Commit Graph

902761 Commits

Author SHA1 Message Date
Hui Wang
9a6418487b ALSA: hda: call runtime_allow() for all hda controllers
Before the pci_driver->probe() is called, the pci subsystem calls
runtime_forbid() and runtime_get_sync() on this pci dev, so only call
runtime_put_autosuspend() is not enough to enable the runtime_pm on
this device.

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

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

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-14 17:15:07 +02:00
Takashi Iwai
f8e4ae10de ALSA: hda: Allow setting preallocation again for x86
The commit c31427d0d2 ("ALSA: hda: No preallocation on x86
platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default
to zero for x86, as the preallocation should work almost all cases.
However, this expectation was too naive; some applications try to
allocate as the max buffer size as possible, and it leads to the
memory exhaustion.  More badly, the commit changed the kconfig no
longer adjustable for x86, so you can't fix it statically (although it
can be still adjusted via procfs).

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

Fixes: c31427d0d2 ("ALSA: hda: No preallocation on x86 platforms")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223
Link: https://lore.kernel.org/r/20200413201919.24241-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-13 22:22:28 +02:00
Roy Spliet
3ba21113bd 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>
2020-04-13 18:03:29 +02:00
Takashi Iwai
c4c8dd6ef8 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>
2020-04-13 18:03:16 +02:00
Takashi Iwai
9479e75fca 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>
2020-04-13 18:02:58 +02:00
Takashi Iwai
2393e7555b 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>
2020-04-13 18:02:36 +02:00
Takashi Iwai
10db5bccc3 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: 3e6db33aaf ("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>
2020-04-13 18:02:21 +02:00
Takashi Iwai
25faa4bd37 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: 5cb543dba9 ("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>
2020-04-13 18:02:09 +02:00
Takashi Iwai
934b96594e 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>
2020-04-12 10:14:54 +02:00
Takashi Iwai
7dc3c5a017 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>
2020-04-12 10:14:48 +02:00
Takashi Iwai
3507245b82 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>
2020-04-12 10:14:44 +02:00
Takashi Iwai
48cc429735 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>
2020-04-12 10:14:39 +02:00
Adam Barber
4963d66b8a 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>
2020-04-11 08:47:49 +02:00
Xu Wang
4df9332528 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>
2020-04-09 14:07:34 +02:00
Takashi Iwai
ddd5609fe8 ASoC: Fixes for v5.7
A collection of fixes that have been accumilated since the merge window,
 mainly relating to x86 platform support.
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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.
2020-04-08 18:08:09 +02:00
Takashi Iwai
3c6fd1f07e 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>
2020-04-08 16:06:04 +02:00
Takashi Iwai
2a48218f8e 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>
2020-04-08 16:05:55 +02:00
Takashi Iwai
1d3aa4a551 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>
2020-04-08 15:57:12 +02:00
Takashi Iwai
c47914c00b 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: 6b8d6e5518 ("[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>
2020-04-07 16:43:46 +02:00
Takashi Iwai
0ad3f0b384 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: 0401e8548e ("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>
2020-04-07 16:42:29 +02:00
Mike Willard
ccfc531695
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>
2020-04-07 15:29:54 +01:00
Kailang Yang
24164f434d 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>
2020-04-07 09:48:57 +02:00
Kailang Yang
431e76c3ed 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>
2020-04-07 09:47:37 +02:00
Hans de Goede
4146575eb0
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>
2020-04-06 17:45:24 +01:00
Hans de Goede
c8b78f24c1
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>
2020-04-06 15:03:28 +01:00
Julia Lawall
7506baeed8
ASoC: stm32: sai: Add missing cleanup
The commit 0d6defc7e0 ("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 0d6defc7e0 ("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>
2020-04-06 15:03:27 +01:00
Emmanuel Pescosta
fd60e0683e 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>
2020-04-04 19:57:09 +02:00
Hans de Goede
c515291d31
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>
2020-04-03 14:39:58 +01:00
Hans de Goede
0bb2be2d1b
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>
2020-04-03 14:39:57 +01:00
Hans de Goede
81630dc042
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: 24c8d14192 ("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>
2020-04-03 14:39:57 +01:00
Hans de Goede
904f353d0e
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>
2020-04-03 14:39:56 +01:00
František Kučera
73d8c94084 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>
2020-04-03 09:41:24 +02:00
Takashi Iwai
b79900a489 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>
2020-04-03 09:39:26 +02:00
Takashi Iwai
ff7e06a556 ALSA: pcm: oss: Fix regression by buffer overflow fix (again)
[ This is essentially the same fix as commit ae769d3556, 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
f2ecf903ef: "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>
2020-04-03 09:38:50 +02:00
Takashi Iwai
ae769d3556 ALSA: pcm: oss: Fix regression by buffer overflow fix
The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef: "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: f2ecf903ef ("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>
2020-04-03 09:25:52 +02:00
Takashi Iwai
dbdd24eaac 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>
2020-04-02 20:42:29 +02:00
Hans de Goede
ca707b3f00 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>
2020-04-02 20:41:57 +02:00
이경택
abca9e4a04
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>
2020-04-01 12:06:28 +01:00
YueHaibing
59564e1173
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: 5549ea6479 ("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>
2020-04-01 12:06:28 +01:00
이경택
21fca8bdbb
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>
2020-04-01 12:06:27 +01:00
이경택
3bbbb7728f
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>
2020-03-31 19:23:55 +01:00
Stephan Gerhold
7f2430cda8
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: 2a9e92d371 ("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>
2020-03-31 19:23:54 +01:00
Andreas Steinmetz
5c6cd7021a 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>
2020-03-31 14:34:28 +02:00
Takashi Iwai
b6f69c7955 Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
This reverts commit 645c08f17f
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: 645c08f17f ("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>
2020-03-31 11:01:12 +02:00
Thomas Hebb
f36938aa74 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: 8c69729b44 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3")
Fixes: 423cd78561 ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360")
Fixes: e4c9fd10eb ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant")
Fixes: 1099f48457 ("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>
2020-03-31 10:54:06 +02:00
Thomas Hebb
c447374494 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 e3743f4311 ("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: e1e8c1fdce ("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>
2020-03-31 10:53:29 +02:00
Thomas Hebb
f128090491 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>
2020-03-31 10:51:45 +02:00
Mark Brown
7f95581187
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
2020-03-30 18:22:38 +01:00
Pierre-Louis Bossart
1ba616bd1a
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: b56be800f1 ("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>
2020-03-30 18:22:37 +01:00
이경택
0ab070917a
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>
2020-03-30 18:22:36 +01:00