This patch is taking care of https://trac.ffmpeg.org/ticket/6834.
It seems that one of the control operations that was available to get
the free decoders input slots was removed.
There is another control operation to retrieve the used slots. Given
that the input slot count is hardcoded to 4 in mpp at this point,
replacing the old control operation by the other one.
This was tested on Rockchip ROCK64.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
The PERSIST_RPARAM_A_RExt_Sony_1 bitstream has an out-of-range value
and has therefore been superseded.
It is otherwise identical, and decodes the same.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: signed integer overflow: 1024 + 2147483640 cannot be represented in type 'int'
Fixes: 4671/clusterfuzz-testcase-minimized-6027464343027712
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It was truncated to int later on anyway. Fate test changes are due to rounding
instead of truncation.
Fixes fate test failures on x86-32 (gcc 4.8 (Ubuntu 4.8.5-2ubuntu1~14.04.1))
after 090b740680.
Signed-off-by: Marton Balint <cus@passwd.hu>
Certain AIFF files encode XA ADPCM compressed audio using a chunk
with the tag `APCM`. Aside from this custom chunk type, they're
otherwise standard AIFF files. I've only observed these files in the
Sega Saturn game Sonic Jam so far.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
w32threads and os2threads both support static mutex initialization now,
so don't limit it to pthreads only.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
At least version 0.6.2 is needed since commit
df3222d4bb.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
If the stream was aborted using the libavformat interrupt callback, we
don't want it to log the reconnect warning. (Exiting after logging this
warning worked well, so this is only for avoiding the ugly warning.)
This prevents a possible crash in CreateDeviceEx when using faulty
response from GetAdapterDisplayModeEx and allows ffmpeg to fallback to
classic d3d9.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
It makes no sense to return an error after the first reconnect, and then
somehow resume the next time it's called. Usually this will lead to
demuxer errors. Make reconnecting block instead, until it has either
successfully reconnected, or given up.
Also make the wait reasonably interruptible. Since there is no mechanism
for this in the API, polling is the best we can do. This behaves roughly
the same as other interruptible network functions in libavformat.
(The original code would work if it returned AVERROR(EAGAIN) or so,
which would make retry_transfer_wrapper() repeat the read call. But I
think having an explicit loop for this is better anyway.)
I also snuck in a fix for reconnect_at_eof. It has to check for
AVERROR_EOF, not 0.
Deprecate the entire library. Merged years ago to provide compatibility
with Libav, it remained unmaintained by the FFmpeg project and duplicated
functionality provided by libswresample.
In order to improve consistency and reduce attack surface, as well as to ease
burden on maintainers, it has been deprecated. Users of this library are asked
to migrate to libswresample, which, as well as providing more functionality,
is faster and has higher accuracy.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This fixes#6940
Although undocumented, AudioToolbox seems to require the data supplied
by the callback (i.e. ffat_encode_callback) being unchanged until the
next time the callback is called. In the old implementation, the
AVBuffer backing the frame is recycled after the frame is freed, and
somebody else (maybe the decoder) will write into the AVBuffer and
change the data. AudioToolbox then encodes some wrong data and noise
is produced. Retaining a frame reference solves this problem.
Signed-off-by: James Almer <jamrial@gmail.com>
