Follows the proper order defined by the spec, even if mostly cosmetic, and is
also preparation for a following change.
Signed-off-by: James Almer <jamrial@gmail.com>
Tests both the Media 100 decoder (using the media100_to_mjpegb BSF
implicitly) as well as using said BSF, followed by the MJPEGB decoder.
(We currently hit a bug when remuxing: The demuxer treats compressorname
as encoded in a Mac character encoding (Mac OS Roman?) and converts
it to UTF-8, yet the muxer just writes it.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The iirfilter is only used in its test tool since
01ecb7172b which
stopped using it in AAC, its only user.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The follow cmd output corrupted file before the patch:
ffmpeg -f lavfi -i color=blue,trim=duration=0.04 \
-f lavfi -i anullsrc,atrim=duration=2 \
-movflags +empty_moov+hybrid_fragmented \
-frag_duration 1000000 \
-frag_interleave 1 \
output.mp4
1. first_track is the first track with track->entry != 0. As in the
command above, video track (track index 0) has a single frame. When
flush the second fragment, first_track is 1, the audio track.
2. write_moof = i == first_track, so write_moof is false for i = 0.
3. When mov->frag_interleave != 0, mov->mdat_buf != NULL, because
it contains audio data. So avio_write is called before write_moof,
that is, the data write before moof, and mov_finish_fragment
executed with wrong mdat_start.
4. With normal fmp4 output, the error isn't obvious. With
hybrid_fragmented, ffplay output.mp4 shows a lot of error messages.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Don't use a 7.1 EAC3 input file for which our decoder is not
bitexact; instead just use the asynth-44100-8.wav file
which (as a 7.1 file) exhibits the same issue fixed by
1b3f4842c1.
(Either the encoder or the resampler are still not completely
bitexact, so we limit the number of frames output.)
Also switch to a framecrc test so that the output channel layout
is directly contained in the ref file.
Reviewed-by: James Almer <jamrial@gmail.com>
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The file has buggy timestamps (it uses B-frames, yet pts==dts)
and therefore the last frame is currently discarded by FFmpeg cli.
Using -fps_mode passthrough avoids this and provides coverage
of the SVQ3 draining logic.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Bitstream generated using the reference encoder, then edited to fix the
colour description and an extra metadata block added. FFmpeg decoder
output is identical to the reference decoder output.
The content used is the first three frames of "Waterfall" from the SVT
Open Content Video Test Suite 2022. This is copyright Sveriges
Television AB and is used under the Creative Commons Attribution 4.0
International License.
This bsf converts AV_PKT_DATA_NEW_EXTRADATA side data in avcc format
to in-band annexb format. However, the side data wasn't been removed
and copied from input packet to output packet. So the output packet
has mixed bitstream format. We don't support mixed bitstream format.
For example, h264_metadata report error in the following case:
ffmpeg -i foo.flv \
-bsf:v "h264_mp4toannexb,h264_metadata" \
-c copy -f null
This patch removed NEW_EXTRADATA side data after process.
This patch also add a check so only NEW_EXTRADATA in avcc format is
processed. NEW_EXTRADATA in annexb format is copied to output as is.
Reported-by: jiangjie <jiangjie618@gmail.com>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
A sample with a particular partitioning structure that could not be read
correctly before 26c5d8cf5d
Signed-off-by: Frank Plowman <post@frankplowman.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Since 45eeb1f785
optimal Huffman tables are the default (without slice-threading).
This made the fate-vsynth*-mjpeg-{trell-,}-huffman tests
identical to their corresponding tests without "-huffman".
This is of course wasteful, so switch the two tests with
"-huffman" counterparts back to the default tables.
Also use one of these tests to test slice threaded encoding.
It has so far been untested.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is currently not due to endianness. This forced to add
workarounds with sed in fate/mxf.mak (which are removed
in this commit).
This is supposed to fix the enhanced-flv-hevc-hdr10 test
on big endian systems.
Reviewed-by: Zhao Zhili <quinkblack-at-foxmail.com@ffmpeg.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
With this modification the test would have caught the regression
introduced in 72bf3d3c12.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Encoding was untested before this.
Notice that the filesize degradation is partially due to
mpegvideo no longer using progressive_sequence and
progressive_frame.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The Main profile of AAC is... terrible.
It enables the use of delta coding across coefficients of two frames
to try to increase compression, and it enabled one more pole for TNS
filters.
What the AAC authors failed to take into account were basic
mathematics, as MDCT leakage (e.g. the spread of each frequency when
represented in a discrete spectrum) is significant in most audio codecs.
This leads to huge variations between each frame, basically rendering
prediction completely pointless.
In fact, enabling AAC-Main prediction does not, in general, even recoup
the metadata losses from signalling the profile and prediction properties
in the first place. So you lose efficiency by using AAC Main.
The rumor is that it was put in the AAC spec for patent reasons, though
patent-wise, it has about as much use as a patent for a bicycle designed
for use by snakes.
The only other thing AAC Main changes is it permits 3-pole TNS filters.
When AAC's bands are absolutely tiny, except for very high frequency bands,
where you're likely to use PNS instead.
Just get rid of it.
