Otherwise the rate emulation logic in `transcode_step` never gets
hit, and the unavailability flag never gets reset, leading to an
eternal loop with some rate emulation use cases.
This change was missed during the rework of ffmpeg.c, in which
encoder initialization was moved further down the time line in
commit 67be1ce0c6 . Previously,
as the encoder initialization had happened earlier, this state was
not possible (flow getting as far as hitting the rate emulation logic,
yet not having the encoder initialized yet).
Fixes#9160
The obstacle to do so was in filter_codec_opts: It uses searches
the AVCodec for options via the AV_OPT_SEARCH_FAKE_OBJ method, which
requires using a void * that points to a pointer to a const AVClass.
When using const AVCodec *, one can not simply use a pointer that points
to the AVCodec's pointer to its AVClass, as said pointer is const, too.
This is fixed by using a temporary pointer to the AVClass.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It only affects the old and deprecated avcodec_decode_(video2|audio4)
API which is no longer used here.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
As per signal() help (man 2 signal) the semantics of using signal may
vary across platforms. It is suggested to use sigaction() instead.
Reviewed-by: Zane van Iperen <zane@zanevaniperen.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
The first stats is printed after the initial stats_period has elapsed. With a large period,
it may appear that ffmpeg has frozen at startup.
The initial stats is now printed after the first transcode_step.
At present, progress stats are updated at a hardcoded interval of
half a second. For long processes, this can lead to bloated
logs and progress reports.
Users can now set a custom period using option -stats_period
Default is kept at 0.5 seconds.
They add considerable complexity to frame-threading implementation,
which includes an unavoidably leaking error path, while the advantages
of this option to the users are highly dubious.
It should be always possible and desirable for the callers to make their
get_buffer2() implementation thread-safe, so deprecate this option.
We now have the possibility of getting AVFrames here, and we should
not touch the muxer's codecpar after writing the header.
Results of FATE tests change as the MXF and Matroska muxers actually
write down the field/frame coding type of a stream in their
respective headers. Before this change, these values in codecpar
would only be set after the muxer was initialized. Now, the
information is also available for encoder and muxer initialization.
Additionally, reap the first rewards by being able to set the
color related encoding values based on the passed AVFrame.
The only tests that seem to have changed their results with this
change seem to be the MXF tests. There, the muxer writes the
limited/full range flag to the output container if the encoder
is not set to "unspecified".
- For video, this means a single initialization point in do_video_out.
- For audio we unfortunately need to do it in two places just
before the buffer sink is utilized (if av_buffersink_get_samples
would still work according to its specification after a call to
avfilter_graph_request_oldest was made, we could at least remove
the one in transcode_step).
Other adjustments to make things work:
- As the AVFrame PTS adjustment to encoder time base needs the encoder
to be initialized, so it is now moved to do_{video,audio}_out,
right after the encoder has been initialized. Due to this,
the additional parameter in do_video_out is removed as it is no
longer necessary.
This way the old max queue size limit based behavior for streams
where each individual packet is large is kept, while for smaller
streams more packets can be buffered (current default is at 50
megabytes per stream).
For some explanation, by default ffmpeg copies packets from before
the appointed seek point/start time and puts them into the local
muxing queue. Before, it getting utilized was much less likely
since as soon as the filter chain was initialized, the encoder
(and thus output stream) was also initialized.
Now, since we will be pushing the encoder initialization to when the
first AVFrame is decoded and filtered - which only happens after
the exact seek point is hit as packets are ignored until then -
this queue will be seeing much more usage.
In more layman's terms, this attempts to fix cases such as where:
- seek point ends up being 5 seconds before requested time.
- audio is set to copy, and thus immediately begins filling the
muxing queue.
- video is being encoded, and thus all received packets are skipped
until the requested time is hit.
The AVFilterInOuts normally get freed in init_output_filter() when
the corresponding streams get created; yet if an error happens before
one reaches said point, they leak. Therefore this commit makes
ffmpeg_cleanup free them, too.
Fixes ticket #8267.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Threaded input can increase smoothness of e.g. x11grab significantly. Before
this patch, in order to activate threaded input the user had to specify a
"dummy" additional input, with this change it is no longer required.
Signed-off-by: Marton Balint <cus@passwd.hu>
This can support encoders which want frames and/or device contexts. For
the device case, it currently picks the first initialised device of the
desired type to give to the encoder - a new option would be needed if it
were necessary to choose between multiple devices of the same type.
Each time the sub2video structure is initialized, the sub2video
subpicture is initialized together with the first received heartbeat.
The heartbeat's PTS is utilized as the subpicture start time.
Additionally, add some documentation on the stages.
Fixes: signed integer overflow: -9223372036854775808 - 9223372036854775807 cannot be represented in type 'long'
Fixes: Ticket8142
Found-by: Suhwan
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
For audio packets with dts != AV_NOPTS_VALUEs the dts was converted
twice to the muxer's timebase during streamcopy, once as a normal
packet and once specifically as an audio packet. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
CPB side_data is copied when stream-copying (see init_output_stream_streamcopy()),
but it shall not be copied when the stream is decoded.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Freeing this was forgotten in ad899522.
Fixes#8315 and #8316.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
do_streamcopy() has a packet that gets zero-initialized first, then gets
initialized via av_init_packet() after which some of its fields are
oerwritten again with the actually desired values (unless it's EOF): The
side data is copied into the packet with av_copy_packet_side_data() and
if the source packet is refcounted, the packet will get a new reference
to the source packet's data. Furthermore, the flags are copied and the
timestamp related fields are overwritten with new values.
This commit replaces this by using av_packet_ref() to both initialize
the packet as well as populate its fields with the right values (unless
it's EOF again in which case the packet will still be initialized). The
differences to the current approach are as follows:
a) There is no call to a deprecated function (av_copy_packet_side_data())
any more.
b) Several fields that weren't copied before are now copied from the source
packet to the new packet (e.g. pos). Some of them (the timestamp related
fields) may be immediately overwritten again and some don't seem to be
used at all (e.g. pos), but in return using av_packet_ref() allows to forgo
the initializations.
c) There was no check for whether copying side data fails or not. This
has been changed: Now the program is exited in this case.
Using av_packet_ref() does not lead to unnecessary copying of data,
because the source packets are already always refcounted (they originate
from av_read_frame()).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch improves the logs when the message "cur_dts is invalid" appears.
If helps to identify which stream generates the trouble,
and the status of the stream.
A lot of users suffers with the message, and the origin varies.
The improved message can help to discover the cause.
Signed-off-by: Andreas Hakon <andreas.hakon@protonmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Right now, the code check for no filter description, but if we use a
filter_complex, the code will use the AVFrame.duration which could be
wrong in case of using fps filter.
How to reproduce the problem:
ffmpeg -f lavfi -i testsrc=duration=1 -vf fps=fps=50 -vsync 1 -f null -
output 50 frames
ffmpeg -f lavfi -i testsrc=duration=1 -filter_complex fps=fps=50 -vsync 1 -f null -
output 51 frames
With this commit, the same command will always output 50 frames.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
