We have test to make sure that certain configurations do print
warnings. However, the normal operation of the muxer within this
test always printed a warning, so those tests to check for
extra warnings didn't essentially guard anything.
The warning that always was printed, "track 1: codec frame size is
not set" was not present in the libav fork where this testcase
originated, it was removed in f234e8a32e.
Set the frame size for the audio stream to silence the warning,
and use this frame size in a couple later calculations, and check
that one test configuration doesn't print warnings.
Setting the frame size apparently changes the rounding of a timestamp
in the ismv muxing testcase.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is based on a spec at https://aomediacodec.github.io/id3-emsg/,
further based on ISO/IEC 23009-1:2019.
Within libavformat, timed ID3 metadata (already supported by the
mpegts demuxer and muxer) is handled as a separate data AVStream
with codec type AV_CODEC_ID_TIMED_ID3. However, it doesn't
have a corresponding track in the mov file - instead, these events
are written as separate toplevel 'emsg' boxes.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously we always assumed that the channels are in native order, even if
they were not. The new channel layout API allows us to signal the proper
channel order, so let's do so.
Signed-off-by: Marton Balint <cus@passwd.hu>
Unlike what the old comment suggested, standard ASS has no character
escape mechanism, but a closing curly bracket doesn't even need one.
For manual authored sub files using a full-width variant of an
appropriate font and with scaling and spacing modifiers is a common
workaround.
This is not an option here, but we can still make things much less bad.
Now the desired opening bracket still shows up in libass, and
standard renders will merely display a backslash in its place
instead of stripping the following text like before.
Creating vsynth_lena.yuv needs the FATE suite,
yet several tests in ffmpeg.mak without a dependency
on samples used it as input file. Fix this by using
vsynth1.yuv (which does not have such a dependency)
instead.
Also use vsynth1.yuv in fate-shortest to avoid
the samples dependency in this test, too.
Fixes ticket #10947.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Both samples rely on a feature our decoder doesn't currently support.
Should fix fate failures on some systems where not even the one single frame
could be generated.
Signed-off-by: James Almer <jamrial@gmail.com>
The fits decoder decodes to native pixel formats; so
the fitsdec-gbrap16be fate test failed on BE despite
its name because the reference file is LE.
This patch fixes this by forcing a pixel format;
the forced pixel format is BE, causing a change
in the reference file.
The fitsdec-gbrp16be test was not affected, because
its source file (lena-rgb48.png from tne FATE suite)
is actually biendian (as if someone had multiplied
8bit content by 257...).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The test depends on the compile option of x265. It failed when
HIGH_BIT_DEPTH isn't enabled. It also failed when asan is enabled
because of memory issue inside of x265, which I don't think can
be fixed within FFmpeg.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
This test muxes two streams into a single pcm file, although
the two streams are of course not recoverable from the output
(unless one has extra information). So use the streamhash muxer
instead (which also provides coverage for it; it was surprisingly
unused in FATE so far). This is in preparation for actually
enforcing a limit of one stream for the PCM muxers.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The newer of these two are the separate integers for content light
level, introduced in 3952bf3e98c76c31594529a3fe34e056d3e3e2ea ,
with X265_BUILD 75. As we already require X265_BUILD of at least
89, no further conditions are required.
Both of these two structures were first available with X264_BUILD
163, so make relevant functionality conditional on the version
being at least such.
Keep handle_side_data available in all cases as this way X264_init
does not require additional version based conditions within it.
Finally, add a FATE test which verifies that pass-through of the
MDCV/CLL side data is working during encoding.
These two were added in 28e23d7f348c78d49a726c7469f9d4e38edec341
and 3558c1f2e97455e0b89edef31b9a72ab7fa30550 for version 0.9.0 of
SVT-AV1, which is also our minimum requirement right now.
In other words, no additional version limiting conditions seem
to be required.
Additionally, add a FATE test which verifies that pass-through of
the MDCV/CLL side data is working during encoding.
In particular, test writing tags with odd strlen.
(These tags are zero-padded to even size.)
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also make use of the av_channel_from_string() function to determine the channel
id. This fixes some parse issues in av_channel_layout_from_string().
Signed-off-by: Marton Balint <cus@passwd.hu>
We lacked tests which supposed to fail, and there are some which should fail
but right now it does not. This will be fixed in a later commit.
Signed-off-by: Marton Balint <cus@passwd.hu>
Deduplicates a lot of code.
Some minor differences (mostly white space and inconsistent use of quotes) are
expected in the fate tests, there was no point aiming for exactly the same
formatting.
Signed-off-by: Marton Balint <cus@passwd.hu>
This makes the wav and pcm demuxer demux bigger packets, which is more
efficient.
As a side effect of the bigger packets, audio durations can become less exact
for command lines such as "ffmpeg -i $INPUT -c:a copy -t 1.0 $OUTPUT".
Signed-off-by: Marton Balint <cus@passwd.hu>
- Remove the 1024 cap on the number of samples, for high sample rate audio it
was suboptimal, calculate the low neighbour power of two for the number of
samples (audio blocks) instead.
- Make the function work correctly also for non-pcm codecs by using the stream
bitrate to estimate the target packet size. A previous version of this patch
used av_get_audio_frame_duration2() the estimate the desired packet size, but
for some codecs that returns the duration of a single audio frame regardless
of frame_bytes.
- Fallback to 4096/block_align*block_align if bitrate is not available.
Signed-off-by: Marton Balint <cus@passwd.hu>
The samples I found all have 2000 sample packets, and by forcing the packet
size with a bsf we could automagically make muxing work for packets containing
more than 3640 samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
Treat it analogously to stream parameters like format/dimensions/etc.
This is functionally different from previous code in 2 ways:
* for non-CFR video, the frame timebase (set by the decoder) is used
rather than the demuxer timebase
* for sub2video, AV_TIME_BASE_Q is used, which is hardcoded by the
subtitle decoding API
These changes should avoid unnecessary and potentially lossy timestamp
conversions from decoder timebase into the demuxer one.
Changes the timebases used in sub2video tests.
