This patch moves clock enable/disable from system resume/suspend to
runtime resume/suspend respectively. Along with this hda controller
chip init or stop is also moved. System resume/suspend can invoke
runtime callbacks and do necessary setup.
chip->running can be used to check for probe completion and device
access during runtime_resume or runtime_suspend can be avoided if
probe is not yet finished. This helps to avoid kernel panic during
boot where runtime PM callbacks can happen from system PM.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Explicit clock enable is not required during probe, as this would be
managed by runtime PM calls. Clock can be enabled/disabled in runtime
resume/suspend. This way it is easier to balance clock enable/disable
counts.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved devm_clk_get() API calls to a separate function and the same
can be called early in the probe. This is done before runtime PM
for the device is enabled. The runtime resume/suspend callbacks can
later enable/disable clocks respectively(the support would be added
in subsequent patches). Clock handles should be available by the
time runtime suspend/resume calls can happen.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables runtime power management(runtime PM) support for
hda. pm_runtime_enable() and pm_runtime_disable() are added during
device probe and remove respectively. The runtime PM callbacks will
be forbidden if hda controller does not have support for runtime PM.
pm_runtime_get_sync() and pm_runtime_put() are added for hda register
access. The callbacks for above will be added in subsequent patches.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface UFX was shipped by RME GmbH in 2012. This model supports later
protocol for management of isochronous communication and synchronization
of sampling transmission frequency.
This commit adds support for the model. At present, it's not clear how
to encode MIDI messages and decide destination address for asynchronous
transaction, thus this commit adds support for isochronous communication
for PCM frames only.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A procedure to retrieve clock configuration is used by two callers.
Each of caller has duplicated code to parse bits.
This commit adds refactoring to remove the duplicated code.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds refactoring for dump of sync status by adding
tables for check bits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a member for a callback function to get clock status
to former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a member for a callback function to switch frame
fetching mode to former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a member for a callback function to dump status and
move existing code to former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a series of Fireface, latter protocol has no way for drivers to
retrieve current clock configuration. On the other hand, this driver
has proc node for it.
This commit removes a proc node to dump both clock configuration
and synchronization status in one proc node.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit moves codes for Fireface 400 to a file of former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a series of Fireface, later model supports different protocol
from former models.
This commit is a preparation to support both of protocols.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unused variable was forgotten to be removed and now we get a
compiler warning:
sound/pci/hda/hda_codec.c: In function 'hda_codec_runtime_suspend':
sound/pci/hda/hda_codec.c:2926:18: warning: unused variable 'pcm'
Fixes: 17bc4815de ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull the PCM suspend improvement / cleanup.
This moves the most of snd_pcm_suspend*() calls into PCM's own device
PM ops. There should be no change from the functionality POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_suspend() is no longer called from outside, so let's make it
local static. Also drop a superfluous NULL check there.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATIIXP driver supports the full PCM resume and saves/restores the
running PCM pointer. This used to be done in the suspend and resume
callbacks together with snd_pcm_suspend() call. But since we moved
the snd_pcm_supsend*() call in PCM device PM ops, this should be moved
to a more appropriate place, i.e. the trigger callback.
Along with the movement of the PCM suspend/resume code, remove the
superfluous snd_pcm_suspend_all() call, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Until now we rely on each driver calling snd_pcm_suspend*() explicitly
at its own PM handling. However, this can be done far more easily by
setting the PM ops to each actual snd_pcm device object.
This patch adds the device_type object for PCM stream and assigns to
each PCM stream object. The type contains only the PM ops for system
suspend; we don't need to deal with the resume in general.
The suspend hook simply calls snd_pcm_suspend_all() for the given PCM
streams. This implies that the PM order is correctly put, i.e. PCM is
suspended before the main (or codec) driver, which should be true in
general. If a special ordering is needed, you'd need to adjust the
device PM order manually later.
This patch introduces a new flag, snd_pcm.no_device_suspend, too.
With this flag set, the PCM device object won't invoke
snd_pcm_suspend_all() by itself. This is needed for ASoC who wants to
manage the PM call orders in its serialized way, and the flag is set
in soc_new_pcm() as default.
For the non-ASoC world, we can get rid of the manual snd_pcm_suspend
calls. This will be done in the later patches.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Program codec stripe through AC_VERB_SET_STRIPE_CONTROL to use multiple
sdo lines if supported. Audio needs to be striped across number of sdo
lines for simultaneous playbacks of higher resolutions to work.
This needs to be implemented only for an Audio Output Converter and only
if the stripe bit(AC_WCAP_STRIPE) of Audio Widget Capabilities parameter
is 1.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Platforms having multiple SORs and hdmi/dp sinks require higher
bandwidth to support simultaneous playbacks of higher resolution.
If hda controller supports multiple SDO lines, STRIPE can be used
to indicate how many of the SDO lines the stream should be striped
across.
During stream start stripe control bits are programmed to use given
number of sdo lines and the same is cleared during stream stop.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Controllers and codecs can support striping of audio out across
multiple SDO lines. The number of supported SDO lines can be
specific to chip. GCAP register can be read to know the maximum
supported SDO lines.
snd_hdac_get_stream_stripe_ctl() is exposed to program stripe bits
on controller and codec side.
stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines, etc.,
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/usb/mixer.c: In function 'parse_audio_feature_unit':
sound/usb/mixer.c:1838:28: warning:
variable 'first_ch_bits' set but not used [-Wunused-but-set-variable]
It never used since 2.6
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To enable SIE(Stream Interrupt Enable) in snd_hdac_stream_start(), we
should set both mask and value to be "1 << azx_dev->index" for register
update, the mask was 0, here fix it.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. for azx_int_enable(), we should set both mask and value to be
"AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN"(the mask was 0) to enable
controller CIE and GIE.
We have similar issues on setting AZX_GCTL_RESET and AZX_GCTL_UNSOL,
here try to correct all of them.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ctl_add() could fail, so let's check its return value and return its
error code upstream upon failure.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix checks if snd_card_register() fails, and if so logs the error
via dev_err() consistent with other patches.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_i2c_sendbytes could fail. The fix checks its return value: if it
fails, issues an error message and returns with its error code.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ctl_add() could fail, so let's check its status and issue an error
message if it indeed fails.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's no reason for us to do that while we initialize dac_mute to
1. Also oxygen_init() has been clearing the OXYGEN_SPDIF_OUT_ENABLE
bit anyway.
Signed-off-by: Tom Yan <tom.ty89@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell has new platform for ALC274.
This will support to enable headset mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In `create_composite_quirk`, the terminating condition of for loops is
`quirk->ifnum < 0`. So any composite quirks should end with `struct
snd_usb_audio_quirk` object with ifnum < 0.
for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) {
.....
}
the data field of Bower's & Wilkins PX headphones usb device device quirks
do not end with {.ifnum = -1}, wihch may result in out-of-bound read.
This Patch fix the bug by adding an ending quirk object.
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places where we access the data without checking the
actual object size from the USB audio descriptor. This may result in
OOB access, as recently reported.
This patch addresses these missing checks. Most of added codes are
simple bLength checks in the caller side. For the input and output
terminal parsers, we put the length check in the parser functions.
For the input terminal, a new argument is added to distinguish between
UAC1 and the rest, as they treat different objects.
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Reported-by: Hui Peng <benquike@163.com>
Tested-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had some sanity checks of the mixer unit descriptors but they
are too loose and some corner cases are overlooked. Add more strict
checks in uac_mixer_unit_get_channels() for avoiding possible OOB
accesses by malformed descriptors.
This also changes the semantics of uac_mixer_unit_get_channels()
slightly. Now it returns zero for the cases where the descriptor
lacks of bmControls instead of -EINVAL. Then the caller side skips
the mixer creation for such unit while it keeps parsing it.
This corresponds to the case like Maya44.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The parser for the processing unit reads bNrInPins field before the
bLength sanity check, which may lead to an out-of-bound access when a
malformed descriptor is given. Fix it by assignment after the bLength
check.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nobody has actually used the type (VERIFY_READ vs VERIFY_WRITE) argument
of the user address range verification function since we got rid of the
old racy i386-only code to walk page tables by hand.
It existed because the original 80386 would not honor the write protect
bit when in kernel mode, so you had to do COW by hand before doing any
user access. But we haven't supported that in a long time, and these
days the 'type' argument is a purely historical artifact.
A discussion about extending 'user_access_begin()' to do the range
checking resulted this patch, because there is no way we're going to
move the old VERIFY_xyz interface to that model. And it's best done at
the end of the merge window when I've done most of my merges, so let's
just get this done once and for all.
This patch was mostly done with a sed-script, with manual fix-ups for
the cases that weren't of the trivial 'access_ok(VERIFY_xyz' form.
There were a couple of notable cases:
- csky still had the old "verify_area()" name as an alias.
- the iter_iov code had magical hardcoded knowledge of the actual
values of VERIFY_{READ,WRITE} (not that they mattered, since nothing
really used it)
- microblaze used the type argument for a debug printout
but other than those oddities this should be a total no-op patch.
I tried to fix up all architectures, did fairly extensive grepping for
access_ok() uses, and the changes are trivial, but I may have missed
something. Any missed conversion should be trivially fixable, though.
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Among a few HD-audio fixes, the only significant one is the
regression fix on some machines like Dell XPS due to the default
binding changes. We ended up reverting the whole since the fix for
ASoC HD-audio driver won't be available immediately.
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Merge tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Among a few HD-audio fixes, the only significant one is the regression
fix on some machines like Dell XPS due to the default binding changes.
We ended up reverting the whole since the fix for ASoC HD-audio driver
won't be available immediately"
* tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Revert DSP detection on legacy HD-audio driver
ALSA: hda/tegra: clear pending irq handlers
ALSA: hda/realtek: Enable the headset mic auto detection for ASUS laptops
This essentially reverts the commits
c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present
and Skylake driver selected")
and
d82b51c855 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+
driver selection")
for the path of legacy HD-audio controller (snd-hda-intel).
The automatic DSP detection and skip of binding with the legacy driver
caused regressions on several machines like Dell XPS13. They give the
PCI class 0x40380 indicating the availability of DSP while they don't
work with ASoC SKL driver (yet).
As the support of ASoC driver for such devices isn't available, it's
better to revert the whole DSP-detection-and-skip behavior of the
legacy driver, so that we can get the old good driver working on such
devices.
The pci_binding option for ASoC SKL driver is still kept so that it
can work without blacklisting.
Fixes: c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected")
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Reported-by: Hans de Goede <hdegoede@redhat.com>
Reported-by: Azat Khuzhin <dohardgopro@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even after disabling interrupts on the module, it could be possible
that irq handlers are still running. System hang is seen during
suspend path. It was found that, there were pending writes on the
HDA bus and clock was disabled by that time.
Above mentioned issue is fixed by clearing any pending irq handlers
before disabling clocks and returning from hda suspend.
Suggested-by: Mohan Kumar <mkumard@nvidia.com>
Suggested-by: Dara Ramesh <dramesh@nvidia.com>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The headset mic of ASUS laptops like UX533FD, UX433FN and UX333FA, whose
CODEC is Realtek ALC294 has jack auto detection feature. This patch
enables the feature.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
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>
