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>
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertently introduced[3] to the codebase from now on.
Also, notice that, dynamic memory allocations won't be affected by
this change:
"Flexible array members have incomplete type, and so the sizeof operator
may not be applied. As a quirk of the original implementation of
zero-length arrays, sizeof evaluates to zero."[1]
sizeof(flexible-array-member) triggers a warning because flexible array
members have incomplete type[1]. There are some instances of code in
which the sizeof operator is being incorrectly/erroneously applied to
zero-length arrays and the result is zero. Such instances may be hiding
some bugs. So, this work (flexible-array member conversions) will also
help to get completely rid of those sorts of issues.
This issue was found with the help of Coccinelle.
[1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html
[2] https://github.com/KSPP/linux/issues/21
[3] commit 7649773293 ("cxgb3/l2t: Fix undefined behaviour")
Signed-off-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Link: https://lore.kernel.org/r/20200507192223.GA16335@embeddedor
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At least POD HD500 uses message-based communication, both sides can
send messages. Add poll callback so application can wait for device
messages without using busy loop.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Link: https://lore.kernel.org/r/20200502193120.79115-3-anarsoul@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently line6 hwdep interface ignores O_NONBLOCK flag when
opening device and it renders it somewhat useless when using poll.
Check for O_NONBLOCK flag when opening device and don't block read()
if it is set.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Link: https://lore.kernel.org/r/20200502193120.79115-2-anarsoul@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The linked list entry from FIFO is peeked at
queue_pending_output_urbs() but the actual element pop-out is
performed outside the spinlock, and it's potentially racy.
Do delete the link at the right place inside the spinlock.
Fixes: 8fdff6a319 ("ALSA: snd-usb: implement new endpoint streaming model")
Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which
increases the refcount of the snd_usb_audio object "chip".
When snd_microii_spdif_default_get() returns, local variable "chip"
becomes invalid, so the refcount should be decreased to keep refcount
balanced.
The reference counting issue happens in several exception handling paths
of snd_microii_spdif_default_get(). When those error scenarios occur
such as usb_ifnum_to_if() returns NULL, the function forgets to decrease
the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak.
Fix this issue by jumping to "end" label when those error scenarios
occur.
Fixes: 447d6275f0 ("ALSA: usb-audio: Add sanity checks for endpoint accesses")
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that ALC1220-VB USB-audio device gives the interrupt
event to some PCM terminals while those don't allow the connector
state request but only the actual I/O terminals return the request.
The recent commit 7dc3c5a017 ("ALSA: usb-audio: Don't create jack
controls for PCM terminals") excluded those phantom terminals, so
those events are ignored, too.
My first thought was that this could be easily deduced from the
associated terminals, but some of them have even no associate terminal
ID, hence it's not too trivial to figure out.
Since the number of such terminals are small and limited, this patch
implements another quirk table for the simple mapping of the
connectors. It's not really scalable, but let's hope that there will
be not many such funky devices in future.
Fixes: 7dc3c5a017 ("ALSA: usb-audio: Don't create jack controls for PCM terminals")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200422113320.26664-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This should be ARRAY_SIZE() instead of sizeof(). The sizeof() limit is
too high so it doesn't work.
Fixes: 093b8494f2 ("ALSA: usb-audio: Print more information in stream proc files")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200422092255.GB195357@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to rounding error driver sometimes incorrectly calculate next packet
size, which results in audible clicks on devices with synchronous playback
endpoints. For example on a high speed bus and a sample rate 44.1 kHz it
loses one sample every ~40.9 seconds. Fortunately playback interface on
Scarlett 2i4 2nd gen has a working explicit feedback endpoint, so we can
switch playback data endpoint to asynchronous mode as a workaround.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200421190908.462860-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error handling code in usX2Y_rate_set() may hit a potential NULL
dereference when an error occurs before allocating all us->urb[].
Add a proper NULL check for fixing the corner case.
Reported-by: Lin Yi <teroincn@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200420075529.27203-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Force it to use asynchronous playback.
Same quirk has already been added for Focusrite Scarlett Solo (2nd gen)
with a commit 46f5710f0b ("ALSA: usb-audio: Add quirk for Focusrite
Scarlett Solo").
This also seems to prevent regular clicks when playing at 44100Hz
on Scarlett 2i2 (2nd gen). I did not notice any side effects.
Moved both quirks to snd_usb_audioformat_attributes_quirk() as suggested.
Signed-off-by: Gregor Pintar <grpintar@gmail.com>
Reviewed-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need
yet more quirks for the proper control names.
This patch provides the mapping table for those boards, correcting the
FU names for volume and mute controls as well as the terminal names
for jack controls. It also improves build_connector_control() not to
add the directional suffix blindly if the string is given from the
mapping table.
With this patch applied, the new UCM profiles will be effective.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Link: https://lore.kernel.org/r/20200420062036.28567-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many Focusrite devices supports a limited set of sample rates per
altsetting. These includes audio interfaces with ADAT ports:
- Scarlett 18i6, 18i8 1st gen, 18i20 1st gen;
- Scarlett 18i8 2nd gen, 18i20 2nd gen;
- Scarlett 18i8 3rd gen, 18i20 3rd gen;
- Clarett 2Pre USB, 4Pre USB, 8Pre USB.
Maximum rate is exposed in the last 4 bytes of Format Type descriptor
which has a non-standard bLength = 10.
Tested-by: Alexey Skobkin <skobkin-ru@ya.ru>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200418175815.12211-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added mixer quirks to allow controlling the internal DSP of the
RME Babyface Pro and its successor Babyface Pro FS.
Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Link: https://lore.kernel.org/r/20200414211019.qprg7whepg2y7nei@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the mapping check to build_connector_control() so that the device
specific quirk can provide the node to skip for the badly behaving
connector controls. As an example, ALC1220-VB-based codec implements
the skip entry for the broken SPDIF connector detection.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mapping table may contain also ignore_ctl_error flag for devices
that are known to behave wild. Since this flag always writes the
card's own ignore_ctl_error flag, it overrides the value already set
by the module option, so it doesn't follow user's expectation.
Let's fix the code not to clear the flag that has been set by user.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ignore_ctl_error option should filter the error at kctl accesses,
but there was an overlook: mixer_ctl_connector_get() returns an error
from the request.
This patch covers the forgotten code path and apply filter_error()
properly. The locking error is still returned since this is a fatal
error that has to be reported even with ignore_ctl_error option.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200412081331.4742-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some recent boards (supposedly with a new AMD platform) contain the
USB audio class 2 device that is often tied with HD-audio. The device
exposes an Input Gain Pad control (id=19, control=12) but this node
doesn't behave correctly, returning an error for each inquiry of
GET_MIN and GET_MAX that should have been mandatory.
As a workaround, simply ignore this node by adding a usbmix_name_map
table entry. The currently known devices are:
* 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi
* 0b05:1916 - ASUS ROG Zenith II
* 0b05:1917 - ASUS ROG Strix
* 0db0:0d64 - MSI TRX40 Creator
* 0db0:543d - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:16d8) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.
Signed-off-by: Emmanuel Pescosta <emmanuelpescosta099@gmail.com>
Link: https://lore.kernel.org/r/20200404153843.9288-1-emmanuelpescosta099@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card.
The MIDI controller part is standard but the PCM part is "vendor specific".
Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE.
Input is not working.
Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra
endpoint descriptor.
The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00
As the code in snd_usbmidi_get_ms_info() looks only at the
first extra descriptor to find USB_DT_CS_ENDPOINT the device
as such is recognized but there is neither input nor output
configured.
The patch iterates through the extra descriptors to find the
proper one. With this patch the device is correctly configured.
Signed-off-by: Andreas Steinmetz <ast@domdv.de>
Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@domdv.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a very big update for the core since Morimoto-san has been
rather busy continuing his refactorings to clean up a lot of the cruft
that we have accumilated over the years. We've also gained several new
drivers, including initial (but still not complete) parts of the Intel
SoundWire support.
- Lots of refactorings to modernize the code from Morimoto-san.
- Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven.
- Continued refactoring and fixing of the Intel support.
- Soundwire and more advanced clocking support for Realtek RT5682.
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and
TLV320ADCX140.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl6B03sTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0Jm5B/4pb6Hz1ifM+ikR4qB7BOdm/UGzxWlD
Mng+qN3kh6Tx+x8/xnQRmfOnhFoKnDSTnlxcDr6ujAtz3bH+CKXbo2jjfvmagQrW
j87tYGdh6T38E6mf6543Eth3zJGphZYUY70REXt0HsaQ8bA912nrs5H8LTFiD2i+
lBJd/MmlCYL3gb/ftN9dpTfYYdxLy9HH9j5tR5zpLN3AD3Hew9jHqqX+ioAk+6lA
Nisbx5grP10pWFklYXyJlUFQOqMOYMtPMblLRA9bjqcEtL3quIXPXKkPNtTh+0Kf
5t7Q39RtVXJAk0DcatjMJUGAGQyM9r5g1Nk8WLB1LLv+NfLcO6cUpLAc
=+z/q
-----END PGP SIGNATURE-----
Merge tag 'asoc-v5.7' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.7
This is a very big update for the core since Morimoto-san has been
rather busy continuing his refactorings to clean up a lot of the cruft
that we have accumilated over the years. We've also gained several new
drivers, including initial (but still not complete) parts of the Intel
SoundWire support.
- Lots of refactorings to modernize the code from Morimoto-san.
- Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven.
- Continued refactoring and fixing of the Intel support.
- Soundwire and more advanced clocking support for Realtek RT5682.
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and
TLV320ADCX140.
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>
Add a new option for specifying the quirk for delayed registration of
the certain device. A list of devices can be passed in a form
ID:IFACE,ID:IFACE,ID:IFACE,....
where ID is the 32bit hex number combo of vendor and device IDs and
IFACE is the interface number to trigger the register.
When a matching device is probed, the card registration is delayed
until the given interface is probed. It's needed for syncing the
registration until the last interface when multiple interfaces are
provided for the same card.
Link: https://lore.kernel.org/r/20200325103322.2508-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A slight refactoring of the registration quirk code. Now it uses the
table lookup for easy additions in future. Also the return type was
changed to bool, and got a few more comments.
Link: https://lore.kernel.org/r/20200325103322.2508-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a quirk that allows special processing and/or
skipping the call to snd_card_register.
For HyperX AMP, which uses two interfaces, but only has
a capture stream in the second, this allows the capture
stream to merge with the first PCM.
Signed-off-by: Chris Wulff <crwulff@gmail.com>
Link: https://lore.kernel.org/r/20200314165449.4086-3-crwulff@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the USB interface of the mixer that the control
was created on instead of the default control interface.
This fixes the Kingston HyperX AMP (0951:16d8) which has
controls on two interfaces.
Signed-off-by: Chris Wulff <crwulff@gmail.com>
Link: https://lore.kernel.org/r/20200314165449.4086-2-crwulff@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MIDI input event parser of the LINE6 driver may enter into an
endless loop when the unexpected data sequence is given, as it tries
to continue the secondary bytes without termination. Also, when the
input data is too short, the parser returns a negative error, while
the caller doesn't handle it properly. This would lead to the
unexpected behavior as well.
This patch addresses those issues by checking the return value
correctly and handling the one-byte event in the parser properly.
The bug was reported by syzkaller.
Reported-by: syzbot+cce32521ee0a824c21f7@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com
Link: https://lore.kernel.org/r/20200309095922.30269-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The struct s1810c_state_packet contains the array in the first field
hence zero-initialization requires a more couple of braces. Fix the
compile warning pointing it out:
sound/usb/mixer_s1810c.c: In function 'snd_sc1810c_get_status_field':
sound/usb/mixer_s1810c.c:178:9: warning: missing braces around initializer [-Wmissing-braces]
Reported-by: kbuild test robot <lkp@intel.com>
Fixes: 8dc5efe3d1 ("ALSA: usb-audio: Add support for Presonus Studio 1810c")
Link: https://lore.kernel.org/r/202002210251.WgMfvKJP%lkp@intel.com
Link: https://lore.kernel.org/r/20200306081231.7940-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Merging the UAC2 effect unit parser improvement. As it's based on the
previous usb-audio driver fix, it was deviated from for-next branch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During parsing the input source, we currently cut off at the Effect
Unit node without parsing further its source id. It's no big problem,
so far, but it should be more consistent to parse it properly.
This patch adds the recursive parsing in parse_term_effect_unit().
It doesn't add anything in the audio unit parser itself, and the
effect unit itself is still skipped, though.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206147
Link: https://lore.kernel.org/r/20200213112059.18745-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for Presonus Studio 1810c, a usb interface
that's UAC2 compliant with a few quirks and a few extra hw-specific
controls. I've tested all 3 altsettings and the added switch
controls and they work as expected.
More infos on the card:
https://www.presonus.com/products/Studio-1810c
Note that this work is based on packet inspection with
usbmon. I just wanted to get this card to work for using
it on our open-source radio station:
https://github.com/UoC-Radio
v2 address issues reported by Takashi:
* Properly get/set enum type controls
* Prevent race condition on switch_get/set
* Various control naming changes
* Various coding style fixes
v3 improve readability of sample rate filtering
and some other minor changes.
Signed-off-by: Nick Kossifidis <mickflemm@gmail.com>
Link: https://lore.kernel.org/r/5e47481a.1c69fb81.befb3.8dac@mx.google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It should be safe to ignore clock validity check result if the following
conditions are met:
- only one single sample rate is supported;
- the terminal is directly connected to the clock source;
- the clock type is internal.
This is to deal with some Denon DJ controllers that always reports that
clock is invalid.
Tested-by: Tobias Oszlanyi <toszlanyi@yahoo.de>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200212235450.697348-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertenly introduced[3] to the codebase from now on.
This issue was found with the help of Coccinelle.
[1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html
[2] https://github.com/KSPP/linux/issues/21
[3] commit 7649773293 ("cxgb3/l2t: Fix undefined behaviour")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Link: https://lore.kernel.org/r/20200211194224.GA9383@embeddedor
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Audioengine D1 (0x2912:0x30c8) does support reading the sample rate,
but it returns the rate in byte-reversed order.
When setting sampling rate, the driver produces these warning messages:
[168840.944226] usb 3-2.2: current rate 4500480 is different from the runtime rate 44100
[168854.930414] usb 3-2.2: current rate 8436480 is different from the runtime rate 48000
[168905.185825] usb 3-2.1.2: current rate 30465 is different from the runtime rate 96000
As can be seen from the hexadecimal conversion, the current rate read
back is byte-reversed from the rate that was set.
44100 == 0x00ac44, 4500480 == 0x44ac00
48000 == 0x00bb80, 8436480 == 0x80bb00
96000 == 0x017700, 30465 == 0x007701
Rather than implementing a new quirk to reverse the order, just skip
checking the rate to avoid spamming the log.
Signed-off-by: Arvind Sankar <nivedita@alum.mit.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200211162235.1639889-1-nivedita@alum.mit.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a regression report about M-Audio Fast Track C400 device,
and the git bisection resulted in the commit e0ccdef926 ("ALSA:
usb-audio: Clean up check_input_term()"). This commit was about the
rewrite of the input terminal parser, and it's not too obvious from
the change what really broke. The answer is: it's the interpretation
of UAC2/3 effect units.
In the original code, UAC2 effect unit is as if through UAC1
processing unit because both UAC1 PU and UAC2/3 EU share the same
number (0x07). The old code went through a complex switch-case
fallthrough, finally bailing out in the middle:
if (protocol == UAC_VERSION_2 &&
hdr[2] == UAC2_EFFECT_UNIT) {
/* UAC2/UAC1 unit IDs overlap here in an
* uncompatible way. Ignore this unit for now.
*/
return 0;
}
... and this special handling was missing in the new code; the new
code treats UAC2/3 effect unit as if it were equivalent with the
processing unit.
Actually, the old code was too confusing. The effect unit has an
incompatible unit description with the processing unit, so we
shouldn't have dealt with EU in the same way.
This patch addresses the regression by changing the effect unit
handling to the own parser function. The own parser function makes
the clear distinct with PU, so it improves the readability, too.
The EU parser just sets the type and the id like the old kernels.
Once when the proper effect unit support is added, we can revisit this
parser function, but for now, let's keep this simple setup as is.
Fixes: e0ccdef926 ("ALSA: usb-audio: Clean up check_input_term()")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206147
Link: https://lore.kernel.org/r/20200211160521.31990-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jabra Evolve 65 headset appears as if supporting lower rates than
48kHz, but it actually doesn't work but with 48kHz for playback.
This patch applies a workaround to enforce the 48kHz like LINE6
devices already did. The workaround is put in a unified helper
function, set_fixed_rate(), to be called from both places now.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206149
Link: https://lore.kernel.org/r/20200211111419.5895-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new macro can fix the sparse warnings gracefully:
sound/usb/proc.c:73:31: warning: restricted snd_pcm_format_t degrades to integer
sound/usb/proc.c:73:38: warning: restricted snd_pcm_format_t degrades to integer
sound/usb/proc.c:73:61: warning: restricted snd_pcm_format_t degrades to integer
No functional changes, just sparse warning fixes.
Link: https://lore.kernel.org/r/20200206163945.6797-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
