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* lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()Justin Ruggles2012-03-052-20/+21
| | | | | | | | | | Also, do not give AVCodecContext.frame_size priority for muxing. Updated 2 FATE references: dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified by -t 2 in the FATE test wmv8-drm-nodec - durations are not needed. previously they were estimated using the packet size and average bit rate.
* aiffdec: do not set AVCodecContext.frame_sizeJustin Ruggles2012-03-051-18/+18
| | | | | | | | | It is unnecessary. Also, for some codecs we're reading more than 1 frame per packet. Instead we use a private context variable to calculate the bit rate, stream duration, and packet durations. Updated FATE seek test, which has slightly different timestamps due to a more accurate bit rate calculation.
* lavf: deobfuscate read_frame_internal().Anton Khirnov2012-03-052-2/+2
| | | | | | | | | | | | | | Split off packet parsing into a separate function. Parse full packets at once and store them in a queue, eliminating the need for tracking parsing state in AVStream. The horrible unreadable loop in read_frame_internal() now isn't weirdly ordered and doesn't contain evil gotos, so it should be much easier to understand. compute_pkt_fields() now invents slightly different timestamps for two raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't be more wrong (or right) than previous ones.
* fate: Add sunrast regression testDerek Buitenhuis2012-03-031-0/+3
| | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* wmaenc: fix m/s stereo encoding for the first frameJustin Ruggles2012-03-032-6/+6
| | | | | | | | We need to set ms_stereo in encode_init() in order to avoid incorrectly encoding the first frame as non-m/s while flagging it as m/s. Fixes an uncomfortable pop in the left channel at the start of playback. CC:libav-stable@libav.org
* tiertexseq: set audio stream start time to 0Justin Ruggles2012-03-031-21/+21
| | | | | Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.
* voc/avs: Do not change the sample rate mid-stream.Justin Ruggles2012-03-031-12/+12
| | | | | Also, set the time base based on the sample rate. lavf-voc seek test updated to reflect slightly different seek points.
* vqf: set packet durationJustin Ruggles2012-03-031-1/+1
| | | | | | Fixes timestamp calculation. The FATE reference is updated because timestamp calculations are now more accurate. Previous timestamps were based on average bit rate.
* mpegaudio_parser: do not ignore information from the first parsed frameJustin Ruggles2012-03-033-18/+14
| | | | Update some demuxing and seeking fate tests.
* g722: Fix the QMF scalingMartin Storsjö2012-03-023-172/+172
| | | | | | | | | | | | | | | This fixes clipping if the encoder input used the full 16 bit input range (samples with a magnitude below 16383 worked fine). The filtered subband samples should be 15 bit maximum, while the code earlier produced them scaled to 16 bit. This makes the decoder output have double the magnitude compared to before. The spec reference samples doesn't test the QMF at all, which was why this part slipped past initially. Signed-off-by: Martin Storsjö <martin@martin.st>
* electronicarts: set timebase for tgv video.Anton Khirnov2012-03-022-87/+87
| | | | | | | | The container has no timestamps and the framerate isn't stored in the data either. The decoder sets codec timebase to experimentally found value 1/15. Do the same for the demuxer too, it should at least be better than the default 1/90000.
* electronicarts: parse the framerate for cmv video.Anton Khirnov2012-03-021-195/+195
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* electronicarts: don't set codec timebaseAnton Khirnov2012-03-023-390/+390
| | | | | | Demuxers are not supposed to set it. Set stream timebase and framerates instead (this is a cfr container with no timestamps).
* FATE: do not decode audio in the nuv test.Justin Ruggles2012-02-291-19/+0
| | | | We already have sufficient coverage for 16-bit pcm.
* fate: add cdxl test for bit line plane arrangementPaul B Mahol2012-02-291-0/+11
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* dpxenc: Don't include the libavcodec ident if bitexact mode is enabledMartin Storsjö2012-02-291-1/+1
| | | | | | This avoids breaking fate every time the lavc version is bumped. Signed-off-by: Martin Storsjö <martin@martin.st>
* prores: store and retrieve extended colourspace informationKostya Shishkov2012-02-292-2/+2
| | | | Based on the patch by Phil Barrett.
* prores: handle 444 chroma in right orderKostya Shishkov2012-02-291-2/+2
| | | | | | | ProRes codes chroma blocks in 444 mode in different order than luma blocks, so make both decoder and encoder read/write chroma blocks in right order. Reported by Phil Barrett
* mpegvideo_enc: add chroma/luma_elim_threshold private options.Anton Khirnov2012-02-291-1/+1
| | | | Deprecate corresponding AVCodecContext fields.
* fate: Overhaul WavPack coverageDerek Buitenhuis2012-02-2727-1/+26
| | | | | | | | WavPack has a comprehensive test suite, and a bunch of corner cases. Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* lavf: don't guess r_frame_rate from either stream or codec timebase.Anton Khirnov2012-02-261-1/+1
| | | | | | | Neither of those is guaranteed to be connected to framerate in any way (if it even exists). Fixes bug 56.
* avconv: saner output video timebase.Anton Khirnov2012-02-2651-3775/+3778
| | | | | | | | | | r_frame_rate should in theory have something to do with input framerate, but in practice it is often made up from thin air by lavf. So unless we are targeting a constant output framerate, it's better to just use input stream timebase. Brings back dropped frames in nuv and cscd tests introduced in cd1ad18a6539bd7fc2dc4c1740fbcbd498c0c0a2
* rawdec: set timebase to 1/fps.Anton Khirnov2012-02-2610-125/+125
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* fate: add tests for cdxl videoPaul B Mahol2012-02-254-0/+78
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* lavf: don't set AVCodecContext.has_b_frames in compute_pkt_fields().Anton Khirnov2012-02-221-1/+1
| | | | | | | | | | | It is not supposed to be done outside lavc. This is basically a revert of 818062f2f346df30f4ec0c0c1f54e8025cc3a80a. It is unclear what issue this was supposed to fix, if it reappears again it will have to be fixed in a more proper place. The wtv-demux test change is because the sample starts with a B-frame.
* Add prores regression test.Reimar Döffinger2012-02-212-0/+8
| | | | | Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* Enable already existing rso regression test.Reimar Döffinger2012-02-211-0/+3
| | | | | Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* Add regression test for "sox" format muxer/demuxer.Reimar Döffinger2012-02-211-0/+3
| | | | | Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* Add dpx encoding regression test.Carl Eugen Hoyos2012-02-211-0/+3
| | | | Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* adpcmenc: Use correct frame_size for Yamaha ADPCM.Justin Ruggles2012-02-201-4/+4
| | | | | | | | | | | Output packet size should match avctx->block_align. The target output packet size is 1024 bytes. Before: mono - 1024 samples -> 512 bytes stereo - 2048 samples -> 2048 bytes After: mono - 2048 samples -> 1024 bytes stereo - 1024 samples -> 1024 bytes
* bethsoftvid: synchronize video timestamps with audio sample rateJustin Ruggles2012-02-161-70/+71
| | | | | | | | According to unofficial documentation, the video rate is locked to the audio sample rate. This results in proper synchronization of audio and video timestamps from the demuxer. This only works if the first audio packet occurs before the first video packet or the audio sample rate is the default rate of 11111 Hz, both of which are true for all samples in our archive.
* bethsoftvid: Set video packet duration instead of accumulating pts.Justin Ruggles2012-02-161-21/+20
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* bethsoftvid: pass palette in side data instead of in a separate packet.Justin Ruggles2012-02-161-70/+71
| | | | | | Update FATE reference to account for now non-existent palette packet. This also fixes the FATE test if frame data is not initialized in get_buffer(), so update comment in avconv accordingly.
* sunrast: Add fate test for gray8.Aneesh Dogra2012-02-131-0/+2
| | | | Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* cljr: set the properties of the coded_frame, not input frame.Anton Khirnov2012-02-122-2/+2
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* FATE: update reference for seek-alac_mp4Justin Ruggles2012-02-111-27/+27
| | | | This should have been updated in b590f3a7bf9103ac7a7a61c48568676201d6824b.
* alacenc: only encode frame size in header for a final smaller frameJustin Ruggles2012-02-111-2/+2
| | | | | Otherwise it is not needed because it matches the frame size as encoded in the extradata.
* rawenc: switch to encode2().Anton Khirnov2012-02-0868-4483/+4479
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | This changes a number of FATE results, since before this commit, the timestamps in all tests using rawenc were made up by lavf. In most cases, the previous timestamps were completely bogus. In some other cases -- raw formats, mostly h264 -- the new timestamps are bogus as well. The only difference is that timestamps invented by the muxer are replaced by timestamps invented by the demuxer. cscd -- avconv sets output codec timebase from r_frame_rate and r_frame_rate is in this case some guessed number 31.42 (377/12), which is not accurate enough to represent all timestamps. This results in some frames having duplicate pts. Therefore, vsync 0 needs to be changed to vsync 2 and avconv drops two frames. A proper fix in the future would be to set output timebase to something saner in avconv. nuv -- previous timestamps for video were wrong AND the cscd comment applies, one frame is dropped. vp8-signbias -- the file contains two frames with identical timestamps, so -vsync 0 needs to be removed/changed to -vsync 2 and avconv drops one frame. vc1-ism -- apparrently either the demuxer lies about timestamps or the file is broken, since dts == pts on all packets, but reordering clearly takes place.
* avconv: rework -t handling for encoding.Anton Khirnov2012-02-0715-40/+45
| | | | | | | | | | | | | Current code compares the desired recording time with InputStream.pts, which has a very unclear meaning. Change the code to use actual timestamps of the frames passed to the encoder. In several tests, one less frame is encoded, which is more correct. In the idroq test one more frame is encoded, which is again more correct. Behavior with stream copy should be unchanged.
* pva-demux test: add -vnAnton Khirnov2012-02-071-27/+26
| | | | | | The output is obviously not supposed to contain video (since only -acodec copy is specified), but that only happens because of the way -t handling is implemented currently.
* frame{crc/md5}: set the stream timebase from codec timebase.Anton Khirnov2012-02-03393-33598/+34039
| | | | | | | | | | | | | Right now those muxers use the default timebase in all cases(1/90000). This patch avoid unnecessary rescaling and makes the printed timestamps more readable. Also, extend the printed information to include the timebases and packet pts/duration and align the columns. Obviously changes the results of all fate tests which use those two muxers.
* fate: make acodec-ac3_fixed test output raw AC3Mans Rullgard2012-02-023-46/+51
| | | | | | There is no point in this test using the RM format. Signed-off-by: Mans Rullgard <mans@mansr.com>
* bethsoftvideo: fix palette reading.Justin Ruggles2012-02-011-71/+70
| | | | | | Return the correct number of consumed bytes and set *data_size = 0. Returned size is 1 too small, leading to that 1 byte being read as the next frame, which results in an extra blank frame at the beginning of the stream.
* fate-golomb: extend golomb-test to get_ue_golomb_long()Janne Grunau2012-02-011-2/+0
| | | | | | | | get_ue_golomb_long() is only tested for values up to 2^15 - 2 since we can not write larger values. Silence the test on success and return a non-zero value on error. Use an heap scratch buffer instead of large stack buffer. Remove unneeded includes.
* fate: add golomb-testDiego Biurrun2012-01-311-0/+2
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* lavc: increase major version to 54.Anton Khirnov2012-01-271-1/+1
| | | | | The lavf-ffm test results change because ffmenc writes AVCodecContext.flags/flags2 and the defaults for those change.
* Revert "sws/pixfmt/pixdesc: add support for yuva444p"Alex Converse2012-01-265-5/+0
| | | | | | This reverts commit fc115c80b7bf6f1f5a937ee7f6e14630d99eb4dd. Tests are broken.
* sws/pixfmt/pixdesc: add support for yuva444pPaul B Mahol2012-01-275-0/+5
| | | | Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* fate: add XWD image regression testPaul B Mahol2012-01-261-0/+3
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* mxfdec: hybrid demuxing/seeking solutionJanne Grunau2012-01-222-50/+45
| | | | | | | | | | | | | | | | | | This uses the old demuxing code for OP1a and separate demuxing code for OPAtom. Timestamp output is added to the old demuxing code. The seeking code is made to seek to the start of the desired EditUnit only, from which the normal demuxing code takes over (if OP1a). This means we do not use delta entries or slices, only StreamOffsets. OPAtom seeking basically works like before. This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other words, we allow seeking before the start or past the end for D-10 too. Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and Reimar Döffinger <Reimar.Doeffinger@gmx.de>. Changed av_calloc to av_mallocz, added overflow checks.
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