ffmpeg audio crash fix (sse2 alignment)

Certain ffmpeg audio decoders use 32 bit float samples internally (wma,
eac3, ...). They are then exported to the calling application as 16 bit
integer.

That conversion is done by faster sse2 code if your processor supports it.
However, sse2 instructions require data buffers to be 16 byte aligned, or
hit a segfault otherwise.

Plain malloc() / realloc() ensures only 8 byte alignment, giving a 50%
chance of a crash.

FFmpeg internally uses aligned buffers a lot. It seems to be a good idea to
do likewise for input buffers as well, even if current version does not
strictly need it yet.

Libavutil/av_realloc() has a bug that can break the alignment when enlarging
an existing buffer. Thus I included a fixed version of it within
ff_audio_decoder.c.

0 files changed, 0 insertions, 0 deletions