| Commit message (Collapse) | Author | Age |
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Fixes FPE
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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This prevents a null ptr dereference.
It could be checked differently but this way it should
be possible to return some data.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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* qatar/master:
xwma: Validate channels and bits_per_coded_sample.
mov: Do not read past the end of the ctts_data table.
mov: Add missing terminator to mov_ch_layout_map_1ch.
asf: reset side data elements on packet copy.
wmavoice: fix stack overread.
wmalossless: error out if a subframe is not used by any channel.
vqa: check palette chunk size before reading data.
wmalossless: reset sample pointer for each subframe.
wmalossless: error out on invalid values for order.
Conflicts:
libavcodec/vqavideo.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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This prevents a SIGFPE later on.
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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Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: Libav-stable@libav.org
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Prevents crash (double free) when free()ing the original packet.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
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* qatar/master:
FATE: Add ZeroCodec test
oggparseogm: fix order of arguments of avpriv_set_pts_info().
pngenc: better upper bound for encoded frame size.
aiffdec: set block_duration to 1 for PCM codecs that are supported in AIFF-C
aiffdec: factor out handling of integer PCM for AIFF-C and plain AIFF
aiffdec: use av_get_audio_frame_duration() to set block_duration for AIFF-C
aiffdec: do not set bit rate if block duration is unknown
wmall: output packet only if we have decoded some samples
Conflicts:
libavcodec/pngenc.c
tests/fate/lossless-video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Anton Khirnov <anton@khirnov.net>
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CC: libav-stable@libav.org
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The comment seemed to indicate that left was used
to limit the size of the recv, but it was not so.
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It allows to exit the thread immediately instead of waiting
for select to timeout (one second in the current code).
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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>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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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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Version from vqa header does not dictate which sound chunks may
appear in file.
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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* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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There is no reason for it to be public, it's only meant to be used
internally.
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The caller can achieve the same effect (i.e. getting raw unparsed/mangled
packets) with av_read_frame() and AVFMT_FLAG_NOPARSE |
AVFMT_FLAG_NOFILLIN
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Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.
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Also fixes stream duration calculation.
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The values in the header refer to decoded data, not compressed data.
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Do not output an extra packet when out_size is reached.
Also return AVERROR_EOF instead of AVERROR(EIO).
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The packet duration is always 28 samples.
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The other fragmentation options (frag_duration, frag_size and
frag_keyframe) are combined with OR, cutting fragments at the
first of the conditions being fulfilled.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This enables reordering of UDP packets by default, unless the caller
explicitly sets -max_delay 0.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This will allow us to automatically generate manpages for them.
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This allows the caller to get them without special code even after
seeking before receiving any data.
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This fixes ticket #1099.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Also add generated key frame in the enum, and doxycomment the existing
ones. Descriptions are directly taken from the public specifications.
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* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
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The private option has not been part of any release yet (and
it is only of use in quite rare cases), so just remove it instead
of keeping it with deprecation warnings.
Signed-off-by: Martin Storsjö <martin@martin.st>
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