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

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

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

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

Merge tag 'asoc-fix-v5.8-rc2' of https://git.kernel.org/pub/scm/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.

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: fc05a5b2 ("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.eduSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-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

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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

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: 10ce77e4 ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

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

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: b0a7043d ("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.comSigned-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

Port commit 6d011d50 ("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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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: c25e295c (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.orgSigned-off-by: Mark Brown <broonie@kernel.org>

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: c25e295c (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.orgSigned-off-by: Mark Brown <broonie@kernel.org>

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: 6e1276a5 ('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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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

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

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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

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

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: 68fd8480 ("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.orgSigned-off-by: Mark Brown <broonie@kernel.org>

 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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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

ASoC: Fixes for v5.8

A small pile of fixes that came in during the merge window, the DPCM
fixes from Pierre are the most notable thing here.

Merge series "ASoC: Fix dailink checks for DPCM" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

We've had a couple of changes that introduce regressions with the
multi-cpu DAI solutions, and while trying to fix them we found
additional inconsistencies that should also go to stable branches.

Bard Liao (1):
  ASoC: core: only convert non DPCM link to DPCM link

Pierre-Louis Bossart (3):
  ASoC: soc-pcm: dpcm: fix playback/capture checks
  ASoC: Intel: boards: replace capture_only by dpcm_capture
  ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags

 sound/soc/intel/boards/glk_rt5682_max98357a.c |  2 +-
 sound/soc/intel/boards/kbl_da7219_max98927.c  |  4 +-
 sound/soc/intel/boards/kbl_rt5663_max98927.c  |  2 +-
 .../intel/boards/kbl_rt5663_rt5514_max98927.c |  2 +-
 sound/soc/soc-core.c                          | 22 ++++++++--
 sound/soc/soc-pcm.c                           | 44 ++++++++++++++-----
 sound/soc/sof/nocodec.c                       |  6 ++-
 7 files changed, 62 insertions(+), 20 deletions(-)

base-commit: 8a9144c1
--
2.20.1

The Asus T101HA uses the default jack-detect mode 3, but instead of
using an analog microphone it is using a DMIC on dmic-data-pin 1,
like the Asus T100HA. Note unlike the T100HA its jack-detect is not
inverted.

Add a DMI quirk with the correct settings for this model.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>

The Toshiba Encore WT10-A tablet almost fully works with the default
settings for Bay Trail CR devices. The only issue is that it uses a
digital mic. connected the the DMIC1 input instead of an analog mic.

Add a quirk for this model using the default settings with the input-map
replaced with BYT_RT5640_DMIC1_MAP.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.comSigned-off-by: Mark Brown <broonie@kernel.org>