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: 307cc9baac ("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>
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: 307cc9baac ("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>
This is another attempt for the reduction of the latency at the start
of a USB audio playback stream. The first attempt in the commit
9ce650a75a caused an unexpected regression (a deadlock with pipewire
usage) and was later reverted by the commit 4b820e167b. The devils
are always living in details, of course; the cause of the deadlock was
the call of snd_pcm_period_elapsed() inside prepare_playback_urb()
callback. In the original code, this callback is never called from
the stream lock context as it's driven solely from the URB complete
callback. Along with the movement of the URB submission into the
trigger START, this prepare call may be also executed in the stream
lock context, hence it deadlocked with the another lock in
snd_pcm_period_elapsed(). (Note that this happens only conditionally
with a small period size that matches with the URB buffer length,
which was a reason I overlooked during my tests. Also, the problem
wasn't seen in the capture stream because the capture stream handles
the period-elapsed only at retire callback that isn't executed at the
trigger.)
If it were only about avoiding the deadlock, it'd be possible to use
snd_pcm_period_elapsed_under_stream_lock() as a solution. However, in
general, the period elapsed notification must be sent after the actual
stream start, and replacing the call wouldn't satisfy the pattern.
A better option is to delay the notification after the stream start
procedure finished, instead. In the case of USB framework, one of the
fitting place would be the complete callback of the first URB.
So, as a workaround of the deadlock and the order fixes above, in
addition to the re-applying the changes in the commit 9ce650a75a,
this patch introduces a new flag indicating the delayed period-elapsed
handling and sets it under the possible deadlock condition
(i.e. prepare callback being called before subs->running is set).
Once when the flag is set, the period-elapsed call is handled at a
later URB complete call instead.
As a reference for the original motivation for the low-latency change,
I cite here again:
| USB-audio driver behaves a bit strangely for the playback stream --
| namely, it starts sending silent packets at PCM prepare state while
| the actual data is submitted at first when the trigger START is
| kicked off. This is a workaround for the behavior where URBs are
| processed too quickly at the beginning. That is, if we start
| submitting URBs at trigger START, the first few URBs will be
| immediately completed, and this would result in the immediate
| period-elapsed calls right after the start, which may confuse
| applications.
|
| OTOH, submitting the data after silent URBs would, of course, result
| in a certain delay of the actual data processing, and this is rather
| more serious problem on modern systems, in practice.
|
| This patch tries to revert the workaround and lets the URB
| submission starting at PCM trigger for the playback again. As far
| as I've tested with various backends (native ALSA, PA, JACK, PW), I
| haven't seen any problems (famous last words :)
|
| Note that the capture stream handling needs no such workaround,
| since the capture is driven per received URB.
Link: https://lore.kernel.org/r/4e71531f-4535-fd46-040e-506a3c256bbd@marcan.st
Link: https://lore.kernel.org/r/s5hbl7li0fe.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20210707112447.27485-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9ce650a75a.
This commit causes watchdog lockups on my machine, and while I have no
idea what the cause is, it bisected right to this commit, and reverting
the change promptly fixes it.
At least occasionally one of the watchdog call traces was
Call Trace:
_raw_spin_lock_irqsave+0x35/0x40
snd_pcm_period_elapsed+0x1b/0xa0 [snd_pcm]
snd_usb_endpoint_start+0x1a0/0x3c0 [snd_usb_audio]
start_endpoints+0x23/0x90 [snd_usb_audio]
snd_usb_substream_playback_trigger+0x7b/0x1a0 [snd_usb_audio]
snd_pcm_common_ioctl+0x1c44/0x2360 [snd_pcm]
snd_pcm_ioctl+0x2e/0x40 [snd_pcm]
__se_sys_ioctl+0x72/0xc0
do_syscall_64+0x4c/0xa0
entry_SYSCALL_64_after_hwframe+0x44/0xae
so presumably it's a locking error on that substream spinlock that
snd_pcm_period_elapsed() takes. But at this point I just want to have a
working system so that I can continue the merge window work tomorrow.
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
USB-audio driver behaves a bit strangely for the playback stream --
namely, it starts sending silent packets at PCM prepare state while
the actual data is submitted at first when the trigger START is kicked
off. This is a workaround for the behavior where URBs are processed
too quickly at the beginning. That is, if we start submitting URBs at
trigger START, the first few URBs will be immediately completed, and
this would result in the immediate period-elapsed calls right after
the start, which may confuse applications.
OTOH, submitting the data after silent URBs would, of course, result
in a certain delay of the actual data processing, and this is rather
more serious problem on modern systems, in practice.
This patch tries to revert the workaround and lets the URB submission
starting at PCM trigger for the playback again. As far as I've tested
with various backends (native ALSA, PA, JACK, PW), I haven't seen any
problems (famous last words :)
Note that the capture stream handling needs no such workaround, since
the capture is driven per received URB.
Link: https://lore.kernel.org/r/20210601162457.4877-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM delay accounting in USB-audio driver is a bit complex to
follow, and this is an attempt to improve the readability and provide
some potential fix.
Basically, the PCM position delay is calculated from two factors: the
in-flight data on URBs and the USB frame counter. For the playback
stream, we advance the hwptr already at submitting URBs. Those
"in-flight" data amount is now tracked, and this is used as the base
value for the PCM delay correction. The in-flight data is decreased
again at URB completion in return. For the capture stream, OTOH,
there is no in-flight data, hence the delay base is zero.
The USB frame counter is used in addition for correcting the current
position. The reference frame counter is updated at each submission
and receiving time, and the difference from the current counter value
is taken into account.
In this patch, each in-flight data bytes is recorded in the new
snd_usb_ctx.queued field, and the total in-flight amount is tracked in
snd_usb_substream.inflight_bytes field, as the replacement of
last_delay field.
Note that updating the hwptr after URB completion doesn't work for
PulseAudio who tries to scratch the buffer on the fly; USB-audio is
basically a double-buffer implementation, hence the scratching the
buffer can't work for the already submitted data. So we always update
hwptr beforehand. It's not ideal, but the delay account should give
enough correctness.
Link: https://lore.kernel.org/r/20210601162457.4877-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a bunch of lines calculating the buffer size in bytes at
each time. Keep the value in subs->buffer_bytes and use it
consistently for the code simplicity.
Link: https://lore.kernel.org/r/20210601162457.4877-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the hw constraints for implicit feedback streams
via commit e4ea77f8e5 ("ALSA: usb-audio: Always apply the hw
constraints for implicit fb sync") added the check of the matching
endpoints and whether those EPs are already opened. This is needed
and correct, per se, even for the normal streams without the implicit
feedback, as the endpoint setup is exclusive.
However, it's reported that there seem applications that behave in
unexpected ways to update the hw_params without clearing the previous
setup via hw_free, and those hit a problem now: then hw_params is
called with still the previous EP setup kept, hence it's restricted
with the previous own setup. Although the obvious fix is to call
snd_pcm_hw_free() API in the application side, it's a kind of
unwelcome change.
This patch tries to ease the situation: in the endpoint check, we add
a couple of more conditions and now skip the endpoint that is being
used only by the stream in question itself. That is, in addition to
the presence check of ep (ep->cur_audiofmt is non-NULL), when the
following conditions are met, we skip such an ep:
- ep->opened == 1, and
- ep->cur_audiofmt == subs->cur_audiofmt.
subs->cur_audiofmt is non-NULL only if it's a re-setup of hw_params,
and ep->cur_audiofmt points to the currently set up parameters. So if
those match, it must be this stream itself.
Fixes: e4ea77f8e5 ("ALSA: usb-audio: Always apply the hw constraints for implicit fb sync")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211941
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210228080138.9936-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the later patch, we're going to issue the PCM sync_stop calls at
disconnection. But currently the USB-audio driver can't handle it
because it has a check of shutdown flag for stopping the URBs. This
is basically superfluous (the stopping URBs are safe at disconnection
state), so let's drop the check.
Fixes: dc5eafe778 ("ALSA: usb-audio: Support PCM sync_stop")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210206203052.15606-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit f274baa49b ("ALSA: usb-audio: Allow non-vmalloc buffer
for PCM buffers") introduced the mode to allocate coherent pages for
PCM buffers, and it used bus->controller device as its DMA device.
It turned out, however, that bus->sysdev is a more appropriate device
to be used for DMA mapping in HCD code.
This patch corrects the device reference accordingly.
Note that, on most platforms, both point to the very same device,
hence this patch doesn't change anything practically. But on
platforms like xhcd-plat hcd, the change becomes effective.
Fixes: f274baa49b ("ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210205144559.29555-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent refactoring, it's been reported that some USB-audio
devices (typically webcams) are no longer detected properly by
PulseAudio. The debug session revealed that it's failing at probing
by PA to try the sample rate 44.1kHz while the device has discrete
sample rates other than 44.1kHz. But the puzzle was that arecord
works as is, and some other devices with the discrete rates work,
either.
After all, this turned out to be the lack of the dependencies in a few
hw constraint rules: snd_pcm_hw_rule_add() has the (variable)
arguments specifying the dependent parameters, and some functions
didn't set the target parameter itself as the dependencies. This
resulted in an invalid parameter that could be generated only in a
certain call pattern. This bug itself has been present in the code,
but it didn't trigger errors just because the rules were casually
avoiding such a corner case. After the recent refactoring and
cleanup, however, the hw constraints work "as expected", and the
problem surfaced now.
For fixing the problem above, this patch adds the missing dependent
parameters to each snd_pcm_hw_rule() call.
Fixes: bc4e94aa8e ("ALSA: usb-audio: Handle discrete rates properly in hw constraints")
BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014
Link: https://lore.kernel.org/r/20210120204554.30177-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the commit 5a6c3e11c9 ("ALSA: usb-audio: Add hw constraint for
implicit fb sync"), we apply the hw constraints for the implicit
feedback sync to make the secondary open aligned with the already
opened stream setup. This change assumed that the secondary open is
performed after the first stream has been already set up, and adds the
hw constraints to sync with the first stream's parameters only when
the EP setup for the first stream was confirmed at the open time.
However, most of applications handling the full-duplex operations do
open both playback and capture streams at first, then set up both
streams. This results in skipping the additional hw constraints since
the counter-part stream hasn't been set up yet at the open of the
second stream, and it eventually leads to "incompatible EP" error in
the end.
This patch corrects the behavior by always applying the hw constraints
for the implicit fb sync. The hw constraint rules are defined so that
they check the sync EP dynamically at each invocation, instead. This
covers the concurrent stream setups better and lets the hw refine
calls resolving to the right configuration.
Also this patch corrects a minor error that has existed in the debug
print that isn't built as default.
Fixes: 5a6c3e11c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code dealing with the implicit feedback mode grew recently, and
it's becoming messy. As we receive more and more devices that need
the similar handling, it's better to be processed through a table
instead of the open code.
This patch moves the code that is relevant with parsing the implicit
feedback mode and some helpers into another file, implicit.c. The
detection and the setup of the implicit feedback sync EPs are
rewritten to use the ID/class matching table instead.
There should be no functional changes.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-38-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture stream of BOSS GT-001 seems always requiring to be tied
with the playback stream. OTOH, the playback stream of this device
doesn't seem working in the implicit fb mode, per se, since the
playback must be running before the capture stream.
This patch tries to address the points above:
- Avoid the implicit fb mode for the playback
- Set up a fake sync EP for the capture stream with the hard-coded
playback stream using the implicit fb mode
Reported-by: Keith Milner <kamilner@superlative.org>
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-37-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two places calculating the next packet size for the playback
stream in the exactly same way. Provide the single helper for this
purpose and use it from both places gracefully.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-32-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fields like interface and alt_idx in snd_usb_substream are mostly
useless now as they can be referred via either cur_audiofmt or
data_endpoint assigned to the substream. Drop those, and also assure
the concurrency about the access of cur_audiofmt field.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-31-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The audioformat is referred in many places but most of usages are
read-only. Let's add const prefix in the possible places.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-28-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an intensive surgery for the endpoint and stream management
for achieving more robust and clean code.
The goals of this patch are:
- More clear endpoint resource changes
- The interface altsetting control in a single place
Below are brief description of the whole changes.
First off, most of the endpoint operations are moved into endpoint.c,
so that the snd_usb_endpoint object is only referred in other places.
The endpoint object is acquired and released via the new functions
snd_usb_endpoint_open() and snd_usb_endpoint_close() that are called
at PCM hw_params and hw_free callbacks, respectively. Those are
ref-counted and EPs can manage the multiple opens.
The open callback receives the audioformat and hw_params arguments,
and those are used for initializing the EP parameters; especially the
endpoint, interface and altset numbers are read from there, as well as
the PCM parameters like the format, rate and channels. Those are
stored in snd_usb_endpoint object. If it's the secondary open, the
function checks whether the given parameters are compatible with the
already opened EP setup, too.
The coupling with a sync EP (including an implicit feedback sync) is
done by the sole snd_usb_endpoint_set_sync() call.
The configuration of each endpoint is done in a single shot via
snd_usb_endpoint_configure() call. This is the place where most of
PCM configurations are done. A few flags and special handling in the
snd_usb_substream are dropped along with this change.
A significant difference wrt the configuration from the previous code
is the order of USB host interface setups. Now the interface is
always disabled at beginning and (re-)enabled at the last step of
snd_usb_endpoint_configure(), in order to be compliant with the
standard UAC2/3. For UAC1, the interface is set before the parameter
setups since there seem devices that require it (e.g. Yamaha THR10),
just like how it was done in the previous driver code.
The start/stop are almost same as before, also single-shots. The URB
callbacks need to be set via snd_usb_endpoint_set_callback() like the
previous code at the trigger phase, too.
Finally, the flag for the re-setup is set at the device suspend
through the full EP list, instead of PCM trigger. This catches the
overlooked cases where the PCM hasn't been running yet but the device
needs the full setup after resume.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-26-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function to evaluate the match of the parameters with an EP
assumes only the discrete rate tables and doesn't handle the
continuous rates properly.
This patch fixes match_endpoint_audioformats() to handle the
continuous rates. Also the almost useless debug prints there are
dropped.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-25-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 92adc96f8e ("ALSA: usb-audio: set the interface format
after resume on Dell WD19") introduced the workaround for the broken
setup after the resume specifically on a Dell dock model. However,
the full setup should have been performed after the resume on all
devices, as we can't guarantee the same state. So this patch removes
the conditional check and applies the workaround always.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-24-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The prepare_data_urb and retire_data_urb fields of the endpoint object
are set dynamically at PCM trigger start/stop. Those are evaluated in
the endpoint handler, but there can be a race, especially if two
different PCM substreams are handling the same endpoint for the
implicit feedback case. Also, the data_subs field of the endpoint is
set and accessed dynamically, too, which has the same risk.
As a slight improvement for the concurrency, this patch introduces the
function to set the callbacks and the data in a shot with the memory
barrier. In the reader side, it's also fetched with the memory
barrier.
There is still a room of race if prepare and retire callbacks are set
during executing the URB completion. But such an inconsistency may
happen only for the implicit fb source, i.e. it's only about the
capture stream. And luckily, the capture stream never sets the
prepare callback, hence the problem doesn't happen practically.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-23-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
start_endpoints() may leave the data endpoint running if an error
happens at starting the sync endpoint. We should stop both streams
properly, instead.
While we're at it, move the debug prints into the endpoint.c that is a
more suitable place.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-22-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A preliminary change for the later big changes. This is a minor code
refactoring to drop the unnecessary arguments that can be retrieved in
a different way.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-21-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A preliminary change for the later big changes. This is a minor code
refactoring to drop the unnecessary arguments that can be retrieved in
a different way.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-20-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A preliminary patch for the later big change. Just a minor code
refactoring.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-19-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a helper function to retrieve the usb_host_interface object from
the given interface and altsetting number pair, which is a commonly
used procedure in the driver code.
No functional changes, just minor code refactoring.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-17-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This behavior turned out to be invalid from the USB spec POV and
shouldn't be applied. As it's an optional flag that is set only via
an card control element that must be hardly used, let's drop it
again.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-16-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently snd_usb_endpoint objects are created at first when the
substream is opened and tries to assign the endpoints corresponding to
the matching audioformat. But since basically the all endpoints have
been already parsed and the information have been obtained, we may
create the endpoint objects statically at the init phase. It's easier
to manage for the implicit fb case, for example.
This patch changes the endpoint object management and lets the parser
to create the all endpoint objects.
This change shouldn't bring any functional changes.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-15-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The implicit feedback mode initializes both the main data stream and
the sync data stream. When a sync stream was already opened, this
would result in the doubly initialization and might screw up things.
Add the check of already opened sync streams and skip the unnecessary
initialization.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-14-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The file debug.h contains a simple macro for debug prints, and it's
used only in two places, the format parser and the hw_params rules.
The former actually should print a more informative message instead,
so the only users are the hw_parmas rules.
This patch moves the contents of debug.h into the hw_params rules
local code and remove the unneeded includes. Also, the debug print in
the format parser is replaced with the information print with more
useful information, and the raw printk() call is replaced with
pr_debug().
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-13-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several hw_params functions narrows the interval via min/max rule in
the very similar way, so factor out those into a helper function and
use commonly.
No functional changes, just minor code refactoring.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, there is no check at the stream open time whether
the endpoint is being already used by others. In the normal
operations, this shouldn't happen, but in the case of the implicit
feedback mode, it's a common problem with the full duplex operation,
because the capture stream is always opened by the playback stream as
an implicit sync source.
Although we recently introduced the check of such a conflict of
parameters at the PCM hw_params time, it doesn't give any hint at the
hw_params itself and just gives the error. This isn't quite
comfortable, and it caused problems on many applications.
This patch attempts to make the parameter handling easier by
introducing the strict hw constraint matching with the counterpart
stream that is being used. That said, when an implicit feedback
playback stream is running before a capture stream is opened, the
capture stream carries the PCM hw-constraint to allow only the same
sample rate, format, periods and period frames as the running playback
stream. If not opened or there is no conflict of endpoints, the
behavior remains as same as before.
Note that this kind of "weak link" should work for most cases, but
this is no concrete solution; e.g. if an application changes the hw
params multiple times while another stream is opened, this would lead
to inconsistencies.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary work for the upcoming hw-constraint change for
the implicit feedback mode.
Currently snd_usb_autoresume() is called at the end of
setup_hwinfo(). It's a bit confusing; because of this implicit
refcount usage, the caller side needs to call snd_usb_autosuspend()
later in the error path although it's not seen inside the function.
Instead, it's clearer to call both snd_usb_autoresume() and suspend()
in the very same function.
It's only refactoring and no functional changes.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of parsing and evaluating the sync endpoint and the implicit
feedback mode at each time the audio stream is opened, let's parse it
once at the probe time, as the all needed information can be obtained
statically from the descriptor or from the quirk.
This patch extends audioformat struct to record the sync endpoint,
interface and altsetting as well as the implicit feedback flag, which
are filled at parsing the streams. Then, set_sync_endpoint() is much
simplified just to follow the already parsed data.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few rooms for improvements wrt the debug prints:
- The EP debug print is shown only at starting, not at stopping
- The EP debug print contains useless object addresses
- Some helpers show the urb and the EP object addresses, too
This patch addresses those shortcomings.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sync EP setup isn't cleared at stopping the stream but expected to
be cleared at the next stream start. This may leave the sync link
setup stale and can spoof wrongly when full duplex streams were
running in the implicit fb sync. Let's initialize them properly at
start and end of the stream.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that many UAC2 devices are with the implicit feedback, but
they couldn't be probed properly because the assumption the driver
takes currently isn't applied: they have the single endpoint for both
data and implicit-fb streams, while we checked only the classical sync
endpoints assigned to the next altsetting in the same interface.
This patch extends the search to match with those typical cases where
the implicit fb stream is found in the next interface number.
While we're at it, slightly refactor the code, not returning 0/-ERROR
but use the standard bool to success/failur, which is more intuitive
in this particular case.
Reported-by: Dylan Robinson <dylan_robinson@motu.com>
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM trigger callback is atomic, hence we must not call a function
like usb_set_interface() there. Calling it from there would lead to a
kernel Oops.
Fix it by moving the usb_set_interface() call to set_sync_endpoint().
Also, apply the snd_usb_set_interface_quirk() for consistency, too.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, when the device provides the discrete sample rate
tables with unusual sample rates, the driver tries to gather the whole
values from the audioformat entries and create a hw-constraint rule to
restrict with this single rate list. This is rather inefficient and
may overlook the rates that are associated only with the certain
audioformat entries.
This patch improves the hw constraint setup by rewriting the existing
hw_rule_rate(). The discrete sample rates (identified by rate_table
and nr_rates of format entry) are checked in the existing
hw_rule_rate() instead of extra rules; in the case of discrete rates,
the function compares with each rate table entry and calculates the
min/max values from there. For the contiguous rates, the behavior
doesn't change.
Along with it, snd_usb_pcm_check_knot() and snb_usb_substream
rate_list field become superfluous, thus those are dropped.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes audio distortion on playback for the Allen&Heath
Qu-16.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201104115717.GA19046@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes audio distortion on playback for the Yamaha MODX.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Tested-by: Frank Slotta <frank.slotta@posteo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201104120705.GA19126@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This patch extends support for DJM-250MK2 and allows recording.
However, DVS is not possible yet (see the comment in code).
Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200825153113.6352-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.
So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
