This allows weird subsampling with progressive JPEGs to be decoded,
such as full-RG and only B subsampled.
Signed-off-by: Leo Izen <leo.izen@gmail.com>
Drop unneeded ctype.h and math.h.
Group all system headers together.
Sort unconditional includes alphabetically.
Group local includes by the library, sort alphabetically.
Several places in the code currently call init_output_stream_wrapper(),
which in turn calls init_output_stream(), which then calls either
enc_open() or init_output_stream_streamcopy(), followed by
of_stream_init(), which tells the muxer the stream is ready for muxing.
All except one of these callers are in the encoding code, which will be
moved to ffmpeg_enc.c. Keeping this structure would then necessitate
ffmpeg_enc.c calling back into the common code in ffmpeg.c, which would
then just call ffmpeg_mux, thus making the already convoluted call chain
even more so.
Simplify the situation by using separate paths for filter-fed output
streams (audio and video encoders) and others (subtitles, streamcopy,
data, attachments).
Encoder initialization is currently split rather arbitrarily between
init_output_stream_encode() and init_output_stream(). Move all of it to
init_output_stream_encode().
The code currently uses lavfi for this, which creates a sort of
configuration dependency loop - the encoder should be ideally
initialized with information from the first audio frame, but to get this
frame one needs to first open the encoder to know the frame size. This
necessitates an awkward workaround, which causes audio handling to be
different from video.
With this change, audio encoder initialization is congruent with video.
For audio AVFrames, nb_samples is typically more trustworthy than
duration. Since sync queues look at durations, make sure they match the
sample count.
The last audio frame in the fate-shortest test is now gone. This is more
correct, since it outlasts the last video frame.
stride value is not relevant with unpadded content and the total count
of pixels (width x height) must be used instead of the rounding based on
width only then multiplied by height
unpadded_10bit value computing is moved sooner in the code in order to
be able to use it during computing of minimal content size. Also make sure to
only set it for 10bit.
Fix 'Overread buffer' error when the content is not lucky enough to have
(enough) padding bytes at the end for not being rejected by the formula
based on the stride value
Fixes ticket #10259.
Signed-off-by: Jerome Martinez <jerome@mediaarea.net>
Signed-off-by: Marton Balint <cus@passwd.hu>
Extend the decklink output to include support for compressed AC-3,
encapsulated using the SMPTE ST 377:2015 standard.
This functionality can be exercised by using the "copy" codec when
the input audio stream is AC-3. For example:
./ffmpeg -i ~/foo.ts -codec:a copy -f decklink 'UltraStudio Mini Monitor'
Note that the default behavior continues to be to do PCM output,
which means without specifying the copy codec a stream containing
AC-3 will be decoded and downmixed to stereo audio before output.
Thanks to Marton Balint for providing feedback.
Signed-off-by: Devin Heitmueller <dheitmueller@ltnglobal.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
The Matroska spec requires it to be equal to the highest BlockAddID value in a
BlockAdditions, but being purely an informative element, only abort if strict
compliance is requested, as demuxing is otherwise unaffected.
Signed-off-by: James Almer <jamrial@gmail.com>
