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[ Upstream commit 1ef8715975de8bd481abbd0839ed4f49d9e5b0ff ]
Fix:
sound/usb/midi.c: In function ‘snd_usbmidi_out_endpoint_create’:
sound/usb/midi.c:1389:2: error: case label does not reduce to an integer constant
case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */
^~~~
See https://lore.kernel.org/r/YkwQ6%2BtIH8GQpuct@zn.tnic for the gory
details as to why it triggers with older gccs only.
[ A slight correction with parentheses around the argument by tiwai ]
Signed-off-by: Borislav Petkov <bp@suse.de>
Link: https://lore.kernel.org/r/20220405151517.29753-3-bp@alien8.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 0665886ad1392e6b5bae85d7a6ccbed48dca1522 upstream.
When a rawmidi output stream is closed, it calls the drain at first,
then does trigger-off only when the drain returns -ERESTARTSYS as a
fallback. It implies that each driver should turn off the stream
properly after the drain. Meanwhile, USB-audio MIDI interface didn't
change the port->active flag after the drain. This may leave the
output work picking up the port that is closed right now, which
eventually leads to a use-after-free for the already released rawmidi
object.
This patch fixes the bug by properly clearing the port->active flag
after the output drain.
Reported-by: syzbot+70e777a39907d6d5fd0a@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/00000000000011555605dceaff03@google.com
Link: https://lore.kernel.org/r/20220420130247.22062-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 24d0c9f0e7de95fe3e3e0067cbea1cd5d413244b ]
In the previous fix, we increased the max buffer bytes from 1MB to 4MB
so that we can use bigger buffers for the modern HiFi devices with
higher rates, more channels and wider formats. OTOH, extending this
has a concern that too big buffer is allowed for the lower rates, less
channels and narrower formats; when an application tries to allocate
as big buffer as possible, it'll lead to unexpectedly too huge size.
Also, we had a problem about the inconsistent max buffer and period
bytes for the implicit feedback mode when both streams have different
channels. This was fixed by the (relatively complex) patch to reduce
the max buffer and period bytes accordingly.
This is an alternative fix for those, a patch to kill two birds with
one stone (*): instead of increasing the max buffer bytes blindly and
applying the reduction per channels, we simply use the hw constraints
for the buffer and period "time". Meanwhile the max buffer and period
bytes are set unlimited instead.
Since the inconsistency of buffer (and period) bytes comes from the
difference of the channels in the tied streams, as long as we care
only about the buffer (and period) time, it doesn't matter; the buffer
time is same for different channels, although we still allow higher
buffer size. Similarly, this will allow more buffer bytes for HiFi
devices while it also keeps the reasonable size for the legacy
devices, too.
As of this patch, the max period and buffer time are set to 1 and 2
seconds, which should be large enough for all possible use cases.
(*) No animals were harmed in the making of this patch.
Fixes: 98c27add5d96 ("ALSA: usb-audio: Cap upper limits of buffer/period bytes for implicit fb")
Fixes: fee2ec8cceb3 ("ALSA: usb-audio: Increase max buffer size")
Link: https://lore.kernel.org/r/20220412130740.18933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit fee2ec8cceb33b8886bc5894fb07e0b2e34148af ]
The current limit of max buffer size 1MB seems too small for modern
devices with lots of channels and high sample rates.
Let's make bigger, 4MB.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20220407212740.17920-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 98c27add5d96485db731a92dac31567b0486cae8 upstream.
In the implicit feedback mode, some parameters are tied between both
playback and capture streams. One of the tied parameters is the
period size, and this can be a problem if the device has different
number of channels to both streams. Assume that an application opens
a playback stream that has an implicit feedback from a capture stream,
and it allocates up to the max period and buffer size as much as
possible. When the capture device supports only more channels than
the playback, the minimum period and buffer sizes become larger than
the sizes the playback stream took. That is, the minimum size will be
over the max size the driver limits, and PCM core sees as if no
available configuration is found, returning -EINVAL mercilessly.
For avoiding this problem, we have to look through the counter part of
audioformat list for each sync ep, and checks the channels. If more
channels are found there, we reduce the max period and buffer sizes
accordingly.
You may wonder that the patch adds only the evaluation of channels
between streams, and what about other parameters? Both the format and
the rate are tied in the implicit fb mode, hence they are always
identical.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215792
Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220407211657.15087-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0f306cca42fe879694fb5e2382748c43dc9e0196 upstream.
For the RODE NT-USB the lowest Playback mixer volume setting mutes the
audio output. But it is not reported as such causing e.g. PulseAudio to
accidentally mute the device when selecting a low volume.
Fix this by applying the existing quirk for this kind of issue when the
device is detected.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220311201400.235892-1-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cd94df1795418056a19ff4cb44eadfc18ac99a57 upstream.
New device id for Corsair Virtuoso SE RGB Wireless that currently is not
in the mixer_map. This entry in the mixer_map is necessary in order to
label its mixer appropriately and allow userspace to pick the correct
volume controls. For instance, my own Corsair Virtuoso SE RGB Wireless
headset has this new ID and consequently, the sidetone and volume are not
working correctly without this change.
> sudo lsusb -v | grep -i corsair
Bus 007 Device 011: ID 1b1c:0a40 Corsair CORSAIR VIRTUOSO SE Wireless Gam
idVendor 0x1b1c Corsair
iManufacturer 1 Corsair
iProduct 2 CORSAIR VIRTUOSO SE Wireless Gaming Headset
Signed-off-by: Reza Jahanbakhshi <reza.jahanbakhshi@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220304212303.195949-1-reza.jahanbakhshi@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 19d20c7a29bf2e46ff1ab8e8c4fcd2da8a4f38e2 upstream.
Commit 83b7dcbc51c930fc2079ab6c6fc9d719768321f1 introduced a generic
implicit feedback parser, which fails to execute for M-Audio FastTrack
Ultra sound cards. The issue is with the ENDPOINT_SYNCTYPE check in
add_generic_implicit_fb() where the SYNCTYPE is ADAPTIVE instead of ASYNC.
The reason is that the sync type of the FastTrack output endpoints are
set to adaptive in the quirks table since commit
65f04443c96dbda11b8fff21d6390e082846aa3c.
Fixes: 83b7dcbc51c9 ("ALSA: usb-audio: Add generic implicit fb parsing")
Signed-off-by: Matteo Martelli <matteomartelli3@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220211224913.20683-2-matteomartelli3@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 3da4b7403db87d39bc2613cfd790de1de99a70ab upstream.
clang static analysis reports this representative issue
mixer.c:1548:35: warning: Assigned value is garbage or undefined
ucontrol->value.integer.value[0] = val;
^ ~~~
The filter_error() macro allows errors to be ignored.
If errors can be ignored, initialize variables
so garbage will not be used.
Fixes: 48cc42973509 ("ALSA: usb-audio: Filter error from connector kctl ops, too")
Signed-off-by: Tom Rix <trix@redhat.com>
Link: https://lore.kernel.org/r/20220126182142.1184819-1-trix@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 4ee02e20893d2f9e951c7888f2284fa608ddaa35 upstream.
This device provides both audio and video. The original quirk added in
commit 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770") used
USB_DEVICE to match the vendor and product ID. Depending on module order,
if snd-usb-audio was asked first, it would match the entire device and
uvcvideo wouldn't get to see it. Change the matching to USB_AUDIO_DEVICE
to restore uvcvideo matching in all cases.
Fixes: 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770")
Reported-by: Jukka Heikintalo <heikintalo.jukka@gmail.com>
Tested-by: Jukka Heikintalo <heikintalo.jukka@gmail.com>
Reported-by: Paweł Susicki <pawel.susicki@gmail.com>
Tested-by: Paweł Susicki <pawel.susicki@gmail.com>
Cc: <stable@vger.kernel.org> # 5.4, 5.10, 5.14, 5.15
Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de>
Link: https://lore.kernel.org/r/20220131183516.61191-1-hahnjo@hahnjo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 1e583aef12aa74afd37c1418255cc4b74e023236 ]
The vendor ID of Presonus Studio 1810c had a superfluous '0' in its
USB ID. Drop it.
Fixes: 8dc5efe3d17c ("ALSA: usb-audio: Add support for Presonus Studio 1810c")
Link: https://lore.kernel.org/r/20211202083833.17784-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit fb1af5bea4670c835e42fc0c14c49d3499468774 upstream.
Olivia Mackintosh has posted to alsa-devel reporting that
there's a potential bug that could break mixer quirks for Pioneer
devices introduced by 6d27788160362a7ee6c0d317636fe4b1ddbe59a7
"ALSA: usb-audio: Add support for the Pioneer DJM 750MK2
Mixer/Soundcard".
This happened because the DJM 750 MK2 was added last to the Pioneer DJM
device table index and defined as 0x4 but was added to snd_djm_devices[]
just after the DJM 750 (MK1) entry instead of last, after the DJM 900
NXS2. This escaped review.
To prevent that from ever happening again, Takashi Iwai suggested to use
C99 array designators in snd_djm_devices[] instead of simply reordering
the entries.
Fixes: 6d2778816036 ("ALSA: usb-audio: Add support for the Pioneer DJM 750MK2")
Reported-by: Olivia Mackintosh <livvy@base.nu>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Geraldo Nascimento <geraldogabriel@gmail.com>
Link: https://lore.kernel.org/r/Yau46FDzoql0SNnW@geday
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 83de8f83816e8e15227dac985163e3d433a2bf9d upstream.
The recent change made mistakenly the stream for capture started at
prepare stage. Add the stream direction check to avoid it.
Fixes: 9c9a3b9da891 ("ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback")
Link: https://lore.kernel.org/r/20211119102629.7476-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit eee5d6f1356a016105a974fb176b491288439efa upstream.
The recent regression report revealed that the judgment of the
low-latency playback mode based on the runtime->stop_threshold cannot
work reliably at the prepare stage, as sw_params call may happen at
any time, and PCM dmix actually sets it up after the prepare call.
This ended up with the stall of the stream as PCM ack won't be issued
at all.
For addressing this, check the free-wheeling mode again at the PCM
trigger right before starting the stream again, and allow switching to
the non-LL mode at a late stage.
Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support")
Reported-and-tested-by: Kirill A. Shutemov <kirill.shutemov@linux.intel.com>
Link: https://lore.kernel.org/r/20211117161855.m45mxcqszkfcetai@box.shutemov.name
Link: https://lore.kernel.org/r/20211119102459.7055-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 53451b6da8271905941eb1eb369db152c4bd92f2 upstream.
The recent support for the improved low-latency playback mode applied
the SNDRV_PCM_INFO_EXPLICIT_SYNC flag for the target streams, but this
was a slight overkill. The use of the flag above disables effectively
both PCM status and control mmaps, while basically what we want to
track is only about the appl_ptr update.
For less restriction, use a more proper flag,
SNDRV_PCM_INFO_SYNC_APPLPTR instead, which disables only the control
mmap.
Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support")
Link: https://lore.kernel.org/r/20211011103650.10182-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 23939115be181bc5dbc33aa8471adcdbffa28910 upstream.
The commit d215f63d49da ("ALSA: usb-audio: Check available frames for
the next packet size") introduced the available frame size check, but
the conversion forgot to initialize the temporary variable properly,
and it resulted in a bogus calculation. This patch fixes it.
Fixes: d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size")
Reported-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20211001104417.14291-1-colin.king@canonical.com
Link: https://lore.kernel.org/r/20211001105425.16191-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 813a17cab9b708bbb1e0db8902e19857b57196ec upstream.
While draining a stream, ALSA PCM core stops the stream by issuing
snd_pcm_stop() after all data has been sent out. And, at PCM trigger
stop, currently USB-audio driver kills the in-flight URBs explicitly,
then at sync-stop ops, sync with the finish of all remaining URBs.
This might result in a drop of the drained samples as most of
USB-audio devices / hosts allow relatively long in-flight samples (as
a sort of FIFO).
For avoiding the trimming, this patch changes the stream-stop behavior
during PCM draining state. Under that condition, the pending URBs
won't be killed. The leftover in-flight URBs are caught by the
sync-stop operation that shall be performed after the trigger-stop
operation.
Link: https://lore.kernel.org/r/20210929080844.11583-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d5f871f89e21bb71827ea57bd484eedea85839a0 upstream.
This is another attempt to improve further the handling of playback
stream in the low latency mode. The latest workaround in commit
4267c5a8f313 ("ALSA: usb-audio: Work around for XRUN with low latency
playback") revealed that submitting URBs forcibly in advance may
trigger XRUN easily. In the classical mode, this problem was avoided
by practically delaying the submission of the actual data with the
pre-submissions of silent data before triggering the stream start.
But that is exactly what we want to avoid.
Now, in this patch, instead of the previous workaround, we take a
similar approach as used in the implicit feedback mode. The URBs are
queued at the PCM trigger start like before, but we check whether the
buffer has been already filled enough before each submission, and
stop queuing if the data overcomes the threshold. The remaining URBs
are kept in the ready list, and they will be retrieved in the URB
complete callback of other (already queued) URBs. In the complete
callback, we try to fill the data and submit as much as possible
again. When there is no more available in-flight URBs that may handle
the pending data, we'll check in PCM ack callback and submit and
process URBs there in addition. In this way, the amount of in-flight
URBs may vary dynamically and flexibly depending on the available data
without hitting XRUN.
The following things are changed to achieve the behavior above:
* The endpoint prepare callback is changed to return an error code;
when there is no enough data available, it may return -EAGAIN.
Currently only prepare_playback_urb() returns the error.
The evaluation of the available data is a bit messy here; we can't
check with snd_pcm_avail() at the point of prepare callback (as
runtime->status->hwptr hasn't been updated yet), hence we manually
estimate the appl_ptr and compare with the internal hwptr_done to
calculate the available frames.
* snd_usb_endpoint_start() doesn't submit full URBs if the prepare
callback returns -EAGAIN, and puts the remaining URBs to the ready
list for the later submission.
* snd_complete_urb() treats the URBs in the low-latency mode similarly
like the implicit feedback mode, and submissions are done in
(now exported) snd_usb_queue_pending_output_urbs().
* snd_usb_queue_pending_output_urbs() again checks the error value
from the prepare callback. If it's -EAGAIN for the normal stream
(i.e. not implicit feedback mode), we push it back to the ready list
again.
* PCM ack callback is introduced for the playback stream, and it calls
snd_usb_queue_pending_output_urbs() if there is no in-flight URB
while the stream is running. This corresponds to the case where the
system needs the appl_ptr update for re-submitting a new URB.
* snd_usb_queue_pending_output_urbs() and the prepare EP callback
receive in_stream_lock argument, which is a bool flag indicating the
call path from PCM ack. It's needed for avoiding the deadlock of
snd_pcm_period_elapsed() calls.
* Set the new SNDRV_PCM_INFO_EXPLICIT_SYNC flag when the new
low-latency mode is deployed. This assures catching each applptr
update even in the mmap mode.
Fixes: 4267c5a8f313 ("ALSA: usb-audio: Work around for XRUN with low latency playback")
Link: https://lore.kernel.org/r/20210929080844.11583-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0ef74366bc150dda4f53c546dfa6e8f7c707e087 upstream.
In theory, stop_urbs() may be called concurrently.
Although we have the state check beforehand, it's safer to apply
ep->lock during the critical list head manipulations.
Link: https://lore.kernel.org/r/20210929080844.11583-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d215f63d49da9a8803af3e81acd6cad743686573 upstream.
This is yet more preparation for the upcoming changes.
Extend snd_usb_endpoint_next_packet_size() to check the available
frames and return -EAGAIN if the next packet size is equal or exceeds
the given size. This will be needed for avoiding XRUN during the low
latency operation.
As of this patch, avail=0 is passed, i.e. the check is skipped and no
behavior change.
Link: https://lore.kernel.org/r/20210929080844.11583-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit bceee75387554f682638e719d1ea60125ea78cea upstream.
When a playback stream runs in the implicit feedback mode, its
operation is passive and won't start unless the capture packet is
received. This behavior contradicts with the low-latency playback
mode, and we should turn off lowlatency_playback flag accordingly.
In theory, we may take the low-latency mode when the playback-first
quirk is set, but it still conflicts with the later operation with the
fixed packet numbers, so it's disabled all together for now.
Link: https://lore.kernel.org/r/20210929080844.11583-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit e581f1cec4f899f788f6c9477f805b1d5fef25e2 upstream.
The free-wheel stream operation like dmix may not update the appl_ptr
appropriately, and it doesn't fit with the low-latency playback mode.
Disable the low-latency playback operation when the stream is set up
in such a mode.
Link: https://lore.kernel.org/r/20210929080844.11583-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 9c9a3b9da891cc70405a544da6855700eddcbb71 upstream.
This is a preparation patch for the upcoming low-latency improvement
changes.
Rename early_playback_start flag with lowlatency_playback as it's more
intuitive. The new flag is basically a reverse meaning.
Along with the rename, factor out the code to set the flag to a
function. This makes the complex condition checks simpler.
Also, the same flag is introduced to snd_usb_endpoint, too, that is
carried from the snd_usb_substream flag. Currently the endpoint flag
isn't still referred, but will be used in later patches.
Link: https://lore.kernel.org/r/20210929080844.11583-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 4e7cf1fbb34ecb472c073980458cbe413afd4d64 upstream.
When a single clock source is shared among several endpoints, we have
to keep the same rate on all active endpoints as long as the clock is
being used. For dealing with such a case, this patch adds one more
check in the hw params constraint for the rate to take the shared
clocks into account. The current rate is evaluated from the endpoint
list that applies the same clock source.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190418
Link: https://lore.kernel.org/r/20210929080844.11583-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit b97053df0f04747c3c1e021ecbe99db675342954 ]
The pointer cs_desc return from snd_usb_find_clock_source could
be null, so there is a potential null pointer dereference issue.
Fix this by adding a null check before dereference.
Signed-off-by: Chengfeng Ye <cyeaa@connect.ust.hk>
Link: https://lore.kernel.org/r/20211024111736.11342-1-cyeaa@connect.ust.hk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 6d27788160362a7ee6c0d317636fe4b1ddbe59a7 ]
The kernel already has support for very similar Pioneer djm products
and this work is based on that.
Added device to quirks-table.h and added control info to
mixer_quirks.c.
Tested on my hardware and all working.
Signed-off-by: William Overton <willovertonuk@gmail.com>
Link: https://lore.kernel.org/r/20211010145841.11907-1-willovertonuk@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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UFX1604
[ Upstream commit 28c369e60827f706cef4604a3e2848198f25bd26 ]
Behringer UFX1204 and UFX1604 have Synchronous endpoints to which
current ALSA code applies implicit feedback sync as if they were
Asynchronous endpoints. This breaks UAC compliance and is unneeded.
The commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 and subsequent
1a15718b41df026cffd0e42cfdc38a1384ce19f9 were meant to clear up noise.
Unfortunately, noise persisted for those using higher sample rates and
this was only solved by commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964
Since there are no more reports of noise, let's get rid of the
implicit-fb quirks breaking UAC compliance.
Signed-off-by: Geraldo Nascimento <geraldogabriel@gmail.com>
Link: https://lore.kernel.org/r/YVYSnoQ7nxLXT0Dq@geday
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 86a42ad07905110f82648853c0ea3434b4eab173 ]
USB-audio driver tries to sync with the clear of all pending URBs in
wait_clear_urbs(), and it waits for all bits in active_mask getting
cleared. This works fine for the normal operations, but when a stream
is managed in the implicit feedback mode, there is still a very thin
race window: namely, in snd_complete_usb(), the active_mask bit for
the current URB is once cleared before re-submitted in
queue_pending_output_urbs(). If wait_clear_urbs() is called during
that period, it may pass the test and go forward even though there may
be a still pending URB.
For covering it, this patch adds a new counter to each endpoint to
keep the number of in-flight URBs, and changes wait_clear_urbs()
checking this number instead. The counter is decremented at the end
of URB complete, hence the reference is kept as long as the URB
complete is in process.
Link: https://lore.kernel.org/r/20210929080844.11583-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 763d92ed5dece7d439fc28a88b2d2728d525ffd9 upstream.
Add another device ID for JBL Quantum 400. It requires the same quirk as
other JBL Quantum devices.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211030174308.1011825-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 8f27b689066113a3e579d4df171c980c54368c4e upstream.
Adding the Line6 HX-Stomp XL USB_ID as it needs this fixed frequency
quirk as well.
The device is basically just the HX-Stomp with some more buttons on
the face. I've done some recording with it after adding it, and it
seems to function properly with this fix. The Midi features appear to
be working as well.
[ a coding style fix and patch reformat by tiwai ]
Signed-off-by: Jason Ormes <skryking@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211030200405.1358678-1-skryking@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f4000b58b64344871d7b27c05e73932f137cfef6 upstream.
USB control and interrupt message timeouts are specified in milliseconds
and should specifically not vary with CONFIG_HZ.
Fixes: 705ececd1c60 ("Staging: add line6 usb driver")
Cc: stable@vger.kernel.org # 2.6.30
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20211025121142.6531-3-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 9b371c6cc37f954360989eec41c2ddc5a6b83917 upstream.
USB control and bulk message timeouts are specified in milliseconds and
should specifically not vary with CONFIG_HZ.
Fixes: c6d43ba816d1 ("ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB")
Cc: stable@vger.kernel.org # 2.6.39
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20211025121142.6531-2-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 55f261b73a7e1cb254577c3536cef8f415de220a upstream.
Add the missing endpoint max-packet sanity check to probe() to avoid
division by zero in alloc_stream_buffers() in case a malicious device
has broken descriptors (or when doing descriptor fuzz testing).
Note that USB core will reject URBs submitted for endpoints with zero
wMaxPacketSize but that drivers doing packet-size calculations still
need to handle this (cf. commit 2548288b4fb0 ("USB: Fix: Don't skip
endpoint descriptors with maxpacket=0")).
Fixes: 63978ab3e3e9 ("sound: add Edirol UA-101 support")
Cc: stable@vger.kernel.org # 2.6.34
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20211026095401.26522-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit df0380b9539b04c1ae8854a984098da06d5f1e67 upstream.
Audient iD14 (2708:0002) may get a control message error that
interferes the operation e.g. with alsactl. Add the quirk to ignore
such errors like other devices.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1191247
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211102161859.19301-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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When a Jieli Technology USB Webcam is connected, the video part works
well, but the mic sound is speeded up. On dmesg there are messages
about different rates from the runtime rates, warnings about volume
resolution and lastly, the log is filled, every 5 seconds, with
retire_capture_urb error messages.
The mic works only when ep packet size is set to wMaxPacketSize (normal
sound and no more retire_capture_urb error messages). Skipping reading
sample rate, fixes the messages about different rates and forcing a volume
resolution, fixes warnings about volume range. I have arbitrarily choosed
the value (16): I read in a comment that there should be no more than 255
levels, so 4096 (max volume) / 16 = 0-255.
Signed-off-by: Marco Giunta <giun7a@gmail.com>
Link: https://lore.kernel.org/r/20211018162552.12082-1-giun7a@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As per discussion at: https://github.com/szszoke/sennheiser-gsp670-pulseaudio-profile/issues/13
The GSP670 has 2 playback and 1 recording device that by default are
detected in an incompatible order for alsa. This may have been done to make
it compatible for the console by the manufacturer and only affects the
latest firmware which uses its own ID.
This quirk will resolve this by reordering the channels.
Signed-off-by: Brendan Grieve <brendan@grieve.com.au>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211015025335.196592-1-brendan@grieve.com.au
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Shciit Hel device responds to the ctl message for the mic capture
switch with a timeout of -EPIPE:
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
This seems safe to ignore as the device works properly with the control
message quirk, so add it to the quirk table so all is good.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-usb@vger.kernel.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/YWgR3nOI1osvr5Yo@kroah.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The device advertises 8 formats, but only a rate of 48kHz is honored
by the hardware and 24 bits give chopped audio, so only report the
one working combination. This fixes out-of-the-box audio experience
with PipeWire which otherwise attempts to choose S24_3LE (while
PulseAudio defaulted to S16_LE).
Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211012200906.3492-1-hahnjo@hahnjo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Scarlett device series from Focusrite Novation seem requiring the
sample rate validations as we've done for MOTU devices; otherwise the
driver probes invalid audioformat entries that contain the sample
rates that actually don't work, and this may result in an incomplete
setup as reported recently.
This patch adds the needed quirk flag for enabling the sample rate
validation for Focusrite Novation devices.
Fixes: fe773b8711e3 ("ALSA: usb-audio: workaround for iface reset issue")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214493
Link: https://lore.kernel.org/r/20211004074050.28241-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The check of the returned error code is missing in
scarlett2_update_monitor_other(). Let's fix it.
Fixes: d5bda7e03982 ("ALSA: usb-audio: scarlett2: Add support for the talkback feature")
Reported-by: kernel test robot <lkp@intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/202109131831.9IodEzRx-lkp@intel.com
Link: https://lore.kernel.org/r/20210929073540.9611-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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USB-audio driver assumes that the normal resume would preserve the
device configuration while reset_resume wouldn't, and tries to restore
the mixer elements only at reset_resume callback. However, this seems
too naive, and some devices do behave differently, resetting the
volume at the normal resume; this resulted in the inconsistent volume
that surprised users.
This patch changes the mixer resume code to handle both the normal and
reset resume in the same way, always restoring the original mixer
element values. This allows us to unify the both callbacks as well as
dropping the no longer used reset_resume field, which ends up with a
good code reduction.
A slight behavior change by this patch is that now we assign
restore_mixer_value() as the default resume callback, and the function
is no longer called at reset-resume when the resume callback is
overridden by the quirk function. That is, if needed, the quirk
resume function would have to handle similarly as
restore_mixer_value() by itself.
Reported-by: En-Shuo Hsu <enshuo@chromium.org>
Cc: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsMew@mail.gmail.com
Link: https://lore.kernel.org/r/20210910105155.12862-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add another device ID for JBL Quantum 800. It requires the same quirk as
other JBL Quantum devices.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210831002531.116957-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For making user to switch back to the old playback mode, this patch
adds a new module option 'lowlatency' to snd-usb-audio driver.
When user face a regression due to the recent low-latency playback
support, they can test easily by passing lowlatency=0 option without
rebuilding the kernel.
Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2")
Link: https://lore.kernel.org/r/20210829073830.22686-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent change for low latency playback works in most of test cases
but it turned out still to hit errors on some use cases, most notably
with JACK with small buffer sizes. This is because USB-audio driver
fills up and submits full URBs at the beginning, while the URBs would
return immediately and try to fill more -- that can easily trigger
XRUN. It was more or less expected, but in the small buffer size, the
problem became pretty obvious.
Fixing this behavior properly would require the change of the
fundamental driver design, so it's no trivial task, unfortunately.
Instead, here we work around the problem just by switching back to the
old method when the given configuration is too fragile with the low
latency stream handling. As a threshold, we calculate the total
buffer bytes in all plus one URBs, and check whether it's beyond the
PCM buffer bytes. The one extra URB is needed because XRUN happens at
the next submission after the first round.
Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210827203311.5987-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent quirk for WALKMAN (commit 7af5a14371c1: "ALSA: usb-audio:
Fix regression on Sony WALKMAN NW-A45 DAC") may be required for other
devices and is worth to be put into the common quirk flags.
This patch adds a new quirk flag bit QUIRK_FLAG_SET_IFACE_FIRST and a
quirk table entry for the device.
Link: https://lore.kernel.org/r/20210824055720.9240-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We've got a regression report for USB-audio with Sony WALKMAN NW-A45
DAC device where no sound is audible on recent kernel. The bisection
resulted in the code change wrt endpoint management, and the further
debug session revealed that it was caused by the order of the USB
audio interface. In the earlier code, we always set up the USB
interface at first before other setups, but it was changed to be done
at the last for UAC2/3, which is more standard way, while keeping the
old way for UAC1. OTOH, this device seems requiring the setup of the
interface at first just like UAC1.
This patch works around the regression by applying the interface setup
specifically for the WALKMAN at the beginning of the endpoint setup
procedure. This change is written straightforwardly to be easily
backported in old kernels. A further cleanup to move the workaround
into a generic quirk section will follow in a later patch.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214105
Link: https://lore.kernel.org/r/20210824054700.8236-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This adds a second mixer control to Digidesign Mbox
to select between Analog/SPDIF capture.
Users will note that selecting the SPDIF input source
automatically switches the clock mode to sync to SPDIF,
which is a feature of the hardware.
(You can change the clock source back to internal if you want
to capture from spdif but not sync to its clock although this mode
is probably not recommended).
Unfortunately, starting the stream resets both modes
to Internally clocked and Analog input mode.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Tested-by: Damien Zammit <damien@zamaudio.com>
Link: https://lore.kernel.org/r/20210813113402.11849-1-damien@zamaudio.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Don't populate array names_to_check on the stack but instead it
static. Makes the object code smaller by 56 bytes. Also clean
up checkpatch warning by adding extra const for names_to_check
and pointer s.
Before:
text data bss dec hex filename
103512 34380 0 137892 21aa4 ./sound/usb/mixer.o
After:
text data bss dec hex filename
103264 34572 0 137836 21a6c ./sound/usb/mixer.o
(gcc version 10.2.0)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210803122839.7143-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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