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>
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>
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>
Add initial support for the Focusrite Scarlett Solo and 2i2 devices:
- They have no mixer
- They don't support reporting sync status or levels
- The configuration space is laid out differently to the other models
- There is no level (line/inst) switch on input 1 of the Solo
Co-developed-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Vladimir Sadovnikov <sadko4u@gmail.com>
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/190b90f6f1f8f8d4dfb5f0a7761ff8ae5c40fdde.1624379707.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add separate init function to call the existing controls_create
function so a custom error can be displayed if initialisation fails.
Use info level instead of error for notifications.
Display the VID/PID so device_setup is targeted to the right device.
Display "enabled" message to easily confirm that the driver is loaded.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/b5d140c65f640faf2427e085fbbc0297b32e5fce.1621584566.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'tags/mute-led-rework' into for-next
ALSA: control - add generic LED API
This patchset tries to resolve the diversity in the audio LED
control among the ALSA drivers. A new control layer registration
is introduced which allows to run additional operations on
top of the elementary ALSA sound controls.
A new control access group (three bits in the access flags)
was introduced to carry the LED group information for
the sound controls. The low-level sound drivers can just
mark those controls using this access group. This information
is not exported to the user space, but user space can
manage the LED sound control associations through sysfs
(last patch) per Mark's request. It makes things fully
configurable in the kernel and user space (UCM).
The actual state ('route') evaluation is really easy
(the minimal value check for all channels / controls / cards).
If there's more complicated logic for a given hardware,
the card driver may eventually export a new read-only
sound control for the LED group and do the logic itself.
The new LED trigger control code is completely separated
and possibly optional (there's no symbol dependency).
The full code separation allows eventually to move this
LED trigger control to the user space in future.
Actually it replaces the already present functionality
in the kernel space (HDA drivers) and allows a quick adoption
for the recent hardware (ASoC codecs including SoundWire).
snd_ctl_led 24576 0
The sound driver implementation is really easy:
1) call snd_ctl_led_request() when control LED layer should be
automatically activated
/ it calls module_request("snd-ctl-led") on demand /
2) mark all related kcontrols with
SNDRV_CTL_ELEM_ACCESS_SPK_LED or
SNDRV_CTL_ELEM_ACCESS_MIC_LED
Link: https://lore.kernel.org/r/20210317172945.842280-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rear Mic on Lenovo P620 cannot record after S3, despite that there's no
error and the other two functions of the USB audio, Line In and Line
Out, work just fine.
The mic starts to work again after running userspace app like "alsactl
store". Following the lead, the evidence shows that as soon as connector
status is queried, the mic can work again.
So also check connector value on resume to "wake up" the USB audio to
make it functional.
This can be device specific, however I think this generic approach may
benefit more than one device.
Now the resume callback checks connector, and a new callback,
reset_resume, to also restore switches and volumes.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210325165918.22593-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Unlike the other DJM, the value to set the "CD/LINE" and "LINE" capture
control options are inverted. This fix makes sure that the displayed
info label while using `alsamixer` matches the input switches label
on the DJM-850 mixer.
Signed-off-by: Nicolas MURE <nicolas.mure2019@gmail.com>
Link: https://lore.kernel.org/r/20210301152729.18094-5-nicolas.mure2019@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit 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>
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>
This patch extends support for DJM-250MK2 and allows mapping
playback and capture channels to available sources.
Configures the card through USB commands.
Signed-off-by: František Kučera <franta-linux@frantovo.cz>
Link: https://lore.kernel.org/r/20200922144206.10472-1-konference@frantovo.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Signed-off-by: Mirko Dietrich <buzz@l4m1.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element. A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members. This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.
This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.
Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com
Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As reported by kbuild test robot, mixer quirks for RME Babyface
Pro used plain integer instead of NULL.
Fixes: 3e8f3bd047 ("ALSA: usb-audio: RME Babyface Pro mixer patch")
Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Reported-by: kbuild test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20200529173248.zzawijfvw73kzjxt@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which
increases the refcount of the snd_usb_audio object "chip".
When snd_microii_spdif_default_get() returns, local variable "chip"
becomes invalid, so the refcount should be decreased to keep refcount
balanced.
The reference counting issue happens in several exception handling paths
of snd_microii_spdif_default_get(). When those error scenarios occur
such as usb_ifnum_to_if() returns NULL, the function forgets to decrease
the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak.
Fix this issue by jumping to "end" label when those error scenarios
occur.
Fixes: 447d6275f0 ("ALSA: usb-audio: Add sanity checks for endpoint accesses")
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1587617711-13200-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added mixer quirks to allow controlling the internal DSP of the
RME Babyface Pro and its successor Babyface Pro FS.
Signed-off-by: Thomas Ebeling <penguins@bollie.de>
Link: https://lore.kernel.org/r/20200414211019.qprg7whepg2y7nei@bollie.ca9.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for Presonus Studio 1810c, a usb interface
that's UAC2 compliant with a few quirks and a few extra hw-specific
controls. I've tested all 3 altsettings and the added switch
controls and they work as expected.
More infos on the card:
https://www.presonus.com/products/Studio-1810c
Note that this work is based on packet inspection with
usbmon. I just wanted to get this card to work for using
it on our open-source radio station:
https://github.com/UoC-Radio
v2 address issues reported by Takashi:
* Properly get/set enum type controls
* Prevent race condition on switch_get/set
* Various control naming changes
* Various coding style fixes
v3 improve readability of sample rate filtering
and some other minor changes.
Signed-off-by: Nick Kossifidis <mickflemm@gmail.com>
Link: https://lore.kernel.org/r/5e47481a.1c69fb81.befb3.8dac@mx.google.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
GCC reports the following warning with W=1
sound/usb/mixer_quirks.c: In function ‘snd_microii_controls_create’:
sound/usb/mixer_quirks.c:1694:2: warning: ‘static’ is not at beginning
of declaration [-Wold-style-declaration]
1694 | const static usb_mixer_elem_resume_func_t resume_funcs[] = {
| ^~~~~
Move static to the beginning of declaration
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200111214736.3002-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply const prefix to the remaining places: the static table for the
unit information, the mixer maps, the validator tables, etc.
Just for minor optimization and no functional changes.
Link: https://lore.kernel.org/r/20200105144823.29547-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of snd_kcontrol_new definitions are read-only and passed as-is.
Let's declare them as const for further optimization.
There should be no functional changes by this patch.
Link: https://lore.kernel.org/r/20200103081714.9560-42-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull USB validation patches. It's based on the latest 5.3 development
branch, so we shall catch up the whole things.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk function snd_emuusb_set_samplerate() has a NULL check for
the mixer element, but this is useless in the current code. It used
to be a check against mixer->id_elems[unitid] but it was changed later
to the value after mixer_eleme_list_to_info() which is always non-NULL
due to the container_of() usage.
This patch fixes the check before the conversion.
While we're at it, correct a typo in the comment in the function,
too.
Fixes: 8c558076c7 ("ALSA: usb-audio: Clean up mixer element list traverse")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2
audio interfaces. Although the interfaces are USB compliant,
additional input/output level controls and hardware routing/mixing
functionality are available using proprietary USB requests.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a very big update, mainly thanks to Morimoto-san's refactoring
work and some fairly large new drivers.
- Lots more work on moving towards a component based framework from
Morimoto-san.
- Support for force disconnecting muxes from Jerome Brunet.
- New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
CX2072X, Realtek RT1011 and RT1308.
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Merge tag 'asoc-v5.3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.3
This is a very big update, mainly thanks to Morimoto-san's refactoring
work and some fairly large new drivers.
- Lots more work on moving towards a component based framework from
Morimoto-san.
- Support for force disconnecting muxes from Jerome Brunet.
- New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
CX2072X, Realtek RT1011 and RT1308.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a couple of left shifts of unsigned 8 bit values that
first get promoted to signed ints and hence get sign extended
on the shift if the top bit of the 8 bit values are set. Fix
this by casting the 8 bit values to unsigned ints to stop the
unintentional sign extension.
Addresses-Coverity: ("Unintended sign extension")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Based on 1 normalized pattern(s):
this program is free software you can redistribute it and or modify
it under the terms of the gnu general public license as published by
the free software foundation either version 2 of the license or at
your option any later version this program is distributed in the
hope that it will be useful but without any warranty without even
the implied warranty of merchantability or fitness for a particular
purpose see the gnu general public license for more details you
should have received a copy of the gnu general public license along
with this program if not write to the free software foundation inc
59 temple place suite 330 boston ma 02111 1307 usa
extracted by the scancode license scanner the SPDX license identifier
GPL-2.0-or-later
has been chosen to replace the boilerplate/reference in 1334 file(s).
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Allison Randal <allison@lohutok.net>
Reviewed-by: Richard Fontana <rfontana@redhat.com>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190527070033.113240726@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds several vendor specific mixer quirks for RME's Class Compliant
USB devices. These provide extra status information from the device
otherwise not available.
These include AES/SPDIF rate and status information, current system
sampling rate and measured frequency. This information is especially
useful in cases where device's clock is slaved to external clock
source.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_usb_feature_unit_ctl properly in mixer.h. Otherwise it's
error-prone.
This fixes the sparse warning:
sound/usb/mixer.c:1464:25: warning: symbol 'snd_usb_feature_unit_ctl' was not declared. Should it be static?
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check on err is redundant as both the true and false paths
end up on a break statement. Remove the redundant check.
Detected by CoverityScan, CID#1268773 ("Identical code for different
branches")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid if ((err = ...) style and expand to multiple lines instead.
No change in the end result, but just the beautification.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new macro for iterating over mixer element list for
avoiding the open codes in many places. Also the open-coded
container_of() and the forced cast to struct usb_mixer_elem_info are
replaced with another simple macro, too.
No functional changes but just readability improvement.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the early commit adcdd0d5a1 ("ALSA: usb-audio: Skip volume
controls triggers hangup on Dell USB Dock"), we add the mixer quirks
for Dell dock to skip two mixer FU's for playback. This supposed that
the device has always the proper initial volume, but it doesn't seem
always correct.
This patch adds the explicit initialization of the volumes to the
fixed 0dB at the device probe time. Also, such a fixup is needed
after the resume, so a new function is hooked to the resume callback
as well.
Bugzilla: http://bugzilla.suse.com/show_bug.cgi?id=1089467
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the mindless scripted replacement of kernel use of POLL*
variables as described by Al, done by this script:
for V in IN OUT PRI ERR RDNORM RDBAND WRNORM WRBAND HUP RDHUP NVAL MSG; do
L=`git grep -l -w POLL$V | grep -v '^t' | grep -v /um/ | grep -v '^sa' | grep -v '/poll.h$'|grep -v '^D'`
for f in $L; do sed -i "-es/^\([^\"]*\)\(\<POLL$V\>\)/\\1E\\2/" $f; done
done
with de-mangling cleanups yet to come.
NOTE! On almost all architectures, the EPOLL* constants have the same
values as the POLL* constants do. But they keyword here is "almost".
For various bad reasons they aren't the same, and epoll() doesn't
actually work quite correctly in some cases due to this on Sparc et al.
The next patch from Al will sort out the final differences, and we
should be all done.
Scripted-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Pull poll annotations from Al Viro:
"This introduces a __bitwise type for POLL### bitmap, and propagates
the annotations through the tree. Most of that stuff is as simple as
'make ->poll() instances return __poll_t and do the same to local
variables used to hold the future return value'.
Some of the obvious brainos found in process are fixed (e.g. POLLIN
misspelled as POLL_IN). At that point the amount of sparse warnings is
low and most of them are for genuine bugs - e.g. ->poll() instance
deciding to return -EINVAL instead of a bitmap. I hadn't touched those
in this series - it's large enough as it is.
Another problem it has caught was eventpoll() ABI mess; select.c and
eventpoll.c assumed that corresponding POLL### and EPOLL### were
equal. That's true for some, but not all of them - EPOLL### are
arch-independent, but POLL### are not.
The last commit in this series separates userland POLL### values from
the (now arch-independent) kernel-side ones, converting between them
in the few places where they are copied to/from userland. AFAICS, this
is the least disruptive fix preserving poll(2) ABI and making epoll()
work on all architectures.
As it is, it's simply broken on sparc - try to give it EPOLLWRNORM and
it will trigger only on what would've triggered EPOLLWRBAND on other
architectures. EPOLLWRBAND and EPOLLRDHUP, OTOH, are never triggered
at all on sparc. With this patch they should work consistently on all
architectures"
* 'misc.poll' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (37 commits)
make kernel-side POLL... arch-independent
eventpoll: no need to mask the result of epi_item_poll() again
eventpoll: constify struct epoll_event pointers
debugging printk in sg_poll() uses %x to print POLL... bitmap
annotate poll(2) guts
9p: untangle ->poll() mess
->si_band gets POLL... bitmap stored into a user-visible long field
ring_buffer_poll_wait() return value used as return value of ->poll()
the rest of drivers/*: annotate ->poll() instances
media: annotate ->poll() instances
fs: annotate ->poll() instances
ipc, kernel, mm: annotate ->poll() instances
net: annotate ->poll() instances
apparmor: annotate ->poll() instances
tomoyo: annotate ->poll() instances
sound: annotate ->poll() instances
acpi: annotate ->poll() instances
crypto: annotate ->poll() instances
block: annotate ->poll() instances
x86: annotate ->poll() instances
...
The E1 has two headphone jacks, one of which can be set as a microphone
input. In the default mode, it uses the built-in microphone as an input.
By sending a special command, the second headphone jack is instead used
as an input.
This might work with the E3 as well, but I don't have one of those to
test it.
Signed-off-by: Ian Douglas Scott <ian@iandouglasscott.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
C-Media devices (at least some models) mute the playback stream when
volumes are set to the minimum value. But this isn't informed via TLV
and the user-space, typically PulseAudio, gets confused as if it's
still played in a low volume.
This patch adds the new flag, min_mute, to struct usb_mixer_elem_info
for indicating that the mixer element is with the minimum-mute volume.
This flag is set for known C-Media devices in
snd_usb_mixer_fu_apply_quirk() in turn.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196669
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new structures as const as they are only passed an
argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const.
Done using Coccinelle:
@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};
@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)
@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p
@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;
Cross compiled these files:
sound/aoa/codecs/tas.c - powerpc
sound/mips/{hal2.c/sgio2audio.c} - mips
sound/ppc/{awacs.c/beep.c/tumbler.c} - powerpc
sound/soc/sh/siu_dai.c - sh
Could not find an architecture to compile sound/sh/aica.c.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer quirk for Tascam US-16x08 usb interface.
Even that this is an usb compliant device,
the input channels and DSP functions (EQ/Compressor) aren't accessible
by default.
Signed-off-by: Detlef Urban <onkel@paraair.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DragonFly quirk added in 42e3121d90 ("ALSA: usb-audio: Add a more
accurate volume quirk for AudioQuest DragonFly") applies a custom dB map
on the volume control when its range is reported as 0..50 (0 .. 0.2dB).
However, there exists at least one other variant (hw v1.0c, as opposed
to the tested v1.2) which reports a different non-sensical volume range
(0..53) and the custom map is therefore not applied for that device.
This results in all of the volume change appearing close to 100% on
mixer UIs that utilize the dB TLV information.
Add a fallback case where no dB TLV is reported at all if the control
range is not 0..50 but still 0..N where N <= 1000 (3.9 dB). Also
restrict the quirk to only apply to the volume control as there is also
a mute control which would match the check otherwise.
Fixes: 42e3121d90 ("ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly")
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: David W <regulars@d-dub.org.uk>
Tested-by: David W <regulars@d-dub.org.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some sanity check codes before actually accessing the endpoint via
get_endpoint() in order to avoid the invalid access through a
malformed USB descriptor. Mostly just checking bNumEndpoints, but in
one place (snd_microii_spdif_default_get()), the validity of iface and
altsetting index is checked as well.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [da6d276957ea: ALSA: usb-audio: Add resume support for
Native Instruments controls] brought a regression where the Native
Instrument audio devices don't get the correct value at update due to
the missing shift at writing. This patch addresses it.
Fixes: da6d276957 ('ALSA: usb-audio: Add resume support for Native Instruments controls')
Reported-and-tested-by: Owen Williams <owilliams@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AudioQuest DragonFly DAC reports a volume control range of 0..50
(0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which
is obviously incorrect and would cause software using the dB information
in e.g. volume sliders to have a massive volume difference in 100..102%
range.
Commit 2d1cb7f658 ("ALSA: usb-audio: add dB range mapping for some
devices") added a dB range mapping for it with range 0..50 dB.
However, the actual volume mapping seems to be neither linear volume nor
linear dB scale, but instead quite close to the cubic mapping e.g.
alsamixer uses, with a range of approx. -53...0 dB.
Replace the previous quirk with a custom dB mapping based on some basic
output measurements, using a 10-item range TLV (which will still fit in
alsa-lib MAX_TLV_RANGE_SIZE).
Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the
range is 0..50, so if this gets fixed/changed in later HW revisions it
will no longer be applied.
v2: incorporated Takashi Iwai's suggestion for the quirk application
method
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
William Overton