ALSA: hda/hdmi: Fix keep_power assignment for non-component devices

It's been reported that, when neither nouveau nor Nvidia graphics
driver is used, the screen starts flickering.  And, after comparing
between the working case (stable 4.4.x) and the broken case, it turned
out that the problem comes from the audio component binding.  The
Nvidia and AMD audio binding code clears the bus->keep_power flag
whenever snd_hdac_acomp_init() succeeds.  But this doesn't mean that
the component is actually bound, but it merely indicates that it's
ready for binding.  So, when both nouveau and Nvidia are blacklisted
or not ready, the driver keeps running without the audio component but
also with bus->keep_power = false.  This made the driver runtime PM
kicked in and powering down when unused, which results in flickering
in the graphics side, as it seems.

For fixing the bug, this patch moves the bus->keep_power flag change
into generic_acomp_notifier_set() that is the function called from the
master_bind callback of component ops; i.e. it's guaranteed that the
binding succeeded.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208609
Fixes: 5a858e79c911 ("ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200728082033.23933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Workaround for spurious wakeups on some Intel platforms

We've received a regression report on Intel HD-audio controller that
wakes up immediately after S3 suspend.  The bisection leads to the
commit c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not
needed").  This commit replaces the system-suspend to use
pm_runtime_force_suspend() instead of the direct call of
__azx_runtime_suspend().  However, by some really mysterious reason,
pm_runtime_force_suspend() causes a spurious wakeup (although it calls
the same __azx_runtime_suspend() internally).

As an ugly workaround for now, revert the behavior to call
__azx_runtime_suspend() and __azx_runtime_resume() for those old Intel
platforms that may exhibit such a problem, while keeping the new
standard pm_runtime_force_suspend() and pm_runtime_force_resume()
pair for the remaining chips.

Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200727164443.4233-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek: Fix add a "ultra_low_power" function for intel reference board (alc256)

Intel requires to enable power saving mode for intel reference board (alc256)

Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200727115647.10967-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek: typo_fix: enable headset mic of ASUS ROG Zephyrus G14(GA401) series with ALC289

This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401.

Fixes: ff53664daff2 ("ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289")
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140837.302763-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G15(GA502) series with ALC289

This patch adds support for headset mic to the ASUS ROG Zephyrus
G15(GA502) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.

Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140616.298892-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Add implicit feedback quirk for SSL2

As expected, this requires the same quirk as the SSL2+ in order for the
clock to sync. This was suggested by, and tested on an SSL2, by Dmitry.

Suggested-by: Dmitry <dpavlushko@gmail.com>
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200621075005.52mjjfc6dtdjnr3h@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S

Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit
14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook).
This laptop uses the same ALC298 but different subsystem id 0x144dc812.
I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c

Signed-off-by: Joonho Wohn <doomsheart@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAHcbMh291aWDKiWSZoxXB4-Eru6OYRwGA4AVEdCZeYmVLo5ZxQ@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge tag 'asoc-fix-v5.8-rc5' of https://git./linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.8

An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them.  For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.

ALSA: info: Drop WARN_ON() from buffer NULL sanity check

snd_info_get_line() has a sanity check of NULL buffer -- both buffer
itself being NULL and buffer->buffer being NULL.  Basically both
checks are valid and necessary, but the problem is that it's with
snd_BUG_ON() macro that triggers WARN_ON().  The latter condition
(NULL buffer->buffer) can be met arbitrarily by user since the buffer
is allocated at the first write, so it means that user can trigger
WARN_ON() at will.

This patch addresses it by simply moving buffer->buffer NULL check out
of snd_BUG_ON() so that spurious WARNING is no longer triggered.

Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: rt5682: Report the button event in the headset type only

The irq work will be manipulated by resume function, and it will report
the wrong jack type while the jack type is headphone in the button event.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200716030123.27122-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: bytcht_es8316: Add missed put_device()

snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.

Fixes: ba49cf6f8e4a ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda/realtek - fixup for yet another Intel reference board

Add headset_jack for the intel reference board support with
10ec:1230.

Signed-off-by: PeiSen Hou <pshou@realtek.com.tw>
Link: https://lore.kernel.org/r/20200716090134.9811-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Enable Speaker for ASUS UX563

ASUS UX563 speaker can't output.
Add quirk to link suitable model will enable it.
This model also could enable headset Mic.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/96dee3ab01a04c28a7b44061e88009dd@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Enable Speaker for ASUS UX533 and UX534

ASUS UX533 and UX534 speaker still can't output.
End User feedback speaker didn't have output.
Add this COEF value will enable it.

Fixes: 4e051106730d ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/80334402a93b48e385f8f4841b59ae09@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: rt5682: Enable Vref2 under using PLL2

Enable Vref2 under long term using PLL2 to avoid clock unstable.

Signed-off-by: derek.fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/1594721600-29994-1-git-send-email-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256

The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256
cannot detect the headset microphone until
ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset
mic pin.

Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200713060421.62435-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289

This patch adds support for headset mic to the ASUS ROG Zephyrus
G14(GA401) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.

Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200711110557.18681-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - change to suitable link model for ASUS platform

ASUS platform couldn't need to use Headset Mode model.
It changes to the suitable model.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/d05bcff170784ec7bb35023407148161@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Fix race against the error recovery URB submission

USB MIDI driver has an error recovery mechanism to resubmit the URB in
the delayed timer handler, and this may race with the standard start /
stop operations.  Although both start and stop operations themselves
don't race with each other due to the umidi->mutex protection, but
this isn't applied to the timer handler.

For fixing this potential race, the following changes are applied:

- Since the timer handler can't use the mutex, we apply the
  umidi->disc_lock protection at each input stream URB submission;
  this also needs to change the GFP flag to GFP_ATOMIC
- Add a check of the URB refcount and skip if already submitted
- Move the timer cancel call at disconnection to the beginning of the
  procedure; this assures the in-flight timer handler is gone properly
  before killing all pending URBs

Reported-by: syzbot+0f4ecfe6a2c322c81728@syzkaller.appspotmail.com
Reported-by: syzbot+5f1d24c49c1d2c427497@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200710160656.16819-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: line6: Sync the pending work cancel at disconnection

Recently syzkaller reported a UAF in LINE6 driver, and it's likely
because we call cancel_delayed_work() at the disconnect callback
instead of cancel_delayed_work_sync().  Let's use the correct one
instead.

Reported-by: syzbot+145012a46658ac00fc9e@syzkaller.appspotmail.com
Suggested-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hlfjr4gio.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: line6: Perform sanity check for each URB creation

LINE6 drivers create stream URBs with a fixed pipe without checking
its validity, and this may lead to a kernel WARNING at the submission
when a malformed USB descriptor is passed.

For avoiding the kernel warning, perform the similar sanity checks for
each pipe type at creating a URB.

Reported-by: syzbot+c190f6858a04ea7fbc52@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hv9iv4hq8.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: rt286: fix unexpected interrupt happens

The HV/VREF should not turn off if the headphone jack plug-in.
This patch could solve the unexpected interrupt issue in some devices.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200709101345.11449-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: wm8974: remove unsupported clock mode

In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie.
inverted frame clock is not support in this mode).

Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657056-4989-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: wm8974: fix Boost Mixer Aux Switch

Clear BIT6 of INPPGA means not muted (Switch On).

Signed-off-by: Puyou Lu <puyou.lu@gmail.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1593657025-4903-1-git-send-email-puyou.lu@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge series "ASoC: more fixes for dpcm checks" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

This is hopefully the last set of fixes to avoid probe errors due to
stricter checks of DAI capabilities introduced late in the 5.8 cycle.

Daniel Baluta (1):
  ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M

Pierre-Louis Bossart (2):
  ASoC: soc-dai: set dai_link dpcm_ flags with a helper
  ASoC: Intel: bdw-rt5677: fix non BE conversion

 include/sound/soc-dai.h              |  1 +
 sound/soc/generic/audio-graph-card.c |  4 +--
 sound/soc/generic/simple-card.c      |  4 +--
 sound/soc/intel/boards/bdw-rt5677.c  |  1 +
 sound/soc/soc-dai.c                  | 38 ++++++++++++++++++++++++++++
 sound/soc/sof/imx/imx8.c             |  8 ++++++
 sound/soc/sof/imx/imx8m.c            |  8 ++++++
 7 files changed, 60 insertions(+), 4 deletions(-)

base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1

Merge series "ASoC: topology: fix error handling flow" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...

Pierre-Louis Bossart (5):
  ASoC: topology: fix kernel oops on route addition error
  ASoC: topology: fix tlvs in error handling for widget_dmixer
  ASoC: topology: use break on errors, not continue
  ASoC: topology: factor kfree(se) in error handling
  ASoC: topology: add more logs when topology load fails.

 sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++----------------
 1 file changed, 58 insertions(+), 39 deletions(-)

base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1

Merge series "ASoC: codecs: add MAX98373 Soundwire driver" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

V2 with a number of cleanups:
split between I2C and SoundWire modes, as done for rt5682, and updated
Kconfigs.
removed useless initializations common to both modes
removed idle_bias on
fixed register classified as volatile in error
fixed SPDX comments

Pierre-Louis Bossart (2):
  ASoC: codecs: max98373: split I2C and common parts
  ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies

Ryan Lee (2):
  ASoC: codecs: max98373: Removed superfluous volume control from chip
    default
  ASoC: codecs: max98373: add SoundWire support

randerwang (2):
  ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
  ASoC: Intel: sdw_max98373: add card_late_probe support

 sound/soc/codecs/Kconfig                  |  20 +-
 sound/soc/codecs/Makefile                 |   4 +
 sound/soc/codecs/max98373-i2c.c           | 612 +++++++++++++++
 sound/soc/codecs/max98373-sdw.c           | 887 ++++++++++++++++++++++
 sound/soc/codecs/max98373-sdw.h           |  72 ++
 sound/soc/codecs/max98373.c               | 619 +--------------
 sound/soc/codecs/max98373.h               |  17 +-
 sound/soc/intel/boards/Kconfig            |   7 +-
 sound/soc/intel/boards/sof_sdw.c          |  19 +-
 sound/soc/intel/boards/sof_sdw_common.h   |   6 +
 sound/soc/intel/boards/sof_sdw_max98373.c |  12 +
 11 files changed, 1668 insertions(+), 607 deletions(-)
 create mode 100644 sound/soc/codecs/max98373-i2c.c
 create mode 100644 sound/soc/codecs/max98373-sdw.c
 create mode 100644 sound/soc/codecs/max98373-sdw.h

base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759
--
2.25.1

ASoC: SOF: core: fix null-ptr-deref bug during device removal

The DSP should be notified for device removal only if the
probe was successful. Fixes the following KASAN bug:
BUG: KASAN: null-ptr-deref in sof_ipc_tx_message+0x80/0x160 [snd_sof]

Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707204027.114169-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend

Idle_bias_on is used to decide bias on/off in standby state by dapm.
When Idle_bias_on is set to one, dapm will keep max98373 active at
idle time. Max98373 is doing nothing in this state, so remove
idle_bias_on setting to let max98373 get suspended when it is idle.

Signed-off-by: randerwang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ryan Lee <ryans.lee@maximintegrated.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: codecs: max98373: Removed superfluous volume control from chip default

Volume control in probe function is not necessary.

Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707205740.114927-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: fix tlvs in error handling for widget_dmixer

we need to free all allocated tlvs, not just the one allocated in
the loop before releasing kcontrols - other the tlvs references will
leak.

Fixes: 9f90af3a995298 ('ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow')
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/20200707203749.113883-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: fix kernel oops on route addition error

When errors happens while loading graph components, the kernel oopses
while trying to remove all topology components. This can be
root-caused to a list pointing to memory that was already freed on
error.

remove_route() is already called on errors and will perform the
required cleanups so there's no need to free the route memory in
soc_tplg_dapm_graph_elems_load() if the route was added to the
list. We do however want to free the routes allocated but not added to
the list.

Fixes: 7df04ea7a31ea ('ASoC: topology: modify dapm route loading routine and add dapm route unloading')
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/20200707203749.113883-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M

This is identical with change for Intel platforms done with
commit 8c05246c0b58 ("ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell")
and fixes a regression on i.MX8/i.MX8M:

[   25.705750]  esai-Codec: ASoC: no backend playback stream
[   27.923378]  esai-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.

Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: bdw-rt5677: fix non BE conversion

When SOF is used, the normal links are converted into DPCM ones. This
generates an error

[ 58.276668] bdw-rt5677 bdw-rt5677: CPU DAI spi-RT5677AA:00 for rtd
Wake on Voice does not support playback
[ 58.276676] bdw-rt5677 bdw-rt5677: ASoC: can't create pcm Wake on
Voice :-22

Fix by forcing the capture direction.

Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Curtis Malainey <curtis@malainey.com>
Link: https://lore.kernel.org/r/20200707210439.115300-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-dai: set dai_link dpcm_ flags with a helper

Add a helper to walk through all the DAIs and set dpcm_playback and
dpcm_capture flags based on the DAIs capabilities, and use this helper
to avoid setting these flags arbitrarily in generic cards.

The commit referenced in the Fixes tag did not introduce the
configuration issue but will prevent the card from probing when
detecting invalid configurations.

Fixes: b73287f0b0745 ('ASoC: soc-pcm: dpcm: fix playback/capture checks')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20200707210439.115300-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl

Add Shengjiu who's actively working on the latest fsl/nxp audio drivers.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://lore.kernel.org/r/20200707045829.10002-1-nicoleotsuka@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: core: Remove only the registered component in devm functions

The ASoC devm_ functions that register a component
(devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will
clean their component by running snd_soc_unregister_component.

snd_soc_unregister_component will then remove all the components for the
device that was used to register the component in the first place.

However, some drivers register several components (such as a DAI and a
dmaengine PCM) on the same device, and if the dmaengine PCM is registered
first, then the DAI will be cleaned up first and
snd_dmaengine_pcm_unregister will be called next.

snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component
on the device, and if there's one unregister that component and release its
dmaengine channels. That doesn't happen in practice though since the first
call to snd_soc_unregister_component removed all the components, so we
never get the chance to release the dmaengine channels.

In order to fix this, instead of removing all the components for a given
device, we can simply remove the component that was registered in the first
place. We should have the same number of component registration than we
have components, so it should work just fine.

Signed-off-by: Maxime Ripard <maxime@cerno.tech>
Link: https://lore.kernel.org/r/20200707074237.287171-1-maxime@cerno.tech
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: compress: fix partial_drain completion state

On partial_drain completion we should be in SNDRV_PCM_STATE_RUNNING
state, so set that for partially draining streams in
snd_compr_drain_notify() and use a flag for partially draining streams

While at it, add locks for stream state change in
snd_compr_drain_notify() as well.

Fixes: f44f2a5417b2 ("ALSA: compress: fix drain calls blocking other compress functions (v6)")
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200629134737.105993-4-vkoul@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Add implicit feedback quirk for RTX6001

USB Audio analyzer RTX6001 uses the same implicit feedback quirk
as other XMOS-based devices.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/822f0f20-1886-6884-a6b2-d11c685cbafa@ivitera.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add quirk for MacroSilicon MS2109

These devices claim to be 96kHz mono, but actually are 48kHz stereo with
swapped channels and unaligned transfers.

Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200702071433.237843-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek: Enable headset mic of Acer Veriton N4660G with ALC269VC

The Acer Veriton N4660G desktop's audio (1025:1248) with ALC269VC cannot
detect the headset microphone until ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE
quirk maps the NID 0x18 as the headset mic pin.

Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-3-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek: Enable headset mic of Acer C20-820 with ALC269VC

The Acer Aspire C20-820 AIO's audio (1025:1065) with ALC269VC can't
detect the headset microphone until ALC269VC_FIXUP_ACER_HEADSET_MIC
quirk maps the NID 0x18 as the headset mic pin.

Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-2-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC

The Acer desktop vCopperbox with ALC269VC cannot detect the MIC of
headset, the line out and internal speaker until
ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied.

Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id

1)
In snd_hda_pick_fixup(), quirks are first matched by PCI SSID and then, if
there is no match, by codec SSID. The Lenovo "ThinkPad X1 Carbon 7th" has
an audio chip with PCI SSID 0x2292 and codec SSID 0x2293[1]. Therefore, fix
the quirk meant for that device to match on .subdevice == 0x2292.

2)
The "Thinkpad X1 Yoga 7th" does not exist. The companion product to the
Carbon 7th is the Yoga 4th. That device has an audio chip with PCI SSID
0x2292 and codec SSID 0x2292[2]. Given the behavior of
snd_hda_pick_fixup(), it is not possible to have a separate quirk for the
Yoga based on SSID. Therefore, merge the quirks meant for the Carbon and
Yoga. This preserves the current behavior for the Yoga.

[1] This is the case on my own machine and can also be checked here
https://github.com/linuxhw/LsPCI/tree/master/Notebook/Lenovo/ThinkPad
https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3225701
[2]
https://github.com/linuxhw/LsPCI/tree/master/Convertible/Lenovo/ThinkPad
https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3176355

Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen")
Fixes: 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen")
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Kailang Yang <kailang@realtek.com>
Tested-by: Vincent Bernat <vincent@bernat.ch>
Tested-by: Even Brenden <evenbrenden@gmail.com>
Signed-off-by: Benjamin Poirier <benjamin.poirier@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200703080005.8942-2-benjamin.poirier@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/hdmi: improve debug traces for stream lookups

The HDMI codec driver has two debug traces printed from different
functions but with identical message content:

"HDMI: hinfo 000000006a6b84d9 not registered"

Fix this duplication and also add a bit more context in addition to raw
object pointer, to help analysis of kernel logs.

Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200703153818.2808592-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later

When HDMI PCM devices are opened in a specific order, with at least one
HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the
connected monitor, on recent Intel platforms (ICL/JSL and newer). While
this is not a typical sequence, at least Pulseaudio does this every time
when it is started, to discover the available PCMs.

The rootcause is an invalid assumption in hdmi_add_pin(), where the
total number of converters is assumed to be known at the time the
function is called. On older Intel platforms this held true, but after
ICL/JSL, the order how pins and converters are in the subnode list as
returned by snd_hda_get_sub_nodes(), was changed. As a result,
information for some converters was not stored to per_pin->mux_nids.
And this means some pins cannot be connected to all converters, and
application instead gets -EBUSY instead at open.

The assumption that converters are always before pins in the subnode
list, is not really a valid one. Fix the problem in hdmi_parse_codec()
by introducing separate loops for discovering converters and pins.

BugLink: https://github.com/thesofproject/linux/issues/1978
BugLink: https://github.com/thesofproject/linux/issues/2216
BugLink: https://github.com/thesofproject/linux/issues/2217
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: opl3: fix infoleak in opl3

The stack object “info” in snd_opl3_ioctl() has a leaking problem.
It has 2 padding bytes which are not initialized and leaked via
“copy_to_user”.

Signed-off-by: xidongwang <wangxidong_97@163.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1594006058-30362-1-git-send-email-wangxidong_97@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

MAINTAINERS: Change Maintainer for some at91 drivers

I hand over the maintenance of these drivers to my colleagues. Claudiu,
Codrin and Tudor already have experience with these controllers and
sub-systems.

Signed-off-by: Nicolas Ferre <nicolas.ferre@microchip.com>
Link: https://lore.kernel.org/r/20200702134224.3750-1-nicolas.ferre@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings

The following build warnings are seen with 'make dt_binding_check':

Documentation/devicetree/bindings/sound/simple-card.example.dts:209.46-211.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@0: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:213.37-215.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@1: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:250.42-261.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@0: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:263.42-288.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:270.32-272.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@0: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:273.23-275.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@1: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:276.23-278.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@2: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:279.23-281.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@3: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:290.42-303.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@2: node has a unit name, but no reg or ranges property

Fix them all.

Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20200630223020.25546-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rockchip: add format and rate constraints on rk3399

S8 and S24 formats does not work on this machine driver so force to use
S16_LE instead.

In addition, add constraint to limit the max value of rate because the
rate higher than 96000(172000, 192000) is not stable either.

Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Link: https://lore.kernel.org/r/20200630091615.4020059-1-yuhsuan@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: amd: Rectifying Unbalanced pm_runtime_enable! issue

When snd_pci_acp3x driver loads we see:

WARNING kernel:snd_pci_acp3x 0000:04:00.5: Unbalanced pm_runtime_enable!
at boot time.
same can be observed in /var/log/messages/.

Modifying pm runtime sequence for fixing unbalanced pm issue.

Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200630092242.7799-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rt5682: cancel jack_detect_work if hs_jack is set to null even soundwire mode

Base on https://patchwork.kernel.org/patch/11237953/
Soundwire mode also should follow it.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Link: https://lore.kernel.org/r/20200701071645.32061-1-oder_chiou@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge tag 'asoc-fix-v5.8-rc3' of https://git./linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.8

A few small driver specific fixes, nothing particularly dramatic.

ALSA: usb-audio: Replace s/frame/packet/ where appropriate

Replace several occurences of "frame" with a "packet" where appropriate.

Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200629025934.154288-2-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Fix packet size calculation

Commit f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation")
introduced a regression for devices which have playback endpoints with
bInterval > 1. Fix this by taking ep->datainterval into account.

Note that frame and fps are actually mean packet and packets per second
in the code introduces by the mentioned commit. This will be fixed in a
follow-up patch.

Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208353
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200629025934.154288-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10

The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.

E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.

Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200628155231.71089-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rt5670: Add new gpio1_is_ext_spk_en quirk and enable it on the Lenovo Miix 2 10

The Lenovo Miix 2 10 has a keyboard dock with extra speakers in the dock.
Rather then the ACL5672's GPIO1 pin being used as IRQ to the CPU, it is
actually used to enable the amplifier for these speakers
(the IRQ to the CPU comes directly from the jack-detect switch).

Add a quirk for having an ext speaker-amplifier enable pin on GPIO1
and replace the Lenovo Miix 2 10's dmi_system_id table entry's wrong
GPIO_DEV quirk (which needs to be renamed to GPIO1_IS_IRQ) with the
new RT5670_GPIO1_IS_EXT_SPK_EN quirk, so that we enable the external
speaker-amplifier as necessary.

Also update the ident field for the dmi_system_id table entry, the
Miix models are not Thinkpads.

Fixes: 67e03ff3f32f ("ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rt5670: Correct RT5670_LDO_SEL_MASK

The RT5670_PWR_ANLG1 register has 3 bits to select the LDO voltage,
so the correct mask is 0x7 not 0x3.

Because of this wrong mask we were programming the ldo bits
to a setting of binary 001 (0x05 & 0x03) instead of binary 101
when moving to SND_SOC_BIAS_PREPARE.

According to the datasheet 001 is a reserved value, so no idea
what it did, since the driver was working fine before I guess we
got lucky and it does something which is ok.

Fixes: 5e8351de740d ("ASoC: add RT5670 CODEC driver")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200628155231.71089-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: cht_bsw_rt5672: Change bus format to I2S 2 channel

The default mode for SSP configuration is TDM 4 slot and so far we were
using this for the bus format on cht-bsw-rt56732 boards.

One board, the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2.
The second piggy-backed, output-only codec is inside the keyboard-dock
(which has extra speakers). Unlike the main rt5672 codec, we cannot
configure this codec, it is hard coded to use 2 channel 24 bit I2S.

Using 4 channel TDM leads to the dock speakers codec (which listens in on
the data send from the SSP to the rt5672 codec) emiting horribly distorted
sound.

Since we only support 2 channels anyways, there is no need for TDM on any
cht-bsw-rt5672 designs. So we can simply use I2S 2ch everywhere.

This commit fixes the Lenovo Miix 2 10 dock speakers issue by changing
the bus format set in cht_codec_fixup() to I2S 2 channel.

This change has been tested on the following devices with a rt5672 codec:

Lenovo Miix 2 10
Lenovo Thinkpad 8
Lenovo Thinkpad 10 (gen 1)

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: qcom: Drop HAS_DMA dependency to fix link failure

When building on allyesconfig kernel for a NO_DMA=y platform (e.g.
Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n,
leading to a link failure:

    sound/soc/qcom/common.o: In function `qcom_snd_parse_of':
    common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port'

While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845
don't, so the following warning is seen:

    WARNING: unmet direct dependencies detected for SND_SOC_QDSP6
      Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n]
      Selected by [y]:
      - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y]
      - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y]

Until recently, this warning was harmless (from a compile-testing
point-of-view), but the new user of q6afe_is_rx_port() turned this into
a hard failure.

As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on
QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is
safe to increase compile testing coverage.  Hence fix the link failure
by dropping the HAS_DMA dependency of SND_SOC_QDSP6.

Fixes: a2120089251f1fe2 ("ASoC: qcom: common: set correct directions for dailinks")
Fixes: 6b1687bf76ef84cb ("ASoC: qcom: add sdm845 sound card support")
Fixes: a6f933f63f2ffdb2 ("ASoC: qcom: apq8096: Add db820c machine driver")
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20200629122443.21736-1-geert@linux-m68k.org
Signed-off-by: Mark Brown <broonie@kernel.org>

AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Alex Deucher <alexander.deucher@amd.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: vijendar.mukunda@amd.com
Cc: Alexander.Deucher@amd.com
Link: https://lore.kernel.org/r/20200625115829.791750-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda - let hs_mic be picked ahead of hp_mic

We have a Dell AIO, there is neither internal speaker nor internal
mic, only a multi-function audio jack on it.

Users reported that after freshly installing the OS and plug
a headset to the audio jack, the headset can't output sound. I
reproduced this bug, at that moment, the Input Source is as below:
Simple mixer control 'Input Source',0
  Capabilities: cenum
  Items: 'Headphone Mic' 'Headset Mic'
  Item0: 'Headphone Mic'

That is because the patch_realtek will set this audio jack as mic_in
mode if Input Source's value is hp_mic.

If it is not fresh installing, this issue will not happen since the
systemd will run alsactl restore -f /var/lib/alsa/asound.state, this
will set the 'Input Source' according to history value.

If there is internal speaker or internal mic, this issue will not
happen since there is valid sink/source in the pulseaudio, the PA will
set the 'Input Source' according to active_port.

To fix this issue, change the parser function to let the hs_mic be
stored ahead of hp_mic.

Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200625083833.11264-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: rt5682: fix the pop noise while OMTP type headset plugin

To turn the headphone output switch off during jack type detection, it
could avoid the pop noise when jack type switches to OMTP type.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200623125312.27896-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usb-audio: Fix OOB access of mixer element list

The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element.  A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members.  This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.

This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.

Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com
Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add quirk for Samsung USBC Headset (AKG)

We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051)
need a tiny delay after each class compliant request.
Otherwise the device might not be able to be recognized each times.

Signed-off-by: Chihhao Chen <chihhao.chen@mediatek.com>
Signed-off-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/1592910203-24035-1-git-send-email-macpaul.lin@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable

Fix unchecked return value for clk_prepare_enable, add error
handler in fsl_mqs_runtime_resume.

Fixes: 9e28f6532c61 ("ASoC: fsl_mqs: Add MQS component driver")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/5edd68d03def367d96268f1a9a00bd528ea5aaf2.1592888591.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: fsl_mqs: Don't check clock is NULL before calling clk API

Because clk_prepare_enable and clk_disable_unprepare should
check input clock parameter is NULL or not internally, then
we don't need to check them before calling the function.

Fixes: 9e28f6532c61 ("ASoC: fsl_mqs: Add MQS component driver")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/743be216bd504c26e8d45d5ce4a84561b67a122b.1592888591.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Flight S

Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:0x16ea) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.

Signed-off-by: Christoffer Nielsen <cn@obviux.dk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAOtG2YHOM3zy+ed9KS-J4HkZo_QGzcUG9MigSp4e4_-13r6B=Q@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge tag 'asoc-fix-v5.8-rc2' of https://git./linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.8

This is a collection of mostly small fixes, mostly fixing fallout from
some of the DPCM changes that went in last time around which shook out
some issues on i.MX and Qualcomm platforms.  The addition of a managed
version of snd_soc_register_dai() is to fix resource leaks.

There's also a few new device IDs for x86 systems.

ASoC: rockchip: Fix a reference count leak.

Calling pm_runtime_get_sync increments the counter even in case of
failure, causing incorrect ref count if pm_runtime_put is not called in
error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails.

Fixes: fc05a5b22253 ("ASoC: rockchip: add support for pdm controller")
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Reviewed-by: Heiko Stuebner <heiko@sntech.de>
Link: https://lore.kernel.org/r/20200613205158.27296-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: amd: closing specific instance.

The steps to reproduce:

Record from the internal mic :
(arecord -D hw:1,2 -f dat /dev/null -V stereos)

Record from the headphone mic:
(arecord -D hw:1,0 -f dat /dev/null -V stereos)

Kill the recording from internal mic.
We can see the recording from the headphone mic is broken.

This patch rectifies the issue reported.

Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200618072653.27103-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda: Intel: add missing PCI IDs for ICL-H, TGL-H and EKL

Mirror PCI ids used for SOF.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164909.18225-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge series "ASoC: SOF: Intel: update PCI IDs" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

Clean-up CometLake and add missing PCI IDs. Changes for the legacy
driver are sent separately.

Pierre-Louis Bossart (3):
  ASoC: Intel: SOF: merge COMETLAKE_LP and COMETLAKE_H
  ASoC: SOF: Intel: add PCI ID for CometLake-S
  ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H

 sound/hda/intel-dsp-config.c   |  4 +---
 sound/soc/intel/boards/Kconfig |  4 ++--
 sound/soc/sof/intel/Kconfig    | 29 ++++++++---------------------
 sound/soc/sof/sof-pci-dev.c    | 24 ++++++++++++++----------
 4 files changed, 25 insertions(+), 36 deletions(-)

--
2.20.1

ASoC: hdac_hda: fix memleak with regmap not freed on remove

kmemleak throws error reports on module load/unload tests, add
snd_hdac_regmap_exit() in .remove().

While we are at it, also fix the error handling flow in .probe() to
use snd_hdac_regmap_exit() if needed.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164144.17859-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H

Usually the DSP is not traditionally enabled on H skews but this might
be used moving forward.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164755.18104-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: add PCI ID for CometLake-S

Mirror ID added for legacy HDaudio

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164755.18104-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: SOF: merge COMETLAKE_LP and COMETLAKE_H

We already have two configurations for CometLake, and a third one
coming. On other platforms, we used a single Kconfig option, so we
should follow the same trend by merging the two cases in a backwards
compatible way.

The backwards compatibility is handled by overloading the COMETLAKE_LP
kconfig as COMETLAKE. In practice we've never seen a case where
COMETLAKE_H is not selected along with COMETLAKE_LP, so keeping one
of the two is enough.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164755.18104-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda/realtek: Add mute LED and micmute LED support for HP systems

There are two more HP systems control mute LED from HDA codec and need
to expose micmute led class so SoF can control micmute LED.

Add quirks to support them.

Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200617102906.16156-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Fix potential use-after-free of streams

With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running.  In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer.  Since
the PCM stream gets closed, this may lead to use-after-free.

This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.

Fixes: 10ce77e4817f ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Add quirk for MSI GE63 laptop

MSI GE63 laptop with ALC1220 codec requires the very same quirk
(ALC1220_FIXUP_CLEVO_P950) as other MSI devices for the proper sound
output.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208057
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200616132150.8778-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: fsl_ssi: Fix bclk calculation for mono channel

For mono channel, SSI will switch to Normal mode.

In Normal mode and Network mode, the Word Length Control bits
control the word length divider in clock generator, which is
different with I2S Master mode (the word length is fixed to
32bit), it should be the value of params_width(hw_params).

The condition "slots == 2" is not good for I2S Master mode,
because for Network mode and Normal mode, the slots can also
be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK)
to check if it is I2S Master mode.

So we refine the formula for mono channel, otherwise there
will be sound issue for S24_LE.

Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/034eff1435ff6ce300b6c781130cefd9db22ab9a.1592276147.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge series "ASoC: topology: fix use-after-free when removing components" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

This patchset fixes a memory allocation issue and removes a 100%
reproducible use-after-free report thrown by KASAN in automated module
removal tests across multiple platforms.

All the credit goes to Bard Liao for root-causing the issue. DAIs may
be registered at the same time as a component, or when the topology is
loaded. This two-step registration causes the memory for
topology-based DAIs to allocated last, and conversely to be released
first by devres, before the component is released and the DAIs removed
from the component DAI list with snd_soc_unregister_dais().

When we remove a component, by the time we walk through its dai list
to unregister all dais, the dais allocated by the topology have been
freed already by devres and the list is corrupted with pointers that
are no longer valid.

The suggestion is to add an explicit devm_ based registration for
topology-based dais, so that each dai is cleanly removed from the
component dai list in the release operation before devres releases the
allocated memory.

Pierre-Louis Bossart (2):
  ASoC: soc-devres: add devm_snd_soc_register_dai()
  ASoC: soc-topology: use devm_snd_soc_register_dai()

 include/sound/soc.h      |  4 ++++
 sound/soc/soc-devres.c   | 37 +++++++++++++++++++++++++++++++++++++
 sound/soc/soc-topology.c |  3 +--
 3 files changed, 42 insertions(+), 2 deletions(-)

--
2.20.1

ASoC: SOF: Intel: hda: Clear RIRB status before reading WP

Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading
WP") from legacy HDA driver to fix the get response timeout issue.
Current SOF driver does not suffer from this issue because sync write
is enabled in hda_init. The issue will come back if the sync write is
disabled for some reason.

Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1591959048-15813-1-git-send-email-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rt1015: Update rt1015 default register value according to spec modification.

Update rt1015 default register value according to spec modification.

Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20200615032433.31061-1-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: qcom: common: set correct directions for dailinks

Currently both FE and BE dai-links are configured bi-directional,
However the DSP BE dais are only single directional,
so set the directions as supported by the BE dais.

Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: John Stultz <john.stultz@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoc: q6afe: add support to get port direction

This patch adds support to q6afe_is_rx_port() to get direction
of DSP BE dai port, this is useful for setting dailink
directions correctly.

Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-pcm: fix checks for multi-cpu FE dailinks

soc_dpcm_fe_runtime_update() is called for all dailinks, and we want
to first discard all back-ends, then deal with front-ends.

The existing code first reports an error with multi-cpu front-ends,
and that check needs to be moved after we know that we are dealing
with a front-end.

Fixes: 6e1276a5e613d ('ASoC: Return error if the function does not support multi-cpu')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/1970
Link: https://lore.kernel.org/r/20200612203507.25621-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rt5682: Let dai clks be registered whether mclk exists or not

According to ideal rt5682 CCF, the root clk is mclk.
But in some platforms, mclk is not exported to CCF.
In this condition, rt5682_register_dai_clks will not be called.
This patch lets dai clks could be registered whether mclk exists or not.

Signed-off-by: derek.fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/1591938925-1070-5-git-send-email-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-topology: use devm_snd_soc_register_dai()

Use devm_ to avoid use-after-free KASAN reports and simplify error
handling.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2186
Link: https://lore.kernel.org/r/20200612205938.26415-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-devres: add devm_snd_soc_register_dai()

The registration of DAIs may be done at two distinct times, once
during a component registration and later when loading a
topology. Since devm_ managed resources are freed in the reverse order
they were allocated, when a component starts unregistering DAIs by
walking through the DAI list, the memory allocated for the
topology-registered DAIs was freed already, which leads to 100%
reproducible KASAN use-after-free reports.

This patch suggests a new devm_ function to force the DAI list to be
updated prior to freeing the memory chunks referenced by the list
pointers.

Suggested-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2186
Link: https://lore.kernel.org/r/20200612205938.26415-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usb-audio: Set 48 kHz rate for Rodecaster

Like the Line6 devices, the Rode Rodecaster Pro does not support
UAC2_CS_RANGE and only supports a sample rate of 48 kHz.

Tested against a Rode Rodecaster Pro.

Tested-by: Christopher Swenson <swenson@swenson.io>
Signed-off-by: Christopher Swenson <swenson@swenson.io>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/ebdb9e72-9649-0b5e-b9b9-d757dbf26927@swenson.io
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add quirk for Denon DCD-1500RE

fix error "clock source 41 is not valid, cannot use"

[] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00
[] New USB device strings: Mfr=1, Product=2, SerialNumber=0
[] Product: DCD-1500RE
[] Manufacturer: D & M Holdings Inc.
[]
[] clock source 41 is not valid, cannot use
[] usbcore: registered new interface driver snd-usb-audio

Signed-off-by: Yick W. Tse <y_w_tse@yahoo.com.hk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1373857985.210365.1592048406997@mail.yahoo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: fsl_asrc_dma: Fix data copying speed issue with EDMA

With EDMA, there is two dma channels can be used for dev_to_dev,
one is from ASRC, one is from another peripheral (ESAI or SAI).

If we select the dma channel of ASRC, there is an issue for ideal
ratio case, the speed of copy data is faster than sample
frequency, because ASRC output data is very fast in ideal ratio
mode.

So it is reasonable to use the dma channel of Back-End peripheral.
then copying speed of DMA is controlled by data consumption
speed in the peripheral FIFO,

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/424ed6c249bafcbe30791c9de0352821c5ea67e2.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End

The dma channel has been requested by Back-End cpu dai driver already.
If fsl_asrc_dma requests dma chan with same dma:tx symlink, then
there will be below warning with SDMA.

[   48.174236] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink

So if we can reuse the dma channel of Back-End, then the issue can be
fixed.

In order to get the dma channel which is already requested in Back-End.
we use the exported two functions (snd_soc_lookup_component_nolocked
and soc_component_to_pcm). If we can get the dma channel, then reuse it,
if can't, then request a new one.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/3a79f0442cb4930c633cf72145cfe95a45b9c78e.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dmaengine_pcm: export soc_component_to_pcm

In DPCM case, Front-End needs to get the dma chan which has
been requested by Back-End and reuse it.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/429c6ae1f3c5b47eb893f475d531d71cdcfe34c0.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-card: export snd_soc_lookup_component_nolocked

snd_soc_lookup_component_nolocked can be used for the DPCM case
that Front-End needs to get the unused platform component but
added by Back-End cpu dai driver.

If the component is gotten, then we can get the dma chan created
by Back-End component and reused it in Front-End.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/55f6e0d76f67a517b9a44136d790ff2a06b5caa8.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usb-audio: Add implicit feedback quirk for SSL2+.

This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the
audio interface's clock. Tested on an SSL2+.

Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200612111807.dgnig6rwhmsl2bod@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Add NVIDIA codec IDs 9a & 9d through a0 to patch table

These IDs are for upcoming NVIDIA chips with audio functions that are largely
similar to the existing ones.

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200611180845.39942-1-aplattner@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: q6asm: handle EOS correctly

Successful send of EOS command does not indicate that EOS is actually
finished, correct event to wait EOS is finished is EOS_RENDERED event.
EOS_RENDERED means that the DSP has finished processing all the buffers
for that particular session and stream.

This patch fixes EOS handling!

Fixes: 68fd8480bb7b ("ASoC: qdsp6: q6asm: Add support to audio stream apis")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200611124159.20742-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: max98390: Update regmap readable reg and volatile

 Update max98390_readable_register and max98390_volatile_reg

Signed-off-by: Steve Lee <steves.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20200611094800.18422-1-steves.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>