Do not pass an options dictionary to the avcodec_open2() in enc_open().
This is cleaner and more robust, as previously various bits of code
would try to interpret the contents of the options dictionary, with
varying degrees of correctness. Now they can just access the encoder
AVCodecContext directly.
Cf. 372c78dd42 - analogous change for
decoding.
A non-progressive field order is now written on the container level in
interlaced ProRes encoding tests.
Found-while-revieweing: CID1520670 Dereference after null check
Sponsored-by: Sovereign Tech Fund
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There is no need to free the already-added items, they will be freed
alongside the codec context. There is also little point in an error
message, as the only reason this can fail is malloc failure.
This allows using WRAPPED_AVFRAME encoders with loopback decoders in
order to connect multiple filtergraphs together.
Clear the flag in muxers, since lavf does not need it for anything and
it would change the results of framecrc FATE tests.
Encoder timebase is equal to the frame timebase, so does not need to be
passed separately.
Also, rename in_picture to frame, which is shorter and more accurate -
it always contains a frame, never a field.
These functions used to be passed directly to pthread_create(), which
required them to return void*. This is no longer the case, so they can
return a plain int.
Change the main loop and every component (demuxers, decoders, filters,
encoders, muxers) to use the previously added transcode scheduler. Every
instance of every such component was already running in a separate
thread, but now they can actually run in parallel.
Changes the results of ffmpeg-fix_sub_duration_heartbeat - tested by
JEEB to be more correct and deterministic.
See the comment block at the top of fftools/ffmpeg_sched.h for more
details on what this scheduler is for.
This commit adds the scheduling code itself, along with minimal
integration with the rest of the program:
* allocating and freeing the scheduler
* passing it throughout the call stack in order to register the
individual components (demuxers/decoders/filtergraphs/encoders/muxers)
with the scheduler
The scheduler is not actually used as of this commit, so it should not
result in any change in behavior. That will change in future commits.
As for the analogous decoding change, this is only a preparatory step to
a fully threaded architecture and does not yet make encoding truly
parallel. The main thread will currently submit a frame and wait until
it has been fully processed by the encoder before moving on. That will
change in future commits after filters are moved to threads and a
thread-aware scheduler is added.
This code suffers from a known issue - if an encoder with a sync queue
receives EOF it will terminate after processing everything it currently
has, even though the sync queue might still be triggered by other
threads. That will be fixed in following commits.
Its function is analogous to that of the fps filter, so filtering is a
more appropriate place for this.
The main practical reason for this move is that it places the encoding
sync queue right at the boundary between filters and encoders. This will
be important when switching to threaded scheduling, as the sync queue
involves multiple streams and will thus need to do nontrivial
inter-thread synchronization.
In addition to framerate conversion, the closely-related
* encoder timebase selection
* applying the start_time offset
are also moved to filtering.
That field is used by the framerate code to track whether any output has
been generated for the last input frame(*). Its use in the last
invocation of print_report() is meant to account for the very last
filtered frame being dropped in the number of dropped frames printed in
the log. However, that is a highly inappropriate place to do so, as it
makes assumptions about vsync logic in completely unrelated code. Move
the increment to encoder flush instead.
(*) the name is misleading, as the input frame has not yet been dropped
and may still be output in the future
Always use the functionality of the latter, which makes more sense as it
avoids losing keyframes due to vsync code dropping frames.
Deprecate the 'source_no_drop' value, as it is now redundant.
Unlike the 'source' mode, which preserves source keyframe-marking as-is,
the 'source_no_drop' mode attempts to keep track of keyframes dropped by
framerate conversion and mark the next output frame as key in such
cases. However,
* c75be06148 broke this functionality entirely, and made it equivalent
to 'source'
* even before it would only work when the frame immediately following
the dropped keyframe is preserved and not dropped as well
Also, drop a redundant check for 'frame' in setting dropped_keyframe, as
it is redundant with the check on the above line.
It is badly named (should have been -top_field_first, or at least -tff),
underdocumented and underspecified, and (most importantly) entirely
redundant with the setfield filter.
Merge three blocks with slightly inconsistent checks into one, treating
encoder input as interlaced when either:
* at least one of ilme/ildct flags is set
* the first frame in the stream is marked as interlaced
* the user specified the -top option
Stop modifying the frame passed to enc_open().
When no output video framerate is specified by the user with -r or can
be inferred from the filtergraph, encoder setup will arbitrarily decide
that the framerate is 25fps. However, making up any framerate value for
VFR encoding is at best unnecessary.
Changes the results of the sub2video tests, where the input timebase is
now used instead of 1/25.
Mainly this fixes handling special values of -enc_time_base ('demux' or
'filter') for audio. It also prints a warning if -enc_time_base is
specified for subtitles, instead of ignoring it silently (current
subtitle encoding API only works with AV_TIME_BASE_Q).
This function converts packet timestamps from the input stream timebase
to OutputStream.mux_timebase, which may or may not be equal to the
actual output AVStream timebase (and even when it is, this may not
always be the optimal choice due to bitstream filtering).
Just keep the timestamps in input stream timebase, they will be rescaled
as needed before bitstream filtering and/or sending the packet to the
muxer.
Move the av_rescale_delta() call for audio (needed to preserve accuracy
with coarse demuxer timebases) to write_packet.
Drop now-unused OutputStream.mux_timebase.
