Our code analyzer reported a uaf.

In snd_emu8000_create_mixer, the callee snd_ctl_add(..,emu->controls[i])
calls snd_ctl_add_replace(.., kcontrol,..). Inside snd_ctl_add_replace(),
if error happens, kcontrol will be freed by snd_ctl_free_one(kcontrol).
Then emu->controls[i] points to a freed memory, and the execution comes
to __error branch of snd_emu8000_create_mixer. The freed emu->controls[i]
is used in snd_ctl_remove(card, emu->controls[i]).

My patch set emu->controls[i] to NULL if snd_ctl_add() failed to avoid
the uaf.

Signed-off-by: Lv Yunlong <lyl2019@mail.ustc.edu.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210426131129.4796-1-lyl2019@mail.ustc.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Remove a duplicate vendor+subvendor pin fixup entry as one is masking
the other and making it unreachable. Consider the more specific newcomer
as a second chance instead.

The generic entry is made less strict to also match for laptops with
slightly different 0x12 pin configuration. Tested on Lenovo Yoga 6 (AMD)
where 0x12 is 0x40000000.

Fixes: 607184cb

 ("ALSA: hda/realtek - Add supported for more Lenovo ALC285 Headset Button")
Signed-off-by: Sami Loone <sami@loone.fi>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/YIXS+GT/dGI/LtK6@yoga
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The recent endpoint management change for implicit feedback mode added
a clearance of ep->sync_sink (formerly ep->sync_slave) pointer at
snd_usb_endpoint_stop() to assure no leftover for the feedback from
the already stopped capture stream.  This turned out to cause a
regression, however, when full-duplex streams were running and only a
capture was stopped.  Because of the above clearance of ep->sync_sink
pointer, no more feedback is done, hence the playback will stall.

This patch fixes the ep->sync_sink clearance to be done only after all
endpoints are released, for addressing the regression.

Reported-and-tested-by: Lucas Endres <jaffa225man@gmail.com>
Fixes: bf6313a0

 ("ALSA: usb-audio: Refactor endpoint management")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210426063349.18601-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Through the examinations and experiments with lots of Roland and BOSS
USB-audio devices, we found out that the recently introduced
full-duplex operations with the implicit feedback mode work fine for
quite a few devices, while the others need only the capture-side quirk
to enforce the full-duplex mode.  The recent commit d86f43b1


("ALSA: usb-audio: Add support for many Roland devices' implicit
feedback quirks") tried to add such quirk entries manually in the
lists, but this turned out to be too many and error-prone, hence it
was reverted again.

This patch is another attempt to cover those missing Roland/BOSS
devices but in a more generic way.  It matches the devices with the
vendor ID 0x0582, and checks whether they are with both ASYNC sync
types or ASYNC is only for capture device.  In the former case, it's
the device with the implicit feedback mode, and applies accordingly.
In both cases, the capture stream requires always the full-duplex
mode, and we apply the known capture quirk for that, too.

Basically the already existing BOSS device quirk entries become
redundant after this generic matching, so those are removed.  Although
the capture_implicit_fb_quirks[] table became empty and superfluous, I
keep it for now, so that people can put a special device easily at any
time later again.

Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212519
Tested-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/20210422120413.457-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This reverts commit d86f43b1 ("ALSA: usb-audio: Add support for
many Roland devices' feedback quirks").

It turned out that many quirk entries there don't contain the proper
EP values and/or the quirk types, which lead to the broken
operations.

As we're going to cover all Roland/BOSS devices in a more generic way
rather the explicit lists, let's revert the previous additions at
first.

Fixes: d86f43b1

 ("ALSA: usb-audio: Add support for many Roland devices' implicit feedback quirks")
Link: https://lore.kernel.org/r/20210422120413.457-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Currently when the call to usb_urb_ep_type_check fails (returning -EINVAL)
the error return path returns -ENOMEM via the exit label "error". Other
uses of the same error exit label set the err variable to -ENOMEM but this
is not being used.  I believe the original intent was for the error exit
path to return the value in err rather than the hard coded -ENOMEM, so
return this rather than the hard coded -ENOMEM.

Addresses-Coverity: ("Unused value")
Fixes: 738d9edc

 ("ALSA: usb-audio: Add sanity checks for invalid EPs")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210420134719.381409-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

On HP EliteBook 845 G8, the audio LEDs can be enabled by
ALC285_FIXUP_HP_MUTE_LED. So use it accordingly.

In addition to that, the mic captures lots of noises, so also limits the
mic boost. The quality of capture audio becomes crystal clear after
limiting the mic boost.

Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210420115530.1349353-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Boot with plugged headset, the Headset Mic will be gone.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/207eecfc3189466a820720bc0c409ea9@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Pioneer devices are supposed to be working with the implicit feedback
mode, but so far the attempt to apply the implicit feedback caused
issues, hence we explicitly skipped the implicit feedback mode for
them.  Recently, Geraldo discovered that the device actually works if
you skip the generic matching of the sync EPs for the capture stream.
That is, we should apply the implicit feedback setup for the playback
like other similar devices, while we need to return 1 from
audioformat_capture_quirk() so that no further matching will be done.

And, later on, Olivia reported later that the fiddling with the
capture quirk alone doesn't suffice for the test with speaker-test
program.  This seems to be a similar case like the recently fixed BOSS
devices.  Indeed, the problem could be addressed by setting
playback_first flag, which indicates that the playback URBs have to be
sent out at first even in the implicit feedback mode.

This patch implements the application of the implicit feedback to
Pioneer devices as described in the above.  The former
skip_pioneer_sync_ep() was dropped, and instead we provide
is_pioneer_implicit_fb() to check the Pioneer devices that need the
implicit feedback.  In the audioformat_implicit_fb_quirk(), simply
apply the implicit fb for playback and set chip->playback_first flag
if matching, and in audioformat_capture_quirk()(), it returns 1 for
skipping the generic EP sync handling.

Reported-by: Geraldo <geraldogabriel@gmail.com>
Tested-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/s5ha6pygqfz.wl-tiwai@suse.de
Link: https://lore.kernel.org/r/20210419153918.450-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add case statement to set sample-rate for the DJM-750 Pioneer
mixer. This was included as part of another patch but I think it has
been archived on Patchwork and hasn't been merged.

Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210418165901.25776-1-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The GA503 has almost exactly the same default setup as the GA401
model with the same issues. The GA401 quirks solve all the issues
so we will use the full quirk chain.

Signed-off-by: Luke D Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210419030411.28304-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>

It makes USB audio capture and playback possible and pristine on my Roland
INTEGRA-7, Boutique D-05, and R-26, along with many more I've encountered
people having had issues with over the last decade or so.

Signed-off-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp
used for ELD notifications), display connection change done during suspend,
can be lost due to following sequence of events:

  1. system in S3 suspend
  2. DP/HDMI receiver connected
  3. system resumed
  4. HDA controller resumed, but card->deferred_resume_work not complete
  5. acomp eld_notify callback
  6. eld_notify ignored as power state is not CTL_POWER_D0
  7. HDA resume deferred work completed, power state set to CTL_POWER_D0

This results in losing the notification, and the jack state reported to
user-space is not correct.

The check on step 6 was added in commit 8ae743e8

 ("ALSA: hda - Skip
ELD notification during system suspend"). It would seem with the deferred
resume logic in ASoC core, this check is not safe.

Fix the issue by modifying the check to use "dev.power.power_state.event"
instead of ALSA specific card power state variable.

BugLink: https://github.com/thesofproject/linux/issues/2825
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210416131157.1881366-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The HP ProBook 445 G7 (17T32ES) uses ALC236. Like ALC236_FIXUP_HP_GPIO_LED,
COEF index 0x34 bit 5 is used to control the playback mute LED, but the
microphone mute LED is controlled using pin VREF instead of a COEF index.

AlsaInfo: https://alsa-project.org/db/?f=0d3f4d1af39cc359f9fea9b550727ee87e5cf45a
Signed-off-by: Jonas Witschel <diabonas@archlinux.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210416105852.52588-1-diabonas@archlinux.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Lenovo Ideapad S740 requires quite a few COEF setups to make its
speakers working.  The verb table was provided from Ryan Prescott as
the result of investigation via qemu:
  https://github.com/ryanprescott/realtek-verb-tools/wiki/How-to-sniff-verbs-from-a-Windows-sound-driver

BugLink: https://github.com/thesofproject/linux/issues/2748
Tested-by: Ryan Prescott <ryan@cousinscomputers.net>
Link: https://lore.kernel.org/r/20210416081211.20059-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Fix two bugs with the Intel HDA Realtek ALC233 sound codec
present in Intel NUC NUC8i7BEH and probably a few other similar
NUC models.

These codecs advertise a 4-level microphone input boost amplifier on
pin 0x19, but the highest two boost settings do not work correctly,
and produce only low analog noise that does not seem to contain any
discernible signal. There is an existing fixup for this exact problem
but for a different PCI subsystem ID, so we re-use that logic.

Changing the boost level also triggers a DC spike in the input signal
that bleeds off over about a second and overwhelms any input during
that time. Thankfully, the existing fixup has the side effect of
making the boost control show up in userspace as a mute/unmute switch,
and this keeps (e.g.) PulseAudio from fiddling with it during normal
input volume adjustments.

Finally, the NUC hardware has built-in inverted stereo mics. This
patch also enables the usual fixup for this so the two channels cancel
noise instead of the actual signal.

[ Re-ordered the quirk entry point by tiwai ]

Signed-off-by: Phil Calvin <phil@philcalvin.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/80dc5663-7734-e7e5-25ef-15b5df24511a@philcalvin.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

We've got a report about the possible race in the user control element
counts (card->user_ctl_count), and it was confirmed that the race
wasn't serious in the old code up to 5.12.  There, the value
modification itself was exclusive and protected via a write semaphore,
hence it's at most concurrent reads and evaluations before the
increment.  Since it's only about the soft-limit to avoid the
exhausting memory usage, one-off isn't a big problem at all.

Meanwhile, the relevant code has been largely modified recently, and
now card->user_ctl_count was replaced with card->user_ctl_alloc_size,
and a few more places were added to access this field.  And, in this
new code, it turned out to be more serious: the modifications are
scattered in various places, and a few of them are without protection.
It implies that it may lead to an inconsistent value by racy
accesses.

For addressing it, this patch extends the range covered by the
card->controls_rwsem write lock at snd_ctl_elem_add() so that the all
code paths that modify and refer to card->user_ctl_alloc_size are
protected by the rwsem properly.

The patch adds also comments in a couple of functions to indicate that
they are under the rwsem lock.

Fixes: 66c6d1ef

 ("ALSA: control: Add memory consumption limit to user controls")
Link: https://lore.kernel.org/r/FEEBF384-44BE-42CF-8FB3-93470933F64F@purdue.edu
Link: https://lore.kernel.org/r/20210415131856.13113-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

During the recent rewrite of the implicit feedback support, we've
tested to apply the implicit fb on BOSS devices, but it failed, as the
capture stream didn't start without the playback.  As the end result,
it got another type of quirk for tying both streams but starts
playback always (commit 6234fdc1

 "ALSA: usb-audio: Quirk for BOSS
GT-001").

Meanwhile, Mike Oliphant has tested the real implicit feedback mode
for the playback again with the latest code, and found out that it
actually works if the initial feedback sync is skipped; that is, on
those BOSS devices, the playback stream has to be started at first
without waiting for the capture URB completions.  Otherwise it gets
stuck.  In the rest operations after the capture stream processed, we
can take them as the implicit feedback source.

This patch is an attempt to improve the support for BOSS devices with
the implicit feedback mode in the way described above.  It adds a new
flag to snd_usb_audio, playback_first, indicating that the playback
stream starts without sync with the initial capture completion.  This
flag is set in the quirk table with the new IMPLICIT_FB_BOTH type.

Reported-and-tested-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

It's a bad idea to allocate big structures on the stack.
Mark the variables as static and add a note for the locking.

Fixes: 22d8de62

 ("ALSA: control - add generic LED trigger module as the new control layer")
Cc: Nathan Chancellor <nathan@kernel.org>
Cc: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210414105858.1937710-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>

In the current code, we have some assumption that the audio clock
selector has been set up implicitly and don't want to touch it unless
it's really needed for the fallback autoclock setup.  This works for
most devices but some seem having a problem.  Partially this was
covered for the devices with a single connector at the initialization
phase (commit 086b957c

 "ALSA: usb-audio: Skip the clock selector
inquiry for single connections"), but also there are cases where the
wrong clock set up is kept silently.  The latter seems to be the cause
of the noises on Behringer devices.

In this patch, we explicitly set up the audio clock selector whenever
the appropriate node is found.

Reported-by: Geraldo Nascimento <geraldogabriel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199327
Link: https://lore.kernel.org/r/CAEsQvcvF7LnO8PxyyCxuRCx=7jNeSCvFAd-+dE0g_rd1rOxxdw@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210413084152.32325-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

module_virtio_driver() makes the code simpler by eliminating
boilerplate code.

Signed-off-by: Chen Huang <chenhuang5@huawei.com>
Reviewed-by: Anton Yakovlev <anton.yakovlev@opensynergy.com>
Link: https://lore.kernel.org/r/20210408125429.1158703-1-chenhuang5@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add description for access parameter in __snd_hda_add_vmaster()
to prevent the compilation warning:
warning: Function parameter or member 'access' not described in '__snd_hda_add_vmaster'

Fixes: e65bf997

 ("ALSA: HDA - remove the custom implementation for the audio LED trigger")
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20210409214616.1539685-1-ranjani.sridharan@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

"group - 1" was intended here instead of "group".  The current error
handling will double free the first item in the array and leak the last
item.

Fixes: cb17fe00

 ("ALSA: control - add sysfs support to the LED trigger module")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/YHBJ4frGxErWB182@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ToneLab EX guitar pedal device requires the same quirk like ToneLab ST
for supporting the MIDI.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212593
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210407144549.1530-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

UA-101 device and co are supported by another driver, snd-ua101, but
the USB audio class driver (snd-usb-audio) catches all and this
resulted in the lack of functionality like missing MIDI devices.

This patch introduces a sort of deny-listing for those devices to just
return -ENODEV at probe in snd-usb-audio driver, so that it falls back
to the probe by snd-ua101.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212477
Link: https://lore.kernel.org/r/20210408075656.30184-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA control interface allows users to add arbitrary control elements
(called "user controls" or "user elements"), and its resource usage is
limited just by the max number of control sets (currently 32).  This
limit, however, is quite loose: each allocation of control set may
have 1028 elements, and each element may have up to 512 bytes (ILP32) or
1024 bytes (LP64) of value data. Moreover, each control set may contain
the enum strings and TLV data, which can be up to 64kB and 128kB,
respectively.  Totally, the whole memory consumption may go over 38MB --
it's quite large, and we'd rather like to reduce the size.

OTOH, there have been other requests even to increase the max number
of user elements; e.g. ALSA firewire stack require the more user
controls, hence we want to raise the bar, too.

For satisfying both requirements, this patch changes the management of
user controls: instead of setting the upper limit of the number of
user controls, we check the actual memory allocation size and set the
upper limit of the total allocation in bytes.  As long as the memory
consumption stays below the limit, more user controls are allowed than
the current limit 32. At the same time, we set the lower limit (8MB)
as default than the current theoretical limit, in order to lower the
risk of DoS.

As a compromise for lowering the default limit, now the actual memory
limit is defined as a module option, 'max_user_ctl_alloc_size', so that
user can increase/decrease the limit if really needed, too.

Link: https://lore.kernel.org/r/s5htur3zl5e.wl-tiwai@suse.de
Co-developed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210408103149.40357-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>

We've got a report about Acer Aspire E1 (PCI SSID 1025:0840) that
loses the speaker output after resume.  With the comparison of COEF
dumps, it was identified that the COEF 0x0d bits 0x6000 corresponds to
the speaker amp.

This patch adds the specific quirk for the device to restore the COEF
bits at the codec (re-)initialization.

BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1183869
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210407095730.12560-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge tag 'asoc-fix-v5.12-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.12

A fairly small batch of driver specific fixes, mainly for various x86
systems with the biggest set being fixes to power down DSPs properly on
x86 SOF systems.

The implicit feedback quirk table contains the entries that also
appear in the capture quirks, and those are all handled to be
skipped.  For the code simplicity, drop the duped entries in the
playback quirk table, and check the match with the capture quirk table
instead.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212519
Link: https://lore.kernel.org/r/20210406113837.32041-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are a few calls of usb_driver_claim_interface() but all of those
miss the proper error checks, as reported by Coverity.  This patch
adds those missing checks.

Along with it, replace the magic pointer with -1 with a constant
USB_AUDIO_IFACE_UNUSED for better readability.

Reported-by: coverity-bot <keescook+coverity-bot@chromium.org>
Addresses-Coverity-ID: 1475943 ("Error handling issues")
Addresses-Coverity-ID: 1475944 ("Error handling issues")
Addresses-Coverity-ID: 1475945 ("Error handling issues")
Fixes: b1ce7ba6 ("ALSA: usb-audio: claim autodetected PCM interfaces all at once")
Fixes: e5779998

 ("ALSA: usb-audio: refactor code")
Link: https://lore.kernel.org/r/202104051059.FB7F3016@keescook
Link: https://lore.kernel.org/r/20210406113534.30455-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add a control to the card before copying the id so that the numid field
is initialized in the copy. Otherwise the numid field of active_id,
format_id, rate_id and channels_id will be the same (0) and
snd_ctl_notify() will not queue the events properly.

Signed-off-by: Jonas Holmberg <jonashg@axis.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210407075428.2666787-1-jonashg@axis.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There is another HP ZBook G5 model with the PCI SSID 103c:844f that
requires the same quirk for controlling the mute LED.  Add the
corresponding entry to the quirk table.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212407
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210401171314.667-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

If count is 16 then this will put the NUL terminator one element beyond
the end of the array.

Fixes: cb17fe00

 ("ALSA: control - add sysfs support to the LED trigger module")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/YGcDOtrimR46vr0k@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When using the driver in I2S TDM mode, the fsl_esai_startup()
function rewrites the number of slots previously set by the
fsl_esai_set_dai_tdm_slot() function to 2.
To fix this, let's use the saved slot count value or, if TDM
is not used and the number of slots is not set, the driver will use
the default value (2), which is set by fsl_esai_probe().

Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>

NPL clock rate is twice the MCLK rate, so set this correctly to
avoid soundwire timeouts.

Fixes: af3d54b9

 ("ASoC: codecs: lpass-rx-macro: add support for lpass rx macro")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210331171235.24824-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

NPL clock rate is twice the MCLK rate, so set this correctly to
avoid soundwire timeouts.

Fixes: c39667dd

 ("ASoC: codecs: lpass-tx-macro: add support for lpass tx macro")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20210331171235.24824-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

Pull mute-LED helper fixes from Jaroslav.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

It may be possible that the string pointer does not move
when parsing. Add a code which detects this state and
simply break the parser loop in this case.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210331180725.663623-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>

As static analysis reported, the prev_lops2 should contain
the previous lops2 pointer in snd_ctl_disconnect_layer().

Link: https://lore.kernel.org/alsa-devel/96e9bd5c-c8db-0db8-b393-fbf4a047dc80@canonical.com/
Fixes: 3f0638a0

 ("ALSA: control - add layer registration routines")
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210331180702.663489-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>