ALSA: usb-audio: Print more information in stream proc files

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

ALSA: Fix misspellings of "Analog Devices"

According to https://www.analog.com/, the company name is spelled
"Analog Devices".

Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20200416103058.15269-6-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: RME Babyface Pro mixer patch

Added mixer quirks to allow controlling the internal DSP of the
RME Babyface Pro and its successor Babyface Pro FS.

Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Link: https://lore.kernel.org/r/20200414211019.qprg7whepg2y7nei@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'topic/for-5.8' into for-next

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: add autodetection for SoundWire

When an ACPI companion device is present and the SoundWire link mask
information is available, use SoundWire instead of legacy HDA or
Skylake drivers.

The SOF driver is selected when SoundWire or DMIC are detected. There
is no precedence at this level. In the SOF driver proper, SoundWire
will be handled first since it is mutually exclusive with HDaudio.

Known devices with an existing DMI quirk bypass this detection to
avoid any dependency on ACPI/DSDT tables.

Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200409190251.16569-1-pierre-louis.bossart@linux.intel.com
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>

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>

ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256

This codec (among others) has a hidden set of audio routes, apparently
designed to allow PC Beep output without a mixer widget on the output
path, which are controlled by an undocumented Realtek vendor register.
The default configuration of these routes means that certain inputs
aren't accessible, necessitating driver control of the register.
However, Realtek has provided no documentation of the register, instead
opting to fix issues by providing magic numbers, most of which have been
at least somewhat erroneous. These magic numbers then get copied by
others into model-specific fixups, leading to a fragmented and buggy set
of configurations.

To get out of this situation, I've reverse engineered the register by
flipping bits and observing how the codec's behavior changes. This
commit documents my findings. It does not change any code.

Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/bd69dfdeaf40ff31c4b7b797c829bb320031739c.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge series "ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai link" from Cezary Rojewski <cezary.rojewski@intel.com>:

As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend

function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.

Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54

Cezary Rojewski (4):
  ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link
  ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link
  ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link
  ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link

 sound/soc/intel/boards/bdw-rt5650.c | 1 -
 sound/soc/intel/boards/bdw-rt5677.c | 1 -
 sound/soc/intel/boards/broadwell.c  | 1 -
 sound/soc/intel/boards/haswell.c    | 1 -
 4 files changed, 4 deletions(-)

--
2.17.1

ASoC: soc-dai: fix DAI startup/shutdown sequence

The addition of a single flag to track the DAI status prevents the DAI
startup sequence from being called on capture if the DAI is already
used for playback.

Fix by extending the existing code with one flag per direction.

Fixes: b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once")
Reported-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200330160602.10180-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: fix regwmask

If regwshift is 32 and the selected architecture compiles '<<' operator
for signed int literal into rotating shift, '1<<regwshift' became 1 and
it makes regwmask to 0x0.
The literal is set to unsigned long to get intended regwmask.

Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/001001d60665$db7af3e0$9270dba0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link

As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend

function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Dominik Brodowski <linux@dominikbrodowski.net>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200319204947.18963-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link

As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend

function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Dominik Brodowski <linux@dominikbrodowski.net>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200319204947.18963-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link

As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend

function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Dominik Brodowski <linux@dominikbrodowski.net>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20200319204947.18963-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link

As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend

function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.

Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54

Reported-by: Dominik Brodowski <linux@dominikbrodowski.net>
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200319204947.18963-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: AMD: Clear format bits before setting them

This avoids residual bit form previous format when the format is changed.
Hence, the resultant format is not an invalid one.

Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200328093921.32211-1-akshu.agrawal@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: bcm: Fix pointer cast warning

The NULL check can be done gracefully without cast.  It fixes a
compile warning like:
  sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast]

Fixes: 88eb404ccc3e ("ASoC: brcm: Add DSL/PON SoC audio driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200330135645.9707-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>

dt-bindings: sound: cs42l51: Remove unneeded I2C unit name

The following warning is seen with 'make dt_binding_check':

Documentation/devicetree/bindings/sound/cirrus,cs42l51.example.dts:18.15-34.11: Warning (unit_address_vs_reg): /example-0/i2c@0: node has a unit name, but no reg or ranges property

Fix it by removing the unneeded i2c unit name.

Signed-off-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200327155721.7596-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>

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

ASoC: Updates for v5.7

This is a very big update for the core since Morimoto-san has been
rather busy continuing his refactorings to clean up a lot of the cruft
that we have accumilated over the years.  We've also gained several new
drivers, including initial (but still not complete) parts of the Intel
SoundWire support.

 - Lots of refactorings to modernize the code from Morimoto-san.
 - Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven.
 - Continued refactoring and fixing of the Intel support.
 - Soundwire and more advanced clocking support for Realtek RT5682.
 - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
   DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and
   TLV320ADCX140.

Merge branch 'for-next' into for-linus

ALSA: hda/realtek - a fake key event is triggered by running shutup

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume()
and rt_suspend() of the codec driver will be called periodically, the
driver can't stay in the rt_suspend state even users doen't use the
sound card.

Through debugging, I found  when running rt_suspend(), it will call
alc225_shutup(), in this function, it will change 3k pull down control
by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a
fake key event and that event will resume the codec, when codec
suspend agin, it will trigger the fake key event one more time, this
process will repeat.

If disable the key event before changing the pull down control, it
will not trigger fake key event. It also needs to restore the pull
down control and re-enable the key event, otherwise the system can't
get key event when codec is in rt_suspend state.

Also move some functions ahead of alc225_shutup(), this can save the
function declaration.

Fixes: 76f7dec08fd6 (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1)
Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200329082018.20486-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda: default enable CA0132 DSP support

If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as
well. Disabled DSP support leads to a hanging alsa system and no sound
output on the card otherwise. Tested on:

  06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01)

Signed-off-by: Rouven Czerwinski <rouven@czerwinskis.de>
Link: https://lore.kernel.org/r/20200329053710.4276-1-r.czerwinski@pengutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'asoc-5.7' into asoc-next

Merge branch 'asoc-5.6' into asoc-linus

Merge series "ASoC: Intel: add SoundWire machine driver" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

To handle multiple hardware combinations, this patchset suggests a
single machine driver which will create and initialize dailinks
dynamically. This allows us to support new configurations easily, as
shown with the TigerLake rt5682 example.

Each configuration updates the card component string, and UCM can test
for the presence of components to configure them as needed.

Since we use a single the machine driver name, all previous ACPI
tables need to be updated. That should have no impact since the
machine drivers listed at the time were not upstreamed and are no
longer maintained.

Naveen Manohar (2):
  ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
  ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper
    function

Pierre-Louis Bossart (1):
  ASoC: Intel: boards: add sof_sdw machine driver

Rander Wang (1):
  ASoC: Intel: soc-acpi: update topology and driver name for SoundWire
    platforms

 sound/soc/intel/boards/Kconfig                |  24 +
 sound/soc/intel/boards/Makefile               |   8 +-
 sound/soc/intel/boards/sof_sdw.c              | 962 ++++++++++++++++++
 sound/soc/intel/boards/sof_sdw_common.h       | 114 +++
 sound/soc/intel/boards/sof_sdw_dmic.c         |  42 +
 sound/soc/intel/boards/sof_sdw_hdmi.c         |  97 ++
 sound/soc/intel/boards/sof_sdw_rt1308.c       | 151 +++
 sound/soc/intel/boards/sof_sdw_rt5682.c       | 126 +++
 sound/soc/intel/boards/sof_sdw_rt700.c        | 125 +++
 sound/soc/intel/boards/sof_sdw_rt711.c        | 156 +++
 sound/soc/intel/boards/sof_sdw_rt715.c        |  42 +
 .../intel/common/soc-acpi-intel-cml-match.c   |  24 +-
 .../intel/common/soc-acpi-intel-icl-match.c   |   6 +-
 .../intel/common/soc-acpi-intel-tgl-match.c   |  30 +-
 14 files changed, 1896 insertions(+), 11 deletions(-)
 create mode 100644 sound/soc/intel/boards/sof_sdw.c
 create mode 100644 sound/soc/intel/boards/sof_sdw_common.h
 create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c
 create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c
 create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c
 create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c
 create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c
 create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c
 create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c

--
2.20.1

ASoC: amd: acp3x-pcm-dma: clean up two indentation issues

There are a couple of statements that are not indented correctly,
add in the missing tab and break the lines to address a checkpatch
warning.

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200327141429.269191-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: tlv320adcx140: Remove undocumented property

Remove undocumented and unneeded ti,use-internal-reg from the example as
it was an artifact from initial development.  The code does not query
for this property and as the document indicates if areg-supply is
undefined then the internal regulator is used.

Fixes: 302c0b7490cd ("dt-bindings: sound: Add TLV320ADCx140 dt
bindings")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
CC: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200327162432.17067-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function

Add support for Google Volteer device. As per new unified soundwire machine
driver, add rt5682-sdw helper function, which configures codec to Link0.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver

RT5682 is in SoundWire mode on link0.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: boards: add sof_sdw machine driver

This machine driver provides support for different configurations:

RT700, RT711, RT1308 (1x and 2x, I2S or SoundWire mode), and RT715
CometLake, Icelake, TigerLake.
PDM digital microphones
HDMI

To avoid introducing one driver per configuration, this common machine
driver relies on platform-specific information, tables and quirks to
dynamically create the relevant dailinks.

Unlike a lot of machine drivers, we use different DAI links for
SoundWire capture and playback since the Cadence PDIs can do capture
OR playback, not both simultaneously.

For each configuration, the card component string is updated so that UCM
can select the relevant parts.

Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms

Update topology and reflect change to unified machine driver for SoundWire.

Signed-off-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325220746.29601-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge branch 'for-5.6' of https://git./linux/kernel/git/broonie/sound into asoc-5.7

Merge series "ASoC: remove rtd->cpu/codec_dai{s}" from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:

Hi Mark

Now, CPU/Codec DAI(s) were replaced by rtd->dais.
Thus, We don't need rtd->cpu/codec_dai{s} anymore.
This pathset replaces it by new macro.

Kuninori Morimoto (36):
  ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro
  ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
  ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime()
  ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups
  ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais

 include/sound/soc.h                           | 30 +++++++------
 sound/arm/pxa2xx-pcm-lib.c                    |  8 ++--
 sound/soc/amd/acp-da7219-max98357a.c          |  2 +-
 sound/soc/amd/acp-rt5645.c                    |  4 +-
 sound/soc/amd/acp3x-rt5682-max9836.c          |  6 +--
 sound/soc/atmel/atmel-pcm-dma.c               |  4 +-
 sound/soc/atmel/atmel-pcm-pdc.c               |  2 +-
 sound/soc/atmel/atmel_wm8904.c                |  2 +-
 sound/soc/atmel/mikroe-proto.c                |  2 +-
 sound/soc/atmel/sam9g20_wm8731.c              |  2 +-
 sound/soc/atmel/sam9x5_wm8731.c               |  2 +-
 sound/soc/au1x/db1200.c                       |  2 +-
 sound/soc/au1x/dbdma2.c                       |  2 +-
 sound/soc/au1x/dma.c                          |  2 +-
 sound/soc/au1x/psc-ac97.c                     |  2 +-
 sound/soc/bcm/bcm63xx-pcm-whistler.c          | 16 +++----
 sound/soc/bcm/cygnus-pcm.c                    | 22 +++++-----
 sound/soc/cirrus/edb93xx.c                    |  4 +-
 sound/soc/cirrus/snappercl15.c                |  4 +-
 sound/soc/codecs/cs47l15.c                    |  4 +-
 sound/soc/codecs/cs47l24.c                    |  6 +--
 sound/soc/codecs/cs47l35.c                    |  6 +--
 sound/soc/codecs/cs47l85.c                    |  6 +--
 sound/soc/codecs/cs47l90.c                    |  6 +--
 sound/soc/codecs/cs47l92.c                    |  4 +-
 sound/soc/codecs/wm5110.c                     |  6 +--
 sound/soc/codecs/wm_adsp.c                    | 10 ++---
 sound/soc/dwc/dwc-pcm.c                       |  2 +-
 sound/soc/fsl/eukrea-tlv320.c                 |  4 +-
 sound/soc/fsl/fsl-asoc-card.c                 | 10 ++---
 sound/soc/fsl/fsl_asrc_dma.c                  |  6 +--
 sound/soc/fsl/fsl_spdif.c                     | 10 ++---
 sound/soc/fsl/fsl_ssi.c                       |  8 ++--
 sound/soc/fsl/imx-audmix.c                    |  8 ++--
 sound/soc/fsl/imx-mc13783.c                   |  4 +-
 sound/soc/fsl/imx-sgtl5000.c                  |  2 +-
 sound/soc/fsl/mpc5200_dma.c                   | 10 ++---
 sound/soc/fsl/mpc5200_psc_i2s.c               |  2 +-
 sound/soc/fsl/mpc8610_hpcd.c                  |  4 +-
 sound/soc/fsl/mx27vis-aic32x4.c               |  4 +-
 sound/soc/fsl/p1022_ds.c                      |  4 +-
 sound/soc/fsl/p1022_rdk.c                     |  4 +-
 sound/soc/fsl/wm1133-ev1.c                    |  6 +--
 sound/soc/generic/simple-card-utils.c         | 12 +++---
 sound/soc/img/img-i2s-in.c                    |  2 +-
 sound/soc/img/img-i2s-out.c                   |  2 +-
 sound/soc/intel/atom/sst-mfld-platform-pcm.c  |  6 +--
 sound/soc/intel/boards/bdw-rt5650.c           |  6 +--
 sound/soc/intel/boards/bdw-rt5677.c           |  6 +--
 sound/soc/intel/boards/broadwell.c            |  4 +-
 sound/soc/intel/boards/bxt_da7219_max98357a.c |  8 ++--
 sound/soc/intel/boards/bxt_rt298.c            |  8 ++--
 sound/soc/intel/boards/byt-max98090.c         |  2 +-
 sound/soc/intel/boards/byt-rt5640.c           |  4 +-
 sound/soc/intel/boards/bytcht_cx2072x.c       | 10 ++---
 sound/soc/intel/boards/bytcht_da7213.c        |  8 ++--
 sound/soc/intel/boards/bytcht_es8316.c        |  8 ++--
 sound/soc/intel/boards/bytcht_nocodec.c       |  4 +-
 sound/soc/intel/boards/bytcr_rt5640.c         |  8 ++--
 sound/soc/intel/boards/bytcr_rt5651.c         |  8 ++--
 sound/soc/intel/boards/cht_bsw_max98090_ti.c  |  6 +--
 sound/soc/intel/boards/cht_bsw_nau8824.c      |  4 +-
 sound/soc/intel/boards/cht_bsw_rt5645.c       | 14 +++----
 sound/soc/intel/boards/cht_bsw_rt5672.c       |  8 ++--
 sound/soc/intel/boards/cml_rt1011_rt5682.c    |  6 +--
 sound/soc/intel/boards/glk_rt5682_max98357a.c | 10 ++---
 sound/soc/intel/boards/haswell.c              |  2 +-
 sound/soc/intel/boards/kbl_da7219_max98357a.c |  8 ++--
 sound/soc/intel/boards/kbl_da7219_max98927.c  |  6 +--
 sound/soc/intel/boards/kbl_rt5660.c           |  6 +--
 sound/soc/intel/boards/kbl_rt5663_max98927.c  |  8 ++--
 .../intel/boards/kbl_rt5663_rt5514_max98927.c |  8 ++--
 .../soc/intel/boards/skl_nau88l25_max98357a.c | 12 +++---
 sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +++----
 sound/soc/intel/boards/skl_rt286.c            |  8 ++--
 sound/soc/intel/boards/sof_da7219_max98373.c  |  8 ++--
 sound/soc/intel/boards/sof_pcm512x.c          |  8 ++--
 sound/soc/intel/boards/sof_rt5682.c           |  6 +--
 sound/soc/intel/haswell/sst-haswell-pcm.c     | 26 ++++++------
 sound/soc/intel/skylake/skl-pcm.c             | 10 ++---
 sound/soc/kirkwood/armada-370-db.c            |  2 +-
 sound/soc/kirkwood/kirkwood-dma.c             |  2 +-
 sound/soc/mediatek/common/mtk-afe-fe-dai.c    | 10 ++---
 .../mediatek/common/mtk-afe-platform-driver.c |  2 +-
 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c    |  2 +-
 sound/soc/mediatek/mt2701/mt2701-cs42448.c    |  4 +-
 sound/soc/mediatek/mt2701/mt2701-wm8960.c     |  4 +-
 sound/soc/mediatek/mt6797/mt6797-afe-pcm.c    |  2 +-
 sound/soc/mediatek/mt8173/mt8173-afe-pcm.c    |  2 +-
 sound/soc/mediatek/mt8173/mt8173-max98090.c   |  4 +-
 .../mediatek/mt8173/mt8173-rt5650-rt5514.c    |  2 +-
 .../mediatek/mt8173/mt8173-rt5650-rt5676.c    |  4 +-
 sound/soc/mediatek/mt8173/mt8173-rt5650.c     |  6 +--
 sound/soc/mediatek/mt8183/mt8183-afe-pcm.c    |  2 +-
 .../mediatek/mt8183/mt8183-da7219-max98357.c  |  4 +-
 .../mt8183/mt8183-mt6358-ts3a227-max98357.c   |  2 +-
 sound/soc/meson/aiu-fifo.c                    |  2 +-
 sound/soc/meson/axg-card.c                    |  8 ++--
 sound/soc/meson/axg-fifo.c                    |  2 +-
 sound/soc/meson/meson-card-utils.c            |  2 +-
 sound/soc/mxs/mxs-sgtl5000.c                  |  4 +-
 sound/soc/pxa/brownstone.c                    |  4 +-
 sound/soc/pxa/corgi.c                         |  4 +-
 sound/soc/pxa/hx4700.c                        |  4 +-
 sound/soc/pxa/imote2.c                        |  4 +-
 sound/soc/pxa/magician.c                      |  8 ++--
 sound/soc/pxa/mioa701_wm9713.c                |  4 +-
 sound/soc/pxa/mmp-pcm.c                       |  2 +-
 sound/soc/pxa/mmp-sspa.c                      |  2 +-
 sound/soc/pxa/poodle.c                        |  4 +-
 sound/soc/pxa/pxa2xx-i2s.c                    |  2 +-
 sound/soc/pxa/spitz.c                         |  4 +-
 sound/soc/pxa/ttc-dkb.c                       |  2 +-
 sound/soc/pxa/z2.c                            |  4 +-
 sound/soc/pxa/zylonite.c                      |  6 +--
 sound/soc/qcom/apq8016_sbc.c                  |  2 +-
 sound/soc/qcom/apq8096.c                      |  6 +--
 sound/soc/qcom/lpass-platform.c               |  2 +-
 sound/soc/qcom/qdsp6/q6asm-dai.c              |  4 +-
 sound/soc/qcom/qdsp6/q6routing.c              |  2 +-
 sound/soc/qcom/sdm845.c                       | 22 +++++-----
 sound/soc/qcom/storm.c                        |  2 +-
 sound/soc/rockchip/rk3288_hdmi_analog.c       |  4 +-
 sound/soc/rockchip/rk3399_gru_sound.c         | 16 +++----
 sound/soc/rockchip/rockchip_max98090.c        |  6 +--
 sound/soc/rockchip/rockchip_rt5645.c          |  6 +--
 sound/soc/samsung/arndale.c                   |  6 +--
 sound/soc/samsung/bells.c                     | 16 +++----
 sound/soc/samsung/h1940_uda1380.c             |  2 +-
 sound/soc/samsung/i2s.c                       |  2 +-
 sound/soc/samsung/jive_wm8750.c               |  4 +-
 sound/soc/samsung/littlemill.c                | 14 +++----
 sound/soc/samsung/lowland.c                   |  4 +-
 sound/soc/samsung/neo1973_wm8753.c            | 10 ++---
 sound/soc/samsung/odroid.c                    |  2 +-
 sound/soc/samsung/pcm.c                       |  4 +-
 sound/soc/samsung/rx1950_uda1380.c            |  2 +-
 sound/soc/samsung/s3c-i2s-v2.c                |  2 +-
 sound/soc/samsung/s3c24xx_simtec.c            |  4 +-
 sound/soc/samsung/s3c24xx_uda134x.c           |  6 +--
 sound/soc/samsung/smartq_wm8987.c             |  4 +-
 sound/soc/samsung/smdk_spdif.c                |  2 +-
 sound/soc/samsung/smdk_wm8580.c               |  2 +-
 sound/soc/samsung/smdk_wm8994.c               |  2 +-
 sound/soc/samsung/smdk_wm8994pcm.c            |  4 +-
 sound/soc/samsung/snow.c                      |  4 +-
 sound/soc/samsung/spdif.c                     |  8 ++--
 sound/soc/samsung/speyside.c                  |  8 ++--
 sound/soc/samsung/tm2_wm5110.c                | 16 +++----
 sound/soc/samsung/tobermory.c                 |  8 ++--
 sound/soc/sh/dma-sh7760.c                     | 16 +++----
 sound/soc/sh/fsi.c                            |  2 +-
 sound/soc/sh/migor.c                          |  6 +--
 sound/soc/sh/rcar/core.c                      |  2 +-
 sound/soc/soc-compress.c                      | 36 ++++++++--------
 sound/soc/soc-core.c                          | 42 +++++++------------
 sound/soc/soc-dapm.c                          |  4 +-
 sound/soc/soc-generic-dmaengine-pcm.c         |  6 +--
 sound/soc/soc-pcm.c                           | 30 ++++++-------
 sound/soc/sof/intel/hda-dai.c                 |  6 +--
 sound/soc/sof/intel/hda-dsp.c                 |  2 +-
 sound/soc/sprd/sprd-pcm-compress.c            |  4 +-
 sound/soc/sprd/sprd-pcm-dma.c                 |  2 +-
 sound/soc/stm/stm32_adfsdm.c                  | 12 +++---
 sound/soc/stm/stm32_sai_sub.c                 |  2 +-
 sound/soc/sunxi/sun4i-spdif.c                 |  2 +-
 sound/soc/tegra/tegra_alc5632.c               |  2 +-
 sound/soc/tegra/tegra_max98090.c              |  2 +-
 sound/soc/tegra/tegra_rt5640.c                |  2 +-
 sound/soc/tegra/tegra_rt5677.c                |  2 +-
 sound/soc/tegra/tegra_sgtl5000.c              |  2 +-
 sound/soc/tegra/tegra_wm8753.c                |  2 +-
 sound/soc/tegra/tegra_wm8903.c                |  6 +--
 sound/soc/tegra/trimslice.c                   |  2 +-
 sound/soc/ti/ams-delta.c                      |  4 +-
 sound/soc/ti/davinci-evm.c                    |  4 +-
 sound/soc/ti/davinci-vcif.c                   |  4 +-
 sound/soc/ti/n810.c                           |  2 +-
 sound/soc/ti/omap-abe-twl6040.c               |  6 +--
 sound/soc/ti/omap-mcbsp-st.c                  |  2 +-
 sound/soc/ti/omap-mcbsp.c                     |  4 +-
 sound/soc/ti/omap-mcpdm.c                     |  2 +-
 sound/soc/ti/omap3pandora.c                   |  4 +-
 sound/soc/ti/osk5912.c                        |  2 +-
 sound/soc/ti/rx51.c                           |  2 +-
 sound/soc/txx9/txx9aclc.c                     |  2 +-
 sound/soc/uniphier/aio-compress.c             | 22 +++++-----
 sound/soc/uniphier/aio-dma.c                  |  6 +--
 sound/soc/ux500/mop500_ab8500.c               |  6 +--
 sound/soc/ux500/ux500_pcm.c                   |  8 ++--
 sound/soc/xtensa/xtfpga-i2s.c                 |  2 +-
 191 files changed, 573 insertions(+), 577 deletions(-)

--
2.17.1

Merge series "ASoC: SOF: Intel: add SoundWire support" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

This patchset provides the support for SoundWire support on Intel
CometLake, IcelLake and TigerLake RVP platforms and form-factor
devices to be released 'soon'.

The bulk of the code is about detecting a valid SoundWire
configuration from ACPI, and implementing the interfaces suggested in
'[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces'
for interrupts, PCI wakes and clock-stop configurations.

Since that SoundWire series will not be in 5.7, the build support for
SOF w/ SoundWire is not provided for now, and fall-back functions will
be used. This code is tested on a daily basis in the SOF tree and is
not expected to change in significant ways.

Changes since v2:
Corrected error in ACPI table (thanks Amadeusz)
Added patch 11 to add reset cycle required on some SoundWire platforms

Bard Liao (1):
  ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt
    handlers

Pierre-Louis Bossart (8):
  ASoC: soc-acpi: expand description of _ADR-based devices
  ASoC: SOF: Intel: add SoundWire configuration interface
  ASoC: SOF: IPC: dai-intel: move ALH declarations in header file
  ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks
  ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect
  ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend
  ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop
    quirks
  ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing
    capabilities

Rander Wang (2):
  ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
  Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread

 include/sound/soc-acpi.h                      |  39 +-
 include/sound/sof/dai-intel.h                 |  18 +-
 .../intel/common/soc-acpi-intel-cml-match.c   |  87 +++-
 .../intel/common/soc-acpi-intel-icl-match.c   |  97 ++++-
 .../intel/common/soc-acpi-intel-tgl-match.c   |  49 ++-
 sound/soc/sof/intel/hda-ctrl.c                |  25 +-
 sound/soc/sof/intel/hda-dsp.c                 |   2 +
 sound/soc/sof/intel/hda-loader.c              |  31 ++
 sound/soc/sof/intel/hda.c                     | 400 ++++++++++++++++++
 sound/soc/sof/intel/hda.h                     |  66 +++
 10 files changed, 750 insertions(+), 64 deletions(-)

--
2.20.1

ASoC: rt5682: move DAI clock registry to I2S mode

The SoundWire mode doesn't need the DAI clocks.
Therefore, the DAI clock registry moves to I2S mode case.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200327073849.18291-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: pxa: magician: convert to use i2c_new_client_device()

Move away from the deprecated API and return the shiny new ERRPTR where
useful.

Signed-off-by: Wolfram Sang <wsa+renesas@sang-engineering.com>
Link: https://lore.kernel.org/r/20200326211010.13471-2-wsa+renesas@sang-engineering.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities

Without this cycle, HDaudio capability parsing fails on some devices.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread

If pci device is in D0, wakeen interrupt will be
aggregated at cAVS level as interrupt. This commit
check the wakeen status and process it in irq thread

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire

When a SoundWire link is in clock stop state, a Slave device may wake
up the Master for some events such as jack detection. The WAKEEN
interrupt will be triggered and processed by the audio pci device.

If audio device is in D3, the interrupt will be routed to PME, or
aggregated at cAVS level as interrupt when audio device is in D0. This
patch only supports D3 case, where the audio pci device will be
resumed by a PME event and the WAKEEN interrupt will be processed
after audio pci device is powered up and ROM is initialized
successfully.

The WAKEEN handling is only enabled after the first boot due to
dependencies on a shim_lock mutex being initialized.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks

Add module parameter so that the different modes can be quickly tested.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers

We have a single irq handler for SOF interrupts. We can further merge
SoundWire ones to completely remove MSI interrupts handling issues
leading to timeouts.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend

Doing this avoid conflicts and errors reported on the bus.

The interrupts are only re-enabled on resume after the firmware is
downloaded, so the behavior is not fully symmetric

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect

For now we have a limited number of machine driver configurations, and
we can detect them based on the link configuration returned after
checking hardware and firmware (BIOS) configurations.

The link configuration is checked with a link_mask as well as a list
of _ADR descriptors for each link.

There is a chance that in extreme cases where the BIOS contains too
much information we would need to detect which Slave devices actually
report as 'attached'. This would be more accurate than static
table-based solutions, but it also introduces timing dependencies
since we don't know when those devices might become attached, so will
only be only be looked at if we see limitations with static methods
and the usual quirks based e.g. on DMI information.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks

These callbacks are invoked when a matching hw_params/hw_free() DAI
operation takes place, and will result in IPC operations with the SOF
firmware.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: IPC: dai-intel: move ALH declarations in header file

ALH was inserted in the wrong place during integration, add after DMIC
to mirror the file used by SOF firmware.

No functional change, just text move in the same file to better track
changes, if any.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: SOF: Intel: add SoundWire configuration interface

Now that the SoundWire core supports the multi-step initialization,
call the relevant APIs.

The actual hardware enablement can be done in two places, ideally we'd
want to startup the SoundWire IP as soon as possible (while still
taking power rail dependencies into account)

However when suspend/resume is implemented, the DSP device will be
resumed first, and only when the DSP firmware is downloaded/booted
would the SoundWire child devices be resumed, so there are only
marginal benefits in starting the IP earlier for the first probe.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: soc-acpi: expand description of _ADR-based devices

For SoundWire, we need to know if endpoints needs to be 'aggregated'
(MIPI parlance, meaning logically grouped), e.g. when two speaker
amplifiers need to be handled as a single logical output.

We don't necessarily have the information at the firmware (BIOS)
level, so add a notion of endpoints and specify if a device/endpoint
is part of a group, with a position.

This may be expanded in future solutions, for now only provide a group
and position information.

Since we modify the header file, change all existing upstream tables
as well to avoid breaking compilation/bisect.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87wo7bhci3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87y2rrhcia.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87zhc7hcih.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/871rpjir34.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87369zir3b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/874kufir3i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/875zevir3p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/877dzbir3w.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/878sjrir43.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87a747ir4b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>

ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87blonir4j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>