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The recent commit c11117b634f4 ("ALSA: usb-audio: Refcount multiple
accesses on the single clock") tries to manage the clock rate shared
by several endpoints. This was intended for avoiding the unmatched
rate by a different endpoint, but unfortunately, it introduced a
regression for PulseAudio and pipewire, too; those applications try to
probe the multiple possible rates (44.1k and 48kHz) and setting up the
normal rate fails but only the last rate is applied.
The cause is that the last sample rate is still left to the clock
reference even after closing the endpoint, and this value is still
used at the next open. It happens only when applications set up via
PCM prepare but don't start/stop the stream; the rate is reset when
the stream is stopped, but it's not cleared at close.
This patch addresses the issue above, simply by clearing the rate set
in the clock reference at the last close of each endpoint.
Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock")
Reported-by: Jason A. Donenfeld <Jason@zx2c4.com>
Tested-by: Jason A. Donenfeld <Jason@zx2c4.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/
Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620
Link: https://lore.kernel.org/r/20220907100421.6443-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There may be a bad USB audio device with a USB ID of (0x04fa, 0x4201) and
the number of it's interfaces less than 4, an out-of-bounds read bug occurs
when parsing the interface descriptor for this device.
Fix this by checking the number of interfaces.
Signed-off-by: Dongxiang Ke <kdx.glider@gmail.com>
Link: https://lore.kernel.org/r/20220906024928.10951-1-kdx.glider@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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One of the former changes for the endpoint management was the more
consistent setup of endpoints at hw_params.
snd_usb_endpoint_configure() is a single function that does the full
setup, and it's called from both PCM hw_params and prepare callbacks.
Although the EP setup at the prepare phase is usually skipped (by
checking need_setup flag), it may be still effective in some cases
like suspend/resume that requires the interface setup again.
As it's a full and single setup, the invocation of
snd_usb_endpoint_configure() includes not only the USB interface setup
but also the buffer release and allocation. OTOH, doing the buffer
release and re-allocation at PCM prepare phase is rather superfluous,
and better to be done only in the hw_params phase.
For those optimizations, this patch splits the endpoint setup to two
phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(),
to be called from hw_params and from prepare, respectively.
Note that this patch changes the driver operation slightly,
effectively moving the USB interface setup again to PCM prepare stage
instead of hw_params stage, while the buffer allocation and such
initializations are still done at hw_params stage.
And, the change of the USB interface setup timing (moving to prepare)
gave an interesting "fix", too: it was reported that the recent
kernels caused silent output at the beginning on playbacks on some
devices on Android, and this change casually fixed the regression.
It seems that those devices are picky about the sample rate change (or
the interface change?), and don't follow the too immediate rate
changes.
Meanwhile, Android operates the PCM in the following order:
- open, then hw_params with the possibly highest sample rate
- close without prepare
- re-open, hw_params with the normal sample rate
- prepare, and start streaming
This procedure ended up the hw_params twice with different rates, and
because the recent kernel did set up the sample rate twice one and
after, it screwed up the device. OTOH, the earlier kernels didn't set
up the USB interface at hw_params, hence this problem didn't appear.
Now, with this patch, the USB interface setup is again back to the
prepare phase, and it works around the problem automagically.
Although we should address the sample rate problem in a more solid
way in future, let's keep things working as before for now.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Cc: <stable@vger.kernel.org>
Reported-by: chihhao chen <chihhao.chen@mediatek.com>
Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@mediatek.com
Link: https://lore.kernel.org/r/20220901124136.4984-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the delayed registration is specified via either delayed_register
option or the quirk, we delay the invocation of snd_card_register()
until the given interface. But if a wrong value has been set there
and there are more interfaces over the given interface number,
snd_card_register() call would be missing for those interfaces.
This patch catches up those missing calls by fixing the comparison of
the interface number. Now the call is skipped only if the processed
interface is less than the given interface, instead of the exact
match.
Fixes: b70038ef4fea ("ALSA: usb-audio: Add delayed_register option")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082
Link: https://lore.kernel.org/r/20220831125901.4660-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The info message that was added in the commit a4aad5636c72 ("ALSA:
usb-audio: Inform devices that need delayed registration") is actually
useful to know the need for the delayed registration. However, it
turned out that this doesn't catch the all cases; namely, this warned
only when a PCM stream is attached onto the existing PCM instance, but
it doesn't count for a newly created PCM instance. This made
confusion as if there were no further delayed registration.
This patch moves the check to the code path for either adding a stream
or creating a PCM instance. Also, make it simpler by checking the
card->registered flag instead of querying each snd_device state.
Fixes: a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082
Link: https://lore.kernel.org/r/20220831125901.4660-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The USB DAC from LH Labs (2522:0007) seems requiring the same quirk as
Sony Walkman to set up the interface like UAC1; otherwise it gets the
constant errors "usb_set_interface failed (-71)". This patch adds a
quirk entry for addressing the buggy behavior.
Reported-by: Lennert Van Alboom <lennert@vanalboom.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/T3VPXtCc4uFws9Gfh2RjX6OdwM1RqfC6VqQr--_LMDyB2x5N3p9_q6AtPna17IXhHwBtcJVdXuS80ZZSCMjh_BafIbnzJPhbrkmhmWS6DlI=@vanalboom.org
Link: https://lore.kernel.org/r/20220828074143.14736-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Don't populate the read-only array marker on the stack but instead make
it static const. Also makes the object code a little smaller.
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://lore.kernel.org/r/20220809181544.3046429-1-colin.i.king@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS ROG Zenith II has two USB interfaces, one for the front headphone
and another for the rest I/O. Currently we provided the mixer mapping
for the latter but with an incomplete form.
This patch corrects and provides more comprehensive mixer mapping, as
well as providing the proper device names for both the front headphone
and main audio.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211005
Fixes: 2a48218f8e23 ("ALSA: usb-audio: Add mixer workaround for TRX40 and co")
Link: https://lore.kernel.org/r/20220809073259.18849-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Focusrite Clarett+ 8Pre uses the same protocol as the Scarlett Gen
2 and Gen 3 product range. This patch adds support for the Clarett+
8Pre by adding appropriate entries to the scarlett2 driver.
The Clarett+ 2Pre and 4Pre, and the Clarett USB product line
presumably use the same protocol as well, so support for them can
easily be added if someone can test.
Signed-off-by: Christian Colglazier <christian@cacolglazier.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/20220809043241.GA2749152@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For sysfs outputs, it's safer to use a new helper, sysfs_emit(),
instead of the raw sprintf() & co. This patch replaces those usages
straightforwardly with a new helper, sysfs_emit().
Link: https://lore.kernel.org/r/20220801165639.26030-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Delete the redundant word 'in'.
Signed-off-by: wangjianli <wangjianli@cdjrlc.com>
Link: https://lore.kernel.org/r/20220724071829.11117-1-wangjianli@cdjrlc.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Delete the redundant word 'in'.
Signed-off-by: wangjianli <wangjianli@cdjrlc.com>
Link: https://lore.kernel.org/r/20220724071644.10630-1-wangjianli@cdjrlc.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Just like other Behringer models, UMC202HD (USB ID 1397:0507) requires
the quirk for the stable streaming, too.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220722143948.29804-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the driver fails in snd_card_register() at probe time, it will free
the 'bcd2k->midi_out_urb' before killing it, which may cause a UAF bug.
The following log can reveal it:
[ 50.727020] BUG: KASAN: use-after-free in bcd2000_input_complete+0x1f1/0x2e0 [snd_bcd2000]
[ 50.727623] Read of size 8 at addr ffff88810fab0e88 by task swapper/4/0
[ 50.729530] Call Trace:
[ 50.732899] bcd2000_input_complete+0x1f1/0x2e0 [snd_bcd2000]
Fix this by adding usb_kill_urb() before usb_free_urb().
Fixes: b47a22290d58 ("ALSA: MIDI driver for Behringer BCD2000 USB device")
Signed-off-by: Zheyu Ma <zheyuma97@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220715010515.2087925-1-zheyuma97@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use atomic_try_cmpxchg instead of atomic_cmpxchg (*ptr, old, new) == old in
ep_state_update. x86 CMPXCHG instruction returns success in ZF flag,
so this change saves a compare after cmpxchg (and related move instruction
in front of cmpxchg).
No functional change intended.
Signed-off-by: Uros Bizjak <ubizjak@gmail.com>
Link: https://lore.kernel.org/r/20220713151946.4743-1-ubizjak@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixes: 4b8ea38fabab ("ALSA: usb-audio: Support jack detection on Dell dock")
Reported-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/202207051932.qUilU0am-lkp@intel.com
Signed-off-by: Jan Schär <jan@jschaer.ch>
Link: https://lore.kernel.org/r/20220705135746.13713-1-jan@jschaer.ch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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: 964af639ad69 ("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>
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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>
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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.
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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>
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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>
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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>
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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>
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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>
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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: 7b0efea4baf0 ("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>
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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: 7b0efea4baf0 ("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>
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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()
...
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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: 5ce0b06ae5e6 ("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>
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At closing a USB MIDI output substream, there might be still a pending
work, which would eventually access the rawmidi runtime object that is
being released. For fixing the race, make sure to cancel the pending
work at closing.
Reported-by: syzbot+6912c9592caca7ca0e7d@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000e7e75005dfd07cf6@google.com
Link: https://lore.kernel.org/r/20220525131203.11299-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch implements a static mapping for Gigabyte B450/550 Mobos so
that the mixer elements appear reasonably and jack detections work
properly.
Reported-and-tested-by: Brock Szuszczewicz <brock@system76.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215988
Link: https://lore.kernel.org/r/20220525122018.3299-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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: bf6313a0ff76 ("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>
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Merge for 5.18-rc1
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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: bf6313a0ff76 ("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>
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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: bf6313a0ff76 ("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>
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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: c11117b634f4 ("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>
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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>
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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: 5436f59bc5bc ("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>
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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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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>
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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>
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Use the new quirk bits to manage the generic implicit fb quirk
entries. This makes easier to compare with other devices.
Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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In the previous fix, we increased the max buffer bytes from 1MB to 4MB
so that we can use bigger buffers for the modern HiFi devices with
higher rates, more channels and wider formats. OTOH, extending this
has a concern that too big buffer is allowed for the lower rates, less
channels and narrower formats; when an application tries to allocate
as big buffer as possible, it'll lead to unexpectedly too huge size.
Also, we had a problem about the inconsistent max buffer and period
bytes for the implicit feedback mode when both streams have different
channels. This was fixed by the (relatively complex) patch to reduce
the max buffer and period bytes accordingly.
This is an alternative fix for those, a patch to kill two birds with
one stone (*): instead of increasing the max buffer bytes blindly and
applying the reduction per channels, we simply use the hw constraints
for the buffer and period "time". Meanwhile the max buffer and period
bytes are set unlimited instead.
Since the inconsistency of buffer (and period) bytes comes from the
difference of the channels in the tied streams, as long as we care
only about the buffer (and period) time, it doesn't matter; the buffer
time is same for different channels, although we still allow higher
buffer size. Similarly, this will allow more buffer bytes for HiFi
devices while it also keeps the reasonable size for the legacy
devices, too.
As of this patch, the max period and buffer time are set to 1 and 2
seconds, which should be large enough for all possible use cases.
(*) No animals were harmed in the making of this patch.
Fixes: 98c27add5d96 ("ALSA: usb-audio: Cap upper limits of buffer/period bytes for implicit fb")
Fixes: fee2ec8cceb3 ("ALSA: usb-audio: Increase max buffer size")
Link: https://lore.kernel.org/r/20220412130740.18933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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In the implicit feedback mode, some parameters are tied between both
playback and capture streams. One of the tied parameters is the
period size, and this can be a problem if the device has different
number of channels to both streams. Assume that an application opens
a playback stream that has an implicit feedback from a capture stream,
and it allocates up to the max period and buffer size as much as
possible. When the capture device supports only more channels than
the playback, the minimum period and buffer sizes become larger than
the sizes the playback stream took. That is, the minimum size will be
over the max size the driver limits, and PCM core sees as if no
available configuration is found, returning -EINVAL mercilessly.
For avoiding this problem, we have to look through the counter part of
audioformat list for each sync ep, and checks the channels. If more
channels are found there, we reduce the max period and buffer sizes
accordingly.
You may wonder that the patch adds only the evaluation of channels
between streams, and what about other parameters? Both the format and
the rate are tied in the implicit fb mode, hence they are always
identical.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215792
Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220407211657.15087-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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Pull 5.18 development branch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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