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

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=207043Tested-by: Roy Spliet <nouveau@spliet.org>
Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

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

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: 3e6db33a ("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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

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: 5cb543db ("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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

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

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

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

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

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

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

Merge tag 'asoc-fix-v5.7' of https://git.kernel.org/pub/scm/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.

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

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

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

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: 6b8d6e55 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

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: 0401e854 ("ALSA: hda - Move beep helper functions to hda_beep.c")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

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

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

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

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

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

The commit 0d6defc7 ("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 0d6defc7 ("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.frSigned-off-by: Mark Brown <broonie@kernel.org>

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

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

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

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: 24c8d141 ("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.comSigned-off-by: Mark Brown <broonie@kernel.org>

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

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

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>

[ This is essentially the same fix as commit ae769d35, 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
f2ecf903: "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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903: "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: f2ecf903 ("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.deSigned-off-by: Takashi Iwai <tiwai@suse.de>

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

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

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

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: 5549ea64 ("ASoC: rt5682: fix unmet dependencies")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200401091055.34112-1-yuehaibing@huawei.comSigned-off-by: Mark Brown <broonie@kernel.org>

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

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

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: 2a9e92d3 ("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.netSigned-off-by: Mark Brown <broonie@kernel.org>

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