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Andreas Rheinhardt bdaa98dd4a avformat/matroskadec: Don't output uninitialized data for RealAudio 28.8
The Matroska demuxer splits every sequence of h Matroska Blocks into
h * w / cfs packets of size cfs; here h (sub_packet_h), w (frame_size)
and cfs (coded_framesize) are parameters from the track's CodecPrivate.

It does this by splitting the Block's data in h/2 pieces of size cfs each
and putting them into a buffer at offset m * 2 * w + n * cfs where
m (range 0..(h/2 - 1)) indicates the index of the current piece in the
current Block and n (range 0..(h - 1)) is the index of the current Block
in the current sequence of Blocks. The data in this buffer is then used
for the output packets.

The problem is that there is currently no check to actually guarantee
that no uninitialized data will be output. One instance where this is
trivially so is if h == 1; another is if cfs * h is so small that the
input pieces do not cover everything that is output. In order to
preclude this, rmdec.c checks for h * cfs == 2 * w and h >= 2. The
former requirement certainly makes much sense, as it means that for
every given m the input pieces (corresponding to the h different values
of n) form a nonoverlapping partition of the two adjacent frames of size w
corresponding to m. But precluding h == 1 is not enough, other odd
values can cause problems, too. That is because the assumption behind
the code is that h frames of size w contain data to be output, although
the real number is h/2 * 2. E.g. for h = 3, cfs = 2 and w = 3 the
current code would output four (== h * w / cfs) packets. although only
data for three (== h/2 * h) packets has been read.

(Notice that if h * cfs == 2 * w, h being even is equivalent to
cfs dividing w; the latter condition also seems very reasonable:
It means that the subframes are a partition of the frames.)

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-05-01 07:42:16 +02:00
2017-01-27 17:06:42 +01:00
2020-03-31 04:14:14 -05:00
2020-04-30 12:18:36 +02:00
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2018-01-06 18:31:37 +00:00

FFmpeg README

FFmpeg is a collection of libraries and tools to process multimedia content such as audio, video, subtitles and related metadata.

Libraries

  • libavcodec provides implementation of a wider range of codecs.
  • libavformat implements streaming protocols, container formats and basic I/O access.
  • libavutil includes hashers, decompressors and miscellaneous utility functions.
  • libavfilter provides a mean to alter decoded Audio and Video through chain of filters.
  • libavdevice provides an abstraction to access capture and playback devices.
  • libswresample implements audio mixing and resampling routines.
  • libswscale implements color conversion and scaling routines.

Tools

  • ffmpeg is a command line toolbox to manipulate, convert and stream multimedia content.
  • ffplay is a minimalistic multimedia player.
  • ffprobe is a simple analysis tool to inspect multimedia content.
  • Additional small tools such as aviocat, ismindex and qt-faststart.

Documentation

The offline documentation is available in the doc/ directory.

The online documentation is available in the main website and in the wiki.

Examples

Coding examples are available in the doc/examples directory.

License

FFmpeg codebase is mainly LGPL-licensed with optional components licensed under GPL. Please refer to the LICENSE file for detailed information.

Contributing

Patches should be submitted to the ffmpeg-devel mailing list using git format-patch or git send-email. Github pull requests should be avoided because they are not part of our review process and will be ignored.

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