Normally we discard things prior to the intended start
for stream copy this is not always possible, and its not done by default
this option allows discarding to be enabled
this is primarely usefull when transcoding a video and stream copying an
audio stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Eliminating finished streams avoids the progress stopping
with the first stream.
Using the max instead of the min avoids the progress stopping
with gaps in sparse streams (subtitles).
Negligible change for normal circumstances.
Rework the transcode() function and its immediate annexes to have
the same structure as in avconv, while still maintaining proper
scheduling to avoid accumulation.
Using -filter_complex without inputs now works.
Fixes Ticket1627
The fate change is due to ffmpeg no longer pushing audio timestamps
aggressively up (which is what caused the AV sync issues in the ticket)
but leaving them as they are.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The old code generates a termination packet with the same regions as the
start packet and page_state set to "only what changed"; the result is
that the termination packet is decoded as identical to the start packet.
The new code does as found in some DVB broadcasts: produce a packet with
no regions. This is done by expecting num_rects to be 0 rather than
using a flip-flop. ffmpeg.c is updated accordingly.
We're now running some of this code through valgrind for the first
time, and a few warnings showed up stemming from two problems.
1) The ASS code assumes the subtitle header is null terminated, but
it wasn't, and passing the size down doesn't look like fun, so I
added a terminator
2) The code wasn't freeing all of its state.
Signed-off-by: Philip Langdale <philipl@overt.org>
With this feature, it becomes possible to perform commonly
requested tasks, such as hardcoding bitmap subtitles.
This will be reverted once libavfilter has proper support
for subtitles. All the changes have the string "sub2video"
in them, it makes it easy to spot the parts.
This patch fixes two things:
- in case of subtitles, check_recording_time() is comparing the current
PTS to the recording time (-t option, set to INT_MAX by default), so
the -ss option needs to be taken into account. It is not required in
do_{audio,video}_out() because this adjustment is set while polling
the filtergraph (see poll_filters()).
- It also adjusts the PTS sent to the encoder (and later transmitted to
the muxer) so the TS in the output make sense and are not kept
verbatim.
Note: this only works for muxers honoring the PTS, such as
lavf/matroskaenc. But for other such as the ASS muxer which just does a
verbatim copy, or the SubRip muxer which doesn't write the TS in some
cases, it will not work yet.
This ensures its not used afterwards.
This commit should not change anything as it should not have been used afterwards.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Filters can change the timings, so input files must not
necessarily be read at the same rythm. This patch select
the input file to read based on the timestamp at output
instead of input. With complex filter graphs, finding the
input for a given output is done by making a request and
checking to what buffer source it has been forwarded.