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* FATE: add a test for the SVQ1 header byte swappingAnton Khirnov2015-02-15
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* fate: Add a test for AAC ELD480.Alex Converse2015-02-06
| | | | The source is er_eld_2100np_48_ep0.mp4 from the official test set.
* nutenc: do not use has_b_framesAnton Khirnov2015-01-27
| | | | | It is unreliable, especially when the stream codec context is not the encoding context. Use the codec descriptor properties instead.
* swscale: fix gbrap to gbrap alpha scalingVittorio Giovara2015-01-16
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* libswscale: GBRAP input & output and GBRAP16 input supportPaul B Mahol2015-01-14
| | | | Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* fate: Use bitexact conversions in the dpxparser testMartin Storsjö2014-12-29
| | | | | | This fixes fate on e.g. ppc. Signed-off-by: Martin Storsjö <martin@martin.st>
* fate: add test for offset theoraVittorio Giovara2014-12-18
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* fate: add a DPX parser testVittorio Giovara2014-12-18
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* fate: add dolby true hd testsJanne Grunau2014-12-09
| | | | | | | | 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.
* Allow reading of growing avi files (currently being written)Joakim Plate2014-12-08
| | | | | | | | | | | | | 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>
* dpx: always write the image offsetZoltan Kovacs2014-12-08
| | | | | | | | | | | | | 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>
* v210enc: Add SIMD optimised 8-bit and 10-bit encodersKieran Kunhya2014-12-05
| | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* swscale: Properly scale YUVDerek Buitenhuis2014-12-05
| | | | | | | | | | 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>
* Replace lena.pnmReinhard Tartler2014-11-28
| | | | | | 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
* vc1dec: refactor check with missing parenthesisVittorio Giovara2014-11-13
| | | | | | | FATE tests updated accordingly, no visual difference. CC: libav-stable@libav.org Bug-Id: CID 1087087
* fate-mpeg4: use TARGET_SAMPLES for resize testsJanne Grunau2014-10-21
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* fate: add mpeg4 tests for frame size changesJanne Grunau2014-10-17
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* lavf: Don't drop both pts and dts if timestamps are invalidMichael Niedermayer2014-10-12
| | | | | | | | | 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>
* fate-vc1_ilaced_twomv: use -flags +bitexactJanne Grunau2014-10-04
| | | | | Also updates the reference since it was generated by the non-bitexact x86 specific code.
* fate: Add VC-1 interlaced twomv testTim Walker2014-09-26
| | | | CC: libav-stable@libav.org
* fate: Switch ra4-288 test from framecrc() to pcm()Katerina Barone-Adesi2014-09-25
| | | | | | The decoder is float-based and the test needs to allow for some fuzz. Signed-off-by: Diego Biurrun <diego@biurrun.de>
* fate: Add tests for RealAudio 1.0 (14.4) and RealAudio 2.0 (with 28.8)Katerina Barone-Adesi2014-09-17
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* xvid: Add C IDCTPascal Massimino2014-09-02
| | | | | | Thanks to Pascal Massimino and Michael Militzer for relicensing as LGPL. Signed-off-by: Diego Biurrun <diego@biurrun.de>
* electronicarts: set the framerate for TGQ/TQIAnton Khirnov2014-08-22
| | | | It is hardcoded to 15fps.
* electronicarts: read the framerate for MADAnton Khirnov2014-08-22
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* cosmetics: Write NULL pointer inequality checks more compactlyGabriel Dume2014-08-15
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* fate: generate tests/pixfmts.mak for all targets requiring itJanne Grunau2014-08-05
| | | | | | All subtargets which should run the fate-filter-pixdesc% need to generate and include tests/pixfmts.mak. Most noteable missing target was fate itself.
* fate: Only generate tests/pixfmts.mak if some pixfmts fate test is runDiego Biurrun2014-08-04
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* fate: png-suiteVittorio Giovara2014-08-04
| | | | Every supported format is converted to RGB.
* fate: explicitly set the default THREADS valueJanne Grunau2014-08-04
| | | | | | 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.
* cpu-test: test av_cpu_countJanne Grunau2014-08-04
| | | | Add CPU count and number threads as informative values for fate.
* fate: add informative cpu testJanne Grunau2014-08-04
| | | | | | | | 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.
* fate: Split fate-pixdesc tests and dispatch them through MakeDiego Biurrun2014-08-01
| | | | This allows running all the tests individually and/or in parallel.
* fate: support testing of release branchesJanne Grunau2014-07-27
| | | | | | | | | | | | | | | | | 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.
* fate: Split audio- and video-related RealMedia targetsKaterina Barone-Adesi2014-07-21
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* build: Conditionally build and run DCT test programDiego Biurrun2014-07-21
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* build: Conditionally build and test iirfilterDiego Biurrun2014-07-21
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* fate: add h264-direct-bff testVittorio Giovara2014-07-20
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* fate: Use the correct, local path to samples for opus reference filesMartin Storsjö2014-07-14
| | | | | | | 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>
* fate: Add dependencies for dct/fft/mdct/rdft testsDiego Biurrun2014-06-26
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* lavf: switch to AVStream.time_base as the hint for the muxer timebaseAnton Khirnov2014-06-18
| | | | | | | | | | | | 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).
* Remove avserver.Anton Khirnov2014-06-18
| | | | | | | 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.
* flvenc: Don't over-count metadata.Josh Allmann2014-06-17
| | | | | | Over-counting occurs if framerate is not set. Signed-off-by: Anton Khirnov <anton@khirnov.net>
* fate: add on2avc audio testVittorio Giovara2014-06-10
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* mpeg2: add sequence display extension informationMarc-Antoine Arnaud2014-06-06
| | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* matroskaenc: set the stream timebase earlierAnton Khirnov2014-05-29
| | | | Fixes calculating the ts offset for audio codecs with delay.
* flvenc: only write the framerate tag based on avg_frame_rateAnton Khirnov2014-05-29
| | | | | Do not fall back on the codec timebase, since that can be anything for VFR video.
* matroskaenc: base DefaultDuration on the framerate, not the codec timebaseAnton Khirnov2014-05-29
| | | | | | 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.
* oggenc: Fix the EOS flagMichael Niedermayer2014-05-28
| | | | | | | | | | | 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>
* oggenc: Flush after writing headersAndrew Kelley2014-05-27
| | | | | | | | | | | | | 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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