The convert_chmap_v3() has a loop with its increment size of
cs_desc->wLength, but we forgot to validate cs_desc->wLength itself,
which may lead to potential endless loop by a malformed descriptor.
Add a proper size check to abort the loop for plugging the hole.
Fixes: ecfd41166b ("ALSA: usb-audio: Validate UAC3 cluster segment descriptors")
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20260427152224.15276-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At parsing UAC3 streams, we allocate a PD object at each time, and
either assign or free it. But there is a case where the PD object may
be leaked; namely, in __snd_usb_parse_audio_interface() loop, when an
audioformat shares the same endpoint with others, it's put to a link
and returns from snd_usb_add_audio_stream(), but the PD is forgotten
afterwards. Overall, the treatment of PD object in the parser code is
a bit flaky, and we should be more careful about the object ownership.
This patch tries to fix the above case and improve the code a bit.
The pd object is now managed with the auto-cleanup in the loop, and
the ownership is updated when the pd object gets assigned to the
stream, which guarantees the release of the leftover object.
Fixes: 7edf3b5e6a ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Link: https://patch.msgid.link/20260427151508.12544-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC3 cluster descriptor length check in
snd_usb_get_audioformat_uac3()was added to
make sure that the buffer is large enough for
a struct uac3_cluster_header_descriptor before the
returned data is cast and used.
However, the check uses sizeof(cluster), where cluster
is a pointer, not the size of the descriptor header.
This makes the validation depend on the architecture
pointer size and does not match the intended object size.
Check against sizeof(*cluster) instead.
Fixes: fb4e2a6e8f ("ALSA: usb-audio: Fix out-of-bounds read in snd_usb_get_audioformat_uac3()")
Cc: stable@vger.kernel.org
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260424-alsa-usb-uac3-cluster-size-v1-1-99a5808898a3@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
convert_chmap_v3() translates UAC3 channel relationship values into
ALSA channel-map positions.
UAC3_CH_FRONT_WIDE_LEFT and UAC3_CH_FRONT_WIDE_RIGHT currently fall
back to SNDRV_CHMAP_UNKNOWN, although ALSA already provides matching
channel-map positions via SNDRV_CHMAP_FLW and SNDRV_CHMAP_FRW.
Map these two UAC3 positions to their ALSA equivalents and update the
comment to clarify that unsupported UAC3 channel relationships remain
reported as SNDRV_CHMAP_UNKNOWN.
No functional change for other channel relationships.
Signed-off-by: Cássio Gabriel <cassiogabrielcontato@gmail.com>
Link: https://patch.msgid.link/20260311200010.103710-1-cassiogabrielcontato@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk flag to skip the usb_set_interface(),
snd_usb_init_pitch(), and snd_usb_init_sample_rate() calls in
__snd_usb_parse_audio_interface(). These are redundant with
snd_usb_endpoint_prepare() at stream-open time.
Enable the quirk for Focusrite devices, as init_sample_rate(rate_max)
sets 192kHz during probing, which disables the internal mixer and Air
and Safe modes.
Fixes: 16f1f83844 ("Revert "ALSA: usb-audio: Drop superfluous interface setup at parsing"")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/65a7909b15f9feb76c2a6f4f8814c240ddc50737.1771594828.git.g@b4.vu
This was done entirely with mindless brute force, using
git grep -l '\<k[vmz]*alloc_objs*(.*, GFP_KERNEL)' |
xargs sed -i 's/\(alloc_objs*(.*\), GFP_KERNEL)/\1)/'
to convert the new alloc_obj() users that had a simple GFP_KERNEL
argument to just drop that argument.
Note that due to the extreme simplicity of the scripting, any slightly
more complex cases spread over multiple lines would not be triggered:
they definitely exist, but this covers the vast bulk of the cases, and
the resulting diff is also then easier to check automatically.
For the same reason the 'flex' versions will be done as a separate
conversion.
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This is the result of running the Coccinelle script from
scripts/coccinelle/api/kmalloc_objs.cocci. The script is designed to
avoid scalar types (which need careful case-by-case checking), and
instead replace kmalloc-family calls that allocate struct or union
object instances:
Single allocations: kmalloc(sizeof(TYPE), ...)
are replaced with: kmalloc_obj(TYPE, ...)
Array allocations: kmalloc_array(COUNT, sizeof(TYPE), ...)
are replaced with: kmalloc_objs(TYPE, COUNT, ...)
Flex array allocations: kmalloc(struct_size(PTR, FAM, COUNT), ...)
are replaced with: kmalloc_flex(*PTR, FAM, COUNT, ...)
(where TYPE may also be *VAR)
The resulting allocations no longer return "void *", instead returning
"TYPE *".
Signed-off-by: Kees Cook <kees@kernel.org>
The maxpacksize field in struct audioformat represents the maximum number
of bytes per isochronous interval. The current implementation only
special-cases high-speed endpoints and does not account for the different
computations required for SuperSpeed, SuperSpeedPlus, or eUSB2. As a
result, USB audio class devices operating at these speeds may fail to
stream correctly. The issue was observed on a MOTU 16A (2025) interface,
which requires more than 1024 bytes per interval at SuperSpeed.
This patch replaces the existing logic with a helper that computes the
correct maximum bytes-per-interval for all USB speeds, borrowing the logic
used in drivers/usb/core/urb.c.
Signed-off-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://patch.msgid.link/20251124210518.90054-1-dylan_robinson@motu.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "p" pointer is void so sizeof(*p) is 1. The intent was to check
sizeof(*cs_desc), which is 3, instead.
Fixes: ecfd41166b ("ALSA: usb-audio: Validate UAC3 cluster segment descriptors")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://patch.msgid.link/aKL5kftC1qGt6lpv@stanley.mountain
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC3 class segment descriptors need to be verified whether their sizes
match with the declared lengths and whether they fit with the
allocated buffer sizes, too. Otherwise malicious firmware may lead to
the unexpected OOB accesses.
Fixes: 11785ef532 ("ALSA: usb-audio: Initial Power Domain support")
Reported-and-tested-by: Youngjun Lee <yjjuny.lee@samsung.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250814081245.8902-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace strcpy() and sprintf() usages in the USB audio drivers with
the safer versions (strscpy() and scnprintf()) with the proper max
size evaluation. Only for safety, no actual behavior change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250710100727.22653-103-tiwai@suse.de
In snd_usb_get_audioformat_uac3(), the length value returned from
snd_usb_ctl_msg() is used directly for memory allocation without
validation. This length is controlled by the USB device.
The allocated buffer is cast to a uac3_cluster_header_descriptor
and its fields are accessed without verifying that the buffer
is large enough. If the device returns a smaller than expected
length, this leads to an out-of-bounds read.
Add a length check to ensure the buffer is large enough for
uac3_cluster_header_descriptor.
Signed-off-by: Youngjun Lee <yjjuny.lee@samsung.com>
Fixes: 9a2fe9b801 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Link: https://patch.msgid.link/20250623-uac3-oob-fix-v1-1-527303eaf40a@samsung.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit adding support for multiple control interfaces expanded struct
snd_usb_power_domain with pointer to control interface for proper control
message routing but missed one initialization point of this structure,
which has left new field with NULL value.
Standard mandates that each device has at least one control interface and
code responsible for power domain does not check for NULL values when
querying for control interface. This caused some USB devices to crash
the kernel.
Fixes: 6aa8700150 ("ALSA: usb-audio: Support multiple control interfaces")
Signed-off-by: Karol Kosik <k.kosik@outlook.com>
Link: https://patch.msgid.link/AS8P190MB1285B563C6B5394DB274813FEC782@AS8P190MB1285.EURP190.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Registering Numark Party Mix II fails with error 'bogus bTerminalLink 1'.
The problem stems from the driver not being able to find input/output
terminals required to configure audio streaming. The information about
those terminals is stored in AudioControl Interface. Numark device
contains 2 AudioControl Interfaces and the driver checks only one of them.
According to the USB standard, a device can have multiple audio functions,
each represented by Audio Interface Collection. Every audio function is
considered to be closed box and will contain unique AudioControl Interface
and zero or more AudioStreaming and MIDIStreaming Interfaces.
The Numark device adheres to the standard and defines two audio functions:
- MIDIStreaming function
- AudioStreaming function
It starts with MIDI function, followed by the audio function. The driver
saves the first AudioControl Interface in `snd_usb_audio` structure
associated with the entire device. It then attempts to use this interface
to query for terminals and clocks. However, this fails because the correct
information is stored in the second AudioControl Interface, defined in the
second Audio Interface Collection.
This patch introduces a structure holding association between each
MIDI/Audio Interface and its corresponding AudioControl Interface,
instead of relying on AudioControl Interface defined for the entire
device. This structure is populated during usb probing phase and leveraged
later when querying for terminals and when sending USB requests.
Alternative solutions considered include:
- defining a quirk for Numark where the order of interface is manually
changed, or terminals are hardcoded in the driver. This solution would
have fixed only this model, though it seems that device is USB compliant,
and it also seems that other devices from this company may be affected.
What's more, it looks like products from other manufacturers have similar
problems, i.e. Rane One DJ console
- keeping a list of all AudioControl Interfaces and querying all of them
to find required information. That would have solved my problem and have
low probability of breaking other devices, as we would always start with
the same logic of querying first AudioControl Interface. This solution
would not have followed the standard though.
This patch preserves the `snd_usb_audio.ctrl_intf` variable, which holds
the first AudioControl Interface, and uses it as a fallback when some
interfaces are not parsed correctly and lack an associated AudioControl
Interface, i.e., when configured via quirks.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217865
Signed-off-by: Karol Kosik <k.kosik@outlook.com>
Link: https://patch.msgid.link/AS8P190MB1285893F4735C8B32AD3886BEC852@AS8P190MB1285.EURP190.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver puts SNDRV_CHMAP_SL and _SR as left and right
surround channels for UAC1 channel map, respectively. But they should
have been SNDRV_CHMAP_RL and _RR; the current value *_SL and _SR are
rather "side" channels, not "surround". I guess I took those
mistakenly when I read the spec mentioning "surround left".
This patch corrects those entries to be the right channels.
Suggested-by: Sylvain BERTRAND <sylvain.bertrand@legeek.net>
Closes: https://lore.kernel.orgZ/qIyJD8lhd8hFhlC@freedom
Fixes: 04324ccc75 ("ALSA: usb-audio: add channel map support")
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20240731142018.24750-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a usb audio device sets more bits than the amount of channels
it could write outside of the map array.
Signed-off-by: Johan Carlsson <johan.carlsson@teenage.engineering>
Fixes: 04324ccc75 ("ALSA: usb-audio: add channel map support")
Message-ID: <20240313081509.9801-1-johan.carlsson@teenage.engineering>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There have been reports of USB-audio driver spewing errors at the
probe time on a few devices like Jabra and Logitech. The suggested
fix there couldn't be applied as is, unfortunately, because it'll
likely break other devices.
But, the patch suggested an interesting point: looking at the current
init code in stream.c, one may notice that it does initialize
differently from the device setup in endpoint.c. Namely, for UAC1, we
should call snd_usb_init_pitch() and snd_usb_init_sample_rate() after
setting the interface, while the init sequence at parsing calls them
before setting the interface blindly.
This patch changes the init sequence at parsing for UAC1 (and other
devices that need a similar behavior) to be aligned with the rest of
the code, setting the interface at first. And, this fixes the
long-standing problems on a few UAC1 devices like Jabra / Logitech,
as reported, too.
Reported-and-tested-by: Joakim Tjernlund <joakim.tjernlund@infinera.com>
Closes: https://lore.kernel.org/r/202bbbc0f51522e8545783c4c5577d12a8e2d56d.camel@infinera.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230821111857.28926-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There may be a bad USB audio device with a USB ID of (0x04fa, 0x4201) and
the number of it's interfaces less than 4, an out-of-bounds read bug occurs
when parsing the interface descriptor for this device.
Fix this by checking the number of interfaces.
Signed-off-by: Dongxiang Ke <kdx.glider@gmail.com>
Link: https://lore.kernel.org/r/20220906024928.10951-1-kdx.glider@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We reset each interface that is being parsed for each stream, but this
is superfluous and even can lead to spurious errors. Since the
interface is set up properly at opening the endpoint for each actual
stream operation, let's drop the superfluous one.
Link: https://lore.kernel.org/r/20220831130021.4762-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The info message that was added in the commit a4aad5636c ("ALSA:
usb-audio: Inform devices that need delayed registration") is actually
useful to know the need for the delayed registration. However, it
turned out that this doesn't catch the all cases; namely, this warned
only when a PCM stream is attached onto the existing PCM instance, but
it doesn't count for a newly created PCM instance. This made
confusion as if there were no further delayed registration.
This patch moves the check to the code path for either adding a stream
or creating a PCM instance. Also, make it simpler by checking the
card->registered flag instead of querying each snd_device state.
Fixes: a4aad5636c ("ALSA: usb-audio: Inform devices that need delayed registration")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082
Link: https://lore.kernel.org/r/20220831125901.4660-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is another quirk for the transfer, and that's currently specific
to Zoom R16/24, handled in create_standard_audio_quirk(). Let's move
this also to the new quirk_flags.
Link: https://lore.kernel.org/r/20210729073855.19043-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The txfr_quirk field was meant for aligning the transfer, and it's set
for certain devices in quirks-table.h. Now we can move that stuff
also to the new quirk_flags gracefully, and reduce the quirks-table.h
entries (that are exposed to module device table).
As the quirks-table.h entries are also with the name string override,
provide the corresponding entries to the usb_audio_names[] table,
too.
Link: https://lore.kernel.org/r/20210729073855.19043-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current channel-map control implementation in USB-audio driver may
lead to an error message like
"control 3:0:0:Playback Channel Map:0: access overflow"
when CONFIG_SND_CTL_VALIDATION is set. It's because the chmap get
callback clears the whole array no matter which count is set, and
rather the false-positive detection.
This patch fixes the problem by clearing only the needed array range
at usb_chmap_ctl_get().
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201211130048.6358-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A preliminary change for the later big changes. This is a minor code
refactoring to drop the unnecessary arguments that can be retrieved in
a different way.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-21-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A preliminary change for the later big changes. This is a minor code
refactoring to drop the unnecessary arguments that can be retrieved in
a different way.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-20-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting the active altsetting at changing sample rate seems
unrecommended. The host should deselect the altsetting at first
before that, then select it again.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-18-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently snd_usb_endpoint objects are created at first when the
substream is opened and tries to assign the endpoints corresponding to
the matching audioformat. But since basically the all endpoints have
been already parsed and the information have been obtained, we may
create the endpoint objects statically at the init phase. It's easier
to manage for the implicit fb case, for example.
This patch changes the endpoint object management and lets the parser
to create the all endpoint objects.
This change shouldn't bring any functional changes.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-15-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of parsing and evaluating the sync endpoint and the implicit
feedback mode at each time the audio stream is opened, let's parse it
once at the probe time, as the all needed information can be obtained
statically from the descriptor or from the quirk.
This patch extends audioformat struct to record the sync endpoint,
interface and altsetting as well as the implicit feedback flag, which
are filled at parsing the streams. Then, set_sync_endpoint() is much
simplified just to follow the already parsed data.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, when the device provides the discrete sample rate
tables with unusual sample rates, the driver tries to gather the whole
values from the audioformat entries and create a hw-constraint rule to
restrict with this single rate list. This is rather inefficient and
may overlook the rates that are associated only with the certain
audioformat entries.
This patch improves the hw constraint setup by rewriting the existing
hw_rule_rate(). The discrete sample rates (identified by rate_table
and nr_rates of format entry) are checked in the existing
hw_rule_rate() instead of extra rules; in the case of discrete rates,
the function compares with each rate table entry and calculates the
min/max values from there. For the contiguous rates, the behavior
doesn't change.
Along with it, snd_usb_pcm_check_knot() and snb_usb_substream
rate_list field become superfluous, thus those are dropped.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.
So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB-audio driver may call snd_card_register() multiple times as
its probe function is per USB interface while some USB-audio devices
may provide multiple interfaces to assign different streams although
they belong to the same device. This works in most cases but the
registration is racy, hence it may miss the device recognition,
e.g. PA doesn't see certain devices when hotplugged.
The recent addition of the delayed registration quirk allows to sync
the registration at the last known interface, and the previous commit
added a new module option to allow the dynamic setup for that
purpose.
Now, this patch tries to find out and notifies for such devices that
require the delayed registration. It shows a message like:
Found post-registration device assignment: 1234abcd:02
If you hit this message, you can pass delayed_register module option
like:
snd_usb_audio.delayed_register=1234abcd:02
by just copying the last shown entry. If this works, it can be added
statically in the quirk list, registration_quirks[] found at the end
of sound/usb/quirks.c.
Link: https://lore.kernel.org/r/20200325103322.2508-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply const prefix to the remaining places: the static table for the
unit information, the mixer maps, the validator tables, etc.
Just for minor optimization and no functional changes.
Link: https://lore.kernel.org/r/20200105144823.29547-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull USB validation patches. It's based on the latest 5.3 development
branch, so we shall catch up the whole things.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new helper to validate each audio descriptor unit before
and check the unit before actually accessing it. This should harden
against the OOB access cases with malformed descriptors that have been
recently frequently reported by fuzzers.
The existing descriptor checks are still kept although they become
superfluous after this patch. They'll be cleaned up eventually
later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some cards have alternate setting with non-PCM format as the first
altsetting in the interface descriptors. This confuses userspace, since
alsa-lib uses device 0 by default. So lets parse interfaces in two steps:
1. Parse altsettings with PCM formats.
2. Parse altsettings with non-PCM formats.
This fixes at least following cards:
- Audinst HUD-mx2
- Audinst HUD-mini
[ Adapted to 5.3 kernel by tiwai ]
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many open code for releasing audioformat object.
Provide a unified helper and call it from the all places.
Only a cleanup, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is
allocated through kzalloc() before the execution goto 'found_clock'.
However, this structure is not deallocated if the memory allocation for
'pd' fails, leading to a memory leak bug.
To fix the above issue, free 'fp->chmap' before returning NULL.
Fixes: 7edf3b5e6a ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Based on 1 normalized pattern(s):
this program is free software you can redistribute it and or modify
it under the terms of the gnu general public license as published by
the free software foundation either version 2 of the license or at
your option any later version this program is distributed in the
hope that it will be useful but without any warranty without even
the implied warranty of merchantability or fitness for a particular
purpose see the gnu general public license for more details you
should have received a copy of the gnu general public license along
with this program if not write to the free software foundation inc
59 temple place suite 330 boston ma 02111 1307 usa
extracted by the scancode license scanner the SPDX license identifier
GPL-2.0-or-later
has been chosen to replace the boilerplate/reference in 1334 file(s).
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Allison Randal <allison@lohutok.net>
Reviewed-by: Richard Fontana <rfontana@redhat.com>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190527070033.113240726@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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>
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>
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>
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>
Power Domains in the UAC3 spec are mainly intended to be
associated to an Input or Output Terminal so the host
changes the power state of the entire capture or playback
path within the topology.
This patch adds support for finding Power Domains associated
to an Audio Streaming Interface (bTerminalLink) and adds a
reference to them in the usb audio substreams (snd_usb_substream).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
