| Commit message (Collapse) | Author | Age | Files | Lines |
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* qatar/master: (22 commits)
als: prevent infinite loop in zero_remaining().
cook: prevent div-by-zero if channels is zero.
pamenc: switch to encode2().
svq1enc: switch to encode2().
dvenc: switch to encode2().
dpxenc: switch to encode2().
pngenc: switch to encode2().
v210enc: switch to encode2().
xwdenc: switch to encode2().
ttadec: use branchless unsigned-to-signed unfolding
avcodec: add a Sun Rasterfile encoder
sunrast: Move common defines to a new header file.
cdxl: fix video decoding for some files
cdxl: fix audio for some samples
apetag: add proper support for binary tags
ttadec: remove dead code
swscale: make access to filter data conditional on filter type.
swscale: update context offsets after removal of AlpMmxFilter.
prores: initialise encoder and decoder parts only when needed
swscale: make monowhite/black RGB-independent.
...
Conflicts:
Changelog
libavcodec/alsdec.c
libavcodec/dpxenc.c
libavcodec/golomb.h
libavcodec/pamenc.c
libavcodec/pngenc.c
libavformat/img2.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
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Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Width is padded for ham encodings too.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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There may be extra padding at and of chunk.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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export as attachment streams
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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The unused code being removed is for encoding only and therefore is not needed
by the decoder.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Prevents crashes on 1-tap filter (unscaled). Also rename "bguf" argument
to "vbuf", seems that was a typo.
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Disadvantage is that it no longer allows modifying brightness through
adjustment of the RGB lookup table. Advantage is that now monowhite/black
no longer need to be identified as a RGB format.
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
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WMApro actually support 13-bits block sizes (potentially even up to 14),
and thus we should support that also. If we get block sizes beyond what
the decoder can handle (14 is possible depending on s->decode_flags),
error out instead of crashing.
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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This allows handling matroska files with errors.
Fixes test4.mkv and test7.mkv from the official Matroska test suite.
These are also trac issues #544 and #545.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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For some reason this always happens with -f ismv.
See trac issue #996.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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Muxing pcm audio in MOV using avcodec_encode_audio() was failing
because avcodec_encode_audio() returns an incorrect packet size of 4
bytes. This can be reproduced by modifying the sample
ffmpeg/doc/examples/muxing.c to encode PCM, see ML patch
muxing-test.diff
I git bisected and commit 89ddff92a385 is the one that broke this. In
mov_write_header() if st->codec->frame_size <= 1 it sets it to 1. Then
avcodec_encode_audio() sets frame->nb_samples = avctx->frame_size, and
frame->nb_samples of 1 is used to compute a packet size of 4 bytes.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
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* qatar/master:
shorten: Use separate pointers for the allocated memory for decoded samples.
atrac3: Fix crash in tonal component decoding.
ws_snd1: Fix wrong samples counts.
movenc: Don't set a default sample duration when creating ismv
rtp: Factorize the check for distinguishing RTCP packets from RTP
golomb: avoid infinite loop on all-zero input (or end of buffer).
bethsoftvid: synchronize video timestamps with audio sample rate
bethsoftvid: add audio stream only after getting the first audio packet
bethsoftvid: Set video packet duration instead of accumulating pts.
bethsoftvid: set packet key frame flag for audio and I-frame video packets.
bethsoftvid: fix read_packet() return codes.
bethsoftvid: pass palette in side data instead of in a separate packet.
sdp: Ignore RTCP packets when autodetecting RTP streams
proresenc: initialise 'sign' variable
mpegaudio: replace memcpy by SIMD code
vc1: prevent using last_frame as a reference for I/P first frame.
Conflicts:
libavcodec/atrac3.c
libavcodec/golomb.h
libavcodec/shorten.c
libavcodec/ws-snd1.c
tests/ref/fate/bethsoft-vid
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Fixes invalid free() if any of the buffers are not allocated due to either
not decoding a header or an error prior to allocating all buffers.
Fixes CVE-2012-0858
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Add a check to avoid writing past the end of the channel_unit.components[]
array.
Bug Found by: cosminamironesei
Fixes CVE-2012-0853
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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This makes the check that avoids overwrite of the samples array actually
work properly.
fixes CVE-2012-0848
CC: libav-stable@libav.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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The binary doesn't change after this patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
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According to unofficial documentation, the video rate is locked to the audio
sample rate. This results in proper synchronization of audio and video
timestamps from the demuxer. This only works if the first audio packet occurs
before the first video packet or the audio sample rate is the default rate of
11111 Hz, both of which are true for all samples in our archive.
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This avoids initializing a stream with dummy values or when the file does not
contain audio.
Also set duration for audio packets, using the sample rate as the time base.
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Fixes avconv video stream copy of bethsoft video, which was skipping all
video frames unless the copyinkf option was used.
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Use proper AVERROR codes, and return 0 for no error.
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Update FATE reference to account for now non-existent palette packet.
This also fixes the FATE test if frame data is not initialized in
get_buffer(), so update comment in avconv accordingly.
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The rtp demuxer which listens for RTP packets and detects the
RTP payload type will currently get confused if the first packet
received is an RTCP packet. Thus ignore such packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
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