The existing meridian audio test does not test
ff_mlp_rematrix_channel_arm. This sample (first 640k of
https://samples.libav.org/A-codecs/TrueHD/TrueHD.raw) uses
ff_mlp_rematrix_channel_arm. Since this sample has 5.1 channels it also
allows testing the integrated downmixing.
This uses the RIFF header stored size to figure out the expected AVI
file size, instead of the actual file. To work fully it requires handling
failed avio_seek() instead of assuming they always succeed.
Some fate file has been cut off and contains half a frame at the end which
previously was not output during demuxing. This frame is now output to
encoder, thus the fate diff update.
Bug-Id: 261
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
According to the DPX file format description found at
http://www.fileformat.info/format/dpx/egff.htm the ImageElement part of
the GenericImageHeader also contains an an offset to the real image data
beside the same member that can be found in the GenericFileHeader.
Libav keeps this member empty (=0) while some applications expects it to
be filled properly. FATE test updated accordingly.
Bug-Id: 742
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Only shift limited range luma, and always only shift chroma
for upconversion.
Based off a patch by Michael Niedermayer.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The new reference.pnm is a freely licensed replacement. The photo has
been taken by Reinhard Tartler on August 28 2014, and is licensed under
the expat license as stated at http://www.jclark.com/xml/copying.txt
In these cases, only drop dts. Because if we drop both we have no
timestamps at all for some files.
This improves playback of HLS streams from GoPro cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes the default of '1' more explicit than defaulting to '1' in
fate-run.sh and regression-funcs.sh if THREADS is not set.
Fixes the reported thread count in fate-cpu if THREADS is not set.
libavutil/cpu-test prints raw and effective cpu flags to STDERR. Detected
cpu flags can be useful for debugging fate errors.
No comparison of the result against a expected result since that would
require fate config specific references.
Adding 'branch=release/10' to the fate config file will check the
release/10 branch instead of master. If no branch is specified it will
use 'master' so that existing config are still valid.
The server side changes are already deployed, see
https://fate.libav.org/v10/ for an example. The server supports only the
release/* branches.
The server enforces that a single slot tests always the same branch.
Please append "-v$RELEASE" to the slot of release branch configs or make
the slot otherwise unique.
A different fate samples dir is needed for each release branch. make
fate-rsync has the correct URL in each branch.
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
This results in DefaultDuration not being written when the framerate is
not known, but as this field is purely informative, this should not
break any sane demuxers.
This corrects the bug that caused the checksums to change in
9767d7c092.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use it instead of checking CODEC_FLAG_BITEXACT in the first stream's
codec context.
Using codec options inside lavf is fragile and can easily break when the
muxing codec context is not the encoding context.
Initial implementation by Andrew D'Addesio <modchipv12@gmail.com> during
GSoC 2012.
Completion by Anton Khirnov <anton@khirnov.net>, sponsored by the
Mozilla Corporation.
Further contributions by:
Christophe Gisquet <christophe.gisquet@gmail.com>
Janne Grunau <janne-libav@jannau.net>
Luca Barbato <lu_zero@gentoo.org>
Also, stop using AVCodecContext.codec_name as fallback, since it will be
deprecated.
Changes the result of the lavf-asf test (and its associated seektest),
since 'msmpeg4v3' gets written instead of just 'msmpeg4'.
Also set the RGBA pixel format correctly as the native endian format,
which is what it returns.
This fixes the tests on big endian.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes conversion of pal8 to rgb formats with alpha.
Updated references for 2 FATE tests which previously encoded fully
transparent images.
Based on a patch by Baptiste Coudurier <baptiste.coudurier@gmail.com>
Extrapolate audio timestamps based on the number of samples demuxed.
Deal with some MXF nastiness involving fractional number of
samples per EditUnit when seeking (the specs handwave this away).
Further fixes from Tomas Härdin.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The official Ut Video decoder only threads with slices, thus until
now any files encoded by the libavcodec encoder have only been
decodable with a single thread. The default slice count is now
set to subsampled_height / 120.
Also sets slices to 1 for the Ut Video encoder tests to keep them
green.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The old one didn't use segmentation. One uses segmentation in all frame
types (--aq-mode=1), and the other uses all segmentation features, but
only in inter frames (mbgraph).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This disables backward probability updates, which makes the codec more
friendly for frame-level multi-threading.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The original test without a forced idct is still useful since it tests
the switching of the idct algorithm/permutation on x86 with MMX. MMXext
or SSE2. Make sure the test runs only if MMX inline asm is available and
force -cpuflags to all.
Add the required bitexact flag for both tests.
They are not measurably faster on x86, they might be somewhat faster on
other platforms due to missing emu edge SIMD, but the gain is not large
enough to justify the added complexity.
They are not measurably faster on x86, they might be somewhat faster on
other platforms due to missing emu edge SIMD, but the gain is not large
enough to justify the added complexity.
The RGB32 pixel format is RGBA/BGRA depending on target
endianness - make sure to convert it to one specific format for
the framecrc tests.
This fixes the pngparser fate test on big endian.
Signed-off-by: Martin Storsjö <martin@martin.st>
The encoder uses almost none of the mpegvideo infrastructure, only some
fields from MpegEncContext.
The FATE results change because now an all-zero quant matrix is written
into the file. Since it is not used for anything for ljpeg, this should
not be a problem.
Originally written by Ronald S. Bultje <rsbultje@gmail.com> and
Clément Bœsch <u@pkh.me>
Further contributions by:
Anton Khirnov <anton@khirnov.net>
Diego Biurrun <diego@biurrun.de>
Luca Barbato <lu_zero@gentoo.org>
Martin Storsjö <martin@martin.st>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
A few fate instances on OS/2, OpenBSD, FreeBSD and IA64 linux currently
still fail a few tests with a maxdiff of 6.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes sync in some samples (e.g. bugs 7581 and 8374 in VLC).
Based on a commit by Matthieu Bouron <matthieu.bouron@gmail.com>
Reported-by: Jean-Baptiste Kempf <jb@videolan.org>
CC: libav-stable@libav.org
The element was only being written when the value == 1. But the default
value of this element is 1, so this has no useful effect. This element
needs to be written when the value == 0.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
QuickTime will play multiple audio tracks concurrently if this flag is
set for multiple audio tracks. And if no subtitle track has this flag
set, QuickTime will show no subtitles in the subtitle menu.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Update the fate reference since the last broken frame is not decoded
anymore.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Only check dependencies if invoking the make targets 'check'
or anything matching 'fate%' except 'fate-rsync'.
Signed-off-by: Martin Storsjö <martin@martin.st>
If building out of tree, make sure the filter scripts are copied
into the build tree before running tests. This makes sure that
SRC_PATH doesn't need to exist on the remote system (or doesn't
need to exist at the same path).
Signed-off-by: Martin Storsjö <martin@martin.st>
If the "build_only" variable is set in the configuration file, the
FATE client will skip running tests and just compile all targets.
Signed-off-by: Martin Storsjö <martin@martin.st>
This replaces a large number of checks for the second field by
fixing the pointers when they are setup.
This should also fix I/BI field pictures.
Changes checksums for vc1_sa10143, the file becomes slightly closer
to what the reference decoder outputs.
Based on "vc1dec: the second field is written wrong to the picture"
by Sebastian Sandberg <sebastiand.sandberg@gmail.com>.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes -t sample-accurate for audio and will allow further
simplication in the future.
Most of the FATE changes are due to audio now being sample accurate. In
some cases a video frame was incorrectly passed with the old code, while
its was over the limit.
