cbs.mak is meant to contain tests strictly for the CBS framework, not for any
bsf that happens to use it under the hood.
Signed-off-by: James Almer <jamrial@gmail.com>
Previously, these tests failed when running on Windows, if the
system is configured with a time zone east of Greenwich, i.e.
with a positive GMT offset.
The muxer converts the creation_date given by the user using
av_parse_time to unix time, as a time_t. The creation_date is
interpreted as a local time, i.e. according to the current time
zone. (This time_t value is then converted back to a broken out
local time form with localtime_r.)
The given reference date/time, "1970-01-01T00:00:00", is the
origin point for unix time, corresponding to time_t zero. However
when interpreted as local time, this doesn't map to exactly zero.
Time zones east of Greenwich reached this time a number of hours
before the point of zero time_t - so the corresponding time_t
value essentially is minus the GMT offset, in seconds.
Windows mktime returns an error, returning (time_t)-1, when given
such a "struct tm", while e.g. glibc mktime happily returns a
negative time_t. av_parse_time doesn't check the return value of
mktime for potential errors.
This is observable with the following test snippet:
struct tm tm = { 0 };
tm.tm_year = 70;
tm.tm_isdst = -1;
tm.tm_mday = 1;
tm.tm_hour = 0;
time_t t = mktime(&tm);
printf("%d-%02d-%02d %02d:%02d:%02d\n", tm.tm_year + 1900, tm.tm_mon + 1, tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec);
printf("t %d\n", (int)t);
By varying the value of tm_hour and the system time zone, one
can observe that Windows mktime returns -1 for all time_t values
that would have been negative.
This range limit is also documented by Microsoft in detail at
https://learn.microsoft.com/en-us/cpp/c-runtime-library/reference/mktime-mktime32-mktime64.
To avoid the issue, pick a different, arbitrary reference time,
which should have a nonnegative time_t for all time zones.
Don't overwrite the bitstream values when updating the top-level loop
filter and segmentation state, instead do the update separately at the
end of the frame parsing.
This also reverts the change to the passthrough tests which made them
have output not matching the input.
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>
- 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>
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>
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 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>
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>
Up until now, our muxer wrote FFV1 in video-for-windows
compatibility mode out of concern for old demuxers that
only support that (whereas the demuxer accepts V_FFV1).
This commit switches to using native mode, because
a) V_FFV1 is around long enough so that old demuxers
should not be an issue (support in FFmpeg has been added
in commit 9ae762da7e
in March 2017/FFmpeg 3.3),
b) using native mode uses fewer bytes for the CodecPrivate,
c) the VfW extradata is zero-padded to an even length
if necessary, but our demuxer forgot to undo the padding
until very recently (92e310eb82),
so that there are many versions of our demuxer around that
are buggy wrt VFW, but not V_FFV1.
This affects the FFV1 extradata checksums, specifically
the (experimental) version 4 files with error check version 2*
as created by
ffmpeg -i ../fate-suite/mpeg2/sony-ct3.bs -c:v ffv1 \
-slices 16 -frames 1 -level 4 -strict experimental ffv1.mkv
VFW files like the above created by this muxer before this patch
would not work with an old demuxer.
*: Without error check version 2, the CRC for the whole extradata
is zero, which is not changed by appending a zero byte.
Reviewed-by: compn <ff@hawaiiantel.net>
Reviewed-by: Dave Rice <dave@dericed.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The structure is padded to an even length with an internal
size field to indicate the real size.
The matroska-matroska-display-metadata test (writing FFV1
in VFW mode) was affected by this.
It should also fix ticket #11613.
Reviewed-by: compn <ff@hawaiiantel.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.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>
I have several .ts captures where video and audio codec changes even
though the PMT version does not change and the PIDs stay the same.
This happens during transition to/from slate (mpeg2 video and audio)
to network broadcast (hevc video and eac3 audio in private PES).
I've updated fate ts-demux expected results.
