This is similar to what was done before for output files and will allow
introducing demuxer-private state in future commits
Unlike for muxing, the code is moved to existing ffmpeg_demux.c rather
than to a new file. The reason is just file size - the demuxing code is
much smaller than muxing.
This enables overriding the rotation as well as horizontal/vertical
flip state of a specific video stream on the input side.
Additionally, switch the singular test that was utilizing the rotation
metadata to instead override the input display rotation, thus leading
to the same result.
ffmpeg_opt.c currently contains code for
- parsing the options provided on the command line
- opening and initializing input files based on these options
- opening and initializing output files based on these options
The code dealing with each of these is for the most part disjoint, so it
makes sense to move them to separate files. Beyond reducing the quite
considerable size of ffmpeg_opt.c, this will also allow exposing muxer
internals (currently private to ffmpeg_mux.c) to the initialization
code, thus removing the awkward separation currently in place.
This simplifies the code as there is no other place the error buffer
is needed, so the av_err2str helper macro can be used.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Currently it would essentially change the find_stream_info setting for
the file it was specified for and all following files, which is unusual
and somewhat unexpected behaviour for a per-file option and not even
documented to behave like this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It has been deprecated in favor of the aresample filter for almost 10
years.
Another thing this option can do is drop audio timestamps and have them
generated by the encoding code or the muxer, but
- for encoding, this can already be done with the setpts filter
- for muxing this should almost never be done as timestamp generation by
the muxer is deprecated, but people who really want to do this can use
the setts bitstream filter
It is either equal to OutputStream.enc_ctx->codec, or NULL when enc_ctx
is NULL. Replace the use of enc with enc_ctx->codec, or the equivalent
enc_ctx->codec_* fields where more convenient.
Don't silently replace it with the default layout for the amount of channels
from the requested layout.
Should fix ticket #9869
Signed-off-by: James Almer <jamrial@gmail.com>
Use it instead of AVStream.codecpar in the main thread. While
AVStream.codecpar is documented to only be updated when the stream is
added or avformat_find_stream_info(), it is actually updated during
demuxing. Accessing it from a different thread then constitutes a race.
Ideally, some mechanism should eventually be provided for signalling
parameter updates to the user. Then the demuxing thread could pick up
the changes and propagate them to the decoder.
There are currently three possible modes for an output stream:
1) The stream is produced by encoding output from some filtergraph. This
is true when ost->enc_ctx != NULL, or equivalently when
ost->encoding_needed != 0.
2) The stream is produced by copying some input stream's packets. This
is true when ost->enc_ctx == NULL && ost->source_index >= 0.
3) The stream is produced by attaching some file directly. This is true
when ost->enc_ctx == NULL && ost->source_index < 0.
OutputStream.stream_copy is currently used to identify case 2), and
sometimes to confusingly (or even incorrectly) identify case 1). Remove
it, replacing its usage with checking enc_ctx/source_index values.
Usually a HW decoder is expected when user specifies a HW acceleration
method via -hwaccel option, however the current implementation doesn't
take HW acceleration method into account, it is possible to select a SW
decoder.
For example:
$ ffmpeg -hwaccel vaapi -i av1.mp4 -f null -
$ ffmpeg -hwaccel nvdec -i av1.mp4 -f null -
$ ffmpeg -hwaccel vdpau -i av1.mp4 -f null -
[...]
Stream #0:0 -> #0:0 (av1 (libdav1d) -> wrapped_avframe (native))
libdav1d is selected in this case even if vaapi, nvdec or vdpau is
specified.
After applying this patch, the native av1 decoder (with vaapi, nvdec or
vdpau support) is selected for decoding(libdav1d is still used for
probing format).
$ ffmpeg -hwaccel vaapi -i av1.mp4 -f null -
$ ffmpeg -hwaccel nvdec -i av1.mp4 -f null -
$ ffmpeg -hwaccel vdpau -i av1.mp4 -f null -
[...]
Stream #0:0 -> #0:0 (av1 (native) -> wrapped_avframe (native))
Tested-by: Mario Roy <marioeroy@gmail.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
After applying this patch, the desired HW acceleration method is known
before selecting decoder, so we may take HW acceleration method into
account when selecting decoder for input stream in the next commit
There should be no functional changes in this patch
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Broken in 9c2b800203a5a8f3d83f3b8f28e8c50d28186b39.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Same issues apply to it as to -shortest.
Changes the results of the following tests:
- matroska-flac-extradata-update
The test reencodes two input FLAC streams into three output FLAC
streams. The last output stream is limited to 8 frames. The current
code results in the first two output streams having 12 frames, after
this commit all three streams have 8 frames and are the same length.
This new result is better, since it is predictable.
- mkv-1242
The test streamcopies one video and one audio stream, video is limited
to 11 frames. The new result shortens the audio stream so that it is
not longer than the video.
The -shortest option (which finishes the output file at the time the
shortest stream ends) is currently implemented by faking the -t option
when an output stream ends. This approach is fragile, since it depends
on the frames/packets being processed in a specific order. E.g. there
are currently some situations in which the output file length will
depend unpredictably on unrelated factors like encoder delay. More
importantly, the present work aiming at splitting various ffmpeg
components into different threads will make this approach completely
unworkable, since the frames/packets will arrive in effectively random
order.
This commit introduces a "sync queue", which is essentially a collection
of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
and are then released for further processing (encoding or muxing) when
it is ensured that the frame in question will not cause its stream to
get ahead of the other streams (the logic is similar to libavformat's
interleaving queue).
These sync queues are then used for encoding and/or muxing when the
-shortest option is specified.
A new option – -shortest_buf_duration – controls the maximum number of
queued packets, to avoid runaway memory usage.
This commit changes the results of the following tests:
- copy-shortest[12]: the last audio frame is now gone. This is
correct, since it actually outlasts the last video frame.
- shortest-sub: the video packets following the last subtitle packet are
now gone. This is also correct.
