The quirk entry for Focusrite Saffire 6 had no proper ep_idx for the
capture endpoint, and this confused the driver, resulting in the
broken sound. This patch adds the missing ep_idx in the entry.
While we are at it, a couple of other entries (for Digidesign MBox and
MOTU MicroBook II) seem to have the same problem, and those are
covered as well.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: André Kapelrud <a.kapelrud@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220521065325.426-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Maris reported that TEAC UD-501 (0644:8043) doesn't work with the
typical "clock source 41 is not valid, cannot use" errors on the
recent kernels. The currently known workaround so far is to restore
(partially) what we've done unconditionally at the clock setup;
namely, re-setup the USB interface immediately after the clock is
changed. This patch re-introduces the behavior conditionally for TEAC
devices.
Further notes:
- The USB interface shall be set later in
snd_usb_endpoint_configure(), but this seems to be too late.
- Even calling usb_set_interface() right after
sne_usb_init_sample_rate() doesn't help; so this must be related
with the clock validation, too.
- The device may still spew the "clock source 41 is not valid" error
at the first clock setup. This seems happening at the very first
try of clock setup, but it disappears at later attempts.
The error is likely harmless because the driver retries the clock
setup (such an error is more or less expected on some devices).
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Reported-and-tested-by: Maris Abele <maris7abele@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220521064627.29292-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix following coccicheck error:
./sound/usb/endpoint.c:1671:8-10: ERROR: reference preceded by free on line 1671.
Here should be 'cp' rather than 'ip'.
Fixes: c11117b634 ("ALSA: usb-audio: Refcount multiple accesses on the single clock")
Signed-off-by: Wan Jiabing <wanjiabing@vivo.com>
Link: https://lore.kernel.org/r/20220518021617.10114-1-wanjiabing@vivo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a clock source is connected to multiple nodes / endpoints, the
current USB-audio driver tries to set up at each time one of them is
configured. Although it reads the current rate and updates only if it
differs, some devices seem unhappy with this behavior and spew the
errors when reading/updating the rate unnecessarily.
This patch tries to reduce the redundant clock setup by introducing a
refcount for each clock source. When the stream is actually running,
a clock rate is "locked", and it bypasses the clock and/or refuse to
change any longer.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934
Link: https://lore.kernel.org/r/20220516104807.16482-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At cleaning up and moving the device rename from the quirk table to
its own table, we removed the entry for Rane SL-1 as we thought it's
only for renaming. It turned out, however, that the quirk is required
for matching with the device that declares itself as no standard
audio but only as vendor-specific.
Restore the quirk entry for Rane SL-1 to fix the regression.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215887
Fixes: 5436f59bc5 ("ALSA: usb-audio: Move device rename and profile quirks to an internal table")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220516103112.12950-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device doesn't support reading the sample rate, so we need to apply
this quirk to avoid a 15-second delay waiting for three timeouts.
Signed-off-by: Forest Crossman <cyrozap@gmail.com>
Link: https://lore.kernel.org/r/20220504002444.114011-2-cyrozap@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge 5.18-rc5 into usb-next
We need the USB fixes in here, and this resolves a merge issue in
drivers/usb/dwc3/drd.c
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The third argument of usb_maxpacket(): in_out has been deprecated
because it could be derived from the second argument (e.g. using
usb_pipeout(pipe)).
N.B. function usb_maxpacket() was made variadic to accommodate the
transition from the old prototype with three arguments to the new one
with only two arguments (so that no renaming is needed). The variadic
argument is to be removed once all users of usb_maxpacket() get
migrated.
CC: Jaroslav Kysela <perex@perex.cz>
CC: Takashi Iwai <tiwai@suse.com>
CC: Clemens Ladisch <clemens@ladisch.de>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Vincent Mailhol <mailhol.vincent@wanadoo.fr>
Link: https://lore.kernel.org/r/20220317035514.6378-8-mailhol.vincent@wanadoo.fr
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
For making easier to test, add the new quirk_flags bits 17 and 18 to
enable and disable the generic implicit feedback mode. The bit 17 is
equivalent with implicit_fb=1 option, applying the generic implicit
feedback sync mode. OTOH, the bit 18 disables the implicit fb mode
forcibly.
Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a rawmidi output stream is closed, it calls the drain at first,
then does trigger-off only when the drain returns -ERESTARTSYS as a
fallback. It implies that each driver should turn off the stream
properly after the drain. Meanwhile, USB-audio MIDI interface didn't
change the port->active flag after the drain. This may leave the
output work picking up the port that is closed right now, which
eventually leads to a use-after-free for the already released rawmidi
object.
This patch fixes the bug by properly clearing the port->active flag
after the output drain.
Reported-by: syzbot+70e777a39907d6d5fd0a@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/00000000000011555605dceaff03@google.com
Link: https://lore.kernel.org/r/20220420130247.22062-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB audio device 0db0:a073 based on the Realtek ALC4080 chipset
exposes all playback volume controls as "PCM". This makes
distinguishing the individual functions hard.
The mapping already adopted for device 0db0:419c based on the same
chipset fixes the issue, apply it for this device too.
Signed-off-by: Maurizio Avogadro <mavoga@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/Yl1ykPaGgsFf3SnW@ryzen
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the previous fix, we increased the max buffer bytes from 1MB to 4MB
so that we can use bigger buffers for the modern HiFi devices with
higher rates, more channels and wider formats. OTOH, extending this
has a concern that too big buffer is allowed for the lower rates, less
channels and narrower formats; when an application tries to allocate
as big buffer as possible, it'll lead to unexpectedly too huge size.
Also, we had a problem about the inconsistent max buffer and period
bytes for the implicit feedback mode when both streams have different
channels. This was fixed by the (relatively complex) patch to reduce
the max buffer and period bytes accordingly.
This is an alternative fix for those, a patch to kill two birds with
one stone (*): instead of increasing the max buffer bytes blindly and
applying the reduction per channels, we simply use the hw constraints
for the buffer and period "time". Meanwhile the max buffer and period
bytes are set unlimited instead.
Since the inconsistency of buffer (and period) bytes comes from the
difference of the channels in the tied streams, as long as we care
only about the buffer (and period) time, it doesn't matter; the buffer
time is same for different channels, although we still allow higher
buffer size. Similarly, this will allow more buffer bytes for HiFi
devices while it also keeps the reasonable size for the legacy
devices, too.
As of this patch, the max period and buffer time are set to 1 and 2
seconds, which should be large enough for all possible use cases.
(*) No animals were harmed in the making of this patch.
Fixes: 98c27add5d ("ALSA: usb-audio: Cap upper limits of buffer/period bytes for implicit fb")
Fixes: fee2ec8cce ("ALSA: usb-audio: Increase max buffer size")
Link: https://lore.kernel.org/r/20220412130740.18933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current limit of max buffer size 1MB seems too small for modern
devices with lots of channels and high sample rates.
Let's make bigger, 4MB.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20220407212740.17920-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the implicit feedback mode, some parameters are tied between both
playback and capture streams. One of the tied parameters is the
period size, and this can be a problem if the device has different
number of channels to both streams. Assume that an application opens
a playback stream that has an implicit feedback from a capture stream,
and it allocates up to the max period and buffer size as much as
possible. When the capture device supports only more channels than
the playback, the minimum period and buffer sizes become larger than
the sizes the playback stream took. That is, the minimum size will be
over the max size the driver limits, and PCM core sees as if no
available configuration is found, returning -EINVAL mercilessly.
For avoiding this problem, we have to look through the counter part of
audioformat list for each sync ep, and checks the channels. If more
channels are found there, we reduce the max period and buffer sizes
accordingly.
You may wonder that the patch adds only the evaluation of channels
between streams, and what about other parameters? Both the format and
the rate are tied in the implicit fb mode, hence they are always
identical.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215792
Fixes: 5a6c3e11c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220407211657.15087-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix:
sound/usb/midi.c: In function ‘snd_usbmidi_out_endpoint_create’:
sound/usb/midi.c:1389:2: error: case label does not reduce to an integer constant
case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */
^~~~
See https://lore.kernel.org/r/YkwQ6%2BtIH8GQpuct@zn.tnic for the gory
details as to why it triggers with older gccs only.
[ A slight correction with parentheses around the argument by tiwai ]
Signed-off-by: Borislav Petkov <bp@suse.de>
Link: https://lore.kernel.org/r/20220405151517.29753-3-bp@alien8.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the RODE NT-USB the lowest Playback mixer volume setting mutes the
audio output. But it is not reported as such causing e.g. PulseAudio to
accidentally mute the device when selecting a low volume.
Fix this by applying the existing quirk for this kind of issue when the
device is detected.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220311201400.235892-1-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Focusrite Scarlett Gen 2/3 interfaces with internal mixers have a
"standalone" mode. When the interface is not connected to a USB host
and standalone mode is enabled, the interface will pass audio as
previously configured. This patch adds an ALSA control to allow
enabling/disabling that mode.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/cd88871c5e77abd5c23a4758a1f2ec9fd427fd69.1646578164.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
scarlett2_config_items[] contains the parameters for the configuration
items. The driver previously had two sets of configurations items; one
for devices with no mixer, and one for devices with a mixer. This
patch splits the latter into two (one set for Gen 2 devices and one
set for Gen 3 devices) in preparation for a new item (standalone)
which is present in both but with a different offset.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20969f9ea500684e978c87067fbdc7e73de1f6ed.1646578164.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New device id for Corsair Virtuoso SE RGB Wireless that currently is not
in the mixer_map. This entry in the mixer_map is necessary in order to
label its mixer appropriately and allow userspace to pick the correct
volume controls. For instance, my own Corsair Virtuoso SE RGB Wireless
headset has this new ID and consequently, the sidetone and volume are not
working correctly without this change.
> sudo lsusb -v | grep -i corsair
Bus 007 Device 011: ID 1b1c:0a40 Corsair CORSAIR VIRTUOSO SE Wireless Gam
idVendor 0x1b1c Corsair
iManufacturer 1 Corsair
iProduct 2 CORSAIR VIRTUOSO SE Wireless Gaming Headset
Signed-off-by: Reza Jahanbakhshi <reza.jahanbakhshi@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220304212303.195949-1-reza.jahanbakhshi@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The default mixer resume code treats the errors at restoring the
modified mixer items as a fatal error, and it returns back to the
caller. This ends up in the resume failure, and the device will be
come unavailable, although basically those errors are intermittent and
can be safely ignored.
The problem itself has been present from the beginning, but it didn't
hit usually because the code tries to resume only the modified items.
But now with the recent commit to forcibly initialize each item at the
probe time, the problem surfaced more often, hence it appears as a
regression.
This patch fixes the regression simply by ignoring the errors at
resume.
Fixes: b96681bd58 ("ALSA: usb-audio: Initialize every feature unit once at probe time")
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215561
Link: https://lore.kernel.org/r/20220214125711.20531-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 83b7dcbc51 introduced a generic
implicit feedback parser, which fails to execute for M-Audio FastTrack
Ultra sound cards. The issue is with the ENDPOINT_SYNCTYPE check in
add_generic_implicit_fb() where the SYNCTYPE is ADAPTIVE instead of ASYNC.
The reason is that the sync type of the FastTrack output endpoints are
set to adaptive in the quirks table since commit
65f04443c9.
Fixes: 83b7dcbc51 ("ALSA: usb-audio: Add generic implicit fb parsing")
Signed-off-by: Matteo Martelli <matteomartelli3@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220211224913.20683-2-matteomartelli3@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable c is being initialized in an outer for-loop and also
re-initialized inside an inner for-loop. The first initialization
is redundant and can be removed.
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://lore.kernel.org/r/20220207140617.341172-1-colin.i.king@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device provides both audio and video. The original quirk added in
commit 48827e1d6a ("ALSA: usb-audio: Add quirk for VF0770") used
USB_DEVICE to match the vendor and product ID. Depending on module order,
if snd-usb-audio was asked first, it would match the entire device and
uvcvideo wouldn't get to see it. Change the matching to USB_AUDIO_DEVICE
to restore uvcvideo matching in all cases.
Fixes: 48827e1d6a ("ALSA: usb-audio: Add quirk for VF0770")
Reported-by: Jukka Heikintalo <heikintalo.jukka@gmail.com>
Tested-by: Jukka Heikintalo <heikintalo.jukka@gmail.com>
Reported-by: Paweł Susicki <pawel.susicki@gmail.com>
Tested-by: Paweł Susicki <pawel.susicki@gmail.com>
Cc: <stable@vger.kernel.org> # 5.4, 5.10, 5.14, 5.15
Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de>
Link: https://lore.kernel.org/r/20220131183516.61191-1-hahnjo@hahnjo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
clang static analysis reports this representative issue
mixer.c:1548:35: warning: Assigned value is garbage or undefined
ucontrol->value.integer.value[0] = val;
^ ~~~
The filter_error() macro allows errors to be ignored.
If errors can be ignored, initialize variables
so garbage will not be used.
Fixes: 48cc429735 ("ALSA: usb-audio: Filter error from connector kctl ops, too")
Signed-off-by: Tom Rix <trix@redhat.com>
Link: https://lore.kernel.org/r/20220126182142.1184819-1-trix@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of the struct_size() helper instead of an open-coded version,
in order to avoid any potential type mistakes or integer overflows that,
in the worst scenario, could lead to heap overflows.
Also, address the following sparse warnings:
sound/usb/mixer_scarlett_gen2.c:1064:28: warning: using sizeof on a flexible structure
sound/usb/mixer_scarlett_gen2.c:1065:29: warning: using sizeof on a flexible structure
Link: https://github.com/KSPP/linux/issues/174
Signed-off-by: Gustavo A. R. Silva <gustavoars@kernel.org>
Reviewed-by: Kees Cook <keescook@chromium.org>
Link: https://lore.kernel.org/r/20220120211600.GA28841@embeddedor
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB audio device 0db0:419c based on the Realtek ALC4080 chip exposes
all playback volume controls as "PCM". This is makes distinguishing the
individual functions hard.
The added mapping distinguishes all playback volume controls as their
respective function:
- Speaker - for back panel output
- Frontpanel Headphone - for front panel output
- IEC958 - for digital output on the back panel
This clarifies the individual volume control functions for users.
Signed-off-by: Johannes Schickel <lordhoto@gmail.com>
Link: https://lore.kernel.org/r/20220115140257.8751-1-lordhoto@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Olivia Mackintosh has posted to alsa-devel reporting that
there's a potential bug that could break mixer quirks for Pioneer
devices introduced by 6d27788160
"ALSA: usb-audio: Add support for the Pioneer DJM 750MK2
Mixer/Soundcard".
This happened because the DJM 750 MK2 was added last to the Pioneer DJM
device table index and defined as 0x4 but was added to snd_djm_devices[]
just after the DJM 750 (MK1) entry instead of last, after the DJM 900
NXS2. This escaped review.
To prevent that from ever happening again, Takashi Iwai suggested to use
C99 array designators in snd_djm_devices[] instead of simply reordering
the entries.
Fixes: 6d27788160 ("ALSA: usb-audio: Add support for the Pioneer DJM 750MK2")
Reported-by: Olivia Mackintosh <livvy@base.nu>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Geraldo Nascimento <geraldogabriel@gmail.com>
Link: https://lore.kernel.org/r/Yau46FDzoql0SNnW@geday
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change made mistakenly the stream for capture started at
prepare stage. Add the stream direction check to avoid it.
Fixes: 9c9a3b9da8 ("ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback")
Link: https://lore.kernel.org/r/20211119102629.7476-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent regression report revealed that the judgment of the
low-latency playback mode based on the runtime->stop_threshold cannot
work reliably at the prepare stage, as sw_params call may happen at
any time, and PCM dmix actually sets it up after the prepare call.
This ended up with the stall of the stream as PCM ack won't be issued
at all.
For addressing this, check the free-wheeling mode again at the PCM
trigger right before starting the stream again, and allow switching to
the non-LL mode at a late stage.
Fixes: d5f871f89e ("ALSA: usb-audio: Improved lowlatency playback support")
Reported-and-tested-by: Kirill A. Shutemov <kirill.shutemov@linux.intel.com>
Link: https://lore.kernel.org/r/20211117161855.m45mxcqszkfcetai@box.shutemov.name
Link: https://lore.kernel.org/r/20211119102459.7055-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices do mute the volume at the minimal volume, and for such
devices, we need to set SNDRV_CTL_TLVT_DB_MINMAX_MUTE to the TLV
information. It corresponds to setting usb_mixer_elem_info.min_mute
flag in the USB-audio driver.
This patch adds a new field min_mute in usbmix_dB_map so that the
mixer map entry can pass the flag.
Link: https://lore.kernel.org/r/20211116065415.11159-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The values in usbmix_dB_map should be rather signed while we're using
u32. As the copied target (usb_mixer_elem_info.dBmin and dBmax) is
int, let's make them also int.
Link: https://lore.kernel.org/r/20211116065415.11159-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding the Line6 HX-Stomp XL USB_ID as it needs this fixed frequency
quirk as well.
The device is basically just the HX-Stomp with some more buttons on
the face. I've done some recording with it after adding it, and it
seems to function properly with this fix. The Midi features appear to
be working as well.
[ a coding style fix and patch reformat by tiwai ]
Signed-off-by: Jason Ormes <skryking@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211030200405.1358678-1-skryking@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add another device ID for JBL Quantum 400. It requires the same quirk as
other JBL Quantum devices.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211030174308.1011825-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the missing endpoint max-packet sanity check to probe() to avoid
division by zero in alloc_stream_buffers() in case a malicious device
has broken descriptors (or when doing descriptor fuzz testing).
Note that USB core will reject URBs submitted for endpoints with zero
wMaxPacketSize but that drivers doing packet-size calculations still
need to handle this (cf. commit 2548288b4f ("USB: Fix: Don't skip
endpoint descriptors with maxpacket=0")).
Fixes: 63978ab3e3 ("sound: add Edirol UA-101 support")
Cc: stable@vger.kernel.org # 2.6.34
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20211026095401.26522-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB control and bulk message timeouts are specified in milliseconds and
should specifically not vary with CONFIG_HZ.
Fixes: c6d43ba816 ("ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB")
Cc: stable@vger.kernel.org # 2.6.39
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20211025121142.6531-2-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pointer cs_desc return from snd_usb_find_clock_source could
be null, so there is a potential null pointer dereference issue.
Fix this by adding a null check before dereference.
Signed-off-by: Chengfeng Ye <cyeaa@connect.ust.hk>
Link: https://lore.kernel.org/r/20211024111736.11342-1-cyeaa@connect.ust.hk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a Jieli Technology USB Webcam is connected, the video part works
well, but the mic sound is speeded up. On dmesg there are messages
about different rates from the runtime rates, warnings about volume
resolution and lastly, the log is filled, every 5 seconds, with
retire_capture_urb error messages.
The mic works only when ep packet size is set to wMaxPacketSize (normal
sound and no more retire_capture_urb error messages). Skipping reading
sample rate, fixes the messages about different rates and forcing a volume
resolution, fixes warnings about volume range. I have arbitrarily choosed
the value (16): I read in a comment that there should be no more than 255
levels, so 4096 (max volume) / 16 = 0-255.
Signed-off-by: Marco Giunta <giun7a@gmail.com>
Link: https://lore.kernel.org/r/20211018162552.12082-1-giun7a@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As per discussion at: https://github.com/szszoke/sennheiser-gsp670-pulseaudio-profile/issues/13
The GSP670 has 2 playback and 1 recording device that by default are
detected in an incompatible order for alsa. This may have been done to make
it compatible for the console by the manufacturer and only affects the
latest firmware which uses its own ID.
This quirk will resolve this by reordering the channels.
Signed-off-by: Brendan Grieve <brendan@grieve.com.au>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211015025335.196592-1-brendan@grieve.com.au
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we used to read the current value of the mixer element
dynamically at the first access, and the error from a GET_CUR message
is treated as a fatal error (unless QUIRK_IGNORE_CTL_ERROR is set).
It's rather inconvenient, as most of GET_CUR errors are no fatal, and
we can continue operation with assumption of some fixed value.
This patch makes the USB-audio driver to change the behavior at probe
time; now it tries to initialize the current value of each mixer
element that is built from a feature unit (those for typically for
mixer volumes and switches). When a read failure happens, it tries to
set the known minimum value. After that point, a cached value is used
always, hence we won't hit GET_CUR message error any longer.
The error from GET_CUR message is still shown as a warning normally,
but only once at the probe time, and it'll keep operating. If the
message is confirmed to be harmless, it can be shut up by
QUIRK_IGNORE_CTL_ERROR quirk flag, too.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error from snd_usb_lock_shutdown() indicates that the device is
disconnected, hence it makes no sense to show any further control
message error in get_ctl_value_v2(). Return the error directly
instead.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error message in get_ctl_value_v2() (for UAC2/3) is shown via
KERN_ERR level but it was intended to be rather a debug message as
seen in get_ctl_value_v1() (for UAC1). This patch downgrade the
printk level.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A back-merge of 5.15 branch into 5.16-devel branch for further
development of USB and ALSA core stuff that depends on 5.15 fixes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Shciit Hel device responds to the ctl message for the mic capture
switch with a timeout of -EPIPE:
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
This seems safe to ignore as the device works properly with the control
message quirk, so add it to the quirk table so all is good.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-usb@vger.kernel.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/YWgR3nOI1osvr5Yo@kroah.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device advertises 8 formats, but only a rate of 48kHz is honored
by the hardware and 24 bits give chopped audio, so only report the
one working combination. This fixes out-of-the-box audio experience
with PipeWire which otherwise attempts to choose S24_3LE (while
PulseAudio defaulted to S16_LE).
Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211012200906.3492-1-hahnjo@hahnjo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent support for the improved low-latency playback mode applied
the SNDRV_PCM_INFO_EXPLICIT_SYNC flag for the target streams, but this
was a slight overkill. The use of the flag above disables effectively
both PCM status and control mmaps, while basically what we want to
track is only about the appl_ptr update.
For less restriction, use a more proper flag,
SNDRV_PCM_INFO_SYNC_APPLPTR instead, which disables only the control
mmap.
Fixes: d5f871f89e ("ALSA: usb-audio: Improved lowlatency playback support")
Link: https://lore.kernel.org/r/20211011103650.10182-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The kernel already has support for very similar Pioneer djm products
and this work is based on that.
Added device to quirks-table.h and added control info to
mixer_quirks.c.
Tested on my hardware and all working.
Signed-off-by: William Overton <willovertonuk@gmail.com>
Link: https://lore.kernel.org/r/20211010145841.11907-1-willovertonuk@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a stream is in the implicit feedback mode, it's more or less tied
with a capture stream. Passing SNDRV_PCM_INFO_JOINT_DUPLEX may help
for user-space to understand the situation.
Link: https://lore.kernel.org/r/20211007083528.4184-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
