If the videos starts with B frame, then the minimum composition time
as computed by stts + ctts will be non-zero. Hence we need to shift
the DTS, so that the first pts is zero. This was the intention of that
code-block. However it was subtracting by the wrong amount.
For example, for one of the videos in the bug nonFormatted.mp4 we have
stts:
sample_count duration
960 1001
ctts:
sample_count duration
1 3003
2 0
1 3003
....
The resulting composition times are : 3003, 1001, 2002, 6006, ...
The minimum composition time or PTS is 1001, which should be used to
offset DTS. However the code block was wrongly using ctts[0] which is
3003. Hence the PTS was negative. This change computes the minimum pts
encountered while fixing the index, and then subtracts it from all the
timestamps after the edit list fixes are applied.
Samples files available from:
https://bugs.chromium.org/p/chromium/issues/detail?id=721451https://bugs.chromium.org/p/chromium/issues/detail?id=723537
fate-suite/h264/twofields_packet.mp4 is a similar file starting with 2
B frames. Before this change the PTS of first two B-frames was -6006
and -3003, and I am guessing one of them got dropped when being decoded
and remuxed to the framecrc before, and now it is not being dropped.
Signed-off-by: Sasi Inguva <isasi@google.com>
This test the demuxer discarding non ADTS frames at the beginning and
end of the input.
As a side effect, this commit also enables fate-adts-demux, which was
accidentally disabled in 324f0fbff1.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This new FATE test for the scale2ref filter makes use of the recently
added scale2ref-specific variables to maintain the aspect ratio of a
test input.
Filtergraph explanation:
[main] has an AR of 4:3. [ref] has an AR of 16:9.
640 / 4 = 160. So the new width for [main] is 160.
160 / ((320 / 240) * (1 / 1)) = 160 / (4 / 3) = 120. So the new
height for [main] is 120.
160 / 120 = 4 / 3 so [main]'s aspect ratio has been maintained while
using [ref]'s width as a reference point.
[ref] is nullsink'd since it is left unchanged by scale2ref (and so
shouldn't need to be tested).
If we were to use "iw/4:-1" in place of "iw/4:ow/mdar":
640 / 4 = 160. So the new width for [main] would be 160.
360 / 4 = 90. So the new height for [main] would be 90.
160 / 90 = 16 / 9 so [main] now has the same aspect ratio as [ref]
which is probably what you do not want.
This is currently the only test for scale2ref.
Signed-off-by: Kevin Mark <kmark937@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This removes the current API violating behavior of overwritting the stream's
extradata during packet filtering, something that should not happen after the
av_bsf_init() call.
The bitstream filter generated extradata is no longer available during
write_header(), and as such not usable with non seekable output. The FATE
tests are updated to reflect this.
Signed-off-by: James Almer <jamrial@gmail.com>
This complex (-1 2 6 2 -1) filter slightly less reduces interlace 'twitter' but better retain detail and subjective sharpness impression compared to the linear (1 2 1) filter.
Signed-off-by: Thomas Mundt <tmundt75@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
the tested sample contain negative value in the red channel
need to be clip to zero, and not set to MAX_RED
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add an option to webm_dash_manifest demuxer to specify a value for
"bandwidth" field in the DASH manifest. The value is then used by
the muxer. Fixes an existing FIXME in the code.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: James Zern <jzern@google.com>
As it gives excellent encoding gains at an insignificant speed increase
and passes fate without problems, it should now be safe to enable by
default.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This merges commits 8e2ea69135 and
096a8effa3 by Anton Khirnov, with the
following change:
- extract_extradata_check() is added to know if the codec is supported
by the bsf before trying to initialize it. This behaviour is similar to
the old AVCodecParser.split checks.
The FATE reference changes are due to the filtered out NAL units that
the old AVCodecParser.split implementation left alone.
Decoding is unchanged as the functions that parse extradata simply
ignored said unnecessary NAL units.
Signed-off-by: James Almer <jamrial@gmail.com>
This field is of little value, and interferes with testing side data,
since sizes can be different on multiple architectures.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Allows to get a more realistic total bitrate (and estimated file size)
in avi_write_header. Previously a static default value of 200k was
assumed.
Adds an internal helper function for bitrate guessing.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Preparation for potentially disabling merged side data by default in the
libs. Do this in particular because it affects fate tests.
The changed tests either reflect added packet side data, or the changed
packet size due to merged side data removal reducing the packet size.
The Chen-Shapiro(CS) test was used to test normality for
Lagged Fibonacci PRNG.
Normality Hypothesis Test:
The null hypothesis formally tests if the population
the sample represents is normally-distributed. For
CS, when the normality hypothesis is True, the
distribution of QH will have a mean close to 1.
Information on CS can be found here:
http://www.stata-journal.com/sjpdf.html?articlenum=st0264http://www.originlab.com/doc/Origin-Help/NormalityTest-Algorithm
Signed-off-by: Thomas Turner <thomastdt@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The constants used in the decoder used floating point precision,
and this caused different values to be generated on different
architectures. Additionally on big endian machines, the fate test
would output bytes in native order, which is different from the one
hardcoded in the test.
So, eradicate floating point numbers and use fixed point (32.32)
arithmetics everywhere, replacing constants with precomputed integer
values, and force the pixel format output to be the same in the fate
test.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This makes sure the actual stream parameters are used, which is
important mainly for hardware decoding+filtering cases, which would
previously require various weird workarounds to handle the fact that a
fake software graph has to be constructed, but never used.
This should also improve behaviour in rare cases where
avformat_find_stream_info() does not provide accurate information.
This merges Libav commit a3a0230. It was previously skipped.
The code in flush_encoders() which sets up a "fake" format wasn't in
Libav. I'm not sure if it's a good idea, but it tends to give
behavior closer to the old one in certain corner cases.
The vp8-size-change gives different result, because now the size of
the first frame is used. libavformat reported the size of the largest
frame for some reason.
The exr tests now use the sample aspect ratio of the first frame. For
some reason libavformat determines 0/1 as aspect ratio, while the
decoder returns the correct one.
The ffm and mxf tests change the field_order values. I'm assuming
another libavformat/decoding mismatch.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
This will be useful in the following commit, after which the muxer
timebase is not always available when encoding.
This merges Libav commit 3e265ca. It was previously skipped.
There are some changes with how/when the mux_timebase field is set,
because the Libav approach often causes a too imprecise time base
to be set. This is hard, because the muxer's write_header function
can readjust the timebase, at which point we might already have
encoded packets buffered. (It might be better to buffer them after
the encoder, instead of after all the timestamp handling logic
before muxing.)
The two FATE tests change because the output time base is raised
for subtitles. (Needed to avoid certain rounding issues in other
cases.)
Includes a minor merge fix by Mark Thompson, and
avconv: Move rescale to stream timebase before monotonisation
also by Mark Thompson <sw@jkqxz.net>.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
Michael Bradshaw
Sasi Inguva
James Almer
Rostislav Pehlivanov
Kevin Mark
Michael Niedermayer
Martin Vignali
Thomas Mundt
Vignesh Venkatasubramanian
Carl Eugen Hoyos
Paul B Mahol
Clément Bœsch
Vittorio Giovara
Tobias Rapp
Muhammad Faiz
Alexis Ballier
wm4
Diego Biurrun
Thomas Turner
Anton Khirnov
Micah Galizia