This aligns declared function types in checkasm with real definition.
Fixes FATE: checkasm-{sw_rgb,sw_scale,sw_yuv2rgb,sw_yuv2yuv}
Fixes: runtime error: call to function <func> through pointer to incorrect function type
Fixes: c1a0e65763
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
In aac/aac-fixed, also remove unnecessary aac demuxer dependency.
Signed-off-by: Nicolas Gaullier <nicolas.gaullier@cji.paris>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When CMD is crc/framecrc, always use the macros CRC/FRAMECRC, even if it
includes unnecessary requirements for rawvideo/pcm_s16le encoders (as
actually noticed in a comment of the Makefile).
Signed-off-by: Nicolas Gaullier <nicolas.gaullier@cji.paris>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
First, always require file protocol when FATE suite is used.
Then, add missing dependencies while removing duplicates.
Signed-off-by: Nicolas Gaullier <nicolas.gaullier@cji.paris>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
A frame graph activation might not produce a frame in the requested sink, so
keep on requesting a frame there unless we encounter a filter activation with
buffersrc empty error.
This makes av_buffersink_get_frame(_flags) work according to its documentation
which claims that EAGAIN is only returned if additional frames must be inserted
into the graph.
Fate changes are because audio frames will have different sizes at segment
boundaries, but content is the same.
Signed-off-by: Marton Balint <cus@passwd.hu>
Before this patch, the DFPWM1a encoder was marked as supporting
variable frame sizes. The DFPWM1a format converts eight bytes
of input into one output byte and so it simply padded the number
of data output by
frame->nb_samples * frame->ch_layout.nb_channels / 8 +
(frame->nb_samples % 8 > 0 ? 1 : 0)
This has several bugs:
a) The additional byte leads to eight additional input byte being
read; this can read into the frame's padding, i.e. the data can
be uninitialized.
b) The criterion for whether one should pad is wrong:
nb_samples * nb_channels should be tested for divisibility by eight.
c) The created frames can be undecodable (at least with our decoder):
Our decoder requires the number of bits per frame to divisible by
the number of channels, yet the above approach does not guarantee this.
d) The padding will be added in the middle of the stream (potentially
for every packet).
This commit fixes all of this by removing the variable frame size cap
and using AVCodecInternal.pad_samples to pad the last frame so that
nb_samples * nb_channels is always a multiple of eight.
The lavf-dfpwm FATE-test was affected by a). The frames originated from
lavfi and were part of an audio frame pool, so that the padding
contained data from an earlier (bigger) frame. Now the last frame is
properly filled with silence.
Reported-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The way streams are coded in an IAMF struct follows a scalable model where the
channel layouts for each layer may not match the channel order our API can
represent in a Native order layout.
For example, an audio element may have six coded streams in the form of two
stereo streams, followed by two mono streams, and then by another two stereo
streams, for a total of 10 channels, and define for them four scalable layers
with loudspeaker_layout values "Stereo", "5.1ch", "5.1.2ch", and "5.1.4ch".
The first layer references the first stream, and each following layer will
reference all previous streams plus extra ones.
In this case, the "5.1ch" layer will reference four streams (the first two
stereo and the two mono) to encompass six channels, which does not match out
native layout 5.1(side) given that FC and LFE come after FL+FR but before
SL+SR, and here, they are at the end.
For this reason, we need to build Custom order layouts that properly represent
what we're exporting.
----
Before:
Stream group #0:0[0x12c]: IAMF Audio Element:
Layer 0: stereo
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Layer 1: 5.1(side)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Layer 2: 5.1.2
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Layer 3: 5.1.4
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:5[0x5]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
----
AFter:
Stream group #0:0[0x12c]: IAMF Audio Element:
Layer 0: stereo
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Layer 1: 6 channels (FL+FR+SL+SR+FC+LFE)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Layer 2: 8 channels (FL+FR+SL+SR+FC+LFE+TFL+TFR)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Layer 3: 10 channels (FL+FR+SL+SR+FC+LFE+TFL+TFR+TBL+TBR)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:5[0x5]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Signed-off-by: James Almer <jamrial@gmail.com>
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>
This is useful for tests that compare dctcoefs which will be either 2 bytes or
4 bytes, depending on bitdepth.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows detecting changes and regressions in side data related code, same as what
framecrc does for before already for packet data itself.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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>
As described in section F.6.1 from ETSI TS 102 366.
Found-by: nyanmisaka
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Improves compatibility with Resolume products by adding an additional
hashtable for DXT color+LUT combinations, and padding the DXT texture
dimensions to the next largest multiple of 16. Produces identical
packets to Resolume Alley in manual tests.
Signed-off-by: Emma Worley <emma@emma.gg>
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>
