Reads color_primaries, color_trc and color_space from mxf
headers. ULs are from https://registry.smpte-ra.org/ site.
Signed-off-by: Harry Mallon <harry.mallon@codex.online>
Previously, the hls-fmp4 and hls-fmp4_ac3 tests used the same file
names for init and segment files, which occasionally could cause
corruption and failed tests, if the input files for both tests are
generated in parallel, as they could overwrite each other.
This happened to work some of the time, as the fmp4_ac3 test actually
only checked the init segment file (which the fmp4 test case never
wrote, due to using the incorrect hls_segment_type option) and the
fmp4 test case always regenerated the input files due to mismatched
target and file names.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, with the file name not matching the target, the files
were regenerated every time fate is rerun - contrary to the other
test targets in the same file. (While regenerating it every time
might be desireable, as that's what the test is about, the file
at least has a dependency on the ffmpeg executable, making them
regenerated every time the executable is updated - and this change
at least makes it consistent with the rest.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Will prevet FATE from breaking once LIBAVCODEC_VERSION_MINOR is bumped to 100.
Reported-by: zane
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Add MMI & MSA runtime detection for MIPS.
Basically there are two code pathes. For systems that
natively support CPUCFG instruction or kernel emulated
that instruction, we'll sense this feature from HWCAP and
report the flags according to values grab from CPUCFG. For
systems that have no CPUCFG (or not export it in HWCAP),
we'll parse /proc/cpuinfo instead.
Signed-off-by: Jiaxun Yang <jiaxun.yang@flygoat.com>
Reviewed-by: Shiyou Yin <yinshiyou-hf@loongson.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When one of output[i] & expected_output is NAN, the unit test will always pass.
Signed-off-by: Ting Fu <ting.fu@intel.com>
Reviewed-by: Guo, Yejun <yejun.guo@intel.com>
add probeaudiostream for get audio stream's codec_name,codec_time_base,
sample_fmt,channels and channel_layout.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Important part of this algorithm is the double threshold step: pixels
above "high" threshold being kept, pixels below "low" threshold dropped,
pixels in between (weak edges) are kept if they are neighboring "high"
pixels.
The weak edge check uses a neighboring context and should not be applied
on the plane's border. The condition was incorrect and has been fixed in
the commit.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
floating point precision will cause rgb*max generate different value on
x86_32 and x86_64. have pass fate test on x86_32 and x86_64 by using
lrintf to get the nearest integral value for rgb * max before av_clip.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Now we just use one ADTS raw frame to calculate the bit rate, it's
lead to a larger error when get the duration from bit rate, the
improvement cumulate Nth ADTS frames to get the average bit rate.
e,g used the command get the duration like:
ffprobe -show_entries format=duration -i fate-suite/aac/foo.aac
before this improvement dump the duration=2.173935
after this improvement dump the duration=1.979267
in fact, the real duration can be get by command like:
ffmpeg -i fate-suite/aac/foo.aac -f null /dev/null with time=00:00:01.97
Also update the fate-adtstoasc_ticket3715.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This is a requirement of the AV1-ISOBMFF spec. Section 2.1.
General Requirements & Brands states:
* It SHALL have the av01 brand among the compatible brands array of the FileTypeBox
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This causes regressions in end to end timestamps with mp3s and ffmpeg.
The revert is to avoid this regression in the 4.3 release
See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted.
This reverts commit 460132c998.
7546ac2fee made it so that the start_time for mp3 files is
adjusted for skip_samples. However, this appears incorrect because
subsequent packet timestamps are not adjusted and skip_samples are
applied by deleting data from a packet without changing the timestamp.
E.g., we are told the start_time is ~25ms and we get a packet with a
timestamp of 0 that has had the skip_samples discarded from it. As such
rendering engines may incorrectly discard everything prior to the
25ms thinking that is where playback should officially start. Since the
samples were deleted without adjusting timestamps though, the true
start_time is still 0.
Other formats like MP4 with edit lists will adjust both the start
time and the timestamps of subsequent packets to avoid this issue.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
A buffer whose size is not a multiple of four has been initialized using
consecutive writes of 32bits. This results in a stack-buffer-overflow
reported by ASAN in the checkasm-sw_scale FATE-test.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
