| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
Acer AIO Veriton Z4860G/Z6860G with the same ALC286 codec has issues
with the input from external microphone. The issue can be fixed by
the fixup ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE for Veriton Z4660G.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Acer AIO Veriton Z4660G with ALC286 codec has issue with the input
from external microphones connecting via 'Front Mic' jack. The fixup
ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE enables the jack sensing of
the headset and fix the audio input issue of external microphone.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The Acer AIO Aspire C24-860 with ALC286 can't detect the headset
microphone. Just like another Acer AIO U27-880, it needs a different
pin value for 0x18 and the headset fixup to make headset mic work.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Acer Aspire U27-880(AIO) with ALC286 codec can not detect headset mic
and internal mic not working either. It needs the similar quirk like
Sony laptops to fix headphone jack sensing and enables use of the
internal microphone.
Unfortunately jack sensing for the headset mic is still not working.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
If a USB sound card reports 0 interfaces, an error condition is triggered
and the function usb_audio_probe errors out. In the error path, there was a
use-after-free vulnerability where the memory object of the card was first
freed, followed by a decrement of the number of active chips. Moving the
decrement above the atomic_dec fixes the UAF.
[ The original problem was introduced in 3.1 kernel, while it was
developed in a different form. The Fixes tag below indicates the
original commit but it doesn't mean that the patch is applicable
cleanly. -- tiwai ]
Fixes: 362e4e49abe5 ("ALSA: usb-audio - clear chip->probing on error exit")
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We've got a regression report for some Thinkpad models (at least
T570s) which shows the too low speaker output volume. The bisection
leaded to the commit 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad
Dock device for ALC298 platform"), and it's basically adding the two
pin configurations for the dock, and looks harmless.
The real culprit seems, though, that the DAC assignment for the
speaker pin is implicitly assumed on these devices, i.e. pin NID 0x14
to be coupled with DAC NID 0x03. When more pins are configured by the
commit above, the auto-parser changes the DAC assignment, and this
resulted in the regression.
As a workaround, just provide the fixed pin / DAC mapping table for
this Thinkpad fixup function. It's no generic solution, but the
problem itself is pretty much device-specific, so must be good
enough.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1554304
Fixes: 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform")
Cc: <stable@vger.kernel.org>
Reported-and-tested-by: Jeremy Cline <jcline@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
It's similar to other AMD audio devices, it also supports D3, which can
save some power drain.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable
Native DSD support.
[ Moved the added entry in numerical order -- tiwai ]
Signed-off-by: Tony Das <tdas444@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Commit 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM
stream") fixes deadlock for non-atomic PCM stream. But, This patch
causes antother stuck.
If writer is RT thread and reader is a normal thread, the reader
thread will be difficult to get scheduled. It may not give chance to
release readlocks and writer gets stuck for a long time if they are
pinned to single cpu.
The deadlock described in the previous commit is because the linux
rwsem queues like a FIFO. So, we might need non-FIFO writelock, not
non-block one.
My suggestion is that the writer gives reader a chance to be scheduled
by using the minimum msleep() instaed of spinning without blocking by
writer. Also, The *_nonblock may be changed to *_nonfifo appropriately
to this concept.
In terms of performance, when trylock is failed, this minimum periodic
msleep will have the same performance as the tick-based
schedule()/wake_up_q().
[ Although this has a fairly high performance penalty, the relevant
code path became already rare due to the previous commit ("ALSA:
pcm: Call snd_pcm_unlink() conditionally at closing"). That is, now
this unconditional msleep appears only when using linked streams,
and this must be a rare case. So we accept this as a quick
workaround until finding a more suitable one -- tiwai ]
Fixes: 67ec1072b053 ("ALSA: pcm: Fix rwsem deadlock for non-atomic PCM stream")
Suggested-by: Wonmin Jung <wonmin.jung@lge.com>
Signed-off-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Currently the PCM core calls snd_pcm_unlink() always unconditionally
at closing a stream. However, since snd_pcm_unlink() invokes the
global rwsem down, the lock can be easily contended. More badly, when
a thread runs in a high priority RT-FIFO, it may stall at spinning.
Basically the call of snd_pcm_unlink() is required only for the linked
streams that are already rare occasion. For normal use cases, this
code path is fairly superfluous.
As an optimization (and also as a workaround for the RT problem
above in normal situations without linked streams), this patch adds a
check before calling snd_pcm_unlink() and calls it only when needed.
Reported-by: Chanho Min <chanho.min@lge.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide
useful string for the vendor and product names too. In order to share
the UCM with WD15, here we keep the profile_name same as the WD15.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.20
Lots of fixes here, the majority of which are driver specific but
there's a couple of core things and one notable driver specific one:
- A core fix for a DAPM regression introduced during the component
refactoring, we'd lost the code that forced a reevaluation of the
DAPM graph after probe (which we suppress during init to save lots
of recalcuation) and have now restored it.
- A core fix for error handling using the newly added
for_each_rtd_codec_dai_rollback() macro.
- A fix for the names of widgets in the newly introduced pcm3060
driver, merged as a fix so we don't have a release with legacy names.
|
|
This patch will enable ALC300.
[ It's almost equivalent with other ALC269-compatible ones, and
apparently has no loopback mixer -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This device makes a loud buzzing sound when a headphone is inserted while
playing audio at full volume through the speaker.
Fixes: bbf8ff6b1d2a ("ALSA: hda/realtek - Fixup for HP x360 laptops with B&O speakers")
Signed-off-by: Girija Kumar Kasinadhuni <gkumar@neverware.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We have several Lenovo laptops with the codec alc285, when playing
sound via headphone, we can hear click/pop noise in the headphone,
if we let the headphone share the DAC of NID 0x2 with the speaker,
the noise disappears.
The Lenovo laptops here include P52, P72, X1 yoda2 and X1 carbon.
I have tried to set preferred_dacs and override_conn, but neither of
them worked. Thanks for Kailang, he told me to invalidate the NID 0x3
through override_wcaps.
BugLink: https://bugs.launchpad.net/bugs/1805079
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The procedure for adding a user control element has some window opened
for race against the concurrent removal of a user element. This was
caught by syzkaller, hitting a KASAN use-after-free error.
This patch addresses the bug by wrapping the whole procedure to add a
user control element with the card->controls_rwsem, instead of only
around the increment of card->user_ctl_count.
This required a slight code refactoring, too. The function
snd_ctl_add() is split to two parts: a core function to add the
control element and a part calling it. The former is called from the
function for adding a user control element inside the controls_rwsem.
One change to be noted is that snd_ctl_notify() for adding a control
element gets called inside the controls_rwsem as well while it was
called outside the rwsem. But this should be OK, as snd_ctl_notify()
takes another (finer) rwlock instead of rwsem, and the call of
snd_ctl_notify() inside rwsem is already done in another code path.
Reported-by: syzbot+dc09047bce3820621ba2@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Some spurious calls of snd_free_pages() have been overlooked and
remain in the error paths of sparc cs4231 driver code. Since
runtime->dma_area is managed by the PCM core helper, we shouldn't
release manually.
Drop the superfluous calls.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Some spurious calls of snd_free_pages() have been overlooked and
remain in the error paths of wss driver code. Since runtime->dma_area
is managed by the PCM core helper, we shouldn't release manually.
Drop the superfluous calls.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
MSI Cubi N 8GL (MS-B171) needs the same fixup as its older model, the
MS-B120, in order for the headset mic to be properly detected.
They both use a single 3-way jack for both mic and headset with an
ALC283 codec, with the same pins used.
Cc: stable@vger.kernel.org
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Power-saving is causing plops on audio start/stop on the built-in audio
of the nForce 430 based ASRock N68C-S UCC motherboard, add this model to
the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Cc: <stable@vger.kernel.org>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The function snd_ac97_put_spsa() gets the bit shift value from the
associated private_value, but it extracts too much; the current code
extracts 8 bit values in bits 8-15, but this is a combination of two
nibbles (bits 8-11 and bits 12-15) for left and right shifts.
Due to the incorrect bits extraction, the actual shift may go beyond
the 32bit value, as spotted recently by UBSAN check:
UBSAN: Undefined behaviour in sound/pci/ac97/ac97_codec.c:836:7
shift exponent 68 is too large for 32-bit type 'int'
This patch fixes the shift value extraction by masking the properly
with 0x0f instead of 0xff.
Reported-and-tested-by: Meelis Roos <mroos@linux.ee>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We need to block sleep states which would require longer time to leave than
the time the DMA must react to the DMA request in order to keep the FIFO
serviced without overrun.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We need to block sleep states which would require longer time to leave than
the time the DMA must react to the DMA request in order to keep the FIFO
serviced without under of overrun.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The latency number is in usec for the pm_qos. Correct the calculation to
give us the time in usec
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The machine_quirk may return NULL which means the acpi entries should be
skipped and search for next matched entry is needed, here add return
check here and continue for NULL case.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The Skylake driver currently has a set of problems supporting
load/unload modules. We need to make the HDaudio codec support
optional to help narrow down the issues.
Support for HDaudio codecs also leads to a Kconfig issue. We want the
hdac_hda codec to be compilable independently of Skylake (e.g. with
ALL_CODECS) but when Skylake is selected as built-in the hdac_hda
codec needs to use the same option due a a code dependency
Solve both problems by adding a user-selectable boolean Kconfig,
select HDAC_HDA as needed and make the HDaudio codec support in the
Skylake driver optional. Tests on a Chell Chromebook device without
HDaudio show no regression for speaker and HDMI playback.
This is submitted as an RFC to allow for comments and more validation.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch fixes the pincfg assignment for the AE-5, which was
previously using the Recon3D pincfg's by mistake.
Fixes: d06feaf02fe6 ("ALSA: hda/ca0132 - Add pincfg for AE-5")
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch adds a new PCI subsys ID for the ZxR, as found and tested by
other users. Without a way to know if any Z's use it as well, it keeps
the quirk of QUIRK_SBZ and goes through the HDA subsys test function.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
According to the current device datasheet (TI Lit # SLAS831D, revised
March 2018) the value written to the device's PAGE register to trigger
a complete register reset should be 0xfe, not 0xff. So go ahead and
update to the correct value.
Reported-by: Stephane Le Provost <stephane.leprovost@mediatek.com>
Tested-by: Stephane Le Provost <stephane.leprovost@mediatek.com>
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
|
|
Audio map are possible in wrong state before card->instantiated has
been set to true. Imaging the following examples:
time 1: at the beginning
in:-1 in:-1 in:-1 in:-1
out:-1 out:-1 out:-1 out:-1
SIGGEN A B Spk
time 2: after someone called snd_soc_dapm_new_widgets()
(e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c)
in:1 in:0 in:0 in:0
out:0 out:0 out:0 out:1
SIGGEN A B Spk
time 3: routes added
in:1 in:0 in:0 in:0
out:0 out:0 out:0 out:1
SIGGEN -----> A -----> B ---> Spk
In the end, the path should be powered on but it did not. At time 3,
"in" of SIGGEN and "out" of Spk did not propagate to their neighbors
because snd_soc_dapm_add_path() will not invalidate the paths if
the card has not instantiated (i.e. card->instantiated is false).
To correct the state of audio map, recalculate the whole map forcely.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The machine driver fails to probe in next-20181113 with:
[ 2.539093] omap-abe-twl6040 sound: ASoC: CODEC DAI twl6040-legacy not registered
[ 2.546630] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -517
...
[ 3.693206] omap-abe-twl6040 sound: ASoC: Both platform name/of_node are set for TWL6040
[ 3.701446] omap-abe-twl6040 sound: ASoC: failed to init link TWL6040
[ 3.708007] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -22
[ 3.715148] omap-abe-twl6040: probe of sound failed with error -22
Bisect pointed to a merge commit:
first bad commit: [0f688ab20a540aafa984c5dbd68a71debebf4d7f] Merge remote-tracking branch 'net-next/master'
and a diff between a working kernel does not reveal anything which would
explain the change in behavior.
Further investigation showed that on the second try of loading fails
because the dai_link->platform is no longer NULL and it might be pointing
to uninitialized memory.
The fix is to move the snd_soc_dai_link and snd_soc_card inside of the
abe_twl6040 struct, which is dynamically allocated every time the driver
probes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
In the initial commit [1], I added differential output of the codec as
separate `+` and `-` widgets:
OUTL+
OUTR+
OUTL-
OUTR-
Later, in the commit [2], I added a device tree property to configure the
output as single-ended or differential. Having this property, the `+` and
`-` separation in widgets seems for me confusing. There are no functional
benefits in such separation, so I find reasonable to get rid of it:
OUTL
OUTR
The new naming is more friendly for sound cards, and is better aligned with
other codec drivers in kernel.
Renaming the output widgets now should not be a problem from the backwards-
compatibility perspective, as the driver for PCM3060 is added into the
mainline very recently, and did not yet appear in any releases.
[1] commit 6ee47d4a8dac ("ASoC: pcm3060: Add codec driver")
[2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended
output")
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Frontend dai_link id is used for closing ADM sessions.
During concurrent usecase when one session is closed,
it closes other ADM session associated with other usecase
too. Dai_link->id should always point to Frontend dai id.
Set cpu_dai id as dai_link id to fix the issue.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
sun8i-codec misses a route from ADC to AIF1 Slot 0 ADC. Add it
to the driver to avoid adding it to every dts.
Fixes: eda85d1fee05d ("ASoC: sun8i-codec: Add ADC support for a33")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
On certain platforms, Display HDMI HDA codec was not going to sleep state
after the use when links are powered down after turning off the display
power. As per the HW recommendation, links are powered down before turning
off the display power to ensure that the codec goes to sleep state.
This patch was updated from an earlier version submitted upstream [1]
which conflicted with the changes merged for HDaudio codec support
with the Intel DSP.
[1] https://patchwork.kernel.org/patch/10540213/
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Stack memory isn't DMA-safe so it isn't safe to use either
regmap_raw_read or regmap_bulk_read to read into stack memory.
The two functions to read the scratch registers were using
stack memory and regmap_raw_read. It's not worth allocating
memory just for this trivial read, and it isn't time-critical.
A simple regmap_read for each register is sufficient.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch adds missing prepare_sleve_config that is needed for
setup the DMA slave channel for I2S.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We need to call pci_iounmap() instead of iounmap() for the regions
obtained via pci_iomap() call for some archs that need special
treatment.
Fixes: aa31704fd81c ("ALSA: hda/ca0132: Add PCI region2 iomap for SBZ")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
HP Pavilion 15 (103c:820d) with ALC295 codec requires the quirk for
the mute LED control over mic3 pin. Added the corresponding quirk
entry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201653
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
drvdata is actually sun8i_codec, not snd_soc_card, so it crashes
when calling snd_soc_card_get_drvdata().
Drop card and scodec vars anyway since we don't need to
disable/unprepare clocks - it's already done by calling
runtime_suspend()
Drop clk_disable_unprepare() calls for the same reason.
Fixes: 36c684936fae7 ("ASoC: Add sun8i digital audio codec")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
PCM OSS layer may allocate a few temporary buffers, one for the core
read/write and another for the conversions via plugins. Currently
both are allocated via vmalloc(). But as the allocation size is
equivalent with the PCM period size, the required size might be quite
small, depending on the application.
This patch replaces these vmalloc() calls with kvzalloc() for covering
small period sizes better. Also, we use "z"-alloc variant here for
addressing the possible uninitialized access reported by syzkaller.
Reported-by: syzbot+1cb36954e127c98dd037@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
For some reason the dapm widgets are incorrectly defined from the start,
Not sure how we ended up with such thing. Fix them now!
Without this fix the backend dais are always powered up even if there
is no active stream.
Reported-by: Jimmy Cheng-Yi Chiang <cychiang@google.com>
Reported-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
q6asm routing gets added multiple times as part of dai probe.
Move this to q6routing routes which has those widgets defined, this also
fixes the issue where these are added each time at dai probe.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Since the commit c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for
mic mute LED controls") we allow enabling the mic mute LED with
multiple ADCs. The commit changed the function return value to be
zero or a negative error, while this change was overlooked in the
thinkpad_acpi helper code where it still expects a positive return
value for success. This eventually leads to a NULL dereference on a
system that has only a mic mute LED.
This patch corrects the return value check in the corresponding code
as well.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201621
Fixes: c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
SD line mask for MI2S starts from BIT 0 instead of BIT 1.
Fix all bit mask for MI2S SD lines.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the
mclk instead of pmc_plt_clk_3.
This commit adds a DMI based quirk for this.
This fixing audio no longer working on these devices after
commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
that commit fixes us unnecessary keeping unused clocks on, but in case
of the Swanky that was breaking audio support since we were not using
the right clock in the cht_bsw_max98090_ti machine driver.
Cc: stable@vger.kernel.org
Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
Reported-and-tested-by: Dean Wallace <duffydack73@gmail.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
SND_SUN50I_CODEC_ANALOG selects SND_SUNXI_ADDA_PR_REGMAP which is leftover
of renaming SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP. Replace
it with SND_SUN8I_ADDA_PR_REGMAP to fix possible link errors for some
configurations:
sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe':
sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init'
Fixes: 42371f327df0 ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few device-specific fixes: a fix for SPDIF on old Creative PCI
board, and two additional fixes for the recent changes in FireWire
audio stack"
* tag 'sound-fix-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size
ALSA: ca0106: Disable IZD on SB0570 DAC to fix audio pops
ALSA: dice: fix to wait for releases of all ALSA character devices
|
|
commit 4d230d12710646 ("ASoC: rsnd: fixup not to call clk_get/set under
non-atomic") fixuped clock start timing. But it exchanged clock start
checker from ssi->usrcnt to ssi->rate.
Current rsnd_ssi_master_clk_start() is called from .prepare,
but some player (for example GStreamer) might calls it many times.
In such case, the checker might returns error even though it was not
error. It should check ssi->usrcnt instead of ssi->rate.
This patch fixup it. Without this patch, GStreamer can't switch
48kHz / 44.1kHz.
Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
In a former commit, PCM constraint based on LCM of SYT_INTERVAL was
obsoleted with PCM rule. However, the new PCM rule brings -EINVAL in
some cases that max/min values of size of buffer/period is not
multiples of one of values of SYT_INTERVAL. For example, pulseaudio
always fail to configure PCM substream.
This commit changes strategy for the PCM rule. Although the buggy rules
had a single dependency (rate from period, period from rate, rate from
buffer, buffer from rate), a revised rule has double dependencies
(period from period/rate, buffer from buffer/rate). A step of value is
calculated with table of SYT_INTERVAL and list of available rates. This
prevents interval template which brings -EINVAL to a call of
snd_interval_refine().
Fixes: 5950229582bc('ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
