The current code performs the cancel of a delayed work at the late
stage of disconnection procedure, which may lead to the access to the
already cleared state.
This patch assures to call cancel_delayed_work_sync() at the beginning
of the disconnection procedure for avoiding that race. The delayed
work object is now assigned in the common line6 object instead of its
derivative, so that we can call cancel_delayed_work_sync().
Along with the change, the startup function is called via the new
callback instead. This will make it easier to port other LINE6
drivers to use the delayed work for startup in later patches.
Reported-by: syzbot+5255458d5e0a2b10bbb9@syzkaller.appspotmail.com
Fixes: 7f84ff68be ("ALSA: line6: toneport: Fix broken usage of timer for delayed execution")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The most significant changes at this cycle are the Sound Open Firmware
support from Intel for the common DSP framework along with its support
for Intel platforms. It's a door opened to a real "free" firmware (in
the sense of FOSS), and other parties show interests in it.
In addition to SOF, we've got a bunch of updates and fixes as usual.
Some highlights are below.
ALSA core:
- Cleanups and fixes in ALSA timer code to cover some races spotted
by syzkaller
- Cleanups and fixes in ALSA sequencer code to cover some races,
again unsurprisingly, spotted by syzkaller
- Optimize the common page allocation helper with alloc_pages_exact()
ASoC:
- Add SOF core support, as well as Intel SOF platform support
- Generic card driver improvements: support for MCLK/sample rate
ratio and pin switches
- A big set of improvements to TLV320AIC32x4 drivers
- New drivers for Freescale audio mixers, several Intel machines,
several Mediatek machines, Meson G12A, Spreadtrum compressed audio
and DMA devices
HD-audio:
- A few Realtek codec fixes for reducing pop noises
- Quirks for Chromebooks
- Workaround for faulty connection report on AMD/Nvidia HDMI
Others:
- A quirk for Focusrite Scarlett Solo USB-audio
- Add support for MOTU 8pre FireWire
- 24bit sample format support in aloop
- GUS patch format support (finally, over a decade) in native
emux synth code
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Merge tag 'sound-5.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"The most significant changes at this cycle are the Sound Open Firmware
support from Intel for the common DSP framework along with its support
for Intel platforms. It's a door opened to a real "free" firmware (in
the sense of FOSS), and other parties show interests in it.
In addition to SOF, we've got a bunch of updates and fixes as usual.
Some highlights are below.
ALSA core:
- Cleanups and fixes in ALSA timer code to cover some races spotted
by syzkaller
- Cleanups and fixes in ALSA sequencer code to cover some races,
again unsurprisingly, spotted by syzkaller
- Optimize the common page allocation helper with alloc_pages_exact()
ASoC:
- Add SOF core support, as well as Intel SOF platform support
- Generic card driver improvements: support for MCLK/sample rate
ratio and pin switches
- A big set of improvements to TLV320AIC32x4 drivers
- New drivers for Freescale audio mixers, several Intel machines,
several Mediatek machines, Meson G12A, Spreadtrum compressed audio
and DMA devices
HD-audio:
- A few Realtek codec fixes for reducing pop noises
- Quirks for Chromebooks
- Workaround for faulty connection report on AMD/Nvidia HDMI
Others:
- A quirk for Focusrite Scarlett Solo USB-audio
- Add support for MOTU 8pre FireWire
- 24bit sample format support in aloop
- GUS patch format support (finally, over a decade) in native emux
synth code"
* tag 'sound-5.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (375 commits)
ASoC: SOF: Fix unused variable warnings
ALSA: line6: toneport: Fix broken usage of timer for delayed execution
ALSA: aica: Fix a long-time build breakage
ALSA: hda/realtek - Support low power consumption for ALC256
ASoC: stm32: i2s: update pcm hardware constraints
ASoC: codec: hdac_hdmi: no checking monitor in hw_params
ASoC: mediatek: mt6358: save PGA for mixer control
ASoC: mediatek: mt6358: save output volume for mixer controls
ASoC: mediatek: mt6358: initialize setting when ramping volume
ASoC: SOF: core: fix undefined nocodec reference
ASoC: SOF: xtensa: fix undefined references
ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly
ALSA: hdea/realtek - Headset fixup for System76 Gazelle (gaze14)
ALSA: hda/intel: add CometLake PCI IDs
ALSA: hda/realtek - Support low power consumption for ALC295
ASoC: rockchip: Fix an uninitialized variable compile warning
ASoC: SOF: Fix a compile warning with CONFIG_PCI=n
ASoC: da7219: Fix a compile warning at CONFIG_COMMON_CLK=n
ASoC: sound/soc/sof/: fix kconfig dependency warning
ASoC: stm32: spdifrx: change trace level on iec control
...
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Merge tag 'media/v5.1-2' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media updates from Mauro Carvalho Chehab:
- remove the deprecated Zoran driver from staging
- new I2C driver: ST MIPID02 CSI-2 camera bridge
- new platform driver: Amlogic Meson AO CEC G12A Controller
- add support for USB audio via the media controller
- au0828 driver is now supported via the media controller on both on
media and on usbaudio
- new kernel test for the media device allocator
- add support for stateless decoder at vicodec driver
- lots of other driver improvements fixes and cleanups
* tag 'media/v5.1-2' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (218 commits)
media: dt-bindings: aspeed-video: Add missing memory-region property
media: platform: Aspeed: Make reserved memory optional
media: platform: Aspeed: Remove use of reset line
media: stm32-dcmi: return appropriate error codes during probe
media: vsp1: Add support for missing 16-bit RGB555 formats
media: vsp1: Add support for missing 16-bit RGB444 formats
media: vsp1: Add support for missing 32-bit RGB formats
media: v4l: Add definitions for missing 16-bit RGB555 formats
media: v4l: Add definitions for missing 16-bit RGB4444 formats
media: v4l: Add definitions for missing 32-bit RGB formats
media: zoran: remove deprecated driver
media: MAINTAINERS: Update AO CEC with ao-cec-g12a driver
media: platform: meson: Add Amlogic Meson G12A AO CEC Controller driver
media: dt-bindings: media: meson-ao-cec: Add G12A AO-CEC-B Compatible
media: cros-ec-cec: decrement HDMI device refcount
media: seco-cec: decrement HDMI device refcount
media: tegra_cec: use new cec_notifier_parse_hdmi_phandle helper
media: stih_cec: use new cec_notifier_parse_hdmi_phandle helper
media: s5p_cec: use new cec_notifier_parse_hdmi_phandle helper
media: meson: ao-cec: use new cec_notifier_parse_hdmi_phandle helper
...
The line6 toneport driver has code for some delayed initialization,
and this hits the kernel Oops because mutex and other sleepable
functions are used in the timer callback. Fix the abuse by a delayed
work instead so that everything works gracefully.
Reported-by: syzbot+a07d0142e74fdd595cfb@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'leds-for-5.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/j.anaszewski/linux-leds
Pull LED updates from Jacek Anaszewski:
"LED core fixes and improvements:
- avoid races with workqueue
- Kconfig: pedantic cleanup
- small fixes for Flash class description
leds-lt3593:
- remove unneeded assignment in lt3593_led_probe
- drop pdata handling code
leds-blinkm:
- clean up double assignment to data->i2c_addr
leds-pca955x, leds-pca963x:
- revert ACPI support, as it turned out that there is no evidence
of officially registered ACPI IDs for these devices.
- make use of device property API
leds-as3645a:
- switch to fwnode property API
LED related addition to ACPI documentation:
- document how to refer to LEDs from remote nodes
LED related fix to ALSA line6/toneport driver:
- avoid polluting led_* namespace
And lm3532 driver relocation from MFD to LED subsystem, accompanied by
various improvements and optimizations; it entails also a change in
omap4-droid4-xt894.dts:
- leds: lm3532: Introduce the lm3532 LED driver
- mfd: ti-lmu: Remove LM3532 backlight driver references
- ARM: dts: omap4-droid4: Update backlight dt properties
- dt: lm3532: Add lm3532 dt doc and update ti_lmu doc"
* tag 'leds-for-5.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/j.anaszewski/linux-leds:
leds: avoid races with workqueue
ALSA: line6: Avoid polluting led_* namespace
leds: lm3532: Introduce the lm3532 LED driver
mfd: ti-lmu: Remove LM3532 backlight driver references
ARM: dts: omap4-droid4: Update backlight dt properties
dt: lm3532: Add lm3532 dt doc and update ti_lmu doc
leds: Small fixes for Flash class description
leds: blinkm: clean up double assignment to data->i2c_addr
leds: pca963x: Make use of device property API
leds: pca955x: Make use of device property API
leds: lt3593: Remove unneeded assignment in lt3593_led_probe
leds: lt3593: drop pdata handling code
leds: pca955x: Revert "Add ACPI support"
leds: pca963x: Revert "Add ACPI support"
drivers: leds: Kconfig: pedantic cleanups
ACPI: Document how to refer to LEDs from remote nodes
leds: as3645a: Switch to fwnode property API
In usX2Y_In04_init(), a new urb is firstly created through usb_alloc_urb()
and saved to 'usX2Y->In04urb'. Then, a buffer is allocated through
kmalloc() and saved to 'usX2Y->In04Buf'. If the allocation of the buffer
fails, the error code ENOMEM is returned after usb_free_urb(), which frees
the created urb. However, the urb is actually freed at card->private_free
callback, i.e., snd_usX2Y_card_private_free(). So the free operation here
leads to a double free bug.
To fix the above issue, simply remove usb_free_urb().
Signed-off-by: Wenwen Wang <wang6495@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The line6 driver uses a lot of USB buffers off of the stack, which is
not allowed on many systems, causing the driver to crash on some of
them. Fix this up by dynamically allocating the buffers with kmalloc()
which allows for proper DMA-able memory.
Reported-by: Christo Gouws <gouws.christo@gmail.com>
Reported-by: Alan Stern <stern@rowland.harvard.edu>
Tested-by: Christo Gouws <gouws.christo@gmail.com>
Cc: stable <stable@vger.kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In parse_audio_selector_unit(), the string array 'namelist' is allocated
through kmalloc_array(), and each string pointer in this array, i.e.,
'namelist[]', is allocated through kmalloc() in the following for loop.
Then, a control instance 'kctl' is created by invoking snd_ctl_new1(). If
an error occurs during the creation process, the string array 'namelist',
including all string pointers in the array 'namelist[]', should be freed,
before the error code ENOMEM is returned. However, the current code does
not free 'namelist[]', resulting in memory leaks.
To fix the above issue, free all string pointers 'namelist[]' in a loop.
Signed-off-by: Wenwen Wang <wang6495@umn.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error from snd_usb_mixer_apply_create_quirk() is ignored in the
current usb-audio driver code, which will continue the probing even
after the error. Let's take it more serious.
Fixes: 7b1eda223d ("ALSA: usb-mixer: factor out quirks")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Media Device Allocator API to allows multiple drivers share a media device.
This API solves a very common use-case for media devices where one physical
device (an USB stick) provides both audio and video. When such media device
exposes a standard USB Audio class, a proprietary Video class, two or more
independent drivers will share a single physical USB bridge. In such cases,
it is necessary to coordinate access to the shared resource.
Using this API, drivers can allocate a media device with the shared struct
device as the key. Once the media device is allocated by a driver, other
drivers can get a reference to it. The media device is released when all
the references are released.
Change the ALSA driver to use the Media Controller API to share media
resources with DVB, and V4L2 drivers on a AU0828 media device.
The Media Controller specific initialization is done after sound card is
registered. ALSA creates Media interface and entity function graph nodes
for Control, Mixer, PCM Playback, and PCM Capture devices.
snd_usb_hw_params() will call Media Controller enable source handler
interface to request the media resource. If resource request is granted,
it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is
returned.
Media specific cleanup is done in usb_audio_disconnect().
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Shuah Khan <shuah@kernel.org>
Signed-off-by: Hans Verkuil <hverkuil-cisco@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung@kernel.org>
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>
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>
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>
usb_alloc_urb() can fail due to kmalloc failure and push the error
upstream. Further this can cause a NULL pointer dereference in
init_pipe_urbs(). This patch avoids such a scenario.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add an entry to the quirks-table to for usb-audio to recognize the
Microbook II (although it only exposes vendor interfaces). A simple boot
quirk is also implemented to set up the sample rate and make sure that
no audio urbs are sent before the device is ready.
This patch only provides audio playback and capture at 96kHz sample
rate. Notice the following shortcomings:
- The sample rate is currently hardcoded to 96k although the device also
supports 48k and 44.1k.
- The various mixer controls of the MicroBook are not made available.
- The keep-iface control should be on by default because the device
shuts down whenever the altsetting is reset which is usually unwanted.
(I don't know the best way to do this)
- The communication format used by the MicroBook for sample rate setting
and also other setup has been reverse engineered by looking at the
usbmon output while running the windows driver through virtualbox. In
this patch the first byte of every message is set to \0 while in the
observed communications the first byte acts as a "message-counter"
increasing its value with every message sent. Leaving it at \0 does
not seem to affect the device.
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At least some USB devices use (MSB-aligned) audio format larger
than the actual resolution of the device. In order to expose the
actual device resolution (bBitResolution), add extra field to the
procfs stream info interface.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit a60945fd08 ("ALSA: usb-audio: move implicit fb quirks to
separate function") introduced an error in the handling of quirks for
implicit feedback endpoints. This commit fixes this.
If a quirk successfully sets up an implicit feedback endpoint, usb-audio
no longer tries to find the implicit fb endpoint itself.
Fixes: a60945fd08 ("ALSA: usb-audio: move implicit fb quirks to separate function")
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the T+A VID to the generic check in order to enable
native DSD support for T+A devices. This works with the new T+A USB
DAC model SD3100HV and will also work with future devices which
support the XMOS/Thesycon style DSD format.
Signed-off-by: Udo Eberhardt <udo.eberhardt@thesycon.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
XMOS/Thesycon family of USB Audio Class firmware flags DSD altsetting
separate from the PCM ones. Thus the DSD altsetting can be auto-detected
based on the flag and doesn't need maintaining specific altsetting
whitelist.
In addition, static VID:PID-to-altsetting whitelisting causes problems
when firmware update changes the altsetting, or same VID:PID is reused
for another device that has different kind of firmware.
This patch removes existing explicit whitelist mappings for XMOS VID
(0x20b1) and Thesycon VID (0x152a).
Also corrects placement of Hegel HD12 and NuPrime DAC-10 to keep list
sorted based on VID.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the Opus #3 DAP (made by 'The Bit')
in order to enable Native DSD support.
[ NOTE: this could be handled in the generic way with fp->dvd_raw if
we add 0x10cb to the vendor whitelist, but since 0x10cb shows a
different vendor name (Erantech), put to the individual entry at
this time -- tiwai ]
Signed-off-by: Olek Poplavsky <woodenbits@gmail.com>
Cc: <stable@vger.kernel.org>
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>
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>
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>
fix typo after a recent commit causing headphones to have no sound
Fixes: ad43d528a7 (ALSA: usb-audio: Define registers for CM6206)
Signed-off-by: Amadeusz Sławiński <amade@asmblr.net>
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>
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>
If a USB sound card reports 0 interfaces, an error condition is triggered
and the function usb_audio_probe errors out. In the error path, there was a
use-after-free vulnerability where the memory object of the card was first
freed, followed by a decrement of the number of active chips. Moving the
decrement above the atomic_dec fixes the UAF.
[ The original problem was introduced in 3.1 kernel, while it was
developed in a different form. The Fixes tag below indicates the
original commit but it doesn't mean that the patch is applicable
cleanly. -- tiwai ]
Fixes: 362e4e49ab ("ALSA: usb-audio - clear chip->probing on error exit")
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the SMSL D1 in order to enable
Native DSD support.
[ Moved the added entry in numerical order -- tiwai ]
Signed-off-by: Tony Das <tdas444@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the Dell WD15 Dock, the WD19 Dock (0bda:402e) doens't provide
useful string for the vendor and product names too. In order to share
the UCM with WD15, here we keep the profile_name same as the WD15.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The register map for CM6206 is known and we can define what
the values written to the different registers actually mean.
I tested to print the hex values before/after this change,
there is no functional change.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotation in caiaq driver.
Note that this seems necessary to be put exactly before the next
label, so it's outside the ifdef block.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds several vendor specific mixer quirks for RME's Class Compliant
USB devices. These provide extra status information from the device
otherwise not available.
These include AES/SPDIF rate and status information, current system
sampling rate and measured frequency. This information is especially
useful in cases where device's clock is slaved to external clock
source.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A quirk in snd-usb-audio was added to automate setting sample rate to
4800k and remove the previously exposed nonfunctional microphone for
the Bowers & Wilkins PX:
commit 240a8af929https://lore.kernel.org/patchwork/patch/919689/
However the headphones where updated shortly after that to remove the
unintentional microphone functionality. I guess because of this the
headphones now crash when connecting them via USB while the quirk is
active. Dmesg:
snd-usb-audio: probe of 2-3:1.0 failed with error -22
usb 2-3: 2:1: cannot get min/max values for control 2 (id 2)
This patch removes the microfone and allows the headphones to connect
and work out of the box. It is based on the current mainline kernel
and successfully applied an tested on my machine (4.18.10.arch1-1).
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Nicolas Huaman <nicolas@herochao.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For discarding the pending bytes on rawmidi, we process with a loop of
snd_rawmidi_transmit() which is just a waste of CPU power.
Implement a lightweight API function to discard the pending bytes and
the proceed the ring buffer instantly, and use it instead of open
codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Addresses-Coverity-ID: 1357413 ("Missing break in switch")
Addresses-Coverity-ID: 114917 ("Missing break in switch")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sizeof() when applied to a pointer typed expression gives the
size of the pointer, not that of the pointed data.
Fixes: 7edf3b5e6a ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Addresses-Coverity-ID: 115084 ("Missing break in switch")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pointer 'urb' is being assigned but is never used hence it is
redundant and can be removed.
Cleans up clang warning:
warning: variable 'urb' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of UAC3 Power Domains associated to an Audio Streaming
path within the PCM's logic. This means, when there is no audio
being transferred (pcm is closed), the host will set the Power Domain
associated to that substream to state D1. When audio is being transferred
(from hw_params onwards), the Power Domain will be set to D0 state.
This is the way the host lets the device know which Terminal
is going to be actively used and it is for the device to
manage its own internal resources on that UAC3 Power Domain.
Note the resume method now sets the Power Domain to D1 state as
resuming the device doesn't mean audio streaming will occur.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai
Linus Torvalds
Wenwen Wang
Greg Kroah-Hartman
Shuah Khan
Jacek Anaszewski
Roope Salmi
Aditya Pakki
Manuel Reinhardt
Jussi Laako
Udo Eberhardt
Olek Poplavsky
YueHaibing
Amadeusz Sławiński
Hui Peng
Hui Peng
Tony Das
Hui Wang
Linus Walleij
Nicolas Huaman
Gustavo A. R. Silva
Wei Yongjun
Colin Ian King
Jorge Sanjuan