led_colors clashes with the array of the same name being added
to the LED class. Do the following amendments to fix this issue
and the other prospective one.
led_colors -> toneport_led_colors
led_init_vals -> toneport_led_init_vals
Fixes: f44edd7b2b ("ALSA: line6/toneport: Implement LED controls via LED class")
Signed-off-by: Jacek Anaszewski <jacek.anaszewski@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Fixes gcc '-Wunused-but-set-variable' warnings:
sound/ppc/snd_ps3.c: In function 'snd_ps3_program_dma':
sound/ppc/snd_ps3.c:236:8: warning: variable 'start_vaddr' set but not used [-Wunused-but-set-variable]
sound/ppc/snd_ps3.c: In function 'snd_ps3_pcm_open':
sound/ppc/snd_ps3.c:529:6: warning: variable 'pcm_index' set but not used [-Wunused-but-set-variable]
They are never used and can be removed.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Geoff Levand <geoff@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have two Dell laptops which have the codec 10ec0236 and 10ec0256
respectively, the headset mic on them can't work, need to apply the
quirk of ALC255_FIXUP_DELL1_MIC_NO_PRESENCE. So adding their pin
configurations in the pin quirk table.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_cards[] array holds the card pointers that have been currently
registered, and it's exported for the external modules that may need
to refer a card object. But accessing to this array can be racy
against the driver probe or removal, as the card registration or free
may happen concurrently.
This patch gets rid of the direct access to snd_cards[] array and
provides a helper function to give the card object from the index
number with a refcount management. Then the caller can access to the
given card object safely, and releases it via snd_card_unref().
While we're at it, add a proper comment to snd_card_unref() and make
it an inlined function for type-safety, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The emu10k1 driver tries to create a unique id string by itself when
it's copied from the card list, but it's rather superfluous, as the
same thing will be done in ALSA core side at the card registration.
Let's drop the code. This allows us removing snd_cards export.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a small race window in the card disconnection code that
allows the registration of another card with the very same card id.
This leads to a warning in procfs creation as caught by syzkaller.
The problem is that we delete snd_cards and snd_cards_lock entries at
the very beginning of the disconnection procedure. This makes the
slot available to be assigned for another card object while the
disconnection procedure is being processed. Then it becomes possible
to issue a procfs registration with the existing file name although we
check the conflict beforehand.
The fix is simply to move the snd_cards and snd_cards_lock clearances
at the end of the disconnection procedure. The references to these
entries are merely either from the global proc files like
/proc/asound/cards or from the card registration / disconnection, so
it should be fine to shift at the very end.
Reported-by: syzbot+48df349490c36f9f54ab@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALSA proc helper manages the child nodes in a linked list, but its
addition and deletion is done without any lock. This leads to a
corruption if they are operated concurrently. Usually this isn't a
problem because the proc entries are added sequentially in the driver
probe procedure itself. But the card registrations are done often
asynchronously, and the crash could be actually reproduced with
syzkaller.
This patch papers over it by protecting the link addition and deletion
with the parent's mutex. There is "access" mutex that is used for the
file access, and this can be reused for this purpose as well.
Reported-by: syzbot+48df349490c36f9f54ab@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The doubly unlock sequence at snd_seq_client_ioctl_unlock() is tricky.
I took a direct unref call since I thought it would avoid
misunderstanding, but rather this seems more confusing. Let's use
snd_seq_client_unlock() consistently even if they look strange to be
called twice, and add more comments for avoiding reader's confusion.
Fixes: 6b580f5231 ("ALSA: seq: Protect racy pool manipulation from OSS sequencer")
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device reports Synch: Synchronous on the playback interface.
This causes regular audible napping on sample rates that are not multiples
of 1 kHz. Fix to Synch: Asynchronous.
Specifically observed on Focusrite Scarlett Solo 2nd generation. I assume
the first generation model has a different device ID. A first generation
Scarlett 2i2 I was able to test advertised Synch: Asynchronous by default.
For example, with a sample rate of 44100 Hz, a silent sample is played
every 40.96 seconds (likely 44.0 samples instead of 44.1 transmitted per
USB frame on average, 4096 being the size of some internal buffer).
There may be some other bug at play here since this doesn't happen
on other platforms. However, a feedback endpoint is listed and using it
fixes the issue. That is the only change in the quirk,
but I didn't find a way to declare only it.
Tested on two units and on two different computers.
Signed-off-by: Roope Salmi <rpsalmi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fields in snd_hdac_bus are ext-bus specific, but they still
should be initialized in snd_hdac_bus_init() for consistency, at
least, for the ones that do need the explicit initialization like the
list head.
Also move the lock field to the more appropriate place and correct the
comment to reflect the recent change where it serves for both the
display power and the link management.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit 98081ca62c ("ALSA: hda - Record the current power
state before suspend/resume calls") made the HD-audio driver to store
the PM state in power_state field. This forgot, however, the
initialization at power up. Although the codec drivers usually don't
need to refer to this field in the normal operation, let's initialize
it properly for consistency.
Fixes: 98081ca62c ("ALSA: hda - Record the current power state before suspend/resume calls")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
OSS sequencer emulation still allows to queue and issue the events
that manipulate the client pool concurrently in a racy way. This
patch serializes the access like the normal sequencer write / ioctl
via taking the client ioctl_mutex. Since the access to the sequencer
client is done indirectly via a client id number, a new helper to
take/release the mutex is introduced.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have two helpers for queuing a sequencer event from the kernel
client, and both are used only from OSS sequencer layer without any
hop and atomic set. Let's simplify and unify two helpers into one.
No functional change, just a call pattern change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of unsubscribe_port() which manages the group count and
module refcount from delete_and_unsubscribe_port() looks racy; it's
not covered by the group list lock, and it's likely a cause of the
reported unbalance at port deletion. Let's move the call inside the
group list_mutex to plug the hole.
Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit feb689025f.
The fix attempt was incorrect, leading to the mutex deadlock through
the close of OSS sequencer client. The proper fix needs more
consideration, so let's revert it now.
Fixes: feb689025f ("ALSA: seq: Protect in-kernel ioctl calls with mutex")
Reported-by: syzbot+47ded6c0f23016cde310@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few core fixes along with the driver specific ones, mainly fixing
small issues that only affect x86 platforms for various reasons (their
unusual machine enumeration mechanisms mainly, plus a fix for error
handling in topology).
There's some of the driver fixes that look larger than they are, like
the hdmi-codec changes which resulted in an indentation change, and most
of the other large changes are for new drivers like the STM32 changes.
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Merge tag 'asoc-fix-v5.1-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.1
A few core fixes along with the driver specific ones, mainly fixing
small issues that only affect x86 platforms for various reasons (their
unusual machine enumeration mechanisms mainly, plus a fix for error
handling in topology).
There's some of the driver fixes that look larger than they are, like
the hdmi-codec changes which resulted in an indentation change, and most
of the other large changes are for new drivers like the STM32 changes.
wcd9335.c: undefined reference to 'devm_regmap_add_irq_chip'
Signed-off-by: Marc Gonzalez <marc.w.gonzalez@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_hdac_display_power() doesn't handle the concurrent calls carefully
enough, and it may lead to the doubly get_power or put_power calls,
when a runtime PM and an async work get called in racy way.
This patch addresses it by reusing the bus->lock mutex that has been
used for protecting the link state change in ext bus code, so that it
can protect against racy display state changes. The initialization of
bus->lock was moved from snd_hdac_ext_bus_init() to
snd_hdac_bus_init() as well accordingly.
Testcase: igt/i915_pm_rpm/module-reload #glk-dsi
Reported-by: Chris Wilson <chris@chris-wilson.co.uk>
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Imre Deak <imre.deak@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Handle error before returning when try_module_get() fails
to prevent inconsistent mutex lock/unlock.
Fixes: 52034add7 (ASoC: pcm: update module refcount if
module_get_upon_open is set)
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When master clock is used, master clock rate is set exclusively.
Parent clocks of master clock cannot be changed after a call to
clk_set_rate_exclusive(). So the parent clock of SAI kernel clock
must be set before.
Ensure also that exclusive rate operations are balanced
in STM32 SAI driver.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix wrong setting on number of channels. The context wants to set
constraint to 2 channels instead of 4.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Just a minor refactoring to use the standard goto for error paths in
snd_timer_open() instead of open code. The first mutex_lock() is
moved to the beginning of the function to make the code clearer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_seq_ioctl_get_subscription() retrieves the port subscriber
information as a pointer, while the object isn't protected, hence it
may be deleted before the actual reference. This race was spotted by
syzkaller and may lead to a UAF.
The fix is simply copying the data in the lookup function that
performs in the rwsem to protect against the deletion.
Reported-by: syzbot+9437020c82413d00222d@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA OSS sequencer calls the ioctl function indirectly via
snd_seq_kernel_client_ctl(). While we already applied the protection
against races between the normal ioctls and writes via the client's
ioctl_mutex, this code path was left untouched. And this seems to be
the cause of still remaining some rare UAF as spontaneously triggered
by syzkaller.
For the sake of robustness, wrap the ioctl_mutex also for the call via
snd_seq_kernel_client_ctl(), too.
Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
spin_lock_irqsave() is used unnecessarily in various places in
sequencer core code although it's pretty obvious that the context is
sleepable. Remove irqsave and use the plain spin_lock_irq() in such
places for simplicity.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a few places in sequencer core, we temporarily unlock / re-lock the
pool spin lock while waiting for the allocation in the blocking mode.
There spin_unlock_irq() / spin_lock_irq() pairs are called while
initially spin_lock_irqsave() is used (and spin_lock_irqrestore() at
the end of the function again). This is likely OK for now, but it's a
bit confusing and error-prone.
This patch replaces these temporary relocking lines with the irqsave
variant to make the lock/unlock sequence more consistently.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use kvmalloc() for allocating cell pools since the pool size can be
relatively small that may be covered better by slab.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is essentially a revert of the commit a7588c896b ("ALSA: timer:
Check ack_list emptiness instead of bit flag"). The intended change
by the commit turns out to be insufficient, as snd_timer_close*()
always calls snd_timer_stop() that deletes the ack_list beforehand.
In theory, we can change the behavior of snd_timer_stop() to sync the
pending ack_list, but this will become a deadlock for the callback
like sequencer that calls again snd_timer_stop() from itself. So,
reverting the change is a more straightforward solution.
Fixes: a7588c896b ("ALSA: timer: Check ack_list emptiness instead of bit flag")
Reported-by: syzbot+58813d77154713f4de15@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we set CONFIG_SND_HDA_POWER_SAVE_DEFAULT to 1 when
configuring the kernel, then two machines were reported to have noise
after installing the new kernel. Put them in the blacklist, the
noise disappears.
https://bugs.launchpad.net/bugs/1821663
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
mmiowb() is now implied by spin_unlock() on architectures that require
it, so there is no reason to call it from driver code. This patch was
generated using coccinelle:
@mmiowb@
@@
- mmiowb();
and invoked as:
$ for d in drivers include/linux/qed sound; do \
spatch --include-headers --sp-file mmiowb.cocci --dir $d --in-place; done
NOTE: mmiowb() has only ever guaranteed ordering in conjunction with
spin_unlock(). However, pairing each mmiowb() removal in this patch with
the corresponding call to spin_unlock() is not at all trivial, so there
is a small chance that this change may regress any drivers incorrectly
relying on mmiowb() to order MMIO writes between CPUs using lock-free
synchronisation. If you've ended up bisecting to this commit, you can
reintroduce the mmiowb() calls using wmb() instead, which should restore
the old behaviour on all architectures other than some esoteric ia64
systems.
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Will Deacon <will.deacon@arm.com>
Fix some cosmetic spacing issues reported by Julia Lawall
<julia.lawall@lip6.fr>.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
kbl_da7219_max98927 supports two configurations - One with da7219 &
max98927/373 and other one with max98927/373 alone. This patch
removes the headset dais from max98927/373 configuration. Since the
snd_soc_dapm_ignore_suspend for DMIC is called from da7219_codec_init,
the code is re-arranged to invoke snd_soc_dapm_ignore_suspend from
kabylake_dmic_init.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Setting the module_get_upon_open field for component driver
prevents the module refcount from being incremented during
component probe(). This could lead to the module being
allowed to be unloaded when a pcm stream is open. So,
if this field is set, the module's refcount should be
incremented during pcm open to prevent module removal
when the component is in use. And, the refcount should
be decremented upon pcm close.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Recently, for Intel platforms the "ignore_module_refcount" field
was introduced for the component driver. In order to avoid a
deadlock preventing the PCI modules from being removed
even when the card was idle, the refcounts were not incremented
for the device driver module during component probe.
However, this change introduced a nasty side effect:
the device driver module can be unloaded while a pcm stream is open.
This patch proposes to change the field to be renamed as
"module_get_upon_open". When this field is set, the module
refcount should be incremented on pcm open amd decremented
upon pcm close. This will enable modules to be removed
when no PCM playback/capture happens and prevent removal
when the component is actually in use.
Also, align with the skylake component driver with the new name.
Fixes: b450b878('ASoC: core: don't increase component module refcount
unconditionally'
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Providing a range for usleep_range() allows the hrtimer subsystem to
coalesce timers - as this delay has no upper limit anyway (interrupts
or context switch is possible) it should not hurt to extend this
from 2 to 2-4 milliseconds.
Signed-off-by: Nicholas Mc Guire <hofrat@opentech.at>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Teclast X98+ II is a Cherrytrail tablet, which require two quirks:
- it has stereo speakers,
- its jack detection mechanism is inverted.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk to support boards whose jack detection mechanism is
inverted.
It will set the 'everest,jack-detect-inverted' boolean device property
for the es8316 codec driver.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Andra Danciu <andradanciu1997@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes the sai driver structure overwriting which results in
a cpu dai name equal NULL.
Fixes: 3e086ed ("ASoC: stm32: add SAI driver")
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The control values and texts of the enum kcontrol associated
with a widget need to be freed when the widget is removed.
However, both struct snd_soc_dapm_widget and struct soc_enum
contain a dobj member, which resulted in a confusion.
The existing code generates a null pointer dereference by
attempting to free the values and texts from the dobj which
belongs to the widget instead of the dobj belonging to the
enum kcontrol.
The suggested fix is to use the correct dobj member (se->dobj)
of the enum kcontrol.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When ioctl calls are made with non-null-terminated userspace strings,
strlcpy causes an OOB-read from within strlen. Fix by changing to use
strscpy instead.
Signed-off-by: Zubin Mithra <zsm@chromium.org>
Reviewed-by: Guenter Roeck <groeck@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use SND_SOC_DAPM_SUPPLY for mic bias DAPM
instead of deprecated SND_SOC_DAPM_MICBIAS.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In ESAI synchronous mode, the clock is generated by Tx, So
we should always set registers of Tx which relate with the
bit clock and frame clock generation (TCCR, TCR, ECR), even
there is only Rx is working.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use an explicit define to avoid Sparse issues coming from the use of
cpu_to_be32
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use le16/32/64_to_cpu() as needed to make Sparse happy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The axg tdmout driver just need a different skew offset to operate
correctly on the g12a SoC family.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The g12a tdmout requires a different signal skew offset than the axg.
With this change, the skew offset is added as a parameter of the tdm
formatters to prepare the addition of the g12a support.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the axg, frddr could only be connected to 1 downstream element, so the
playback was possible on 1 interface only at a time.
On the g12a, the frddr may connect and wait for the request of up to 3
downstream elements. With this, it possible for single playback to be
played on several interfaces at the same time.
Like the toddr fifo, the g12a frddr also need to take care of resetting
the read pointer to the initial fifo address when preparing a playback.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the g12a SoC fifo can set the fifo initial start address, we must
make sure to actually reset the write pointer to this address when
starting a capture.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The g12a fifos gained the ability to set the initial address of the
pointer within the buffer, instead of defaulting to the buffer start
address.
It is not very useful to us (yet) but we need to put a copy the buffer
start address in the related register for the fifo to work properly on the
g12a SoC family
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the card is registered by the machine driver,
dai link components are probed after the snd_card is
created. This is done in snd_soc_bind_card() which calls
snd_soc_instantiate_card() to first create the snd_card
and then probes the link components by calling
soc_probe_link_components(). The snd_card is used by the
component driver to add the kcontrols associated
with dapm widgets to the card.
When the machine driver is unregistered, the snd_card
is freed when the card resources are cleaned up.
But the snd_card needs to be valid while unloading the
topology dapm widgets in order to remove the kcontrols
from the card.
Since, unloading topology is done when the component
driver is removed, the link components should be removed
in snd_soc_unbind_card(). This will ensure that the kcontrols
are removed before the card resources are cleaned up and
the snd_card itself is freed.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology is not unloaded in the core during unregister_component()
anymore. So, add the remove() callback that will unload the
topology.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The chips main power supplies VA and VP are enabled during probe but
then never disabled, this will cause warnings from the regulator
framework on driver removal. Fix this by adding a remove callback and
disabling the supplies, whilst doing so follow best practice and put the
chip back into reset as well.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add sleep PM callbacks to support system low power modes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support of master mode for cs42l51 cirrus audio codec.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add cs42l51 audio codec power supply management
through regulator framework.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is unsafe to call snd_compr_stop_error from outside of the
compressed ops. Firstly the compressed device lock needs to be held
and secondly it queues error work to issue a trigger stop which
should not happen after the stream has been freed. To avoid these
issues use the same trick used for the IRQ handling, simply send a
snd_compr_fragment_elapsed to cause user-space to wake on the poll,
then report the error when user-space issues the pointer request
after it wakes.
Fixes: a2bcbc1b9a ("ASoC: wm_adsp: Shutdown any compressed streams on DSP watchdog timeout")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@kernel.org
According the publicly available datasheet (and some test) the max98357a
also supports 32, 44.1 and 88.2 kHz sample rate.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PowerTune controls the power level of the chip. On playback this
indirectly controls things like the gain of the various output
amplifiers. This can allow for the decrease of output levels
from the codec. This adds controls for those power levels to
the driver.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a switch for setting common mode voltage. This can allow
for higher drive levels on the amplifier outputs.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch marks RXFIFO_DATA as precious to avoid being read
outside a call from the driver, such as regmap debugfs
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are two identical spelling mistakes in dev_err messages. Fix them.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Mukesh Ojha <mojha@codeaurora.org>
Reviewed-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fix the wrong reg value for rk322x/rk322xh,
cuz there is no STORE JUSTIFIED MODE on it.
on rk322x/rk322xh, the same bit means PDM_MODE/RESERVED,
if the bit is set to RESERVED, the controller will not work.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch make the waterlevel more reasonable, because the pdm
controller share the single FIFO(128 entries) with each channel.
adjust waterlevel in frame to meet the vad or dma frames request.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for rk1808, the pdm controller
is the same as rk3308.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support fractional div for rk3308.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to reset controller every time, do this
once in pdm_probe.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch add default regs value for controller.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch set left justified store mode default.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch decreases the transfer bursts to avoid the fifo overrun.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is because set_fmt ops maybe called when PD is off,
and in such case, regmap_ops will lead system hang.
enale PD before doing regmap_ops.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit da215354eb ("ASoC: simple-card: merge simple-scu-card")
merged simple-scu-audio-card which can handle DPCM into
simple-audio-card.
By this patch, the judgement to select "normal sound card" or
"DPCM sound card" is based on its CPU/Codec DAI count.
But, because of it, existing "simple-audio-card" user who is
assuming "normal sound card" might select DPCM unintentionally.
To solve this issue, this patch allows "simple-audio-card" user
can select "normal sound card", and "simple-scu-audio-card" user
can select both "normal sound card" and "DPCM sound card".
This keeps compatibility collectry.
Fixes: da215354eb ("ASoC: simple-card: merge simple-scu-card")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit ae3cb57909 ("ASoC: audio-graph-card: merge
audio-graph-scu-card") merged audio-graph-scu-card which can
handle DPCM into audio-graph-card.
By this patch, the judgement to select "normal sound card" or
"DPCM sound card" is based on its OF-graph endpoint connection.
But, because of it, existing "audio-graph-card" user who is
assuming "normal sound card" might select DPCM unintentionally.
To solve this issue, this patch allows "audio-graph-card" user
can select "normal sound card", and "audio-graph-scu-card" user
can select both "normal sound card" and "DPCM sound card".
This keeps compatibility collectry.
Fixes: ae3cb57909 ("ASoC: audio-graph-card: merge audio-graph-scu-card")
Reported-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update the copyright dates and use the SPDX identifier instead
of reciting the license.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the unnecessary validation of the 'cstream' variable to fix
below smatch warning:
sprd_platform_compr_drain_notify() warn: variable dereferenced
before check 'cstream' (see line 105)
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The author of these files has changed her name. Update
instances in the code of her dead name to current legal
name.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds a SND_PCI_QUIRK(...) line for the Tuxedo XC 1509.
The Tuxedo XC 1509 and the System76 oryp5 are the same barebone
notebooks manufactured by Clevo. To name the fixups both use after the
actual underlying hardware, this patch also changes System76_orpy5
to clevo_pb51ed in 2 enum symbols and one function name,
matching the other pci_quirk entries which are also named after the
device ODM.
Fixes: 7f665b1c32 ("ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5")
Signed-off-by: Richard Sailer <rs@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Just Implement of reset combo jack JD verb for ACT_PRE_PROBE state.
Intel test result was also failed.
It test passed until changed the initial state to ACT_INIT.
Mic JD will show every time.
This patch also changed the model name as 'alc-chrome-book' for
application of Chrome OS.
Fixes: 10f5b1b85e ("ALSA: hda/realtek - Fixed Headset Mic JD not stable")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Suggested-by: Baolin Wang <baolin.wang@linaro.org>
Tested-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Users have been seeing sound stability issues with max98090 codecs since:
commit 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
At first that commit broke sound for Chromebook Swanky and Clapper models,
the problem was that the machine-driver has been controlling the wrong
clock on those models since support for them was added. This was hidden by
clk-pmc-atom.c keeping the actual clk on unconditionally.
With the machine-driver controlling the proper clock, sound works again
but we are seeing bug reports describing it as: low volume,
"sounds like played at 10x speed" and instable.
When these issues are hit the following message is seen in dmesg:
"max98090 i2c-193C9890:00: PLL unlocked".
Attempts have been made to fix this by inserting a delay between enabling
the clk and enabling and checking the pll, but this has not helped.
It seems that at least on boards which use pmc_plt_clk_0 as clock,
if we ever disable the clk, the pll looses its lock and after that we get
various issues.
This commit fixes this by enabling the clock once at probe time on
these boards. In essence this restores the old behavior of clk-pmc-atom.c
always keeping the clk on on these boards.
Fixes: 648e921888 ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL")
Reported-by: Mogens Jensen <mogens-jensen@protonmail.com>
Reported-by: Dean Wallace <duffydack73@gmail.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When building CONFIG_SND_SOC_MT8183_DA7219_MAX98357A=m
gcc warn this:
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c: In function mt8183_da7219_max98357_dev_probe:
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: struct snd_soc_dai_link has no member named platform; did you mean platforms?
dai_link->platform = NULL;
^~~~~~~~
platforms
use 'dai_link->platforms' instead of 'dai_link->platform'.
Fixes: 11c0269017 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When building CONFIG_SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=m the
following error pops up:
../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c: In function ‘mt8183_mt6358_ts3a227_max98357_dev_probe’:
../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: ‘struct snd_soc_dai_link’ has no member named ‘platform’; did you mean ‘platforms’?
dai_link->platform = NULL;
^~~~~~~~
platforms
Rework to use 'dai_link->platforms' instead of 'dai_link->platform'.
Fixes: 11c0269017 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227")
Signed-off-by: Anders Roxell <anders.roxell@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tidy up some instances of dereferencing to obtain things that are
already stored in local variables.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm_adsp_compr_detach is NULL aware so there is no need to check for NULL
before calling it, remove the redundant check.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trigger stop can be called in situations where trigger start failed
and as such it can't be assumed the buffer is already attached to
the compressed stream or a NULL pointer may be dereferenced.
Fixes: 639e5eb3c7 ("ASoC: wm_adsp: Correct handling of compressed streams that restart")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Acer TravelMate B114-21 laptop cannot detect and record sound from
headset MIC. This patch adds the ALC233_FIXUP_ACER_HEADSET_MIC HDA verb
quirk chained with ALC233_FIXUP_ASUS_MIC_NO_PRESENCE pin quirk to fix
this issue.
[ fixed the missing brace and reordered the entry -- tiwai ]
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Vinod Koul <vkoul@kernel.org>
Cc: Dan Carpenter <dan.carpenter@oracle.com> (commit_signer:1/11=9%,authored:1/11=9%)
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This aligns all kcontrol tplg pointer increments to be consistent
in the respective create methods and ensures that the position is
pointing to the next widget rather the current invalid widget.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the same machine driver is reused between platforms but with a
different alias, using the driver name is not enough. Add additional
fallback case to use the card device name.
Tested on GeminiLake with bxt_da7219_max98357a machine driver
Suggested-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We use 2-stage DMA mode to support Spreadtrum audio compress offload,
which means we use one DMA source channel to transfer data from IRAM
buffer to the DSP fifo to do decoding/encoding, once IRAM buffer is
empty by transferring done, another DMA destination channel will be
triggered automatically to start to transfer data from DDR buffer to
the IRAM buffer. This can reduce the AP subsystem wakeup times to save
power.
Co-developed-by: Yintang Ren <yintang.ren@unisoc.com>
Signed-off-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, buffers, schedulers, src's, encoders, decoders
and effect type dapm widgets remain always on as their
power_check method is not set. Setting this callback allows these
widgets in the audio path to be powered managed properly.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This device can optionally detect headset or microphone button presses.
Add support for this by passing this event to the jack layer.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This device can detect the insertion/removal of headphones and headsets.
Enable reporting this status by enabling this interrupt and forwarding
this to upper-layers if a jack has been defined.
This jack definition and the resulting operation from a jack detection
event must currently be defined by sound card platform code until CODEC
outputs to jack mappings can be defined generically.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If for any reason, the backend does not have the requested substream
(like capture on a playback only backend), the BE will be skipped in
dpcm_be_dai_startup().
However, dpcm_apply_symmetry() does not skip those BE and will
dereference the be_substream (NULL) pointer anyway.
Like in dpcm_be_dai_startup(), just skip those BE.
Fixes: 906c7d690c ("ASoC: dpcm: Apply symmetry for DPCM")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
acpi_dev_get_first_match_name() is deprecated and going to be removed
because it leaks a reference.
Convert the driver to use acpi_dev_get_first_match_dev() instead.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_parse_clk':
sound/soc/generic/simple-card-utils.c:164:18: warning:
parameter 'dai_name' set but not used [-Wunused-but-set-parameter]
It's not used since commit 0580dde594 ("ASoC: simple-card-utils: add
asoc_simple_debug_info()"), so can be removed.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Mukesh Ojha <mojha@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On some devices (Teclast X98+ II tablet, maybe others), the jack
detection has been wired backwards, so when the ES8316 reports
headphones being present it means they are actually not plugged.
Use a quirk around this incorrect behaviour, which can be enabled
through the 'everest,jack-detect-inverted' boolean device property.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_err message. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Mukesh Ojha <mojha@codeaurora.org>
Acked-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for the machine board with
mt6358, da7219 and max98357 codecs.
Signed-off-by: Shunli Wang <shunli.wang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for the machine board with TS3A227.
Signed-off-by: Shunli Wang <shunli.wang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the ACPI ID for the product "chromebook pixel 2015" to match the
coreboot settings.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Support multiple endpoints on cs42L51 codec port
when used in of_graph context.
This patch allows to share the codec port between two CPU DAIs.
Example:
STM32MP157C-DK2 board uses CS42L51 audio codec.
This codec is connected to two serial audio interfaces,
which are configured either as rx or tx.
From AsoC point of view the topolgy is the following:
// 2 CPU DAIs (SAI2A/B), 1 Codec (CS42L51)
Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0
Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0
In the DT two endpoints have to be associated to the codec port:
cs42l51_port: port {
cs42l51_tx_endpoint: endpoint@0 {
remote-endpoint = <&sai2a_endpoint>;
};
cs42l51_rx_endpoint: endpoint@1 {
remote-endpoint = <&sai2b_endpoint>;
};
};
However, when the audio graph card parses the codec nodes, it expects
to find DAI interface indexes matching the endpoints indexes.
The current patch forces the use of DAI id 0 for both endpoints,
which allows to share the codec DAI between the two CPU DAIs
for playback and capture streams respectively.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The common pins were mistakenly not added to the DAPM graph.
Adding these pins will allow valid graphs to be created.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The important fixes at this time are a couple fixes in ALSA core:
a fix for PCM is about the OOB access in PCM OSS plugins that has
been for long time, but hasn't hit so often until now just because
we allocated a large buffer via vmalloc(), and surfaced more often
after switching to kvmalloc(). Another fix is for a long-standing
PCM problem wrt racy PM resume. Others are trivial nospec coverage
and usual HD-audio quirks.
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Merge tag 'sound-5.1-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The important fixes at this time are a couple fixes in ALSA core: a
fix for PCM is about the OOB access in PCM OSS plugins that has been
for long time, but hasn't hit so often until now just because we
allocated a large buffer via vmalloc(), and surfaced more often after
switching to kvmalloc(). Another fix is for a long-standing PCM
problem wrt racy PM resume.
Others are trivial nospec coverage and usual HD-audio quirks"
* tag 'sound-5.1-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fix speakers on Acer Predator Helios 500 Ryzen laptops
ALSA: pcm: Don't suspend stream in unrecoverable PCM state
ALSA: hda/ca0132 - Simplify alt firmware loading code
ALSA: pcm: Fix possible OOB access in PCM oss plugins
ALSA: hda/realtek: Enable headset MIC of ASUS X430UN and X512DK with ALC256
ALSA: hda/realtek: Enable headset mic of ASUS P5440FF with ALC256
ALSA: hda/realtek: Enable ASUS X441MB and X705FD headset MIC with ALC256
ALSA: hda/realtek - Add support for Acer Aspire E5-523G/ES1-432 headset mic
ALSA: hda/realtek: Enable headset MIC of Acer Aspire Z24-890 with ALC286
ALSA: seq: oss: Fix Spectre v1 vulnerability
ALSA: rawmidi: Fix potential Spectre v1 vulnerability
alloc_pages_exact() is more suitable choice for allocating the sound
buffers, as it doesn't need to align with power-of-two. Along with
the conversion, we can drop __GFP_COMP as well.
The patch also replace the error messages to be more explicit.
Acked-by: Michal Hocko <mhocko@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_malloc_pages() and snd_free_pages() are merely thin wrappers of
the standard page allocator / free functions. Even the arguments are
compatible with some standard helpers, so there is little merit of
keeping these wrappers.
This patch replaces the all existing callers of snd_malloc_pages() and
snd_free_pages() with the direct calls of the standard helper
functions. In this version, we use a recently introduced one,
alloc_pages_exact(), which suits better than the old
snd_malloc_pages() implementation for our purposes. Then we can avoid
the waste of pages by alignment to power-of-two.
Since alloc_pages_exact() does split pages, we need no longer
__GFP_COMP flag; or better to say, we must not pass __GFP_COMP to
alloc_pages_exact(). So the former unconditional addition of
__GFP_COMP flag in snd_malloc_pages() is dropped, as well as in most
other places.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Michal Hocko <mhocko@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_timer_close() is supposed to close the timer instance and sync
with the deactivation of pending actions. However, there are still
some overlooked cases:
- It calls snd_timer_stop() at the beginning, but some other might
re-trigger the timer right after that.
- snd_timer_stop() calls del_timer_sync() only when all belonging
instances are closed. If multiple instances were assigned to a
timer object and one is closed, the timer is still running. Then
the pending action assigned to this timer might be left.
Actually either of the above is the likely cause of the reported
syzkaller UAF.
This patch plug these holes by introducing SNDRV_TIMER_IFLG_DEAD
flag. This is set at the beginning of snd_timer_close(), and the flag
is checked at snd_timer_start*() and else, so that no longer new
action is left after snd_timer_close().
Reported-by: syzbot+d5136d4d3240cbe45a2a@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For checking the pending timer instance that is still left on the
timer object that is being closed, we set/clear a bit flag
SNDRV_TIMER_IFLG_CALLBACK around the call of callbacks. This can be
simplified by replace with the list_empty() call for ti->ack_list.
This covers the existence more comprehensively and safely.
A gratis bonus is that we can get rid of SNDRV_TIMER_IFLG_CALLBACK bit
flag definition as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a card is under disconnection, we bail out immediately at each
timer interrupt or tasklet. This might leave some items left in ack
list. For a better integration of the upcoming change to check
ack_list emptiness, clear out the whole list upon the emergency exit
route.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timer core has two almost identical code for processing callbacks:
once in snd_timer_interrupt() for fast callbacks and another in
snd_timer_tasklet() for delayed callbacks. Let's unify them.
In the new version, the resolution is read from ti->resolution at each
call, and this must be fine; ti->resolution is set in the preparation
step in snd_timer_interrupt().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's a feature request for the ancient sutff, but it's still valid;
the loading of a GUS-patch isn't available via hwdep device although
it's supported over OSS sequencer. The only missing piece is the call
of snd_soundfont_load_guspatch() in synth emux hwdep code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers
don't work out of the box.
The problem can be worked around with hdajackretask, remapping the
"Black Headphone, Right side" pin (0x21) to the Internal speaker.
This patch adds a quirk to change this mapping by default.
[ corrected ALC299_FIXUP_PREDATOR_SPK definition and adapted for the
latest tree by tiwai ]
Signed-off-by: Bernhard Rosenkraenzer <bero@lindev.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements Audio Mixer machine driver for NXP iMX8 SOCs.
It connects together Audio Mixer and related SAI instances.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs.
The Audio Mixer is a on-chip functional module that allows mixing of
two audio streams into a single audio stream.
Audio Mixer datasheet is available here:
https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To enable S24_LE format, sample_type in topology fw has to be set to 1.
But sample_type defined in topology firmware configuration is not
getting reflected in the dsp param. This patch sets sample_type in base
config so that the sample type defined in the topology firmware is reflected
in the dsp params. This issues was uncovered while debugging the S24_LE format
which require the MSB byte in 32 bit word to be skipped. Setting sample_type
in topology firmware to 1 helps to skip MSB byte word.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some architectures do not yet support the common clock API at all but
the tlv320aic32x4 driver now requires it.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@kernel.org>
UAPI Changes:
- Report an error early instead of SIGBUS later when mmap beyond BO size
Core Changes:
- This includes backmerge of drm-next and two merges of Maarten's
topic/hdr-formats
Driver Changes:
- Add Comet Lake (Gen9) PCI IDs to Coffee Lake ID list (Anusha)
- Add missing ICL PCI ID (Jose)
- Fix legacy gamma mode for ICL (Ville)
- Assume eDP is present on port A when there is no VBT (Thomas)
- Corrections to eDP training patterns (Jose)
- Fix PSR2 selective update corruption after PSR1 setup (Jose)
- Fix CRC mismatch error for DP link layer compliance (Aditya)
- Fix CNL DPLL readout and clean up code (Ville)
- Turn off the CUS when turning off a HDR plane (Ville)
- Avoid a race with execlist tasklet during race (Chris)
- Add missing CSC readout and clean up code (Ville)
- Avoid unnecessary wakeref during debugfs/drop_caches/set (Chris, Caz)
- Hold references to ring/HW context/context explicitly when used (Chris)
- Assume next platforms inherit old platform (Rodrigo)
- Use HWS indices rather than addresses for breadcrumbs (Chris)
- Add REG_BIT/REG_GENMASK and REG_FIELD_PREP macros (Jani)
- Convert crept in C99 types to kernel fixed size types (Jani)
- Avoid passing full dev_priv in forcewake functions (Daniele)
- Reset GuC on GPU reset (Sujaritha)
- Rework MG and Combo PLLs to vfuncs (Lucas)
- Explicitly track ppGTT size (Chris, Bob)
- Coding style improvements and code modularization (Ville)
- Selftest and debugging improvements (Chris)
Signed-off-by: Dave Airlie <airlied@redhat.com>
# Conflicts:
# drivers/gpu/drm/i915/intel_hdmi.c
From: Joonas Lahtinen <joonas.lahtinen@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20190325124925.GA12726@jlahtine-desk.ger.corp.intel.com
The clocking and processing blocks are now properly set up to
support 192000 sample rates. Allow drivers to ask for that.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
mclk is not used by anything anymore. Remove support for it.
All that information now comes from the clock tree.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The sysclk is now managed by the CCF. Change this function
to merely find the system clock and set it using
clk_set_rate.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code uses a static lookup table to determine the
settings of the various clock devices on board the chip. This is
limiting in a couple of ways. First, this doesn't allow for any
master clock rates other than the three that have been
precalculated. Additionally, new sample rates are difficult to
add to the table. Witness that the chip is capable of 192000 Hz
sampling, but it is not provided by this driver. Last, if the
driver is clocked by something that isn't a crystal, the
upstream clock may not be able to achieve exactly the rate
requested in the driver. This will mean that clocking will be
slightly off for the sampling clock or that it won't work at all.
This patch determines the settings for all of the clocks at
runtime considering the real conditions of the clocks in the
system. The rules for the clocks are in TI's SLAA557 application
guide on pages 37, 51 and 77.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move these to separate helper functions. This looks cleaner and fits
better with the new clock setting in CCF.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Control the clock gating to the various clock components to use
the CCF. This allows us to prepare_enalbe only 3 clocks and the
relationships assigned to them will cause upstream clockss to
enable automatically. Additionally we can do this in a single
call to the CCF.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage BDIV divider as components in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage DAC/ADC dividers as components in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage codec clock input as a component in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage the on-board PLL as a component in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of single config, private_value is left uninitialized.
The private_value does need to be initialized or in
snd_soc_dapm_new_control_unlocked() call failure case, it leads to a
bogus free in snd_soc_dapm_free_kcontrol()
Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently snd_aloop supports only S16 and S32 audio sample formats. With
this patch the S24 formats are also supported.
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently PCM core sets each opened stream forcibly to SUSPENDED state
via snd_pcm_suspend_all() call, and the user-space is responsible for
re-triggering the resume manually either via snd_pcm_resume() or
prepare call. The scheme works fine usually, but there are corner
cases where the stream can't be resumed by that call: the streams
still in OPEN state before finishing hw_params. When they are
suspended, user-space cannot perform resume or prepare because they
haven't been set up yet. The only possible recovery is to re-open the
device, which isn't nice at all. Similarly, when a stream is in
DISCONNECTED state, it makes no sense to change it to SUSPENDED
state. Ditto for in SETUP state; which you can re-prepare directly.
So, this patch addresses these issues by filtering the PCM streams to
be suspended by checking the PCM state. When a stream is in either
OPEN, SETUP or DISCONNECTED as well as already SUSPENDED, the suspend
action is skipped.
To be noted, this problem was originally reported for the PCM runtime
PM on HD-audio. And, the runtime PM problem itself was already
addressed (although not intended) by the code refactoring commits
3d21ef0b49 ("ALSA: pcm: Suspend streams globally via device type PM
ops") and 17bc4815de ("ALSA: pci: Remove superfluous
snd_pcm_suspend*() calls"). These commits eliminated the
snd_pcm_suspend*() calls from the runtime PM suspend callback code
path, hence the racy OPEN state won't appear while runtime PM.
(FWIW, the race window is between snd_pcm_open_substream() and the
first power up in azx_pcm_open().)
Although the runtime PM issue was already "fixed", the same problem is
still present for the system PM, hence this patch is still needed.
And for stable trees, this patch alone should suffice for fixing the
runtime PM problem, too.
Reported-and-tested-by: Jon Hunter <jonathanh@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
w_text_param can be NULL and it is being dereferenced without checking.
Add the missing sanity check to prevent NULL pointer dereference.
Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Unlike other drivers probe method, of_match_node return value
is not used or checked. This patch removes the redundant code.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Reviewed-by: Steven Price <steven.price@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support of low power modes to STM32 SAI driver.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When using bare externs outside include files that types should
at least match. This fixes a type confusion between bool
and int.
Cc: broonie@kernel.org
Signed-off-by: Andi Kleen <ak@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do division with div_u64 for the PLL calculation.
These errors are fixed and list as follows:
1."__udivdi3" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
2."__aeabi_uldivmod" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
3. nau8810.c:(.text.nau8810_calc_pll+0xd8): undefined reference to
`__udivdi3'
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The only significant change is the regression fixes for the jack
detection at resume on HD-audio, while others are all small or
trivial fixes like the coverage of missing error code or usual
HD-audio quirk.
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Merge tag 'sound-5.1-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The only significant change is the regression fixes for the jack
detection at resume on HD-audio, while others are all small or trivial
fixes like the coverage of missing error code or usual HD-audio quirk"
* tag 'sound-5.1-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: Enable headset MIC of Acer AIO with ALC286
ALSA: hda - Enforces runtime_resume after S3 and S4 for each codec
ALSA: hda - Don't trigger jackpoll_work in azx_resume
ALSA: opl3: fix mismatch between snd_opl3_drum_switch definition and declaration
ALSA: hda - add Lenovo IdeaCentre B550 to the power_save_blacklist
ALSA: firewire-motu: use 'version' field of unit directory to identify model
ALSA: sb8: add a check for request_region
ALSA: echoaudio: add a check for ioremap_nocache
ca0132 codec driver loads the firmware selectively depending on the
model in addition to the fallback of the default firmware. The code
works good, but a minor problem is that the current code seems
confusing for Clang where it spews a warning about uninitialized
variable.
This patch simplifies the code flow for such a false-positive
warning. After this refactoring, the ca0132_spec.alt_firmware_present
field is no longer used, hence it's eliminated as well.
Reported-and-tested-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM OSS emulation converts and transfers the data on the fly via
"plugins". The data is converted over the dynamically allocated
buffer for each plugin, and recently syzkaller caught OOB in this
flow.
Although the bisection by syzbot pointed out to the commit
65766ee0bf ("ALSA: oss: Use kvzalloc() for local buffer
allocations"), this is merely a commit to replace vmalloc() with
kvmalloc(), hence it can't be the cause. The further debug action
revealed that this happens in the case where a slave PCM doesn't
support only the stereo channels while the OSS stream is set up for a
mono channel. Below is a brief explanation:
At each OSS parameter change, the driver sets up the PCM hw_params
again in snd_pcm_oss_change_params_lock(). This is also the place
where plugins are created and local buffers are allocated. The
problem is that the plugins are created before the final hw_params is
determined. Namely, two snd_pcm_hw_param_near() calls for setting the
period size and periods may influence on the final result of channels,
rates, etc, too, while the current code has already created plugins
beforehand with the premature values. So, the plugin believes that
channels=1, while the actual I/O is with channels=2, which makes the
driver reading/writing over the allocated buffer size.
The fix is simply to move the plugin allocation code after the final
hw_params call.
Reported-by: syzbot+d4503ae45b65c5bc1194@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS X430UN and X512DK with ALC256 cannot detect the headset MIC
until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS laptop P5440FF with ALC256 can't detect the headset microphone
until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
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>
The ASUS laptop X441MB and X705FD with ALC256 cannot detect the headset
MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
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>
li->conf will be 0 if it was not DPCM case.
Then, 1) we shouldn't call devm_kcalloc() with size 0,
2) we need NULL pointer check if li->conf was not 0.
This patch fixed above issues.
Special thanks to Pierre-Louis Bossart
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Lochnagar is an evaluation and development board for Cirrus
Logic Smart CODEC and Amp devices. It allows the connection of
most Cirrus Logic devices on mini-cards, as well as allowing
connection of various application processor systems to provide a
full evaluation platform.
Lochnagar 2 provides a set of line inputs/outputs, and a USB audio
device. This driver adds support for these analog line connections and
the Lochnagar side of the USB audio link.
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Different processing blocks are required for different sampling
rates and power parameters. Set the processing blocks based
on this information.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Inter-IC Sound Controller (I2SMCC) provides a 5-wire, bidirectional,
synchronous, digital audio link to external audio devices: I2SMCC_DIN,
I2SMCC_DOUT, I2SMCC_WS, I2SMCC_CK, and I2SMCC_MCK pins.
The I2SMCC complies with the Inter-IC Sound (I2S) bus specification and
supports a Time Division Multiplexed (TDM) interface with external
multi-channel audio codecs.
The I2SMCC consists of a receiver, a transmitter and a common clock
generator that can be enabled separately to provide Master, Slave or
Controller modes with receiver and/or transmitter active.
DMA Controller channels, separate for the receiver and for the transmitter,
allow a continuous high bit rate data transfer without processor
intervention to the following:
- Audio CODECs in Master, Slave, or Controller mode
- Stereo DAC or ADC through a dedicated I2S serial interface
- Multi-channel or multiple stereo DACs or ADCs, using the TDM format
This IP is embedded in Microchip's new sam9x60 SoC.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils is using asoc_simple_card_xxx() for each
function naming, but it is very verbose.
Thus it is easy to be over 80 char.
This patch renames it to asoc_simple_xxx().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph are using
asoc_simple_card_parse_dai() which is different implementation.
But, these are implemanted at simple-card-utils.
It should be implemanted at each files.
This patch separate these into each files.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph are initializing each priv,
but it is same operation.
This patch adds new asoc_simple_card_init_priv() and initialize
priv by same operation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_be_hw_params_fixup() between in these
2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_dai_init() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_hw_param() between in these 2 drivers.
One note is that only simple-card supports simple_set_clk_rate()
at hw_param from commit e9be4ffd4f ("ASoC: simple-card: set cpu
dai clk in hw_params").
By this patch, audio-graph has same feature.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_shutdown() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_startup() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Historically, simple-card/simple-scu-card/audio-graph/audio-graph-scu
are similar but different generic sound card.
simple-scu-card which was for DPCM was merged into simple-card, and
audio-graph-scu which was for DPCM was merged into audio-graph.
simple-card is for non OF graph sound card, and
audio-graph is for OF graph sound card.
And, small detail difference (= function parameter, naming, etc)
between simple-card/audio-graph has been unified.
So today, the difference between simple-card/audio-graph are
just using OF graph style, or not.
In other words, there should no difference other than OF graph sytle.
simple-card/audio-graph are using own priv today , but we can merge it.
This patch merge it at simple_card_utils.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils has dev_dbg(), but people want to
add #define DEBUG at simple-card/audio-graph, not simple-card-utils.
And, people want to get all information.
This patch adds new asoc_simple_debug_info() to indicates information.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the DAI clocks for DA7219 have now been split into BCLK and WCLK,
the clock lookup name needs to be udpated here to select BCLK to
achieve the same functionality as before with regards to DAI clock
gating.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For the purposes of platforms which use the codec as DAI clock
master for the CPU and other codec devices, there is the need to
not only expose the clock gating of BCLK and WCLK but also the
ability to set those rates without going through the ASoC APIs.
To make this possible, the previous CCF implementation in the
driver has been extended to separate BCLK and WCLK out. WCLK is
the parent clock to BCLK, and is also the clock gate for both.
BCLK in HW is a factor/multiplier of WCLK so derives from whatever
SR is chosen for WCLK, hence the need to make it a child of WCLK
for the purposes of CCF. Enabling/disabling either BCLK or WCLK
will result in clocks being ungated/gated accordingly. To simplify
matters, these clocks can only be configured if the codec is set
as master, otherwise CCF control is disallowed.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is very low possibility ( < 0.1% ) that channel swap happened
in beginning when multi output/input pin is enabled. The issue is
that hardware can't send data to correct pin in the beginning with
the normal enable flow.
This is hardware issue, but there is no errata, the workaround flow
is that: Each time playback/recording, firstly clear the xSMA/xSMB,
then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled
before xSMA). Which is to use the xSMA as the trigger start register,
previously the xCR_TE or xCR_RE is the bit for starting.
Fixes commit 43d24e76b6 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Cc: <stable@vger.kernel.org>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a constraint for the channel number setting on the
asrc of older version (e.g. imx35), the channel number should
be even, odd number isn't valid.
So add this constraint when the asrc of older version is used.
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The CS4270 does not by default increment the register address on
consecutive writes. During normal operation it doesn't matter as all
register accesses are done individually. At resume time after suspend,
however, the regcache code gathers the biggest possible block of
registers to sync and sends them one on one go.
To fix this, set the INCR bit in all cases.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect
the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk
applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
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>
The Acer Aspire Z24-890 cannot detect the headset MIC until
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap)
Fix this by sanitizing dev before using it to index dp->synths.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info->stream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap)
Fix this by sanitizing info->stream before using it to index
rmidi->streams.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot
record sound from headset MIC. This patch adds the
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue.
Fixes: 9f8aefed96 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G")
Fixes: b72f936f6b ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register platform component with a prefix, to avoid warnings
on debugfs entries creation, due to component name
redundancy.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DFSDM must be stopped when a new setting is applied.
restart systematically DFSDM on multiple prepare calls,
to apply changes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The wm_adsp_ops structures should be static and correct two printf
specifiers.
Fixes: 170b1e123f ("ASoC: wm_adsp: Add support for new Halo core DSPs")
Fixes: 4e08d50d1f ("ASoC: wm_adsp: Factor out DSP specific operations")
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can simplify the code by caching the CPU DAI master/slave
information rather than reading previously set register bit.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Break the clock setting logic out from the main hw_params. It's
rather large and unweildy and makes for a large function. This
also better enables some of the following changes to the clock
tree access in the driver.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Depending on MACH_JZ4740 prevent us from creating a generic kernel that
works on more than one MIPS board. Instead, we just depend on MIPS being
set.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A Halo Core DSP has a memory protection unit that can trap and signal
memory access faults. This patch adds a function that dumps the fault
information.
The interrupt reaches the host via the parent codec interrupt controller
so this fault function is exported to be called by the codec driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Halo core is a new generation of audio DSP architecture from
Cirrus Logic. A new iteration of the WMFW file format (v3) is also
added, for this new architecture. Currently this format is not
supported on the old ADSP2 architecture however support may be
added for it in the future.
Signed-off-by: Wen Shi <wenshi@opensource.cirrus.com>
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the signature of mtk_regmap_update_bits to also take a shift, and
warn when reg >= 0 but shift < 0. This reduce the code repetition
on the calling side, and prevent future UBSAN warning when some of the
xxx_shift and xxx_reg are both set to -1.
Signed-off-by: Pi-Hsun Shih <pihsun@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In sound/soc/mediatek/common/mtk-afe-fe-dai.c, when xxx_reg is -1, it's
a no-op to call mtk_regmap_update_bits, but since both xxx_reg and
xxx_shift are set to -1, the (1 << xxx_shift) in the argument would
trigger a UBSAN warning.
Fix the warning by setting those xxx_shift to 0 instead.
Note that since the code explicitly checks .mono_shift >= 0 and
.fs_shift >= 0 before using them in '<<' operator, those two members are
not set to 0.
Signed-off-by: Pi-Hsun Shih <pihsun@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for the addition of more types of DSP core refactor the
handling of DSP specific operations such as starting the memory or
enabling the core into a set of callbacks. This should make it easier to
add new core types and allow for more code reuse between them.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to duplicate this code for both ADSP1 and 2 as the
handling is exactly the same.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for further additions refactor the reading of the
firmware status.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The original wm_adsp2_early_event took an additional frequency
argument for clocking control so could not be used directly as a
DAPM callback. But this setup could equally be done by the codec
driver function wrapping wm_adsp2_early event. In preparation
for adding support for new core types wm_adsp2_set_dspclk has
been exported, and the freq argument removed so that it can
be used directly as a DAPM callback.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This function is not presently called from outside the adsp code and nor
should it be, as such stop exporting it.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a watchdog timeout is received from the DSP it is safe to assume the
DSP is not functioning anymore and as such any active compressed streams
should be put into an error state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Best to lock across handling the bus error to ensure the DSP doesn't
change power state as we are reading the status registers.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During recent logging improvements it seems two error messages lost
their updates during patch application/rebasing. Add these back in.
Fixes: 0d3fba3e7a ("ASoC: wm_adsp: Improve logging messages")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously support was added to allow streams to be stopped and
started again without the DSP being power cycled and this was done
by clearing the buffer state in trigger start. Another supported
use-case is using the DSP for a trigger event then opening the
compressed stream later to receive the audio, unfortunately clearing
the buffer state in trigger start destroys the data received
from such a trigger. Correct this issue by moving the call to
wm_adsp_buffer_clear to be in trigger stop instead.
Fixes: 61fc060c40 ("ASoC: wm_adsp: Support streams which can start/stop with DSP active")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some strings are allocated by kstrdup, but not freed when error
happened.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The stream_name is allocated by kstrdup. We have to free it when the
dai is removed or return from error.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d ("ALSA: hda - Use standard runtime PM for codec power-save control")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 3baffc4a84 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Fixes: 3baffc4a84 ("ALSA: hda/intel: Refactoring PM code")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sylwester Nawrocki