| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
commit 0c5086f5699906ec8e31ea6509239489f060f2dc upstream.
The HP Thunderbolt Dock has two separate USB devices, one is for speaker
and one is for headset. Add names for them so userspace can apply UCM
settings.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200608062630.10806-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 862b2509d157c629dd26d7ac6c6cdbf043d332eb upstream.
When a USB-audio interface gets runtime-suspended via auto-pm feature,
the driver suspends all functionality and increment
chip->num_suspended_intf. Later on, when the system gets suspended to
S3, the driver increments chip->num_suspended_intf again, skips the
device changes, and sets the card power state to
SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from
S3, the resume callback decrements chip->num_suspended_intf. Since
this refcount is still not zero (it's been runtime-suspended), the
whole resume is skipped. But there is a small pitfall here.
The problem is that the driver doesn't restore the card power state
after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So,
even after the system resume finishes, the card instance still appears
as if it were system-suspended, and this confuses many ioctl accesses
that are blocked unexpectedly.
In details, we have two issues behind the scene: one is that the card
power state is changed only when the refcount becomes zero, and
another is that the prior auto-suspend check is kept in a boolean
flag. Although the latter problem is almost negligible since the
auto-pm feature is imposed only on the primary interface, but this can
be a potential problem on the devices with multiple interfaces.
This patch addresses those issues by the following:
- Replace chip->autosuspended boolean flag with chip->system_suspend
counter
- At the first system-suspend, chip->num_suspended_intf is recorded to
chip->system_suspend
- At system-resume, the card power state is restored when the
chip->num_suspended_intf refcount reaches to chip->system_suspend,
i.e. the state returns to the auto-suspended
Also, the patch fixes yet another hidden problem by the code
refactoring along with the fixes above: namely, when some resume
procedure failed, the driver left chip->num_suspended_intf that was
already decreased, and it might lead to the refcount unbalance.
In the new code, the refcount decrement is done after the whole resume
procedure, and the problem is avoided as well.
Fixes: 0662292aec05 ("ALSA: usb-audio: Handle normal and auto-suspend equally")
Reported-and-tested-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200603153709.6293-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit e18035cf5cb3d2bf8e4f4d350a23608bd208b934 upstream.
Add and use snd_pcm_stream_lock_nested() in snd_pcm_link/unlink
implementation. The code is fine, but generates a lockdep complaint:
============================================
WARNING: possible recursive locking detected
5.7.1mq+ #381 Tainted: G O
--------------------------------------------
pulseaudio/4180 is trying to acquire lock:
ffff888402d6f508 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xda8/0xee0 [snd_pcm]
but task is already holding lock:
ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
other info that might help us debug this:
Possible unsafe locking scenario:
CPU0
----
lock(&group->lock);
lock(&group->lock);
*** DEADLOCK ***
May be due to missing lock nesting notation
2 locks held by pulseaudio/4180:
#0: ffffffffa1a05190 (snd_pcm_link_rwsem){++++}-{3:3}, at: snd_pcm_common_ioctl+0xca0/0xee0 [snd_pcm]
#1: ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
[...]
Cc: stable@vger.kernel.org
Fixes: f57f3df03a8e ("ALSA: pcm: More fine-grained PCM link locking")
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Link: https://lore.kernel.org/r/37252c65941e58473b1219ca9fab03d48f47e3e3.1591610330.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
commit 951e2736f4b11b58dc44d41964fa17c3527d882a upstream.
Prevent SNDRV_PCM_IOCTL_LINK linking stream to itself - the code
can't handle it. Fixed commit is not where bug was introduced, but
changes the context significantly.
Cc: stable@vger.kernel.org
Fixes: 0888c321de70 ("pcm_native: switch to fdget()/fdput()")
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Link: https://lore.kernel.org/r/89c4a2487609a0ed6af3ecf01cc972bdc59a7a2d.1591634956.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 573fcbfd319ccef26caa3700320242accea7fd5c upstream.
A couple of Lenovo ThinkCentre machines all have 2 front mics and they
use the same codec alc623 and have the same pin config, so add a
pintbl entry for those machines to apply the fixup
ALC283_FIXUP_HEADSET_MIC.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200608115541.9531-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit f4588cc425beb62e355bc2a5de5d5c83e26a74ca upstream.
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: acfedcbe1ce4 ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit bbd6aac3ae15bef762af03bf62e35ace5c4292bd upstream.
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: 60aec494b389 ("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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 27a7c67012cfa6d79f87fbb51afa13c6c0e24e34 upstream.
dp/hdmi ati hda is not shown in audio settings
[ rearranged to a more appropriate place per device number order
-- tiwai ]
Signed-off-by: Hersen Wu <hersenxs.wu@amd.com>
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200603013137.1849404-1-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit d9b8fbf15d05350b36081eddafcf7b15aa1add50 upstream.
snd_es968_pnp_detect() misses a snd_card_free() in a failed path.
Add the missed function call to fix it.
Fixes: a20971b201ac ("ALSA: Merge es1688 and es968 drivers")
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200603092459.1424093-1-hslester96@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 8ba4dc3cff8cbe2c571063a5fd7116e8bde563ca upstream.
The xmax values for Master Playback Volume and Mic Boost
Capture Volume are specified incorrectly (one greater)
which results in the wrong dB gain being shown to the user
in the case of Master Playback Volume.
Signed-off-by: Pavel Dobias <dobias@2n.cz>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200515120757.24669-1-dobias@2n.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit be8499c48f115b912f5747c420f66a5e2c31defe upstream.
Fix the mic gain registers for channels 2-4.
The incorret register was being set as it was touching the CH1 config
registers.
Fixes: 37bde5acf040 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200427203608.7031-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Only a few last-minute small fixes: the change in ALSA core hwdep is
about the undefined behavior of bit shift, which is almost harmless
but still worth to pick up quickly.
The rest are all device-specific fixes for HD-audio and USB-audio, and
safe to apply at the late stage"
* tag 'sound-5.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Add new codec supported for ALC287
ALSA: usb-audio: Quirks for Gigabyte TRX40 Aorus Master onboard audio
ALSA: usb-audio: mixer: volume quirk for ESS Technology Asus USB DAC
ALSA: hda/realtek - Add a model for Thinkpad T570 without DAC workaround
ALSA: hwdep: fix a left shifting 1 by 31 UB bug
|
|
Enable new codec supported for ALC287.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/dcf5ce5507104d0589a917cbb71dc3c6@realtek.com
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 54947cd64c1b ("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: 54947cd64c1b ("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>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Just a few small fixes: the only significant one is a slight
improvement for PCM running position update with no-period-elapsed
case while the rest are HD-audio fixups and ice1712 model quirk"
* tag 'sound-5.7-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Add more fixup entries for Clevo machines
ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio option
ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Xtreme
ALSA: pcm: fix incorrect hw_base increase
|
|
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
e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII") we
enabled the ADCIII ports using the model=staudio option but forgot
this part to ensure the STDSP24 card is initialized properly.
Fixes: e8a91ae18bdc ("ALSA: ice1712: Add support for STAudio ADCIII")
Signed-off-by: Scott Bahling <sbahling@suse.com>
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1048934
Link: https://lore.kernel.org/r/20200518175728.28766-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec
requires a similar workaround for Clevo laptops to enforce the
DAC/mixer connection path. Set up a quirk entry for that.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275
Signed-off-by: Christian Lachner <gladiac@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200518053844.42743-2-gladiac@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
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>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Things look good and calming down; the only change to ALSA core is the
fix for racy rawmidi buffer accesses spotted by syzkaller, and the
rest are all small device-specific quirks for HD-audio and USB-audio
devices"
* tag 'sound-5.7-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Limit int mic boost for Thinkpad T530
ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DA
ALSA: hda/realtek: Enable headset mic of ASUS UX581LV with ALC295
ALSA: hda/realtek - Enable headset mic of ASUS UX550GE with ALC295
ALSA: hda/realtek - Enable headset mic of ASUS GL503VM with ALC295
ALSA: hda/realtek: Add quirk for Samsung Notebook
ALSA: rawmidi: Fix racy buffer resize under concurrent accesses
ALSA: usb-audio: add mapping for ASRock TRX40 Creator
ALSA: hda/realtek - Fix S3 pop noise on Dell Wyse
Revert "ALSA: hda/realtek: Fix pop noise on ALC225"
ALSA: firewire-lib: fix 'function sizeof not defined' error of tracepoints format
ALSA: usb-audio: Add control message quirk delay for Kingston HyperX headset
|
|
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>
|
|
Some models of the Samsung Notebook 9 have very quiet and distorted
headphone output. This quirk changes the VREF value of the ALC298
codec NID 0x1a from default HIZ to new 100.
[ adjusted to 5.7-base and rearranged in SSID order -- tiwai ]
Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Link: https://lore.kernel.org/r/20200510032838.1989130-1-pozulp.kernel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
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 317d9313925c ("ALSA: hda/realtek - Set default power save node to
0") makes the ALC225 have pop noise on S3 resume and cold boot.
The previous fix enable power save node universally for ALC225, however
it makes some ALC225 systems unable to produce any sound.
So let's only enable power save node for the affected Dell Wyse
platform.
Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0")
BugLink: https://bugs.launchpad.net/bugs/1866357
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200503152449.22761-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This reverts commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757.
Enable power save node breaks some systems with ACL225. Revert the patch
and use a platform specific quirk for the original issue isntead.
Fixes: 3b36b13d5e69 ("ALSA: hda/realtek: Fix pop noise on ALC225")
BugLink: https://bugs.launchpad.net/bugs/1875916
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200503152449.22761-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
format
The snd-firewire-lib.ko has 'amdtp-packet' event of tracepoints. Current
printk format for the event includes 'sizeof(u8)' macro expected to be
extended in compilation time. However, this is not done. As a result,
perf tools cannot parse the event for printing:
$ mount -l -t debugfs
debugfs on /sys/kernel/debug type debugfs (rw,nosuid,nodev,noexec,relatime)
$ cat /sys/kernel/debug/tracing/events/snd_firewire_lib/amdtp_packet/format
...
print fmt: "%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u %s",
REC->second, REC->cycle, REC->src, REC->dest, REC->channel,
REC->payload_quadlets, REC->data_blocks, REC->data_block_counter,
REC->packet_index, REC->irq, REC->index,
__print_array(__get_dynamic_array(cip_header),
__get_dynamic_array_len(cip_header),
sizeof(u8))
$ sudo perf record -e snd_firewire_lib:amdtp_packet
[snd_firewire_lib:amdtp_packet] function sizeof not defined
Error: expected type 5 but read 0
This commit fixes it by obsoleting the macro with actual size.
Cc: <stable@vger.kernel.org>
Fixes: bde2bbdb307a ("ALSA: firewire-lib: use dynamic array for CIP header of tracing events")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200503045718.86337-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Just a collection of small fixes around this time:
- One more try for fixing PCM OSS regression
- HD-audio: a new quirk for Lenovo, the improved driver blacklisting,
a lock fix in the minor error path, and a fix for the possible race
at monitor notifiaction
- USB-audio: a quirk ID fix, a fix for POD HD500 workaround"
* tag 'sound-5.7-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Correct a typo of NuPrime DAC-10 USB ID
ALSA: opti9xx: shut up gcc-10 range warning
ALSA: hda/hdmi: fix without unlocked before return
ALSA: hda/hdmi: fix race in monitor detection during probe
ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter
ALSA: line6: Fix POD HD500 audio playback
ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)
ALSA: pcm: oss: Place the plugin buffer overflow checks correctly
ALSA: hda: Match both PCI ID and SSID for driver blacklist
|
|
Kingston HyperX headset with 0951:16ad also needs the same quirk for
delaying the frequency controls.
Signed-off-by: Jesus Ramos <jesus-ramos@live.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/BY5PR19MB3634BA68C7CCA23D8DF428E796AF0@BY5PR19MB3634.namprd19.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The USB vendor ID of NuPrime DAC-10 is not 16b0 but 16d0.
Fixes: f656891c6619 ("ALSA: usb-audio: add more quirks for DSD interfaces")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200430124755.15940-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
gcc-10 points out a few instances of suspicious integer arithmetic
leading to value truncation:
sound/isa/opti9xx/opti92x-ad1848.c: In function 'snd_opti9xx_configure':
sound/isa/opti9xx/opti92x-ad1848.c:322:43: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_opti9xx_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow]
322 | (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/opti92x-ad1848.c:351:3: note: in expansion of macro 'snd_opti9xx_write_mask'
351 | snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
| ^~~~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/miro.c: In function 'snd_miro_configure':
sound/isa/opti9xx/miro.c:873:40: error: overflow in conversion from 'int' to 'unsigned char' changes value from '(int)snd_miro_read(chip, 3) & -256 | 240' to '240' [-Werror=overflow]
873 | (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask)))
| ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~
sound/isa/opti9xx/miro.c:1010:3: note: in expansion of macro 'snd_miro_write_mask'
1010 | snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
| ^~~~~~~~~~~~~~~~~~~
These are all harmless here as only the low 8 bit are passed down
anyway. Change the macros to inline functions to make the code
more readable and also avoid the warning.
Strictly speaking those functions also need locking to make the
read/write pair atomic, but it seems unlikely that anyone would
still run into that issue.
Fixes: 1841f613fd2e ("[ALSA] Add snd-miro driver")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20200429190216.85919-1-arnd@arndb.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Fix the following coccicheck warning:
sound/pci/hda/patch_hdmi.c:1852:2-8: preceding lock on line 1846
After add sanity check to pass klockwork check,
The spdif_mutex should be unlock before return true
in check_non_pcm_per_cvt().
Fixes: 960a581e22d9 ("ALSA: hda: fix some klockwork scan warnings")
Signed-off-by: Wu Bo <wubo40@huawei.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587907042-694161-1-git-send-email-wubo40@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
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 4285de0725b1) to 5.7-base, hence neither Fixes nor
Cc-to-stable tags are included here -- tiwai ]
The checks of the plugin buffer overflow in the previous fix by commit
f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow")
are put in the wrong places mistakenly, which leads to the expected
(repeated) sound when the rate plugin is involved. Fix in the right
places.
Also, at those right places, the zero check is needed for the
termination node, so added there as well, and let's get it done,
finally.
Link: https://lore.kernel.org/r/20200424193843.20397-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This became a slightly big pull request, as the accumulated ASoC fixes
are included here. Some highlights:
- Revert of ASoC DAI startup changes that caused regression on some
x86 platforms
- Regression fix in HD-audio power management and driver blacklist
- A collection of ASoC DAPM and topology fixes
- Continued USB-audio fixes and quirks
- Lots of small device-specific fixes
- Rockchip S/PDIF DT stuff update for validation issues"
* tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (51 commits)
ALSA: hda: Always use jackpoll helper for jack update after resume
ALSA: hda/realtek - Add new codec supported for ALC245
ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
ALSA: usb-audio: Add connector notifier delegation
ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen
ASoC: wm8960: Fix wrong clock after suspend & resume
ALSA: usx2y: Fix potential NULL dereference
ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2
ASoC: wm89xx: Add missing dependency
ASoC: dapm: fixup dapm kcontrol widget
ASoC: rsnd: Fix "status check failed" spam for multi-SSI
ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent
ASoC: meson: gx-card: fix codec-to-codec link setup
ASoC: meson: axg-card: fix codec-to-codec link setup
ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
ALSA: hda: Remove ASUS ROG Zenith from the blacklist
ALSA: hda/realtek - Fix unexpected init_amp override
ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
ASoC: stm32: sai: fix sai probe
...
|
|
The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.
Since the empty codec problem appear on the certain AMD platform (PCI
ID 1022:1487), this patch changes the blacklist matching to both PCI
ID and SSID using pci_match_id(). Also, the entry that was removed by
the previous fix for ASUS ROG Zenigh II is re-added.
Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
HD-audio codec driver applies a tricky procedure to forcibly perform
the runtime resume by mimicking the usage count even if the device has
been runtime-suspended beforehand. This was needed to assure to
trigger the jack detection update after the system resume.
And recently we also applied the similar logic to the HD-audio
controller side. However this seems leading to some inconsistency,
and eventually PCI controller gets screwed up.
This patch is an attempt to fix and clean up those behavior: instead
of the tricky runtime resume procedure, the existing jackpoll work is
scheduled when such a forced codec resume is required. The jackpoll
work will power up the codec, and this alone should suffice for the
jack status update in usual cases. If the extra polling is requested
(by checking codec->jackpoll_interval), the manual update is invoked
after that, and the codec is powered down again.
Also, we filter the spurious wake up of the codec from the controller
runtime resume by checking codec->relaxed_resume flag. If this flag
is set, basically we don't need to wake up explicitly, but it's
supposed to be done via the audio component notifier.
Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed")
Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Enable new codec supported for ALC245.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/8c0804738b2c42439f59c39c8437817f@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which
increases the refcount of the snd_usb_audio object "chip".
When snd_microii_spdif_default_get() returns, local variable "chip"
becomes invalid, so the refcount should be decreased to keep refcount
balanced.
The reference counting issue happens in several exception handling paths
of snd_microii_spdif_default_get(). When those error scenarios occur
such as usb_ifnum_to_if() returns NULL, the function forgets to decrease
the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak.
Fix this issue by jumping to "end" label when those error scenarios
occur.
Fixes: 447d6275f0c2 ("ALSA: usb-audio: Add sanity checks for endpoint accesses")
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
It turned out that ALC1220-VB USB-audio device gives the interrupt
event to some PCM terminals while those don't allow the connector
state request but only the actual I/O terminals return the request.
The recent commit 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack
controls for PCM terminals") excluded those phantom terminals, so
those events are ignored, too.
My first thought was that this could be easily deduced from the
associated terminals, but some of them have even no associate terminal
ID, hence it's not too trivial to figure out.
Since the number of such terminals are small and limited, this patch
implements another quirk table for the simple mapping of the
connectors. It's not really scalable, but let's hope that there will
be not many such funky devices in future.
Fixes: 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
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.
|
