Kai Vehmanen [Mon, 12 Oct 2020 10:27:04 +0000 (13:27 +0300)]
ALSA: hda: fix jack detection with Realtek codecs when in D3
In case HDA controller becomes active, but codec is runtime suspended,
jack detection is not successful and no interrupt is raised. This has
been observed with multiple Realtek codecs and HDA controllers from
different vendors. Bug does not occur if both codec and controller are
active, or both are in suspend. Bug can be easily hit on desktop systems
with no built-in speaker.
The problem can be fixed by powering up the codec once after every
controller runtime resume. Even if codec goes back to suspend later, the
jack detection will continue to work. Add a flag to 'hda_codec' to
describe codecs that require this flow from the controller driver.
Modify __azx_runtime_resume() to use pm_request_resume() to make the
intent clearer.
Mark all Realtek codecs with the new forced_resume flag.
Takashi Iwai [Tue, 6 Oct 2020 16:17:22 +0000 (19:17 +0300)]
ALSA: hda/i915 - fix list corruption with concurrent probes
Current hdac_i915 uses a static completion instance to wait
for i915 driver to complete the component bind.
This design is not safe if multiple HDA controllers are active and
communicating with different i915 instances, and can lead to list
corruption and failed audio driver probe.
Fix the design by moving completion mechanism to common acomp
code and remove the related code from hdac_i915.
Jian-Hong Pan [Wed, 7 Oct 2020 05:22:25 +0000 (13:22 +0800)]
ALSA: hda/realtek: Enable audio jacks of ASUS D700SA with ALC887
The ASUS D700SA desktop's audio (1043:2390) with ALC887 cannot detect
the headset microphone and another headphone jack until
ALC887_FIXUP_ASUS_HMIC and ALC887_FIXUP_ASUS_AUDIO quirks are applied.
The NID 0x15 maps as the headset microphone and NID 0x19 maps as another
headphone jack. Also need the function like alc887_fixup_asus_jack to
enable the audio jacks.
Qiu Wenbo [Fri, 2 Oct 2020 12:44:54 +0000 (20:44 +0800)]
ALSA: hda/realtek - Add mute Led support for HP Elitebook 845 G7
After installing archlinux, the mute led and micmute led are not working
at all. This patch fix this issue by applying a fixup from similar
model. These mute leds are confirmed working on HP Elitebook 845 G7.
Kai Vehmanen [Wed, 30 Sep 2020 11:41:40 +0000 (14:41 +0300)]
ASoC: hdac_hda: allow runtime pm at end of probe
Align with recent change to forbid runtime suspend during codec
init in snd_hda_codec_device_new(), with matching call to
allow suspend at end of hdac_hda_codec_probe().
In snd-hda-intel, call to snd_hda_set_power_save() at end of
controller probe does the same thing, but ASoC controller drivers
do not modify runtime settings for codecs, so this has to be done
in codec drivers, and in this case in hdac_hda.
For certain codecs (like Realtek), pm_runtime_forbid() is invoked
in the probe function after build_controls(). In a stress test,
its observed occasionally that runtime PM calls are invoked
before controls are built. This causes the codec to be
runtime suspended before probe completes. Because of this, not all
controls are enumerated correctly, and audio does not work until
system is rebooted.
This issue being common across all codecs, pm_runtime_forbid() is
called when the codec object is created to fix this issue.
A codec enables or disables runtime pm in its own probe function.
Multiple stress tests of 2000+ cycles has been done to test the fix.
Hui Wang [Wed, 30 Sep 2020 05:51:46 +0000 (13:51 +0800)]
ALSA: hda - Don't register a cb func if it is registered already
If the caller of enable_callback_mst() passes a cb func, the callee
function will malloc memory and link this cb func to the list
unconditionally. This will introduce problem if caller is in the
hda_codec_ops.init() since the init() will be repeatedly called in the
codec rt_resume().
So far, the patch_hdmi.c and patch_ca0132.c call enable_callback_mst()
in the hda_codec_ops.init().
Hui Wang [Mon, 28 Sep 2020 08:01:17 +0000 (16:01 +0800)]
ALSA: hda/realtek - set mic to auto detect on a HP AIO machine
Recently we enabled a HP AIO machine, we found the mic on the machine
couldn't record any sound and it couldn't detect plugging and
unplugging as well.
Through debugging we found the mic is set to manual detect mode, after
setting it to auto detect mode, it could detect plugging and
unplugging and could record sound.
Kai Vehmanen [Thu, 24 Sep 2020 16:10:27 +0000 (19:10 +0300)]
ALSA: hda - remove kerneldoc for internal hdac_i915 function
Drop the kerneldoc markup for connectivity_check() as it's an
static helper function. Fixes the following make W=1 warning:
sound/hda/hdac_i915.c:80: warning: Function parameter or member 'i915' not described in 'connectivity_check'
sound/hda/hdac_i915.c:80: warning: Function parameter or member 'hdac' not described in 'connectivity_check'
ALSA: seq: oss: Avoid mutex lock for a long-time ioctl
Recently we applied a fix to cover the whole OSS sequencer ioctls with
the mutex for dealing with the possible races. This works fine in
general, but in theory, this may lead to unexpectedly long stall if an
ioctl like SNDCTL_SEQ_SYNC is issued and an event with the far future
timestamp was queued.
For fixing such a potential stall, this patch changes the mutex lock
applied conditionally excluding such an ioctl command. Also, change
the mutex_lock() with the interruptible version for user to allow
escaping from the big-hammer mutex.
František Kučera [Tue, 22 Sep 2020 14:42:06 +0000 (16:42 +0200)]
ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
This patch extends support for DJM-250MK2 and allows mapping
playback and capture channels to available sources.
Configures the card through USB commands.
ALSA: ctl: Workaround for lockdep warning wrt card->ctl_files_rwlock
The recent change in lockdep for read lock caused the deadlock
warnings in ALSA control code which uses the read_lock() for
notification and else while write_lock_irqsave() is used for adding
and removing the list entry. Although a deadlock would practically
never hit in a real usage (the addition and the deletion can't happen
with the notification), it's better to fix the read_lock() usage in a
semantically correct way.
This patch replaces the read_lock() calls with read_lock_irqsave()
version for avoiding a reported deadlock. The notification code path
takes the irq disablement in anyway, and other code paths are very
short execution, hence there shouldn't be any big performance hit by
this change.
Fixes: e918188611f0 ("locking: More accurate annotations for read_lock()") Reported-by: syzbot+561a74f84100162990b2@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20200922084953.29018-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kai Vehmanen [Mon, 21 Sep 2020 14:17:41 +0000 (17:17 +0300)]
ALSA: hda - fix CONTROLLER_IN_GPU macro name
The CONTROLLER_IN_GPU() macro has different semantics than
the similarly named macro in hda_intel.c. The name is also
misleading as the macro is used to apply a Intel HSW/BDW
programming logic for HDA controller clock configuration.
Rename macro to reflect the actual implementation.
Kai Vehmanen [Mon, 21 Sep 2020 14:17:40 +0000 (17:17 +0300)]
ALSA: hda - handle multiple i915 device instances
Currently i915_component_master_match() will return the first matching
i915 instance. This does not work in case system has multiple i915
and HDA audio controller instances.
Add a new connectivity check that handles following cases:
- i915 and HDA controller on same PCI bus
- discrete GPU with embedded HDA audio controller connected
via PCI bridge
Hui Wang [Mon, 14 Sep 2020 06:51:18 +0000 (14:51 +0800)]
ALSA: hda/realtek - Couldn't detect Mic if booting with headset plugged
We found a Mic detection issue on many Lenovo laptops, those laptops
belong to differnt models and they have different audio design like
internal mic connects to the codec or PCH, they all have this problem,
the problem is if plugging a headset before powerup/reboot the
machine, after booting up, the headphone could be detected but Mic
couldn't. If we plug out and plug in the headset, both headphone and
Mic could be detected then.
Through debugging we found the codec on those laptops are same, it is
alc257, and if we don't disable the 3k pulldown in alc256_shutup(),
the issue will be fixed. So far there is no pop noise or power
consumption regression on those laptops after this change.
Tom Rix [Sun, 13 Sep 2020 16:52:30 +0000 (09:52 -0700)]
ALSA: asihpi: fix iounmap in error handler
clang static analysis flags this problem
hpioctl.c:513:7: warning: Branch condition evaluates to
a garbage value
if (pci.ap_mem_base[idx]) {
^~~~~~~~~~~~~~~~~~~~
If there is a failure in the middle of the memory space loop,
only some of the memory spaces need to be cleaned up.
At the error handler, idx holds the number of successful
memory spaces mapped. So rework the handler loop to use the
old idx.
There is a second problem, the memory space loop conditionally
iomaps()/sets the mem_base so it is necessay to initize pci.
Merge tag 'asoc-fix-v5.9-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.9
Most of this is various driver specific fixes, none of which are
terribly exciting in themselves, plus one core fix adding and using a
new DAI lookup function to deal with a lockdep warning.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In FireWire driver, a tasklet is
still used for offloading the AMDTP PCM stream handling. It can be
achieved gracefully with a work queued, too.
This patch replaces the tasklet usage in firewire-lib driver with a
simple work. The conversion is fairly straightforward but for the
in_interrupt() checks that are replaced with the check using the
current_work().
Note that in_interrupt() in amdtp_packet tracepoint is still kept as
is. This is the place that is probed by both softirq of 1394 OHCI and
a user task of a PCM application, and the work handling is already
filtered in amdtp_domain_stream_pcm_pointer().
The miXart driver has been already converted to use the threaded IRQ
instead of tasklet while there is a remaining comment still mentioning
a tasklet. Update the comment appropriately.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In ASIHPI driver, a tasklet is
still used for offloading the PCM IRQ handling. It can be achieved
gracefully with a threaded IRQ, too.
This patch replaces the tasklet usage in asihpi driver with a threaded
IRQ. It also simplified some call patterns.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In Riptide driver, a tasklet is
still used for offloading the PCM IRQ handling. It can be achieved
gracefully with a threaded IRQ, too.
This patch replaces the tasklet usage in riptide driver with a
threaded IRQ.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In HDSP-MADI driver, a tasklet is
still used for offloading the MIDI I/O handling (optional via mixer
switch). It can be achieved gracefully with a work queued, too.
This patch replaces the tasklet usage in HDSP-MADI driver with a
simple work. The conversion is fairly straightforward. The only
significant difference is that the work initialization is moved to the
right place in snd_hdspm_create() and cancel_work_sync() is always
called in snd_hdspm_free() to assure killing the pending works.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In HDSP driver, a tasklet is still
used for offloading the MIDI I/O handling (optional via mixer
switch). It can be achieved gracefully with a work queued, too.
This patch replaces the tasklet usage in HDSP driver with a simple
work. The conversion is fairly straightforward. The only significant
difference is that a superfluous tasklet_kill() call is removed from
snd_hdap_midi_input_trigger().
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In aloop driver, a tasklet is still
used for offloading the timer event task. It can be achieved
gracefully with a work queued, too.
This patch replaces the tasklet usage in aloop driver with a simple
work.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In UA101 driver, a tasklet is still
used for handling the output URBs. It can be achieved gracefully with
a work queued in the high-prio system workqueue, too.
This patch replaces the tasklet usage in UA101 driver with a simple
work.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In USB-audio driver, a tasklet is
still used in MIDI interface code for handling the output byte
stream. It can be achieved gracefully with a work queued in the
high-prio system workqueue.
This patch replaces the tasklet usage in USB-audio driver with a
simple work.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In ALSA core timer API, the
callbacks can be offlined to a tasklet when a flag is set in the timer
backend. It can be achieved gracefully with a work queued in the
high-prio system workqueue.
This patch replaces the usage of tasklet in ALSA timer API with a
simple work. Currently the tasklet feature is used only in the system
timer and hrtimer backends, so both are patched to use the new flag
name SNDRV_TIMER_HW_WORK, too.
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. This patch replaces the usage of
tasklet in pcsp driver with a simple work. In pcsp driver, a global
tasklet is used for offloading the period-elapse handling in the
hrtimer callback (introduced in commit 96c7d478efad "ALSA: pcsp - Fix
locking messes in snd-pcsp"). It can be achieved gracefully with a
work queued in the high-prio system workqueue.
This also changes tasklet_kill() with cancel_work_sync() in the
sync_stop callback, which is anyway better to assure canceling the
pending tasks.
Hui Wang [Wed, 9 Sep 2020 02:00:41 +0000 (10:00 +0800)]
ALSA: hda/realtek - The Mic on a RedmiBook doesn't work
The Mic connects to the Nid 0x19, but the configuration of Nid 0x19
is not defined to Mic, and also need to set the coeff to enable the
auto detection on the Nid 0x19. After this change, the Mic plugging
in or plugging out could be detected and could record the sound from
the Mic.
And the coeff value is suggested by Kailang of Realtek.
ASoC: tlv320adcx140: Wake up codec before accessing register
According to its datasheet, after reset this codec goes into sleep
mode. In this mode, any register accessing should be avoided except for
exiting sleep mode. Hence this commit moves SLEEP_CFG access before any
register accessing.
ASoC: core: Do not cleanup uninitialized dais on soc_pcm_open failure
Introduce for_each_rtd_dais_rollback macro which behaves exactly like
for_each_codec_dais_rollback and its cpu_dais equivalent but for all
dais instead.
Use newly added macro to fix soc_pcm_open error path and prevent
uninitialized dais from being cleaned-up.
Luke D Jones [Mon, 7 Sep 2020 08:19:59 +0000 (20:19 +1200)]
ALSA: hda: fixup headset for ASUS GX502 laptop
The GX502 requires a few steps to enable the headset i/o: pincfg,
verbs to enable and unmute the amp used for headpone out, and
a jacksense callback to toggle output via internal or jack using
a verb.
ALSA: hda/realtek - Improved routing for Thinkpad X1 7th/8th Gen
There've been quite a few regression reports about the lowered volume
(reduced to ca 65% from the previous level) on Lenovo Thinkpad X1
after the commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker
on Lenovo Carbon X1 7th gen"). Although the commit itself does the
right thing from HD-audio POV in order to have a volume control for
bass speakers, it seems that the machine has some secret recipe under
the hood.
Through experiments, Benjamin Poirier found out that the following
routing gives the best result:
* DAC1 (NID 0x02) -> Speaker pin (NID 0x14)
* DAC2 (NID 0x03) -> Shared by both Bass Speaker pin (NID 0x17) &
Headphone pin (0x21)
* DAC3 (NID 0x06) -> Unused
DAC1 seems to have some equalizer internally applied, and you'd get
again the output in a bad quality if you connect this to the
headphone pin. Hence the headphone is connected to DAC2, which is now
shared with the bass speaker pin. DAC3 has no volume amp, hence it's
not connected at all.
For achieving the routing above, this patch introduced a couple of
workarounds:
* The connection list of bass speaker pin (NID 0x17) is reduced not to
include DAC3 (NID 0x06)
* Pass preferred_pairs array to specify the fixed connection
Here, both workarounds are needed because the generic parser prefers
the individual DAC assignment over others.
When the routing above is applied, the generic parser creates the two
volume controls "Front" and "Bass Speaker". Since we have only two
DACs for three output pins, those are not fully controlling each
output individually, and it would confuse PulseAudio. For avoiding
the pitfall, in this patch, we rename those volume controls to some
unique ones ("DAC1" and "DAC2"). Then PulseAudio ignore them and
concentrate only on the still good-working "Master" volume control.
If a user still wants to control each DAC volume, they can still
change manually via "DAC1" and "DAC2" volume controls.
sound/drivers/vx/vx_pcm.c:63:7: style: Variable 'buf' is assigned a
value that is never used. [unreadVariable]
buf = (unsigned char *)runtime->dma_area;
^
sound/drivers/vx/vx_pcm.c:539:30: style: Variable
'chip->playback_pipes[audio]' is reassigned a value before the old one
has been used. [redundantAssignment]
chip->playback_pipes[audio] = pipe;
^
sound/drivers/vx/vx_pcm.c:533:31: note: chip->playback_pipes[audio] is
assigned
chip->playback_pipes[audio] = pipe;
^
sound/drivers/vx/vx_pcm.c:539:30: note: chip->playback_pipes[audio] is
overwritten
chip->playback_pipes[audio] = pipe;
^
sound/pci/hda/hda_auto_parser.c:353:7: style: Local variable 'i'
shadows outer variable [shadowVariable]
int i = 0;
^
sound/pci/hda/hda_auto_parser.c:182:6: note: Shadowed declaration
int i;
^
sound/pci/hda/hda_auto_parser.c:353:7: note: Shadow variable
int i = 0;
^
It's not clear why a new declaration was added, remove and reuse
variable declared with larger scope.
sound/core/compress_offload.c:1044:6: style: Redundant initialization
for 'ret'. The initialized value is overwritten before it is
read. [redundantInitialization]
ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS,
^
sound/core/compress_offload.c:1034:10: note: ret is initialized
int ret = -EINVAL;
^
sound/core/compress_offload.c:1044:6: note: ret is overwritten
ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS,
^
ALSA: compress_offload: dereference after checking for NULL pointer
Fix cppcheck warning and only dereference once the initial checks are
done:
sound/core/compress_offload.c:516:38: warning: Either the condition
'!stream' is redundant or there is possible null pointer dereference:
stream. [nullPointerRedundantCheck]
struct snd_compr_runtime *runtime = stream->runtime;
^
sound/core/compress_offload.c:518:17: note: Assuming that condition
'!stream' is not redundant
if (snd_BUG_ON(!(stream) || !(stream)->runtime))
^
sound/core/compress_offload.c:516:38: note: Null pointer dereference
struct snd_compr_runtime *runtime = stream->runtime;
^
Cppcheck complains about a possible NULL pointer dereference but it
actually looks like the NULL assignment is not needed (same loop is
used in other parts of the file without it).
ALSA: core: pcm_memory: dereference pointer after NULL checks
Fix cppcheck warnings:
sound/core/pcm_memory.c:380:26: warning: Either the condition
'!substream' is redundant or there is possible null pointer
dereference: substream. [nullPointerRedundantCheck]
struct snd_card *card = substream->pcm->card;
^
sound/core/pcm_memory.c:384:6: note: Assuming that condition
'!substream' is not redundant
if (PCM_RUNTIME_CHECK(substream))
^
sound/core/pcm_memory.c:380:26: note: Null pointer dereference
struct snd_card *card = substream->pcm->card;
^
sound/core/pcm_memory.c:433:26: warning: Either the condition
'!substream' is redundant or there is possible null pointer
dereference: substream. [nullPointerRedundantCheck]
struct snd_card *card = substream->pcm->card;
^
sound/core/pcm_memory.c:436:6: note: Assuming that condition
'!substream' is not redundant
if (PCM_RUNTIME_CHECK(substream))
^
sound/core/pcm_memory.c:433:26: note: Null pointer dereference
struct snd_card *card = substream->pcm->card;
^
Fix cppcheck, the fallthrough only makes sense within the conditional
block
sound/core/memalloc.c:161:3: style:inconclusive: Statements following
return, break, continue, goto or throw will never be
executed. [unreachableCode]
fallthrough;
^
Hans de Goede [Tue, 1 Sep 2020 08:06:23 +0000 (10:06 +0200)]
ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1
The MPMAN Converter9 2-in-1 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.
ALSA: hda: add dev_dbg log when driver is not selected
On SKL+ Intel platforms, the driver selection is handled by the
snd_intel_dspcfg, and when the HDaudio legacy driver is not selected,
be it with the auto-selection or user preferences with a kernel
parameter, the probe aborts with no logs, only a -ENODEV return value.
Having no dmesg trace, even with dynamic debug enabled, makes support
more complicated than it needs to be, and even experienced users can
be fooled. A simple dev_dbg() trace solves this problem.
Rander Wang [Wed, 2 Sep 2020 15:42:18 +0000 (18:42 +0300)]
ALSA: hda: fix a runtime pm issue in SOF when integrated GPU is disabled
In snd_hdac_device_init pm_runtime_set_active is called to
increase child_count in parent device. But when it is failed
to build connection with GPU for one case that integrated
graphic gpu is disabled, snd_hdac_ext_bus_device_exit will be
invoked to clean up a HD-audio extended codec base device. At
this time the child_count of parent is not decreased, which
makes parent device can't get suspended.
This patch calls pm_runtime_set_suspended to decrease child_count
in parent device in snd_hdac_device_exit to match with
snd_hdac_device_init. pm_runtime_set_suspended can make sure that
it will not decrease child_count if the device is already suspended.
Allen Pais [Wed, 2 Sep 2020 04:02:21 +0000 (09:32 +0530)]
ALSA: ua101: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:20 +0000 (09:32 +0530)]
ALSA: usb-audio: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:19 +0000 (09:32 +0530)]
ASoC: txx9: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:18 +0000 (09:32 +0530)]
ASoC: siu: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:17 +0000 (09:32 +0530)]
ASoC: fsl_esai: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:16 +0000 (09:32 +0530)]
ALSA: hdsp: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:15 +0000 (09:32 +0530)]
ALSA: riptide: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:14 +0000 (09:32 +0530)]
ALSA: pci/asihpi: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:13 +0000 (09:32 +0530)]
ALSA: firewire: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Allen Pais [Wed, 2 Sep 2020 04:02:12 +0000 (09:32 +0530)]
ALSA: core: convert tasklets to use new tasklet_setup() API
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
ASoC: Intel: haswell: Fix power transition refactor
While addressing existing power-cycle limitations for
sound/soc/intel/haswell solution, change brings regression for standard
audio userspace flows e.g.: when using PulseAudio.
Occasional sound-card initialization fail is still better than
permanent audio distortions, so revert the change.
Fixes: 8ec7d6043263 ("ASoC: Intel: haswell: Power transition refactor") Reported-by: Christian Bundy <christianbundy@fraction.io> Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200901153041.14771-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA: pcm: oss: Remove superfluous WARN_ON() for mulaw sanity check
The PCM OSS mulaw plugin has a check of the format of the counter part
whether it's a linear format. The check is with snd_BUG_ON() that
emits WARN_ON() when the debug config is set, and it confuses
syzkaller as if it were a serious issue. Let's drop snd_BUG_ON() for
avoiding that.
While we're at it, correct the error code to a more suitable, EINVAL.
ASoC: wm8994: Ensure the device is resumed in wm89xx_mic_detect functions
When the wm8958_mic_detect, wm8994_mic_detect functions get called from
the machine driver, e.g. from the card's late_probe() callback, the CODEC
device may be PM runtime suspended and any regmap writes have no effect.
Add PM runtime calls to these functions to ensure the device registers
are updated as expected.
This suppresses an error during boot
"wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec"
caused by the regmap access error due to the cache_only flag being set.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Acked-by: Krzysztof Kozlowski <krzk@kernel.org> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200827173357.31891-2-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
The WM8994_MICBIAS register is not available in the WM1811 CODEC so skip
initialization of that register for that device.
This suppresses an error during boot:
"wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec"
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Acked-by: Krzysztof Kozlowski <krzk@kernel.org> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20200827173357.31891-1-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie@kernel.org>
Dan Crawford [Sat, 29 Aug 2020 02:49:46 +0000 (12:49 +1000)]
ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO
Following Christian Lachner's patch for Gigabyte X570-based motherboards,
also patch the MSI X570-A PRO motherboard; the ALC1220 codec requires the
same workaround for Clevo laptops to enforce the DAC/mixer connection
path. Set up a quirk entry for that.
I suspect most if all X570 motherboards will require similar patches.
[ The entries reordered in the SSID order -- tiwai ]
Jerome Brunet [Fri, 28 Aug 2020 15:14:38 +0000 (17:14 +0200)]
ASoC: meson: axg-toddr: fix channel order on g12 platforms
On g12 and following platforms, The first channel of record with more than
2 channels ends being placed randomly on an even channel of the output.
On these SoCs, a bit was added to force the first channel to be placed at
the beginning of the output. Apparently the behavior if the bit is not set
is not easily predictable. According to the documentation, this bit is not
present on the axg series.
Set the bit on g12 and fix the problem.
Fixes: a3c23a8ad4dc ("ASoC: meson: axg-toddr: add g12a support") Reported-by: Nicolas Belin <nbelin@baylibre.com> Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200828151438.350974-1-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie@kernel.org>
commit 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper")
added snd_soc_dai_link_set_capabilities().
But it is using snd_soc_find_dai() (A) which is required client_mutex (B).
And client_mutex is soc-core.c local.
void snd_soc_dai_link_set_capabilities(xxx)
{
...
for_each_pcm_streams(direction) {
...
for_each_link_cpus(dai_link, i, cpu) {
(A) dai = snd_soc_find_dai(cpu);
...
}
...
for_each_link_codecs(dai_link, i, codec) {
(A) dai = snd_soc_find_dai(codec);
...
}
}
...
}
Because of these background, we will get WARNING if .config has CONFIG_LOCKDEP.
WARNING: CPU: 2 PID: 53 at sound/soc/soc-core.c:814 snd_soc_find_dai+0xf8/0x100
CPU: 2 PID: 53 Comm: kworker/2:1 Not tainted 5.7.0-rc1+ #328
Hardware name: Renesas H3ULCB Kingfisher board based on r8a77951 (DT)
Workqueue: events deferred_probe_work_func
pstate: 60000005 (nZCv daif -PAN -UAO)
pc : snd_soc_find_dai+0xf8/0x100
lr : snd_soc_find_dai+0xf4/0x100
...
Call trace:
snd_soc_find_dai+0xf8/0x100
snd_soc_dai_link_set_capabilities+0xa0/0x16c
graph_dai_link_of_dpcm+0x390/0x3c0
graph_for_each_link+0x134/0x200
graph_probe+0x144/0x230
platform_drv_probe+0x5c/0xb0
really_probe+0xe4/0x430
driver_probe_device+0x60/0xf4
snd_soc_find_dai() will be used from (X) CPU/Codec/Platform driver with
mutex lock, and (Y) Card driver without mutex lock.
This snd_soc_dai_link_set_capabilities() is for Card driver,
this means called without mutex.
This patch adds snd_soc_find_dai_with_mutex() to solve it.
Kai Vehmanen [Wed, 26 Aug 2020 17:03:06 +0000 (20:03 +0300)]
ALSA: hda/hdmi: always check pin power status in i915 pin fixup
When system is suspended with active audio playback to HDMI/DP, two
alternative sequences can happen at resume:
a) monitor is detected first and ALSA prepare follows normal
stream setup sequence, or
b) ALSA prepare is called first, but monitor is not yet detected,
so PCM is restarted without a pin,
In case of (b), on i915 systems, haswell_verify_D0() is not called at
resume and the pin power state may be incorrect. Result is lack of audio
after resume with no error reported back to user-space.
Fix the problem by always verifying converter and pin state in the
i915_pin_cvt_fixup().
Dinghao Liu [Thu, 20 Aug 2020 04:28:27 +0000 (12:28 +0800)]
ASoC: qcom: common: Fix refcount imbalance on error
for_each_child_of_node returns a node pointer np with
refcount incremented. So when devm_kzalloc fails, a
pairing refcount decrement is needed to keep np's
refcount balanced.
Vinod Koul [Wed, 26 Aug 2020 16:33:40 +0000 (22:03 +0530)]
ASoC: rt700: Fix return check for devm_regmap_init_sdw()
devm_regmap_init_sdw() returns a valid pointer on success or ERR_PTR on
failure which should be checked with IS_ERR. Also use PTR_ERR for
returning error codes.
Vinod Koul [Wed, 26 Aug 2020 16:33:39 +0000 (22:03 +0530)]
ASoC: rt715: Fix return check for devm_regmap_init_sdw()
devm_regmap_init_sdw() returns a valid pointer on success or ERR_PTR on
failure which should be checked with IS_ERR. Also use PTR_ERR for
returning error codes.
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