| Commit message (Collapse) | Author | Age |
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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It is now possible to use lavfi sources.
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The new name is shorter and more accurate, since this variable is no
longer used only for checking recording time constraint.
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Operate with a pointer to InputFile instead of its index in input_files.
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Replace for (;foo == 0;) with while (!foo)
This is prettier.
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This simplifies splitting code for reading from input out of
transcode().
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Before this commit, poll_filters() reads all frames available on each
lavfi output. This does not work for lavfi sources that produce
an unlimited number of frames, e.g. color and similar.
With this commit, poll_filters() reads from output with the lowest
timestamp and returns to wait for more input if no frames are available
on it.
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This allows simplifying a few expressions.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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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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Also exit in such a case if -xerror is used.
Fixes bug 329.
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start time is already substracted from the frame timestamp, so it needs
to be checked against 0, not start time.
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Currently it always exits with an error when more than
one position is specified.
CC: libav-stable@libav.org
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If the output frame size is smaller than the input sample rate,
and the input stream time base corresponds exactly to the input
frame size (getting input packet timestamps like 0, 1, 2, 3, 4 etc),
the output timestamps from the filter will be like
0, 1, 2, 3, 4, 4, 5 ..., leadning to non-monotone timestamps later.
A concrete example is input mp3 data having frame sizes of 1152
samples, transcoded to aac with 1024 sample frames.
By setting the audio filter time base to the sample rate, we will
get sensible timestamps for all output packets, regardless of
the ratio between the input and output frame sizes.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This allows passing the right options to encoders when there's more
than one encoder for a certain codec id.
Signed-off-by: Martin Storsjö <martin@martin.st>
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The warning silenced was:
avconv.c: In function ‘opt_output_file’:
avconv.c:3380:21: warning: ‘meta_out’ may be used uninitialized in this function [-Wuninitialized]
avconv.c:3315:20: note: ‘meta_out’ was declared here
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The warning silenced was:
avconv.c: In function ‘configure_filtergraph’:
avconv.c:603:8: warning: ‘ist’ may be used uninitialized in this function [-Wuninitialized]
avconv.c:549:18: note: ‘ist’ was declared here
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parse_forced_keyframes() relies in encoder timebase being set, so call
it from transcode_init() after it is known.
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Also ensure that all such filters get a non-NULL name. Should fix FATE
failures on some architectures after
58b049f2fa4f192b00baadb5f1f32ca366f936ea.
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Because of a mistake during merging the code for simple and complex
filtergraphs, -async inserts an asyncts filter both on input and output.
Remove the output hunk.
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This reduces the dependency on unistd.h which is not available
on all systems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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It will allow adding new fields to it without ABI breaks.
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Our w32pthreads wrapper has various issues and is only supposed to be
used in libavcodec.
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When there are multiple input files, run demuxing for each input file in
a separate thread, so reading packets does not block.
This is useful for achieving low latency when reading from multiple
(possibly slow) input streams.
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It will be used by avplay.
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Use just the pointer to the head of the buffer pool.
This will allow sharing the code with avplay.
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frame number should never be strictly larger than max frames, so the
if() was never triggered.
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filters
There may be multiple frames returned, so with just one check we can
write more than requested to the output.
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This is done by automatically inserting a setpts filter.
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Some tests change because -s now inserts the scaler to the end instead
of beginning of the filtergraph.
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Signed-off-by: Martin Storsjö <martin@martin.st>
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