Things seem calming down, only usual device-specific fixes for
HD-audio and USB-audio at this time.
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Merge tag 'sound-5.12-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Things seem calming down, only usual device-specific fixes for
HD-audio and USB-audio at this time"
* tag 'sound-5.12-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: fix mute/micmute LEDs for HP 640 G8
ALSA: hda: Add missing sanity checks in PM prepare/complete callbacks
ALSA: hda: Re-add dropped snd_poewr_change_state() calls
ALSA: usb-audio: Apply sample rate quirk to Logitech Connect
ALSA: hda/realtek: call alc_update_headset_mode() in hp_automute_hook
ALSA: hda/realtek: fix a determine_headset_type issue for a Dell AIO
Logitech ConferenceCam Connect is a compound USB device with UVC and
UAC. Not 100% reproducible but sometimes it keeps responding STALL to
every control transfer once it receives get_freq request.
This patch adds 046d:0x084c to a snd_usb_get_sample_rate_quirk list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=203419
Signed-off-by: Ikjoon Jang <ikjn@chromium.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210324105153.2322881-1-ikjn@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Majority of changes are various ASoC device/platform-specific small
fixes (including a removal of stale file) while the only common
change is a clk management fix in ASoC simple-card driver.
The rest are usual HD-audio quirks.
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Merge tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The majority of changes are various ASoC device/platform-specific
small fixes (including a removal of stale file) while the only common
change is a clk management fix in ASoC simple-card driver.
The rest are the usual HD-audio quirks"
* tag 'sound-5.12-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (44 commits)
ALSA: usb-audio: Fix unintentional sign extension issue
ALSA: hda/realtek: fix mute/micmute LEDs for HP 850 G8
ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms
ASoC: rt711: add snd_soc_component remove callback
ASoC: rt5659: Update MCLK rate in set_sysclk()
ASoC: simple-card-utils: Do not handle device clock
ALSA: hda/realtek: fix mute/micmute LEDs for HP 440 G8
ALSA: hda/realtek: fix mute/micmute LEDs for HP 840 G8
ALSA: hda/realtek: apply pin quirk for XiaomiNotebook Pro
ALSA: hda/realtek: Apply headset-mic quirks for Xiaomi Redmibook Air
ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge
ALSA: dice: fix null pointer dereference when node is disconnected
ALSA: hda: generic: Fix the micmute led init state
ASoC: qcom: lpass-cpu: Fix lpass dai ids parse
spi: cadence: set cqspi to the driver_data field of struct device
ASoC: SOF: intel: fix wrong poll bits in dsp power down
ASoC: codecs: wcd934x: add a sanity check in set channel map
ASoC: qcom: sdm845: Fix array out of range on rx slim channels
ASoC: qcom: sdm845: Fix array out of bounds access
ASoC: remove remnants of sirf prima/atlas audio codec
...
The shifting of the u8 integer device by 24 bits to the left will
be promoted to a 32 bit signed int and then sign-extended to a
64 bit unsigned long. In the event that the top bit of device is
set then all then all the upper 32 bits of the unsigned long will
end up as also being set because of the sign-extension. Fix this
by casting device to an unsigned long before the shift.
Addresses-Coverity: ("Unintended sign extension")
Fixes: a07df82c79 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210318132008.15266-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MODULE_SUPPORTED_DEVICE was added in pre-git era and never was
implemented. We can safely remove it, because the kernel has grown
to have many more reliable mechanisms to determine if device is
supported or not.
Signed-off-by: Leon Romanovsky <leonro@nvidia.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The problem was in wrong "if" placement. chip->quirk_type is freed
in snd_card_free_when_closed(), but inside if statement it's accesed.
Fixes: 9799110825 ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()")
Signed-off-by: Pavel Skripkin <paskripkin@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/16da19126ff461e5e64a9aec648cce28fb8ed73e.1615242183.git.paskripkin@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Other Plantronics headset models seem requiring the same workaround as
C320-M to add the 20ms delay for the control messages, too. Apply the
workaround generically for devices with the vendor ID 0x047f.
Note that the problem didn't surface before 5.11 just with luck.
Since 5.11 got a big code rewrite about the stream handling, the
parameter setup procedure has changed, and this seemed triggering the
problem more often.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1182552
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304085009.4770-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rear audio on Lenovo ThinkStation P620 stops working after commit
1965c4364b ("ALSA: usb-audio: Disable autosuspend for Lenovo
ThinkStation P620"):
[ 6.013526] usbcore: registered new interface driver snd-usb-audio
[ 6.023064] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023083] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023090] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023098] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.023103] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.023110] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045846] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045866] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045877] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045886] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
[ 6.045894] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x0, type = 1
[ 6.045908] usb 3-6: cannot get ctl value: req = 0x81, wValue = 0x202, wIndex = 0x0, type = 4
I overlooked the issue because when I was working on the said commit,
only the front audio is tested. Apology for that.
Changing supports_autosuspend in driver is too late for disabling
autosuspend, because it was already used by USB probe routine, so it can
break the balance on the following code that depends on
supports_autosuspend.
Fix it by using usb_disable_autosuspend() helper, and balance the
suspend count in disconnect callback.
Fixes: 1965c4364b ("ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210304043419.287191-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit only contains the fix about the `URB_CONTROL` request
direction to set the samplerate of Pioneer DJM devices (`URB_CONTROL out`).
Fixes: 3b85f5fc75 ("ALSA: usb-audio: Add DJM450 to Pioneer format quirk")
Signed-off-by: Nicolas MURE <nicolas.mure2019@gmail.com>
Link: https://lore.kernel.org/r/20210301142927.14552-1-nicolas.mure2019@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Corsair Virtuoso SE RGB Wireless is a USB headset with a mic and a
sidetone feature. Assign the Corsair Virtuoso name map to the SE product
ids as well, in order to label its mixer appropriately and allow
userspace to pick the correct volume controls.
Signed-off-by: Andrea Fagiani <andfagiani@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/40bbdf55-f854-e2ee-87b4-183e6451352c@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the hw constraints for implicit feedback streams
via commit e4ea77f8e5 ("ALSA: usb-audio: Always apply the hw
constraints for implicit fb sync") added the check of the matching
endpoints and whether those EPs are already opened. This is needed
and correct, per se, even for the normal streams without the implicit
feedback, as the endpoint setup is exclusive.
However, it's reported that there seem applications that behave in
unexpected ways to update the hw_params without clearing the previous
setup via hw_free, and those hit a problem now: then hw_params is
called with still the previous EP setup kept, hence it's restricted
with the previous own setup. Although the obvious fix is to call
snd_pcm_hw_free() API in the application side, it's a kind of
unwelcome change.
This patch tries to ease the situation: in the endpoint check, we add
a couple of more conditions and now skip the endpoint that is being
used only by the stream in question itself. That is, in addition to
the presence check of ep (ep->cur_audiofmt is non-NULL), when the
following conditions are met, we skip such an ep:
- ep->opened == 1, and
- ep->cur_audiofmt == subs->cur_audiofmt.
subs->cur_audiofmt is non-NULL only if it's a re-setup of hw_params,
and ep->cur_audiofmt points to the currently set up parameters. So if
those match, it must be this stream itself.
Fixes: e4ea77f8e5 ("ALSA: usb-audio: Always apply the hw constraints for implicit fb sync")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211941
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210228080138.9936-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some USB audio firmware seem to report broken dB values for the volume
controls, and this screws up applications like PulseAudio who blindly
trusts the given data. For example, Edifier G2000 reports a PCM
volume from -128dB to -127dB, and this results in barely inaudible
sound.
This patch adds a sort of sanity check at parsing the dB values in
USB-audio driver and disables the dB reporting if the range looks
bogus. Here, we assume -96dB as the bottom line of the max dB.
Note that, if one can figure out that proper dB range later, it can be
patched in the mixer maps.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211929
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210227105737.3656-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 93db51d06b ("ALSA: usb-audio: Check valid altsetting at
parsing rates for UAC2/3") changed the behavior of the function
set_sample_rate_v2v3() slightly to treat the inconsistent sample rate
as an error. It was done by assumption that the sample rate
validation should have been done at the parser phase as implemented in
that patch. But the validation is later selectively enabled only for
certain devices as it causes a regression (the commit fe773b8711
"ALSA: usb-audio: workaround for iface reset issue"), and now the
inconsistency surfaced as a fatal error while it worked in the past as
is, as reported for FiiO M3K DAC.
For recovering from the regression, change set_sample_rate_v2v3()
again to ignore the sample rate difference as non-error.
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1182633
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210227082002.21185-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BOSS GP-10 with 0582:0185 requires the similar quirk to make the
implicit feedback working like other BOSS devices.
Reported-by: Keith Milner <kamilner@superlative.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210214154251.10750-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the later patch, we're going to issue the PCM sync_stop calls at
disconnection. But currently the USB-audio driver can't handle it
because it has a check of shutdown flag for stopping the URBs. This
is basically superfluous (the stopping URBs are safe at disconnection
state), so let's drop the check.
Fixes: dc5eafe778 ("ALSA: usb-audio: Support PCM sync_stop")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210206203052.15606-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The endpoint management has bit flags to indicate the current state,
and we're dealing two things: the running bit and the stopping bit.
There is a thin window in transition from the running to the stopping
in stop_urbs(), and as long as the bit flags are used, it's difficult
to plug.
This patch modifies the state management code to use the atomic int
and follow the explicit three states, STOPPED, RUNNING and STOPPING.
The state change is done via atomic_cmpxhg() for avoiding possible
races, and check the state change more strictly. The unexpected state
change is now handled as an error.
Fixes: d0f09d1e4a ("ALSA: usb-audio: Refactoring endpoint URB deactivation")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210206203052.15606-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we stop an endpoint in release_urbs(), it ignores the
inconsistent endpoint state and tries to release the resources.
This shouldn't happen in theory, but it's still safer to abort the
release and let the caller proper error handling.
Also, stop_and_unlink_urbs() called from release_urbs() does two step
works, and it's more straightforward to split this to two functions
again, so that the call from the PCM trigger won't take the path with
sleeping.
This patch modifies the EP management code to adapt two points above.
Fixes: d0f09d1e4a ("ALSA: usb-audio: Refactoring endpoint URB deactivation")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210206203052.15606-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds mixer quirks for the Pioneer DJM-900NXS2 mixer. This
device has 6 capture channels, 5 of them allow setting the signal
source. This adds controls for these, similar to the DJM-250Mk2.
However, playpack channels are not controllable via software like on the
250Mk2, as they can only be set manually on the mixing console.
Read-only controls showing the currently selected playback channels are
omitted.
Signed-off-by: Fabian Lesniak <fabian@lesniak-it.de>
Link: https://lore.kernel.org/r/20210205215116.258724-2-fabian@lesniak-it.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows for N different devices to use the pioneer mixer quirk for
setting capture/record type and recording level. The impementation has
not changed much with the exception of an additional mask on
private_value to allow storing of a device index:
DEVICE MASK 0xff000000
GROUP_MASK 0x00ff0000
VALUE_MASK 0x0000ffff
This could be improved by changing the arrays of wValues for each
channel to contain named definitions (e.g. SND_DJM_CAP_LINE). It would
improve readability and perhaps would allow using the same array for
multiple channels. The channel number can be specified on the control
next to the wIndex.
Feedback is very much appreciated as I'm not the most proficient C
programmer but am learning as I go.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210205184256.10201-2-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit f274baa49b ("ALSA: usb-audio: Allow non-vmalloc buffer
for PCM buffers") introduced the mode to allocate coherent pages for
PCM buffers, and it used bus->controller device as its DMA device.
It turned out, however, that bus->sysdev is a more appropriate device
to be used for DMA mapping in HCD code.
This patch corrects the device reference accordingly.
Note that, on most platforms, both point to the very same device,
hence this patch doesn't change anything practically. But on
platforms like xhcd-plat hcd, the change becomes effective.
Fixes: f274baa49b ("ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210205144559.29555-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The kerndoc comment for the new function snd_usb_endpoint_free_all()
had a typo wrt the argument name. Fix it.
Fixes: 00272c6182 ("ALSA: usb-audio: Avoid unnecessary interface re-setup")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210205082837.6327-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As with most Pioneer devices, the device descriptor is vendor specific
and as such, the number of channels, the PCM format, endpoints and
sample rate need to be specified. This device has 8 inputs and 8 outputs
and a sample rate of 48000 only. The PCM format is S24_3LE like other
devices.
There seems to be an appetite for reducing duplication amongs these
Pioneer patches but again, I feel this is a step to be taken after
support has been added as it's not completely clear where the
commonalities are.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210202134225.3217-3-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the DJM-750, ensure that the format control message is passed to
the device when opening a stream. It seems as though fmt->sync_ep is not
always set when this function is called hence the passing of the value
at the call site. If this can be fixed, fmt->sync_up should be used as
the wvalue.
There doesn't seem to be a "cpu_to_le24" type function defined hence for
the open code but I did see a similar thing done in Bluez lib. Perhaps
we can get these definitions defined in byteorder.h. See hci_cpu_to_le24
in include/net/bluetooth/hci.h:2543 for similar usage.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210202134225.3217-2-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced sample rate validation code seems causing a
problem on some devices; namely, after performing this, the bus gets
screwed and it influences even on other USB devices.
As a quick workaround, perform it only for the necessary devices;
currently MOTU devices are known to need the valid altset checks, so
filter out other devices.
Fixes: 93db51d06b ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3")
Reported-by: Jamie Heilman <jamie@audible.transient.net>
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1178203
Link: https://lore.kernel.org/r/20210123155842.22652-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At probing a UAC2/UAC3 device like NUX MG-300 USB interface, we get
error messages "RANGE setting not yet supported". It comes the place
where the driver tries to determine the resolution of mixer volumes
via SET_CUR_RES and GET_CUR_RES verbs. Those verbs aren't supported
on UAC2 and UAC3, hence the driver warns like the above. Although the
driver handles this error and works as expected, it's still ugly to
show such errors unnecessarily.
This patch papers over the errors by applying the resolution detection
only for UAC1 and skipping it for UAC2/UAC3.
Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current USB-audio driver gets an error at probing NUX MG-300 about
parsing the clocks. This is because the firmware doesn't return the
proper connection of the clock selector that is connected to a single
clock; it's likely that the firmware was lazy^w optimized and the
inquiry wasn't handled. Actually it makes little sense to inquire and
set up the single connection explicitly.
This patch fixes the issue by simply skipping the clock selector
inquiry if it's a single connection.
Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent refactoring, it's been reported that some USB-audio
devices (typically webcams) are no longer detected properly by
PulseAudio. The debug session revealed that it's failing at probing
by PA to try the sample rate 44.1kHz while the device has discrete
sample rates other than 44.1kHz. But the puzzle was that arecord
works as is, and some other devices with the discrete rates work,
either.
After all, this turned out to be the lack of the dependencies in a few
hw constraint rules: snd_pcm_hw_rule_add() has the (variable)
arguments specifying the dependent parameters, and some functions
didn't set the target parameter itself as the dependencies. This
resulted in an invalid parameter that could be generated only in a
certain call pattern. This bug itself has been present in the code,
but it didn't trigger errors just because the rules were casually
avoiding such a corner case. After the recent refactoring and
cleanup, however, the hw constraints work "as expected", and the
problem surfaced now.
For fixing the problem above, this patch adds the missing dependent
parameters to each snd_pcm_hw_rule() call.
Fixes: bc4e94aa8e ("ALSA: usb-audio: Handle discrete rates properly in hw constraints")
BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014
Link: https://lore.kernel.org/r/20210120204554.30177-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds the Pioneer DJ DJM-750 to the quirks table and ensures
skip_pioneer_sync_ep() is (also) called: this device uses the vendor
ID of 0x08e4 (I'm not sure why they use multiple vendor IDs but many
just like to be awkward it seems).
Playback on all 8 channels works. I'll likely keep this working in the
future and submit futher patches and improvements as necessary.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210118130621.77miiie47wp7mump@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For addressing the regression on Pioneer devices, we recently
corrected the quirk code to enable the implicit feedback mode on those
devices properly. However, the devices still showed problems with the
full duplex operations with JACK, and after debug sessions, we figured
out that the older kernels that had worked with JACK also didn't use
the implicit feedback mode at all although they had the quirk code to
enable it; instead, the old code worked just to skip the normal sync
endpoint setup that would have been detected without it. IOW, what
broke without the implicit-fb quirk in the past was the application of
the normal sync endpoint that is actually the capture data endpoint on
these devices.
This patch covers the overseen piece: it modifies the quirk code again
not to enable the implicit feedback mode but just to make the driver
skipping the sync endpoint detection. This made the driver working
with JACK full-duplex mode again.
Still it's not quite clear why the implicit feedback doesn't work on
those devices yet; maybe it's about some issues in the URB setup. But
at least, with this patch, the driver should work in the level of the
older kernels again.
Fixes: 167c9dc84e ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices")
Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC2/3 sample rate setup is based on the clock node, which is
usually shared in the interface, and can't be re-setup without
deselecting the interface once, and that's how the current code
behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence
we basically need to call for each endpoint usage even if those share
the same interface.
This patch fixes the behavior of UAC1 to call always
snd_usb_init_sample_rate() in snd_usb_endpoint_configure().
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current sample rate setup function for UAC1 assumes only the first
endpoint retrieved from the interface:altset pair, but the rate set up
may be needed also for the secondary endpoint. Also, retrieving the
endpoint number from the interface descriptor is redundant; we have
already the target endpoint in the given audioformat object.
This patch simplifies the code and corrects the target endpoint as
described in the above. It simply refers to fmt->endpoint directly.
Also, this patch drops the pioneer_djm_set_format_quirk() that is
caleld from snd_usb_set_format_quirk(); this function does the sample
rate setup but for the capture endpoint (0x82), and that's exactly
what the change above fixes.
Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last remaining usage of strlcpy() in USB-audio driver is the setup
of the card longname string. Basically we need to know whether any
non-empty string is set or not, and no real length is needed.
Refactor the code and use strscpy() instead. After this change,
strlcpy() is gone from all sound/* code.
Link: https://lore.kernel.org/r/20210115100437.20906-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver still contains two calls of strlcpy() because the
return size is evaluated. Basically it just checks whether the string
is copied or not, but since strcpy() may return a negative error code,
we should check the negative value and treat as filled.
Link: https://lore.kernel.org/r/20210115095758.19707-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the commit 5a6c3e11c9 ("ALSA: usb-audio: Add hw constraint for
implicit fb sync"), we apply the hw constraints for the implicit
feedback sync to make the secondary open aligned with the already
opened stream setup. This change assumed that the secondary open is
performed after the first stream has been already set up, and adds the
hw constraints to sync with the first stream's parameters only when
the EP setup for the first stream was confirmed at the open time.
However, most of applications handling the full-duplex operations do
open both playback and capture streams at first, then set up both
streams. This results in skipping the additional hw constraints since
the counter-part stream hasn't been set up yet at the open of the
second stream, and it eventually leads to "incompatible EP" error in
the end.
This patch corrects the behavior by always applying the hw constraints
for the implicit fb sync. The hw constraint rules are defined so that
they check the sync EP dynamically at each invocation, instead. This
covers the concurrent stream setups better and lets the hw refine
calls resolving to the right configuration.
Also this patch corrects a minor error that has existed in the debug
print that isn't built as default.
Fixes: 5a6c3e11c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pioneer devices have both playback and capture streams sharing the
same iface/altsetting, and those need to be paired as implicit
feedback. Instead of a half-baked (and broken) static quirk entry,
set up more generically for those devices by checking the number of
endpoints and the attribute of the secondary EP.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are devices that have multiple endpoints sharing the same
iface/altset not only for sync but also for the actual streams, and
the audioformat for such an endpoint needs to be handled with the
proper endpoint index; otherwise it confuses the endpoint management.
This patch extends the audioformat to annotate the endpoint index, and
put the proper ep_idx=1 to Pioneer device quirk entries accordingly.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current endpoint handling assumed (more or less) a unique 1:1
relation between the endpoint and the iface/altset. The exception was
the sync EP without the implicit feedback which has usually the
secondary EP of the same altset. This works fine for most devices,
but it turned out that some unusual devices like Pinoeer's ones have
both playback and capture endpoints in the same iface/altsetting and
use both for the implicit feedback mode. For handling such a case, we
need to extend the endpoint management to take the shared interface
into account.
This patch does that: it adds a new object snd_usb_iface_ref for
managing the reference counts of the each USB interface that is used
by each endpoint. The interface setup is performed only once for the
(sharing) endpoints, and the doubly initialization is avoided.
Along with this, the resource release of endpoints and interface
refcounts are put into a single function, snd_usb_endpoint_free_all()
instead of looping in the caller side.
Fixes: bf6313a0ff ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The implicit feedback mode needs to handle two endpoints and the
choice of the audioformat object for the sync EP is important since
this determines the compatibility of the hw_params. The current code
uses the same audioformat object if both the main EP and the sync EP
point to the same iface/altsetting. This was done in consideration of
the non-implicit-fb sync EP handling, and it doesn't match well with
the cases where actually to endpoints are defined in the sameiface /
altsetting like a few Pioneer devices.
Modify snd_usb_find_implicit_fb_sync_format() to pick up the
audioformat that is assigned in the counter-part substreams primarily,
so that the actual capture stream can be opened properly. We keep the
same audioformat object only as a fallback in case nothing found,
though.
Fixes: 9fddc15e80 ("ALSA: usb-audio: Factor out the implicit feedback quirk code")
Link: https://lore.kernel.org/r/20210108075219.21463-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change in the endpoint management moved the endpoint object
creation from the stream open time to the parser of the audio
descriptor. It works fine for the standard audio, but it overlooked
the other places that create audio streams via quirks
(QUIRK_AUDIO_FIXED_ENDPOINT) like the reported a few Pioneer devices;
those call snd_usb_add_audio_stream() manually, hence they miss the
endpoints, eventually resulting in the error at opening streams.
Moreover, now the sync EP setup was moved to the explicit call of
snd_usb_audioformat_set_sync_ep(), and this needs to be added for
those places, too.
This patch addresses those regressions for quirks. It adds a local
helper function add_audio_stream_from_fixed_fmt(), which does the all
needed tasks, and replaces the calls of snd_usb_add_audio_stream()
with this new function.
Fixes: 54cb31901b ("ALSA: usb-audio: Create endpoint objects at parsing phase")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
strlcpy is deprecated. see: Documentation/process/deprecated.rst
Change the calls that do not use the strlcpy return value to the
preferred strscpy.
Done with cocci script:
@@
expression e1, e2, e3;
@@
- strlcpy(
+ strscpy(
e1, e2, e3);
This cocci script leaves the instances where the return value is
used unchanged.
After this patch, sound/ has 3 uses of strlcpy() that need to be
manually inspected for conversion and changed one day.
$ git grep -w strlcpy sound/
sound/usb/card.c: len = strlcpy(card->longname, s, sizeof(card->longname));
sound/usb/mixer.c: return strlcpy(buf, p->name, buflen);
sound/usb/mixer.c: return strlcpy(buf, p->names[index], buflen);
Miscellenea:
o Remove trailing whitespace in conversion of sound/core/hwdep.c
Link: https://lore.kernel.org/lkml/CAHk-=wgfRnXz0W3D37d01q3JFkr_i_uTL=V6A6G1oUZcprmknw@mail.gmail.com/
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/22b393d1790bb268769d0bab7bacf0866dcb0c14.camel@perches.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BOSS RC-505 (shown by lsusb as "Roland Corp. RC-505") does require the
same quirk as these other BOSS devices.
Without this quirk it is neither possible to capture audio from nor to
write audio to the RC-505. Both just result in an empty audio
stream. With these changes both capture and playback seem to work
quite fine. MIDI funtionality was not tested.
Tested-by: Harry Reinold <harry.reinold@posteo.de>
Signed-off-by: Timon Reinold <tirei@agon.one>
Link: https://lore.kernel.org/r/20210102210835.21268-1-tirei@agon.one
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BOSS AD-10 requires the very same quirk like other BOSS devices to
enable the special implicit feedback mode.
Reported-and-tested-by: Martin Passing <martin@passing.name>
Link: https://lore.kernel.org/r/20201229083428.20467-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DIV_ROUND_UP() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.
Generated using the following the Coccinelle semantic patch.
// <smpl>
@@
expression x, y;
@@
-(((x) + (y) - 1) / (y))
+DIV_ROUND_UP(x, y)
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-11-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calculation of in_cables and out_cables bitmaps are done with the
bit shift by the value from the descriptor, which is an arbitrary
value, and can lead to UBSAN shift-out-of-bounds warnings.
Fix it by filtering the bad descriptor values with the check of the
upper bound 0x10 (the cable bitmaps are 16 bits).
Reported-by: syzbot+92e45ae45543f89e8c88@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201223174557.10249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
