Stripe control programming is governed by following formula, which is
referenced from the HD Audio specification(Revision 1.0a).
  { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }

Currently above is implemented in snd_hdac_get_stream_stripe_ctl().
This patch introduces a structure member to store the default factor
of '8'. If any HW wants to use a different value, this member can be
easily updated.
Signed-off-by: NSameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1588580176-2801-3-git-send-email-spujar@nvidia.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Tegra194 supports 4 SDO lines but GCAP register indicates 2 lines. Thus it
does not reflect the true capability of the HW. This patch presents a
workaround by updating NSDO value accordingly in T_AZA_DBG_CFG_2 register.
Signed-off-by: NSameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1588580176-2801-2-git-send-email-spujar@nvidia.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

At least POD HD500 uses message-based communication, both sides can
send messages. Add poll callback so application can wait for device
messages without using busy loop.
Signed-off-by: NVasily Khoruzhick <anarsoul@gmail.com>
Link: https://lore.kernel.org/r/20200502193120.79115-3-anarsoul@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Currently line6 hwdep interface ignores O_NONBLOCK flag when
opening device and it renders it somewhat useless when using poll.

Check for O_NONBLOCK flag when opening device and don't block read()
if it is set.
Signed-off-by: NVasily Khoruzhick <anarsoul@gmail.com>
Link: https://lore.kernel.org/r/20200502193120.79115-2-anarsoul@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The "header->number" comes from the ioctl and it needs to be clamped to
prevent out of bounds writes.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200501094011.GA960082@mwandaSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Cover with a proper ifdef around the variable declaration for fixing
the following compilation warning without CONFIG_LEDS_TRIGGER_AUDIO:
  sound/pci/hda/patch_realtek.c: In function 'alc_fixup_hp_gpio_led':
  sound/pci/hda/patch_realtek.c:4134:6: warning: unused variable 'err' [-Wunused-variable]
Reported-by: NStephen Rothwell <sfr@canb.auug.org.au>
Fixes: 87dc3648 ("ALSA: hda/realtek - Add LED class support for micmute LED")
Link: https://lore.kernel.org/r/20200501072857.13720-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Currently DMIC controls micmute LED via "audio mute LED trigger".

However, unlike Dell and Lenovo platforms, HP platforms don't provide a
way to control micmute LED via ACPI, it's controlled by HDA codec
instead.

So let's register an LED class for micmute so other subsystems like DMIC
can facilitate the codec-controlled LED.
Signed-off-by: NKai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200430135209.14703-1-kai.heng.feng@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Though the system uses DMIC, headset mic still uses the HDA, let's use
GPIO 0x1 to control the micmute LED.

The micmute LED GPIO has a different polarity to the mute LED GPIO, we
can use the newly added micmute_led_polarity to indicate that.
Signed-off-by: NKai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200430083255.5093-2-kai.heng.feng@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Currently mute LED and micmute LED share the same GPIO polarity.

So split the polarity for mute and micmute, in case they have different
polarities.
Signed-off-by: NKai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

There's no callers in-tree.
Signed-off-by: NYueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200429132805.18712-1-yuehaibing@huawei.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The linked list entry from FIFO is peeked at
queue_pending_output_urbs() but the actual element pop-out is
performed outside the spinlock, and it's potentially racy.

Do delete the link at the right place inside the spinlock.

Fixes: 8fdff6a3 ("ALSA: snd-usb: implement new endpoint streaming model")
Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Frame size computation has been fixed and the workaround is no longer
needed.
Signed-off-by: NAlexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200424022449.14972-2-alexander@tsoy.meSigned-off-by: NTakashi Iwai <tiwai@suse.de>

For computation of the the next frame size current value of fs/fps and
accumulated fractional parts of fs/fps are used, where values are stored
in Q16.16 format. This is quite natural for computing frame size for
asynchronous endpoints driven by explicit feedback, since in this case
fs/fps is a value provided by the feedback endpoint and it's already in
the Q format. If an error is accumulated over time, the device can
adjust fs/fps value to prevent buffer overruns/underruns.

But for synchronous endpoints the accuracy provided by these computations
is not enough. Due to accumulated error the driver periodically produces
frames with incorrect size (+/- 1 audio sample).

This patch fixes this issue by implementing a different algorithm for
frame size computation. It is based on accumulating of the remainders
from division fs/fps and it doesn't accumulate errors over time. This
new method is enabled for synchronous and adaptive playback endpoints.
Signed-off-by: NAlexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.meSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Back-merge 5.7-rc devel branch for further changes.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The commit 3c6fd1f0 ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.

Since the empty codec problem appear on the certain AMD platform (PCI
ID 1022:1487), this patch changes the blacklist matching to both PCI
ID and SSID using pci_match_id().  Also, the entry that was removed by
the previous fix for ASUS ROG Zenigh II is re-added.

Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Merge NHLT init cleanup.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

NHLT fetch based on _DSM prevents ACPI table override mechanism from
being utilized. Make use of acpi_get_table to enable it and get rid of
redundant code. In consequence, NHLT can be overridden just like any
other ACPI table, e.g.: DSDT or SSDT.

Change has been verified on all Intel AVS architecture platforms, RVP
and production laptops both.

Change possible due to addition of NHLT signature to the list of
standard ACPI tables:
https://patchwork.kernel.org/patch/11463235/

Override helps not only with debug purposes but also allows user for
table adjustment when one found on their production hardware is invalid.
Shared official NHLT spec is now available to community at:
https://01.org/blogs/intel-smart-sound-technology-audio-dsp

NHLT support for iASL is still ongoing subject but should be available
in nearest future.
Signed-off-by: NCezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200423160310.28019-1-cezary.rojewski@intel.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

HD-audio codec driver applies a tricky procedure to forcibly perform
the runtime resume by mimicking the usage count even if the device has
been runtime-suspended beforehand.  This was needed to assure to
trigger the jack detection update after the system resume.

And recently we also applied the similar logic to the HD-audio
controller side.  However this seems leading to some inconsistency,
and eventually PCI controller gets screwed up.

This patch is an attempt to fix and clean up those behavior: instead
of the tricky runtime resume procedure, the existing jackpoll work is
scheduled when such a forced codec resume is required.  The jackpoll
work will power up the codec, and this alone should suffice for the
jack status update in usual cases.  If the extra polling is requested
(by checking codec->jackpoll_interval), the manual update is invoked
after that, and the codec is powered down again.

Also, we filter the spurious wake up of the codec from the controller
runtime resume by checking codec->relaxed_resume flag.  If this flag
is set, basically we don't need to wake up explicitly, but it's
supposed to be done via the audio component notifier.

Fixes: c4c8dd6e ("ALSA: hda: Skip controller resume if not needed")
Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Enable new codec supported for ALC245.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/8c0804738b2c42439f59c39c8437817f@realtek.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which
increases the refcount of the snd_usb_audio object "chip".

When snd_microii_spdif_default_get() returns, local variable "chip"
becomes invalid, so the refcount should be decreased to keep refcount
balanced.

The reference counting issue happens in several exception handling paths
of snd_microii_spdif_default_get(). When those error scenarios occur
such as usb_ifnum_to_if() returns NULL, the function forgets to decrease
the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak.

Fix this issue by jumping to "end" label when those error scenarios
occur.

Fixes: 447d6275 ("ALSA: usb-audio: Add sanity checks for endpoint accesses")
Signed-off-by: NXiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: NXin Tan <tanxin.ctf@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cnSigned-off-by: NTakashi Iwai <tiwai@suse.de>

It turned out that ALC1220-VB USB-audio device gives the interrupt
event to some PCM terminals while those don't allow the connector
state request but only the actual I/O terminals return the request.
The recent commit 7dc3c5a0 ("ALSA: usb-audio: Don't create jack
controls for PCM terminals") excluded those phantom terminals, so
those events are ignored, too.

My first thought was that this could be easily deduced from the
associated terminals, but some of them have even no associate terminal
ID, hence it's not too trivial to figure out.

Since the number of such terminals are small and limited, this patch
implements another quirk table for the simple mapping of the
connectors.  It's not really scalable, but let's hope that there will
be not many such funky devices in future.

Fixes: 7dc3c5a0 ("ALSA: usb-audio: Don't create jack controls for PCM terminals")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the following coccicheck warning:

sound/pci/oxygen/xonar_pcm179x.c:463:1-17: WARNING: Assignment of 0/1 to
bool variable
sound/pci/oxygen/xonar_pcm179x.c:505:1-17: WARNING: Assignment of 0/1 to
bool variable
Signed-off-by: NJason Yan <yanaijie@huawei.com>
Link: https://lore.kernel.org/r/20200422071646.48436-1-yanaijie@huawei.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

This should be ARRAY_SIZE() instead of sizeof().  The sizeof() limit is
too high so it doesn't work.

Fixes: 093b8494 ("ALSA: usb-audio: Print more information in stream proc files")
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200422092255.GB195357@mwandaSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Merge tag 'asoc-fix-v5.7-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.7

Quite a lot of fixes here, a lot of driver specific ones but the biggest
one is the revert of changes to the startup and shutdown sequence for
DAIs that went in during the merge window - they broke some older x86
platforms and attempts to fix them didn't succeed so it's safer to just
roll them back and try to make sure those platforms are handled properly
in any future attempt.

The rockchip S/PDIF DT stuff was IIRC for validation issues.

Due to rounding error driver sometimes incorrectly calculate next packet
size, which results in audible clicks on devices with synchronous playback
endpoints. For example on a high speed bus and a sample rate 44.1 kHz it
loses one sample every ~40.9 seconds. Fortunately playback interface on
Scarlett 2i4 2nd gen has a working explicit feedback endpoint, so we can
switch playback data endpoint to asynchronous mode as a workaround.
Signed-off-by: NAlexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200421190908.462860-1-alexander@tsoy.meSigned-off-by: NTakashi Iwai <tiwai@suse.de>

After suspend & resume, wm8960_hw_params may be called when
bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking
is not called. But if sample rate is changed at that time, then
the output clock rate will be not correct.

So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params
is not necessary and it causes above issue.

Fixes: 3176bf2d ("ASoC: wm8960: update pll and clock setting function")
Signed-off-by: NShengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: NCharles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1587468525-27514-1-git-send-email-shengjiu.wang@nxp.comSigned-off-by: NMark Brown <broonie@kernel.org>

The error handling code in usX2Y_rate_set() may hit a potential NULL
dereference when an error occurs before allocating all us->urb[].
Add a proper NULL check for fixing the corner case.
Reported-by: NLin Yi <teroincn@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

Force it to use asynchronous playback.

Same quirk has already been added for Focusrite Scarlett Solo (2nd gen)
with a commit 46f5710f ("ALSA: usb-audio: Add quirk for Focusrite
Scarlett Solo").

This also seems to prevent regular clicks when playing at 44100Hz
on Scarlett 2i2 (2nd gen). I did not notice any side effects.

Moved both quirks to snd_usb_audioformat_attributes_quirk() as suggested.
Signed-off-by: NGregor Pintar <grpintar@gmail.com>
Reviewed-by: NAlexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.comSigned-off-by: NTakashi Iwai <tiwai@suse.de>

sound/soc/codecs/wm8900.o: In function `wm8900_i2c_probe':
wm8900.c:(.text+0xa36): undefined reference to `__devm_regmap_init_i2c'
sound/soc/codecs/wm8900.o: In function `wm8900_modinit':
wm8900.c:(.init.text+0xb): undefined reference to `i2c_register_driver'
sound/soc/codecs/wm8900.o: In function `wm8900_exit':
wm8900.c:(.exit.text+0x8): undefined reference to `i2c_del_driver'
sound/soc/codecs/wm8988.o: In function `wm8988_i2c_probe':
wm8988.c:(.text+0x857): undefined reference to `__devm_regmap_init_i2c'
sound/soc/codecs/wm8988.o: In function `wm8988_modinit':
wm8988.c:(.init.text+0xb): undefined reference to `i2c_register_driver'
sound/soc/codecs/wm8988.o: In function `wm8988_exit':
wm8988.c:(.exit.text+0x8): undefined reference to `i2c_del_driver'
sound/soc/codecs/wm8995.o: In function `wm8995_i2c_probe':
wm8995.c:(.text+0x1c4f): undefined reference to `__devm_regmap_init_i2c'
sound/soc/codecs/wm8995.o: In function `wm8995_modinit':
wm8995.c:(.init.text+0xb): undefined reference to `i2c_register_driver'
sound/soc/codecs/wm8995.o: In function `wm8995_exit':
wm8995.c:(.exit.text+0x8): undefined reference to `i2c_del_driver'

Add SND_SOC_I2C_AND_SPI dependency to fix this.

Fixes: ea00d952 ("ASoC: Use imply for SND_SOC_ALL_CODECS")
Reported-by: NHulk Robot <hulkci@huawei.com>
Signed-off-by: NYueHaibing <yuehaibing@huawei.com>
Acked-by: NCharles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200420125343.20920-1-yuehaibing@huawei.comSigned-off-by: NMark Brown <broonie@kernel.org>

Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz <matthias.blankertz@cetitec.com>:

Fix rsnd_dai_call() operations being performed twice for the master SSI
in multi-SSI setups, and fix the rsnd_ssi_stop operation for multi-SSI
setups.
The only visible effect of these issues was some "status check failed"
spam when the rsnd_ssi_stop was called, but overall the code is cleaner
now, and some questionable writes to the SSICR register which did not
lead to any observable misbehaviour but were contrary to the datasheet
are fixed.

Mark:
The first patch kind of reverts my "ASoC: rsnd: Fix parent SSI
start/stop in multi-SSI mode" from a few days ago and achieves the same
effect in a simpler fashion, if you would prefer a clean patch series
based on v5.6 drop me a note.

Greetings,
	Matthias

Matthias Blankertz (2):
  ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent
  ASoC: rsnd: Fix "status check failed" spam for multi-SSI

 sound/soc/sh/rcar/ssi.c | 18 +++++++++++++-----
 1 file changed, 13 insertions(+), 5 deletions(-)

base-commit: 15a5760c
--
2.26.1

Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet <jbrunet@baylibre.com>:

This patchset fixes the problem reported by Marc in this thread [0]
The problem was due to an error in the meson card drivers which had
the "no_pcm" dai_link property set on codec-to-codec links

[0]: https://lore.kernel.org/r/20200417122732.GC5315@sirena.org.uk

Jerome Brunet (2):
  ASoC: meson: axg-card: fix codec-to-codec link setup
  ASoC: meson: gx-card: fix codec-to-codec link setup

 sound/soc/meson/axg-card.c | 4 +++-
 sound/soc/meson/gx-card.c  | 4 +++-
 2 files changed, 6 insertions(+), 2 deletions(-)

--
2.25.2

snd_soc_dapm_kcontrol widget which is created by autodisable control
should contain correct on_val, mask and shift because it is set when the
widget is powered and changed value is applied on registers by following
code in dapm_seq_run_coalesced().

		mask |= w->mask << w->shift;
		if (w->power)
			value |= w->on_val << w->shift;
		else
			value |= w->off_val << w->shift;

Shift on the mask in dapm_kcontrol_data_alloc() is removed to prevent
double shift.
And, on_val in dapm_kcontrol_set_value() is modified to get correct
value in the dapm_seq_run_coalesced().
Signed-off-by: NGyeongtaek Lee <gt82.lee@samsung.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/000001d61537$b212f620$1638e260$@samsung.comSigned-off-by: NMark Brown <broonie@kernel.org>

Fix the rsnd_ssi_stop function to skip disabling the individual SSIs of
a multi-SSI setup, as the actual stop is performed by rsnd_ssiu_stop_gen2
- the same logic as in rsnd_ssi_start. The attempt to disable these SSIs
was harmless, but caused a "status check failed" message to be printed
for every SSI in the multi-SSI setup.
The disabling of interrupts is still performed, as they are enabled for
all SSIs in rsnd_ssi_init, but care is taken to not accidentally set the
EN bit for an SSI where it was not set by rsnd_ssi_start.
Signed-off-by: NMatthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.comSigned-off-by: NMark Brown <broonie@kernel.org>

The master SSI of a multi-SSI setup was attached both to the
RSND_MOD_SSI slot and the RSND_MOD_SSIP slot of the rsnd_dai_stream.
This is not correct wrt. the meaning of being "parent" in the rest of
the SSI code, where it seems to indicate an SSI that provides clock and
word sync but is not transmitting/receiving audio data.

Not treating the multi-SSI master as parent allows removal of various
special cases to the rsnd_ssi_is_parent conditions introduced in commit
a09fb3f2 ("ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode").
It also fixes the issue that operations performed via rsnd_dai_call()
were performed twice for the master SSI. This caused some "status check
failed" spam when stopping a multi-SSI stream as the driver attempted to
stop the master SSI twice.
Signed-off-by: NMatthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.comSigned-off-by: NMark Brown <broonie@kernel.org>

Since the addition of commit 9b5db059 ("ASoC: soc-pcm: dpcm: Only allow
playback/capture if supported"), meson-axg cards which have codec-to-codec
links fail to init and Oops.

  Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128
  Internal error: Oops: 96000044 [#1] PREEMPT SMP
  CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1
  pc : invalidate_paths_ep+0x30/0xe0
  lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
  Call trace:
   invalidate_paths_ep+0x30/0xe0
   snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
   dpcm_path_get+0x38/0xd0
   dpcm_fe_dai_open+0x70/0x920
   snd_pcm_open_substream+0x564/0x840
   snd_pcm_open+0xfc/0x228
   snd_pcm_capture_open+0x4c/0x78
   snd_open+0xac/0x1a8
   ...

While this error was initially reported the axg-card type, it also applies
to the gx-card type.

While initiliazing the links, ASoC treats the codec-to-codec links of this
card type as a DPCM backend. This error eventually leads to the Oops.

Most of the card driver code is shared between DPCM backends and
codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on
codec-to-codec links, leading to this problem. This commit fixes that.

Fixes: e37a0c31 ("ASoC: meson: gx: add sound card support")
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200420114511.450560-3-jbrunet@baylibre.comSigned-off-by: NMark Brown <broonie@kernel.org>

Since the addition of commit 9b5db059 ("ASoC: soc-pcm: dpcm: Only allow
playback/capture if supported"), meson-axg cards which have codec-to-codec
links fail to init and Oops:

  Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128
  Internal error: Oops: 96000044 [#1] PREEMPT SMP
  CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1
  pc : invalidate_paths_ep+0x30/0xe0
  lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
  Call trace:
   invalidate_paths_ep+0x30/0xe0
   snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8
   dpcm_path_get+0x38/0xd0
   dpcm_fe_dai_open+0x70/0x920
   snd_pcm_open_substream+0x564/0x840
   snd_pcm_open+0xfc/0x228
   snd_pcm_capture_open+0x4c/0x78
   snd_open+0xac/0x1a8
   ...

While initiliazing the links, ASoC treats the codec-to-codec links of this
card type as a DPCM backend. This error eventually leads to the Oops.

Most of the card driver code is shared between DPCM backends and
codec-to-codec links. The property "no_pcm" marking DCPM BE was left set on
codec-to-codec links, leading to this problem. This commit fixes that.

Fixes: 0a8f1117 ("ASoC: meson: axg-card: add basic codec-to-codec link support")
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.comSigned-off-by: NMark Brown <broonie@kernel.org>

TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need
yet more quirks for the proper control names.

This patch provides the mapping table for those boards, correcting the
FU names for volume and mute controls as well as the terminal names
for jack controls.  It also improves build_connector_control() not to
add the directional suffix blindly if the string is given from the
mapping table.

With this patch applied, the new UCM profiles will be effective.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

For more debug and usability information, add the entry showing the
DSD raw states and the channel mapping in each stream proc file.

Link: https://lore.kernel.org/r/20200419212134.14200-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>

The commit 3c6fd1f0 ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.

This patch reverts the corresponding entry as a temporary solution.
Although Zenith II and co will see get the empty HD-audio bus again,
it'd be merely resource wastes and won't affect the functionality,
so it's no end of the world.  We'll need to address this later,
e.g. by either switching to DMI string matching or using PCI ID &
SSID pairs.

Fixes: 3c6fd1f0 ("ALSA: hda: Add driver blacklist")
Reported-by: NJohnathan Smithinovic <johnathan.smithinovic@gmx.at>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.deSigned-off-by: NTakashi Iwai <tiwai@suse.de>