Remove now-obsolete code setting packet durations pre-muxing for CFR
encoded video.
Changes output in the following FATE tests:
* numerous adpcm tests
* ffmpeg-filter_complex_audio
* lavf-asf
* lavf-mkv
* lavf-mkv_attachment
* matroska-encoding-delay
All of these change due to the fact that the output duration is now
the actual input data duration and does not include padding added by
the encoder.
* apng-osample: less wrong packet durations are now passed to the muxer.
They are not entirely correct, because the first frame duration should
be 3 rather than 2. This is caused by the vsync code and should be
addressed later, but this change is a step in the right direction.
* tscc2-mov: last output frame has a duration of 11 rather than 1 - this
corresponds to the duration actually returned by the demuxer.
* film-cvid: video frame durations are now 2 rather than 1 - this
corresponds to durations actually returned by the demuxer and matches
the timestamps.
* mpeg2-ticket6677: durations of some video frames are now 2 rather than
1 - this matches the timestamps.
That field was added to store timestamp conversion state for audio
decoding code. Later it started being used by streamcopy as well, but
that use is wrong because, for a given input stream, both decoding and
an arbitrary number of streamcopies may be performed simultaneously.
They would then all overwrite the same state variable.
Store this state in MuxStream instead.
This is the last use of InputStream in of_streamcopy(), so the ist
parameter can now be removed.
It stores codec parameters of the stream submitted to the muxer, which
may be different from the codec parameters in AVStream due to bitstream
filtering.
This avoids the confusing back and forth synchronisation between the
encoder, bitstream filters, and the muxer, now information flows only in
one direction. It also reduces the need for non-muxing code to access
AVStream.
Reduces access to a deeply nested muxer property
OutputStream.st->codecpar->codec_type for this fundamental and immutable
stream property.
Besides making the code shorter, this will allow making the AVStream
(OutputStream.st) private to the muxer in the future.
Set InputStream.decoding_needed/discard/etc. only from
ist_{filter,output},add() functions. Reduces the knowledge of
InputStream internals in muxing/filtering code.
init_input_stream() can print log messages directly, there is no need to
ship them to the caller.
Also, log errors to the InputStream and avoid duplicate information in
the message.
Changing AVCodecContext.sample_aspect_ratio after the encoder was opened
is by itself questionable, but if anywhere it belongs in encoding rather
than filtering code.
Creating a new output stream of a given type is currently done by
calling new_<type>_stream(), which all start by calling
new_output_stream() to allocate the stream and do common init, followed
by type-specific init.
Reverse this structure - the caller now calls the common function
ost_add() with the type as a parameter, which then calls the
type-specific function internally. This will allow adding common code
that runs after type-specific code in future commits.
In most cases this should only occur once or so per stream in an
input, and in case the logic ends up in an eternal loop, it should
be visible to the end user instead of being completely invisible.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
When no packet dts values are available from the container, video
decoding code will currently use its own guessed values, which will then
be propagated to pkt_dts on decoded frames and used as pts in certain
cases. This is inaccurate, fragile, and unnecessarily convoluted.
Simply removing this allows the extrapolation code introduced in the
previous commit to do a better job.
Changes the results of numerous h264 and hevc FATE tests, where no
spurious timestamp gaps are generated anymore. Several tests no longer
need -vsync passthrough.
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
This field contains different values depending on whether the stream is
being decoded or not. When it is, InputStream.pts is set to the
timestamp of the last decoded frame. Otherwise, it is made equal to
InputStream.dts.
Since a given InputStream can be at the same time decoded and
streamcopied to any number of output streams, this use is incorrect, as
decoded frame timestamps can be delayed with respect to input packets by
an arbitrary amount (e.g. depending on the thread count when frame
threading is used).
Replace all uses of InputStream.pts for streamcopy with InputStream.dts,
which is its value when decoding is not performed. Stop setting
InputStream.pts for pure streamcopy.
Also, pass InputStream.dts as a parameter to do_streamcopy(), which
will allow that function to be decoupled from InputStream completely in
the future.
Which is subtitle encoding. Also, check for AVSubtitle.pts rather than
InputStream.pts, since that is the more authoritative value and is
guaranteed to be valid.
That function only contains two checks now - whether the muxer returned
EOF and whether the packet timestamp is before requested output start
time.
The first check is unnecessary, since the packet will just be rejected
by the muxer. The second check is better combined with a related check
directly in do_streamcopy().
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.
The channel layout is set before opening the encoder, in enc_open().
Messing with it in configure_output_audio_filter() cannot accomplish
anything meaningful.
This option adds a long string of numbers to the progress line, where
i-th number contains the base-2 logarithm of the number of times a frame
with this QP value was seen by print_report().
There are multiple problems with this feature:
* despite this existing since 2005, web search shows no indication
that it was ever useful for any meaningful purpose;
* the format of what is printed is entirely undocumented, one has to
find it out from the source code;
* QP values above 31 are silently ignored;
* it only works with one video stream;
* as it relies on global state, it is in conflict with ongoing
architectural changes.
It then seems that the nontrivial cost of maintaining this option is not
worth its negligible (or possibly negative - since it pollutes the
already large option space) value.
Users who really need similar functionality can also implement it
themselves using -vstats.
