ALSA: hda/realtek - Add new codec supported for ALC245

Enable new codec supported for ALC245.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/8c0804738b2c42439f59c39c8437817f@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif

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

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

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

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

Fixes: 447d6275f0c2 ("ALSA: usb-audio: Add sanity checks for endpoint accesses")
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Add connector notifier delegation

It turned out that ALC1220-VB USB-audio device gives the interrupt
event to some PCM terminals while those don't allow the connector
state request but only the actual I/O terminals return the request.
The recent commit 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack
controls for PCM terminals") excluded those phantom terminals, so
those events are ignored, too.

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

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

Fixes: 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge tag 'asoc-fix-v5.7-rc2' of https://git./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.

ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen

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: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200421190908.462860-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: wm8960: Fix wrong clock after suspend & resume

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: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1587468525-27514-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usx2y: Fix potential NULL dereference

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: Lin Yi <teroincn@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2

Force it to use asynchronous playback.

Same quirk has already been added for Focusrite Scarlett Solo (2nd gen)
with a commit 46f5710f0b88 ("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: Gregor Pintar <grpintar@gmail.com>
Reviewed-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: wm89xx: Add missing dependency

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: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200420125343.20920-1-yuehaibing@huawei.com
Signed-off-by: Mark 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: 15a5760cb8b6d5c1ebbf1d2e1f0b77380ab68a82
--
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

ASoC: dapm: fixup dapm kcontrol widget

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: Gyeongtaek Lee <gt82.lee@samsung.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/000001d61537$b212f620$1638e260$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rsnd: Fix "status check failed" spam for multi-SSI

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: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent

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
a09fb3f28a60 ("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: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: gx-card: fix codec-to-codec link setup

Since the addition of commit 9b5db059366a ("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: e37a0c313a0f ("ASoC: meson: gx: add sound card support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200420114511.450560-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: meson: axg-card: fix codec-to-codec link setup

Since the addition of commit 9b5db059366a ("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: 0a8f1117a680 ("ASoC: meson: axg-card: add basic codec-to-codec link support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200420114511.450560-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos

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

ALSA: hda: Remove ASUS ROG Zenith from the blacklist

The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.

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: 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist")
Reported-by: Johnathan Smithinovic <johnathan.smithinovic@gmx.at>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Fix unexpected init_amp override

The commit 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping
spec->init_amp detection") changed the way to assign spec->init_amp
field that specifies the way to initialize the amp.  Along with the
change, the commit also replaced a few fixups that set spec->init_amp
in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE.  This was rather
aligning to the other fixups, and not supposed to change the actual
behavior.

However, this change turned out to cause a regression on FSC S7020,
which hit exactly the above.  The reason was that there is still one
place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE
call, namely in alc_ssid_check().

This patch fixes the regression by adding the proper spec->init_amp
override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED.

Fixes: 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329
Link: https://lore.kernel.org/r/20200418190639.10082-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices

Many Focusrite devices supports a limited set of sample rates per
altsetting. These includes audio interfaces with ADAT ports:
 - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen;
 - Scarlett 18i8 2nd gen, 18i20 2nd gen;
 - Scarlett 18i8 3rd gen, 18i20 3rd gen;
 - Clarett 2Pre USB, 4Pre USB, 8Pre USB.

Maximum rate is exposed in the last 4 bytes of Format Type descriptor
which has a non-standard bLength = 10.

Tested-by: Alexey Skobkin <skobkin-ru@ya.ru>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell

Major regressions were detected by SOF CI on CherryTrail and Broadwell:

[   25.705750]  SSP2-Codec: ASoC: no backend playback stream
[   27.923378]  SSP2-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.

For some reason the SSP structures used for SKL+ did provide this
information but legacy platforms didn't.

Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
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/20200417172014.11760-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: stm32: sai: fix sai probe

pcm config must be set before snd_dmaengine_pcm_register() call.

Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue")
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200417142122.10212-1-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda/hdmi: Add module option to disable audio component binding

As the recent regression showed, we want sometimes to turn off the
audio component binding just for debugging.  This patch adds the
module option to control it easily without compilation.

Fixes: ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200415162523.27499-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge series "ASoC: rsnd: Fixes for multichannel HDMI audio output" from Matthias Blankertz <matthias.blankertz@cetitec.com>:

This fixes two issues in the snd-soc-rcar driver blocking multichannel
HDMI audio out: The parent SSI in a multi-SSI configuration is not
correctly set up and started, and the SSI->HDMI channel mapping is
wrong.

With these patches, the following device tree snippet can be used on an
r8a7795-based platform (Salvator-X) to enable multichannel HDMI audio on
HDMI0:

rsnd_port1: port@1 {
rsnd_endpoint1: endpoint {
remote-endpoint = <&dw_hdmi0_snd_in>;

dai-format = "i2s";
bitclock-master = <&rsnd_endpoint1>;
frame-master = <&rsnd_endpoint1>;

playback = <&ssi0 &ssi1 &ssi2 &ssi9>;
};
};

With a capable receiver attached, all of 2ch (stereo), 6ch (e.g. 5.1)
and 8ch audio output should work.

Matthias Blankertz (2):
  ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode
  ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode

 sound/soc/sh/rcar/ssi.c  | 8 ++++----
 sound/soc/sh/rcar/ssiu.c | 2 +-
 2 files changed, 5 insertions(+), 5 deletions(-)

base-commit: 7111951b8d4973bda27ff663f2cf18b663d15b48
--
2.26.0

ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry

If we don't find any pcm, pcm will point at address at an offset from
the the list head and not a meaningful structure. Fix this by returning
correct pcm if found and NULL if not. Found with coccinelle.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode

The HDMI?_SEL register maps up to four stereo SSI data lanes onto the
sdata[0..3] inputs of the HDMI output block. The upper half of the
register contains four blocks of 4 bits, with the most significant
controlling the sdata3 line and the least significant the sdata0 line.

The shift calculation has an off-by-one error, causing the parent SSI to
be mapped to sdata3, the first multi-SSI child to sdata0 and so forth.
As the parent SSI transmits the stereo L/R channels, and the HDMI core
expects it on the sdata0 line, this causes no audio to be output when
playing stereo audio on a multichannel capable HDMI out, and
multichannel audio has permutated channels.

Fix the shift calculation to map the parent SSI to sdata0, the first
child to sdata1 etc.

Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode

The parent SSI of a multi-SSI setup must be fully setup, started and
stopped since it is also part of the playback/capture setup. So only
skip the SSI (as per commit 203cdf51f288 ("ASoC: rsnd: SSI parent cares
SWSP bit") and commit 597b046f0d99 ("ASoC: rsnd: control SSICR::EN
correctly")) if the SSI is parent outside of a multi-SSI setup.

Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence

On Baytrail/Cherrytrail, the Atom/SST driver fails miserably:

[    9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01
[    9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034
[    9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4
[    9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5
[    9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5
[    9.833037]  Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed
[    9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034
[    9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4
[    9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5
[    9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5
[    9.853994]  Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed

Commit b56be800f1292 ("ASoC: soc-pcm: call
snd_soc_dai_startup()/shutdown() once") was the initial problematic
commit.

Commit 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence")
was an attempt to fix things but it does not work on Baytrail,
reverting all changes seems necessary for now.

Fixes: 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200415030437.23803-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: Fix endianness issue

As done in already existing cases, we should use le32_to_cpu macro while
accessing hdr->magic. Found with sparse.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162435.31859-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs

For some reason, the MI2S DAIs do not have channels_min/max defined.
This means that snd_soc_dai_stream_valid() returns false,
i.e. the DAIs have neither valid playback nor capture stream.

It's quite surprising that this ever worked correctly,
but in 5.7-rc1 this is now failing badly: :)

Commit 0e9cf4c452ad ("ASoC: pcm: check if cpu-dai supports a given stream")
introduced a check for snd_soc_dai_stream_valid() before calling
hw_params(), which means that the q6i2s_hw_params() function
was never called, eventually resulting in:

    qcom-q6afe aprsvc:q6afe:4:4: no line is assigned

... even though "qcom,sd-lines" is set in the device tree.

Commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported")
now even avoids creating PCM devices if the stream is not supported,
which means that it is failing even earlier with e.g.:

    Primary MI2S: ASoC: no backend playback stream

Avoid all that trouble by adding channels_min/max for the MI2S DAIs.

Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200415150050.616392-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-pcm: dpcm: Only allow playback/capture if supported

At the moment, PCM devices for DPCM are only created based on the
dpcm_playback/capture parameters of the DAI link, without considering
if the CPU/FE DAI is actually capable of playback/capture.

Normally the dpcm_playback/capture parameter should match the
capabilities of the CPU DAI. However, there is no way to set that
parameter from the device tree (e.g. with simple-audio-card or
qcom sound cards). dpcm_playback/capture are always both set to 1.

This causes problems when the CPU DAI does only support playback
or capture. Attemting to open that PCM device with an unsupported
stream type then results in a null pointer dereference:

    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
     ...

... because the DAI playback/capture_widget is not set in that case.

We could add checks there to fix the problem (maybe we should
anyway), but much easier is to not expose the device as
playback/capture in the first place. Attemting to use that
device would always fail later anyway.

Add checks for snd_soc_dai_stream_valid() to the DPCM case
to avoid exposing playback/capture if it is not supported.

Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: sgtl5000: Fix VAG power-on handling

As mentioned slightly out of patch context in the code, there
is no reset routine for the chip. On boards where the chip is
supplied by a fixed regulator, it might not even be resetted
during (e.g. watchdog) reboot and can be in any state.

If the device is probed with VAG enabled, the driver's probe
routine will generate a loud pop sound when ANA_POWER is
being programmed. Avoid this by properly disabling just the
VAG bit and waiting the required power down time.

Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.com>
Reviewed-by: Fabio Estevam <festivem@gmail.com>
Link: https://lore.kernel.org/r/20200414181140.145825-1-sebastian.reichel@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda: call runtime_allow() for all hda controllers

Before the pci_driver->probe() is called, the pci subsystem calls
runtime_forbid() and runtime_get_sync() on this pci dev, so only call
runtime_put_autosuspend() is not enough to enable the runtime_pm on
this device.

For controllers with vgaswitcheroo feature, the pci/quirks.c will call
runtime_allow() for this dev, then the controllers could enter
rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like
Intel hda controllers, the runtime_pm is not enabled because the
runtime_allow() is not called.

Since it is no harm calling runtime_allow() twice, here let hda
driver call runtime_allow() for all controllers. Then the runtime_pm
is enabled on all controllers after the put_autosuspend() is called.

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: rockchip-i2s: add power-domains property

In the old txt situation we add/describe only properties that are used
by the driver/hardware itself. With yaml it also filters things in a
node that are used by other drivers like 'power-domains' for rk3399,
so add it to 'rockchip-i2s.yaml'.

Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200324094149.6904-3-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: madera: Remove a couple of stray blank lines

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200409181311.30247-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: wsa881x: mark read_only_wordlength flag

WSA881x works in PDM mode so the wordlength is fixed, which also makes
the only field "WordLength" in DPN_BlockCtrl1 register a read-only.
Writing to this register will throw up errors with Qualcomm Controller.
So use ro_blockctrl1_reg flag to mark this field as read-only so that
core will not write to this register.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: qcom: common: Silence duplicate parse error messages

All error paths in qcom_snd_parse_of() prints more specific error
messages, so silence the one in apq8096_platform_probe() and
sdm845_snd_platform_probe() to avoid spamming the kernel log.

Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org>
Link: https://lore.kernel.org/r/20200406003229.2354631-1-bjorn.andersson@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>

soc/stm/stm32_sub_sai: Add missing '\n' in log messages

Message logged by 'dev_xxx()' or 'pr_xxx()' should end with a '\n'.

Fixes: 3e086ed("ASoC: stm32: add SAI drivers")
Signed-off-by: Sebastian Fricke <sebastian.fricke.linux@gmail.com>
Link: https://lore.kernel.org/r/20200413042952.7675-1-sebastian.fricke.linux@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dapm: Remove dapm_connect_dai_link_widgets helper

This helper is adding very little both it and is one caller are very
small functions simply combine the two.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200409181209.30130-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dapm: Move error message to avoid some duplication

Move the error message into snd_soc_dapm_new_dai from
dapm_connect_dai_pair, since the two copies are almost identical and
are the only callers.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200409181209.30130-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rockchip-spdif: add power-domains property

In the old txt situation we add/describe only properties that are used
by the driver/hardware itself. With yaml it also filters things in a
node that are used by other drivers like 'power-domains' for rk3399,
so add it to 'rockchip-spdif.yaml'.

Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200404115225.4314-3-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rockchip-spdif: add #sound-dai-cells property

'#sound-dai-cells' is required to properly interpret
the list of DAI specified in the 'sound-dai' property,
so add them to 'rockchip-spdif.yaml'

Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200404115225.4314-2-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: convert rockchip spdif bindings to yaml

Current dts files with 'spdif' nodes are manually verified.
In order to automate this process rockchip-spdif.txt
has to be converted to yaml.

Also rk3188.dtsi, rk3288.dtsi use an extra fallback string,
so change this in the documentation.

Changed:
"rockchip,rk3188-spdif", "rockchip,rk3066-spdif"
"rockchip,rk3288-spdif", "rockchip,rk3066-spdif"

Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200404115225.4314-1-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: soc-acpi-intel-cml-match: remove useless 'rt1308_2_adr'

Fix the following gcc warning:

sound/soc/intel/common/soc-acpi-intel-cml-match.c:116:45: warning:
‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=]
 static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
                                             ^~~~~~~~~~~~

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Jason Yan <yanaijie@huawei.com>
Link: https://lore.kernel.org/r/20200410081117.21319-2-yanaijie@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: intel: soc-acpi-intel-icl-match: remove useless 'rt1308_2_adr'

Fix the following gcc warning:

sound/soc/intel/common/soc-acpi-intel-icl-match.c:90:45: warning:
‘rt1308_2_adr’ defined but not used [-Wunused-const-variable=]
 static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
                                             ^~~~~~~~~~~~

Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Jason Yan <yanaijie@huawei.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200410081117.21319-1-yanaijie@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: amd: Fix button configuration

RT5682 buttons were incorrectly mapped.

Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Link: https://lore.kernel.org/r/20200414113527.13532-1-akshu.agrawal@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: tas571x: disable regulators on failed probe

If probe fails after enabling the regulators regulator_put is called for
each supply without having them disabled before. This produces some
warnings like

WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c
[<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14)
[<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4)
[<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4)
[<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c)
[<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38)
[<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38)
[<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c)
[<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c)
[<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c)
[<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60)
[<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc)
[<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0)
[<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0)
[<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108)
[<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88)
[<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250)
[<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244)
[<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540)
[<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8)
[<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20)

Fixes: 3fd6e7d9a146 (ASoC: tas571x: New driver for TI TAS571x power amplifiers)
Signed-off-by: Philipp Puschmann <p.puschmann@pironex.de>
Link: https://lore.kernel.org/r/20200414112754.3365406-1-p.puschmann@pironex.de
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dapm: Fix regression introducing multiple copies of DAI widgets

Refactoring was done to factor out the linking of DAI widgets into
a helper function, dapm_add_valid_dai_widget. However when this was
done, a regression was introduced for CODEC to CODEC links. It was
over looked that the playback and capture variables persisted across
all CODEC DAIs being processed, which ensured that the special DAI
widget that is added for CODEC to CODEC links was only created once.
This bug causes kernel panics during DAPM shutdown.

To stick with the spirit of the original refactoring whilst fixing the
issue, variables to hold the DAI widgets are added to snd_soc_dai_link.
Furthermore the dapm_add_valid_dai_widget function is renamed to
dapm_connect_dai_pair, the function only adds DAI widgets in the CODEC
to CODEC case and its primary job is to add routes connecting two DAI
widgets, making the original name quite misleading.

Fixes: 6c4b13b51aa3 ("ASoC: Add dapm_add_valid_dai_widget helper")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200409181209.30130-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume

Commit 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume")
removed the DAI side suspend/resume hooks and switched entirely to
component suspend/resume.  However the Samsung SoC s3c-i2s-v2 driver was
not updated.

Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while
changing dai to component which allows to keep the struct
snd_soc_component_driver const.

This fixes build errors:

    sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’:
    sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’
      dai_drv->suspend = s3c2412_i2s_suspend;

Reported-by: Arnd Bergmann <arnd@arndb.de>
Fixes: 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume")
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20200413124548.28197-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda: Allow setting preallocation again for x86

The commit c31427d0d21e ("ALSA: hda: No preallocation on x86
platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default
to zero for x86, as the preallocation should work almost all cases.
However, this expectation was too naive; some applications try to
allocate as the max buffer size as possible, and it leads to the
memory exhaustion.  More badly, the commit changed the kconfig no
longer adjustable for x86, so you can't fix it statically (although it
can be still adjusted via procfs).

So, practically seen, it's more recommended to set a reasonable limit
for x86, too.  This patch follows to that experience, and changes the
default to 2048 and allow the kconfig adjustable again.

Fixes: c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223
Link: https://lore.kernel.org/r/20200413201919.24241-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported

This fixes runtime PM not working after a suspend-to-RAM cycle at least for
the codec-less HDA device found on NVIDIA GPUs.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Signed-off-by: Roy Spliet <nouveau@spliet.org>
Link: https://lore.kernel.org/r/20200413082034.25166-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Skip controller resume if not needed

The HD-audio controller does system-suspend and resume operations by
directly calling its helpers __azx_runtime_suspend() and
__azx_runtime_resume().  However, in general, we don't have to resume
always the device fully at the system resume; typically, if a device
has been runtime-suspended, we can leave it to runtime resume.

Usually for achieving this, the driver would call
pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the
system suspend and resume ops.  Unfortunately, this doesn't work for
the resume path in our case.  For handling the jack detection at the
system resume, a child codec device may need the (literally) forcibly
resume even if it's been runtime-suspended, and for that, the
controller device must be also resumed even if it's been suspended.

This patch is an attempt to improve the situation.  It replaces the
direct __azx_runtime_suspend()/_resume() calls with with
pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight
trick as we've done for the codec side.  More exactly:

- azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and
  azx_runtime_resume(), so that it can be properly executed from the
  system-suspend/resume path

- The WAKEEN handling depends on the card's power state now; it's set
  and cleared only for the runtime-suspend

- azx_resume() checks whether any codec may need the forcible resume
  beforehand.  If the forcible resume is required, it does temporary
  PM refcount up/down for actually triggering the runtime resume.

- A new helper function, hda_codec_need_resume(), is introduced for
  checking whether the codec needs a forcible runtime-resume, and the
  existing code is rewritten with that.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Keep the controller initialization even if no codecs found

Currently, when the HD-audio controller driver doesn't detect any
codecs, it tries to abort the probe.  But this abort happens at the
delayed probe, i.e. the primary probe call already returned success,
hence the driver is never unbound until user does so explicitly.
As a result, it may leave the HD-audio device in the running state
without the runtime PM.  More badly, if the device is a HD-audio bus
that is tied with a GPU, GPU cannot reach to the full power down and
consumes unnecessarily much power.

This patch changes the logic after no-codec situation; it continues
probing without the further codec initialization but keep the
controller driver running normally.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Tested-by: Roy Spliet <nouveau@spliet.org>
Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Release resources at error in delayed probe

snd-hda-intel driver handles the most of its probe task in the delayed
work (either via workqueue or via firmware loader).  When an error
happens in the later delayed probe, we can't deregister the device
itself because the probe callback already returned success and the
device was bound.  So, for now, we set hda->init_failed flag and make
the rest untouched until the device gets really unbound.
However, this leaves the device up running, keeping the resources
without any use that prevents other operations.

In this patch, we release the resources at first when a probe error
happens in the delayed probe stage, but keeps the top-level object, so
that the PM and other ops can still refer to the object itself.

Also for simplicity, snd_hda_intel object is allocated via devm, so
that we can get rid of the explicit kfree calls.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq ops

freeze_noirq and thaw_noirq need to check the PM availability like
other PM ops.  There are cases where the device got disabled due to
the error, and the PM operation should be ignored for that.

Fixes: 3e6db33aaf1d ("ALSA: hda - Set SKL+ hda controller power at freeze() and thaw()")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Don't release card at firmware loading error

At the error path of the firmware loading error, the driver tries to
release the card object and set NULL to drvdata.  This may be referred
badly at the possible PM action, as the driver itself is still bound
and the PM callbacks read the card object.

Instead, we continue the probing as if it were no option set.  This is
often a better choice than the forced abort, too.

Fixes: 5cb543dba986 ("ALSA: hda - Deferred probing with request_firmware_nowait()")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Check mapping at creating connector controls, too

Add the mapping check to build_connector_control() so that the device
specific quirk can provide the node to skip for the badly behaving
connector controls.  As an example, ALC1220-VB-based codec implements
the skip entry for the broken SPDIF connector detection.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Don't create jack controls for PCM terminals

Some funky firmwares set the connector flag even on PCM terminals
although it doesn't make sense (and even actually the firmware doesn't
react properly!).  Let's skip creation of jack controls in such a
case.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Don't override ignore_ctl_error value from the map

The mapping table may contain also ignore_ctl_error flag for devices
that are known to behave wild.  Since this flag always writes the
card's own ignore_ctl_error flag, it overrides the value already set
by the module option, so it doesn't follow user's expectation.
Let's fix the code not to clear the flag that has been set by user.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Filter error from connector kctl ops, too

The ignore_ctl_error option should filter the error at kctl accesses,
but there was an overlook: mixer_ctl_connector_get() returns an error
from the request.

This patch covers the forgotten code path and apply filter_error()
properly.  The locking error is still returned since this is a fatal
error that has to be reported even with ignore_ctl_error option.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Enable the headset mic on Asus FX505DT

On Asus FX505DT with Realtek ALC233, the headset mic is connected
to pin 0x19, with default 0x411111f0.

Enable headset mic by reconfiguring the pin to an external mic
associated with the headphone on 0x21. Mic jack detection was also
found to be working.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207131
Signed-off-by: Adam Barber <barberadam995@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200410090032.2759-1-barberadam995@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ctxfi: Remove unnecessary cast in kfree

Remove unnecassary casts in the argument to kfree.

Signed-off-by: Xu Wang <vulab@iscas.ac.cn>
Link: https://lore.kernel.org/r/20200409112052.13402-1-vulab@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: topology: Check return value of soc_tplg_dai_config

Function soc_tplg_dai_config can fail, check for and handle possible
failure.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-7-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: Check return value of pcm_new_ver

Function pcm_new_ver can fail, so we should check it's return value and
handle possible error.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: Check soc_tplg_add_route return value

Function soc_tplg_add_route can propagate error code from callback, we
should check its return value and handle fail in correct way.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: Check return value of soc_tplg_*_create

Functions soc_tplg_denum_create, soc_tplg_dmixer_create,
soc_tplg_dbytes_create can fail, so their return values should be
checked and error should be propagated.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: Check return value of soc_tplg_create_tlv

Function soc_tplg_create_tlv can fail, so we should check if it succeded
or not and proceed appropriately.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: topology: Add missing memory checks

kstrdup is an allocation function and it can fail, so its return value
should be checked and handled appropriately.

In order to check all cases, we need to modify set_stream_info to return
a value, so check that everything went correctly when doing kstrdup().
Later add proper checks and error handlers.

Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200327204729.397-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

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

ASoC: Fixes for v5.7

A collection of fixes that have been accumilated since the merge window,
mainly relating to x86 platform support.

ALSA: hda: Add driver blacklist

The recent AMD platform exposes an HD-audio bus but without any actual
codecs, which is internally tied with a USB-audio device, supposedly.
It results in "no codecs" error of HD-audio bus driver, and it's
nothing but a waste of resources.

This patch introduces a static blacklist table for skipping such a
known bogus PCI SSID entry.  As of writing this patch, the known SSIDs
are:
* 1043:874f - ASUS ROG Zenith II / Strix
* 1462:cb59 - MSI TRX40 Creator
* 1462:cb60 - MSI TRX40

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Add mixer workaround for TRX40 and co

Some recent boards (supposedly with a new AMD platform) contain the
USB audio class 2 device that is often tied with HD-audio.  The device
exposes an Input Gain Pad control (id=19, control=12) but this node
doesn't behave correctly, returning an error for each inquiry of
GET_MIN and GET_MAX that should have been mandatory.

As a workaround, simply ignore this node by adding a usbmix_name_map
table entry.  The currently known devices are:
* 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi
* 0b05:1916 - ASUS ROG Zenith II
* 0b05:1917 - ASUS ROG Strix
* 0db0:0d64 - MSI TRX40 Creator
* 0db0:543d - MSI TRX40

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Add quirk for MSI GL63

MSI GL63 laptop requires the similar quirk like other MSI models,
ALC1220_FIXUP_CLEVO_P950.  The board BIOS doesn't provide a PCI SSID
for the device, hence we need to take the codec SSID (1462:1275)
instead.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ice1724: Fix invalid access for enumerated ctl items

The access to Analog Capture Source control value implemented in
prodigy_hifi.c is wrong, as caught by the recently introduced sanity
check; it should be accessing value.enumerated.item[] instead of
value.integer.value[].  This patch corrects the wrong access pattern.

Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: Fix potential access overflow in beep helper

The beep control helper function blindly stores the values in two
stereo channels no matter whether the actual control is mono or
stereo.  This is practically harmless, but it annoys the recently
introduced sanity check, resulting in an error when the checker is
enabled.

This patch corrects the behavior to store only on the defined array
member.

Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: cs4270: pull reset GPIO low then high

Pull the RST line low then high when initializing the driver,
in order to force a reset of the chip.
Previously, the line was not pulled low, which could result in
the chip registers not resetting to their default values on boot.

Signed-off-by: Mike Willard <mwillard@izotope.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200401205454.79792-1-mwillard@izotope.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: hda/realtek - Add HP new mute led supported for ALC236

HP new platform has new mute led feature.
COEF index 0x34 bit 5 to control playback mute led.
COEF index 0x35 bit 2 and bit 3 to control Mic mute led.

[ corrected typos by tiwai ]

Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Add supported new mute Led for HP

HP Note Book supported new mute Led.
Hardware PIN was not enough to meet old LED rule.
JD2 to control playback mute led.
GPO3 to control capture mute led.
(ALC285 didn't control GPO3 via verb command)
This two PIN just could control by COEF registers.

[ corrected typos by tiwai ]

Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: rt5645: Add platform-data for Medion E1239T

The Medion E1239T 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 other models following Intel's Brasswell's reference design.

This commit adds a DMI quirk pointing to the intel_braswell_platform_data
for this model.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200402185257.3355-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet

The MPMAN MPWIN895CL tablet almost fully works with out default settings.
The only problem is that it has only 1 speaker so any sounds only playing
on the right channel get lost.

Add a quirk for this model using the default settings + MONO_SPEAKER.

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/20200405133726.24154-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: stm32: sai: Add missing cleanup

The commit 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue")
converts some function calls to their non-devm equivalents.  The
appropriate cleanup code was added to the remove function, but not
to the probe function.  Add a call to snd_dmaengine_pcm_unregister
to compensate for the call to snd_dmaengine_pcm_register in case
of subsequent failure.

Fixes: commit 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue")
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Acked-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/1586099028-5104-1-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>

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

Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:16d8) 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: Emmanuel Pescosta <emmanuelpescosta099@gmail.com>
Link: https://lore.kernel.org/r/20200404153843.9288-1-emmanuelpescosta099@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: Intel: atom: Fix uninitialized variable compiler warning

GCC 10 gives a "variable might be used uninitialized" warning for the
block variable in sst_prepare_and_post_msg().

This is a false-positive warning, but lets fix it anyways.

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/20200402185359.3424-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: atom: Check drv->lock is locked in sst_fill_and_send_cmd_unlocked

sst_fill_and_send_cmd_unlocked must be called with the drv->lock mutex
locked already. In the past there have been cases where this was not the
case, add a WARN_ON to check for drv->lock being locked.

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/20200402185359.3424-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()

sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some
places it is called with the drv->lock mutex already held.

So it must always be called with the mutex locked. This commit adds missing
locking in the sst_set_be_modules() code-path.

Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls")
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/20200402185359.3424-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Turn "firmware boot complete" message into a dbg message

Using a Canon Lake machine with the SOF driver causes dmesg to fill
up with a ton of these messages:

[  275.902194] sof-audio-pci 0000:00:1f.3: firmware boot complete
[  351.529358] sof-audio-pci 0000:00:1f.3: firmware boot complete
[  560.049047] sof-audio-pci 0000:00:1f.3: firmware boot complete
etc.

Since the DSP is powered down when not in used this happens everytime
e.g. a notification plays, polluting dmesg.

Turn this messages into a debug message, matching what the code already
does for the ""booting DSP firmware" message.

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/20200402184948.3014-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk

Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card.
The MIDI controller part is standard but the PCM part is "vendor specific".
Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE.
Input is not working.

Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/pcm-oss-fix' into for-linus

An empty merge for the original fix for PCM OSS regression where the
same fix is already applied in a different form.

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

ALSA: pcm: oss: Fix regression by buffer overflow fix (again)

[ This is essentially the same fix as commit ae769d355664, but it's
  adapted to the latest code for 5.7; hence it contains no Fixes or
  other tags for avoid backport confusion -- tiwai ]

The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef06: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a
regression on OSS applications.  The patch introduced the size check
in client and slave size calculations to limit to each plugin's buffer
size, but I overlooked that some code paths call those without
allocating the buffer but just for estimation.

This patch fixes the bug by skipping the size check for those code
paths while keeping checking in the actual transfer calls.

Link: https://lore.kernel.org/r/20200403073818.27943-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: pcm: oss: Fix regression by buffer overflow fix

The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef06: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a
regression on OSS applications.  The patch introduced the size check
in client and slave size calculations to limit to each plugin's buffer
size, but I overlooked that some code paths call those without
allocating the buffer but just for estimation.

This patch fixes the bug by skipping the size check for those code
paths while keeping checking in the actual transfer calls.

Fixes: f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow")
Tested-and-reported-by: Jari Ruusu <jari.ruusu@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200403072515.25539-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

edd: Use scnprintf() for avoiding potential buffer overflow

Since snprintf() returns the would-be-output size instead of the
actual output size, the succeeding calls may go beyond the given
buffer limit.  Fix it by replacing with scnprintf().

Link: https://lore.kernel.org/r/20200320084429.1803-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen

The audio setup on the Lenovo Carbon X1 8th gen is the same as that on
the Lenovo Carbon X1 7th gen, as such it needs the same
ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk.

This fixes volume control of the speaker not working among other things.

BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196
Cc: stable@vger.kernel.org
Suggested-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: topology: use name_prefix for new kcontrol

Current topology doesn't add prefix of component to new kcontrol.

Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/009b01d60804$ae25c2d0$0a714870$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: rt5682: Fix build error without CONFIG_I2C

If I2C is n but SoundWire is m, building fails:

sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class
 module_i2c_driver(rt5682_i2c_driver);
 ^~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5682.c:3716:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration

Guard this use #ifdef CONFIG_I2C.

Fixes: 5549ea647997 ("ASoC: rt5682: fix unmet dependencies")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200401091055.34112-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dpcm: allow start or stop during pause for backend

soc_compr_trigger_fe() allows start or stop after pause_push.
In dpcm_be_dai_trigger(), however, only pause_release is allowed
command after pause_push.
So, start or stop after pause in compress offload is always
returned as error if the compress offload is used with dpcm.
To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed
for start or stop command.

Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: dapm: connect virtual mux with default value

Since a virtual mixer has no backing registers
to decide which path to connect,
it will try to match with initial state.
This is to ensure that the default mixer choice will be
correctly powered up during initialization.
Invert flag is used to select initial state of the virtual switch.
Since actual hardware can't be disconnected by virtual switch,
connected is better choice as initial state in many cases.

Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/01a301d60731$b724ea10$256ebe30$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag

At the moment, playing audio with PulseAudio with the qdsp6 driver
results in distorted sound. It seems like its timer-based scheduling
does not work properly with qdsp6 since setting tsched=0 in
the PulseAudio configuration avoids the issue.

Apparently this happens when the pointer() callback is not accurate
enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop
PulseAudio from using timer-based scheduling by default.

According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html:

    The flag is being used in the sense explained in the previous audio
    meeting -- the data transfer granularity isn't fine enough but aligned
    to the period size (or less).

q6asm-dai reports the position as multiple of

    prtd->pcm_count = snd_pcm_lib_period_bytes(substream)

so it indeed just a multiple of the period size.

Therefore adding the flag here seems appropriate and makes audio
work out of the box.

Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>

ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor

The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra
endpoint descriptor.

The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00

As the code in snd_usbmidi_get_ms_info() looks only at the
first extra descriptor to find USB_DT_CS_ENDPOINT the device
as such is recognized but there is neither input nor output
configured.

The patch iterates through the extra descriptors to find the
proper one. With this patch the device is correctly configured.

Signed-off-by: Andreas Steinmetz <ast@domdv.de>
Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@domdv.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"

This reverts commit 645c08f17f477915f6d900b767e789852f150054
which was reported to break the build a program using this header.

The original issue was addressed in the alsa-lib side recently, so we
can make the header more self-contained again.

Reported-by: Dmitry V. Levin <ldv@altlinux.org>
Fixes: 645c08f17f47 ("ALSA: uapi: Drop asound.h inclusion from asoc.h")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200331090023.8112-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups

patch_realtek.c has historically failed to properly configure the PC
Beep Hidden Register for the ALC256 codec (among others). Depending on
your kernel version, symptoms of this misconfiguration can range from
chassis noise, picked up by a poorly-shielded PCBEEP trace, getting
amplified and played on your internal speaker and/or headphones to loud
feedback, which responds to the "Headphone Mic Boost" ALSA control,
getting played through your headphones. For details of the problem, see
the patch in this series titled "ALSA: hda/realtek - Set principled PC
Beep configuration for ALC256", which fixes the configuration.

These symptoms have been most noticed on the Dell XPS 13 9350 and 9360,
popular laptops that use the ALC256. As a result, several model-specific
fixups have been introduced to try and fix the problem, the most
egregious of which locks the "Headphone Mic Boost" control as a hack to
minimize noise from a feedback loop that shouldn't have been there in
the first place.

Now that the underlying issue has been fixed, remove all these fixups.
Remaining fixups needed by the XPS 13 are all picked up by existing pin
quirks.

This change should, for the XPS 13 9350/9360

 - Significantly increase volume and audio quality on headphones
 - Eliminate headphone popping on suspend/resume
 - Allow "Headphone Mic Boost" to be set again, making the headphone
   jack fully usable as a microphone jack too.

Fixes: 8c69729b4439 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3")
Fixes: 423cd785619a ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360")
Fixes: e4c9fd10eb21 ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant")
Fixes: 1099f48457d0 ("ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/b649a00edfde150cf6eebbb4390e15e0c2deb39a.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda/realtek - Set principled PC Beep configuration for ALC256

The Realtek PC Beep Hidden Register[1] is currently set by
patch_realtek.c in two different places:

In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to
non-beep input on 1Ah and no 1Ah loopback to either headphones or
speakers. (Although, curiously, the loopback amp is still enabled.) This
write was added fairly recently by commit e3743f431143 ("ALSA:
hda/realtek - Dell headphone has noise on unmute for ALC236") and is a
safe default. However, it happens in the wrong place:
alc_fill_eapd_coef() runs on module load and cold boot but not on S3
resume, meaning the register loses its value after suspend.

Conversely, in alc256_init(), the register is updated to unset bit 13
(disable speaker loopback) and set bit 5 (set non-beep input on 1Ah).
Although this write does run on S3 resume, it's not quite enough to fix
up the register's default value of 0x3717. What's missing is a set of
bit 14 to disable headphone loopback. Without that, we end up with a
feedback loop where the headphone jack is being driven by amplified
samples of itself[2].

This change eliminates the update in alc256_init() and replaces it with
the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is
supposed to be the codec's default value, so using it will make
debugging easier for Realtek.

Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs.

[1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst

[2] Setting the "Headphone Mic Boost" control from userspace changes
this feedback loop and has been a widely-shared workaround for headphone
noise on laptops like the Dell XPS 13 9350. This commit eliminates the
feedback loop and makes the workaround unnecessary.

Fixes: e1e8c1fdce8b ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>