The layout of the UAC2 Control request and response varies depending on
the request type. With the current implementation, only the Layout 2
Parameter Block (with the 2-byte sized RANGE attribute) is handled
properly. For the Control requests with the 1-byte sized RANGE attribute
(Bass Control, Mid Control, Tremble Control), the response is parsed
incorrectly.
This commit:
* fixes the wLength field value in the request
* fixes parsing the range values from the response
Fixes: 23caaf19b1 ("ALSA: usb-mixer: Add support for Audio Class v2.0")
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add quirk to ensure a sync endpoint is properly configured.
This patch is a fix for same symptoms on Behringer UFX1204 as patch
from Albertto Aquirre on Dec 8 2016 for Axe-Fx II.
Signed-off-by: Lassi Ylikojola <lassi.ylikojola@gmail.com>
Cc: <stable@vger.kernel.org>
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 misc vfs updates from Al Viro:
"All kinds of misc stuff, without any unifying topic, from various
people.
Neil's d_anon patch, several bugfixes, introduction of kvmalloc
analogue of kmemdup_user(), extending bitfield.h to deal with
fixed-endians, assorted cleanups all over the place..."
* 'work.misc' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (28 commits)
alpha: osf_sys.c: use timespec64 where appropriate
alpha: osf_sys.c: fix put_tv32 regression
jffs2: Fix use-after-free bug in jffs2_iget()'s error handling path
dcache: delete unused d_hash_mask
dcache: subtract d_hash_shift from 32 in advance
fs/buffer.c: fold init_buffer() into init_page_buffers()
fs: fold __inode_permission() into inode_permission()
fs: add RWF_APPEND
sctp: use vmemdup_user() rather than badly open-coding memdup_user()
snd_ctl_elem_init_enum_names(): switch to vmemdup_user()
replace_user_tlv(): switch to vmemdup_user()
new primitive: vmemdup_user()
memdup_user(): switch to GFP_USER
eventfd: fold eventfd_ctx_get() into eventfd_ctx_fileget()
eventfd: fold eventfd_ctx_read() into eventfd_read()
eventfd: convert to use anon_inode_getfd()
nfs4file: get rid of pointless include of btrfs.h
uvc_v4l2: clean copyin/copyout up
vme_user: don't use __copy_..._user()
usx2y: don't bother with memdup_user() for 16-byte structure
...
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>
The capture interface does not work, and the playback interface
actually supports only 48kHz unlike what is advertised (44.1, 32, 22,
16, 8).
The only unknown here is if there are other devices that use the same
product ID, but given that this ID is currently unknown, I would assume
it is specially allocated for the nura headset.
Signed-off-by: Martin Peres <martin.peres@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
memdup_user() checks it, so the only effect would be failing with
-EINVAL instead of -EFAULT in case when access_ok() is false.
However, the caller has already checked access_ok() itself (and
would have buggered off with -EFAULT), so the check is completely
pointless. Removing it both simplifies the only instance
of ->dsp_load() and allows to get rid of the check in caller -
its sole effect used to be in preventing a bogus error value
from access_ok() in the instance. Let memdup_user() do the
right thing instead...
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
get_term_name() calls snd_usb_copy_string_desc() for retrieving the
name when a specific ID (name field) is given. When this returns an
error (zero), however, it simply returns as is. This will end up in a
fixed name string in the caller side, which often is meaningless.
For giving a bit more useful name string depending on the terminal
type, change the get_term_name() function to go through the fallback
mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 89b89d121f ("ALSA: usb-audio: Add check return value for
usb_string()") added the check of the return value from
snd_usb_copy_string_desc(), which is correct per se, but it introduced
a regression. In the original code, either the "Clock Source",
"Playback Source" or "Capture Source" suffix is added after the
terminal string, while the commit changed it to add the suffix only
when get_term_name() is failing. It ended up with an incorrect ctl
name like "PCM" instead of "PCM Capture Source".
Also, even the original code has a similar bug: when the ctl name is
generated from snd_usb_copy_string_desc() for the given iSelector, it
also doesn't put the suffix.
This patch addresses these issues: the suffix is added always when no
static mapping is found. Also the patch tries to put more comments
and cleans up the if/else block for better readability in order to
avoid the same pitfall again.
Fixes: 89b89d121f ("ALSA: usb-audio: Add check return value for usb_string()")
Reported-and-tested-by: Mauro Santos <registo.mailling@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds VID:PID of Esoteric D-05X to the TEAC device id's.
Renames the is_teac_50X_dac() function to is_teac_dsd_dac() to cover
broader device family from the same corporation sharing the same USB
audio implementation.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_copy_string_desc() returns zero if usb_string() fails.
In case of failure, we need to check the snd_usb_copy_string_desc()'s
return value and add an exception case
Signed-off-by: Jaejoong Kim <climbbb.kim@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_usb_copy_string_desc() retrieves the usb string corresponding to
the index number through the usb_string(). The problem is that the
usb_string() returns the length of the string (>= 0) when successful, but
it can also return a negative value about the error case or status of
usb_control_msg().
If iClockSource is '0' as shown below, usb_string() will returns -EINVAL.
This will result in '0' being inserted into buf[-22], and the following
KASAN out-of-bound error message will be output.
AudioControl Interface Descriptor:
bLength 8
bDescriptorType 36
bDescriptorSubtype 10 (CLOCK_SOURCE)
bClockID 1
bmAttributes 0x07 Internal programmable Clock (synced to SOF)
bmControls 0x07
Clock Frequency Control (read/write)
Clock Validity Control (read-only)
bAssocTerminal 0
iClockSource 0
To fix it, check usb_string()'return value and bail out.
==================================================================
BUG: KASAN: stack-out-of-bounds in parse_audio_unit+0x1327/0x1960 [snd_usb_audio]
Write of size 1 at addr ffff88007e66735a by task systemd-udevd/18376
CPU: 0 PID: 18376 Comm: systemd-udevd Not tainted 4.13.0+ #3
Hardware name: LG Electronics 15N540-RFLGL/White Tip Mountain, BIOS 15N5
Call Trace:
dump_stack+0x63/0x8d
print_address_description+0x70/0x290
? parse_audio_unit+0x1327/0x1960 [snd_usb_audio]
kasan_report+0x265/0x350
__asan_store1+0x4a/0x50
parse_audio_unit+0x1327/0x1960 [snd_usb_audio]
? save_stack+0xb5/0xd0
? save_stack_trace+0x1b/0x20
? save_stack+0x46/0xd0
? kasan_kmalloc+0xad/0xe0
? kmem_cache_alloc_trace+0xff/0x230
? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio]
? usb_audio_probe+0x4de/0xf40 [snd_usb_audio]
? usb_probe_interface+0x1f5/0x440
? driver_probe_device+0x3ed/0x660
? build_feature_ctl+0xb10/0xb10 [snd_usb_audio]
? save_stack_trace+0x1b/0x20
? init_object+0x69/0xa0
? snd_usb_find_csint_desc+0xa8/0xf0 [snd_usb_audio]
snd_usb_mixer_controls+0x1dc/0x370 [snd_usb_audio]
? build_audio_procunit+0x890/0x890 [snd_usb_audio]
? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio]
? kmem_cache_alloc_trace+0xff/0x230
? usb_ifnum_to_if+0xbd/0xf0
snd_usb_create_mixer+0x25b/0x4b0 [snd_usb_audio]
? snd_usb_create_stream+0x255/0x2c0 [snd_usb_audio]
usb_audio_probe+0x4de/0xf40 [snd_usb_audio]
? snd_usb_autosuspend.part.7+0x30/0x30 [snd_usb_audio]
? __pm_runtime_idle+0x90/0x90
? kernfs_activate+0xa6/0xc0
? usb_match_one_id_intf+0xdc/0x130
? __pm_runtime_set_status+0x2d4/0x450
usb_probe_interface+0x1f5/0x440
Cc: <stable@vger.kernel.org>
Signed-off-by: Jaejoong Kim <climbbb.kim@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the semantics of the enable option for snd-usb-audio
in order to allow users to disable a device specified by either or both
of the vendor id and the product id.
Signed-off-by: Tamaki Nishino <otamachan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 8428a8ebde ("ALSA: usb-audio: Fix potential zero-division
at parsing FU") is utterly bogus and breaks the case with csize=1
instead of fixing anything. Just take it back again.
Reported-by: Jörg Otte <jrg.otte@gmail.com>
Fixes: 8428a8ebde ("ALSA: usb-audio: Fix potential zero-division at parsing FU"
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Pull timer updates from Thomas Gleixner:
- The final conversion of timer wheel timers to timer_setup().
A few manual conversions and a large coccinelle assisted sweep and
the removal of the old initialization mechanisms and the related
code.
- Remove the now unused VSYSCALL update code
- Fix permissions of /proc/timer_list. I still need to get rid of that
file completely
- Rename a misnomed clocksource function and remove a stale declaration
* 'timers-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (27 commits)
m68k/macboing: Fix missed timer callback assignment
treewide: Remove TIMER_FUNC_TYPE and TIMER_DATA_TYPE casts
timer: Remove redundant __setup_timer*() macros
timer: Pass function down to initialization routines
timer: Remove unused data arguments from macros
timer: Switch callback prototype to take struct timer_list * argument
timer: Pass timer_list pointer to callbacks unconditionally
Coccinelle: Remove setup_timer.cocci
timer: Remove setup_*timer() interface
timer: Remove init_timer() interface
treewide: setup_timer() -> timer_setup() (2 field)
treewide: setup_timer() -> timer_setup()
treewide: init_timer() -> setup_timer()
treewide: Switch DEFINE_TIMER callbacks to struct timer_list *
s390: cmm: Convert timers to use timer_setup()
lightnvm: Convert timers to use timer_setup()
drivers/net: cris: Convert timers to use timer_setup()
drm/vc4: Convert timers to use timer_setup()
block/laptop_mode: Convert timers to use timer_setup()
net/atm/mpc: Avoid open-coded assignment of timer callback function
...
All commits found here are small fixes for regression or stable:
- PCM timestamp behavior fix that could be seen as a regression
- Remove spurious WARN_ON() from ALSA timer 32bit compat ioctl
- HD-audio HDMI/DP channel mapping fix for 32bit archs
- Fix the previous fix for HD-audio initialization code
- More hardening USB-audio against malicious USB descriptors
- HD-audio quirks/fixes (Realtek codec, AMD controller)
- Missing help text for the recent Intel SST kconfig change
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Merge tag 'sound-fix-4.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"All commits found here are small fixes for regression or stable:
- PCM timestamp behavior fix that could be seen as a regression
- Remove spurious WARN_ON() from ALSA timer 32bit compat ioctl
- HD-audio HDMI/DP channel mapping fix for 32bit archs
- Fix the previous fix for HD-audio initialization code
- More hardening USB-audio against malicious USB descriptors
- HD-audio quirks/fixes (Realtek codec, AMD controller)
- Missing help text for the recent Intel SST kconfig change"
* tag 'sound-fix-4.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda: Add Raven PCI ID
ALSA: hda/realtek - Fix ALC700 family no sound issue
ALSA: hda - Fix yet remaining issue with vmaster 0dB initialization
ALSA: usb-audio: Add sanity checks in v2 clock parsers
ALSA: usb-audio: Fix potential zero-division at parsing FU
ALSA: usb-audio: Fix potential out-of-bound access at parsing SU
ALSA: usb-audio: Add sanity checks to FE parser
ALSA: timer: Remove kernel warning at compat ioctl error paths
ALSA: pcm: update tstamp only if audio_tstamp changed
ALSA: hda/realtek: Add headset mic support for Intel NUC Skull Canyon
ALSA: hda: Fix too short HDMI/DP chmap reporting
ALSA: usb-audio: uac1: Invalidate ctl on interrupt
ALSA: hda/realtek - Fix ALC275 no sound issue
ASoC: Intel: Add help text for SND_SOC_INTEL_SST_TOPLEVEL
With all callbacks converted, and the timer callback prototype
switched over, the TIMER_FUNC_TYPE cast is no longer needed,
so remove it. Conversion was done with the following scripts:
perl -pi -e 's|\(TIMER_FUNC_TYPE\)||g' \
$(git grep TIMER_FUNC_TYPE | cut -d: -f1 | sort -u)
perl -pi -e 's|\(TIMER_DATA_TYPE\)||g' \
$(git grep TIMER_DATA_TYPE | cut -d: -f1 | sort -u)
The now unused macros are also dropped from include/linux/timer.h.
Signed-off-by: Kees Cook <keescook@chromium.org>
The helper functions to parse and look for the clock source, selector
and multiplier unit may return the descriptor with a too short length
than required, while there is no sanity check in the caller side.
Add some sanity checks in the parsers, at least, to guarantee the
given descriptor size, for avoiding the potential crashes.
Fixes: 79f920fbff ("ALSA: usb-audio: parse clock topology of UAC2 devices")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
parse_audio_feature_unit() contains a code dividing potentially with
zero when a malformed FU descriptor is passed. Although there is
already a sanity check, it checks only the value zero, hence it can
still lead to a zero-division when a value 1 is passed there.
Fix it by correcting the sanity check (and the error message
thereof).
Fixes: 23caaf19b1 ("ALSA: usb-mixer: Add support for Audio Class v2.0")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver may trigger an out-of-bound access at parsing a
malformed selector unit, as it checks the header length only after
evaluating bNrInPins field, which can be already above the given
length. Fix it by adding the length check beforehand.
Fixes: 99fc86450c ("ALSA: usb-mixer: parse descriptors with structs")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the usb-audio descriptor contains the malformed feature unit
description with a too short length, the driver may access
out-of-bounds. Add a sanity check of the header size at the beginning
of parse_audio_feature_unit().
Fixes: 23caaf19b1 ("ALSA: usb-mixer: Add support for Audio Class v2.0")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an interrupt occurs, the value of at least one of the belonging
controls should have changed. To make sure they get re-read from device
on the next read, invalidate the cache. This was correctly implemented
for uac2 already, but missing for uac1.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are no big surprising changes in this cycle, yet not too
boring, either. The biggest change from diffstat POV is the removal
of the legacy OSS driver codes that have been already disabled for a
long time. This will bring a few trivial merge conflicts.
As new features in ASoC side, there are two things: a new AC97 bus
implementation and AMD Stony platform support. Both include the
relevant changes shared with other subsystems, e.g. AC97 MFD changes
and DRM AMD changes.
Some other highlighted topics are:
- A bunch of USB-audio drivers got the hardening against the malicious
device accesses with a new helper code for endpoint sanity check.
- Lots of cleanups for ASoC Intel platform code, including support for
their open source audio firmware.
- Continued ASoC core componentization works.
- Support for scaling MCLK with sample rate in ASoC simple-card.
- Stabler PCM hot-unplug capability, especially for ASoC usages.
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Merge tag 'sound-4.15-rc1' of ssh://gitolite.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There are no big surprising changes in this cycle, yet not too boring,
either. The biggest change from diffstat POV is the removal of the
legacy OSS driver codes that have been already disabled for a long
time. This will bring a few trivial merge conflicts.
As new features in ASoC side, there are two things: a new AC97 bus
implementation and AMD Stony platform support. Both include the
relevant changes shared with other subsystems, e.g. AC97 MFD changes
and DRM AMD changes.
Some other highlighted topics are:
- A bunch of USB-audio drivers got the hardening against the
malicious device accesses with a new helper code for endpoint
sanity check
- Lots of cleanups for ASoC Intel platform code, including support
for their open source audio firmware
- Continued ASoC core componentization works
- Support for scaling MCLK with sample rate in ASoC simple-card
- Stabler PCM hot-unplug capability, especially for ASoC usages"
* tag 'sound-4.15-rc1' of ssh://gitolite.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (302 commits)
Documentation: sound: hd-audio: notes.rst
ASoC: bcm2835: Support left/right justified and DSP modes
ASoC: bcm2835: Enforce full symmetry
ASoC: bcm2835: Support additional samplerates up to 384kHz
ASoC: bcm2835: Add support for TDM modes
ASoC: add mclk-fs support to audio graph card
ASoC: add mclk-fs to audio graph card binding
ASoC: rt5514: work around link error
ASoC: rt5514: mark PM functions as __maybe_unused
ASoC: rt5663: Check the JD status in the button pushing
ASoC: amd: Modified DMA transfer Mechanism for Playback
ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2
ASoC: sun4i-codec: fixed 32bit audio capture support for H3/H2+
ASoC: da7213: add support for DSP modes
ASoC: sun8i-codec: Add a comment on the LRCK inversion
ASoC: sun8i-codec: Set the BCLK divider
ASoC: rt5663: Delay and retry reading rt5663 ID register
ASoC: amd: use do_div rather than 64 bit division to fix 32 bit builds
ASoC: cs42l56: Fix reset GPIO name in example DT binding
ASoC: rt5514-spi: check irq status to schedule data copy in resume function
...
Pull core locking updates from Ingo Molnar:
"The main changes in this cycle are:
- Another attempt at enabling cross-release lockdep dependency
tracking (automatically part of CONFIG_PROVE_LOCKING=y), this time
with better performance and fewer false positives. (Byungchul Park)
- Introduce lockdep_assert_irqs_enabled()/disabled() and convert
open-coded equivalents to lockdep variants. (Frederic Weisbecker)
- Add down_read_killable() and use it in the VFS's iterate_dir()
method. (Kirill Tkhai)
- Convert remaining uses of ACCESS_ONCE() to
READ_ONCE()/WRITE_ONCE(). Most of the conversion was Coccinelle
driven. (Mark Rutland, Paul E. McKenney)
- Get rid of lockless_dereference(), by strengthening Alpha atomics,
strengthening READ_ONCE() with smp_read_barrier_depends() and thus
being able to convert users of lockless_dereference() to
READ_ONCE(). (Will Deacon)
- Various micro-optimizations:
- better PV qspinlocks (Waiman Long),
- better x86 barriers (Michael S. Tsirkin)
- better x86 refcounts (Kees Cook)
- ... plus other fixes and enhancements. (Borislav Petkov, Juergen
Gross, Miguel Bernal Marin)"
* 'locking-core-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (70 commits)
locking/x86: Use LOCK ADD for smp_mb() instead of MFENCE
rcu: Use lockdep to assert IRQs are disabled/enabled
netpoll: Use lockdep to assert IRQs are disabled/enabled
timers/posix-cpu-timers: Use lockdep to assert IRQs are disabled/enabled
sched/clock, sched/cputime: Use lockdep to assert IRQs are disabled/enabled
irq_work: Use lockdep to assert IRQs are disabled/enabled
irq/timings: Use lockdep to assert IRQs are disabled/enabled
perf/core: Use lockdep to assert IRQs are disabled/enabled
x86: Use lockdep to assert IRQs are disabled/enabled
smp/core: Use lockdep to assert IRQs are disabled/enabled
timers/hrtimer: Use lockdep to assert IRQs are disabled/enabled
timers/nohz: Use lockdep to assert IRQs are disabled/enabled
workqueue: Use lockdep to assert IRQs are disabled/enabled
irq/softirqs: Use lockdep to assert IRQs are disabled/enabled
locking/lockdep: Add IRQs disabled/enabled assertion APIs: lockdep_assert_irqs_enabled()/disabled()
locking/pvqspinlock: Implement hybrid PV queued/unfair locks
locking/rwlocks: Fix comments
x86/paravirt: Set up the virt_spin_lock_key after static keys get initialized
block, locking/lockdep: Assign a lock_class per gendisk used for wait_for_completion()
workqueue: Remove now redundant lock acquisitions wrt. workqueue flushes
...
The biggest thing this release has been the conversion of the AC98 bus
to the driver model, that's been a long time coming so thanks to Robert
Jarzmik for his dedication there. Due to there being some AC97 MFD
there's a few fairly large changes in input and the MFD layer, mainly to
the wm97xx driver.
There's also some drivers/drm changes to support the new AMD Stoney
platform, these are shared with the DRM subsystem and should be being
merged via both.
Within the subsystem the overwhelming bulk of the changes is in the
Intel drivers which continue to need lots of cleanups and fixes, this
release they've also gained support for their open source firmware.
There's also some large changs in the core as Morimoto-san continues to
mirror operations into the component level in preparation for conversion
of drivers to that.
- The AC97 bus has finally caught up with the driver model thanks to
some dedicated and persistent work from Robert Jarzmik.
- Continued work from Morimoto-san on moving us towards being able to
use components for everything.
- Lots of cleanups for the Intel platform code, including support for
their open source audio firmware.
- Support for scaling MCLK with sample rate in simple-card.
- Support for AMD Stoney platform.
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Merge tag 'asoc-v4.15' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.15
The biggest thing this release has been the conversion of the AC98 bus
to the driver model, that's been a long time coming so thanks to Robert
Jarzmik for his dedication there. Due to there being some AC97 MFD
there's a few fairly large changes in input and the MFD layer, mainly to
the wm97xx driver.
There's also some drivers/drm changes to support the new AMD Stoney
platform, these are shared with the DRM subsystem and should be being
merged via both.
Within the subsystem the overwhelming bulk of the changes is in the
Intel drivers which continue to need lots of cleanups and fixes, this
release they've also gained support for their open source firmware.
There's also some large changs in the core as Morimoto-san continues to
mirror operations into the component level in preparation for conversion
of drivers to that.
- The AC97 bus has finally caught up with the driver model thanks to
some dedicated and persistent work from Robert Jarzmik.
- Continued work from Morimoto-san on moving us towards being able to
use components for everything.
- Lots of cleanups for the Intel platform code, including support for
their open source audio firmware.
- Support for scaling MCLK with sample rate in simple-card.
- Support for AMD Stoney platform.
The amount of the changes isn't as quite small as wished, nevertheless
they are straight fixes that deserve merging to 4.14 final.
Most of fixes are about ALSA core bugs spotted by fuzzer: a follow-up
fix for the previous nested rwsem patch, a fix to avoid the resource
hogs due to too many concurrent ALSA timer invocations, and a fix for
a crash with SYSEX MIDI transfer over OSS sequencer emulation that is
used by none but fuzzer.
The rest are usual HD-audio and USB-audio device-specific quirks,
which are safe to apply.
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Merge tag 'sound-4.14' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The amount of the changes isn't as quite small as wished, nevertheless
they are straight fixes that deserve merging to 4.14 final.
Most of fixes are about ALSA core bugs spotted by fuzzer: a follow-up
fix for the previous nested rwsem patch, a fix to avoid the resource
hogs due to too many concurrent ALSA timer invocations, and a fix for
a crash with SYSEX MIDI transfer over OSS sequencer emulation that is
used by none but fuzzer.
The rest are usual HD-audio and USB-audio device-specific quirks,
which are safe to apply"
* tag 'sound-4.14' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - fix headset mic problem for Dell machines with alc274
ALSA: seq: Fix OSS sysex delivery in OSS emulation
ALSA: seq: Avoid invalid lockdep class warning
ALSA: timer: Limit max instances per timer
ALSA: usb-audio: support new Amanero Combo384 firmware version
The us122l driver creates URBs per the fixed endpoints, and this may
end up with URBs with inconsistent pipes when a fuzzer or a malicious
program deals with the manipulated endpoints. It ends up with a
kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 0 != type 3
------------[ cut here ]------------
WARNING: CPU: 0 PID: 24 at drivers/usb/core/urb.c:471
usb_submit_urb+0x113e/0x1400
Call Trace:
usb_stream_start+0x48a/0x9f0 sound/usb/usx2y/usb_stream.c:690
us122l_start+0x116/0x290 sound/usb/usx2y/us122l.c:365
us122l_create_card sound/usb/usx2y/us122l.c:502
us122l_usb_probe sound/usb/usx2y/us122l.c:588
....
For avoiding the bad access, this patch adds a few sanity checks of
the validity of created URBs like previous similar fixes using the new
usb_urb_ep_type_check() helper function.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many source files in the tree are missing licensing information, which
makes it harder for compliance tools to determine the correct license.
By default all files without license information are under the default
license of the kernel, which is GPL version 2.
Update the files which contain no license information with the 'GPL-2.0'
SPDX license identifier. The SPDX identifier is a legally binding
shorthand, which can be used instead of the full boiler plate text.
This patch is based on work done by Thomas Gleixner and Kate Stewart and
Philippe Ombredanne.
How this work was done:
Patches were generated and checked against linux-4.14-rc6 for a subset of
the use cases:
- file had no licensing information it it.
- file was a */uapi/* one with no licensing information in it,
- file was a */uapi/* one with existing licensing information,
Further patches will be generated in subsequent months to fix up cases
where non-standard license headers were used, and references to license
had to be inferred by heuristics based on keywords.
The analysis to determine which SPDX License Identifier to be applied to
a file was done in a spreadsheet of side by side results from of the
output of two independent scanners (ScanCode & Windriver) producing SPDX
tag:value files created by Philippe Ombredanne. Philippe prepared the
base worksheet, and did an initial spot review of a few 1000 files.
The 4.13 kernel was the starting point of the analysis with 60,537 files
assessed. Kate Stewart did a file by file comparison of the scanner
results in the spreadsheet to determine which SPDX license identifier(s)
to be applied to the file. She confirmed any determination that was not
immediately clear with lawyers working with the Linux Foundation.
Criteria used to select files for SPDX license identifier tagging was:
- Files considered eligible had to be source code files.
- Make and config files were included as candidates if they contained >5
lines of source
- File already had some variant of a license header in it (even if <5
lines).
All documentation files were explicitly excluded.
The following heuristics were used to determine which SPDX license
identifiers to apply.
- when both scanners couldn't find any license traces, file was
considered to have no license information in it, and the top level
COPYING file license applied.
For non */uapi/* files that summary was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 11139
and resulted in the first patch in this series.
If that file was a */uapi/* path one, it was "GPL-2.0 WITH
Linux-syscall-note" otherwise it was "GPL-2.0". Results of that was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 WITH Linux-syscall-note 930
and resulted in the second patch in this series.
- if a file had some form of licensing information in it, and was one
of the */uapi/* ones, it was denoted with the Linux-syscall-note if
any GPL family license was found in the file or had no licensing in
it (per prior point). Results summary:
SPDX license identifier # files
---------------------------------------------------|------
GPL-2.0 WITH Linux-syscall-note 270
GPL-2.0+ WITH Linux-syscall-note 169
((GPL-2.0 WITH Linux-syscall-note) OR BSD-2-Clause) 21
((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) 17
LGPL-2.1+ WITH Linux-syscall-note 15
GPL-1.0+ WITH Linux-syscall-note 14
((GPL-2.0+ WITH Linux-syscall-note) OR BSD-3-Clause) 5
LGPL-2.0+ WITH Linux-syscall-note 4
LGPL-2.1 WITH Linux-syscall-note 3
((GPL-2.0 WITH Linux-syscall-note) OR MIT) 3
((GPL-2.0 WITH Linux-syscall-note) AND MIT) 1
and that resulted in the third patch in this series.
- when the two scanners agreed on the detected license(s), that became
the concluded license(s).
- when there was disagreement between the two scanners (one detected a
license but the other didn't, or they both detected different
licenses) a manual inspection of the file occurred.
- In most cases a manual inspection of the information in the file
resulted in a clear resolution of the license that should apply (and
which scanner probably needed to revisit its heuristics).
- When it was not immediately clear, the license identifier was
confirmed with lawyers working with the Linux Foundation.
- If there was any question as to the appropriate license identifier,
the file was flagged for further research and to be revisited later
in time.
In total, over 70 hours of logged manual review was done on the
spreadsheet to determine the SPDX license identifiers to apply to the
source files by Kate, Philippe, Thomas and, in some cases, confirmation
by lawyers working with the Linux Foundation.
Kate also obtained a third independent scan of the 4.13 code base from
FOSSology, and compared selected files where the other two scanners
disagreed against that SPDX file, to see if there was new insights. The
Windriver scanner is based on an older version of FOSSology in part, so
they are related.
Thomas did random spot checks in about 500 files from the spreadsheets
for the uapi headers and agreed with SPDX license identifier in the
files he inspected. For the non-uapi files Thomas did random spot checks
in about 15000 files.
In initial set of patches against 4.14-rc6, 3 files were found to have
copy/paste license identifier errors, and have been fixed to reflect the
correct identifier.
Additionally Philippe spent 10 hours this week doing a detailed manual
inspection and review of the 12,461 patched files from the initial patch
version early this week with:
- a full scancode scan run, collecting the matched texts, detected
license ids and scores
- reviewing anything where there was a license detected (about 500+
files) to ensure that the applied SPDX license was correct
- reviewing anything where there was no detection but the patch license
was not GPL-2.0 WITH Linux-syscall-note to ensure that the applied
SPDX license was correct
This produced a worksheet with 20 files needing minor correction. This
worksheet was then exported into 3 different .csv files for the
different types of files to be modified.
These .csv files were then reviewed by Greg. Thomas wrote a script to
parse the csv files and add the proper SPDX tag to the file, in the
format that the file expected. This script was further refined by Greg
based on the output to detect more types of files automatically and to
distinguish between header and source .c files (which need different
comment types.) Finally Greg ran the script using the .csv files to
generate the patches.
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Reviewed-by: Philippe Ombredanne <pombredanne@nexb.com>
Reviewed-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Support DSD_U32_BE sample format on new Amanero Combo384 firmware
version on older VID/PID.
Fixes: 3eff682d76 ("ALSA: usb-audio: Support both DSD LE/BE Amanero firmware versions")
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Please do not apply this to mainline directly, instead please re-run the
coccinelle script shown below and apply its output.
For several reasons, it is desirable to use {READ,WRITE}_ONCE() in
preference to ACCESS_ONCE(), and new code is expected to use one of the
former. So far, there's been no reason to change most existing uses of
ACCESS_ONCE(), as these aren't harmful, and changing them results in
churn.
However, for some features, the read/write distinction is critical to
correct operation. To distinguish these cases, separate read/write
accessors must be used. This patch migrates (most) remaining
ACCESS_ONCE() instances to {READ,WRITE}_ONCE(), using the following
coccinelle script:
----
// Convert trivial ACCESS_ONCE() uses to equivalent READ_ONCE() and
// WRITE_ONCE()
// $ make coccicheck COCCI=/home/mark/once.cocci SPFLAGS="--include-headers" MODE=patch
virtual patch
@ depends on patch @
expression E1, E2;
@@
- ACCESS_ONCE(E1) = E2
+ WRITE_ONCE(E1, E2)
@ depends on patch @
expression E;
@@
- ACCESS_ONCE(E)
+ READ_ONCE(E)
----
Signed-off-by: Mark Rutland <mark.rutland@arm.com>
Signed-off-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Cc: Linus Torvalds <torvalds@linux-foundation.org>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: davem@davemloft.net
Cc: linux-arch@vger.kernel.org
Cc: mpe@ellerman.id.au
Cc: shuah@kernel.org
Cc: snitzer@redhat.com
Cc: thor.thayer@linux.intel.com
Cc: tj@kernel.org
Cc: viro@zeniv.linux.org.uk
Cc: will.deacon@arm.com
Link: http://lkml.kernel.org/r/1508792849-3115-19-git-send-email-paulmck@linux.vnet.ibm.com
Signed-off-by: Ingo Molnar <mingo@kernel.org>
In preparation for unconditionally passing the struct timer_list pointer to
all timer callbacks, switch to using the new timer_setup() and from_timer()
to pass the timer pointer explicitly.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Back-merge for applying the timer API conversion patch for line6
driver that conflicts with the recent fix in upstream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pointer card is being assigned a value but it is never used.
Remove this redundant variable. Cleans up clang warning:
Value stored to 'card' is never read
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add native DSD support quirk for Pro-Ject Pre Box S2 Digital USB id
2772:0230.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
caiaq driver doesn't kill the URB properly at its error path during
the probe, which may lead to a use-after-free error later. This patch
addresses it.
Reported-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Johan Hovold <johan@kernel.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pulling the EP validity checks in USB audio drivers.
It also adds a new helper in USB core, which was acked by Greg.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few other places calling usb_submit_urb() with the URB
composed from the fixed endpoint without validation. For avoiding the
spurious kernel warnings, add the sanity checks to appropriate
places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few other places in caiaq driver have the URB handling with the
fixed endpoints without checking the validity, too. Add the sanity
check with the new helper function at each appropriate place for
avoiding the spurious kernel warnings due to invalid EPs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hiface usb-audio driver sets up URBs containing the fixed endpoints
without validation. This may end up with an oops-like kernel warning
when submitted.
For avoiding it, this patch adds the calls of the new sanity-check
helper for URBs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usx2y driver sets up URBs containing the fixed endpoints without
validation. This may end up with an oops-like kernel warning when
submitted.
For avoiding it, this patch adds the calls of the new sanity-check
helper for URBs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver may set up a URB containing the fixed EP without
validating its presence for some non-class-compliant devices. This
may end up with an oops-like kernel warning when submitted.
For avoiding it, this patch adds the call of the new sanity-check
helper for URBs. The checks are needed only for MIDI I/O as the other
places have already some other checks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As syzkaller spotted, currently line6 drivers submit a URB with the
fixed EP without checking whether it's actually available, which may
result in a kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 3 != type 1
------------[ cut here ]------------
WARNING: CPU: 0 PID: 24 at drivers/usb/core/urb.c:449
usb_submit_urb+0xf8a/0x11d0
Modules linked in:
CPU: 0 PID: 24 Comm: kworker/0:1 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
line6_start_listen+0x55f/0x9e0 sound/usb/line6/driver.c:82
line6_init_cap_control sound/usb/line6/driver.c:690
line6_probe+0x7c9/0x1310 sound/usb/line6/driver.c:764
podhd_probe+0x64/0x70 sound/usb/line6/podhd.c:474
usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361
....
This patch adds a sanity check of validity of EPs at the device
initialization phase for avoiding the call with an invalid EP.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As syzkaller spotted, currently caiaq driver submits a URB with the
fixed EP without checking whether it's actually available, which may
result in a kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 3 != type 1
------------[ cut here ]------------
WARNING: CPU: 1 PID: 1150 at drivers/usb/core/urb.c:449
usb_submit_urb+0xf8a/0x11d0
Modules linked in:
CPU: 1 PID: 1150 Comm: kworker/1:1 Not tainted
4.14.0-rc2-42660-g24b7bd59eec0 #277
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
init_card sound/usb/caiaq/device.c:467
snd_probe+0x81c/0x1150 sound/usb/caiaq/device.c:525
usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361
....
This patch adds a sanity check of validity of EPs at the device
initialization phase for avoiding the call with an invalid EP.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As syzkaller spotted, currently bcd2000 driver submits a URB with the
fixed EP without checking whether it's actually available, which may
result in a kernel warning like:
usb 1-1: BOGUS urb xfer, pipe 1 != type 3
------------[ cut here ]------------
WARNING: CPU: 0 PID: 1846 at drivers/usb/core/urb.c:449
usb_submit_urb+0xf8a/0x11d0
Modules linked in:
CPU: 0 PID: 1846 Comm: kworker/0:2 Not tainted
4.14.0-rc2-42613-g1488251d1a98 #238
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
bcd2000_init_device sound/usb/bcd2000/bcd2000.c:289
bcd2000_init_midi sound/usb/bcd2000/bcd2000.c:345
bcd2000_probe+0xe64/0x19e0 sound/usb/bcd2000/bcd2000.c:406
usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361
....
This patch adds a sanity check of validity of EPs at the device
initialization phase for avoiding the call with an invalid EP.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver may leave a stray URB for the mixer interrupt when it
exits by some error during probe. This leads to a use-after-free
error as spotted by syzkaller like:
==================================================================
BUG: KASAN: use-after-free in snd_usb_mixer_interrupt+0x604/0x6f0
Call Trace:
<IRQ>
__dump_stack lib/dump_stack.c:16
dump_stack+0x292/0x395 lib/dump_stack.c:52
print_address_description+0x78/0x280 mm/kasan/report.c:252
kasan_report_error mm/kasan/report.c:351
kasan_report+0x23d/0x350 mm/kasan/report.c:409
__asan_report_load8_noabort+0x19/0x20 mm/kasan/report.c:430
snd_usb_mixer_interrupt+0x604/0x6f0 sound/usb/mixer.c:2490
__usb_hcd_giveback_urb+0x2e0/0x650 drivers/usb/core/hcd.c:1779
....
Allocated by task 1484:
save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59
save_stack+0x43/0xd0 mm/kasan/kasan.c:447
set_track mm/kasan/kasan.c:459
kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551
kmem_cache_alloc_trace+0x11e/0x2d0 mm/slub.c:2772
kmalloc ./include/linux/slab.h:493
kzalloc ./include/linux/slab.h:666
snd_usb_create_mixer+0x145/0x1010 sound/usb/mixer.c:2540
create_standard_mixer_quirk+0x58/0x80 sound/usb/quirks.c:516
snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560
create_composite_quirk+0x1c4/0x3e0 sound/usb/quirks.c:59
snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560
usb_audio_probe+0x1040/0x2c10 sound/usb/card.c:618
....
Freed by task 1484:
save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59
save_stack+0x43/0xd0 mm/kasan/kasan.c:447
set_track mm/kasan/kasan.c:459
kasan_slab_free+0x72/0xc0 mm/kasan/kasan.c:524
slab_free_hook mm/slub.c:1390
slab_free_freelist_hook mm/slub.c:1412
slab_free mm/slub.c:2988
kfree+0xf6/0x2f0 mm/slub.c:3919
snd_usb_mixer_free+0x11a/0x160 sound/usb/mixer.c:2244
snd_usb_mixer_dev_free+0x36/0x50 sound/usb/mixer.c:2250
__snd_device_free+0x1ff/0x380 sound/core/device.c:91
snd_device_free_all+0x8f/0xe0 sound/core/device.c:244
snd_card_do_free sound/core/init.c:461
release_card_device+0x47/0x170 sound/core/init.c:181
device_release+0x13f/0x210 drivers/base/core.c:814
....
Actually such a URB is killed properly at disconnection when the
device gets probed successfully, and what we need is to apply it for
the error-path, too.
In this patch, we apply snd_usb_mixer_disconnect() at releasing.
Also introduce a new flag, disconnected, to struct usb_mixer_interface
for not performing the disconnection procedure twice.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While line6_probe() may kick off URB for a control MIDI endpoint, the
function doesn't clean up it properly at its error path. This results
in a leftover URB action that is eventually triggered later and causes
an Oops like:
general protection fault: 0000 [#1] PREEMPT SMP KASAN
CPU: 1 PID: 0 Comm: swapper/1 Not tainted
RIP: 0010:usb_fill_bulk_urb ./include/linux/usb.h:1619
RIP: 0010:line6_start_listen+0x3fe/0x9e0 sound/usb/line6/driver.c:76
Call Trace:
<IRQ>
line6_data_received+0x1f7/0x470 sound/usb/line6/driver.c:326
__usb_hcd_giveback_urb+0x2e0/0x650 drivers/usb/core/hcd.c:1779
usb_hcd_giveback_urb+0x337/0x420 drivers/usb/core/hcd.c:1845
dummy_timer+0xba9/0x39f0 drivers/usb/gadget/udc/dummy_hcd.c:1965
call_timer_fn+0x2a2/0x940 kernel/time/timer.c:1281
....
Since the whole clean-up procedure is done in line6_disconnect()
callback, we can simply call it in the error path instead of
open-coding the whole again. It'll fix such an issue automagically.
The bug was spotted by syzkaller.
Fixes: eedd0e95d3 ("ALSA: line6: Don't forget to call driver's destructor at error path")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When podhd_init() failed with the acquiring a ctrl i/f, the line6
helper still calls the disconnect callback that eventually calls again
usb_driver_release_interface() with the NULL intf.
Put the proper NULL check before calling it for avoiding an Oops.
Fixes: fc90172ba2 ("ALSA: line6: Claim pod x3 usb data interface")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error path in podhd_init() tries to clear the pending timer, while
the timer object is initialized at the end of init sequence, thus it
may hit the uninitialized object, as spotted by syzkaller:
INFO: trying to register non-static key.
the code is fine but needs lockdep annotation.
turning off the locking correctness validator.
CPU: 1 PID: 1845 Comm: kworker/1:2 Not tainted
4.14.0-rc2-42613-g1488251d1a98 #238
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011
Workqueue: usb_hub_wq hub_event
Call Trace:
__dump_stack lib/dump_stack.c:16
dump_stack+0x292/0x395 lib/dump_stack.c:52
register_lock_class+0x6c4/0x1a00 kernel/locking/lockdep.c:769
__lock_acquire+0x27e/0x4550 kernel/locking/lockdep.c:3385
lock_acquire+0x259/0x620 kernel/locking/lockdep.c:4002
del_timer_sync+0x12c/0x280 kernel/time/timer.c:1237
podhd_disconnect+0x8c/0x160 sound/usb/line6/podhd.c:299
line6_probe+0x844/0x1310 sound/usb/line6/driver.c:783
podhd_probe+0x64/0x70 sound/usb/line6/podhd.c:474
....
For addressing it, assure the initializations of timer and work by
moving them to the beginning of podhd_init().
Fixes: 790869dacc ("ALSA: line6: Add support for POD X3")
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, lots of Plantronics, MS and Phoenix Audio devices need the
quirk not to read sample rate back, and the list just grows.
In this patch, instead of adding each device, apply the quirk by
matching with these vendors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like other Plantronics devices, P610 does not support sample
rate reading. Apply sample rate quirk to it.
BugLink: https://bugs.launchpad.net/bugs/1719853
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usx2y driver allocates the stream read/write buffers in continuous
pages depending on the stream setup, and this may spew the kernel
warning messages with a stack trace like:
WARNING: CPU: 1 PID: 1846 at mm/page_alloc.c:3883
__alloc_pages_slowpath+0x1ef2/0x2d70
Modules linked in:
CPU: 1 PID: 1846 Comm: kworker/1:2 Not tainted
....
It may confuse user as if it were any serious error, although this is
no fatal error and the driver handles the error case gracefully.
Since the driver has already some sanity check of the given size (128
and 256 pages), it can't pass any crazy value. So it's merely page
fragmentation.
This patch adds __GFP_NOWARN to each caller for suppressing such
kernel warnings. The original issue was spotted by syzkaller.
Reported-by: Andrey Konovalov <andreyknvl@google.com>
Tested-by: Andrey Konovalov <andreyknvl@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make these const as they are only passed to a const argument of the
function snd_pcm_set_ops in the file referencing them. Also, add const
to the declaration in the headers.
Structures found using Coccinelle and changes done by hand.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like other Plantronics devices, C310 and C520-M do not support sample
rate reading. Add them to the sample rate quirk accordingly.
BugLink: https://bugs.launchpad.net/bugs/1708499
BugLink: https://bugs.launchpad.net/bugs/1709282
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Line6 POD HD500X multi effect processor for playback
and capture (in/out audio) through USB.
Signed-off-by: Hans P. Moller <hmoller@uc.cl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a jump target so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a jump target so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a jump target so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usx2y driver has a debug printk code without proper KERN_ prefix.
On recent kernels, KERN_CONT prefix is mandatory for continued output
lines. Put it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver has a debug printk code without proper KERN_
prefix. On recent kernels, KERN_CONT prefix is mandatory for
continued output lines. Put it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add jump targets so that a bit of exception handling can be better reused
at the end of this function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These headsets reports a lot of: cannot set freq 44100 to ep 0x81
and need a small delay between sample rate settings, just like
Zoom R16/24. Add both headsets to the Zoom R16/24 quirk for
a 1 ms delay between control msgs.
Signed-off-by: Joakim Tjernlund <joakim.tjernlund@infinera.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/pcm.h> work with
const snd_pcm_ops. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DSD support for new Amanero Combo384 firmware version with a new
PID. This firmware uses DSD_U32_BE.
Fixes: 3eff682d76 ("ALSA: usb-audio: Support both DSD LE/BE Amanero firmware versions")
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
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>
Make this const as it is only used in a copy operation.
Done using Coccinelle.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A Senheisser headset requires the typical sample-rate quirk for
avoiding spurious errors from inquiring the current sample rate like:
usb 1-1: 2:1: cannot get freq at ep 0x4
usb 1-1: 3:1: cannot get freq at ep 0x83
The USB ID 1395:740a has to be added to the entries in
snd_usb_get_sample_rate_quirk().
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1052580
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Omit extra messages for a memory allocation failure in this function.
This issue was detected by using the Coccinelle software.
[Fixed the unused variable warning by this change as well -- tiwai]
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_device_id are not supposed to change at runtime. All functions
working with usb_device_id provided by <linux/usb.h> work with
const usb_device_id. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove Initialization from POD HD500X because it's not needed.
Every time the device is connected dmesg gives the following output:
"receive length failed (error -11)".
To solve this problem, another flags is introduced
(LINE6_CAP_CONTROL_INFO) and it is only used for PODX3 in: sysfs
entries, call podhd_startup_finalize(pod) and disconnection.
With this patch the error disappear.
Signed-off-by: Hans P. Moller <hmoller@uc.cl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Line6 POD HD500X multi effect processor for
playback and capture (in/out audio) through USB.
Signed-off-by: Hans P. Moller <hmoller@uc.cl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another attempt to work around the VLA used in
mixer_us16x08.c. Basically the temporary array is used individually
for two cases, and we can declare locally in each block, instead of
hackish max() usage.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A mixer element created in a quirk for Tascam US-16x08 contains a
typo: it should be "EQ MidLow Q" instead of "EQ MidQLow Q".
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195875
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Cc: <stable@vger.kernel.org> # v4.11+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 89b593c30e ("ALSA: usb-audio: purge needless
variable length array"). The patch turned out to cause a severe
regression, triggering an Oops at snd_usb_ctl_msg(). It was overseen
that snd_usb_ctl_msg() writes back the response to the given buffer,
while the patch changed it to a read-only const buffer. (One should
always double-check when an extra pointer cast is present...)
As a simple fix, just revert the affected commit. It was merely a
cleanup. Although it brings VLA again, it's clearer as a fix. We'll
address the VLA later in another patch.
Fixes: 89b593c30e ("ALSA: usb-audio: purge needless variable length array")
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195875
Cc: <stable@vger.kernel.org> # v4.11+
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>
This USB driver currently depends on X86 for some undocumented reason.
Since it compiles just fine for arm, we can at least enable compile
testing.
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the device-id table and a private flag to determine the device type
(US122 or US144) rather than spreading product-id conditionals
throughout the driver.
This USB driver currently depends on X86 (why?), but we should still add
the missing endianness conversions when accessing the USB
device-descriptor fields.
Compile-tested only.
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add missing endianness conversion when using the USB device-descriptor
bcdDevice field when applying the Amanero Combo384 (endianness!) quirk.
Fixes: 3eff682d76 ("ALSA: usb-audio: Support both DSD LE/BE Amanero firmware versions")
Cc: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The /proc/bus/usb devices don't exist anymore, since when we
got rid of usbfs. Those devices are now seen at
/dev/bus/usb.
Signed-off-by: Mauro Carvalho Chehab <mchehab@s-opensource.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In kernel APIs of ALSA control interface, drivers can create a control
element set by a call of snd_ctl_new1() with a template. This template
is known to have const qualifier in general cases.
This commit adds the qualifier to template array, for safer program and
runtime. Application of this change moves the symbol from .data section
to .rodata section.
Cc: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new strcutures as const as they are only passed as
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 too.
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;
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new strcutures as const as they are only passed as
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 too.
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;
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently snd-usb-audio driver received a new option, quirk_alias, to
allow user to apply the existing quirk for a different device. This
works for many quirks as is, but some still need more tune-ups:
namely, some quirks check the USB vendor/device IDs in various places,
thus it doesn't work as long as the ID is different from the expected
one.
With this patch, the driver stores the aliased USB ID, so that these
rest quirks per device ID are applied. The transition to use the
cached USB ID was already done in the past, so what we needed now is
only to overwrite chip->usb_id.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
->fault(), ->page_mkwrite(), and ->pfn_mkwrite() calls do not need to
take a vma and vmf parameter when the vma already resides in vmf.
Remove the vma parameter to simplify things.
[arnd@arndb.de: fix ARM build]
Link: http://lkml.kernel.org/r/20170125223558.1451224-1-arnd@arndb.de
Link: http://lkml.kernel.org/r/148521301778.19116.10840599906674778980.stgit@djiang5-desk3.ch.intel.com
Signed-off-by: Dave Jiang <dave.jiang@intel.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Ross Zwisler <ross.zwisler@linux.intel.com>
Cc: Theodore Ts'o <tytso@mit.edu>
Cc: Darrick J. Wong <darrick.wong@oracle.com>
Cc: Matthew Wilcox <mawilcox@microsoft.com>
Cc: Dave Hansen <dave.hansen@intel.com>
Cc: Christoph Hellwig <hch@lst.de>
Cc: Jan Kara <jack@suse.com>
Cc: Dan Williams <dan.j.williams@intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
A few more cleanups and improvements that have been overlooked:
- Use ARRAY_SIZE() macro appropriately
- Code shuffling for minor optimization
- Omit superfluous variable initializations
- Get rid of superfluous NULL checks
- Add const to snd_us16x08_control_params definitions
No functional changes.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places leaking memory and doing double-free in
mixer_us16x08.c.
The driver allocates a usb_mixer_elem_info object at each
add_new_ctl() call. This has to be freed via kctl->private_free, but
currently this is done properly only for some controls.
Also, the driver allocates three external objects (comp_store,
eq_store, meter_store), and these are referred in elem->private_data
(it's not kctl->private_data). And these have to be released, but
there are none doing it. Moreover, these extra objects have to be
released only once. Thus the release should be done only by the first
kctl element that refers to it.
For fixing these, we call either snd_usb_mixer_elem_free() (only for
kctl->private_data) or elem_private_free() (for both
kctl->private_data and elem->private_data) via kctl->private_free
appropriately.
Last but not least, snd_us16x08_controls_create() may return in the
middle without releasing the allocated *_store objects due to an
error. For fixing this, we shuffle the allocation code so that it's
called just before its reference.
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Reported-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Variable length array is used in 'snd_us16x08_meter_get()', while there
is no need. It's better to purge it because variable length array has
overhead for stack handling.
This commit replaces the array with static length. Sparse generated below
warning.
sound/usb/mixer_us16x08.c:714:18: warning: Variable length array is used.
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new structures as const as they are only passed as
an argument to the function add_new_ctl. This agrument is of type const,
so snd_kcontrol_new structures having this property can be made const
too.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When accessed inner a file, functions should have static qualifier for
local-linkage.
This commit fixes the bug. Sparse generated below warning.
sound/usb/mixer_us16x08.c:1043:32: warning: symbol 'snd_us16x08_create_meter_store' was not declared. Should it be static?
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When accessed by one referrer inner a file, variables should have static
qualifier to declare local-linkage.
This commit fixes the bug. Sparse generated below warnings.
sound/usb/mixer_us16x08.c:156:13: warning: duplicate const
sound/usb/mixer_us16x08.c:156:18: warning: symbol 'route_names' was not declared. Should it be static?
Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
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>
While not all line6 devices currently support PCM, it causes no
harm to 'have it prepared'.
This also fixes toneport, which only has PCM - in which case
we previously skipped the USB transfer properties detection completely.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit f6a0dd107a.
The commit caused a regression on LINE6 Transport that has no control
caps. Although reverting the commit may result back in a spurious
error message for some device again, it's the simplest regression fix,
hence it's taken as is at first. The further code fix will follow
later.
Fixes: f6a0dd107a ("ALSA: line6: Only determine control port properties if needed")
Reported-by: Igor Zinovev <zinigor@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plantronics BT600 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1" and "cannot get freq at
ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion
logic allows us to save a few cpu cycles by returning early, skipping the
pending urb in case the stream was stopped; the stop logic handles the urb
and sets the completion callbacks to NULL.
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream") was
incomplete causing another more severe kernel panic, so it got reverted.
This fixes both the original problem and its fallout kernel race/crash.
The original fix is to move the endpoint member NULL clearing logic inside
wait_clear_urbs() so the irq triggering the urb completion doesn't call
retire_capture/playback_urb() after the NULL clearing and generate a panic.
However this creates a new race between snd_usb_endpoint_start()'s call
to wait_clear_urbs() and the irq urb completion handler which again calls
retire_capture/playback_urb() leading to a new NULL dereference.
We keep the EP deactivation code in snd_usb_endpoint_start() because
removing it will break the EP reference counting (see [1] [2] for info),
however we don't need the "can_sleep" mechanism anymore because a new
function was introduced (snd_usb_endpoint_sync_pending_stop()) which
synchronizes pending stops and gets called inside the pcm prepare callback.
It also makes sense to remove can_sleep because it was also removed from
deactivate_urbs() signature in [3] so we benefit from more simplification.
[1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start")
[2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream")
[3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code")
Fixes: f8114f8583 ("Revert "ALSA: usb-audio: Fix race at stopping the stream"")
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DSD support for both little endian (DSD_U32_LE) and big endian
(DSD_U32_BE) version of the Amanero firmware.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 16200948d8.
The commit was intended to cover the race condition, but it introduced
yet another regression for devices with the implicit feedback, leading
to a kernel panic due to NULL-dereference in an irq context.
As the race condition that was addressed by the commit is very rare
and the regression is much worse, let's revert the commit for rc1, and
fix the issue properly in a later patch.
Fixes: 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream")
Reported-by: Ioan-Adrian Ratiu <adi@adirat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Sampling rate changes after first set one are not reflected to the
hardware, while driver and ALSA think the rate has been changed.
Fix the problem by properly stopping the interface at the beginning of
prepare call, allowing new rate to be set to the hardware. This keeps
the hardware in sync with the driver.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[Problem]
In some USB DACs, a terrible pop noise comes to be heard
at the start of DSD playback (in the following situations).
- play first DSD track
- change from PCM track to DSD track
- change from DSD64 track to DSD128 track (and etc...)
- seek DSD track
- Fast-Forward/Rewind DSD track
[Cause]
At the start of playback, there is a little silence.
The silence bit pattern "0x69" is required on DSD mode,
but it is not like that.
[Solution]
This patch adds DSD silence pattern to the endpoint settings.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the following devices.
- TEAC NT-503
- TEAC UD-503
- TEAC UD-501
(1) Add quirks for native DSD support for TEAC devices.
(2) A specific vendor command is needed to switch between PCM/DOP and
DSD mode, same as Denon/Marantz devices.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Logitech QuickCam Communicate Deluxe/S7500 microphone fails with the
following warning.
[ 6.778995] usb 2-1.2.2.2: Warning! Unlikely big volume range (=3072),
cval->res is probably wrong.
[ 6.778996] usb 2-1.2.2.2: [5] FU [Mic Capture Volume] ch = 1, val =
4608/7680/1
Adding it to the list of devices in volume_control_quirks makes it work
properly, fixing related typo.
Signed-off-by: Con Kolivas <kernel@kolivas.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Axe-Fx II implicit feedback end point and the data sync endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
since commit 57e6dae108 ("ALSA: usb-audio: do not trust too-big
wMaxPacketSize values"), the expected packetsize is always limited
to nominal + 25%. It was discovered, that some devices (Android audio
accessory) have a much higher jitter in used packetsizes than 25%
which would result in BABBLE condition and dropping of packets.
A better solution is so assume the jitter to be the nominal packetsize:
-one nearly empty packet followed by a almost 150% sized one.
V2: changed to assume max frequency is +50 of nominal packetsize.
Signed-off-by: Andreas Pape <apape@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of userland applications call 'snd_pcm_hw_params()' and
'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()'
is called twice and the second 'snd_pcm_hw_prepare()' is called in
'SNDRV_PCM_STATE_PREPARED' state.
Some devices are not able to manage this and they will stop playback
if the sample rate will be configured several times over USB protocol.
V2: updated Changelog
Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com>
Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a kernel crash report showing like:
Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = a1d7c000
[00000008] *pgd=31c93831, *pte=00000000, *ppte=00000000
Internal error: Oops: 17 [#1] PREEMPT SMP ARM
CPU: 0 PID: 250 Comm: dbus-daemon Not tainted 3.14.51-03479-gf50bdf4 #1
task: a3ae61c0 ti: a08c8000 task.ti: a08c8000
PC is at retire_capture_urb+0x10/0x1f4 [snd_usb_audio]
LR is at snd_complete_urb+0x140/0x1f0 [snd_usb_audio]
pc : [<7f0eb22c>] lr : [<7f0e57fc>] psr: 200e0193
sp : a08c9c98 ip : a08c9ce8 fp : a08c9ce4
r10: 0000000a r9 : 00000102 r8 : 94cb3000
r7 : 94cb3000 r6 : 94d0f000 r5 : 94d0e8e8 r4 : 94d0e000
r3 : 7f0eb21c r2 : 00000000 r1 : 94cb3000 r0 : 00000000
Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user
Control: 10c5387d Table: 31d7c04a DAC: 00000015
Process dbus-daemon (pid: 250, stack limit = 0xa08c8238)
Stack: (0xa08c9c98 to 0xa08ca000)
...
Backtrace:
[<7f0eb21c>] (retire_capture_urb [snd_usb_audio]) from [<7f0e57fc>] (snd_complete_urb+0x140/0x1f0 [snd_usb_audio])
[<7f0e56bc>] (snd_complete_urb [snd_usb_audio]) from [<80371118>] (__usb_hcd_giveback_urb+0x78/0xf4)
[<803710a0>] (__usb_hcd_giveback_urb) from [<80371514>] (usb_giveback_urb_bh+0x8c/0xc0)
[<80371488>] (usb_giveback_urb_bh) from [<80028e3c>] (tasklet_hi_action+0xc4/0x148)
[<80028d78>] (tasklet_hi_action) from [<80028358>] (__do_softirq+0x190/0x380)
[<800281c8>] (__do_softirq) from [<80028858>] (irq_exit+0x8c/0xfc)
[<800287cc>] (irq_exit) from [<8000ea88>] (handle_IRQ+0x8c/0xc8)
[<8000e9fc>] (handle_IRQ) from [<800085e8>] (gic_handle_irq+0xbc/0xf8)
[<8000852c>] (gic_handle_irq) from [<80509044>] (__irq_svc+0x44/0x78)
[<80508820>] (_raw_spin_unlock_irq) from [<8004b880>] (finish_task_switch+0x5c/0x100)
[<8004b824>] (finish_task_switch) from [<805052f0>] (__schedule+0x48c/0x6d8)
[<80504e64>] (__schedule) from [<805055d4>] (schedule+0x98/0x9c)
[<8050553c>] (schedule) from [<800116c8>] (do_work_pending+0x30/0xd0)
[<80011698>] (do_work_pending) from [<8000e160>] (work_pending+0xc/0x20)
Code: e1a0c00d e92ddff0 e24cb004 e24dd024 (e5902008)
Kernel panic - not syncing: Fatal exception in interrupt
There is a race between retire_capture_urb() and stop_endpoints().
The latter is called at stopping the stream and it sets some endpoint
fields to NULL. But its call is asynchronous, thus the pending
complete callback might get called after these NULL clears, and it
leads the NULL dereference like the above.
The fix is to move the NULL clearance after the synchronization,
i.e. wait_clear_urbs(). This is called at prepare and hw_free
callbacks, so it's assured to be called before the restart of the
stream or the release of the stream.
Also, while we're at it, put the EP_FLAG_RUNNING flag check at the
beginning of snd_complete_urb() to skip the pending complete after the
stream is stopped.
Fixes: b2eb950de2 ("ALSA: usb-audio: stop both data and sync...")
Reported-by: Jiada Wang <jiada_wang@mentor.com>
Reported-by: Mark Craske <Mark_Craske@mentor.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function returns -EINVAL even if it builds the stream properly.
The bogus error code sneaked in during the code refactoring, but it
wasn't noticed until now since the returned error code itself is
ignored in anyway. Kill it here, but there is no behavior change by
this patch, obviously.
Fixes: e5779998bf ('ALSA: usb-audio: refactor code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Userspace apps have to claim USB interfaces before using endpoints in
them (drivers/usb/core/devio.c:checkintf()). It's a lock mechanism so
that two "drivers" don't steal data from each other. Kernel drivers don't
have to claim interfaces to work - but they should, to lock out userspace.
While there, fix line6_properties struct to match checkpatch.pl.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usb-audio driver implements the deferred device disconnection for
the device in use. In this mode, the disconnection callback returns
immediately while the actual ALSA card object removal happens later
when all files get closed. As Shuah reported, this code flow,
however, leads to a use-after-free, detected by KASAN:
BUG: KASAN: use-after-free in snd_usb_audio_free+0x134/0x160 [snd_usb_audio] at addr ffff8801c863ce10
Write of size 8 by task pulseaudio/2244
Call Trace:
[<ffffffff81b31473>] dump_stack+0x67/0x94
[<ffffffff81564ef1>] kasan_object_err+0x21/0x70
[<ffffffff8156518a>] kasan_report_error+0x1fa/0x4e0
[<ffffffff81564ad7>] ? kasan_slab_free+0x87/0xb0
[<ffffffff81565733>] __asan_report_store8_noabort+0x43/0x50
[<ffffffffa0fc0f54>] ? snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
[<ffffffffa0fc0f54>] snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
[<ffffffffa0fc0fb1>] snd_usb_audio_dev_free+0x31/0x40 [snd_usb_audio]
[<ffffffff8243c78a>] __snd_device_free+0x12a/0x210
[<ffffffff8243d1f5>] snd_device_free_all+0x85/0xd0
[<ffffffff8242cae4>] release_card_device+0x34/0x130
[<ffffffff81ef1846>] device_release+0x76/0x1e0
[<ffffffff81b37ad7>] kobject_release+0x107/0x370
.....
Object at ffff8801c863cc80, in cache kmalloc-2048 size: 2048
Allocated:
[<ffffffff810804eb>] save_stack_trace+0x2b/0x50
[<ffffffff81564296>] save_stack+0x46/0xd0
[<ffffffff8156450d>] kasan_kmalloc+0xad/0xe0
[<ffffffff81560d1a>] kmem_cache_alloc_trace+0xfa/0x240
[<ffffffff8214ea47>] usb_alloc_dev+0x57/0xc90
[<ffffffff8216349d>] hub_event+0xf1d/0x35f0
....
Freed:
[<ffffffff810804eb>] save_stack_trace+0x2b/0x50
[<ffffffff81564296>] save_stack+0x46/0xd0
[<ffffffff81564ac1>] kasan_slab_free+0x71/0xb0
[<ffffffff81560929>] kfree+0xd9/0x280
[<ffffffff8214de6e>] usb_release_dev+0xde/0x110
[<ffffffff81ef1846>] device_release+0x76/0x1e0
....
It's the code trying to clear drvdata of the assigned usb_device where
the usb_device itself was already released in usb_release_dev() after
the disconnect callback.
This patch fixes it by checking whether the code path is via the
disconnect callback, i.e. chip->shutdown flag is set.
Fixes: 79289e2419 ('ALSA: usb-audio: Refer to chip->usb_id for quirks...')
Reported-and-tested-by: Shuah Khan <shuahkh@osg.samsung.com>
Cc: <stable@vger.kernel.org> # v4.6+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error checking here is messed up so we could end up dereferencing
-EFAULT.
Fixes: a16039cbf1 ('ALSA: line6: Add hwdep interface to access the POD control messages')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit c039aaa77a was incomplete,
missing part of the setup for Live. This makes also audio input work,
in addition to audio output.
Fixes: c039aaa77a
Reported-by: Eddi De Pieri <eddi@depieri.net>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, usb-line6 module exports an array of MIDI manufacturer ID and
usb-pod module uses it. However, the declaration is not the definition in
common header. The difference is explicit length of array. Although
compiler calculates it and everything goes well, it's better to use the
same representation between definition and declaration.
This commit fills the length of array for usb-line6 module. As a small
good sub-effect, this commit suppress below warnings from static analysis
by sparse v0.5.0.
sound/usb/line6/driver.c:274:43: error: cannot size expression
sound/usb/line6/driver.c:275:16: error: cannot size expression
sound/usb/line6/driver.c:276:16: error: cannot size expression
sound/usb/line6/driver.c:277:16: error: cannot size expression
Fixes: 705ececd1c ("Staging: add line6 usb driver")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
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>
sound/usb/line6/driver.c:484:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values.
NULL check before some freeing functions is not needed.
Based on checkpatch warning
"kfree(NULL) is safe this check is probably not required"
and kfreeaddr.cocci by Julia Lawall.
Generated by: scripts/coccinelle/free/ifnullfree.cocci
CC: Andrej Krutak <dev@andree.sk>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We must do it this way, because e.g. POD X3 won't play any sound unless
the host listens on the bulk EP, so we cannot export it only via libusb.
The driver currently doesn't use the bulk EP messages in other way,
in future it could e.g. sense/modify volume(s).
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only initialize PCM for POD HD devices that support it.
No POD HD seems to support MIDI, thus drop the initialization.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes audio in/out and basic initialization via control EP (emulates
what original driver does). The initialization is done similarly to original
POD, firmware and serial IDs are read and exported via sysfs.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all PODs use MIDI via USB data interface, thus allow avoiding
that code and instead using direct processing.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
POD X3 can initialize similarly to older PODs, but it doesn't have the MIDI
interface. Instead, configuration is done via proprietary bulk EP messages.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. POD X3 seems to require playback data to be sent to it to generate
capture data. Otherwise the device stalls and doesn't send any more capture
data until it's reset.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Splits max_packet_size to max_packet_size_in/out (e.g. for
different channel counts).
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This has two parts:
* intervals_per_second setup
(high speed needs 8000, instead of 1000)
* iso_buffers setup (count of iso buffers depends on
USB speed, 2 is not enough for high speed)
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This basically changes LINE6_ISO_BUFFERS constant to a configurable
iso_buffers property.
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid getting sample rate on B850V3 CP2114 as it is unsupported and
causes noisy "current rate is different from the runtime rate" messages
when playback starts.
Signed-off-by: Ken Lin <ken.lin@advantech.com.tw>
Signed-off-by: Akshay Bhat <akshay.bhat@timesys.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kmalloc already print similar error once failing to alloc
enough memory, so let's remove this dump here.
Signed-off-by: Shawn Lin <shawn.lin@rock-chips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Users of devices affected by the Tenor feedback data error report
buffer underruns, even with the +/- 0x1.0000 quirk applied.
Compensating the error with 0xf000 instead seems to reliably fix
that issue.
See
https://sourceforge.net/p/alsa/mailman/message/35230259/
Reported-and-tested-by: Norman Nolte <norman.nolte@gmx.net>
Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk seems to be necessary not only for TEAC UD-H01 devices, but to
more that are based on the Tenor 8802TL chipset. Devices built by T+A
are affected too, and they apparently all use the same USB PID:PID.
Extend the quirky handling for that device as well, and rename the
quirks flag.
Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
That's a quirk, after all, so move it where to all the other quirks
live.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 02fc76f6a changed base of the sysfs attributes from device to card.
The "show" callbacks dereferenced wrong objects because of this.
Fixes: 02fc76f6a7 ('ALSA: line6: Create sysfs via snd_card_add_dev_attr()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there's an error, pcm is released in line6_pcm_acquire already.
Fixes: 247d95ee6d ('ALSA: line6: Handle error from line6_pcm_acquire()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
kmalloc will print enough information in case of failure.
Signed-off-by: Wolfram Sang <wsa-dev@sang-engineering.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ELP HD USB Camera (05a3:9420) needs this quirk for suppressing
the unsupported sample rate inquiry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481
Cc: <stable@vger.kernel.org>
Signed-off-by: Vittorio Gambaletta <linuxbugs@vittgam.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VF0610 does not support reading the sample rate which leads to many
lines of "cannot get freq at ep 0x82". This patch adds the USB ID
(0x041E:4080) to snd_usb_get_sample_rate_quirk() list.
Signed-off-by: Piotr Karasinski <peter.karasinski@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To allow for structure randomisation, replace the in order struct
initialisation style with explicit field style.
The Coccinelle semantic patch used to make this change is as follows:
@decl@
identifier i1,fld;
type T;
field list[n] fs;
@@
struct i1 {
fs
T fld;
...};
@@
identifier decl.i1,i2,decl.fld;
expression e;
position bad.p, bad.fix;
@@
struct i1 i2@p = { ...,
+ .fld = e
- e@fix
,...};
Signed-off-by: Amitoj Kaur Chawla <amitoj1606@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phoenix Audio has yet another device with another id (even a different
vendor id, 0556:0014) that requires the same quirk for the sample
rate.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the
same workaround as TMX320 for avoiding the firmware bug. It fixes the
frequent error about the sample rate inquiries and the slow device
probe as consequence.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many USB audio devices with buggy firmware that don't react
with the sample rate reading properly. This often results in the
flood of error messages and slowing down the operation.
The sample rate read back is basically only for confirming the sample
rate setup, and it's not critically important. As a compromise, in
this patch, we stop the sample rate read back once when the device
gives errors more than tolerance (twice, as of now). This should
improve most of error cases while we still can catch the firmware
bugginess.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
we've had a very calm development cycle, so far. Here are the few
fixes for HD-audio and USB-audio, all of which are small and easy.
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Merge tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"We've had a very calm development cycle, so far. Here are the few
fixes for HD-audio and USB-audio, all of which are small and easy"
* tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix inconsistent monitor_present state until repoll
ALSA: hda - Fix regression of monitor_present flag in eld proc file
ALSA: usb-audio: Skip volume controls triggers hangup on Dell USB Dock
ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
ALSA: sscape: Use correct format identifier for size_t
ALSA: usb-audio: Add a quirk for Plantronics BT300
ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320
ALSA: hda - Bind with i915 only when Intel graphics is present
This is Dell usb dock audio workaround.
It was fixed the master volume keep lower.
[Some background: the patch essentially skips the controls of a couple
of FU volumes. Although the firmware exposes the dB and the value
information via the usb descriptor, changing the values (we set the
min volume as default) screws up the device. Although this has been
fixed in the newer firmware, the devices are shipped with the old
firmware, thus we need the workaround in the driver side. -- tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
miniDSP USBStreamer UAC2 devices send clock validity changes with the
control field set to zero. The current interrupt handler ignores all
packets if the control field does not match the mixer element's, but
it really should only do that in case that field is needed to
distinguish multiple elements with the same ID.
This patch implements a logic that lets notifications packets pass
if the element ID is unique for a given device.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 specifies clock sources that optionally have validity controls.
This patch exposes them as mixer controls, so they can be read (and
at least in theory even be written) by userspace applications in order
to make clock selection policy decisions.
This implementation does nothing if the device is not UAC2 compliant,
or if the clock source does not define said validity control bits.
Tested with a miniDSP USBStreamer (0x2752/0x0016).
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plantronics BT300 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1". This patch adds the USB
ID of the BT300 to quirks.c and avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media fixes from Mauro Carvalho Chehab:
"Some bug fixes on au0828 and snd-usb-audio:
- the au0828+snd-usb-audio MC patch broke several things and produced
some race conditions. Better to revert the patches, and re-work on
them for a next version
- fix a regression at tuner disable links logic
- properly handle dev_state as a bitmask"
* tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
[media] Revert "[media] media: au0828 change to use Managed Media Controller API"
[media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
[media] au0828: Fix dev_state handling
[media] au0828: fix au0828_v4l2_close() dev_state race condition
[media] media: au0828 fix to clear enable/disable/change source handlers
[media] v4l2-mc: cleanup a warning
[media] au0828: disable tuner links and cache tuner/decoder
Phoenix Audio TMX320 gives the similar error when the sample rate is
asked:
usb 2-1.3: 2:1: cannot get freq at ep 0x85
usb 2-1.3: 1:1: cannot get freq at ep 0x2
....
Add the corresponding USB-device ID (1de7:0014) to
snd_usb_get_sample_rate_quirk() list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes:
- A fix in ALSA timer core to avoid possible BUG() trigger
- A fix in ALSA timer core 32bit compat layer
- A few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- Fixes of USB-audio double-free at some error paths
- A fix for memory leak in DICE driver at hotunplug
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Merge tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes:
- a fix in ALSA timer core to avoid possible BUG() trigger
- a fix in ALSA timer core 32bit compat layer
- a few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- fixes of USB-audio double-free at some error paths
- a fix for memory leak in DICE driver at hotunplug"
* tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: timer: Use mod_timer() for rearming the system timer
ALSA: hda - fix front mic problem for a HP desktop
ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call
ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
ALSA: dice: fix memory leak when unplugging
ALSA: hda - Apply fix for white noise on Asus N550JV, too
ALSA: hda - Fix white noise on Asus N750JV headphone
ALSA: hda - Asus N750JV external subwoofer fixup
ALSA: timer: fix gparams ioctl compatibility for different architectures
Unfortunately, this patch caused several regressions at au0828 and
snd-usb-audio, like this one:
https://bugzilla.kernel.org/show_bug.cgi?id=115561
It also showed several troubles at the MC core that handles pretty
poorly the memory protections and data lifetime management.
So, better to revert it and fix the core before reapplying this
change.
This reverts commit aebb2b89bf ("[media] sound/usb: Use Media
Controller API to share media resources")'
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.
This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.
Based on a patch by Takashi Iwai <tiwai@suse.de>
[Note for stable backports:
this patch requires the commit 902eb7fd1e ('ALSA: usb-audio: Minor
code cleanup in create_fixed_stream_quirk()')]
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358
Reported-by: Ralf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: Vladis Dronov <vdronov@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous pull request introduced a few WARN_ON() for Intel
HD-audio HDMI. Indeed it caught bugs, and now users get annoyed.
So this request came up: a collection of small fixes to paper over
the inconsistencies on (mostly) old Intel chipsets.
In addition, a trivial USB-audio quirk is included, too.
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Merge tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"The previous pull request introduced a few WARN_ON() for Intel
HD-audio HDMI. Indeed it caught bugs, and now users get annoyed. So
this request came up: a collection of small fixes to paper over the
inconsistencies on (mostly) old Intel chipsets.
In addition, a trivial USB-audio quirk is included, too"
* tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix missing ELD update at unplugging
ALSA: usb-audio: add Microsoft HD-5001 to quirks
ALSA: hda - Workaround for unbalanced i915 power refcount by concurrent probe
ALSA: hda - Fix spurious kernel WARNING on Baytrail HDMI
ALSA: hda - Fix forgotten HDMI monitor_present update
ALSA: hda - Really restrict i915 notifier to HSW+
The Microsoft HD-5001 webcam microphone does not support sample rate
reading as the HD-5000 one.
This results in dmesg errors and sound hanging with pulseaudio.
Signed-off-by: Victor Clément <victor.clement@openmailbox.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514
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Merge tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514"
* tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits)
ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug
ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
ALSA: mixart: silence an uninitialized variable warning
ALSA: usb-audio: Add sanity checks for endpoint accesses
ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
ALSA: hda - Limit i915 HDMI binding only for HSW and later
ALSA: hda - Fix unconditional GPIO toggle via automute
ALSA: mixart: silence unitialized variable warnings
ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
ASoC: rsnd: add simplified module explanation
ASoC: hdac_hdmi: Add broxton device ID
ASoC: Intel: Bxtn: Add Broxton PCI ID
ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops
ASoC: Intel: add dmabuffer to common sst_dsp
ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core
ASoC: Intel: Skylake: Fix whitepsace issues
ASoC: Intel: Skylake: Move module id defines
...
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>
create_fixed_stream_quirk() may cause a NULL-pointer dereference by
accessing the non-existing endpoint when a USB device with a malformed
USB descriptor is used.
This patch avoids it simply by adding a sanity check of bNumEndpoints
before the accesses.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* commit '840f5b0572ea': (381 commits)
media: au0828 disable tuner to demod link in au0828_media_device_register()
[media] touptek: cast char types on %x printk
[media] touptek: don't DMA at the stack
[media] mceusb: use %*ph for small buffer dumps
[media] v4l: exynos4-is: Drop unneeded check when setting up fimc-lite links
[media] v4l: vsp1: Check if an entity is a subdev with the right function
[media] hide unused functions for !MEDIA_CONTROLLER
[media] em28xx: fix Terratec Grabby AC97 codec detection
[media] media: add prefixes to interface types
[media] media: rc: nuvoton: switch attribute wakeup_data to text
[media] v4l2-ioctl: fix YUV422P pixel format description
[media] media: fix null pointer dereference in v4l_vb2q_enable_media_source()
[media] v4l2-mc.h: fix yet more compiler errors
[media] staging/media: add missing TODO files
[media] media.h: always start with 1 for the audio entities
[media] sound/usb: Use meaninful names for goto labels
[media] v4l2-mc.h: fix compiler warnings
[media] media: au0828 audio mixer isn't connected to decoder
[media] sound/usb: Use Media Controller API to share media resources
[media] dw2102: add support for TeVii S662
...
Change ALSA driver to use Media Controller API to share media resources
with DVB and V4L2 drivers on a AU0828 media device. Media Controller
specific initialization is done after sound card is registered. ALSA
creates Media interface and entity function graph nodes for Control,
Mixer, PCM Playback, and PCM Capture devices.
snd_usb_hw_params() will call Media Controller enable source handler
interface to request the media resource. If resource request is
granted, it will release it from snd_usb_hw_free(). If resource is
busy, -EBUSY is returned.
Media specific cleanup is done in usb_audio_disconnect().
Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
Plantronics DA45 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x4" and "cannot get freq at
ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 'umidi' object will be free'd on the error path by snd_usbmidi_free()
when tearing down the rawmidi interface. So we shouldn't try to free it
in snd_usbmidi_create() after having registered the rawmidi interface.
Found by KASAN.
Signed-off-by: Andrey Konovalov <andreyknvl@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microsoft LifeCam HD-6000 (045e:076f) requires the similar quirk for
avoiding the stall due to the invalid sample rate reads.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=111491
Signed-off-by: Lev Lybin <lev.lybin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In my patch adding native DSD support for the Oppo HA-1, the wrong vendor ID got
through. This patch fixes the vendor ID and aligns the comment.
Fixes: a4eae3a506 ('ALSA: usb: Add native DSD support for Oppo HA-1')
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new option "quirk_alias" to snd-usb-audio driver for
allowing user to pass the quirk alias list. A quirk alias consists of
a string form like 0123abcd:5678beef, which makes to apply a quirk to
a device with USB ID 0123:abcd treated as if it were 5678:beef.
This feature is useful to test an existing quirk, typically for a
newer model of the same vendor, without patching / rebuilding the
kernel driver.
The current implementation is fairly simplistic: since there is no API
for matching a usb_device_id to the given ID pair, it has an open code
to loop over the id table and matches only with vendor:product pair.
So far, this is OK, as all existing entries are with vendor:product
pairs, indeed. Once when we have another matching entry, however,
we'd need to update get_alias_quirk() as well.
Note that this option is provided only for testing / development. If
you want to have a proper support, contact to upstream for adding the
matching quirk in the driver code statically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for the later change to allow a better
quirk ID management. In the current USB-audio code, there are a few
places looking at usb_device idVendor and idProduct fields directly
even though we have already a static member in snd_usb_audio.usb_id.
This patch modifies such codes to refer to the latter field.
For achieving this, two slightly intensive changes have been done:
- The snd_usb_audio object is set/reset via dev_getdrv() for the given
USB device; it's needed for minimizing the changes for some existing
quirks that take only usb_device object.
- __snd_usbmidi_create() is introduced to receive the pre-given usb_id
argument. The exported snd_usbmidi_create() is unchanged by calling
this new function internally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TEAC UD-501/UD-503/NT-503 fail to switch properly between different
rate/format. Similar to 'Playback Design', this patch corrects the
invalid clock source error for TEAC products and avoids complete
freeze of the usb interface of 503 series.
Signed-off-by: Guillaume Fougnies <guillaume@eulerian.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had quite busy weeks in this cycle. Looking at ALSA core, the
significant changes are a few fixes wrt timer and sequencer ioctls
that have been revealed by fuzzer recently. Other than that, ASoC
core got a few updates about DAI link handling, but these are rather
straightforward refactoring.
In drivers scene, ASoC received quite lots of new drivers in addition
to bunch of updates for still ongoing Intel Skylake support and
topology API. HD-audio gained a new HDMI/DP hotplug notification via
component. FireWire got a pile of code refactoring/updates with
SCS.1x driver integration.
More highlights are shown below.
[NOTE: this contains also many commits for DRM. This is due to the
pull of drm stable branch into sound tree, as the base of i915 audio
component work for HD-audio. The highlights below don't contain
these DRM changes, as these are supposed to be pulled via drm tree in
anyway sooner or later.]
Core
- Handful fixes to harden ALSA timer and sequencer ioctls against
races reported by syzkaller fuzzer
- Irq description string can be unique to each card; only for
HD-audio for now
ASoC
- Conversion of the array of DAI links to a list for supporting
dynamically adding and removing DAI links
- Topology API enhancements to make everything more component based
and being able to specify PCM links via topology
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production; we really need to get to the
point where that can be done
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come
- Lots of new features and cleanups for the Renesas drivers
- ANC support for WM5110
- New drivers: Imagination Technologies IPs, Atmel class D speaker,
Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
- Rename PCM1792a driver to be generic pcm179x
HD-Audio
- Use audio component for i915 HDMI/DP hotplug handling
- On-demand binding with i915 driver
- bdl_pos_adj parameter adjustment for Baytrail controllers
- Enable power_save_node for CX20722; this shouldn't lead to
regression, hopefully
- Kabylake HDMI/DP codec support
- Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
machines
- A few code refactoring
FireWire
- Lots of code cleanup and refactoring
- Integrate the support of SCS.1x devices into snd-oxfw driver;
snd-scs1x driver is obsoleted
USB-audio
- Fix possible NULL dereference at disconnection
- A regression fix for Native Instruments devices
Misc
- A few code cleanups of fm801 driver
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Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We've had quite busy weeks in this cycle. Looking at ALSA core, the
significant changes are a few fixes wrt timer and sequencer ioctls
that have been revealed by fuzzer recently. Other than that, ASoC
core got a few updates about DAI link handling, but these are rather
straightforward refactoring.
In drivers scene, ASoC received quite lots of new drivers in addition
to bunch of updates for still ongoing Intel Skylake support and
topology API. HD-audio gained a new HDMI/DP hotplug notification via
component. FireWire got a pile of code refactoring/updates with
SCS.1x driver integration.
More highlights are shown below.
[ NOTE: this contains also many commits for DRM. This is due to the
pull of drm stable branch into sound tree, as the base of i915 audio
component work for HD-audio. The highlights below don't contain
these DRM changes, as these are supposed to be pulled via drm tree
in anyway sooner or later. ]
Core:
- Handful fixes to harden ALSA timer and sequencer ioctls against
races reported by syzkaller fuzzer
- Irq description string can be unique to each card; only for
HD-audio for now
ASoC:
- Conversion of the array of DAI links to a list for supporting
dynamically adding and removing DAI links
- Topology API enhancements to make everything more component based
and being able to specify PCM links via topology
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production; we really need to get to the
point where that can be done
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come
- Lots of new features and cleanups for the Renesas drivers
- ANC support for WM5110
- New drivers: Imagination Technologies IPs, Atmel class D speaker,
Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
- Rename PCM1792a driver to be generic pcm179x
HD-Audio:
- Use audio component for i915 HDMI/DP hotplug handling
- On-demand binding with i915 driver
- bdl_pos_adj parameter adjustment for Baytrail controllers
- Enable power_save_node for CX20722; this shouldn't lead to
regression, hopefully
- Kabylake HDMI/DP codec support
- Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
machines
- A few code refactoring
FireWire:
- Lots of code cleanup and refactoring
- Integrate the support of SCS.1x devices into snd-oxfw driver;
snd-scs1x driver is obsoleted
USB-audio:
- Fix possible NULL dereference at disconnection
- A regression fix for Native Instruments devices
Misc:
- A few code cleanups of fm801 driver"
* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
ALSA: timer: Code cleanup
ALSA: timer: Harden slave timer list handling
ALSA: hda - Add fixup for Dell Latitidue E6540
ALSA: timer: Fix race among timer ioctls
ALSA: hda - add codec support for Kabylake display audio codec
ALSA: timer: Fix double unlink of active_list
ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
ALSA: hda - fix the headset mic detection problem for a Dell laptop
ALSA: hda - Fix white noise on Dell Latitude E5550
ALSA: hda_intel: add card number to irq description
ALSA: seq: Fix race at timer setup and close
ALSA: seq: Fix missing NULL check at remove_events ioctl
ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
ASoC: AMD: Add missing include file
ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
ALSA: usb: Add native DSD support for Oppo HA-1
ASoC: Make aux_dev more like a generic component
ASoC: bcm2835: cleanup includes by ordering them alphabetically
ASoC: AMD: Manage ACP 2.x SRAM banks power
...
Pull trivial tree updates from Jiri Kosina.
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial:
floppy: make local variable non-static
exynos: fixes an incorrect header guard
dt-bindings: fixes some incorrect header guards
cpufreq-dt: correct dead link in documentation
cpufreq: ARM big LITTLE: correct dead link in documentation
treewide: Fix typos in printk
Documentation: filesystem: Fix typo in fs/eventfd.c
fs/super.c: use && instead of & for warn_on condition
Documentation: fix sysfs-ptp
lib: scatterlist: fix Kconfig description
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>
ALSA PCM may still have a leftover instance after disconnection and
it delays its release. The problem is that the PCM close code path of
USB-audio driver has a call of snd_usb_autosuspend(). This involves
with the call of usb_autopm_put_interface() and it may lead to a
kernel Oops due to the NULL object like:
BUG: unable to handle kernel NULL pointer dereference at 0000000000000190
IP: [<ffffffff815ae7ef>] usb_autopm_put_interface+0xf/0x30 PGD 0
Call Trace:
[<ffffffff8173bd94>] snd_usb_autosuspend+0x14/0x20
[<ffffffff817461bc>] snd_usb_pcm_close.isra.14+0x5c/0x90
[<ffffffff8174621f>] snd_usb_playback_close+0xf/0x20
[<ffffffff816ef58a>] snd_pcm_release_substream.part.36+0x3a/0x90
[<ffffffff816ef6b3>] snd_pcm_release+0xa3/0xb0
[<ffffffff816debb0>] snd_disconnect_release+0xd0/0xe0
[<ffffffff8114d417>] __fput+0x97/0x1d0
[<ffffffff8114d589>] ____fput+0x9/0x10
[<ffffffff8109e452>] task_work_run+0x72/0x90
[<ffffffff81088510>] do_exit+0x280/0xa80
[<ffffffff8108996a>] do_group_exit+0x3a/0xa0
[<ffffffff8109261f>] get_signal+0x1df/0x540
[<ffffffff81040903>] do_signal+0x23/0x620
[<ffffffff8114c128>] ? do_readv_writev+0x128/0x200
[<ffffffff810012e1>] prepare_exit_to_usermode+0x91/0xd0
[<ffffffff810013ba>] syscall_return_slowpath+0x9a/0x120
[<ffffffff817587cd>] ? __sys_recvmsg+0x5d/0x70
[<ffffffff810d2765>] ? ktime_get_ts64+0x45/0xe0
[<ffffffff8115dea0>] ? SyS_poll+0x60/0xf0
[<ffffffff818d2327>] int_ret_from_sys_call+0x25/0x8f
We have already a check of disconnection in snd_usb_autoresume(), but
the check is missing its counterpart. The fix is just to put the same
check in snd_usb_autosuspend(), too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=109431
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Oppo HA-1. It uses a XMOS chipset
but they use their own vendor ID.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().
Signed-off-by: Geliang Tang <geliangtang@163.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoid getting sample rate on AudioQuest DragonFly as it is unsupported
and causes noisy "cannot get freq at ep 0x1" messages when playback
starts.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
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>
The usb_protocol_ops structures are never modified, so declare them as
const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fix multiple spelling typos found in
various part of kernel.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The snd_rawmidi_global_ops structures are never modified, so declare them
as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 939f325f4a ("usb: add usb_endpoint_maxp() macro") and commit
29cc88979a ("USB: use usb_endpoint_maxp() instead of le16_to_cpu()")
introduced a new helper macro. This trivial patch convert remaining
users found in ua101 driver.
Signed-off-by: Cheah Kok Cheong <thrust73@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of the many faults of the QinHeng CH345 USB MIDI interface chip is
that it does not handle received SysEx messages correctly -- every second
event packet has a wrong code index number, which is the one from the last
seen message, instead of 4. For example, the two messages "FE F0 01 02 03
04 05 06 07 08 09 0A 0B 0C 0D 0E F7" result in the following event
packets:
correct: CH345:
0F FE 00 00 0F FE 00 00
04 F0 01 02 04 F0 01 02
04 03 04 05 0F 03 04 05
04 06 07 08 04 06 07 08
04 09 0A 0B 0F 09 0A 0B
04 0C 0D 0E 04 0C 0D 0E
05 F7 00 00 05 F7 00 00
A class-compliant driver must interpret an event packet with CIN 15 as
having a single data byte, so the other two bytes would be ignored. The
message received by the host would then be missing two bytes out of six;
in this example, "F0 01 02 03 06 07 08 09 0C 0D 0E F7".
These corrupted SysEx event packages contain only data bytes, while the
CH345 uses event packets with a correct CIN value only for messages with
a status byte, so it is possible to distinguish between these two cases by
checking for the presence of this status byte.
(Other bugs in the CH345's input handling, such as the corruption resulting
from running status, cannot be worked around.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CH345 USB MIDI chip has two output ports. However, they are
multiplexed through one pin, and the number of ports cannot be reduced
even for hardware that implements only one connector, so for those
devices, data sent to either port ends up on the same hardware output.
This becomes a problem when both ports are used at the same time, as
longer MIDI commands (such as SysEx messages) are likely to be
interrupted by messages from the other port, and thus to get lost.
It would not be possible for the driver to detect how many ports the
device actually has, except that in practice, _all_ devices built with
the CH345 have only one port. So we can just ignore the device's
descriptors, and hardcode one output port.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device has no mixer (and identifies itself as such), so just skip
the mixer definition.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum
sample frequency, consideration must be made for the fact that four bytes
of the packet contain a length descriptor and consequently must not be
counted as part of the audio data.
This is corroborated by the wMaxPacketSize for this device, which is 108
bytes according for the USB playback endpoint descriptor. The frame size
is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out
as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte
length descriptor.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)
The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).
In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.
For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.
The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.
In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.
Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.
The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.
Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Tested-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices.
Tested that the Nocturn shows up in aconnect, and that it can be used
as a control surface (using the xtor synthesizer patch editor).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.
We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.
Detailed explanation and rationale:
The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:
maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
>> (16 - ep->datainterval);
Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.
The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.
In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.
The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.
Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).
This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.
The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.
For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.
Rephrasing the maxsize expression to:
maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
(frame_bits >> 3);
for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.
We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):
Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56
This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .
(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)
Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables interrupt transfer mode for MIDI ports on newer
Boss/Roland devices such as the GT-100/001 which support interrupt
transfer on both IN and OUT MIDI endpoints. Previously this wasn't being
enabled for these devices as the code was specifically looking for the
scenario where the IN endpoint supported interrupt transfer and the OUT
endpoint was bulk transfer. Newer devices support interrupt transfer for
both endpoints.
This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland
VS-20.
It would benefit from some regresison testing with other devices if
possible.
Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to verify that "value" is either zero or one, so we test if it
is greater than one. Unfortunately, this is a signed int so it could
also be negative. I think this is harmless but it introduces a static
checker warning. Let's make "value" unsigned.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.
This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.
Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check of cval->cached should be zero-based (including master channel).
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In theory, the device may get suspended even at runtime PM suspend.
Currently we don't save the mixer state for autopm, and it may bring
inconsistency.
This patch removes the special handling for autosuspend.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent fix of runtime PM for USB-audio driver, we got a
lockdep warning like:
=============================================
[ INFO: possible recursive locking detected ]
4.2.0-rc8+ #61 Not tainted
---------------------------------------------
pulseaudio/980 is trying to acquire lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
but task is already holding lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
This comes from snd_usb_autoresume() invoking down_read() and it's
used in a nested way. Although it's basically safe, per se (as these
are read locks), it's better to reduce such spurious warnings.
The read lock is needed to guarantee the execution of "shutdown"
(cleanup at disconnection) task after all concurrent tasks are
finished. This can be implemented in another better way.
Also, the current check of chip->in_pm isn't good enough for
protecting the racy execution of multiple auto-resumes.
This patch rewrites the logic of snd_usb_autoresume() & co; namely,
- The recursive call of autopm is avoided by the new refcount,
chip->active. The chip->in_pm flag is removed accordingly.
- Instead of rwsem, another refcount, chip->usage_count, is introduced
for tracking the period to delay the shutdown procedure. At
the last clear of this refcount, wake_up() to the shutdown waiter is
called.
- The shutdown flag is replaced with shutdown atomic count; this is
for reducing the lock.
- Two new helpers are introduced to simplify the management of these
refcounts; snd_usb_lock_shutdown() increases the usage_count, checks
the shutdown state, and does autoresume. snd_usb_unlock_shutdown()
does the opposite. Most of mixer and other codes just need this,
and simply returns an error if it receives an error from lock.
Fixes: 9003ebb13f ('ALSA: usb-audio: Fix runtime PM unbalance')
Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Gustard DAC-X20U.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input terminal parser recurses into the referenced clock entity to verify
it is existant and thus the terminal descriptor is valid. The actual property
values of the term instance which is initially parsed must not be overriden by
the recursion. For this to work the term properties have to be assigned after
recursing into the referenced clock entity descriptors.
Signed-off-by: Julian Scheel <julian@jusst.de>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix for deadlock in PM in commit [1ee23fe07ee8: ALSA: usb-audio:
Fix deadlocks at resuming] introduced a new check of in_pm flag.
However, the brainless patch author evaluated it in a wrong way
(logical AND instead of logical OR), thus usb_autopm_get_interface()
is wrongly called at probing, leading to unbalance of runtime PM
refcount.
This patch fixes it by correcting the logic.
Reported-by: Hans Yang <hansy@nvidia.com>
Fixes: 1ee23fe07e ('ALSA: usb-audio: Fix deadlocks at resuming')
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M-Audio Transit exposes an interface with a SYNC_NONE attribute.
This is not a valid value according to the USB audio classspec. However
there is a sync endpoint associated to this record. Changing the logic to
try to use this sync endpoint allows for seamless transitions between
altset 2 and altset 3. If any errors happen, the behavior remains the same.
$ more /proc/asound/card1/stream0
M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (ADAPTIVE)
Rates: 48001 - 96000 (continuous)
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (NONE)
Rates: 8000 - 48000 (continuous)
Interface 1
Altset 3
Format: S16_LE
Channels: 2
Endpoint: 3 OUT (ASYNC)
Rates: 8000 - 48000 (continuous)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a transition occurs between alternate settings that do not use the
same synchronization method, the substream pointers were not reset.
This prevents audio from being played during the second transition.
Identified and tested with M-Audio Transit device
(0763:2006 Midiman M-Audio Transit)
Details of the issue:
First playback to adaptive endpoint:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
[ 3169.297556] usb 1-2: setting usb interface 1:1
[ 3169.297568] usb 1-2: Creating new playback data endpoint #3
[ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0
[ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000
first playback to asynchronous endpoint:
$ aplay -Dhw:1,0 ~/16_48.wav
Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian,
Rate 48000 Hz, Stereo
[ 3204.520251] usb 1-2: setting usb interface 1:3
[ 3204.520264] usb 1-2: Creating new playback data endpoint #3
[ 3204.520272] usb 1-2: Creating new capture sync endpoint #83
[ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
[ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000
[ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000
second playback to adaptive endpoint: no audio and error on terminal:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
aplay: pcm_write:1939: write error: Input/output error
[ 3239.483589] usb 1-2: setting usb interface 1:1
[ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000
[ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
This last line shows that a sync endpoint is used when it shouldn't.
The sync endpoint is no longer valid and the pointers are corrupted
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB Audio Class version 2.0 supports three different parameter block sizes for
CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes
(5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter
Block). Use the correct size according to the specific control as it was
already done for UACv1. The allocated block size for control requests is
increased to support the 4 byte worst case.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2.
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a monitor stream is active, the next PCM stream access results in
EBUSY error because of the check in line6_stream_start(). Fix this by
just skipping the submission of pending URBs when the stream is
already running instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101431
Cc: <stable@vger.kernel.org> # v4.0+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Steinberg MI2 and MI4 interfaces are compatible with the USB class
audio spec, but the MIDI part of the devices is reported as a vendor
specific interface.
This patch adds entries to quirks-table.h to recognize the MIDI
endpoints. Audio functionality was already working and is unaffected by
this change.
Signed-off-by: Dominic Sacré <dominic.sacre@gmx.de>
Signed-off-by: Albert Huitsing <albert@huitsing.nl>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kirill Marinushkin