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: fe773b8711 ("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>
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 5e35dc0338 and subsequent
1a15718b41 were meant to clear up noise.
Unfortunately, noise persisted for those using higher sample rates and
this was only solved by commit d2e8f64125
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>
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>
This is another attempt to improve further the handling of playback
stream in the low latency mode. The latest workaround in commit
4267c5a8f3 ("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: 4267c5a8f3 ("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>
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>
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>
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>
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>
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>
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>
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>
snd_usb_find_clock_source and snd_usb_find_clock_selector are helper
macros that look at an entity id and validate that this entity id is
in fact a clock source or a clock selector. The present comments
inside __uac_clock_find_source give the reader the impression we're
looking for an entity id.
We're looking for an entity id indeed, the clock source, but since
__uac_clock_find_source is recursive, we're also looking *at* the
entity ids, in the search for the one clock source.
Fix the comment so we don't give readers a wrong idea.
Signed-off-by: Geraldo Nascimento <geraldogabriel@gmail.com>
Link: https://lore.kernel.org/r/YU6Kj05oOqRmhJDf@geday
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As noted in the "Deprecated Interfaces, Language Features, Attributes,
and Conventions" documentation [1], size calculations (especially
multiplication) should not be performed in memory allocator (or similar)
function arguments due to the risk of them overflowing. This could lead
to values wrapping around and a smaller allocation being made than the
caller was expecting. Using those allocations could lead to linear
overflows of heap memory and other misbehaviors.
In this case this is not actually dynamic size: all the operands
involved in the calculation are constant values. However it is better to
refactor this anyway, just to keep the open-coded math idiom out of
code.
So, use the struct_size() helper to do the arithmetic instead of the
argument "size + size * count" in the kzalloc() function.
Also, take the opportunity to refactor the declaration variables to make
it more easy to read.
[1] https://www.kernel.org/doc/html/latest/process/deprecated.html#open-coded-arithmetic-in-allocator-arguments
Signed-off-by: Len Baker <len.baker@gmx.com>
Link: https://lore.kernel.org/r/20210919133727.44694-1-len.baker@gmx.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
The recent quirk for WALKMAN (commit 7af5a14371: "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>
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: bf6313a0ff ("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>
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>
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>
This patch adds a new module option, quirk_flags, for allowing user to
try some additional device-specific quirk behavior more easily.
When this option is set to non-zero, it overrides the quirk_flags, and
the specific workaround is applied.
Link: https://lore.kernel.org/r/20210729074404.19728-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer code has a flag ignore_ctl_error for ignoring the errors
returned from the device wrt mixer accesses, and this is set from the
entries in mixer_maps.c, as well as ignore_ctl_error module option.
Those can be well integrated into the new quirk_flags field, too.
Link: https://lore.kernel.org/r/20210729074404.19728-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-suspend suppression workaround for Lenovo machines are
handled in quirks-table.h. Now it's more easier to handle with
quirk_flags.
Link: https://lore.kernel.org/r/20210729074404.19728-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The rate validation at the device probe is applied only to the
specific devices (currently only for MOTU devices), and this check can
be moved to quirk_flags gracefully, too.
Link: https://lore.kernel.org/r/20210729074404.19728-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We apply some delay for the control messages on certain devices as a
workaround, and this can be moved into the quirk_flags as well.
Currently there are three different delay periods (1ms, 5ms and 20ms),
so three different quirk bits are assigned for them.
Link: https://lore.kernel.org/r/20210729073855.19043-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is another quirk for the transfer, and that's currently specific
to Zoom R16/24, handled in create_standard_audio_quirk(). Let's move
this also to the new quirk_flags.
Link: https://lore.kernel.org/r/20210729073855.19043-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The txfr_quirk field was meant for aligning the transfer, and it's set
for certain devices in quirks-table.h. Now we can move that stuff
also to the new quirk_flags gracefully, and reduce the quirks-table.h
entries (that are exposed to module device table).
As the quirks-table.h entries are also with the name string override,
provide the corresponding entries to the usb_audio_names[] table,
too.
Link: https://lore.kernel.org/r/20210729073855.19043-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices that can have media-controller API entries are currently
specified via tables in quirks-table.h, as a part of descriptor
override. This can fit better to the new quirk_flags, as we just need
a matching with the given ID and create the MC entries accordingly.
Link: https://lore.kernel.org/r/20210729073855.19043-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As more and more device-specific workarounds came up and gathered in
various places, it becomes harder to manage. Now it's time to clean
up and collect workarounds more consistently and make them more easily
applicable.
This patch is the first step for that: a new field quirk_flags is
introduced in snd_usb_audio struct to contain the bit flags for
various device-specific quirks. Those are separate one from the
quirks in quirks-table.h; the quirks-table.h entries are for more
intrusive stuff that needs the descriptor override, while the new
quirk_flags is for easier ones that are tied with the vendor:product
IDs.
In this patch, as the first example, we convert the list of devices
and vendors to ignore GET_SAMPLE_RATE, formerly defined in
snb_usb_get_sample_rate_quirk().
Link: https://lore.kernel.org/r/20210729073855.19043-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the resume on Lenovo machines seems causing a
regression on others. It's because the change always triggers the
connector selection no matter which widget node type is.
This patch addresses the regression by setting the resume callback
selectively only for the connector widget.
Fixes: 44609fc01f ("ALSA: usb-audio: Check connector value on resume")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213897
Link: https://lore.kernel.org/r/20210729185126.24432-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apparently JBL Quantum 600 has multiple hardware revisions. Apply
registration quirk to another device id as well.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210727093326.1153366-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The change to restore the autosuspend from the disabled state uses a
wrong check: namely, it should have been the exact comparison of the
quirk_type instead of the bitwise and (&). Otherwise it matches
wrongly with the other quirk types.
Although re-enabling the autosuspend for the already enabled device
shouldn't matter much, it's better to fix the unbalanced call.
Fixes: 9799110825 ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hr1flh9ov.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The following scenario describes an echo test for
Samsung USBC Headset (AKG) with VID/PID (0x04e8/0xa051).
We first start a capture stream(USB IN transfer) in 96Khz/24bit/1ch mode.
In clock find source function, we get value 0x2 for clock selector
and 0x1 for clock source.
Kernel-4.14 behavior
Since clock source is valid so clock selector was not set again.
We pass through this function and start a playback stream(USB OUT transfer)
in 48Khz/32bit/2ch mode. This time we get value 0x1 for clock selector
and 0x1 for clock source. Finally clock id with this setting is 0x9.
Kernel-5.10 behavior
Clock selector was always set one more time even it is valid.
When we start a playback stream, we will get 0x2 for clock selector
and 0x1 for clock source. In this case clock id becomes 0xA.
This is an incorrect clock source setting and results in severe noises.
We see wrong data rate in USB IN transfer.
(From 288 bytes/ms becomes 144 bytes/ms) It should keep in 288 bytes/ms.
This earphone works fine on older kernel version load because
this is a newly-added behavior.
Fixes: d2e8f64125 ("ALSA: usb-audio: Explicitly set up the clock selector")
Signed-off-by: chihhao.chen <chihhao.chen@mediatek.com>
Link: https://lore.kernel.org/r/1627100621-19225-1-git-send-email-chihhao.chen@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The values of the line output controls can change when the SW/HW
switches are set to HW, and also when speaker switching is enabled.
These notifications were sent with a mask of only
SNDRV_CTL_EVENT_MASK_INFO. Change the notifications to set the
SNDRV_CTL_EVENT_MASK_VALUE mask bit as well.
When the mute control is updated, the notification was sent with a
mask of SNDRV_CTL_EVENT_MASK_INFO. Change the mask to the correct
value of SNDRV_CTL_EVENT_MASK_VALUE.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/8192e15ba62fa4bc90425c005f265c0de530be20.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the hardware mute button is pressed, private->vol_updated is set
so that the mute status is invalidated. As the channel mute values may
be affected by the global mute value, update scarlett2_mute_ctl_get()
to call scarlett2_update_volumes() if private->vol_updated is set.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/aa18ddbf8d8bd7f31832ab1b6b6057c00b931202.1626959758.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These devices has two interfaces, but only the second interface
contains the capture endpoint, thus quirk is required to delay the
registration until the second interface appears.
Tested-by: Jakub Fišer <jakub@ufiseru.cz>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210721235605.53741-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we've added a new usb_mixer element type, USB_MIXER_BESPOKEN,
but it wasn't added in the table in snd_usb_mixer_dump_cval(). This
is no big problem since each bespoken type should have its own dump
method, but it still isn't disallowed to use the standard one, so we
should cover it as well. Along with it, define the table with the
explicit array initializer for avoiding other pitfalls.
Fixes: 785b6f29a7 ("ALSA: usb-audio: scarlett2: Fix wrong resume call")
Reported-by: Pavel Machek <pavel@denx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210714084836.1977-1-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>
Clang warns:
sound/usb/mixer_scarlett_gen2.c:1189:32: warning: expression result
unused [-Wunused-value]
for (i = 0; i < count; i++, (u16 *)buf++)
^ ~~~~~
1 warning generated.
It appears the intention was to cast the void pointer to a u16 pointer
so that the data could be iterated through like an array of u16 values.
However, the cast happens after the increment because a cast is an
rvalue, whereas the post-increment operator only works on lvalues, so
the loop does not iterate as expected. This is not a bug in practice
because count is not greater than one at the moment but this could
change in the future so this should be fixed.
Replace the cast with a temporary variable of the proper type, which is
less error prone and fixes the iteration. Do the same thing for the
'u8 *' below this if block.
Fixes: ac34df733d ("ALSA: usb-audio: scarlett2: Update get_config to do endian conversion")
Link: https://github.com/ClangBuiltLinux/linux/issues/1408
Acked-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://lore.kernel.org/r/20210627051202.1888250-1-nathan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mixer control put callbacks should return 1 if the value is changed.
Fix the mute, air, phantom, direct monitor, speaker switch, talkback,
and MSD controls accordingly.
Fix scarlett2_speaker_switch_enable() to not ignore the return value
of scarlett2_sw_hw_change().
Reported-by: Aaron Wolf <aaron@wolftune.com>
Tested-by: Aaron Wolf <aaron@wolftune.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/76643f7ac81aef93351122d07881e30d51dcb1b9.1624798436.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 Gen 3 has 4 inputs with a pad control, not 2. Update
s18i8_gen3_info.pad_input_count.
Reported-by: Aaron Wolf <aaron@wolftune.com>
Tested-by: Aaron Wolf <aaron@wolftune.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/29a6ce412a42373daab7c96c395560461fcf08c6.1624798436.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For configuration items with a size of 16, scarlett2_usb_get_config()
was filling *buf with little-endian data. Update it to convert to CPU
endian. This function is not currently used so affects nothing yet;
will be used by the upcoming talkback feature.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/cbc8b6eedd859dd27086ab4126d724a86dd50bcb.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 and 18i20 Gen 3 support "speaker switching". Add a Speaker
Switch control which can be set to Off/Main/Alt.
When speaker switching is enabled or disabled, the interface may
change the state of the Analog Outputs 3 and 4 routing and the global
mute button, so use a flag private->speaker_switching_switched to note
that those should be checked when the next "monitor other"
notification is received.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/874193a534cd0aeb6f2e108ae761cadd2dc25ad2.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enabling/disabling speaker switching will update the mux
configuration. To prepare for this, add a private->mux_updated flag
and update the scarlett2_mux_src_enum_ctl_get() callback to check it.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/5ce3bb9fe4006b550d18c783c5ff640fe0bfbfcb.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Save the struct snd_kcontrol pointers for the sw_hw and mux controls.
This is in preparation for speaker switching support which needs to be
able to update those controls.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/269d89181bf29dbea80ba6f8cfff84fb23b77f86.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split part of scarlett2_sw_hw_enum_ctl_put() out into
scarlett2_sw_hw_change() so that the code which actually makes the
change is available in its own function. This will be used by the
speaker switching support which needs to set the SW/HW switch to HW
when speaker switching is enabled.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/f2cf91841ba067b490e7709bc4b14f4532b4ddd5.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Solo and 2i2 devices don't have a mixer but they do have a "direct
monitor" switch. Add support for getting and setting the state of this
switch.
Co-developed-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/61d23dc4feb3b046d870ad7203e66ff2bd1d278c.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some inputs on Gen 3 models support software-selectable phantom power.
Add support for getting and setting the state of those switches and
the "Phantom Power Persistence" switch.
Co-developed-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/5837ce8a8c686560fc8f40b4204dd2a10721869b.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some inputs on Gen 3 models have an "air" feature which can be enabled
from the driver or (model-dependent) from the front panel. Add support
for getting and setting the state of those switches.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/44d448a4150b9c068754759c9fdd2bfe21484487.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add initial support for the Focusrite Scarlett Solo and 2i2 devices:
- They have no mixer
- They don't support reporting sync status or levels
- The configuration space is laid out differently to the other models
- There is no level (line/inst) switch on input 1 of the Solo
Co-developed-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/190b90f6f1f8f8d4dfb5f0a7761ff8ae5c40fdde.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move scarlett2_usb_get() and scarlett2_usb_get_config() above the
functions relating to updating the configuration so that
scarlett2_usb_set_config() can call scarlett2_usb_get() in a
subsequent patch.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/1549f8e44548be679119f0b1462f888f4a03812d.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models allow the level and pad settings to be controlled from the
front-panel of the device. For these, the device will send an
"input-other" notification to prompt the driver to re-read the status
of those settings.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/06289a7697455e96b7dbdfd2d384d4b20f8df6e0.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current way of the scarlett2 mixer code managing the
usb_mixer_elem_info object is wrong in two ways: it passes its
internal index to the head.id field, and the val_type field is
uninitialized. This ended up with the wrong execution at the resume
because a bogus unit id is passed wrongly. Also, in the later code
extensions, we'll have more mixer elements, and passing the index will
overflow the unit id size (of 256).
This patch corrects those issues. It introduces a new value type,
USB_MIXER_BESPOKEN, which indicates a non-standard mixer element, and
use this type for all scarlett2 mixer elements, as well as
initializing the fixed unit id 0 for avoiding the overflow.
Tested-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/49721219f45b7e175e729b0d9d9c142fd8f4342a.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The scarlett2_ports struct contains both generic (hardware IDs and
descriptions) and model-specific (port count) data. Remove the generic
data from the scarlett2_device_info struct so it is not repeated for
every model.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/7a9e57e4e55a482390c692a9e60731d72b664a15.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Gen 3 devices do not put all of the mux entries for the same port
types together in order in the "set mux" message data. To prepare for
this, replace the struct scarlett2_ports num[] array and the
assignment_order[] array with mux_assignment[], a list of port types
and ranges that is defined in the struct scarlett2_device_info.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/08e8d784d78262cb57496d28ef1ad7b6213a90ab.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For each analogue output, in addition to the output volume (gain)
control, the hardware also has a mute control. Add ALSA mute controls
for each analogue output.
If the device has the line_out_hw_vol feature, then the mute control
is disabled along with the output volume control when the switch is
set to HW.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/6fad82174b44633e46cfd96332a038de74d544f2.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the USB device ID to the scarlett2_device_info struct so that the
switch statement which finds the appropriate struct can be replaced
with a loop that looks through an array of pointers to those structs.
Suggested-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/474c408c29fb280a611e47e49e59ca2fb9810d27.1624294591.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At extending the available mixer values for 32bit types, we forgot to
add the corresponding entries for the format dump in the proc output.
This may result in OOB access. Here adds the missing entries.
Fixes: bc18e31c30 ("ALSA: usb-audio: Fix parameter block size for UAC2 control requests")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210622090647.14021-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the hard-coded interface number and related constants for the
vendor-specific interface and look them up from the USB endpoint
descriptor.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164652.GA9237@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
scarlett2_usb_set_config() and scarlett2_usb_get_config() were copying
struct scarlett2_config. Use a pointer instead.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164648.GA9231@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix mixer control callbacks to use the correct members of the struct
snd_ctl_elem_value. The use of value.integer and value.enumerated were
swapped in a few places.
Update scarlett2_mux_src_enum_ctl_put() to use min() instead of
clamp() as value.enumerated.item is unsigned.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164647.GA9226@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mixer control put callbacks should return 1 if the value is changed.
Fix the sw_hw, level, pad, and button controls accordingly.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164645.GA9221@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The private->vol_updated flag was being checked outside of the
mutex_lock/unlock() of private->data_mutex leading to the volume data
being fetched twice from the device unnecessarily or old volume data
being returned.
Update scarlett2_*_ctl_get() and include the private->vol_updated flag
check inside the critical region.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164643.GA9216@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add index temporary variable to scarlett2_mixer_ctl_put() for
consistency with the other *_ctl_put() functions.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164641.GA9211@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename struct scarlett2_mixer_data to struct scarlett2_data. A
less-wordy name is better because it is used everywhere, and although
this is a mixer driver, it also controls other vendor-specific
features.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164639.GA9206@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To match the vendor's terminology, change #defines, identifiers, and
comments:
- mute/dim/hardware buttons are now called dim/mute
- mixer status/interrupt is now notify
- vol is now monitor
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164636.GA9199@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The per-model button_count value was used to determine whether
dim/mute controls should be added, but these are present iff
line_out_hw_vol is true. Remove button_count and replace with
SCARLETT2_BUTTON_MAX and a check for line_out_hw_vol true.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164634.GA9193@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just ignore instead of printing an error if the interrupt data is not
the expected length. This check was for development and the condition
has not been observed.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164632.GA9186@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 Gen 2 has 8 PCM Inputs, not 20. Fix the ports entry in
s18i8_gen2_info.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164625.GA9165@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 18i8 Gen 2 S/PDIF outputs are available at 192kHz, unlike
the 18i20 Gen 2. Remove the comment that says otherwise.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20210620164622.GA9155@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This provides support for Denon DN-X1600 hardware mixer.
The device itself supports 44100, 48000 and 96000 (Hz)
sample rates, but switching rates via software is currently not working.
Therefore, this patch hardcodes the sample rate to 48000Hz which
enables all 8 channels to function correctly when the correct
sample rate is selected on the hardware itself.
MIDI also tested and works.
Signed-off-by: Damien Zammit <damien@zamaudio.com>
Tested-by: xalmoxis@gmail.com
Link: https://lore.kernel.org/r/20210610083528.603942-2-damien@zamaudio.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for retrieving the mux configuration from the hardware
when the driver is initialising. Previously the ALSA controls were
initialised to a default hard-coded state instead of being initialised
to match the hardware state.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Suggested-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Tested-by: Markus Schroetter <project.m.schroetter@gmail.com>
Tested-by: Alex Fellows <alex.fellows@gmail.com>
Tested-by: Daniel Sales <daniel.sales.z@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/15b17c60a2bca174bcddcec41c9419b746f21c1d.1623091570.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for reading the mixer volumes from the hardware when the
driver is initialising. Previously these ALSA volume controls were
initialised to zero instead of being initialised to match the hardware
state.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Suggested-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Tested-by: Markus Schroetter <project.m.schroetter@gmail.com>
Tested-by: Alex Fellows <alex.fellows@gmail.com>
Tested-by: Daniel Sales <daniel.sales.z@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/bb33fa9b79efc6f7a0f0e6fb7018cc8d4d59b3ba.1623091570.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 power_state argument of snd_power_wait() is superfluous, receiving
only SNDRV_POWER_STATE_D0. Let's drop it in all callers for
simplicity.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20210523090920.15345-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/mixer_scarlett_gen2.c:2000:5: warning: symbol 'snd_scarlett_gen2_controls_create' was not declared. Should it be static?
Fixes: 265d1a90e4 ("ALSA: usb-audio: scarlett2: Improve driver startup messages")
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20210522180900.GA83915@f59a3af2f1d9
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add separate init function to call the existing controls_create
function so a custom error can be displayed if initialisation fails.
Use info level instead of error for notifications.
Display the VID/PID so device_setup is targeted to the right device.
Display "enabled" message to easily confirm that the driver is loaded.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/b5d140c65f640faf2427e085fbbc0297b32e5fce.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use usb_rcvctrlpipe() not usb_sndctrlpipe() for USB control input in
the Scarlett Gen 2 mixer driver. This fixes the device hang during
initialisation when used with the ehci-pci host driver.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/66a3d05dac325d5b53e4930578e143cef1f50dbe.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The direction of the pipe argument must match the request-type direction
bit or control requests may fail depending on the host-controller-driver
implementation.
Fix the UAC2_CS_CUR request which erroneously used usb_sndctrlpipe().
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: stable@vger.kernel.org # 5.10
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20210521133742.18098-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cur variable indicating the currently selected clock source can be
theoretically used as uninitialized after the recent commit
481f17c418 ("ALSA: usb-audio: Handle error for the current selector
gracefully"). For addressing it, initialize it before use.
Also, one place seems setting 0 to a wrong variable ret, instead of
cur; otherwise it makes little sense. Since the initialization is
done beforehand, we can get rid of this line, too.
Fixes: 481f17c418 ("ALSA: usb-audio: Handle error for the current selector gracefully")
Reported-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/4b261d68-f53f-240d-2d8a-2f88b337849d@canonical.com
Link: https://lore.kernel.org/r/s5hfsyhh97t.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The initialization of MIDI devices that are found on some LINE6
drivers are currently done in a racy way; namely, the MIDI buffer
instance is allocated and initialized in each private_init callback
while the communication with the interface is already started via
line6_init_cap_control() call before that point. This may lead to
Oops in line6_data_received() when a spurious event is received, as
reported by syzkaller.
This patch moves the MIDI initialization to line6_init_cap_control()
as well instead of the too-lately-called private_init for avoiding the
race. Also this reduces slightly more lines, so it's a win-win
change.
Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com
Link: https://lore.kernel.org/r/000000000000a4be9405c28520de@google.com
Link: https://lore.kernel.org/r/20210517132725.GA50495@hyeyoo
Cc: Hyeonggon Yoo <42.hyeyoo@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210518083939.1927-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we bail out when the device returns an error or an invalid
value for the current clock selector value via
uac_clock_selector_get_val(). But it's possible that the device is
really uninitialized and waits for the setup of the proper route at
first.
For handling such a case, this patch lets the driver dealing with the
error or the invalid error more gracefully, choosing the clock source
automatically instead.
Link: https://lore.kernel.org/r/20210518152112.8016-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just does refactoring of the UAC2/3 clock setup code.
There should be no functional changes. The major changes are:
* Provide union objects for pointing both UAC2 and UAC3 objects
* Unify clock source, selector and multiplier helper functions
* Unify __uac_clock_find_source() to deal with both UAC2 and UAC3
equally
Link: https://lore.kernel.org/r/20210518152112.8016-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor code refactoring by merging the superfluous function calls.
The functions were split in the past for covering pre-history USB
driver code, but this is utterly useless.
Link: https://lore.kernel.org/r/20210517131545.27252-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently us428ctls_shmem pages are allocated dynamically upon the
mmap call, but this is quite racy. Since the shared memory itself is
mandatory for the mmap, let's allocate it at the beginning of the card
initialization. Also, fix the initialization of the wait queue, too.
Link: https://lore.kernel.org/r/20210517131545.27252-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM shmem pages are allocated in snd_usx2y_usbpcm_prepare().
Theoretically the prepare callback may be called simultaneously for
both playback and capture, hence this allocation can be racy.
Make sure that the allocation is performed exclusively by extending
the pcm_mutex lock to cover the allocation code, too.
Link: https://lore.kernel.org/r/20210517131545.27252-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Theoretically the initialization functions in usx2y drivers may be
called multiple times as the driver gets initialized via hwpdep
ioctl. Meanwhile, those functions including memory allocations don't
check whether they are called twice, and they forget the old
resources, which would lead to memory leaks.
This patch adds the sanity checks about the doubly initializations to
give kernel WARNING, and returns an error in such a case. Also, each
allocation assures to release the resources at its error path
properly.
Link: https://lore.kernel.org/r/20210517131545.27252-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The initialization os usx2y driver is multi-staged, and the PCM and
other device creations are done after the DSP is loaded and
initialized. Upon the initialization, when an error happens, the
driver tries to call snd_card_free(). But this is dangerous, and in
general, the driver cannot kill itself during its operation.
Hence better to drop the snd_card_free() call from there.
Link: https://lore.kernel.org/r/20210517131545.27252-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usx2y drivers may expose the allocated pages via mmap, but it performs
zero-clear only for the struct size, not aligned with the page size.
This leaves out some uninitialized trailing bytes.
This patch fixes the clearance to cover all memory that are exposed to
user-space.
Link: https://lore.kernel.org/r/20210517131545.27252-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes various trivial coding-style issues in usx2y code,
such as:
* the assginments in if condition
* comparison order with constants
* NULL / zero checks
* unsigned -> unsigned int
* addition of braces in control blocks
* debug print with function names
* move local variables in block into function head
* reduction of too nested indentations
No functional changes.
Link: https://lore.kernel.org/r/20210517131545.27252-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch corrects merely the spaces in the usx2y code, including the
superfluous trailing space in the debug prints and a slight reformat
of some comment lines. Nothing really touches about the code itself.
Link: https://lore.kernel.org/r/20210517131545.27252-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For improving readability, convert camelCase fields, variables and
functions to the plain names with underscore. Also align the macros
to be capital letters.
All done via sed, no functional changes.
Note that you'll still see many coding style issues even after this
patch; the fixes will follow.
Link: https://lore.kernel.org/r/20210517131545.27252-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced MIDI endpoint parser code has an access to the
field without the size validation, hence it might lead to
out-of-bounce access. Add the sanity checks for the descriptor
sizes.
Fixes: eb596e0fd1 ("ALSA: usb-audio: generate midi streaming substream names from jack names")
Link: https://lore.kernel.org/r/20210511090500.2637-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Variable len is set to zero but this value is never read as it is
overwritten with a new value later on, hence it is a redundant
assignment and can be removed.
Cleans up the following clang-analyzer warning:
sound/usb/mixer.c:2713:3: warning: Value stored to 'len' is never read
[clang-analyzer-deadcode.DeadStores].
Reported-by: Abaci Robot <abaci@linux.alibaba.com>
Signed-off-by: Jiapeng Chong <jiapeng.chong@linux.alibaba.com>
Link: https://lore.kernel.org/r/1619519194-57806-1-git-send-email-jiapeng.chong@linux.alibaba.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent endpoint management change for implicit feedback mode added
a clearance of ep->sync_sink (formerly ep->sync_slave) pointer at
snd_usb_endpoint_stop() to assure no leftover for the feedback from
the already stopped capture stream. This turned out to cause a
regression, however, when full-duplex streams were running and only a
capture was stopped. Because of the above clearance of ep->sync_sink
pointer, no more feedback is done, hence the playback will stall.
This patch fixes the ep->sync_sink clearance to be done only after all
endpoints are released, for addressing the regression.
Reported-and-tested-by: Lucas Endres <jaffa225man@gmail.com>
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210426063349.18601-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the examinations and experiments with lots of Roland and BOSS
USB-audio devices, we found out that the recently introduced
full-duplex operations with the implicit feedback mode work fine for
quite a few devices, while the others need only the capture-side quirk
to enforce the full-duplex mode. The recent commit d86f43b17e
("ALSA: usb-audio: Add support for many Roland devices' implicit
feedback quirks") tried to add such quirk entries manually in the
lists, but this turned out to be too many and error-prone, hence it
was reverted again.
This patch is another attempt to cover those missing Roland/BOSS
devices but in a more generic way. It matches the devices with the
vendor ID 0x0582, and checks whether they are with both ASYNC sync
types or ASYNC is only for capture device. In the former case, it's
the device with the implicit feedback mode, and applies accordingly.
In both cases, the capture stream requires always the full-duplex
mode, and we apply the known capture quirk for that, too.
Basically the already existing BOSS device quirk entries become
redundant after this generic matching, so those are removed. Although
the capture_implicit_fb_quirks[] table became empty and superfluous, I
keep it for now, so that people can put a special device easily at any
time later again.
Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212519
Tested-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/20210422120413.457-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit d86f43b17e ("ALSA: usb-audio: Add support for
many Roland devices' feedback quirks").
It turned out that many quirk entries there don't contain the proper
EP values and/or the quirk types, which lead to the broken
operations.
As we're going to cover all Roland/BOSS devices in a more generic way
rather the explicit lists, let's revert the previous additions at
first.
Fixes: d86f43b17e ("ALSA: usb-audio: Add support for many Roland devices' implicit feedback quirks")
Link: https://lore.kernel.org/r/20210422120413.457-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently when the call to usb_urb_ep_type_check fails (returning -EINVAL)
the error return path returns -ENOMEM via the exit label "error". Other
uses of the same error exit label set the err variable to -ENOMEM but this
is not being used. I believe the original intent was for the error exit
path to return the value in err rather than the hard coded -ENOMEM, so
return this rather than the hard coded -ENOMEM.
Addresses-Coverity: ("Unused value")
Fixes: 738d9edcfd ("ALSA: usb-audio: Add sanity checks for invalid EPs")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210420134719.381409-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pioneer devices are supposed to be working with the implicit feedback
mode, but so far the attempt to apply the implicit feedback caused
issues, hence we explicitly skipped the implicit feedback mode for
them. Recently, Geraldo discovered that the device actually works if
you skip the generic matching of the sync EPs for the capture stream.
That is, we should apply the implicit feedback setup for the playback
like other similar devices, while we need to return 1 from
audioformat_capture_quirk() so that no further matching will be done.
And, later on, Olivia reported later that the fiddling with the
capture quirk alone doesn't suffice for the test with speaker-test
program. This seems to be a similar case like the recently fixed BOSS
devices. Indeed, the problem could be addressed by setting
playback_first flag, which indicates that the playback URBs have to be
sent out at first even in the implicit feedback mode.
This patch implements the application of the implicit feedback to
Pioneer devices as described in the above. The former
skip_pioneer_sync_ep() was dropped, and instead we provide
is_pioneer_implicit_fb() to check the Pioneer devices that need the
implicit feedback. In the audioformat_implicit_fb_quirk(), simply
apply the implicit fb for playback and set chip->playback_first flag
if matching, and in audioformat_capture_quirk()(), it returns 1 for
skipping the generic EP sync handling.
Reported-by: Geraldo <geraldogabriel@gmail.com>
Tested-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/s5ha6pygqfz.wl-tiwai@suse.de
Link: https://lore.kernel.org/r/20210419153918.450-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add case statement to set sample-rate for the DJM-750 Pioneer
mixer. This was included as part of another patch but I think it has
been archived on Patchwork and hasn't been merged.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210418165901.25776-1-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It makes USB audio capture and playback possible and pristine on my Roland
INTEGRA-7, Boutique D-05, and R-26, along with many more I've encountered
people having had issues with over the last decade or so.
Signed-off-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the recent rewrite of the implicit feedback support, we've
tested to apply the implicit fb on BOSS devices, but it failed, as the
capture stream didn't start without the playback. As the end result,
it got another type of quirk for tying both streams but starts
playback always (commit 6234fdc1ce "ALSA: usb-audio: Quirk for BOSS
GT-001").
Meanwhile, Mike Oliphant has tested the real implicit feedback mode
for the playback again with the latest code, and found out that it
actually works if the initial feedback sync is skipped; that is, on
those BOSS devices, the playback stream has to be started at first
without waiting for the capture URB completions. Otherwise it gets
stuck. In the rest operations after the capture stream processed, we
can take them as the implicit feedback source.
This patch is an attempt to improve the support for BOSS devices with
the implicit feedback mode in the way described above. It adds a new
flag to snd_usb_audio, playback_first, indicating that the playback
stream starts without sync with the initial capture completion. This
flag is set in the quirk table with the new IMPLICIT_FB_BOTH type.
Reported-and-tested-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, we have some assumption that the audio clock
selector has been set up implicitly and don't want to touch it unless
it's really needed for the fallback autoclock setup. This works for
most devices but some seem having a problem. Partially this was
covered for the devices with a single connector at the initialization
phase (commit 086b957cc1 "ALSA: usb-audio: Skip the clock selector
inquiry for single connections"), but also there are cases where the
wrong clock set up is kept silently. The latter seems to be the cause
of the noises on Behringer devices.
In this patch, we explicitly set up the audio clock selector whenever
the appropriate node is found.
Reported-by: Geraldo Nascimento <geraldogabriel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199327
Link: https://lore.kernel.org/r/CAEsQvcvF7LnO8PxyyCxuRCx=7jNeSCvFAd-+dE0g_rd1rOxxdw@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210413084152.32325-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UA-101 device and co are supported by another driver, snd-ua101, but
the USB audio class driver (snd-usb-audio) catches all and this
resulted in the lack of functionality like missing MIDI devices.
This patch introduces a sort of deny-listing for those devices to just
return -ENODEV at probe in snd-usb-audio driver, so that it falls back
to the probe by snd-ua101.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212477
Link: https://lore.kernel.org/r/20210408075656.30184-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few calls of usb_driver_claim_interface() but all of those
miss the proper error checks, as reported by Coverity. This patch
adds those missing checks.
Along with it, replace the magic pointer with -1 with a constant
USB_AUDIO_IFACE_UNUSED for better readability.
Reported-by: coverity-bot <keescook+coverity-bot@chromium.org>
Addresses-Coverity-ID: 1475943 ("Error handling issues")
Addresses-Coverity-ID: 1475944 ("Error handling issues")
Addresses-Coverity-ID: 1475945 ("Error handling issues")
Fixes: b1ce7ba619 ("ALSA: usb-audio: claim autodetected PCM interfaces all at once")
Fixes: e5779998bf ("ALSA: usb-audio: refactor code")
Link: https://lore.kernel.org/r/202104051059.FB7F3016@keescook
Link: https://lore.kernel.org/r/20210406113534.30455-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'tags/mute-led-rework' into for-next
ALSA: control - add generic LED API
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Logitech ConferenceCam Connect is a compound USB device with UVC and
UAC. Not 100% reproducible but sometimes it keeps responding STALL to
every control transfer once it receives get_freq request.
This patch adds 046d:0x084c to a snd_usb_get_sample_rate_quirk list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=203419
Signed-off-by: Ikjoon Jang <ikjn@chromium.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210324105153.2322881-1-ikjn@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rear Mic on Lenovo P620 cannot record after S3, despite that there's no
error and the other two functions of the USB audio, Line In and Line
Out, work just fine.
The mic starts to work again after running userspace app like "alsactl
store". Following the lead, the evidence shows that as soon as connector
status is queried, the mic can work again.
So also check connector value on resume to "wake up" the USB audio to
make it functional.
This can be device specific, however I think this generic approach may
benefit more than one device.
Now the resume callback checks connector, and a new callback,
reset_resume, to also restore switches and volumes.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210325165918.22593-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Majority of changes are various ASoC device/platform-specific small
fixes (including a removal of stale file) while the only common
change is a clk management fix in ASoC simple-card driver.
The rest are usual HD-audio quirks.
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Merge tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The majority of changes are various ASoC device/platform-specific
small fixes (including a removal of stale file) while the only common
change is a clk management fix in ASoC simple-card driver.
The rest are the usual HD-audio quirks"
* tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (44 commits)
ALSA: usb-audio: Fix unintentional sign extension issue
ALSA: hda/realtek: fix mute/micmute LEDs for HP 850 G8
ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms
ASoC: rt711: add snd_soc_component remove callback
ASoC: rt5659: Update MCLK rate in set_sysclk()
ASoC: simple-card-utils: Do not handle device clock
ALSA: hda/realtek: fix mute/micmute LEDs for HP 440 G8
ALSA: hda/realtek: fix mute/micmute LEDs for HP 840 G8
ALSA: hda/realtek: apply pin quirk for XiaomiNotebook Pro
ALSA: hda/realtek: Apply headset-mic quirks for Xiaomi Redmibook Air
ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge
ALSA: dice: fix null pointer dereference when node is disconnected
ALSA: hda: generic: Fix the micmute led init state
ASoC: qcom: lpass-cpu: Fix lpass dai ids parse
spi: cadence: set cqspi to the driver_data field of struct device
ASoC: SOF: intel: fix wrong poll bits in dsp power down
ASoC: codecs: wcd934x: add a sanity check in set channel map
ASoC: qcom: sdm845: Fix array out of range on rx slim channels
ASoC: qcom: sdm845: Fix array out of bounds access
ASoC: remove remnants of sirf prima/atlas audio codec
...
The shifting of the u8 integer device by 24 bits to the left will
be promoted to a 32 bit signed int and then sign-extended to a
64 bit unsigned long. In the event that the top bit of device is
set then all then all the upper 32 bits of the unsigned long will
end up as also being set because of the sign-extension. Fix this
by casting device to an unsigned long before the shift.
Addresses-Coverity: ("Unintended sign extension")
Fixes: a07df82c79 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210318132008.15266-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MODULE_SUPPORTED_DEVICE was added in pre-git era and never was
implemented. We can safely remove it, because the kernel has grown
to have many more reliable mechanisms to determine if device is
supported or not.
Signed-off-by: Leon Romanovsky <leonro@nvidia.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The problem was in wrong "if" placement. chip->quirk_type is freed
in snd_card_free_when_closed(), but inside if statement it's accesed.
Fixes: 9799110825 ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Signed-off-by: Pavel Skripkin <paskripkin@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/16da19126ff461e5e64a9aec648cce28fb8ed73e.1615242183.git.paskripkin@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Other Plantronics headset models seem requiring the same workaround as
C320-M to add the 20ms delay for the control messages, too. Apply the
workaround generically for devices with the vendor ID 0x047f.
Note that the problem didn't surface before 5.11 just with luck.
Since 5.11 got a big code rewrite about the stream handling, the
parameter setup procedure has changed, and this seemed triggering the
problem more often.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304085009.4770-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rear audio on Lenovo ThinkStation P620 stops working after commit
1965c4364b ("ALSA: usb-audio: Disable autosuspend for Lenovo
ThinkStation P620"):
[ 6.013526] usbcore: registered new interface driver snd-usb-audio
[ 6.023064] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023083] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023090] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023098] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023103] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023110] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045846] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045866] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045877] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045886] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045894] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045908] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
I overlooked the issue because when I was working on the said commit,
only the front audio is tested. Apology for that.
Changing supports_autosuspend in driver is too late for disabling
autosuspend, because it was already used by USB probe routine, so it can
break the balance on the following code that depends on
supports_autosuspend.
Fix it by using usb_disable_autosuspend() helper, and balance the
suspend count in disconnect callback.
Fixes: 1965c4364b ("ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304043419.287191-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unlike the other DJM, the value to set the "CD/LINE" and "LINE" capture
control options are inverted. This fix makes sure that the displayed
info label while using `alsamixer` matches the input switches label
on the DJM-850 mixer.
Signed-off-by: Nicolas MURE <nicolas.mure2019@gmail.com>
Link: https://lore.kernel.org/r/20210301152729.18094-5-nicolas.mure2019@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit only contains the fix about the `URB_CONTROL` request
direction to set the samplerate of Pioneer DJM devices (`URB_CONTROL out`).
Fixes: 3b85f5fc75 ("ALSA: usb-audio: Add DJM450 to Pioneer format quirk")
Signed-off-by: Nicolas MURE <nicolas.mure2019@gmail.com>
Link: https://lore.kernel.org/r/20210301142927.14552-1-nicolas.mure2019@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Corsair Virtuoso SE RGB Wireless is a USB headset with a mic and a
sidetone feature. Assign the Corsair Virtuoso name map to the SE product
ids as well, in order to label its mixer appropriately and allow
userspace to pick the correct volume controls.
Signed-off-by: Andrea Fagiani <andfagiani@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/40bbdf55-f854-e2ee-87b4-183e6451352c@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A number of devices have named substreams which are hard to remember /
decypher from <device> MIDI n names. Eg. Korg puts a pass through on
one substream and iConnectivity devices name the connections.
This makes it easier to connect to the correct device. Devices which
handle naming through quirks are unaffected by this change.
Addresses TODO comment in sound/usb/midi.c
Signed-off-by: George Harker <george@george-graphics.co.uk>
Link: https://lore.kernel.org/r/20210226212617.24616-1-george@george-graphics.co.uk
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>
Some USB audio firmware seem to report broken dB values for the volume
controls, and this screws up applications like PulseAudio who blindly
trusts the given data. For example, Edifier G2000 reports a PCM
volume from -128dB to -127dB, and this results in barely inaudible
sound.
This patch adds a sort of sanity check at parsing the dB values in
USB-audio driver and disables the dB reporting if the range looks
bogus. Here, we assume -96dB as the bottom line of the max dB.
Note that, if one can figure out that proper dB range later, it can be
patched in the mixer maps.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211929
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210227105737.3656-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 93db51d06b ("ALSA: usb-audio: Check valid altsetting at
parsing rates for UAC2/3") changed the behavior of the function
set_sample_rate_v2v3() slightly to treat the inconsistent sample rate
as an error. It was done by assumption that the sample rate
validation should have been done at the parser phase as implemented in
that patch. But the validation is later selectively enabled only for
certain devices as it causes a regression (the commit fe773b8711
"ALSA: usb-audio: workaround for iface reset issue"), and now the
inconsistency surfaced as a fatal error while it worked in the past as
is, as reported for FiiO M3K DAC.
For recovering from the regression, change set_sample_rate_v2v3()
again to ignore the sample rate difference as non-error.
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1182633
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210227082002.21185-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BOSS GP-10 with 0582:0185 requires the similar quirk to make the
implicit feedback working like other BOSS devices.
Reported-by: Keith Milner <kamilner@superlative.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210214154251.10750-1-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 endpoint management has bit flags to indicate the current state,
and we're dealing two things: the running bit and the stopping bit.
There is a thin window in transition from the running to the stopping
in stop_urbs(), and as long as the bit flags are used, it's difficult
to plug.
This patch modifies the state management code to use the atomic int
and follow the explicit three states, STOPPED, RUNNING and STOPPING.
The state change is done via atomic_cmpxhg() for avoiding possible
races, and check the state change more strictly. The unexpected state
change is now handled as an error.
Fixes: d0f09d1e4a ("ALSA: usb-audio: Refactoring endpoint URB deactivation")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210206203052.15606-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we stop an endpoint in release_urbs(), it ignores the
inconsistent endpoint state and tries to release the resources.
This shouldn't happen in theory, but it's still safer to abort the
release and let the caller proper error handling.
Also, stop_and_unlink_urbs() called from release_urbs() does two step
works, and it's more straightforward to split this to two functions
again, so that the call from the PCM trigger won't take the path with
sleeping.
This patch modifies the EP management code to adapt two points above.
Fixes: d0f09d1e4a ("ALSA: usb-audio: Refactoring endpoint URB deactivation")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210206203052.15606-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds mixer quirks for the Pioneer DJM-900NXS2 mixer. This
device has 6 capture channels, 5 of them allow setting the signal
source. This adds controls for these, similar to the DJM-250Mk2.
However, playpack channels are not controllable via software like on the
250Mk2, as they can only be set manually on the mixing console.
Read-only controls showing the currently selected playback channels are
omitted.
Signed-off-by: Fabian Lesniak <fabian@lesniak-it.de>
Link: https://lore.kernel.org/r/20210205215116.258724-2-fabian@lesniak-it.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows for N different devices to use the pioneer mixer quirk for
setting capture/record type and recording level. The impementation has
not changed much with the exception of an additional mask on
private_value to allow storing of a device index:
DEVICE MASK 0xff000000
GROUP_MASK 0x00ff0000
VALUE_MASK 0x0000ffff
This could be improved by changing the arrays of wValues for each
channel to contain named definitions (e.g. SND_DJM_CAP_LINE). It would
improve readability and perhaps would allow using the same array for
multiple channels. The channel number can be specified on the control
next to the wIndex.
Feedback is very much appreciated as I'm not the most proficient C
programmer but am learning as I go.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210205184256.10201-2-livvy@base.nu
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>
The kerndoc comment for the new function snd_usb_endpoint_free_all()
had a typo wrt the argument name. Fix it.
Fixes: 00272c6182 ("ALSA: usb-audio: Avoid unnecessary interface re-setup")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210205082837.6327-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As with most Pioneer devices, the device descriptor is vendor specific
and as such, the number of channels, the PCM format, endpoints and
sample rate need to be specified. This device has 8 inputs and 8 outputs
and a sample rate of 48000 only. The PCM format is S24_3LE like other
devices.
There seems to be an appetite for reducing duplication amongs these
Pioneer patches but again, I feel this is a step to be taken after
support has been added as it's not completely clear where the
commonalities are.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210202134225.3217-3-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the DJM-750, ensure that the format control message is passed to
the device when opening a stream. It seems as though fmt->sync_ep is not
always set when this function is called hence the passing of the value
at the call site. If this can be fixed, fmt->sync_up should be used as
the wvalue.
There doesn't seem to be a "cpu_to_le24" type function defined hence for
the open code but I did see a similar thing done in Bluez lib. Perhaps
we can get these definitions defined in byteorder.h. See hci_cpu_to_le24
in include/net/bluetooth/hci.h:2543 for similar usage.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210202134225.3217-2-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced sample rate validation code seems causing a
problem on some devices; namely, after performing this, the bus gets
screwed and it influences even on other USB devices.
As a quick workaround, perform it only for the necessary devices;
currently MOTU devices are known to need the valid altset checks, so
filter out other devices.
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Reported-by: Jamie Heilman <jamie@audible.transient.net>
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1178203
Link: https://lore.kernel.org/r/20210123155842.22652-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At probing a UAC2/UAC3 device like NUX MG-300 USB interface, we get
error messages "RANGE setting not yet supported". It comes the place
where the driver tries to determine the resolution of mixer volumes
via SET_CUR_RES and GET_CUR_RES verbs. Those verbs aren't supported
on UAC2 and UAC3, hence the driver warns like the above. Although the
driver handles this error and works as expected, it's still ugly to
show such errors unnecessarily.
This patch papers over the errors by applying the resolution detection
only for UAC1 and skipping it for UAC2/UAC3.
Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current USB-audio driver gets an error at probing NUX MG-300 about
parsing the clocks. This is because the firmware doesn't return the
proper connection of the clock selector that is connected to a single
clock; it's likely that the firmware was lazy^w optimized and the
inquiry wasn't handled. Actually it makes little sense to inquire and
set up the single connection explicitly.
This patch fixes the issue by simply skipping the clock selector
inquiry if it's a single connection.
Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-1-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>
This adds the Pioneer DJ DJM-750 to the quirks table and ensures
skip_pioneer_sync_ep() is (also) called: this device uses the vendor
ID of 0x08e4 (I'm not sure why they use multiple vendor IDs but many
just like to be awkward it seems).
Playback on all 8 channels works. I'll likely keep this working in the
future and submit futher patches and improvements as necessary.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210118130621.77miiie47wp7mump@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For addressing the regression on Pioneer devices, we recently
corrected the quirk code to enable the implicit feedback mode on those
devices properly. However, the devices still showed problems with the
full duplex operations with JACK, and after debug sessions, we figured
out that the older kernels that had worked with JACK also didn't use
the implicit feedback mode at all although they had the quirk code to
enable it; instead, the old code worked just to skip the normal sync
endpoint setup that would have been detected without it. IOW, what
broke without the implicit-fb quirk in the past was the application of
the normal sync endpoint that is actually the capture data endpoint on
these devices.
This patch covers the overseen piece: it modifies the quirk code again
not to enable the implicit feedback mode but just to make the driver
skipping the sync endpoint detection. This made the driver working
with JACK full-duplex mode again.
Still it's not quite clear why the implicit feedback doesn't work on
those devices yet; maybe it's about some issues in the URB setup. But
at least, with this patch, the driver should work in the level of the
older kernels again.
Fixes: 167c9dc84e ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices")
Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC2/3 sample rate setup is based on the clock node, which is
usually shared in the interface, and can't be re-setup without
deselecting the interface once, and that's how the current code
behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence
we basically need to call for each endpoint usage even if those share
the same interface.
This patch fixes the behavior of UAC1 to call always
snd_usb_init_sample_rate() in snd_usb_endpoint_configure().
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current sample rate setup function for UAC1 assumes only the first
endpoint retrieved from the interface:altset pair, but the rate set up
may be needed also for the secondary endpoint. Also, retrieving the
endpoint number from the interface descriptor is redundant; we have
already the target endpoint in the given audioformat object.
This patch simplifies the code and corrects the target endpoint as
described in the above. It simply refers to fmt->endpoint directly.
Also, this patch drops the pioneer_djm_set_format_quirk() that is
caleld from snd_usb_set_format_quirk(); this function does the sample
rate setup but for the capture endpoint (0x82), and that's exactly
what the change above fixes.
Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last remaining usage of strlcpy() in USB-audio driver is the setup
of the card longname string. Basically we need to know whether any
non-empty string is set or not, and no real length is needed.
Refactor the code and use strscpy() instead. After this change,
strlcpy() is gone from all sound/* code.
Link: https://lore.kernel.org/r/20210115100437.20906-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver still contains two calls of strlcpy() because the
return size is evaluated. Basically it just checks whether the string
is copied or not, but since strcpy() may return a negative error code,
we should check the negative value and treat as filled.
Link: https://lore.kernel.org/r/20210115095758.19707-1-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>
Pioneer devices have both playback and capture streams sharing the
same iface/altsetting, and those need to be paired as implicit
feedback. Instead of a half-baked (and broken) static quirk entry,
set up more generically for those devices by checking the number of
endpoints and the attribute of the secondary EP.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are devices that have multiple endpoints sharing the same
iface/altset not only for sync but also for the actual streams, and
the audioformat for such an endpoint needs to be handled with the
proper endpoint index; otherwise it confuses the endpoint management.
This patch extends the audioformat to annotate the endpoint index, and
put the proper ep_idx=1 to Pioneer device quirk entries accordingly.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current endpoint handling assumed (more or less) a unique 1:1
relation between the endpoint and the iface/altset. The exception was
the sync EP without the implicit feedback which has usually the
secondary EP of the same altset. This works fine for most devices,
but it turned out that some unusual devices like Pinoeer's ones have
both playback and capture endpoints in the same iface/altsetting and
use both for the implicit feedback mode. For handling such a case, we
need to extend the endpoint management to take the shared interface
into account.
This patch does that: it adds a new object snd_usb_iface_ref for
managing the reference counts of the each USB interface that is used
by each endpoint. The interface setup is performed only once for the
(sharing) endpoints, and the doubly initialization is avoided.
Along with this, the resource release of endpoints and interface
refcounts are put into a single function, snd_usb_endpoint_free_all()
instead of looping in the caller side.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The implicit feedback mode needs to handle two endpoints and the
choice of the audioformat object for the sync EP is important since
this determines the compatibility of the hw_params. The current code
uses the same audioformat object if both the main EP and the sync EP
point to the same iface/altsetting. This was done in consideration of
the non-implicit-fb sync EP handling, and it doesn't match well with
the cases where actually to endpoints are defined in the sameiface /
altsetting like a few Pioneer devices.
Modify snd_usb_find_implicit_fb_sync_format() to pick up the
audioformat that is assigned in the counter-part substreams primarily,
so that the actual capture stream can be opened properly. We keep the
same audioformat object only as a fallback in case nothing found,
though.
Fixes: 9fddc15e80 ("ALSA: usb-audio: Factor out the implicit feedback quirk code")
Link: https://lore.kernel.org/r/20210108075219.21463-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change in the endpoint management moved the endpoint object
creation from the stream open time to the parser of the audio
descriptor. It works fine for the standard audio, but it overlooked
the other places that create audio streams via quirks
(QUIRK_AUDIO_FIXED_ENDPOINT) like the reported a few Pioneer devices;
those call snd_usb_add_audio_stream() manually, hence they miss the
endpoints, eventually resulting in the error at opening streams.
Moreover, now the sync EP setup was moved to the explicit call of
snd_usb_audioformat_set_sync_ep(), and this needs to be added for
those places, too.
This patch addresses those regressions for quirks. It adds a local
helper function add_audio_stream_from_fixed_fmt(), which does the all
needed tasks, and replaces the calls of snd_usb_add_audio_stream()
with this new function.
Fixes: 54cb31901b ("ALSA: usb-audio: Create endpoint objects at parsing phase")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
strlcpy is deprecated. see: Documentation/process/deprecated.rst
Change the calls that do not use the strlcpy return value to the
preferred strscpy.
Done with cocci script:
@@
expression e1, e2, e3;
@@
- strlcpy(
+ strscpy(
e1, e2, e3);
This cocci script leaves the instances where the return value is
used unchanged.
After this patch, sound/ has 3 uses of strlcpy() that need to be
manually inspected for conversion and changed one day.
$ git grep -w strlcpy sound/
sound/usb/card.c: len = strlcpy(card->longname, s, sizeof(card->longname));
sound/usb/mixer.c: return strlcpy(buf, p->name, buflen);
sound/usb/mixer.c: return strlcpy(buf, p->names[index], buflen);
Miscellenea:
o Remove trailing whitespace in conversion of sound/core/hwdep.c
Link: https://lore.kernel.org/lkml/CAHk-=wgfRnXz0W3D37d01q3JFkr_i_uTL=V6A6G1oUZcprmknw@mail.gmail.com/
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/22b393d1790bb268769d0bab7bacf0866dcb0c14.camel@perches.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BOSS RC-505 (shown by lsusb as "Roland Corp. RC-505") does require the
same quirk as these other BOSS devices.
Without this quirk it is neither possible to capture audio from nor to
write audio to the RC-505. Both just result in an empty audio
stream. With these changes both capture and playback seem to work
quite fine. MIDI funtionality was not tested.
Tested-by: Harry Reinold <harry.reinold@posteo.de>
Signed-off-by: Timon Reinold <tirei@agon.one>
Link: https://lore.kernel.org/r/20210102210835.21268-1-tirei@agon.one
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BOSS AD-10 requires the very same quirk like other BOSS devices to
enable the special implicit feedback mode.
Reported-and-tested-by: Martin Passing <martin@passing.name>
Link: https://lore.kernel.org/r/20201229083428.20467-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DIV_ROUND_UP() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.
Generated using the following the Coccinelle semantic patch.
// <smpl>
@@
expression x, y;
@@
-(((x) + (y) - 1) / (y))
+DIV_ROUND_UP(x, y)
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-11-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calculation of in_cables and out_cables bitmaps are done with the
bit shift by the value from the descriptor, which is an arbitrary
value, and can lead to UBSAN shift-out-of-bounds warnings.
Fix it by filtering the bad descriptor values with the check of the
upper bound 0x10 (the cable bitmaps are 16 bits).
Reported-by: syzbot+92e45ae45543f89e8c88@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201223174557.10249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BOSS GT-1 (USB ID 0582:01d6) requires implicit feedback
like other similar BOSS devices. This patch adds this support.
[ rearranged the table entry in the ID order -- tiwai ]
Signed-off-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20201221215533.2511-1-oliphant@nostatic.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some buggy firmware don't give the current sample rate but leaves
zero. Handle this case more gracefully without warning but just skip
the current rate verification from the next time.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201218145858.2357-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current channel-map control implementation in USB-audio driver may
lead to an error message like
"control 3:0:0:Playback Channel Map:0: access overflow"
when CONFIG_SND_CTL_VALIDATION is set. It's because the chmap get
callback clears the whole array no matter which count is set, and
rather the false-positive detection.
This patch fixes the problem by clearing only the needed array range
at usb_chmap_ctl_get().
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201211130048.6358-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Steinberg UR22 (with USB ID 0499:1509) requires the implicit feedback
for the proper playback, otherwise it causes occasional cracks.
This patch adds the corresponding the quirk table entry with the
recently added generic implicit fb support.
Reported-and-tested-by: Kilian <meschi@posteo.de>
Link: https://lore.kernel.org/r/20201209161835.13625-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another quirk for Pioneer DJ DDJ-SR2, which is quite similar like
other DJ DDJ models but with slightly different EPs or channels.
Reported-by: Geraldo <geraldogabriel@gmail.com>
Link: https://lore.kernel.org/r/20201130083714.10640-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch allows the Pioneer DJ DDJ-RR to be seen as a USB audio
device under Linux and therefore usable in such applications as
Mixxx.
Tested Master Audio out, headphones (both output jacks) and microphone
input. All work perfectly.
Signed-off-by: Daniel Martin <dmanlfc@gmail.com>
Link: https://lore.kernel.org/r/20201128084035.2958-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the usb audio driver correctly finds implicit feedback endpoints,
the implicit feedback quirk for the MOTU M-Series is no longer required.
This also removes some unnecessary vendor specific messages from the MOTU
M-Series boot quirk. The removed vendor specific messages turned on vendor
specific interrupts to the host every 32 samples. The only thing the boot
quirk needs to do is wait for 2 seconds.
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Signed-off-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-42-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few other BOSS devices (BR-80, GT-100v2, Katana) seem requiring the
same quirk as BOSS GT-001, i.e. no implicit feedback for playback but
tying with capture. Add and correct the corresponding quirk table
entries for them.
Reported-and-tested-by: Keith Milner <kamilner@superlative.org>
Link: https://lore.kernel.org/r/20201123085347.19667-41-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A new module option, implicit_fb, is added to specify the driver
looking for the implicit feedback sync. This can be useful for a
device that could be working better in the implicit feed back mode and
user wants to test it quickly. When this works, we can add the quirk
entry easily.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-40-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch extends the implicit feedback mode parser code to check the
description more generically, so that the quirk entries can be added
without the explicit EP and interface numbers. The search is done for
the next and the previous interface of the given altset, and if both
entries are ASYNC mode and the direction matches, it just takes as the
sync endpoint. The generic parser is applicable only for the playback
stream.
As of now, only a few M-Audio devices have been converted to use this
mode.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-39-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>
Now the sync endpoints have been parsed at the beginning and won't be
changed dynamically, let's show them in the proc outputs for helping
debugging.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-36-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just for consistency, use unsigned char for iface and altsetting in
allover places. Also rearrange the field positions of
snd_usb_endpiont and tidy up with some comments.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-35-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Follow the inclusive terminology, just replace sync_master/sync_slave
with sync_source/sync_sink. It's also a bit clearer from its meaning,
too.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-34-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are multiple places in format.c performing the similar code for
setting the rate_min, rate_max and rates fields. This patch unifies
those in a helper function and calls it at the end of the parser phase
so that all rate_table entries have been already determined.
No functional changes, 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/20201123085347.19667-33-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>
Minor code refactoring to consolidate the URB deactivation code in
endpoint.c. A slight behavior change is that the error handling in
snd_usb_endpoint_start() leaves EP_FLAG_STOPPING now. This should be
synced with the later PCM sync_stop callback.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-30-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The endpoint objects may be started/stopped concurrently by different
substreams in the case of implicit feedback mode, while the current
code handles the reference counter without any protection.
This patch changes the refcount to atomic_t for avoiding the
inconsistency. We need no reference_t here as the refcount goes only
up to 2.
Also the name "use_count" is renamed to "running" since this is about
actually the running status, not the open refcount.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-29-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>
The implicit feedback mode uses a ring buffer for storing the received
packet sizes from the feedback source, and the code has a slight flaw;
when a playback stream stalls by some reason and the URBs aren't
processed, the next_packet FIFO might become empty, but the driver
can't distinguish whether it's empty or full because it's managed with
read_poss and write_pos.
This patch addresses those by changing the next_packet array
management. Instead of keeping read and write positions, now the head
position and the queued amount are kept. It's easier to understand
about the emptiness. Also, the URB active flag is now cleared before
calling queue_pending_output_urbs() for avoiding (theoretically)
possible inconsistency.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-27-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>
Setting the active altsetting at changing sample rate seems
unrecommended. The host should deselect the altsetting at first
before that, then select it again.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-18-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Factor out the code to obtain snd_usb_endpoint object matching with
the given endpoint. It'll be used in the later patch to add the
implicit feedback hw-constraint.
No functional change by this patch itself.
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
Link: https://lore.kernel.org/r/20201123085347.19667-6-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 current driver code assumes blindly that all found sample rates for
the same endpoint from the UAC2 and UAC3 descriptors can be used no
matter which altsetting, but actually this was wrong: some devices
accept only limited sample rates in each altsetting. For determining
which altsetting supports which rate, we need to verify each sample rate
and check the validity via UAC2_AS_VAL_ALT_SETTINGS. This control
reports back the available altsettings as a bitmap.
This patch implements the missing piece above, the verification and
reconstructs the sample rate tables based on the result.
An open question is how to deal with the altsettings that ended up
with no valid sample rates after verification. At least, there is a
device that showed this problem although the sample rates did work in
the later usage (see bug link). For now, we accept such an altset as
is, assuming that it's a firmware bug.
Reported-by: Dylan Robinson <dylan_robinson@motu.com>
Tested-by: Keith Milner <kamilner@superlative.org>
Tested-by: Dylan Robinson <dylan_robinson@motu.com>
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1178203
Link: https://lore.kernel.org/r/20201123085347.19667-4-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>
Found one more Logitech device, BCC950 ConferenceCam, which needs
the same delay here. This makes 3 out of 3 devices I have tried.
Therefore, add a delay for all Logitech devices as it does not hurt.
Signed-off-by: Joakim Tjernlund <joakim.tjernlund@infinera.com>
Cc: <stable@vger.kernel.org> # 4.19.y, 5.4.y
Link: https://lore.kernel.org/r/20201117122803.24310-1-joakim.tjernlund@infinera.com
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>
Khadas audio devices ( USB_ID_VENDOR 0x3353 )
have DSD-capable implementations from XMOS
need add new usb vendor id for recognition
Signed-off-by: Artem Lapkin <art@khadas.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201103103311.5435-1-art@khadas.com
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>
Just a few additional small and trivial fixes.
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Merge tag 'sound-fix-5.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Just a few additional small and trivial fixes"
* tag 'sound-fix-5.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix the return value if cb func is already registered
ALSA: usb-audio: Line6 Pod Go interface requires static clock rate quirk
ALSA: hda/ca0132: make some const arrays static, makes object smaller
ALSA: sparc: dbri: fix repeated word 'the'
Recently released Line6 Pod Go requires static clock rate quirk to make
its usb audio interface working. Added its usb id to the list of similar
line6 devices.
Signed-off-by: Lukasz Halman <lukasz.halman@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201020061409.GA24382@TAG009442538903
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amount of changes is smaller at this round (what a surprise),
but lots of activity is seen. Most of changes are about ASoC
driver development, especially Intel platforms.
Here are some highlights:
General:
* Replace all tasklet usages with other alternatives
* Cleanup of the ASoC error unwinding code
* Fixes for trivial issues caught by static checker
* Spell fixes allover the places
ALSA Core:
* Lockdep fix for control devices
* Fix for potential OSS sequencer mutex stalls
HD-audio and USB-audio:
* SoundBlaster AE-7 support
* Changes in quirk table for the rename handling
* Quirks for HP and ASUS machines, Pioneer DJ DJM-250MK2.
ASoC:
* Lots of updates for Intel SOF and SoundWire enablement
* Replacement of the DSP driver for some older x86 systems;
the new code was written from scratch, better maintenance
expected
* Helpers for parsing auxiluary devices from the device tree
* New support for AllWinner A64, Cirrus Logic CS4234, Mediatek
MT6359 Microchip S/PDIF TX and RX controllers, Realtek RT1015P,
and Texas Instruments J721E, TAS2110, TAS2564 and TAS2764
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Merge tag 'sound-5.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"The amount of changes is smaller at this round (what a surprise), but
lots of activity is seen. Most of changes are about ASoC driver
development, especially Intel platforms. Here are some highlights:
General:
- Replace all tasklet usages with other alternatives
- Cleanup of the ASoC error unwinding code
- Fixes for trivial issues caught by static checker
- Spell fixes allover the places
ALSA Core:
- Lockdep fix for control devices
- Fix for potential OSS sequencer mutex stalls
HD-audio and USB-audio:
- SoundBlaster AE-7 support
- Changes in quirk table for the rename handling
- Quirks for HP and ASUS machines, Pioneer DJ DJM-250MK2.
ASoC:
- Lots of updates for Intel SOF and SoundWire enablement
- Replacement of the DSP driver for some older x86 systems; the new
code was written from scratch, better maintenance expected
- Helpers for parsing auxiluary devices from the device tree
- New support for AllWinner A64, Cirrus Logic CS4234, Mediatek MT6359
Microchip S/PDIF TX and RX controllers, Realtek RT1015P, and Texas
Instruments J721E, TAS2110, TAS2564 and TAS2764"
* tag 'sound-5.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (498 commits)
ALSA: hda/hdmi: fix incorrect locking in hdmi_pcm_close
ALSA: hda: fix jack detection with Realtek codecs when in D3
ALSA: fireworks: use semicolons rather than commas to separate statements
ALSA: hda: use semicolons rather than commas to separate statements
ALSA: hda/i915 - fix list corruption with concurrent probes
ASoC: dmaengine: Document support for TX only or RX only streams
ASoC: mchp-spdiftx: remove 'TX' from playback stream name
ASoC: ti: davinci-mcasp: Use &pdev->dev for early dev_warn
ASoC: tas2764: Add the driver for the TAS2764
dt-bindings: tas2764: Add the TAS2764 binding doc
ASoC: Intel: catpt: Add explicit DMADEVICES kconfig dependency
ASoC: Intel: catpt: Fix compilation when CONFIG_MODULES is disabled
ASoC: stm32: dfsdm: add actual resolution trace
ASoC: stm32: dfsdm: change rate limits
ASoC: qcom: sc7180: Add support for audio over DP
Asoc: qcom: lpass-platform : Increase buffer size
ASoC: qcom: Add support for lpass hdmi driver
Asoc: qcom: lpass:Update lpaif_dmactl members order
Asoc:qcom:lpass-cpu:Update dts property read API
ASoC: dt-bindings: Add dt binding for lpass hdmi
...
The usb_control_msg_send() and usb_control_msg_recv() calls can return
an error if a "short" write/read happens, and they can handle data off
of the stack, so move the driver over to using those calls instead,
saving some logic when dynamically allocating memory.
v2: API change of use usb_control_msg_send() and usb_control_msg_recv()
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Vasily Khoruzhick <anarsoul@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-9-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20200923134348.23862-13-oneukum@suse.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The usb_control_msg_send() and usb_control_msg_recv() calls can return
an error if a "short" write/read happens, so move the driver over to
using those calls instead, saving some logic in the wrapper functions
that were being used in this driver.
This also resolves a long-staging bug where data on the stack was being
sent in a USB control message, which was not allowed.
v2: API change of usb_control_msg_send()
Cc: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-8-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20200923134348.23862-11-oneukum@suse.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The usb_control_msg_send() call can handle data on the stack, as well as
returning an error if a "short" write happens, so move the driver over
to using that call instead. This ends up removing a helper function
that is no longer needed.
v2: API change in usb_control_msg_send()
Cc: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-7-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20200923134348.23862-10-oneukum@suse.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
This patch extends support for DJM-250MK2 and allows mapping
playback and capture channels to available sources.
Configures the card through USB commands.
Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Link: https://lore.kernel.org/r/20200922144206.10472-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 34dedd2a83.
According to Realtek, volume FU works for line-in.
I can confirm volume control works after device firmware is updated.
Fixes: 34dedd2a83 ("ALSA: usb-audio: Disable Lenovo P620 Rear line-in volume control")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200915103925.12777-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A USB device will always haev a bi-directional endpoint 0, that's just
how the devices work, so no need to check for that in a few quirk tests
as it will always pass.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Alexander Tsoy <alexander@tsoy.me>
Reported-by: Alan Stern <stern@rowland.harvard.edu>
Link: https://lore.kernel.org/r/20200914153756.3412156-12-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The usb_control_msg_send() call can return an error if a "short" write
happens, so move the driver over to using that call instead.
Cc: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-10-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The usb_control_msg_send() and usb_control_msg_recv() calls can return
an error if a "short" write/read happens, and they can handle data off
of the stack, so move the driver over to using those calls instead,
saving some logic when dynamically allocating memory.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Vasily Khoruzhick <anarsoul@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-9-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The usb_control_msg_send() and usb_control_msg_recv() calls can return
an error if a "short" write/read happens, so move the driver over to
using those calls instead, saving some logic in the wrapper functions
that were being used in this driver.
This also resolves a long-staging bug where data on the stack was being
sent in a USB control message, which was not allowed.
Cc: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-8-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The usb_control_msg_send() call can handle data on the stack, as well as
returning an error if a "short" write happens, so move the driver over
to using that call instead. This ends up removing a helper function
that is no longer needed.
Cc: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-7-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
snd_usb_pipe_sanity_check() is a great function, so let's move it into
the USB core so that other parts of the kernel, including the USB core,
can call it.
Name it usb_pipe_type_check() to match the existing
usb_urb_ep_type_check() call, which now uses this function.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: "Gustavo A. R. Silva" <gustavoars@kernel.org>
Cc: Eli Billauer <eli.billauer@gmail.com>
Cc: Emiliano Ingrassia <ingrassia@epigenesys.com>
Cc: Alan Stern <stern@rowland.harvard.edu>
Cc: Alexander Tsoy <alexander@tsoy.me>
Cc: "Geoffrey D. Bennett" <g@b4.vu>
Cc: Jussi Laako <jussi@sonarnerd.net>
Cc: Nick Kossifidis <mickflemm@gmail.com>
Cc: Dmitry Panchenko <dmitry@d-systems.ee>
Cc: Chris Wulff <crwulff@gmail.com>
Cc: Jesus Ramos <jesus-ramos@live.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200914153756.3412156-2-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In UA101 driver, a tasklet is still
used for handling the output URBs. It can be achieved gracefully with
a work queued in the high-prio system workqueue, too.
This patch replaces the tasklet usage in UA101 driver with a simple
work.
Link: https://lore.kernel.org/r/20200903104131.21097-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The tasklet is an old API that should be deprecated, usually can be
converted to another decent API. In USB-audio driver, a tasklet is
still used in MIDI interface code for handling the output byte
stream. It can be achieved gracefully with a work queued in the
high-prio system workqueue.
This patch replaces the tasklet usage in USB-audio driver with a
simple work.
Link: https://lore.kernel.org/r/20200903104131.21097-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Back-merge to apply the tasklet conversion patches that are based
on the already applied tasklet API changes on 5.9-rc4.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Signed-off-by: Romain Perier <romain.perier@gmail.com>
Signed-off-by: Allen Pais <allen.lkml@gmail.com>
Link: https://lore.kernel.org/r/20200902040221.354941-11-allen.lkml@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for unconditionally passing the
struct tasklet_struct pointer to all tasklet
callbacks, switch to using the new tasklet_setup()
and from_tasklet() to pass the tasklet pointer explicitly.
Signed-off-by: Romain Perier <romain.perier@gmail.com>
Signed-off-by: Allen Pais <allen.lkml@gmail.com>
Link: https://lore.kernel.org/r/20200902040221.354941-10-allen.lkml@gmail.com
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>
The commit 51ab5d77dc ("ALSA: usb-audio: Properly match with audio
interface class") converted the quirk entries that have both vid/pid
pair and bInterface fields to match with all those with a new macro
USB_AUDIO_CLASS(). However, it turned out that those are false
conversions; all those (but the unknown KeithMcMillen device) are
actually with vendor-specific interface class, hence the conversions
broke the matching.
This patch corrects those entries to the right one,
USB_DEVICE_VENDOR_SPEC() (and USB_DEVICE() for KeithMcMillen to be
sure), and drop the unused USB_AUDIO_CLASS macro again.
Fixes: 51ab5d77dc ("ALSA: usb-audio: Properly match with audio interface class")
Reported-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200823113251.10175-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct quirk table entries for Lenovo ThinkStation P620, too.
The name and profile strings are now set from a different table, hence
removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If USB autosuspend is enabled, both front and rear panel can no longer
detect jack insertion.
Enable USB remote wakeup, i.e. needs_remote_wakeup = 1, doesn't help
either.
So disable USB autosuspend to prevent missing jack detection event.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200823105854.26950-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes over several drivers, but all are driver-
specific and nothing looks scary. Slightly large changes are seen in
ASoC qcom driver for the bugs that were revealed by the recent ASoC
core change to report the invalid register access errors. Also ASoC
fsl got a slight intensive change for the distortion fix. Others are
only trivial fixes or device-specific quirks.
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Merge tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes over several drivers, but all are driver-
specific and nothing looks scary.
Slightly large changes are seen in ASoC qcom driver for the bugs that
were revealed by the recent ASoC core change to report the invalid
register access errors. Also ASoC fsl got a slight intensive change
for the distortion fix.
Others are only trivial fixes or device-specific quirks"
* tag 'sound-5.9-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits)
ALSA: hda: avoid reset of sdo_limit
ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
ALSA: usb-audio: ignore broken processing/extension unit
ASoC: intel: Fix memleak in sst_media_open
ASoC: wm8994: Avoid attempts to read unreadable registers
ASoC: msm8916-wcd-analog: fix register Interrupt offset
ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
ALSA: hda/realtek: Add model alc298-samsung-headphone
ALSA: usb-audio: Update documentation comment for MS2109 quirk
ALSA: isa: fix spelling mistakes in the comments
ALSA: usb-audio: Add capture support for Saffire 6 (USB 1.1)
ALSA: hda/realtek: Add quirk for Samsung Galaxy Flex Book
ASoC: q6routing: add dummy register read/write function
ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM
ASoC: Make soc_component_read() returning an error code again
ASoC: amd: Replacing component->name with codec_dai->name.
ASoC: fsl: Fix unused variable warning
ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n
ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n
ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n
...
There are a few entries in the quirk table that set the device ID with
USB_DEVICE() macro while having an extra bInterfaceClass field. But
bInterfaceClass field is never checked unless the proper match_flags
is set, so those may match incorrectly with all interfaces.
Introduce another macro to match with the vid/pid pair and the audio
class interface, and apply it to such entries, so that they can match
properly.
Link: https://lore.kernel.org/r/20200817082140.20232-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new macro USB_AUDIO_DEVICE() for the entries matching with
the pid/vid pair and the class/subclass, and remove the open-code.
Link: https://lore.kernel.org/r/20200817082140.20232-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we've added the devices that need vendor/product string renames
or the profile setup into the standard quirk table in quirks-table.h.
This table is imported into the primary USB audio device entry, hence
it's all exported for the probing so that udev and co can take a look
at it. OTOH, for renaming or profile setup, we don't need to expose
those explicit entries because the probe itself follows the standard
way. That said, we're exposing unnecessarily too many entries.
This patch moves such internal quirk entries into the own table, and
reduces the exported device table size. Along with the moving items,
re-arrange the entries in the proper order.
Link: https://lore.kernel.org/r/20200817082140.20232-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have broken extension unit where getting current value
doesn't work. Attempt that once when creating mixer control for it. If
it fails, just ignore it, so that it won't cripple the device entirely
(and/or make the error floods).
Signed-off-by: Tom Yan <tom.ty89@gmail.com>
Link: https://lore.kernel.org/r/5f3abc52.1c69fb81.9cf2.fe91@mx.google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the recent fix addressed the channel swap problem more properly,
update the comment as well.
Fixes: 1b7ecc241a ("ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109")
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200816084431.102151-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same
interface. This was not supported by the composite quirk back in the day
when initial support for this device was added, thus only playback was
enabled until now.
Fixes: 11e424e88b ("ALSA: usb-audio: Add support for Focusrite Saffire 6 USB")
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable.vger.kernel.org>
Link: https://lore.kernel.org/r/20200815002103.29247-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All device-specific small fixes and quirks mostly for usual
suspects, USB-audio and HD-audio.
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Merge tag 'sound-fix-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All device-specific small fixes and quirks mostly for usual suspects,
USB-audio and HD-audio"
* tag 'sound-fix-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: echoaudio: Fix potential Oops in snd_echo_resume()
ALSA: hda/hdmi: Use force connectivity quirk on another HP desktop
ALSA: hda/realtek - Fix unused variable warning
ALSA: hda - reverse the setting value in the micmute_led_set
ALSA: echoaduio: Drop superfluous volatile modifier
ALSA: usb-audio: Disable Lenovo P620 Rear line-in volume control
ALSA: usb-audio: add quirk for Pioneer DDJ-RB
ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109
ALSA: hda - fix the micmute led status for Lenovo ThinkCentre AIO
ALSA: usb-audio: fix overeager device match for MacroSilicon MS2109
ALSA: hda/realtek: Fix pin default on Intel NUC 8 Rugged
ALSA: usb-audio: Creative USB X-Fi Pro SB1095 volume knob support
ALSA: usb-audio: fix spelling mistake "buss" -> "bus"
The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel
line-in claim to support volume control, but it doens't seem to have an
AMP, so when line-in volume lowers below 80, nothing gets recorded
anymore.
Disable the volume control to workaround the issue.
Fixes: f8c11eb7da ("ALSA: usb-audio: Add support for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200810133108.31580-1-kai.heng.feng@canonical.com
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>
This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
* API cleanups and conversions to the unified mute_stream() call
* Simplify I/O helper functions
* Use helper macros to retrieve RTD from substreams
ASoC drivers:
* Lots of fixes and cleanups in Intel ASoC drivers
* Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
* Minor code refacotring for SG-buffer handling
HD-audio:
* Generalization of mute-LED handling with LED classdev
* Intel silent stream support for HDMI
* Device-specific fixes: CA0132, Loongson-3
Others:
* Usual USB- and HD-audio quirks for various devices
* Fixes for echoaudio DMA position handling
* Various documents and trivial fixes for sparse warnings
* Conversion to adapt inclusive terms
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Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
- API cleanups and conversions to the unified mute_stream() call
- Simplify I/O helper functions
- Use helper macros to retrieve RTD from substreams
ASoC drivers:
- Lots of fixes and cleanups in Intel ASoC drivers
- Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
- Minor code refacotring for SG-buffer handling
HD-audio:
- Generalization of mute-LED handling with LED classdev
- Intel silent stream support for HDMI
- Device-specific fixes: CA0132, Loongson-3
Others:
- Usual USB- and HD-audio quirks for various devices
- Fixes for echoaudio DMA position handling
- Various documents and trivial fixes for sparse warnings
- Conversion to adopt inclusive terms"
* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
ALSA: pci: delete repeated words in comments
ALSA: isa: delete repeated words in comments
ALSA: hda/tegra: Add 100us dma stop delay
ALSA: hda: Add dma stop delay variable
ASoC: hda/tegra: Set buffer alignment to 128 bytes
ALSA: seq: oss: Serialize ioctls
ALSA: hda/hdmi: Add quirk to force connectivity
ALSA: usb-audio: add startech usb audio dock name
ALSA: usb-audio: Add support for Lenovo ThinkStation P620
Revert "ALSA: hda: call runtime_allow() for all hda controllers"
ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
ALSA: docs: fix typo
ALSA: doc: use correct config variable name
ASoC: core: Two step component registration
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
...
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Signed-off-by: Mirko Dietrich <buzz@l4m1.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a spelling mistake in a usb_audio_dbg debug message. Also
replace "param" with "parameter". Fix these.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200806105134.46447-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name
and shows up currently as "USB Sound Device" in ALSA.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20200804010616.3399256-1-cujomalainey@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with
two USB audio cards.
USB device (17aa:104d) provides functionality for Internal Speaker and
Front Headset. It's UAC v2, so it supports insertion control (jack
detection). However, when trying to get the connector status of the
speaker, an error occurs:
[ 5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0
Since the insertion control works perfectly for the headset, the error
for speaker is probably casued by connecting internally. So let's relax
the error for a bit if it's a speaker, and always reports it's connected.
USB device (17aa:1046) is for rear Line-in, Line-out and Microphone.
The insertion control works for all three jacks. However, there's an
Function Unit that doesn't work:
[ 5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4
[ 5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19)
So turn off the FU to avoid the error.
Also, add specific card name for both devices, so userspace can easily
indentify both cards.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200803142612.17156-1-kai.heng.feng@canonical.com
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>
Follow the recent inclusive terminology guidelines and replace the
word "slave" in vmaster API. I chose the word "follower" at this time
since it seems fitting for the purpose.
Note that the word "master" is kept in API, since it refers rather to
audio master volume control.
Also, while we're at it, a typo in comments is corrected, too.
Link: https://lore.kernel.org/r/20200717154517.27599-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200719151705.59624-1-grandmaster@al2klimov.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using uninitialized_var() is dangerous as it papers over real bugs[1]
(or can in the future), and suppresses unrelated compiler warnings
(e.g. "unused variable"). If the compiler thinks it is uninitialized,
either simply initialize the variable or make compiler changes.
In preparation for removing[2] the[3] macro[4], remove all remaining
needless uses with the following script:
git grep '\buninitialized_var\b' | cut -d: -f1 | sort -u | \
xargs perl -pi -e \
's/\buninitialized_var\(([^\)]+)\)/\1/g;
s:\s*/\* (GCC be quiet|to make compiler happy) \*/$::g;'
drivers/video/fbdev/riva/riva_hw.c was manually tweaked to avoid
pathological white-space.
No outstanding warnings were found building allmodconfig with GCC 9.3.0
for x86_64, i386, arm64, arm, powerpc, powerpc64le, s390x, mips, sparc64,
alpha, and m68k.
[1] https://lore.kernel.org/lkml/20200603174714.192027-1-glider@google.com/
[2] https://lore.kernel.org/lkml/CA+55aFw+Vbj0i=1TGqCR5vQkCzWJ0QxK6CernOU6eedsudAixw@mail.gmail.com/
[3] https://lore.kernel.org/lkml/CA+55aFwgbgqhbp1fkxvRKEpzyR5J8n1vKT1VZdz9knmPuXhOeg@mail.gmail.com/
[4] https://lore.kernel.org/lkml/CA+55aFz2500WfbKXAx8s67wrm9=yVJu65TpLgN_ybYNv0VEOKA@mail.gmail.com/
Reviewed-by: Leon Romanovsky <leonro@mellanox.com> # drivers/infiniband and mlx4/mlx5
Acked-by: Jason Gunthorpe <jgg@mellanox.com> # IB
Acked-by: Kalle Valo <kvalo@codeaurora.org> # wireless drivers
Reviewed-by: Chao Yu <yuchao0@huawei.com> # erofs
Signed-off-by: Kees Cook <keescook@chromium.org>
Follow the recent inclusive terminology guidelines and replace the
word "blacklist" appropriately.
Only a comment fix, no functional changes.
Link: https://lore.kernel.org/r/20200714172631.25371-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add hw monitor volume control for POD HD500. The same change may
work for HD500X but I don't have it to test.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Link: https://lore.kernel.org/r/20200713152852.65832-1-anarsoul@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB MIDI driver has an error recovery mechanism to resubmit the URB in
the delayed timer handler, and this may race with the standard start /
stop operations. Although both start and stop operations themselves
don't race with each other due to the umidi->mutex protection, but
this isn't applied to the timer handler.
For fixing this potential race, the following changes are applied:
- Since the timer handler can't use the mutex, we apply the
umidi->disc_lock protection at each input stream URB submission;
this also needs to change the GFP flag to GFP_ATOMIC
- Add a check of the URB refcount and skip if already submitted
- Move the timer cancel call at disconnection to the beginning of the
procedure; this assures the in-flight timer handler is gone properly
before killing all pending URBs
Reported-by: syzbot+0f4ecfe6a2c322c81728@syzkaller.appspotmail.com
Reported-by: syzbot+5f1d24c49c1d2c427497@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200710160656.16819-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently syzkaller reported a UAF in LINE6 driver, and it's likely
because we call cancel_delayed_work() at the disconnect callback
instead of cancel_delayed_work_sync(). Let's use the correct one
instead.
Reported-by: syzbot+145012a46658ac00fc9e@syzkaller.appspotmail.com
Suggested-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hlfjr4gio.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
LINE6 drivers create stream URBs with a fixed pipe without checking
its validity, and this may lead to a kernel WARNING at the submission
when a malformed USB descriptor is passed.
For avoiding the kernel warning, perform the similar sanity checks for
each pipe type at creating a URB.
Reported-by: syzbot+c190f6858a04ea7fbc52@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hv9iv4hq8.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few places (except for ASoC) are left unconverted for the new
fallthrough pseudo keyword. Now replace them all.
Reviewed-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Link: https://lore.kernel.org/r/20200709111750.8337-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix W=1 warning. Variables are declared in a header file included from
multiple C files, replace by #defines as suggested by Takashi
sound/usb/line6/driver.h:70:18: warning: ‘SYSEX_EXTRA_SIZE’ defined
but not used [-Wunused-const-variable=]
70 | static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4;
| ^~~~~~~~~~~~~~~~
sound/usb/line6/driver.h:69:18: warning: ‘SYSEX_DATA_OFS’ defined but
not used [-Wunused-const-variable=]
69 | static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3;
| ^~~~~~~~~~~~~~
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200707184924.96291-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB Audio analyzer RTX6001 uses the same implicit feedback quirk
as other XMOS-based devices.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/822f0f20-1886-6884-a6b2-d11c685cbafa@ivitera.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit f0bd62b640 ("ALSA: usb-audio: Improve frames size computation")
introduced a regression for devices which have playback endpoints with
bInterval > 1. Fix this by taking ep->datainterval into account.
Note that frame and fps are actually mean packet and packets per second
in the code introduces by the mentioned commit. This will be fixed in a
follow-up patch.
Fixes: f0bd62b640 ("ALSA: usb-audio: Improve frames size computation")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208353
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200629025934.154288-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element. A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members. This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.
This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.
Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com
Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051)
need a tiny delay after each class compliant request.
Otherwise the device might not be able to be recognized each times.
Signed-off-by: Chihhao Chen <chihhao.chen@mediatek.com>
Signed-off-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/1592910203-24035-1-git-send-email-macpaul.lin@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:0x16ea) 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: Christoffer Nielsen <cn@obviux.dk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAOtG2YHOM3zy+ed9KS-J4HkZo_QGzcUG9MigSp4e4_-13r6B=Q@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running. In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer. Since
the PCM stream gets closed, this may lead to use-after-free.
This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.
Fixes: 10ce77e481 ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the Line6 devices, the Rode Rodecaster Pro does not support
UAC2_CS_RANGE and only supports a sample rate of 48 kHz.
Tested against a Rode Rodecaster Pro.
Tested-by: Christopher Swenson <swenson@swenson.io>
Signed-off-by: Christopher Swenson <swenson@swenson.io>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/ebdb9e72-9649-0b5e-b9b9-d757dbf26927@swenson.io
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix error "clock source 41 is not valid, cannot use"
[] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00
[] New USB device strings: Mfr=1, Product=2, SerialNumber=0
[] Product: DCD-1500RE
[] Manufacturer: D & M Holdings Inc.
[]
[] clock source 41 is not valid, cannot use
[] usbcore: registered new interface driver snd-usb-audio
Signed-off-by: Yick W. Tse <y_w_tse@yahoo.com.hk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1373857985.210365.1592048406997@mail.yahoo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the
audio interface's clock. Tested on an SSL2+.
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200612111807.dgnig6rwhmsl2bod@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently USB-audio driver manages the auto-pm of the primary
interface although a card may consist of multiple interfaces.
This may leave the secondary and other interfaces left running
unnecessarily after the auto-suspend.
This patch allows the driver managing the auto-pm of all bundled
interfaces per card. The chip->pm_intf field is extended as
chip->intf[] to contain the array of assigned interfaces, and the
runtime-PM is performed to all those interfaces.
Tested-by: Macpaul Lin <macpaul.lin@mediatek.com>
Link: https://lore.kernel.org/r/20200605064117.28504-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the open-code with the new QUIRK_DEVICE_PROFILE() macro for
simplicity.
Fixes: 0c5086f569 ("ALSA: usb-audio: Add vendor, product and profile name for HP Thunderbolt Dock")
Link: https://lore.kernel.org/r/20200608071513.570-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP Thunderbolt Dock has two separate USB devices, one is for speaker
and one is for headset. Add names for them so userspace can apply UCM
settings.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200608062630.10806-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a USB-audio interface gets runtime-suspended via auto-pm feature,
the driver suspends all functionality and increment
chip->num_suspended_intf. Later on, when the system gets suspended to
S3, the driver increments chip->num_suspended_intf again, skips the
device changes, and sets the card power state to
SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from
S3, the resume callback decrements chip->num_suspended_intf. Since
this refcount is still not zero (it's been runtime-suspended), the
whole resume is skipped. But there is a small pitfall here.
The problem is that the driver doesn't restore the card power state
after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So,
even after the system resume finishes, the card instance still appears
as if it were system-suspended, and this confuses many ioctl accesses
that are blocked unexpectedly.
In details, we have two issues behind the scene: one is that the card
power state is changed only when the refcount becomes zero, and
another is that the prior auto-suspend check is kept in a boolean
flag. Although the latter problem is almost negligible since the
auto-pm feature is imposed only on the primary interface, but this can
be a potential problem on the devices with multiple interfaces.
This patch addresses those issues by the following:
- Replace chip->autosuspended boolean flag with chip->system_suspend
counter
- At the first system-suspend, chip->num_suspended_intf is recorded to
chip->system_suspend
- At system-resume, the card power state is restored when the
chip->num_suspended_intf refcount reaches to chip->system_suspend,
i.e. the state returns to the auto-suspended
Also, the patch fixes yet another hidden problem by the code
refactoring along with the fixes above: namely, when some resume
procedure failed, the driver left chip->num_suspended_intf that was
already decreased, and it might lead to the refcount unbalance.
In the new code, the refcount decrement is done after the whole resume
procedure, and the problem is avoided as well.
Fixes: 0662292aec ("ALSA: usb-audio: Handle normal and auto-suspend equally")
Reported-and-tested-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200603153709.6293-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pioneer DJ DJM-900NXS2 is a widely used DJ mixer with 2 audio USB
interfaces. Both have a MIDI controller, 10 playback and 12 capture
channels. Audio endpoints are vendor-specific and 3 files need to be
patched. All playback and capture channels work fine with all supported
sample rates (44.1k, 48k, 96k). Patches are attached.
Signed-off-by: Dmitry Panchenko <dmitry@d-systems.ee>
Link: https://lore.kernel.org/r/48ab19ff-3303-9bf8-ed0e-bcb31d8537eb@d-systems.ee
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As reported by kbuild test robot, mixer quirks for RME Babyface
Pro used plain integer instead of NULL.
Fixes: 3e8f3bd047 ("ALSA: usb-audio: RME Babyface Pro mixer patch")
Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Reported-by: kbuild test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20200529173248.zzawijfvw73kzjxt@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices,
a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC
(USB ID 0414:a000). The latter serves solely for the headphone output
on the front panel while the former serves for the rest I/Os (mostly
for the I/Os in the rear panel but also including the front mic).
Both chips do work more or less with the unmodified USB-audio driver,
but there are a few glitches. The ALC1220-VB returns an error for an
inquiry to some jacks, as already seen on other TRX40-based mobos.
However this machine has a slightly incompatible configuration, hence
the existing mapping cannot be used as is.
Meanwhile the ESS chip seems working without any quirk. But since
both audio devices don't provide any specific names, both cards appear
as "USB-Audio", and it's quite confusing for users.
This patch is an attempt to overcome those issues:
- The specific mapping table for ALC1220-VB is provided, reducing the
non-working nodes and renaming the badly chosen controls.
The connector map isn't needed here unlike other TRX40 quirks.
- For both USB IDs (0414:a000 and 0414:a001), provide specific card
name strings, so that user-space can identify more easily; and more
importantly, UCM profile can be applied to each.
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526082810.29506-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Asus USB DAC is a USB type-C audio dongle for connecting to
the headset and headphone. The volume minimum value -23040 which
is 0xa600 in hexadecimal with the resolution value 1 indicates
this should be endianness issue caused by the firmware bug. Add
a volume quirk to fix the volume control problem.
Also fixes this warning:
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062613.55401-1-chiu@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This is another TRX40 based motherboard with ALC1220-VB USB-audio
that requires a static mapping table.
This motherboard also has a PCI device which advertises no codecs. The
PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD
vendor ID, don't blacklist for now in case other boards have a working
audio device with the same ssid.
alsa-info.sh report for this board:
http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c
Signed-off-by: Andrew Oakley <andrew@adoakley.name>
Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name
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>
Apparently interface 1 is control interface akin to HD500X,
setting LINE6_CAP_CONTROL and choosing it as ctrl_if fixes
audio playback on POD HD500.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200425201115.3430-1-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.
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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>
The Scarlett gen2 mixer quirk code defines a few record types to
communicate via USB hub, and those must be all little-endian.
This patch changes the field types to LE to annotate endianess
properly. It also fixes the incorrect usage of leXX_to_cpu() in a
couple of places, which was caught by sparse after this change.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200201080530.22390-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I overlooked that some fields are words and need the converts from
LE in the recently added USB descriptor validation code.
This patch fixes those with the proper macro usages.
Fixes: 57f8770620 ("ALSA: usb-audio: More validations of descriptor units")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200201080530.22390-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With firmware 2.82 Line6 changed the usb id of some of the Helix
devices but the quirks is still needed.
Add it to the quirk list for line6 helix family of devices.
Thanks to Jens for pointing out the missing ids.
Signed-off-by: Nicola Lunghi <nick83ola@gmail.com>
Link: https://lore.kernel.org/r/20200125150917.5040-1-nick83ola@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes crackling sound during playback.
Further note: MOTU is known for reusing Product IDs for different
devices or different generations of the device (e.g. MicroBook
I/II/IIc shares a single Product ID). This patch was only tested with
M4 audio interface, but the same Product ID is also used by M2. Hope
it will work for M2 as well.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200115151358.56672-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was
checking for there to be at least one altsetting but then went on to
access the second one, which may not exist.
This could lead to random slab data being used to initialise the sync
endpoint in snd_usb_add_endpoint().
Fixes: c75a8a7ae5 ("ALSA: snd-usb: add support for implicit feedback")
Fixes: ca10a7ebdf ("ALSA: usb-audio: FT C400 sync playback EP to capture EP")
Fixes: 5e35dc0338 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204")
Fixes: 17f08b0d9a ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II")
Fixes: 103e962564 ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk")
Cc: stable <stable@vger.kernel.org> # 3.5
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20200114083953.1106-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add delay to make sure that audio urbs are not sent too early.
Otherwise the device hangs. Windows driver makes ~2s delay, so use
about the same time delay value.
snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which
is an overkill. Thus a quirk that is called only once is implemented.
Also send two vendor-specific control messages before and after
the delay. This behaviour is blindly copied from the Windows driver.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200112102358.18085-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
GCC reports the following warning with W=1
sound/usb/mixer_quirks.c: In function ‘snd_microii_controls_create’:
sound/usb/mixer_quirks.c:1694:2: warning: ‘static’ is not at beginning
of declaration [-Wold-style-declaration]
1694 | const static usb_mixer_elem_resume_func_t resume_funcs[] = {
| ^~~~~
Move static to the beginning of declaration
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200111214736.3002-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply const prefix to each possible place: the string array and the
parameter tables and callers.
Just for minor optimization and no functional changes.
Link: https://lore.kernel.org/r/20200105144823.29547-23-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply const prefix to each possible place: the rate table, the
controller tables, and the key tables.
Just for minor optimization and no functional changes.
Link: https://lore.kernel.org/r/20200105144823.29547-13-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply const prefix to the remaining places: the static table for the
unit information, the mixer maps, the validator tables, etc.
Just for minor optimization and no functional changes.
Link: https://lore.kernel.org/r/20200105144823.29547-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For consistency reason, make all hex numbers with lower alphabets for
USB ID entries. It improves grep-ability and reduces careless
mistakes.
Link: https://lore.kernel.org/r/20200105081900.21870-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk entries used in us122l and usx2y drivers can be declared as
const as they are read-only.
There should be no functional changes by this patch.
Link: https://lore.kernel.org/r/20200103081714.9560-52-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_ratden definitions used in line6 drivers are all read-only, so
they can be marked as const.
There should be no functional changes by this patch.
Link: https://lore.kernel.org/r/20200103081714.9560-51-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of snd_kcontrol_new definitions are read-only and passed as-is.
Let's declare them as const for further optimization.
There should be no functional changes by this patch.
Link: https://lore.kernel.org/r/20200103081714.9560-42-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we may declare const for snd_device_ops definitions, so let's do
it for optimization.
There should be no functional changes by this patch.
Link: https://lore.kernel.org/r/20200103081714.9560-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of snd_pcm_hardware definitions are just copied to another object
as-is, hence we can define them as const for further optimization.
There should be no functional changes by this patch.
Link: https://lore.kernel.org/r/20200103081714.9560-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Corsair Virtuoso RGB Wireless is a USB headset with a mic and a
sidetone feature. Label its mixer appropriately instead of all
"Headset", so that applications such as Pulseaudio don't just move
the sidetone control when they intend the main Headset control.
Signed-off-by: Chris Boyle <chris@boyle.name>
Link: https://lore.kernel.org/r/20191227094053.GA12167@nova.chris.boyle.name
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure to check the return value of usb_altnum_to_altsetting() to
avoid dereferencing a NULL pointer when the requested alternate settings
is missing.
The format altsetting number may come from a quirk table and there does
not seem to be any other validation of it (the corresponding index is
checked however).
Fixes: b099b9693d ("ALSA: usb-audio: Avoid superfluous usb_set_interface() calls")
Cc: stable <stable@vger.kernel.org> # 4.18
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20191220093134.1248-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Taking the 5.5 devel branch back into the main devel branch.
A USB-audio fix needs to be adjusted to adapt the changes that have
been formerly applied for stop_sync.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we found the headset-mic on the Dell Dock WD19 doesn't work
anymore after s3 (s2i or deep), this problem could be workarounded by
closing (pcm_close) the app and then reopening (pcm_open) the app, so
this bug is not easy to be detected by users.
When problem happens, retire_capture_urb() could still be called
periodically, but the size of captured data is always 0, it could be
a firmware bug on the dock. Anyway I found after resuming, the
snd_usb_pcm_prepare() will be called, and if we forcibly run
set_format() to set the interface and its endpoint, the capture
size will be normal again. This problem and workaound also apply to
playback.
To fix it in the kernel, add a quirk to let set_format() run
forcibly once after resume.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191218132650.6303-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clang warns:
../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation;
statement is not part of the previous 'if' [-Wmisleading-indentation]
info->version = USX2Y_DRIVER_VERSION;
^
../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here
if (us428->chip_status & USX2Y_STAT_CHIP_INIT)
^
1 warning generated.
This warning occurs because there is a space before the tab on this
line. Remove it so that the indentation is consistent with the Linux
kernel coding style and clang no longer warns.
This was introduced before the beginning of git history so no fixes tag.
Link: https://github.com/ClangBuiltLinux/linux/issues/831
Signed-off-by: Nathan Chancellor <natechancellor@gmail.com>
Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the driver with the new managed buffer allocation API.
The superfluous snd_pcm_lib_malloc_pages() and
snd_pcm_lib_free_pages() calls are dropped.
Link: https://lore.kernel.org/r/20191209094943.14984-71-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the driver with the new managed buffer allocation API.
The superfluous snd_pcm_lib_malloc_pages() and
snd_pcm_lib_free_pages() calls are dropped.
Link: https://lore.kernel.org/r/20191209094943.14984-70-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the drivers with the new managed buffer allocation API.
The superfluous snd_pcm_lib_malloc_pages() and
snd_pcm_lib_free_pages() calls are dropped.
Link: https://lore.kernel.org/r/20191209094943.14984-68-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the driver with the new managed buffer allocation API.
The hw_params callback became superfluous and dropped.
The hw_free callback still remains because of the substream
deactivation sync call.
Link: https://lore.kernel.org/r/20191209094943.14984-66-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage.
This patch introduces this functionality as to be seen in the mac
or windows scarlett control.
The correct address could already be found in the dump info, but was
never used. Without this patch inputs 3/4 are quite unusable else.
Signed-off-by: Jens Verwiebe <info@jensverwiebe.de>
Link: https://lore.kernel.org/r/384d65cd-5e87-91eb-9fc3-e57226f534c6@jensverwiebe.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_mixer_controls_badd() that parses UAC3 BADD profiles misses a
NULL check for the given interfaces. When a malformed USB descriptor
is passed, this may lead to an Oops, as spotted by syzkaller.
Skip the iteration if the interface doesn't exist for avoiding the
crash.
Fixes: 17156f23e9 ("ALSA: usb: add UAC3 BADD profiles support")
Reported-by: syzbot+a36ab65c6653d7ccdd62@syzkaller.appspotmail.com
Suggested-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191122112840.24797-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The s6i6_gen2_info.ports[] array had the Mixer and PCM port type
entries in the wrong place. Use designators to explicitly specify the
array elements being set.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Tested-by: Alex Fellows <alex.fellows@gmail.com>
Tested-by: Markus Schroetter <project.m.schroetter@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191110134356.GA31589@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unit descriptor validation may lead to a probe error when the
device provides a buggy descriptor or the validator detected
incorrectly. For identifying such an error and band-aiding, give a
new module option, skip_validation. With this option, the driver
ignores the validation errors with the hexdump of the unit
descriptor, so we can check it in a bit more details.
Link: https://lore.kernel.org/r/20191114165613.7422-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced unit descriptor validation had some bug for
processing and extension units, it counts a bControlSize byte twice so
it expected a bigger size than it should have been. This seems
resulting in a probe error on a few devices.
Fix the calculation for proper checks of PU and EU.
Fixes: 57f8770620 ("ALSA: usb-audio: More validations of descriptor units")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191114165613.7422-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 60849562a5 ("ALSA: usb-audio: Fix possible NULL
dereference at create_yamaha_midi_quirk()") added NULL checks in
create_yamaha_midi_quirk(), but there was an overlook. The code
allows one of either injd or outjd is NULL, but the second if check
made returning -ENODEV if any of them is NULL. Fix it in a proper
form.
Fixes: 60849562a5 ("ALSA: usb-audio: Fix possible NULL dereference at create_yamaha_midi_quirk()")
Reported-by: Pavel Machek <pavel@denx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191113111259.24123-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While output urb's snd_complete_urb() is executing, calling
prepare_outbound_urb() may cause endpoint stopped before
prepare_outbound_urb() returns and result in next urb submitted
to stopped endpoint. usb-audio driver cannot re-use it afterwards as
the urb is still hold by usb stack.
This change checks EP_FLAG_RUNNING flag after prepare_outbound_urb() again
to let snd_complete_urb() know the endpoint already stopped and does not
submit next urb. Below kind of error will be fixed:
[ 213.153103] usb 1-2: timeout: still 1 active urbs on EP #1
[ 213.164121] usb 1-2: cannot submit urb 0, error -16: unknown error
Signed-off-by: Henry Lin <henryl@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191113021420.13377-1-henryl@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A check of the return value from get_cur_mix_raw() is missing at the
resolution test code in get_min_max_with_quirks(), which may leave the
variable untouched, leading to a random uninitialized value, as
detected by syzkaller fuzzer.
Add the missing return error check for fixing that.
Reported-and-tested-by: syzbot+abe1ab7afc62c6bb6377@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191109181658.30368-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change (*) in the ALSA memalloc core allows us to drop the
special vmalloc-specific allocation and page handling. This patch
coverts to the common code.
(*) 1fe7f397cf: ALSA: memalloc: Add vmalloc buffer allocation
support
7e8edae39f: ALSA: pcm: Handle special page mapping in the
default mmap handler
Link: https://lore.kernel.org/r/20191105151856.10785-15-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change (*) in the ALSA memalloc core allows us to drop the
special vmalloc-specific allocation and page handling. This patch
coverts to the common code.
(*) 1fe7f397cf: ALSA: memalloc: Add vmalloc buffer allocation
support
7e8edae39f: ALSA: pcm: Handle special page mapping in the
default mmap handler
Link: https://lore.kernel.org/r/20191105151856.10785-14-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change (*) in the ALSA memalloc core allows us to drop the
special vmalloc-specific allocation and page handling. This patch
coverts to the common code.
(*) 1fe7f397cf: ALSA: memalloc: Add vmalloc buffer allocation
support
7e8edae39f: ALSA: pcm: Handle special page mapping in the
default mmap handler
Link: https://lore.kernel.org/r/20191105151856.10785-13-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change (*) in the ALSA memalloc core allows us to drop the
special vmalloc-specific allocation and page handling. This patch
coverts to the common code.
(*) 1fe7f397cf: ALSA: memalloc: Add vmalloc buffer allocation
support
7e8edae39f: ALSA: pcm: Handle special page mapping in the
default mmap handler
Link: https://lore.kernel.org/r/20191105151856.10785-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change (*) in the ALSA memalloc core allows us to drop the
special vmalloc-specific allocation and page handling. This patch
coverts to the common code.
(*) 1fe7f397cf: ALSA: memalloc: Add vmalloc buffer allocation
support
7e8edae39f: ALSA: pcm: Handle special page mapping in the
default mmap handler
Also, since the SG-buffer-specific PCM ops becomes identical with the
normal PCM ops, unify them again to the single ops, too.
Link: https://lore.kernel.org/r/20191105151856.10785-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change (commit 08422d2c55: "ALSA: memalloc: Allow NULL
device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation
helper accepting NULL as the device pointer for the default usage.
Drop the snd_dma_continuous_data() usage that became superfluous from
the callers.
Link: https://lore.kernel.org/r/20191105151856.10785-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use true/false for bool type return in uac_clock_source_is_valid().
Signed-off-by: Saurav Girepunje <saurav.girepunje@gmail.com>
Link: https://lore.kernel.org/r/20191029175200.GA7320@saurav
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced USB-audio descriptor validator had a stupid
copy&paste error that may lead to an unexpected overlook of too short
descriptors for processing and extension units. It's likely the cause
of the report triggered by syzkaller fuzzer. Let's fix it.
Fixes: 57f8770620 ("ALSA: usb-audio: More validations of descriptor units")
Reported-by: syzbot+0620f79a1978b1133fd7@syzkaller.appspotmail.com
Link: https://lore.kernel.org/r/s5hsgnkdbsl.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adjust indentation from spaces to tab (+optional two spaces) as in
coding style with command like:
$ sed -e 's/^ /\t/' -i */Kconfig
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191004144931.3851-1-krzk@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EVGA NU Audio is actually a USB audio device on a PCIexpress card,
with it's own USB controller. It supports both PCM and DSD.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190924071143.30911-1-jussi@sonarnerd.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID ID for Hiby portable players family with
native DSD playback support.
Signed-off-by: Ilya Pshonkin <sudokamikaze@protonmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190917074937.157802-1-ilya.pshonkin@netforce.ua
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Oppo has issued firmware updates that change alt setting used for DSD
support. However, these devices seem to support auto-detection, so
support is moved from explicit whitelisting to auto-detection.
Also Rotel devices have USB interfaces that support DSD with
auto-detection.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DSD support auto-detection for newer Playback Designs devices. Older
device generations have a different USB interface implementation.
Keep the auto-detection VID whitelist sorted.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We recently cleaned up the error handling in commit 52c3e317a8 ("ALSA:
usb-audio: Unify the release of usb_mixer_elem_info objects") but
accidentally left this stray return.
Fixes: 52c3e317a8 ("ALSA: usb-audio: Unify the release of usb_mixer_elem_info objects")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous addition of descriptor validation may lead to a NULL
dereference at create_yamaha_midi_quirk() when either injd or outjd is
NULL. Add proper non-NULL checks.
Fixes: 57f8770620 ("ALSA: usb-audio: More validations of descriptor units")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The primary changes in this patch are cleanups of __check_input_term()
and move to a non-nested switch-case block by evaluating the pair of
UAC version and the unit type, as we've done for parse_audio_unit().
Also each parser is split into the function for readability.
Now, a slight behavior change by this cleanup is the handling of
processing and extension units. Formerly we've dealt with them
differently between UAC1/2 and UAC3; the latter returns an error if no
input sources are available, while the former continues to parse.
In this patch, unify the behavior in all cases: when input sources are
available, it parses recursively, then override the type and the id,
as well as channel information if not provided yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull USB validation patches. It's based on the latest 5.3 development
branch, so we shall catch up the whole things.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we got the more comprehensive validation code for USB-audio
descriptors, the check of overflow in each descriptor unit parser
became superfluous. Drop some of the obvious cases.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the direct kfree() calls, introduce a new local helper to
release the usb_mixer_elem_info object. This will be extended to do
more than a single kfree() in the later patches.
Also, use the standard goto instead of multiple calls in
parse_audio_selector_unit() error paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor code refactoring by combining the UAC version and the type in
the switch-case flow, so that we reduce the indentation and
redundancy. One good bonus is that the duplicated definition of the
same type value (e.g. UAC2_EFFECT_UNIT) can be handled more cleanly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new helper to validate each audio descriptor unit before
and check the unit before actually accessing it. This should harden
against the OOB access cases with malformed descriptors that have been
recently frequently reported by fuzzers.
The existing descriptor checks are still kept although they become
superfluous after this patch. They'll be cleaned up eventually
later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a
variable size depending on both input and output pins. Its size is to
fit with input * output bits. The problem is that the input size
can't be determined simply from the unit descriptor itself but it
needs to parse the whole connected sources. Although the
uac_mixer_unit_get_channels() tries to check some possible overflow of
this bitmap, it's incomplete due to the lack of the evaluation of
input pins.
For covering possible overflows, this patch adds the bitmap overflow
check in the loop of input pins in parse_audio_mixer_unit().
Fixes: 0bfe5e434e ("ALSA: usb-audio: Check mixer unit descriptors more strictly")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to release the allocated object at the early error path in
line6_init_pcm(). For addressing it, slightly shuffle the code so
that the PCM destructor (pcm->private_free) is assigned properly
before all error paths.
Fixes: 3450121997 ("ALSA: line6: Fix write on zero-sized buffer")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk function snd_emuusb_set_samplerate() has a NULL check for
the mixer element, but this is useless in the current code. It used
to be a check against mixer->id_elems[unitid] but it was changed later
to the value after mixer_eleme_list_to_info() which is always non-NULL
due to the container_of() usage.
This patch fixes the check before the conversion.
While we're at it, correct a typo in the comment in the function,
too.
Fixes: 8c558076c7 ("ALSA: usb-audio: Clean up mixer element list traverse")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Behringer UFX1604 requires the similar quirk to apply implicit fb like
another Behringer model UFX1204 in order to fix the noisy playback.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
`check_input_term` recursively calls itself with input from
device side (e.g., uac_input_terminal_descriptor.bCSourceID)
as argument (id). In `check_input_term`, if `check_input_term`
is called with the same `id` argument as the caller, it triggers
endless recursive call, resulting kernel space stack overflow.
This patch fixes the bug by adding a bitmap to `struct mixer_build`
to keep track of the checked ids and stop the execution if some id
has been checked (similar to how parse_audio_unit handles unitid
argument).
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The `uac_mixer_unit_descriptor` shown as below is read from the
device side. In `parse_audio_mixer_unit`, `baSourceID` field is
accessed from index 0 to `bNrInPins` - 1, the current implementation
assumes that descriptor is always valid (the length of descriptor
is no shorter than 5 + `bNrInPins`). If a descriptor read from
the device side is invalid, it may trigger out-of-bound memory
access.
```
struct uac_mixer_unit_descriptor {
__u8 bLength;
__u8 bDescriptorType;
__u8 bDescriptorSubtype;
__u8 bUnitID;
__u8 bNrInPins;
__u8 baSourceID[];
}
```
This patch fixes the bug by add a sanity check on the length of
the descriptor.
Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later
on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In
hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through
kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and
'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs.
Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails.
To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'.
Fixes: a91c3fb2f8 ("Add M2Tech hiFace USB-SPDIF driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Pioneer DDJ-SX3 is a plain 12 32bit channel out and 10 channel in
PCM/midi controller. The PCM part is "vendor specific".
It needs the "ignore invalid bsynchaddress" patch as it uses 0 for that.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Linux kernel assumes that get_endpoint(alts,0) and
get_endpoint(alts,1) are eachothers feedback endpoints.
To reassure that validity it will test bsynchaddress to comply with that
assumption. But if the bsyncaddress is 0 (invalid), it will flag that as
a wrong assumption and return an error.
Fix: Skip the test if bSynchAddress is 0.
Note: those with a valid bSynchAddress should have a code quirck added.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some cards have alternate setting with non-PCM format as the first
altsetting in the interface descriptors. This confuses userspace, since
alsa-lib uses device 0 by default. So lets parse interfaces in two steps:
1. Parse altsettings with PCM formats.
2. Parse altsettings with non-PCM formats.
This fixes at least following cards:
- Audinst HUD-mx2
- Audinst HUD-mini
[ Adapted to 5.3 kernel by tiwai ]
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many open code for releasing audioformat object.
Provide a unified helper and call it from the all places.
Only a cleanup, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is
allocated through kzalloc() before the execution goto 'found_clock'.
However, this structure is not deallocated if the memory allocation for
'pd' fails, leading to a memory leak bug.
To fix the above issue, free 'fp->chmap' before returning NULL.
Fixes: 7edf3b5e6a ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
syzbot found the following crash on:
general protection fault: 0000 [#1] SMP KASAN
RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75
Call Trace:
snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb sound/usb/quirks.c:1007
snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline]
snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280
usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576
usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361
really_probe+0x281/0x650 drivers/base/dd.c:548
....
It was introduced in commit 801ebf1043 for checking pipe and endpoint
types. It is fixed by adding a check of the ep pointer in question.
BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2
Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2@syzkaller.appspotmail.com>
Fixes: 801ebf1043 ("ALSA: usb-audio: Sanity checks for each pipe and EP types")
Cc: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Hillf Danton <hdanton@sina.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2
audio interfaces. Although the interfaces are USB compliant,
additional input/output level controls and hardware routing/mixing
functionality are available using proprietary USB requests.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
