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>
[ This is again a forward-port of the fix applied for 5.6-base code
(commit 4285de0725) 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
f2ecf903ef ("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>
The checks of the plugin buffer overflow in the previous fix by commit
f2ecf903ef ("ALSA: pcm: oss: Avoid plugin buffer overflow")
are put in the wrong places mistakenly, which leads to the expected
(repeated) sound when the rate plugin is involved. Fix in the right
places.
Also, at those right places, the zero check is needed for the
termination node, so added there as well, and let's get it done,
finally.
Fixes: f2ecf903ef ("ALSA: pcm: oss: Avoid plugin buffer overflow")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200424193350.19678-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 3c6fd1f07e ("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>
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: c4c8dd6ef8 ("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>
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: 447d6275f0 ("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>
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 7dc3c5a017 ("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: 7dc3c5a017 ("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>
Quite a lot of fixes here, a lot of driver specific ones but the biggest
one is the revert of changes to the startup and shutdown sequence for
DAIs that went in during the merge window - they broke some older x86
platforms and attempts to fix them didn't succeed so it's safer to just
roll them back and try to make sure those platforms are handled properly
in any future attempt.
The rockchip S/PDIF DT stuff was IIRC for validation issues.
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Merge tag 'asoc-fix-v5.7-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.7
Quite a lot of fixes here, a lot of driver specific ones but the biggest
one is the revert of changes to the startup and shutdown sequence for
DAIs that went in during the merge window - they broke some older x86
platforms and attempts to fix them didn't succeed so it's safer to just
roll them back and try to make sure those platforms are handled properly
in any future attempt.
The rockchip S/PDIF DT stuff was IIRC for validation issues.
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>
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: 3176bf2d7c ("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>
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>
Force it to use asynchronous playback.
Same quirk has already been added for Focusrite Scarlett Solo (2nd gen)
with a commit 46f5710f0b ("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>
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: ea00d95200 ("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>
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: 15a5760cb8
--
2.26.1
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
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>
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>
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
a09fb3f28a ("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>
Since the addition of commit 9b5db05936 ("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: e37a0c313a ("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>
Since the addition of commit 9b5db05936 ("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: 0a8f1117a6 ("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>
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>
The commit 3c6fd1f07e ("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: 3c6fd1f07e ("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>
