Should set "number of frames" to bytes 24-27 of IVF header, not
duration.
It is described by [1], and confirmed by parsing all IVF files in [2].
This commit also updates the md5sum of refs to pass fate-cbs.
[1] Duck IVF - MultimediaWiki
https://wiki.multimedia.cx/index.php/Duck_IVF
[2] webm/vp8-test-vectors
https://chromium.googlesource.com/webm/vp8-test-vectors
Signed-off-by: Jianhui Dai <jianhui.j.dai@intel.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Changes the result of fate-mxf-probe-dv25, where the bitrate is now
exported.
Also changes the result of fate-bsf-dv-error-marker, where the exported
bitrate is now different. Note that the codec layer bitrate does not
match the container bitrate, because container timing is 25fps, while
the DV profile is 50.
Add an optional filter_line3 to the available optimisations.
filter_line3 is equivalent to filter_line, memcpy, filter_line
filter_line shares quite a number of loads and some calculations in
common with its next iteration and testing shows that using aarch64
neon filter_line3s performance is 30% better than two filter_lines
and a memcpy.
Adds a test for vf_bwdif filter_line3 to checkasm
Rounds job start lines down to a multiple of 4. This means that if
filter_line3 exists then filter_line will not sometimes be called
once at the end of a slice depending on thread count. The final slice
may do up to 3 extra lines but filter_edge is faster than filter_line
so it is unlikely to create any noticable thread load variation.
Signed-off-by: John Cox <jc@kynesim.co.uk>
Signed-off-by: Martin Storsjö <martin@martin.st>
No need to generate intermediate files and probe them. We only care to know that the
output of the bsf excludes the frames in question, and a simple checksum is enough.
Signed-off-by: James Almer <jamrial@gmail.com>
Should fix integer overflows, and improve encoding results.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixed-point AAC decoder currently does not produce the same output on
all platforms. Until that is fixed, silence the audio stream using the
volume filter.
Also, actually use the aac_fixed decoder as was the original intent.
The code will currently add a small offset to avoid exact midpoints, but
this can cause inexact results when a float timestamp is exactly
representable as an integer.
Fixes off-by-one in the first frame duration in multiple FATE tests.
Use the next I/P/B or start code as the end of current frame.
Before the patch, extension start code, user data start code,
sequence end code and so on are treated as the start of next
frame.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Since this is an external encoder not under our control, we cannot test
the encoded output exactly as is done for internal encoders. We can
still test however that the output is decodable and produces the
expected number of frames with expected dimensions, pixel formats, and
timestamps.
Currently those are set in different ways depending on whether the
stream is decoded or not, using some values from the decoder if it is.
This is wrong, because there may be arbitrary amount of delay between
input packets and output frames (depending e.g. on the thread count when
frame threading is used).
Always use the path that was previously used only for streamcopy. This
should not cause any issues, because these values are now used only for
streamcopy and discontinuity handling.
This change will allow to decouple discontinuity processing from
decoding and move it to ffmpeg_demux. It also makes the code simpler.
Changes output in fate-cover-art-aiff-id3v2-remux and
fate-cover-art-mp3-id3v2-remux, where attached pictures are now written
in the correct order. This happens because InputStream.dts is no longer
reset to AV_NOPTS_VALUE after decoding, so streamcopy actually sees
valid dts values.
Stop using InputStream.dts for generating missing timestamps for decoded
frames, because it contains pre-decoding timestamps and there may be
arbitrary amount of delay between input packets and output frames (e.g.
dependent on the thread count when frame threading is used). It is also
in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary
rounding issues.
New code maintains a timebase that is the inverse of the LCM of all the
samplerates seen so far, and thus can accurately represent every audio
sample. This timebase is used to generate missing timestamps after
decoding.
Changes the result of the following FATE tests
* pcm_dvd-16-5.1-96000
* lavf-smjpeg
* adpcm-ima-smjpeg
In all of these the timestamps now better correspond to actual frame
durations.
One that is fine enough to represent all DV audio sample rates. Audio
packet durations are now sample-accurate.
This largely undoes commit 76fbb0052d. To
avoid breaking the issue fixed by that commit, resync audio timestamps
against video if they get more than one frame apart. The sample from
issue #8762 still works correctly after this commit.
Slightly changes the results of the lavf-dv seektest, due to the audio
timebase being more granular.
Current code will call avpriv_set_pts_info() for each video frame,
possibly setting a different timebase if the stream framerate changes.
This violates API conventions, as the timebase is supposed to stay
constant after stream creation.
Change the demuxer to set a single timebase that is fine enough to
handle all supported DV framerates.
The seek tests change slightly because the new timebase is more
granular.
Previously they would only be used with trivial filtergraphs, because
filters did not handle frame durations. That is no longer true - most
filters process frame durations properly (there may still be some that
don't - this change will help finding and fixing them).
Improves output video frame durations in a number of FATE tests.
