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.
* 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.
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.
An AVFormatContext leaks on errors that happen before it is attached
to its permanent place (an InputFile). Fix this by attaching
it earlier.
Given that it is not documented that avformat_close_input() is usable
with an AVFormatContext that has only been allocated with
avformat_alloc_context() and not opened with avformat_open_input(),
one error path before avformat_open_input() had to be treated
specially: It uses avformat_free_context().
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The av_opt_eval family of functions emits errors messages on error
and can therefore not be used with fake objects when the AVClass
has a custom item_name callback. The AVClass for AVCodecContext
has such a custom callback (it searches whether an AVCodec is set
to use its name). In practice it means that whatever is directly
after the "cc" pointer to the AVClass for AVCodec in the stack frame
of ist_add() will be treated as a pointer to an AVCodec with
unpredictable consequences.
Fix this by using an actual AVCodecContext instead of a fake object.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is no longer needed as the side data is available for decoders in the
AVCodecContext.
The tests affected reflect the removal of useless CPB and Stereo 3D side
data in packets.
Signed-off-by: James Almer <jamrial@gmail.com>
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.
Make the function process just one input stream at a time and save an
indentation level. Also rename it to ist_add() to be consistent with an
analogous function in ffmpeg_mux_init.
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.
Set them in ifilter_parameters_from_dec(), similarly to audio/video
streams. This reduces the extent to which sub2video filters need to be
treated specially.
Export the corresponding flag in InputFile instead. This will allow
making the demuxer AVFormatContext private in future commits, similarly
to what was previously done for muxers.
There is no point in having a per-stream wallclock start time, since
they are all computed at the same instant. Keep a per-file start time
instead, initialized when the demuxer thread starts.
That is a more appropriate place for this code and will allow hiding
more of InputStream.
The value of repeat_pict extracted from libavformat internal parser no
longer needs to be trasmitted outside of the demuxing thread.
Move readrate handling to the demuxer thread. This has to be done in the
same commit, since it reads InputStream.dts,nb_packets, which are now
set in the demuxer thread.
This way computing it and using it for streamcopy does not need to
happen in sync. Will be useful in following commits, where updating
InputStream.dts will be moved to the demuxing thread.
When an input stream terminates and no frames were successfully decoded,
filtering code will currently configure the filtergraph using demuxer
stream parameters. Use decoder parameters instead, which should be more
reliable. Also, initialize them immediately when an input stream is
bound to a filtergraph input, so that these parameters are always
available (if at all) and filtering code does not need to reach into the
decoder at some arbitrary later point.
Stop using InputStream.dts for generating missing timestamps for decoded
frames, because it contains pre-decoding timestamps and there may be
arbitrary amount of delay between input packets and output frames (e.g.
dependent on the thread count when frame threading is used). It is also
in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary
rounding issues.
New code maintains a timebase that is the inverse of the LCM of all the
samplerates seen so far, and thus can accurately represent every audio
sample. This timebase is used to generate missing timestamps after
decoding.
Changes the result of the following FATE tests
* pcm_dvd-16-5.1-96000
* lavf-smjpeg
* adpcm-ima-smjpeg
In all of these the timestamps now better correspond to actual frame
durations.
Set InputStream.decoding_needed/discard/etc. only from
ist_{filter,output},add() functions. Reduces the knowledge of
InputStream internals in muxing/filtering code.
When no timestamps are available from the container, the video decoding
code will currently use fake dts values - generated in
process_input_packet() based on a combination of information from the
decoder and the parser (obtained via the demuxer) - to generate
timestamps during decoder flushing. This is fragile, hard to follow, and
unnecessarily convoluted, since more reliable information can be
obtained directly from post-decoding values.
The new code keeps track of the last decoded frame pts and estimates its
duration based on a number of heuristics. Timestamps generated when both
pts and pkt_dts are missing are then simple pts+duration of the last frame.
The heuristics are somewhat complicated by the fact that lavf insists on
making up packet timestamps based on its highly incomplete information.
That should be removed in the future, allowing to further simplify this
code.
The results of the following tests change:
* h264-3386 now requires -fps_mode passthrough to avoid dropping frames
at the end; this is a pathology of the interaction of the new and old
code, and the fact that the sample switches from field to frame coding
in the last packet, and will be fixed in following commits
* hevc-conformance-DELTAQP_A_BRCM_4 stops inventing an arbitrary
timestamp gap at the end
* hevc-small422chroma - the single frame output by this test now has a
timestamp of 0, rather than an arbitrary 7
Currently, output streams where an input stream is sent directly (i.e.
not through lavfi) are determined by iterating over ALL the output
streams and skipping the irrelevant ones. This is awkward and
inefficient.
Replace it with an array of streams in each InputFile. This is a more
accurate reflection of the actual relationship between InputStream and
InputFile.
Analogous to what was previously done to output streams in
7ef7a22251.
The current adjustment of input start times just adjusts the tsoffset.
And it does so, by resetting the tsoffset to nullify the new start time.
This leads to breakage of -copyts, ignoring of input_ts_offset, breaking
of -isync as well as breaking wrap correction.
Fixed by taking cognizance of these parameters, and by correcting start times
just before sync offsets are applied.
Anton Khirnov