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>
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.
The muxing queue currently lives in OutputStream, which is a very large
struct storing the state for both encoding and muxing. The muxing queue
is only used by the code in ffmpeg_mux, so it makes sense to restrict it
to that file.
This makes the first step towards reducing the scope of OutputStream.
Figure out earlier whether the output stream/file should be bitexact and
store this information in a flag in OutputFile/OutputStream.
Stop accessing the muxer in set_encoder_id(), which will become
forbidden in future commits.
Move the file size checking code to ffmpeg_mux. Use the recently
introduced of_filesize(), making this code consistent with the size
shown by print_report().
Move header_written into it, which is not (and should not be) used by
any code outside of ffmpeg_mux.
In the future this context will contain more muxer-private state that
should not be visible to other code.
This is a per-file input option that adjusts an input's timestamps
with reference to another input, so that emitted packet timestamps
account for the difference between the start times of the two inputs.
Typical use case is to sync two or more live inputs such as from capture
devices. Both the target and reference input source timestamps should be
based on the same clock source.
If either input lacks starting timestamps, then no sync adjustment is made.
