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,
* c75be061487 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.
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>
Also, avoid spurious end-of-line after side data entries, and improve
rendering of compact output, by adding an indication of the side data
type for each entry.
Also fixes issue:
http://trac.ffmpeg.org/ticket/9266
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.
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().
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
These defines are also used in other contexts than just AVCodecContext
ones, e.g. in libavformat. Furthermore, given that these defines are
public, the AV-prefix is the right one, so deprecate (and not just move)
the FF-macros.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The word "monotonous" means "spoken in a monotone" which is not what we
mean here. We mean "monotonic" i.e. nondecreasing.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
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.
Bitstream filtering input timebase is not always necessarily equal to
OutputStream.mux_timebase. Also, set AVPacket.time_base correctly for
packets output by bitstream filters
Do not rescale at all in of_output_packet() when not doing bitstream
filtering, as it's unnecessary - write_packet() will rescale to the
actual muxer timebase.
This is a bit cleaner as int need not be the underlying type
of an enum if a smaller type can hold all its values.
Also declare the children_ids array as const as it never changes.
Reviewed-by: Stefano Sabatini <stefasab@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes Coverity issue #1524491.
Regression since e6126abc6997058ca49ee596b70611bbe367163e.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
