Signed-off-by: Paul B Mahol <onemda@gmail.com>

(cherry picked from commit 5cc0057f)
Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: James Almer <jamrial@gmail.com>

Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: James Almer <jamrial@gmail.com>

Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: James Almer <jamrial@gmail.com>

Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: James Almer <jamrial@gmail.com>

Signed-off-by: Paul B Mahol <onemda@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/ffmpegReviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>

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 090b7406.
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>

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>

At least version 0.6.2 is needed since commit
df3222d4.
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.

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Fixes a crash which happened when someone tried to encode mono.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>

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 is more correct.
Signed-off-by: Paul B Mahol <onemda@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>

Signed-off-by: Paul B Mahol <onemda@gmail.com>

This fixes a double free in he error case.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

This fixes a segfault caused by passing NULL to ff_filter_frame
when an error occurs.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>

Signed-off-by: Paul B Mahol <onemda@gmail.com>

These are new modes which are supposed to be more SIMD friendly.
Signed-off-by: Paul B Mahol <onemda@gmail.com>

Reviewed-by: Steven Liu <lq@chinaffmpeg.org>

This can avoid OOM for min_size close to FFmpeg's arbitrary alloc limits.