| Commit message (Collapse) | Author | Age |
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It is unreliable, especially when the stream codec context is not the
encoding context. Use the codec descriptor properties instead.
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Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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This fixes fate on e.g. ppc.
Signed-off-by: Martin Storsjö <martin@martin.st>
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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.
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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>
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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>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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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>
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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
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FATE tests updated accordingly, no visual difference.
CC: libav-stable@libav.org
Bug-Id: CID 1087087
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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>
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Also updates the reference since it was generated by the non-bitexact
x86 specific code.
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CC: libav-stable@libav.org
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The decoder is float-based and the test needs to allow for some fuzz.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Thanks to Pascal Massimino and Michael Militzer for relicensing as LGPL.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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It is hardcoded to 15fps.
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Signed-off-by: Diego Biurrun <diego@biurrun.de>
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All subtargets which should run the fate-filter-pixdesc% need to
generate and include tests/pixfmts.mak. Most noteable missing target was
fate itself.
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Every supported format is converted to RGB.
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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.
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Add CPU count and number threads as informative values for fate.
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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.
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This allows running all the tests individually and/or in parallel.
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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.
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Signed-off-by: Diego Biurrun <diego@biurrun.de>
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This fixes running fate in configs where the samples are located
in a different path on the target.
Signed-off-by: Martin Storsjö <martin@martin.st>
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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).
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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.
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Over-counting occurs if framerate is not set.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Fixes calculating the ts offset for audio codecs with delay.
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Do not fall back on the codec timebase, since that can be anything for
VFR video.
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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.
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This corrects the bug that caused the checksums to change in
9767d7c092c890ecc5953452e8a951fd902dd67b.
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>
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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>
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This seems to be needed on some 32bit arches.
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