TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need
yet more quirks for the proper control names.
This patch provides the mapping table for those boards, correcting the
FU names for volume and mute controls as well as the terminal names
for jack controls. It also improves build_connector_control() not to
add the directional suffix blindly if the string is given from the
mapping table.
With this patch applied, the new UCM profiles will be effective.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 3c6fd1f07e ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.
This patch reverts the corresponding entry as a temporary solution.
Although Zenith II and co will see get the empty HD-audio bus again,
it'd be merely resource wastes and won't affect the functionality,
so it's no end of the world. We'll need to address this later,
e.g. by either switching to DMI string matching or using PCI ID &
SSID pairs.
Fixes: 3c6fd1f07e ("ALSA: hda: Add driver blacklist")
Reported-by: Johnathan Smithinovic <johnathan.smithinovic@gmx.at>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200419071926.22683-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 1c76aa5fb4 ("ALSA: hda/realtek - Allow skipping
spec->init_amp detection") changed the way to assign spec->init_amp
field that specifies the way to initialize the amp. Along with the
change, the commit also replaced a few fixups that set spec->init_amp
in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather
aligning to the other fixups, and not supposed to change the actual
behavior.
However, this change turned out to cause a regression on FSC S7020,
which hit exactly the above. The reason was that there is still one
place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE
call, namely in alc_ssid_check().
This patch fixes the regression by adding the proper spec->init_amp
override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED.
Fixes: 1c76aa5fb4 ("ALSA: hda/realtek - Allow skipping spec->init_amp detection")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329
Link: https://lore.kernel.org/r/20200418190639.10082-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many Focusrite devices supports a limited set of sample rates per
altsetting. These includes audio interfaces with ADAT ports:
- Scarlett 18i6, 18i8 1st gen, 18i20 1st gen;
- Scarlett 18i8 2nd gen, 18i20 2nd gen;
- Scarlett 18i8 3rd gen, 18i20 3rd gen;
- Clarett 2Pre USB, 4Pre USB, 8Pre USB.
Maximum rate is exposed in the last 4 bytes of Format Type descriptor
which has a non-standard bLength = 10.
Tested-by: Alexey Skobkin <skobkin-ru@ya.ru>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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.
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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 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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
