The index of the motion vector has to be checked before being
multiplied by 2 for the array index.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
aacenc: Fix identification padding when the bitstream is already aligned.
aacenc: Write correct length for long identification strings.
aud: remove unneeded field, audio_stream_index from context
aud: fix time stamp calculation for ADPCM IMA WS
aud: simplify header parsing
aud: set pts_wrap_bits to 64.
cosmetics: indentation
aud: support Westwood SND1 audio in AUD files.
adpcm_ima_ws: fix stereo decoding
avcodec: add a new codec_id for CRYO APC IMA ADPCM.
vqa: remove unused context fields, audio_samplerate and audio_bits
vqa: clean up audio header parsing
vqa: set time base to frame rate as coded in the header.
vqa: set packet duration.
vqa: use 1/sample_rate as the audio stream time base
vqa: set stream start_time to 0.
lavc: postpone the removal of AVCodecContext.request_channels.
lavf: postpone removing av_close_input_file().
lavc: postpone removing old audio encoding and decoding API
avplay: remove the -er option.
...
Conflicts:
Changelog
libavcodec/version.h
libavdevice/v4l.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Although it has been deprecated for a long time, its intended
replacement (request_channel_layout) is not actually used anywhere, so
request_channels is currently the only way to access that functionality.
Previously this was just checked in case of slice threads,
but frame threads do not support this either currently.
Making them support this is of course the long term goal
Fixes bug155
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Remove ffmpeg.
aacenc: Simplify windowing
aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
aacenc: Deinterleave input samples before processing.
aacenc: Store channel count in AACEncContext.
aacenc: Move Q^3/4 calculation to it's own table
aacenc: Request normalized float samples instead of converting s16 samples to float.
aacpsy: Replace an if with FFMAX in LAME windowing.
aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
aacenc: cosmetics: move init() and end() to the bottom of the file.
aacenc: aac_encode_init() cleanup
XWD encoder and decoder
vc1: don't read the interpfrm and bfraction elements for interlaced frames
mxfdec: fix memleak on mxf_read_close()
westwood: split the AUD and VQA demuxers into separate files.
Conflicts:
.gitignore
Changelog
Makefile
configure
doc/ffmpeg.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/aacenc.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/img2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous implementation assumed that a new picture would always
supersede the previous picture. Similarly, presentation segments
were assumed to pertain to the most-recently-read picture.
However, each presentation segment may refer to 0 or more pictures
by their ID. Picture IDs may repeat, and a repeated picture ID
indicates that the old picture for that ID is no longer needed
and may be discarded.
The new implementation allocates a buffer with one slot for each
possible picture ID (the picture ID is a 16-bit field) and
properly decodes presentation segments so that all relevant
pictures are output upon encountering a display segment.
Given that most PGS streams are unlikely to use more than a small
fraction of the available picture IDs, it would probably be better
to use a more memory-efficient data structure. I'm lazy though, so
I leave this to a more motivated individual.
I've tested the code with MKV files in VLC (a recent revision from
their git repo) and with HandBrake (a version that I hacked up to
use ffmpeg's PGS subtitle decoder).
Review-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes memory corruption when seeking in broken streams.
a random mpeg4 in nut file was used to debug.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This matches the spec as well as the reference decoder, and fixes a bug
with interlaced frame decoding.
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* qatar/master: (25 commits)
riff: fix invalid av_freep() calls on EOF in ff_read_riff_info
pam: Fix a typo that broke writing and reading PAM files.
mxfdec: fix memleak on av_realloc failures
mxfdec: Do not parse slices or DeltaEntryArrays.
mxfdec: hybrid demuxing/seeking solution
mxfdec: Add Avid's essence element key.
mfxdec: Separate mxf_essence_container_uls for audio and video.
mxfdec: Compute packet offsets properly.
mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack.
mxfdec: use av_dlog() for 'no corresponding source package found'
mxfdec: Make mxf->partitions sorted by offset.
mxfdec: parse ThisPartition
mxfdec: Speed up metadata and index parsing.
mxfdec: Make sure DataDefinition is consistent between material track and source track.
mxfdec: add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
mxfdec: Add hack that adjusts the n_delta calculation when system items are present.
mxfdec: Parse IndexTableSegments and convert them into AVIndexEntry arrays.
mxfdec: Move FooterPartition to MXFContext and make sure it is never zero.
mxfdec: check return value of avio_seek
mxfdec: skip to end of structural sets
...
Conflicts:
configure
libavcodec/pnm.c
libavformat/mxfdec.c
libavformat/riff.c
libavformat/rtsp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This was a regression that came in when I switched to using the
h.264 annex b filter all the time. As the filter modifies extradata,
its use violates the statelessness assumption that exists in the
'ffmpeg' command line tool, and maybe elsewhere. It assumes that
a docoder can be reinitalised and pointed to an existing stream and
get the same results.
For now, the only way to meet this requirement is to backup the
extradata.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
the written length was off by 2 causing aac decoders to fail with the data.
lucky the encoder was marked as experimental and not used much
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
