As reported by kbuild test robot, mixer quirks for RME Babyface
Pro used plain integer instead of NULL.
Fixes: 3e8f3bd047 ("ALSA: usb-audio: RME Babyface Pro mixer patch")
Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Reported-by: kbuild test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20200529173248.zzawijfvw73kzjxt@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This commit splits the method to switch fetching mode for protocol
version 2 so that model-dependent operations are explicitly defined.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-15-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit splits the method to get clock source for protocol
version 3 so that model-dependent operations are explicitly defined.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-14-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit splits the method to get clock source for protocol
version 2 so that model-dependent operations are explicitly defined.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-13-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds alternative functions to detect packet format so that
each function corresponds to each model.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-9-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds alternative functions to detect packet format so that
each function corresponds to each model.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-8-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In MOTU protocol, data block consists of SPH and 24-bit chunks
aligned to quadlet. The number of chunks per data block is specific
to model. For models with optical interface, the number differs
depending on I/O settings for the interface (ADAT, TOSLINK).
Currently the number is calculated from flags in model-specific
data. However this is weak in the case that the model has quirks.
Actually, for quirks of some models, flags are used against their
original meanings.
This commit adds model-specific table of chunk count. For future
integration, this table is based on the calculation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The most of members in spec data is used in each protocol file. It's
better to capsulate the data to the file.
This commit moves the data to the file for protocol version 3.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The most of members in spec data is used in each protocol file. It's
better to capsulate the data to the file.
This commit moves the data to the file for protocol version 2.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-2-o-takashi@sakamocchi.jp
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>
snd_hdac_bus_queue_event() and snd_hdac_bus_exec_verb() are used only
internally in HD-audio core. Let's drop the exports and move the
declarations into local.h.
Link: https://lore.kernel.org/r/20200516062854.22141-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unsol event handling code has a loop retrieving the read/write
indices and the arrays without locking while the append to the array
may happen concurrently. This may lead to some inconsistency.
Although there hasn't been any proof of this bad results, it's still
safer to protect the racy accesses.
This patch adds the spinlock protection around the unsol handling loop
for addressing it. Here we take bus->reg_lock as the writer side
snd_hdac_bus_queue_event() is also protected by that lock.
Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For USB sound devices using implicit feedback the endpoint used for
this feedback should be able to be opened twice, once for required
feedback and second time for audio data. This way these devices can be
put in duplex audio mode. Since this only works if the settings of the
endpoint don't change a check is included for this.
This fixes bug 207023 ("MOTU M2 regression on duplex audio") and
should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device
will not operate full-duplex")
Fixes: c249177944 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series")
Signed-off-by: Erwin Burema <e.burema@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751
Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf
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>
This is trivial commit but fulfill missing entries in Kconfig for devices
supported by ALSA firewire-motu driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200511022540.161363-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface UFX was shipped by RME GmbH in 2010, and now discontinued.
Although this model has some enhanced feature which Fireface 802
doesn't have (e.g. on-board USB mass storage device class, configuration
interface with color display), the functionality relevant to
packet communication on IEEE 1394 bus seems to be the same as
Fireface 802 (e.g. available number of channels for PCM frame in
each sampling transfer frequency).
With the assumption, this commit adds support for Fireface UFX. In ALSA
fireface driver, these two models are handled as the same one.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200510074301.116224-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface 802 was shipped by RME GmbH in 2014. This model supports later
protocol for management of isochronous communication and synchronization
of sampling transmission frequency.
This model consists of below ICs:
* TI TSB41AB2
* Xilinx Spartan-6 FPGA XC6SLX16
* TI TMS320 C6747
* SMSC USB3250
Especially, this model just supports IEEE 1394a, against its name which
evokes Fireface 800.
This commit adds support for Fireface 802 (tested). Userspace applications
can transfer PCM frames and MIDI messages via ALSA PCM/Rawmidi interface.
I note that 4 channels for ADAt1 and ADAT2 are disabled at higher sampling
transfer frequency since isochronous resources reservation fails due to
bandwidth limitation of IEEE 1394a.
The value read from LATTER_SYNC_STATUS register is slightly different
from the one of Fireface UCX. The higher 4 bits and lower 4 bits are
swapped within the same byte.
Without any assist of userspace application, transmitted MIDI messages
from the device are not going to be processed. For detail, please refer
to my comment in code of latter protocol.
$ python crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 0404ffff bus_info_length 4, crc_length 4, crc 65535 (should be 26805)
404 31333934 bus_name "1394"
408 20008000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 8 (512)
40c 000a3504 company_id 000a35 |
410 38077423 device_id 0438077423 | EUI-64 000a350438077423
root directory
-----------------------------------------------------------------
414 0005ffff directory_length 5, crc 65535 (should be 9514)
418 0c0083c0 node capabilities per IEEE 1394
41c 03000a35 vendor
420 8100000b --> descriptor leaf at 44c
424 8d000007 --> eui-64 leaf at 440
428 d1000001 --> unit directory at 42c
unit directory at 42c
-----------------------------------------------------------------
42c 0004ffff directory_length 4, crc 65535 (should be 45134)
430 12000a35 specifier id
434 13000005 version
438 17101800 model
43c 81000008 --> descriptor leaf at 45c
eui-64 leaf at 440
-----------------------------------------------------------------
440 0002ffff leaf_length 2, crc 65535 (should be 60131)
444 000a3504 company_id 000a35 |
448 38077423 device_id 0438077423 | EUI-64 000a350438077423
descriptor leaf at 44c
-----------------------------------------------------------------
44c 0003ffff leaf_length 3, crc 65535 (should be 469)
450 00000000 textual descriptor
454 00000000 minimal ASCII
458 524d4521 "RME!"
descriptor leaf at 45c
-----------------------------------------------------------------
45c 0005ffff leaf_length 5, crc 65535 (should be 10561)
460 00000000 textual descriptor
464 00000000 minimal ASCII
468 46697265 "Fire"
46c 66616365 "face"
470 20383032 " 802"
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200510074301.116224-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In RME fireface series, version field of unit directory in configuration
ROM is used to distinguish each model. The value of field is known and
it's better to use enumeration constants for code representation.
This commit adds enumeration constants for model identification.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200510074301.116224-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the latter models of RME Fireface series, device start to transfer
packets several dozens of milliseconds. On the other hand, ALSA fireface
driver starts IR context 2 milliseconds after the start. This results
in loss to handle incoming packets on the context.
This commit changes to start IR context immediately instead of
postponement. For Fireface 800, this affects nothing because the device
transfer packets 100 milliseconds or so after the start and this is
within wait timeout.
Cc: <stable@vger.kernel.org>
Fixes: acfedcbe1c ("ALSA: firewire-lib: postpone to start IR context")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200510074301.116224-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
128000 and 192000 are congruence modulo 32000, thus it's wrong to
distinguish them as multiple of 32000 and 48000 by modulo 32000 at
first.
Additionally, used condition statement to detect quadruple speed can
cause missing bit flag.
Furthermore, counter to ensure the configuration is wrong and it
causes false positive.
This commit fixes the above three bugs.
Cc: <stable@vger.kernel.org>
Fixes: 60aec494b3 ("ALSA: fireface: support allocate_resources operation in latter protocol")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200510074301.116224-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, the sequence of syt offset and the number of data
blocks per packet is calculated for pool in AMDTP domain structure in
advance of processing outgoing packets.
This commit uses the sequence for outgoing packet processing to obsolete
per-stream processing of the sequence.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-11-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, sequence of syt offset and the number of data
blocks is generated when packets for outgoing stream are going to be
queued.
This commit generates and pools the sequence independently of the
processing of outgoing packets for future extension.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-10-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For future extension, storage is required to store packet sequence in
incoming AMDTP stream to recover media clock for outgoing AMDTP stream.
This commit adds the storage to AMDTP domain for this purpose. The
packet sequence is represented by 'struct seq_desc' which has two
members; syt_offset and the number of data blocks. The size of storage
is decided according to the size of packet queue.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-9-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When calculating the number of data blocks per packet, some states are
stored in AMDTP stream structure. This is inconvenient when reuse the
calculation from non-stream structure.
This commit applies refactoring to helper function for the calculation
so that the function doesn't touch AMDTP stream structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-8-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When calculating syt offset, some states are stored in AMDTP stream
structure. This is inconvenient when reuse the calculation from
non-stream structure.
This commit applies refactoring to helper function for the calculation
so that the function doesn't touch AMDTP stream structure.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
