This removes the need to power cycle the Dell WD15 dock if it has been
attached to a Windows machine.
The Windows driver puts the ALC4020 USB audio controller into
'manual mode', and then does all the power management and other
configuration itself, by sending HD audio commands directly to the
ALC3263 audio codec via vendor-type USB messages. If manual mode is off,
this is all handled by the firmware, and works well enough.
If manual mode is turned on, the latency of the SET INTERFACE command
goes from several hundred ms to less than 1 ms
(see https://bugzilla.suse.com/show_bug.cgi?id=1089467), but I'm not
sure if the additional code that would be required is worth it.
Funnily enough, the Windows driver tries to turn off manual mode when
the dock is disconnected, which doesn't work for obvious reasons.
Additionally, fix a bug in dell_dock_init_vol, which didn't work because
the Control Selector was missing.
Now, it properly resets the volume to 0dB.
Fixes: 964af639ad ("ALSA: usb-audio: Initialize Dell Dock playback volumes")
Signed-off-by: Jan Schär <jan@jschaer.ch>
Link: https://lore.kernel.org/r/20220627171855.42338-2-jan@jschaer.ch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Dell WD15 dock has a headset and a line out port. Add support for
detecting if a jack is inserted into one of these ports.
For the headset jack, additionally determine if a mic is present.
The WD15 contains an ALC4020 USB audio controller and ALC3263 audio codec
from Realtek. It is a UAC 1 device, and UAC 1 does not support jack
detection. Instead, jack detection works by sending HD Audio commands over
vendor-type USB messages.
I found out how it works by looking at USB captures on Windows.
The audio codec is very similar to the one supported by
sound/soc/codecs/rt298.c / rt298.h, some constant names and the mic
detection are adapted from there. The realtek_add_jack function is adapted
from build_connector_control in sound/usb/mixer.c.
I tested this on a WD15 dock with the latest firmware.
Signed-off-by: Jan Schär <jan@jschaer.ch>
Link: https://lore.kernel.org/r/20220627171855.42338-1-jan@jschaer.ch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of fixes for v5.19, quite large but nothing major - a good
chunk of it is more stuff that was identified by mixer-test regarding
event generation.
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Merge tag 'asoc-fix-v5.19-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.19
A collection of fixes for v5.19, quite large but nothing major - a good
chunk of it is more stuff that was identified by mixer-test regarding
event generation.
The patch applies the same quirks used for SC-01 at firmware v1.1.0 to
the ones running v1.0.0, with respect to hard-coded sample rates.
I got two more units and successfully tested the patch series with both
firmwares.
The support is now complete (not accounting ASIO).
Signed-off-by: Egor Vorontsov <sdoregor@sdore.me>
Link: https://lore.kernel.org/r/20220627100041.2861494-2-sdoregor@sdore.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fiero SC-01 is a USB sound card with two mono inputs and a single
stereo output. The inputs are composed into a single stereo stream.
The device uses a vendor-provided driver on Windows and does not work
at all without it. The driver mostly provides ASIO functionality, but
also alters the way the sound card is queried for sample rates and
clocks.
ALSA queries those failing with an EPIPE (same as Windows 10 does).
Presumably, the vendor-provided driver does not query it at all, simply
matching by VID:PID. Thus, I consider this a buggy firmware and adhere
to a set of fixed endpoint quirks instead.
The soundcard has an internal clock. Implicit feedback mode is required
for the playback.
I have updated my device to v1.1.0 from a Windows 10 VM using a vendor-
provided binary prior to the development, hoping for it to just begin
working. The device provides no obvious way to downgrade the firmware,
and regardless, there's no binary available for v1.0.0 anyway.
Thus, I will be getting another unit to extend the patch with support
for that. Expected to be a simple copy-paste of the existing one,
though.
There were no previous reports of that device in context of Linux
anywhere. Other issues have been reported though, but that's out of the
scope.
Signed-off-by: Egor Vorontsov <sdoregor@sdore.me>
Link: https://lore.kernel.org/r/20220627100041.2861494-1-sdoregor@sdore.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Treat the claimed 96kHz 1ch in the descriptors as 48kHz 2ch, so that
the audio stream doesn't sound mono. Also fix initial stream
alignment, so that left and right channels are in the correct order.
Signed-off-by: John Veness <john-linux@pelago.org.uk>
Link: https://lore.kernel.org/r/20220624140757.28758-1-john-linux@pelago.org.uk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both Behringer UMC 202 HD and 404 HD need explicit quirks to enable
the implicit feedback mode and start the playback stream primarily.
The former seems fixing the stuttering and the latter is required for
a playback-only case.
Note that the "clock source 41 is not valid" error message still
appears even after this fix, but it should be only once at probe.
The reason of the error is still unknown, but this seems to be mostly
harmless as it's a one-off error and the driver retires the clock
setup and it succeeds afterwards.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220624101132.14528-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Buffer overflow could occur in the loop "while", due to accessing an
array element before checking the index.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Signed-off-by: Daniil Dementev <d.dementev@ispras.ru>
Reviewed-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Link: https://lore.kernel.org/r/20220610165732.2904-1-d.dementev@ispras.ru
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Focusrite Saffire 6 has fixed audioformat quirks with multiple
endpoints assigned to a single altsetting. Unfortunately the generic
parser couldn't detect the sync endpoint correctly as the implicit
sync due to the missing EP attribute bits. In the former kernels, it
used to work somehow casually, but it's been broken for a while after
the large code change in 5.11.
This patch cures the regression by the following:
- Allow the static quirk table to provide the sync EP information;
we just need to fill the fields and let the generic parser skipping
parsing if sync_ep is already set.
- Add the sync endpoint information to the entry for Saffire 6.
Fixes: 7b0efea4ba ("ALSA: usb-audio: Add missing ep_idx in fixed EP quirks")
Reported-and-tested-by: André Kapelrud <a.kapelrud@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220606160910.6926-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When ep_idx is already non-zero, it means usually a capture stream
that is set up explicity by a fixed-format quirk, and applying the
check for generic (non-implicit-fb) sync EPs might hit incorrectly,
resulting in a bogus sync endpoint for the capture stream.
This patch adds a check for the ep_idx and skip if it's a secondary
endpoint. It's a part of the fixes for regressions on Saffire 6.
Fixes: 7b0efea4ba ("ALSA: usb-audio: Add missing ep_idx in fixed EP quirks")
Reported-and-tested-by: André Kapelrud <a.kapelrud@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220606160910.6926-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here is the "big" set of USB and Thunderbolt driver changes for
5.18-rc1. For the most part it's been a quiet development cycle for the
USB core, but there are the usual "hot spots" of development activity.
Included in here are:
- Thunderbolt driver updates:
- fixes for devices without displayport adapters
- lane bonding support and improvements
- other minor changes based on device testing
- dwc3 gadget driver changes. It seems this driver will never
be finished given that the IP core is showing up in zillions
of new devices and each implementation decides to do something
different with it...
- uvc gadget driver updates as more devices start to use and
rely on this hardware as well
- usb_maxpacket() api changes to remove an unneeded and unused
parameter.
- usb-serial driver device id updates and small cleanups
- typec cleanups and fixes based on device testing
- device tree updates for usb properties
- lots of other small fixes and driver updates.
All of these have been in linux-next for weeks with no reported
problems.
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge tag 'usb-5.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb
Pull USB / Thunderbolt updates from Greg KH:
"Here is the "big" set of USB and Thunderbolt driver changes for
5.18-rc1. For the most part it's been a quiet development cycle for
the USB core, but there are the usual "hot spots" of development
activity.
Included in here are:
- Thunderbolt driver updates:
- fixes for devices without displayport adapters
- lane bonding support and improvements
- other minor changes based on device testing
- dwc3 gadget driver changes.
It seems this driver will never be finished given that the IP core
is showing up in zillions of new devices and each implementation
decides to do something different with it...
- uvc gadget driver updates as more devices start to use and rely on
this hardware as well
- usb_maxpacket() api changes to remove an unneeded and unused
parameter.
- usb-serial driver device id updates and small cleanups
- typec cleanups and fixes based on device testing
- device tree updates for usb properties
- lots of other small fixes and driver updates.
All of these have been in linux-next for weeks with no reported
problems"
* tag 'usb-5.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb: (154 commits)
USB: new quirk for Dell Gen 2 devices
usb: dwc3: core: Add error log when core soft reset failed
usb: dwc3: gadget: Move null pinter check to proper place
usb: hub: Simplify error and success path in port_over_current_notify
usb: cdns3: allocate TX FIFO size according to composite EP number
usb: dwc3: Fix ep0 handling when getting reset while doing control transfer
usb: Probe EHCI, OHCI controllers asynchronously
usb: isp1760: Fix out-of-bounds array access
xhci: Don't defer primary roothub registration if there is only one roothub
USB: serial: option: add Quectel BG95 modem
USB: serial: pl2303: fix type detection for odd device
xhci: Allow host runtime PM as default for Intel Alder Lake N xHCI
xhci: Remove quirk for over 10 year old evaluation hardware
xhci: prevent U2 link power state if Intel tier policy prevented U1
xhci: use generic command timer for stop endpoint commands.
usb: host: xhci-plat: omit shared hcd if either root hub has no ports
usb: host: xhci-plat: prepare operation w/o shared hcd
usb: host: xhci-plat: create shared hcd after having added main hcd
xhci: prepare for operation w/o shared hcd
xhci: factor out parts of xhci_gen_setup()
...
Maris found out that the quirk for TEAC devices to work around the
clock setup is needed to apply only when the base clock is changed,
e.g. from 48000-based clocks (48000, 96000, 192000, 384000) to
44100-based clocks (44100, 88200, 176400, 352800), or vice versa,
while switching to another clock with the same base clock doesn't need
the (forcible) interface setup.
This patch implements the optimization for the TEAC clock quirk to
avoid the unnecessary interface re-setup.
Fixes: 5ce0b06ae5 ("ALSA: usb-audio: Workaround for clock setup on TEAC devices")
Reported-by: Maris Abele <maris7abele@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220531130749.30357-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Devices such as the TC-Helicon GoXLR require the sync endpoint to be
configured in advance of the data endpoint in order for sound output
to work.
This patch simply changes the ordering of EP configuration to resolve
this.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215079
Signed-off-by: Craig McLure <craig@mclure.net>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220524062115.25968-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
