Initially, avcodec/srtenc.c was outputting CRLF [1]. Later, a real SRT
muxer was added [2], which outputs LF. The original srtenc.c was
converted to use the muxer [3], changing its output to LF, except for
newline characters within subtitle text.
Fix this to avoid producing SRT files with mixed line endings.
[1] 8e43b6fed9
[2] 9e63c30daa
[3] 55180b3299
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
This was introduced in commit 9c43703, to support a codec "extension"
in the prores_aw encoder.
This removes the chroma fill loop, and instead performs the inverse
transform on null coefficients, which achieves the same result and
fixes an off-by-one in the chroma values produced.
Updated test to reflect this change.
The first sample in the stsc box may not refer to the first stsd entry.
This is the case in h264/thezerotheorem-cut.mp4, and as such the
fate-h264_redundant_pps-side_data test is updated accordingly.
Signed-off-by: James Almer <jamrial@gmail.com>
The new logic should be easier to follow.
It also uses ff_inlink_consume_frame() for all simple passthrough operations
making custom get_audio_buffer callback unnecessary.
Fate changes are because the new logic does not repacketize input audio up
until the crossfade. Content is the same.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also ensure that the dst buffers are not too big
(they had the right size for >8 bit depths and were therefore
too big for eight bit, letting potential buffer overflows
in the eight bit version go undetected).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
- Changes in mov_write_video_tag function to handle APV elementary stream
- Provided structure APVDecoderConfigurationRecord that specifies the decoder configuration information for APV video content
Co-Authored-by: James Almer <jamrial@gmail.com>
Signed-off-by: Dawid Kozinski <d.kozinski@samsung.com>
Signed-off-by: James Almer <jamrial@gmail.com>
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>
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>