Chris Chiu [Tue, 26 May 2020 06:26:13 +0000 (14:26 +0800)]
ALSA: usb-audio: mixer: volume quirk for ESS Technology Asus USB DAC
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
Takashi Iwai [Tue, 26 May 2020 06:24:06 +0000 (08:24 +0200)]
ALSA: hda/realtek - Add a model for Thinkpad T570 without DAC workaround
We fixed the regression of the speaker volume for some Thinkpad models
(e.g. T570) by the commit 5cd14c02e265 ("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.
Scott Bahling [Mon, 18 May 2020 17:57:28 +0000 (19:57 +0200)]
ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio option
The ST Audio ADCIII is an STDSP24 card plus extension box. With commit de36595f0655 ("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.
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.
Brent Lu [Mon, 18 May 2020 04:30:38 +0000 (12:30 +0800)]
ALSA: pcm: fix incorrect hw_base increase
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.
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.
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.
Takashi Iwai [Thu, 14 May 2020 16:05:33 +0000 (18:05 +0200)]
ALSA: hda/realtek - Limit int mic boost for Thinkpad T530
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.
Takashi Iwai [Tue, 12 May 2020 07:32:03 +0000 (09:32 +0200)]
ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DA
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.
Mike Pozulp [Sun, 10 May 2020 03:28:37 +0000 (20:28 -0700)]
ALSA: hda/realtek: Add quirk for Samsung Notebook
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 ]
Takashi Iwai [Thu, 7 May 2020 11:44:56 +0000 (13:44 +0200)]
ALSA: rawmidi: Fix racy buffer resize under concurrent accesses
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.
Andrew Oakley [Sun, 3 May 2020 14:16:39 +0000 (15:16 +0100)]
ALSA: usb-audio: add mapping for ASRock TRX40 Creator
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: firewire-lib: fix 'function sizeof not defined' error of tracepoints 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:
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.
Kai Vehmanen [Tue, 28 Apr 2020 12:38:36 +0000 (15:38 +0300)]
ALSA: hda/hdmi: fix race in monitor detection during probe
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.
Hui Wang [Mon, 27 Apr 2020 03:00:39 +0000 (11:00 +0800)]
ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter
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.
ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)
[ 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 2a86cec927f1 ("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.
ALSA: pcm: oss: Place the plugin buffer overflow checks correctly
The checks of the plugin buffer overflow in the previous fix by commit 2a86cec927f1 ("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.
ALSA: hda: Match both PCI ID and SSID for driver blacklist
The commit 122728626b74 ("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.
ALSA: hda: Always use jackpoll helper for jack update after resume
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.
Xiyu Yang [Thu, 23 Apr 2020 04:54:19 +0000 (12:54 +0800)]
ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
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: 0074a231fe8c ("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 6f2b4ce37644 ("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.
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.
Alexander Tsoy [Tue, 21 Apr 2020 19:09:08 +0000 (22:09 +0300)]
ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen
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.
Shengjiu Wang [Tue, 21 Apr 2020 11:28:45 +0000 (19:28 +0800)]
ASoC: wm8960: Fix wrong clock after suspend & resume
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.
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.
Gregor Pintar [Mon, 20 Apr 2020 21:40:30 +0000 (23:40 +0200)]
ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2
Force it to use asynchronous playback.
Same quirk has already been added for Focusrite Scarlett Solo (2nd gen)
with a commit 060910e2a1d6 ("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.
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: 4eaceddd7fb51595 ("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>
Mark Brown [Mon, 20 Apr 2020 13:35:08 +0000 (14:35 +0100)]
Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz <matthias.blankertz@cetitec.com>:
Fix rsnd_dai_call() operations being performed twice for the master SSI
in multi-SSI setups, and fix the rsnd_ssi_stop operation for multi-SSI
setups.
The only visible effect of these issues was some "status check failed"
spam when the rsnd_ssi_stop was called, but overall the code is cleaner
now, and some questionable writes to the SSICR register which did not
lead to any observable misbehaviour but were contrary to the datasheet
are fixed.
Mark:
The first patch kind of reverts my "ASoC: rsnd: Fix parent SSI
start/stop in multi-SSI mode" from a few days ago and achieves the same
effect in a simpler fashion, if you would prefer a clean patch series
based on v5.6 drop me a note.
Greetings,
Matthias
Matthias Blankertz (2):
ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent
ASoC: rsnd: Fix "status check failed" spam for multi-SSI
Mark Brown [Mon, 20 Apr 2020 13:35:07 +0000 (14:35 +0100)]
Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet <jbrunet@baylibre.com>:
This patchset fixes the problem reported by Marc in this thread [0]
The problem was due to an error in the meson card drivers which had
the "no_pcm" dai_link property set on codec-to-codec links
Gyeongtaek Lee [Sat, 18 Apr 2020 04:13:20 +0000 (13:13 +0900)]
ASoC: dapm: fixup dapm kcontrol widget
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().
ASoC: rsnd: Fix "status check failed" spam for multi-SSI
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.
ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent
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 eb91a8bedc3d ("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.
ASoC: meson: gx-card: fix codec-to-codec link setup
Since the addition of commit ae8b0ee8a07a ("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.
ASoC: meson: axg-card: fix codec-to-codec link setup
Since the addition of commit ae8b0ee8a07a ("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.
ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
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.
ALSA: hda: Remove ASUS ROG Zenith from the blacklist
The commit 122728626b74 ("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.
The commit 9a08cb3c3c0c ("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.
ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
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: ae8b0ee8a07ae6 ("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>
ALSA: hda/hdmi: Add module option to disable audio component binding
As the recent regression showed, we want sometimes to turn off the
audio component binding just for debugging. This patch adds the
module option to control it easily without compilation.
Mark Brown [Thu, 16 Apr 2020 12:01:34 +0000 (13:01 +0100)]
Merge series "ASoC: rsnd: Fixes for multichannel HDMI audio output" from 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:
ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry
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.
ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode
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.
ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode
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 443342d4897e ("ASoC: rsnd: SSI parent cares
SWSP bit") and commit 07b90ac87de1 ("ASoC: rsnd: control SSICR::EN
correctly")) if the SSI is parent outside of a multi-SSI setup.
ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence
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 b8db8fc315ab9 ("ASoC: soc-pcm: call
snd_soc_dai_startup()/shutdown() once") was the initial problematic
commit.
Commit dccc1dc601adc5 ("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.
Stephan Gerhold [Wed, 15 Apr 2020 15:00:50 +0000 (17:00 +0200)]
ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs
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 ec9e33f0fe69 ("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 ae8b0ee8a07a ("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: 72dcc863eb64 ("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>
Stephan Gerhold [Wed, 15 Apr 2020 10:49:28 +0000 (12:49 +0200)]
ASoC: soc-pcm: dpcm: Only allow playback/capture if supported
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.
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.
Hui Wang [Tue, 14 Apr 2020 14:27:25 +0000 (22:27 +0800)]
ALSA: hda: call runtime_allow() for all hda controllers
Before the pci_driver->probe() is called, the pci subsystem calls
runtime_forbid() and runtime_get_sync() on this pci dev, so only call
runtime_put_autosuspend() is not enough to enable the runtime_pm on
this device.
For controllers with vgaswitcheroo feature, the pci/quirks.c will call
runtime_allow() for this dev, then the controllers could enter
rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like
Intel hda controllers, the runtime_pm is not enabled because the
runtime_allow() is not called.
Since it is no harm calling runtime_allow() twice, here let hda
driver call runtime_allow() for all controllers. Then the runtime_pm
is enabled on all controllers after the put_autosuspend() is called.
Johan Jonker [Tue, 24 Mar 2020 09:41:49 +0000 (10:41 +0100)]
ASoC: rockchip-i2s: add power-domains property
In the old txt situation we add/describe only properties that are used
by the driver/hardware itself. With yaml it also filters things in a
node that are used by other drivers like 'power-domains' for rk3399,
so add it to 'rockchip-i2s.yaml'.
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.
All error paths in qcom_snd_parse_of() prints more specific error
messages, so silence the one in apq8096_platform_probe() and
sdm845_snd_platform_probe() to avoid spamming the kernel log.
Johan Jonker [Sat, 4 Apr 2020 11:52:25 +0000 (13:52 +0200)]
ASoC: rockchip-spdif: add power-domains property
In the old txt situation we add/describe only properties that are used
by the driver/hardware itself. With yaml it also filters things in a
node that are used by other drivers like 'power-domains' for rk3399,
so add it to 'rockchip-spdif.yaml'.
sound/soc/intel/common/soc-acpi-intel-cml-match.c:116:45: warning:
‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=]
static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
^~~~~~~~~~~~
sound/soc/intel/common/soc-acpi-intel-icl-match.c:90:45: warning:
‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=]
static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
^~~~~~~~~~~~
Reported-by: Hulk Robot <hulkci@huawei.com> Signed-off-by: Jason Yan <yanaijie@huawei.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200410081117.21319-1-yanaijie@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
If probe fails after enabling the regulators regulator_put is called for
each supply without having them disabled before. This produces some
warnings like
WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c
[<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14)
[<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4)
[<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4)
[<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c)
[<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38)
[<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38)
[<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c)
[<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c)
[<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c)
[<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60)
[<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc)
[<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0)
[<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0)
[<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108)
[<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88)
[<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250)
[<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244)
[<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540)
[<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8)
[<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20)
Charles Keepax [Thu, 9 Apr 2020 18:12:07 +0000 (19:12 +0100)]
ASoC: dapm: Fix regression introducing multiple copies of DAI widgets
Refactoring was done to factor out the linking of DAI widgets into
a helper function, dapm_add_valid_dai_widget. However when this was
done, a regression was introduced for CODEC to CODEC links. It was
over looked that the playback and capture variables persisted across
all CODEC DAIs being processed, which ensured that the special DAI
widget that is added for CODEC to CODEC links was only created once.
This bug causes kernel panics during DAPM shutdown.
To stick with the spirit of the original refactoring whilst fixing the
issue, variables to hold the DAI widgets are added to snd_soc_dai_link.
Furthermore the dapm_add_valid_dai_widget function is renamed to
dapm_connect_dai_pair, the function only adds DAI widgets in the CODEC
to CODEC case and its primary job is to add routes connecting two DAI
widgets, making the original name quite misleading.
ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume
Commit cbe62165a219 ("ASoC: soc-core: remove DAI suspend/resume")
removed the DAI side suspend/resume hooks and switched entirely to
component suspend/resume. However the Samsung SoC s3c-i2s-v2 driver was
not updated.
Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while
changing dai to component which allows to keep the struct
snd_soc_component_driver const.
This fixes build errors:
sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’:
sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’
dai_drv->suspend = s3c2412_i2s_suspend;
Reported-by: Arnd Bergmann <arnd@arndb.de> Fixes: cbe62165a219 ("ASoC: soc-core: remove DAI suspend/resume") Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org> Link: https://lore.kernel.org/r/20200413124548.28197-1-krzk@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA: hda: Allow setting preallocation again for x86
The commit 9c699afeb873 ("ALSA: hda: No preallocation on x86
platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default
to zero for x86, as the preallocation should work almost all cases.
However, this expectation was too naive; some applications try to
allocate as the max buffer size as possible, and it leads to the
memory exhaustion. More badly, the commit changed the kconfig no
longer adjustable for x86, so you can't fix it statically (although it
can be still adjusted via procfs).
So, practically seen, it's more recommended to set a reasonable limit
for x86, too. This patch follows to that experience, and changes the
default to 2048 and allow the kconfig adjustable again.
The HD-audio controller does system-suspend and resume operations by
directly calling its helpers __azx_runtime_suspend() and
__azx_runtime_resume(). However, in general, we don't have to resume
always the device fully at the system resume; typically, if a device
has been runtime-suspended, we can leave it to runtime resume.
Usually for achieving this, the driver would call
pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the
system suspend and resume ops. Unfortunately, this doesn't work for
the resume path in our case. For handling the jack detection at the
system resume, a child codec device may need the (literally) forcibly
resume even if it's been runtime-suspended, and for that, the
controller device must be also resumed even if it's been suspended.
This patch is an attempt to improve the situation. It replaces the
direct __azx_runtime_suspend()/_resume() calls with with
pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight
trick as we've done for the codec side. More exactly:
- azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and
azx_runtime_resume(), so that it can be properly executed from the
system-suspend/resume path
- The WAKEEN handling depends on the card's power state now; it's set
and cleared only for the runtime-suspend
- azx_resume() checks whether any codec may need the forcible resume
beforehand. If the forcible resume is required, it does temporary
PM refcount up/down for actually triggering the runtime resume.
- A new helper function, hda_codec_need_resume(), is introduced for
checking whether the codec needs a forcible runtime-resume, and the
existing code is rewritten with that.
ALSA: hda: Keep the controller initialization even if no codecs found
Currently, when the HD-audio controller driver doesn't detect any
codecs, it tries to abort the probe. But this abort happens at the
delayed probe, i.e. the primary probe call already returned success,
hence the driver is never unbound until user does so explicitly.
As a result, it may leave the HD-audio device in the running state
without the runtime PM. More badly, if the device is a HD-audio bus
that is tied with a GPU, GPU cannot reach to the full power down and
consumes unnecessarily much power.
This patch changes the logic after no-codec situation; it continues
probing without the further codec initialization but keep the
controller driver running normally.
ALSA: hda: Release resources at error in delayed probe
snd-hda-intel driver handles the most of its probe task in the delayed
work (either via workqueue or via firmware loader). When an error
happens in the later delayed probe, we can't deregister the device
itself because the probe callback already returned success and the
device was bound. So, for now, we set hda->init_failed flag and make
the rest untouched until the device gets really unbound.
However, this leaves the device up running, keeping the resources
without any use that prevents other operations.
In this patch, we release the resources at first when a probe error
happens in the delayed probe stage, but keeps the top-level object, so
that the PM and other ops can still refer to the object itself.
Also for simplicity, snd_hda_intel object is allocated via devm, so
that we can get rid of the explicit kfree calls.
ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq ops
freeze_noirq and thaw_noirq need to check the PM availability like
other PM ops. There are cases where the device got disabled due to
the error, and the PM operation should be ignored for that.
ALSA: hda: Don't release card at firmware loading error
At the error path of the firmware loading error, the driver tries to
release the card object and set NULL to drvdata. This may be referred
badly at the possible PM action, as the driver itself is still bound
and the PM callbacks read the card object.
Instead, we continue the probing as if it were no option set. This is
often a better choice than the forced abort, too.
ALSA: usb-audio: Check mapping at creating connector controls, too
Add the mapping check to build_connector_control() so that the device
specific quirk can provide the node to skip for the badly behaving
connector controls. As an example, ALC1220-VB-based codec implements
the skip entry for the broken SPDIF connector detection.
ALSA: usb-audio: Don't create jack controls for PCM terminals
Some funky firmwares set the connector flag even on PCM terminals
although it doesn't make sense (and even actually the firmware doesn't
react properly!). Let's skip creation of jack controls in such a
case.
ALSA: usb-audio: Don't override ignore_ctl_error value from the map
The mapping table may contain also ignore_ctl_error flag for devices
that are known to behave wild. Since this flag always writes the
card's own ignore_ctl_error flag, it overrides the value already set
by the module option, so it doesn't follow user's expectation.
Let's fix the code not to clear the flag that has been set by user.
ALSA: usb-audio: Filter error from connector kctl ops, too
The ignore_ctl_error option should filter the error at kctl accesses,
but there was an overlook: mixer_ctl_connector_get() returns an error
from the request.
This patch covers the forgotten code path and apply filter_error()
properly. The locking error is still returned since this is a fatal
error that has to be reported even with ignore_ctl_error option.
ASoC: topology: Check return value of soc_tplg_*_create
Functions soc_tplg_denum_create, soc_tplg_dmixer_create,
soc_tplg_dbytes_create can fail, so their return values should be
checked and error should be propagated.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200327204729.397-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
kstrdup is an allocation function and it can fail, so its return value
should be checked and handled appropriately.
In order to check all cases, we need to modify set_stream_info to return
a value, so check that everything went correctly when doing kstrdup().
Later add proper checks and error handlers.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200327204729.397-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
The recent AMD platform exposes an HD-audio bus but without any actual
codecs, which is internally tied with a USB-audio device, supposedly.
It results in "no codecs" error of HD-audio bus driver, and it's
nothing but a waste of resources.
This patch introduces a static blacklist table for skipping such a
known bogus PCI SSID entry. As of writing this patch, the known SSIDs
are:
* 1043:874f - ASUS ROG Zenith II / Strix
* 1462:cb59 - MSI TRX40 Creator
* 1462:cb60 - MSI TRX40
projects / kernel.git / log