| Commit message (Collapse) | Author | Age | Files | Lines |
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MM_PREAMBLE_SIZE is subtracted from buf_size almost immediately.
The original size is still in avpkt->size.
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Add "data" and "extradata" attributes to the schema.
Add "tag" element in "streams".
Print extradata before tags to avoid closing the element.
Fix trac ticket #1588.
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This ensures that the output buffers will have the channel layout set,
which is required by filters.
Also consistent with ffmpeg.c behavior.
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This is required otherwise it is not always possible to guess the number
of channels from the layout, for example if the channel layout is
unknown.
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There is no need to duplicate the input reference, since a filter should
not pass along a reference which is later modified. If this happens the
filter passing the reference should be fixed.
Also improve consistency with start_frame() of settb, allowing a pending
factorization.
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The documentation was erroneously removed in 7cf9aadd.
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Fixes Ticket1586
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Fixes Ticket1502
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:
avformat: Drop pointless "format" from container long names
swscale: bury one more piece of inline asm under HAVE_INLINE_ASM.
wv: K&R formatting cosmetics
configure: Add missing descriptions to help output
h264_ps: declare array of colorspace strings on its own line.
fate: amix: specify f32 sample format for comparison
tiny_psnr: support 32-bit float samples
eamad/eatgq/eatqi: call special EA IDCT directly
eamad: remove use of MpegEncContext
mpegvideo: remove unnecessary inclusions of faandct.h
af_asyncts: avoid overflow in out_size with large delta values
af_asyncts: add first_pts option
Conflicts:
configure
libavcodec/eamad.c
libavcodec/h264_ps.c
libavformat/crcenc.c
libavformat/ffmdec.c
libavformat/ffmenc.c
libavformat/framecrcenc.c
libavformat/md5enc.c
libavformat/nutdec.c
libavformat/rawenc.c
libavformat/yuv4mpeg.c
tests/tiny_psnr.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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These decoders use a special non-MPEG2 IDCT. Call it directly
instead of going through dsputil. There is never any reason
to use a regular IDCT with these decoders or to use the EA IDCT
with other codecs.
This also fixes the bizarre situation of eamad and eatqi decoding
incorrectly if eatgq is disabled.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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There is no sense in pulling in this monster struct just for
a handful of fields. The code does not call any functions
expecting an MpegEncContext.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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This allows for padding/trimming at the start of stream. By default, no
assumption is made about the first frame's expected pts, so no padding or
trimming is done.
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When start_frame has returned, the buffer reference's PTS
may have been modified by another filter.
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Based on code by: carl
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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broken by: cbb2c95e47cd83cb65b2ce77cbb0296e85fa506b
i had fixed that commit but pushed from an old branch
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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output.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Also bump minor version and add APIchanges entry.
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While here refactor code.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
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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:
flvdec: remove spurious use of stream id
lavf: deprecate r_frame_rate.
lavf: round estimated average fps to a "standard" fps.
Conflicts:
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/electronicarts.c
libavformat/flvdec.c
libavformat/rawdec.c
libavformat/utils.c
tests/ref/fate/iv8-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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We match streams by codec id now.
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According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
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avfilter_get_audio_buffer_ref_from_frame()
The new name is more meaningful in that context.
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Keeping the set options in the context when the component is still used
can be useful for debugging purposes.
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Avoid to fill the same picture again and again with the same content.
Optimize computation, and provides an example for the use of the
AV_PERM_REUSE permission flag.
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* commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23':
lavf: use dts difference instead of AVPacket.duration in find_stream_info()
avf: introduce nobuffer option
fate: make yadif tests consistent across systems
vf_hqdn3d: support 9 and 10bit colordepth
vf_hqdn3d: reduce intermediate precision
vf_hqdn3d: simplify and optimize
factor identical ff_inplace_start_frame out of two filters
vf_hqdn3d: cosmetics
avprobe/avconv: fix tentative declaration compile errors on MSVS.
Conflicts:
doc/APIchanges
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/options_table.h
libavformat/utils.c
libavformat/version.h
tests/fate/filter.mak
tests/ref/fate/filter-yadif-mode0
tests/ref/fate/filter-yadif-mode1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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AVPacket.duration is mostly made up and thus completely useless, this is
especially true for video streams.
Therefore use dts difference for framerate estimation and
the max_analyze_duration check.
The asyncts test now needs -analyzeduration, because the default is 5
seconds and the audio stream in the sample appears at ~10 seconds.
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Useful in cases where a significant analyzeduration is
still needed, while minimizing buffering before output.
An example is processing low-latency streams where all
media types won't necessarily come in if the
analyzeduration is small.
Additional changes by Josh Allmann <joshua.allmann@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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