commit f15cfca818 upstream.
The Zoom UAC-2 USB audio interface provides an async playback endpoint
("1 OUT (ASYNC)") and capture endpoint ("2 IN (ASYNC)"), both with
2-channel S32_LE in 44.1, 48, 88.2, 96, 176.4, or 192
kilosamples/s. The device provides explicit feedback to adjust the
host's playback rate, but the feedback appears unstable and biased
relative to the device's capture rate.
"alsaloop -t 1000" experiences playback underruns and tries to
resample the captured audio to match the varying playback
rate. Forcing the kernel to use implicit feedback appears to
produce more stable results. This causes the host to transmit one
playback sample for each capture sample received. (Zoom North America
has been notified of this change.)
Signed-off-by: Keith Winstein <keithw@cs.stanford.edu>
Tested-by: Keith Winstein <keithw@cs.stanford.edu>
Cc: <stable@vger.kernel.org>
BugLink: https://lore.kernel.org/r/20201027071841.GA164525@trolley.csail.mit.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1b7ecc241a upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3da87ec67a upstream.
As expected, this requires the same quirk as the SSL2+ in order for the
clock to sync. This was suggested by, and tested on an SSL2, by Dmitry.
Suggested-by: Dmitry <dpavlushko@gmail.com>
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200621075005.52mjjfc6dtdjnr3h@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 2edb84e304 ]
MicroBook IIc operates in UAC2 mode by default. This patch addresses
several issues with it:
- MicroBook II and IIc shares the same USB ID. We can distinguish them
by interface class.
- MaxPacketsOnly attribute is erroneously set in endpoint descriptors.
As a result this card produces noise with all sample rates other than
96 KHz. This also causes issues like IOMMU page faults and other
problems with host controller.
- Sample rate changes takes more than 2 seconds for this device. Clock
validity request returns false during that period, so the clock validity
quirk is required.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200229151815.14199-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
This reverts commit aba41867dd which is
commit f0bd62b640 upstream.
It causes a number of reported issues and a fix for it has not hit
Linus's tree yet. Revert this to resolve those problems.
Cc: Alexander Tsoy <alexander@tsoy.me>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Sasha Levin <sashal@kernel.org>
Cc: Hans de Goede <jwrdegoede@fedoraproject.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e7585db1b0 upstream.
This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the
audio interface's clock. Tested on an SSL2+.
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200612111807.dgnig6rwhmsl2bod@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit ff58bbc7b9 ]
With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running. In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer. Since
the PCM stream gets closed, this may lead to use-after-free.
This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.
Fixes: 10ce77e481 ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 10ce77e481 ]
For USB sound devices using implicit feedback the endpoint used for
this feedback should be able to be opened twice, once for required
feedback and second time for audio data. This way these devices can be
put in duplex audio mode. Since this only works if the settings of the
endpoint don't change a check is included for this.
This fixes bug 207023 ("MOTU M2 regression on duplex audio") and
should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device
will not operate full-duplex")
Fixes: c249177944 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series")
Signed-off-by: Erwin Burema <e.burema@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751
Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f0bd62b640 ]
For computation of the the next frame size current value of fs/fps and
accumulated fractional parts of fs/fps are used, where values are stored
in Q16.16 format. This is quite natural for computing frame size for
asynchronous endpoints driven by explicit feedback, since in this case
fs/fps is a value provided by the feedback endpoint and it's already in
the Q format. If an error is accumulated over time, the device can
adjust fs/fps value to prevent buffer overruns/underruns.
But for synchronous endpoints the accuracy provided by these computations
is not enough. Due to accumulated error the driver periodically produces
frames with incorrect size (+/- 1 audio sample).
This patch fixes this issue by implementing a different algorithm for
frame size computation. It is based on accumulating of the remainders
from division fs/fps and it doesn't accumulate errors over time. This
new method is enabled for synchronous and adaptive playback endpoints.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c249177944 ]
This fixes crackling sound during playback.
Further note: MOTU is known for reusing Product IDs for different
devices or different generations of the device (e.g. MicroBook
I/II/IIc shares a single Product ID). This patch was only tested with
M4 audio interface, but the same Product ID is also used by M2. Hope
it will work for M2 as well.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200115151358.56672-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 5d1b71226d upstream.
The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was
checking for there to be at least one altsetting but then went on to
access the second one, which may not exist.
This could lead to random slab data being used to initialise the sync
endpoint in snd_usb_add_endpoint().
Fixes: c75a8a7ae5 ("ALSA: snd-usb: add support for implicit feedback")
Fixes: ca10a7ebdf ("ALSA: usb-audio: FT C400 sync playback EP to capture EP")
Fixes: 5e35dc0338 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204")
Fixes: 17f08b0d9a ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II")
Fixes: 103e962564 ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk")
Cc: stable <stable@vger.kernel.org> # 3.5
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20200114083953.1106-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 92adc96f8e upstream.
Recently we found the headset-mic on the Dell Dock WD19 doesn't work
anymore after s3 (s2i or deep), this problem could be workarounded by
closing (pcm_close) the app and then reopening (pcm_open) the app, so
this bug is not easy to be detected by users.
When problem happens, retire_capture_urb() could still be called
periodically, but the size of captured data is always 0, it could be
a firmware bug on the dock. Anyway I found after resuming, the
snd_usb_pcm_prepare() will be called, and if we forcibly run
set_format() to set the interface and its endpoint, the capture
size will be normal again. This problem and workaound also apply to
playback.
To fix it in the kernel, add a quirk to let set_format() run
forcibly once after resume.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191218132650.6303-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0141254b0a upstream.
Make sure to check the return value of usb_altnum_to_altsetting() to
avoid dereferencing a NULL pointer when the requested alternate settings
is missing.
The format altsetting number may come from a quirk table and there does
not seem to be any other validation of it (the corresponding index is
checked however).
Fixes: b099b9693d ("ALSA: usb-audio: Avoid superfluous usb_set_interface() calls")
Cc: stable <stable@vger.kernel.org> # 4.18
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20191220093134.1248-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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>
Behringer UFX1604 requires the similar quirk to apply implicit fb like
another Behringer model UFX1204 in order to fix the noisy playback.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Linux kernel assumes that get_endpoint(alts,0) and
get_endpoint(alts,1) are eachothers feedback endpoints.
To reassure that validity it will test bsynchaddress to comply with that
assumption. But if the bsyncaddress is 0 (invalid), it will flag that as
a wrong assumption and return an error.
Fix: Skip the test if bSynchAddress is 0.
Note: those with a valid bSynchAddress should have a code quirck added.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
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>
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>
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>
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>
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>
Set the UAC3 Power Domain state for an Audio Streaming interface
to D2 state before suspending the device (usb_driver callback).
This lets the device know there is no intention to use any of the
Units in the Audio Function and that the host is not going to
even listen for wake-up events (interrupts) on the units.
When the usb_driver gets resumed, the state D0 (fully powered) will
be set. This ties up the UAC3 Power Domains to the runtime PM.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with
somewhat special for vmalloc handling, but in the end, this turned to
just the default handler, i.e. NULL. As the situation has never
changed over decades, let's rip it off.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
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>
Avoid if ((err = ...) style and expand to multiple lines instead.
No change in the end result, but just the beautification.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that we can avoid the extra goto lines.
Also beautify the code to follow the standard codex.
No functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The stream direction in open and close callbacks can be retrieved from
substream->direction, hence we don't have to stick with the unique PCM
ops hard-coded for each direction. Rewrite the common open/close
callback functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
retire_capture_urb() may print warning messages when the given URB
doesn't align, and this may flood the system log easily.
Put the rate limit to the message for avoiding it.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1093485
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new flag to struct snd_usb_audio for allowing the device
to skip usb_set_interface() calls at changing or closing the stream.
As of this patch, the flag is nowhere set, so it's just a place
holder. The dynamic switching will be added in the following patch.
A background information for this change:
Dell WD15 dock with Realtek chip gives a very long pause at each time
the driver changes the altset, which eventually happens at every PCM
stream open/close and parameter change. As the long pause happens in
each usb_set_interface() call, there is nothing we can do as long as
it's called. The workaround is to reduce calling it as much as
possible, and this flag indicates that behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary change for the upcoming quirk implementation.
Currently USB-audio driver tries to call usb_set_interface() whenever
the format change with interface/altset modification happens. In this
patch, the check is replaced with the comparison of cur_altsetting and
the targeted altsetting pointer, so that the driver may skip the
unnecessary function calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Axe-Fx III implicit feedback end point and the data sink endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add quirk to ensure a sync endpoint is properly configured.
This patch is a fix for same symptoms on Behringer UFX1204 as patch
from Albertto Aquirre on Dec 8 2016 for Axe-Fx II.
Signed-off-by: Lassi Ylikojola <lassi.ylikojola@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/pcm.h> work with
const snd_pcm_ops. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make this const as it is only used in a copy operation.
Done using Coccinelle.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream") was
incomplete causing another more severe kernel panic, so it got reverted.
This fixes both the original problem and its fallout kernel race/crash.
The original fix is to move the endpoint member NULL clearing logic inside
wait_clear_urbs() so the irq triggering the urb completion doesn't call
retire_capture/playback_urb() after the NULL clearing and generate a panic.
However this creates a new race between snd_usb_endpoint_start()'s call
to wait_clear_urbs() and the irq urb completion handler which again calls
retire_capture/playback_urb() leading to a new NULL dereference.
We keep the EP deactivation code in snd_usb_endpoint_start() because
removing it will break the EP reference counting (see [1] [2] for info),
however we don't need the "can_sleep" mechanism anymore because a new
function was introduced (snd_usb_endpoint_sync_pending_stop()) which
synchronizes pending stops and gets called inside the pcm prepare callback.
It also makes sense to remove can_sleep because it was also removed from
deactivate_urbs() signature in [3] so we benefit from more simplification.
[1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start")
[2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream")
[3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code")
Fixes: f8114f8583 ("Revert "ALSA: usb-audio: Fix race at stopping the stream"")
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Axe-Fx II implicit feedback end point and the data sync endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of userland applications call 'snd_pcm_hw_params()' and
'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()'
is called twice and the second 'snd_pcm_hw_prepare()' is called in
'SNDRV_PCM_STATE_PREPARED' state.
Some devices are not able to manage this and they will stop playback
if the sample rate will be configured several times over USB protocol.
V2: updated Changelog
Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com>
Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2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=Cc+v
-----END PGP SIGNATURE-----
Merge tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514"
* tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits)
ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug
ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
ALSA: mixart: silence an uninitialized variable warning
ALSA: usb-audio: Add sanity checks for endpoint accesses
ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
ALSA: hda - Limit i915 HDMI binding only for HSW and later
ALSA: hda - Fix unconditional GPIO toggle via automute
ALSA: mixart: silence unitialized variable warnings
ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
ASoC: rsnd: add simplified module explanation
ASoC: hdac_hdmi: Add broxton device ID
ASoC: Intel: Bxtn: Add Broxton PCI ID
ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops
ASoC: Intel: add dmabuffer to common sst_dsp
ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core
ASoC: Intel: Skylake: Fix whitepsace issues
ASoC: Intel: Skylake: Move module id defines
...
Add some sanity check codes before actually accessing the endpoint via
get_endpoint() in order to avoid the invalid access through a
malformed USB descriptor. Mostly just checking bNumEndpoints, but in
one place (snd_microii_spdif_default_get()), the validity of iface and
altsetting index is checked as well.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
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>
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>
Geoffrey D. Bennett