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As long as investigating packet dumps from Fireface 400/800, bits on
status registers for clock synchronization are the same.
This commit moves a parser for the registers to obsolete model-specific
operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As long as investigating packet dumps from Fireface 400/800, status
registers for clock synchronization is common.
This commit moves some macros for them to header file.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When building without CONFIG_PCI, we can (depending on the architecture)
get a link failure:
ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined!
Adding a compile-time check for PCI gets it to work correctly on
32-bit ARM.
Fixes: d99501b8575d ("ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Back-merge for resolving the conflict of fixup entries added in both
branches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS UX433FN and UX333FA with ALC294 cannot detect the headset MIC
and output through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS UX533FD with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The known ALC256_FIXUP_ASUS_MIC fixup can fix the headphone jack
sensing and enable use of the internal microphone on this laptop
X542UN. However, it's ALC294 so create a new fixup named
ALC294_FIXUP_ASUS_MIC to avoid confusion.
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>
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In an initial commit, 'SYNC_STATUS' register is referred to get
clock configuration, however this is wrong, according to my local
note at hand for reverse-engineering about packet dump. It should
be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config()
is correctly programmed.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 76fdb3a9e13a ('ALSA: fireface: add support for Fireface 400')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Users reported a mute LED regression on Lenovo X1 Carbon, the root
cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE
to this machine, then the machine can't apply the fixup of
ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup
together.
Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch will enable headset button for new Chrome platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Extend some structs to add the support for jack button changes.
Now snd_hda_jack_add_kctl() receives two more arguments: the jack type
and the jack keymaps. Both are optional, and when zero are passed,
the function behaves just like before.
For reporting button state changes, you'd need to update
jack->button_state bits accordingly, typically in the jack callback.
Then the value OR'ed with button_state and the jack plug state is
passed to snd_jack_report().
Note that currently the code assumes only the one-shot button events,
i.e. it tries to send the button release soon after sending the button
event. If a driver really supports the button release handling by
itself, we may need to introduce some flag to control this behavior in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For allowing the callee to evaluate the associated jack information
and the unsolicited event data, add the new fields to
hda_jack_callback. They can be used, for example, to retrieve the
headset button state in the callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Back-merge for applying the more HD-audio quirks on top of the latest
code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If it plugged headphone or headset into the jack, then
do the reboot, it will have a chance to cause headphone no sound.
It just need to run the headphone mode procedure after boot time.
The issue will be fixed.
It also suitable for ALC234 ALC274 and ALC294.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
A couple of open coded iterating thru the child node names are converted
to use for_each_child_of_node() instead.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Convert soundbus uevent and sysfs OF node name and device type usage to
use printf specifier and helper functions instead of directly accessing
the name and type pointers. This will allow the eventual removal of the
pointers.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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Tegra186 and Tegra194 contain the same codecs as earlier chips and can
be supported using the same patch function.
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recent devices support more than the 4 codecs that the AZX core will
probe by default. Probe up to 8 codecs to make sure all of them are
enumerated.
Suggested-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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This is a series of patches for conversion to LEDs audio-mute
trigger. It's based on 4.20-rc3 to be an immutable branch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As addressed in alsa-lib (commit b420056604f0), we need to fix the
case where the evaluation of PCM interval "(x x+1]" leading to
-EINVAL. After applying rules, such an interval may be translated as
"(x x+1)".
Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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By default HDA sound card is registered with shortname "tegra-hda".
Same driver is used across tegra platforms and it is necessary to
distinguish between platforms to use platform specific settings from
userspace. One such example is, hdmi port on different platforms use
different alsa pcm device ID. For hdmi playback to work it should
open correct pcm device depending on the platform.
This patch applies shortname from first compatible string provided
in root node of device tree. Userspace then can use this card name
to apply specific settings.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since we've switched to the LED trigger for binding with HD-audio,
we can drop the exported function as well as the whole
linux/thinkpad_acpi.h.
The own TPACPI_LED_MUTE and TPACPI_LED_MICMUTE definitions are
replaced with the identical ones for LEDS, i.e. LED_AUDIO_MUTE and
LED_AUDIO_MICMUTE, respectively. They are no longer needed as
referred only locally.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Henrique de Moraes Holschuh <hmh@hmh.eng.br>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since we've switched to the LED trigger for binding with HD-audio,
we can drop the exported function as well as the whole
linux/dell-led.h.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now all relevant platform drivers are providing the LED audio trigger,
we can switch the mute LED control with the LED trigger, finally.
For the mic-mute LED trigger, a common fixup function,
snd_hda_gen_fixup_micmute_led(), is provided to be called for the
corresponding quirk entries. This sets up the capture sync hook with
ledtrig_audio_set() call appropriately.
For the mute LED trigger, which is done currently only for
thinkpad_acpi, the call is replaced with ledtrig_audio_set() as well.
Overall, the beauty of the new implementation is that the whole ugly
bindings with request_symbol() are dropped, and also that it provides
more flexibility to users.
One potential behavior change by this patch is that the mute LED enum
may be created on machines that actually have no LED device. In the
former code, we did test-call and abort binding if the test failed.
But with the LED-trigger binding, this test isn't possible, and the
actual check is done in the LED class device side. So it's the
downside of simpleness.
Also, note that the HD-audio codec driver doesn't select CONFIG_LEDS
and co by itself. It's supposed to be selected by the platform
drivers instead.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In the upcoming change, the binding of audio mute / mic-mute LED
controls will be switched with LED trigger. This patch is the last
piece of preparation: adding the audio mute / mic-mute LED class
devices to thinkpad_acpi driver.
Two devices, platform::mute and platform::micmute, will be added for
controlling the mute LED and mic-mute LED, respectively. The new
prefix "platform" is the suggestion by upstream for indicating the
generic laptop attribute.
Also this selects CONFIG_LEDS_TRIGGERS and CONFIG_LEDS_TRIGGERS_AUDIO
unconditionally. Strictly speaking, these aren't 100% mandatory, but
leaving these manual selections would lead to a functional regression
easily once after converting from the dynamic symbol binding to the
LEDs trigger in a later patch.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Henrique de Moraes Holschuh <hmh@hmh.eng.br>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds the LED trigger support for audio mic-mute control.
As of this patch, the LED device isn't tied with the audio driver, and
can be changed via user-space at "platform::micmute" sysfs entry.
(This new prefix "platform" is the agreement among people for
indicating the generic laptop / system-wide attribute.)
The binding with HD-audio is still done via the existing exported
dell_micmute_led_set(). It will be replaced with the LED trigger
binding in later patches.
Also this selects CONFIG_LEDS_TRIGGERS and CONFIG_LEDS_TRIGGERS_AUDIO
unconditionally. Strictly speaking, these aren't 100% mandatory, but
leaving these manual selections would lead to a functional regression
easily once after converting from the dynamic symbol binding to the
LEDs trigger in a later patch.
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds a new LED trigger for coupling the audio mixer change
with the LED on laptops or other devices. Currently there are two
trigger types, "audio-mute" and "audio-micmute".
The audio driver triggers the LED brightness change via
ledtrig_audio_set() call with the proper type (either mute or
mic-mute). OTOH, the consumers may call ledtrig_audio_get() for the
initial brightness value that may have been set by the audio driver
beforehand.
This new stuff will be used by HD-audio codec driver and some platform
drivers (thinkpad_acpi and dell-laptop, also upcoming huawei-wmi).
Acked-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Acked-by: Pavel Machek <pavel@ucw.cz>
Acked-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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.
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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>
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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>
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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>
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Both snd_ctl_add() and snd_ctl_replace() process the things in a
fairly similar way, and indeed the most of the codes can be unified.
This patch is a refactoring to consolidate the both functions to call
a single helper with an extra "mode" argument. There should be no
functional difference, except for one additional sanity check applied
now to snd_ctl_replace() (which was rather overlooking, IMO), too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull the user control race fix, so that we can continue working on the
code refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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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>
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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>
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