| Commit message (Collapse) | Author | Age | Files | Lines |
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FATE reference updated due timestamp rounding because of resampling from
44100 Hz to 16000 Hz in avconv.
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Update FATE references due to encoder delay.
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This simplifies matching of timestamps between input frames and output
packets.
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Signed-off-by: Michael Niedermayer <[email protected]>
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This parly reverts 85469f.
Fixes ticket #1091.
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This is even potentially faster in this use-case.
Should fix AAC SBR decoding on machines with SSE but not
SSE2, fixing track issue #1041.
Signed-off-by: Reimar Döffinger <[email protected]>
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Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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This is how it is defined in Amiga Developer CD from year 1992 and
is consistent with files created with ADPro.
Signed-off-by: Paul B Mahol <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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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 <[email protected]>
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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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This fixes decoding of Bink files with non-multiple-of-16 width.
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Documentation for those will be generated automatically.
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This will allow us to automatically generate manpages for them.
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This change avoids accessing the segment map of the previous frame if
segmentation is not enabled for the current frame. The caller of
decode_mb_mode() only calls ff_thread_await_progress() on the reference
segmentation index array if segmentation is enabled, so Chromium's TSAN
will report a race when accessing this data while segmentation is not
enabled.
Signed-off-by: Ronald S. Bultje <[email protected]>
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Fixes Ticket1109
Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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Signed-off-by: Michael Niedermayer <[email protected]>
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ISC doesn't contain this line, so remove it to
prevent confusion.
Signed-off-by: Derek Buitenhuis <[email protected]>
Signed-off-by: Michael Niedermayer <[email protected]>
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Fixes crash
Found-by: durandal_1707
Signed-off-by: Michael Niedermayer <[email protected]>
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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 <[email protected]>
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An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <[email protected]>
Signed-off-by: Kostya Shishkov <[email protected]>
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Changing flush type from Z_FINISH is needed since encoder compresses fixed
amount of data and doesn't care about writing end of stream marker.
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