Fix a dead lock under certain conditions. Let's assume we have a queue of 1
message max, 2 senders, and 1 receiver.
Scenario (real record obtained with debug added):
[...]
SENDER #0: acquired lock
SENDER #0: queue is full, wait
SENDER #1: acquired lock
SENDER #1: queue is full, wait
RECEIVER: acquired lock
RECEIVER: reading a msg from the queue
RECEIVER: signal the cond
RECEIVER: acquired lock
RECEIVER: queue is empty, wait
SENDER #0: writing a msg the queue
SENDER #0: signal the cond
SENDER #0: acquired lock
SENDER #0: queue is full, wait
SENDER #1: queue is full, wait
Translated:
- initially the queue contains 1/1 message with 2 senders blocking on
it, waiting to push another message.
- Meanwhile the receiver is obtaining the lock, read the message,
signal & release the lock. For some reason it is able to acquire the
lock again before the signal wakes up one of the sender. Since it
just emptied the queue, the reader waits for the queue to fill up
again.
- The signal finally reaches one of the sender, which writes a message
and then signal the condition. Unfortunately, instead of waking up
the reader, it actually wakes up the other worker (signal = notify
the condition just for 1 waiter), who can't push another message in
the queue because it's full.
- Meanwhile, the receiver is still waiting. Deadlock.
This scenario can be triggered with for example:
tests/api/api-threadmessage-test 1 2 100 100 1 1000 1000
One working solution is to make av_thread_message_queue_{send,recv}()
call pthread_cond_broadcast() instead of pthread_cond_signal() so both
senders and receivers are unlocked when work is done (be it reading or
writing).
This second solution replaces the condition with two: one to notify the
senders, and one to notify the receivers. This prevents senders from
notifying other senders instead of a reader, and the other way around.
It also avoid broadcasting to everyone like the first solution, and is,
as a result in theory more optimized.
fix default basefreq/endfreq comparison
on platform that does not do comparison
in double type
found on zeranoe 32-bit build, where
default freq range is detected as non-default
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If the chroma components are subsampled, smaller buffers are allocated
for them. In that case the maximal block_offset for the chroma
components is not as large as for the luma component.
This fixes out of bounds writes causing segmentation faults or memory
corruption.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
There are a couple of major changes here:
1. Start using TNS coefficient compression.
2. Start using 3 bits per coefficient maximum for short windows.
The bits we save from these 2 changes seem to make a nice impact on the
rest of the file/windows.
3. Remove special case gain checking for short windows.
4. Modify the coefficient loop to support up to 3 windows.
The additional restrictions on TNS were something that was no in the
specifications and furthermore restricting TNS to only low energy short
windows was done to compensate for bugs elsewhere in the code.
Overall, the improvements here reduce crackling artifacts heard in very
noisy tracks.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
The original plan was to have TNS use data from the PNS search to better
tune itself to noise but this was never used nor necessary. This should
slightly boost the PNS accuracy if TNS was used.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
It is required to call va_end for each invocation of va_start within the
same function.
Fixes: CID 1341583.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Pretty standard macros, these should help libav*
users avoid repeating ver.si.on parsing code,
which aids in compatibility-checking tasks like
identifying FFmpeg from Libav (_MICRO >= 100 check).
Something many are doing since we are not
intercompatible anymore.
Signed-off-by: Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
Fixes out of array read
Fixes: d41d8cd98f00b204e9800998ecf8427e/signal_sigsegv_321165b_7641_077dfcd8cbc80b1c0b470c8554cd6ffb.bit
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes out of array read
Fixes: 99d142c47e6ba3510a74b872a1a2ae72/asan_heap-oob_11b36f4_3811_0f5c69e7609a88a580135678de1df844.dxa
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Thiss commit removes the experimental flag from the native AAC Encoder
and thus makes it the default.
After a lot of work, done by myself and Claudio Freire, the quality of
this encoder rivals and surpasses libfdk_aac in some situations. The
encoder had instability issues earlier which prevented it from having
its experimental flag removed, however the last commits done by Claudio
removed the last known source of instability and solved a lot of
problems which were previously observed. The issues were caused by the
various coding tools interfering with the scalefactor indices. Thus,
with these problems solved, it should now be possible to declare this
encoder as the default and recommend that the users should use it
instead of others provided by external libraries, as it is both faster
and has a subjectively higher quality with selected tracks.
The encoder has still yet to be fine tuned for every possible audio file
type like music or voice, so it is hoped that with the experimental flag
removed the users should be able to provide feedback and make the
encoder better than the alternatives for every type of audio and at
every bitrate.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Since the next commit removes the experimental flag from the encoder
it's better to update the documentation which has been around in its
current form for as long as the encoder itself.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
ANMR has some interesting things coming up but is currently not in a
shape fit for non-experimental usage. Same with "FAST".
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This coder produces a much lower quality audio than the rest, is much
slower and is unstable. Hasn't been updated for a very long time as
well, hence it is more appropriate to remove it since it also depends on
a big burden of a code (the encode_window_bands_info function which is
just as old, just as unstable and bad and in no way modifiable or
fixable).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
QuickTime metadata can come after trak data. Add indicator for which trak is being parsed (-1 if none) so that global metadata after the trak can be parsed.
Signed-off-by: Neil Birkbeck <neil.birkbeck@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Using -ss as an input option shifts timestamps down by the seek, so it
doesn't have to be added to the recording time when checking whether to
stop.
Fixes#977
Signed-off-by: Simon Thelen <ffmpeg-dev@c-14.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
