Unfortunately the output buffer size check assumes that the
input buffer is never over-consumed, thus this actually
also allowed to write outside the output buffer if "lucky".
Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
CONFIG_VDPAU is the condition on which ff_vdpau_mpeg_picture_complete
is compiled in, so it is more appropriate, particularly since the
separate VDPAU decoder should be removed in the longer term.
Since the code already supports both little- and big-endian
audio for recording, do not fail just because the endianness is not
what we expect.
It is possible that 8-bit recording would not have worked at all on
some systems without that change.
This allows the AC-3 decoder to be used directly with RealMedia
decoders that unlike the libavformat one do not byte-swap automatically.
Since the new code is only used in case we would fail directly otherwise
there should be no risk for regressions.
This allows the AC-3 decoder to be used directly with RealMedia
decoders that unlike the libavformat one do not byte-swap automatically.
Since the new code is only used in case we would fail directly otherwise
there should be no risk for regressions.
The "buf" pointer needs to be overwritten since otherwise the CRC check fails.
Instead of returning an error when bytes are left over, just return
the number of actually used bytes as other decoders do.
Instead add a special case so an error will be returned when none
of the data looks valid to avoid making debugging a pain.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 4a72765a1c)
check AVCodecContext->sample_fmt against AVCodec->sample_fmts[] to ensure
that the encoder supports the specified sample format. Error out if it doesn't.
Previously, it would continue and output garbage. Fixes issue 2587.
(cherry picked from commit 2cfa2d9258)
Instead of returning an error when bytes are left over, just return
the number of actually used bytes as other decoders do.
Instead add a special case so an error will be returned when none
of the data looks valid to avoid making debugging a pain.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
check AVCodecContext->sample_fmt against AVCodec->sample_fmts[] to ensure
that the encoder supports the specified sample format. Error out if it doesn't.
Previously, it would continue and output garbage. Fixes issue 2587.
Currently (since the data_offset fix) the ogg demuxer assumes that
after the first non-header packets in any stream no more header packets
will follow.
This is not guaranteed, so change the code back again to wait until it
has finished the headers for all streams before returning from ogg_get_headers.
This fixes issue 2428.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6bd69e6ada)
This validate the length of a mkv element directly after reading
it.
This has the advantage that it is easy to add new limits and makes
it less likely to forget to add checks and also avoids issues like
bits of the length value above the first 32 being ignored because
the parsing functions only takes an int.
Previously discussed in the "mkv 0-byte integer parsing" thread.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 95ec3d4cac)
Currently (since the data_offset fix) the ogg demuxer assumes that
after the first non-header packets in any stream no more header packets
will follow.
This is not guaranteed, so change the code back again to wait until it
has finished the headers for all streams before returning from ogg_get_headers.
This fixes issue 2428.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This validate the length of a mkv element directly after reading
it.
This has the advantage that it is easy to add new limits and makes
it less likely to forget to add checks and also avoids issues like
bits of the length value above the first 32 being ignored because
the parsing functions only takes an int.
Previously discussed in the "mkv 0-byte integer parsing" thread.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The first part of the metadata, the "vendor" string, is required by
libvorbis, it will refuse to play when it is not available.
Also we do not currently parse that part into metadata so it would also
be lost if we removed it as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8cb3c557a9)
The first part of the metadata, the "vendor" string, is required by
libvorbis, it will refuse to play when it is not available.
Also we do not currently parse that part into metadata so it would also
be lost if we removed it as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
VC1 test container always uses time-base 1 ms, so we must convert
from whatever time-base the application gave us to that, otherwise
the video will play at ridiculous speeds.
It would be possible to signal that a container supports only one
time-base and have code in a layer above do the conversion, but
for a single format this seems over-engineered.
color value instead of always taking 0 (resulting in green frames).
Fixes issue issue2531.
Originally committed as revision 26363 to svn://svn.ffmpeg.org/ffmpeg/trunk
of returning packets with uninitialized data.
Returning partial packets as for other demuxers is problematice due to
packet scrambling and thus is not done.
Originally committed as revision 25931 to svn://svn.ffmpeg.org/ffmpeg/trunk
ID3v2 tags which no longer works since ID3v2 handling was moved to
generic code.
In addition, in caused false-positives for all files starting with
one or more 0-bytes.
Originally committed as revision 25929 to svn://svn.ffmpeg.org/ffmpeg/trunk
PaletteControl.
This also fixes playback of some files with ffplay (images were
corrupted for a short time after a palette change).
Originally committed as revision 25778 to svn://svn.ffmpeg.org/ffmpeg/trunk
the object number is, it determines whether we should continue
parsing the presentation description and whether we should
clear the subtitles on the next display command.
Based on patch by Mark Goodman [mark goodman gmail com]
Originally committed as revision 25682 to svn://svn.ffmpeg.org/ffmpeg/trunk
Contrary to progressive, just being able to crop up to 14/15 pixels
is not enough to encode all supported resolutions, and the new
behaviour is also consistent with e.g. MPEG-2 etc.
Originally committed as revision 25669 to svn://svn.ffmpeg.org/ffmpeg/trunk
but worse it did not set up destination dimensions, thus every user
of it would necessarily fail.
Originally committed as revision 32424 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
and add a test for regular GSM as fate-gsm.
Fixes a 8kHz sample from issue 113.
Originally committed as revision 25313 to svn://svn.ffmpeg.org/ffmpeg/trunk
bitstream endianness, so they cannot reuse the same code
without a lot of function call overhead.
Thus use templating to compile two different binaries.
Originally committed as revision 25312 to svn://svn.ffmpeg.org/ffmpeg/trunk
between ffmpeg and ffplay and avoids a valgrind error by freeing
avformat_opts->key.
Originally committed as revision 25309 to svn://svn.ffmpeg.org/ffmpeg/trunk
have been accepted before).
Also do not fail if they are invalid but instead override them to 0.
This allows decoding e.g. MPEG video when only the container values are corrupted.
For encoding a value of 0,0 of course makes no sense, but was allowed
through before and will be caught by an extra check in the encode function.
Originally committed as revision 25124 to svn://svn.ffmpeg.org/ffmpeg/trunk
case the stride must be aligned to a multiple of 4.
The original CSCD encoder just compresses bitmaps it gets via Windows
API functions as-is, thus it uses exactly those alignment rules.
Originally committed as revision 25096 to svn://svn.ffmpeg.org/ffmpeg/trunk
