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 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.
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.
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*
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.
It now contains data from multiple sources, so group those items that
always come from the decoder. Also, initialize them to invalid values,
so that frames that did not originate from a decoder can be
distinguished.
This is only a preparatory step to a fully threaded architecture and
does not yet make decoding truly parallel - the main thread will
currently submit a packet and wait until it has been fully processed by
the decoding thread before moving on. Decoder behavior as observed by
the rest of the program should remain unchanged. That will change in
future commits after encoders and filters are moved to threads and a
thread-aware scheduler is added.
It is only used for flushing the subtitle decoder, so allocate a
dedicated packet for that.
Keep Decoder.pkt unused for now, it will be repurposed in future
commits.
Make all relevant state per-filtergraph input, rather than per-input
stream. Refactor the code to make it work and avoid leaking memory when
a single subtitle stream is sent to multiple filters.
This code is a sub2video analogue of ifilter_send_frame(), so it
properly belongs to the filtering code.
Note that using sub2video with more than one target for a given input
subtitle stream is currently broken and this commit does not change
that. It will be addressed in following commits.
This function is entangled with decoder setup, so it is more decoding
code rather than ffmpeg_hw code. This will allow making more decoder
state private in the future.
As the comment in the code mentions, EAGAIN is not an expected value here
because we call avcodec_receive_frame() until all frames have been returned.
avcodec_send_packet() returning EAGAIN means a packet is still buffered, which
hints that the underlying decoder is buggy and not fetching packets as it
should.
An example of this behavior was in the libdav1d wrapper before f209614290,
where feeding it split frames (or individual OBUs) would result in the CLI
eventually printing the confusing "Error submitting packet to decoder: Resource
temporarily unavailable" error message, and just keep going until EOF without
returning new frames.
Signed-off-by: James Almer <jamrial@gmail.com>
It currently emulates the long-removed
avcodec_decode_audio4/avcodec_decode_video2 APIs, which obfuscates the
actual decoding flow. Restructure the decoding calls so that they
naturally follow the new avcodec_send_packet()/avcodec_receive_frame()
design.
This is not only significantly easier to read, but also shorter.
It is currently handled in the same loop as audio and video, but this
obscures the actual flow, because only one iteration is ever performed
for subtitles.
Also, avoid a pointless packet reference.
Anton Khirnov
James Almer