The {br}/{abr} directives are not limited to post-encoding, they can
also be used pre-muxing. The already-present {packet} tag describes this
more accurately, so just drop the assertions.
They cannot store 1 as signed, only 0 and -1.
Avoids warnings such as:
implicit truncation from 'int' to a one-bit wide bit-field changes value from 1 to -1 [-Wsingle-bit-bitfield-constant-conversion]
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.
* the code is made shorter and simpler
* avoids constantly allocating and freeing AVPackets, thanks to
ThreadQueue integration with ObjPool
* is consistent with decoding/filtering/muxing
* reduces the diff in the future switch to thread-aware scheduling
This makes ifile_get_packet() always block. Any potential issues caused
by this will be resolved by the switch to thread-aware scheduling in
future commits.
Otherwise they'd be silently ignored if received by the filtering thread
before the filtergraph can be initialized, which would make the output
dependent on the order in which frames from different inputs arrive.
As previously for decoding, this is merely "scaffolding" for moving to a
fully threaded architecture and does not yet make filtering truly
parallel - the main thread will currently wait for the filtering thread
to finish its work before continuing. That will change in future commits
after encoders are also moved to threads and a thread-aware scheduler is
added.
Avoid making decisions based on current graph input state, which makes
the output dependent on the order in which the frames from different
inputs are interleaved.
Makes the output of fate-filter-overlay-dvdsub-2397 more correct - the
subtitle appears two frames later, which is closer to its PTS as stored
in the file.
Current code tracks min/max pts for each stream separately; then when
the file ends it combines them with last frame's duration to compute the
total duration of each stream; finally it selects the longest of those
durations as the file duration.
This is incorrect - the total file duration is the largest timestamp
difference between any frames, regardless of the stream.
Also change the way the last frame information is reported from decoders
to the muxer - previously it would be just the last frame's duration,
now the end timestamp is sent, which is simpler.
Changes the result of the fate-ffmpeg-streamloop-transcode-av test,
where the timestamps are shifted slightly forward. Note that the
matroska demuxer does not return the first audio packet after seeking
(due to buggy interaction betwen the generic code and the demuxer), so
there is a gap in audio.
This ensures that tq_receive() will always return EOF after all streams
were receive-finished, even though the sending side might not have
closed them yet. This may allow the receiver to avoid manually tracking
which streams it has already closed.
It is common for subtitle streams to have large gaps between packets.
When the caller is interleaving packets from multiple files, it can
easily happen that two successive subtitle packets trigger this limit,
even though no excessive buffering is happening.
Should fix#7064
By intent, and in practice, the "active contributor" criterion applies
to the person authoring the commits, not the one pushing them into the
repository.
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.
ifilter_send_eof() will fail if the input has no real or fallback
parameters, so there is no need to handle the case of some inputs being
in EOF state yet having no parameters.
In this case any timestamps are guessed by compute_pkt_fields() in
libavformat. Since we are decoding the stream, we have more accurate
information from the decoder and do not need any guesses.
Eliminates spurious PTS gaps in a number of FATE tests.
Also avoids dropping the majority of frames in fate-dirac*
* export AVCodecParserContext.picture_structure.
* when there are two field pictures in the packet, set
the interlacing parameters accordingly:
* repeat_pict=1 and picture_structure=FRAME to indicate 2 fields
* field_order to indicate the first field of the two
The demuxer does not set packet timestamps itself after
c6b6356635 and instead relies on the
parser to do it. However, this does not matter from the caller
perspective as it still happens inside the demuxer. The demuxer should
thus not be flagged as not having timestamps.
The parser does not have a timebase associated with it, so in general it
makes no sense for it to be exporting durations. Longer-term this
should be handled more cleanly with a new parser API.
It dates back to pre-2005 days, when people generally tended to commit
their work directly without going through the mailing list. Few
developers do it today, and never outside of their standalone modules.
This item is thus confusing and misleading and is better removed.
The section consistes of a single short paragraph linking to other
chapters. The enclosing chapter also has no other sections, all other
text is placed in the chapter directly.
Keeping a separate section for this paragraph just adds more clutter.
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().
It was added in cb114ed464 with the comment
"This will allow fixing several bugs with the -shortest option".
Since
* there is no explanation of what these bugs are
* libavformat is not the place to work around ffmpeg CLI bugs
* there is no indication that this feature is actually in use
deprecate it without replacement.
This is not an error condition, but would be treated like one if the
program terminates on the next transcode loop iteration because of a
signal or keyboard input.
Fixes#10504
Tested-by: https://github.com/0Ky
Anton Khirnov