[ Upstream commit a39d51ff1f52cd0b6fe7d379ac93bd8b4237d1b7 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 5fadc941d0 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 7e1afce586 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit e53f47f6c1 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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>
Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as
this serves merely as an intermediate buffer that is copied to each
URB transfer buffer. This works well in general on x86, but on some
archs this may result in cache coherency issues when mmap is used.
OTOH, it works also on such arch unless mmap is used.
This patch is a step for mitigating the inconvenience; a new module
option "use_vmalloc" is provided so that user can choose to allocate
the DMA coherent buffer instead of the existing vmalloc buffer.
The drawback is that it'd be the standard dma_alloc_coherent() calls
and the system would require contiguous pages on non-x86 archs.
Note that it's a global option and not dynamically switchable since
the buffer is pre-allocated at the probe time. In theory, it's
possible to be switchable, but it'd be trickier and racier.
As default use_vmalloc option is set to true, so that the old behavior
is kept. For allowing the coherent mmap on ARM or MIPS, pass
use_vmalloc=0 option explicitly.
Reported-and-tested-by: Daniel Danzberger <daniel@dd-wrt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC3 channel map is created during interface parsing,
and in some cases was not freed in failure paths.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
bmAtributes offset doesn't exist in the UAC3 CS_EP descriptor.
Hence, checking for pitch control as if it was UAC2 doesn't make
any sense. Use the defined UAC3 offsets instead.
Fixes: 9a2fe9b801 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently released USB Audio Class 3.0 specification
contains BADD (Basic Audio Device Definition) document
which describes pre-defined UAC3 configurations.
BADD support is mandatory for UAC3 devices, it should be
implemented as a separate USB device configuration.
As per BADD document, class-specific descriptors
shall not be included in the Device’s Configuration
descriptor ("inferred"), but host can guess them
from BADD profile number, number of endpoints and
their max packed sizes.
This patch adds support of all BADD profiles from the spec
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload snd_usb_parse_audio_interface() function
which became quite long after adding UAC3 spec support.
Move class-specific parts of uac3 parsing to separate
function which now produce audioformat structure that
is ready to be fed to snd_usb_add_audio_stream().
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload snd_usb_parse_audio_interface() function
which became quite long after adding UAC3 spec support.
Move class-specific parts of uac1/2 parsing to separate
function which now produce audioformat structure that
is ready to be fed to snd_usb_add_audio_stream().
This also broke Blue Microphones workaround (which
relies on audioformat decoded from previous altsetting)
into two parts: prepare quirk flag analyzing previous
altsetting then use it with current altsetting.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload snd_usb_parse_audio_interface() function which
became quite long after adding UAC3 spec support.
Move audioformat allocation and initialization
into separate function, this will make easier
future refactoring.
Attributes left in the original func because it'll
be used for UAC3 BADD profiles suport in the future
There is no functional change.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The channel mapping is defined by bChRelationship, not bChPurpose.
Fixes: 9a2fe9b801 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Michael Drake <michael.drake@codethink.co.uk>
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently released USB Audio Class 3.0 specification
introduces many significant changes comparing to
previous versions, like
- new Power Domains, support for LPM/L1
- new Cluster descriptor
- changed layout of all class-specific descriptors
- new High Capability descriptors
- New class-specific String descriptors
- new and removed units
- additional sources for interrupts
- removed Type II Audio Data Formats
- ... and many other things (check spec)
It also provides backward compatibility through
multiple configurations, as well as requires
mandatory support for BADD (Basic Audio Device
Definition) on each ADC3.0 compliant device
This patch adds initial support of UAC3 specification
that is enough for Generic I/O Profile (BAOF, BAIF)
device support from BADD document.
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Offload USB audio interface parsing function by
moving quirks to a specially designed location (quirks.c)
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media fixes from Mauro Carvalho Chehab:
"Some bug fixes on au0828 and snd-usb-audio:
- the au0828+snd-usb-audio MC patch broke several things and produced
some race conditions. Better to revert the patches, and re-work on
them for a next version
- fix a regression at tuner disable links logic
- properly handle dev_state as a bitmask"
* tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
[media] Revert "[media] media: au0828 change to use Managed Media Controller API"
[media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
[media] au0828: Fix dev_state handling
[media] au0828: fix au0828_v4l2_close() dev_state race condition
[media] media: au0828 fix to clear enable/disable/change source handlers
[media] v4l2-mc: cleanup a warning
[media] au0828: disable tuner links and cache tuner/decoder
A collection of small fixes:
- A fix in ALSA timer core to avoid possible BUG() trigger
- A fix in ALSA timer core 32bit compat layer
- A few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- Fixes of USB-audio double-free at some error paths
- A fix for memory leak in DICE driver at hotunplug
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Merge tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes:
- a fix in ALSA timer core to avoid possible BUG() trigger
- a fix in ALSA timer core 32bit compat layer
- a few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- fixes of USB-audio double-free at some error paths
- a fix for memory leak in DICE driver at hotunplug"
* tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: timer: Use mod_timer() for rearming the system timer
ALSA: hda - fix front mic problem for a HP desktop
ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call
ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
ALSA: dice: fix memory leak when unplugging
ALSA: hda - Apply fix for white noise on Asus N550JV, too
ALSA: hda - Fix white noise on Asus N750JV headphone
ALSA: hda - Asus N750JV external subwoofer fixup
ALSA: timer: fix gparams ioctl compatibility for different architectures
Unfortunately, this patch caused several regressions at au0828 and
snd-usb-audio, like this one:
https://bugzilla.kernel.org/show_bug.cgi?id=115561
It also showed several troubles at the MC core that handles pretty
poorly the memory protections and data lifetime management.
So, better to revert it and fix the core before reapplying this
change.
This reverts commit aebb2b89bf ("[media] sound/usb: Use Media
Controller API to share media resources")'
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.
This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.
Based on a patch by Takashi Iwai <tiwai@suse.de>
[Note for stable backports:
this patch requires the commit 902eb7fd1e ('ALSA: usb-audio: Minor
code cleanup in create_fixed_stream_quirk()')]
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358
Reported-by: Ralf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: Vladis Dronov <vdronov@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change ALSA driver to use Media Controller API to share media resources
with DVB and V4L2 drivers on a AU0828 media device. 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().
Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().
Signed-off-by: Geliang Tang <geliangtang@163.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)
The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).
In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.
For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.
The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.
In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.
Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.
The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.
Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Tested-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.
This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.
Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert with dev_err() and co from snd_printk(), etc.
As there are too deep indirections (e.g. ep->chip->dev->dev),
a few new local macros, usb_audio_err() & co, are introduced.
Also, the device numbers in some messages are dropped, as they are
shown in the prefix automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Johan Carlsson
Takashi Iwai
Dongxiang Ke
Hector Martin
Gustavo A. R. Silva
Alexander Tsoy
Wenwen Wang
Thomas Gleixner
Shuah Khan
Wei Yongjun
Jorge Sanjuan
Ruslan Bilovol
Michael Drake
Markus Elfring
Linus Torvalds
Mauro Carvalho Chehab
Vladis Dronov
Shuah Khan
Geliang Tang
Ricard Wanderlof
Johan Rastén