* commit '5caf039ba2b4be067568a30146f29008d8db28d0':
Prepare for 11_beta2 Release
Conflicts:
RELEASE
Not merged as theres no FFmpeg 11_beta2
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f9f34cb9983ec6f4ef119c34b726d3b39c143110':
ogg: Use separate classes for the aliases
Conflicts:
libavformat/oggenc.c
See: 2ccc6ff03a
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This option facilitates testing shared libarary builds: for instance
fate builders do no longer need to set LD_LIBRARY_PATH as the binaries will
get the right search paths hardcoded into their executable file.
This option is only meant to be used for testing purposes: The installed
libraries must not move around in the file system, and doing so will
cause a lot of subtle problems. For more information why using RPATH is
dangerous, please refer to
https://blog.flameeyes.eu/2010/06/the-why-and-how-of-rpath
Unfortunately this was not explicitly documented and thus
might be risky.
But all uses I could find in FFmpeg and one in VLC had a memleak
in these cases, and I could not find any that relied on the previous
behaviour.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
~560 → ~500 decicycles
This is following the comments from Michael in
https://ffmpeg.org/pipermail/ffmpeg-devel/2014-August/160599.html
Using 2 registers for accumulator didn't help. On the other hand,
some re-ordering between the movs and psadbw allowed going ~538 to ~500.
Not having allocated it is not a good reason to leave the object
in an undetermined state. Though a particular setting like the
AV_EF_* flags could be useful to control that behaviour.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also correctly namespace other functions in vidstabutils, and decrease
difference from Libav.
Initial-patch-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3526ab891c28396ada8b58bf7647309bab30de1d':
qt-faststart: Undefine fseeko/ftello before defining them
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '051aadeed104ecbe8ee4850ec2d7e5394f5e1ccd':
ogg: Provide aliases for Speex, Opus and audio-only ogg
Conflicts:
Changelog
libavformat/oggenc.c
libavformat/version.h
See: 2ccc6ff03a
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In some cases, in particular if several blocks are needed because of
the channel layout (e.g. 2.1), the information used to write the
trailing bits terminating the sample data was not reset.
This would cause potential desync on the decoder, although decoded
samples were actually mostly fine.
Fixes ticket #3879.
* commit '4d6c5152849e23a4cc0f6a6ac2880c01ebcd301b':
electronicarts: do not fail on zero-sized chunks
Conflicts:
libavformat/electronicarts.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'dc4b2e7d33903a6b9380e8a84b22b3a20facbb08':
rv34: use ff_mpeg_update_thread_context only when decoder is fully initialized
Merged-by: Michael Niedermayer <michaelni@gmx.at>
At least one FATE sample contains such chunks and happens to work simply
by accident (due to find_stream_info() swallowing the error).
CC: libav-stable@libav.org
MpegEncContext based decoders are only fully initialized after the first
ff_thread_get_buffer() call. The RV30/40 decoders may fail before a frame
buffer was requested. ff_mpeg_update_thread_context() fails on half
initialized MpegEncContexts. Since this can only happen before a the
first frame was decoded there is no need to call
ff_mpeg_update_thread_context().
Based on patches by John Stebbins and tested by John Stebbins.
CC: libav-stable@libav.org
It was only validating that normal data wasn't filling the buffer.
However, extra data may be written afterwards.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Currently, the encoder will try to reduce it down to 150000, but the
decoder will complain starting at 131072 (WV_MAX_SAMPLES). Therefore,
change the loop limit.
Fixes ticket #3881.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Buffers containing copies of the AAC and AC3 header bits were not padded
before parsing, violating init_get_bits() buffer padding requirement,
leading to potential buffer read overflows.
This change adds FF_INPUT_BUFFER_PADDING_SIZE bytes to the bit buffer
for parsing the header in each of aac_parser.c and ac3_parser.c.
Based on patch by: Matt Wolenetz <wolenetz@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some streams were found to have what appeared to be truncated SPS.
Their syntax seem to be valid at least until the end of the VUI, so
try that syntax if the parsing would overflow the SPS in the
conforming syntax.
Fixes ticket #3872.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
