Same issues apply to it as to -shortest.
Changes the results of the following tests:
- matroska-flac-extradata-update
The test reencodes two input FLAC streams into three output FLAC
streams. The last output stream is limited to 8 frames. The current
code results in the first two output streams having 12 frames, after
this commit all three streams have 8 frames and are the same length.
This new result is better, since it is predictable.
- mkv-1242
The test streamcopies one video and one audio stream, video is limited
to 11 frames. The new result shortens the audio stream so that it is
not longer than the video.
The -shortest option (which finishes the output file at the time the
shortest stream ends) is currently implemented by faking the -t option
when an output stream ends. This approach is fragile, since it depends
on the frames/packets being processed in a specific order. E.g. there
are currently some situations in which the output file length will
depend unpredictably on unrelated factors like encoder delay. More
importantly, the present work aiming at splitting various ffmpeg
components into different threads will make this approach completely
unworkable, since the frames/packets will arrive in effectively random
order.
This commit introduces a "sync queue", which is essentially a collection
of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
and are then released for further processing (encoding or muxing) when
it is ensured that the frame in question will not cause its stream to
get ahead of the other streams (the logic is similar to libavformat's
interleaving queue).
These sync queues are then used for encoding and/or muxing when the
-shortest option is specified.
A new option – -shortest_buf_duration – controls the maximum number of
queued packets, to avoid runaway memory usage.
This commit changes the results of the following tests:
- copy-shortest[12]: the last audio frame is now gone. This is
correct, since it actually outlasts the last video frame.
- shortest-sub: the video packets following the last subtitle packet are
now gone. This is also correct.
The muxing queue currently lives in OutputStream, which is a very large
struct storing the state for both encoding and muxing. The muxing queue
is only used by the code in ffmpeg_mux, so it makes sense to restrict it
to that file.
This makes the first step towards reducing the scope of OutputStream.
Figure out earlier whether the output stream/file should be bitexact and
store this information in a flag in OutputFile/OutputStream.
Stop accessing the muxer in set_encoder_id(), which will become
forbidden in future commits.
Move the file size checking code to ffmpeg_mux. Use the recently
introduced of_filesize(), making this code consistent with the size
shown by print_report().
Move header_written into it, which is not (and should not be) used by
any code outside of ffmpeg_mux.
In the future this context will contain more muxer-private state that
should not be visible to other code.
This is a per-file input option that adjusts an input's timestamps
with reference to another input, so that emitted packet timestamps
account for the difference between the start times of the two inputs.
Typical use case is to sync two or more live inputs such as from capture
devices. Both the target and reference input source timestamps should be
based on the same clock source.
If either input lacks starting timestamps, then no sync adjustment is made.
Provide a header based inline reimplementation of it.
Using av_fopen_utf8 doesn't work outside of the libraries when built
with MSVC as shared libraries (in the default configuration, where
each DLL gets a separate statically linked CRT).
Signed-off-by: Martin Storsjö <martin@martin.st>
The earlier code has ignored it for all stream types except
video and subtitles, probably because audio was presumed
to only consist of keyframes. Yet this assumption is not true
for e.g. TrueHD.
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Bitstream filters inserted between the input and output were never drained,
resulting in packets being lost if the bsf had any buffered.
Signed-off-by: James Almer <jamrial@gmail.com>
Do this by switching from the dynamic buffer API to the AVBPrint API;
the former has no defined way to check for errors.
This also avoids allocating an AVIOContext.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Do this by switching from the dynamic buffer API to the AVBPrint API;
the former has no defined way to check for errors.
This also avoids allocating an AVIOContext.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Anton Khirnov
Gyan Doshi
Nil Admirari
Martin Storsjö
James Almer
Andreas Rheinhardt