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* Revert "Revert "lavfi/buffersrc: push the frame deeper if requested.""Nicolas George2017-07-301-1/+1
| | | | | | | | | | | This reverts commit 04aa09c4bcf2d5a634a35da3a3ae3fc1abe30ef8 and reintroduces 0ff5567a30be6d7c804e95997ae282d6bacd76c3 that was temporarily reverted due to minor regressions. It also reverts e5bce8b4ce7b1f3a83998febdfa86a3771df96ce that fixed FATE refs. The fate-ffm change is caused by field_order now being set on the output format because the first frame arrives earlier. The fate-mxf change is assumed to be the same.
* fate: update checksums for fate-lavf-ffm and fate-lavf-mxfJames Almer2017-06-241-1/+1
| | | | | | | | | | | | | <@jamrial> durandal_1707: 04aa09c4bc broke fate-lavf-ffm and fate-lavf-mxf <@durandal_1707> how so? <@jamrial> one byte changes <@durandal_1707> jamrial: just update checksums <@jamrial> durandal_1707: but why did they change at all? the commit you reverted didn't affect them <@jamrial> why does reverting it affect these tests? <@jamrial> i don't think updating the checksum without knowing what changed is a good idea <@durandal_1707> jamrial: the lavfi core is in weird state after removal of recursive code <@durandal_1707> jamrial: the change is that older ones would get progressive flag set and new one doesnt <@jamrial> alright
* ffmpeg: init filtergraphs only after we have a frame on each inputAnton Khirnov2017-03-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* avformat/ffmenc: Make ffm_write_header_codec_ctx() use codecparMichael Niedermayer2016-12-021-1/+1
| | | | | | | | | This would be simpler if codecpar supported AVOptions modern ffserver should be unaffected by this, older ffserver which required the muxer to directly access the encoder could have issues with this, but this direct access is just wrong and unsafe Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avformat/ffmenc: set bitexact mode for old API without accessing the encoderMichael Niedermayer2016-12-021-1/+1
| | | | Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avformat/ffmenc: Drop ffm_write_header_codec_private_ctx()Michael Niedermayer2016-12-021-1/+1
| | | | | | | | | This accesses the private encoder context, it should not be used by the current ffserver it may affect old ffserver versions but i believe there is consens that accessing the private encoder context from the muxer is completely wrong. Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avcodec: Add bits_per_raw_sample to AVCodecParametersMichael Niedermayer2016-04-151-1/+1
| | | | | | | | | | | | | | | The bits_per_raw_sample represents the number of bits of precision per sample. The field is added at the logical place, not at the end as the code was just recently added This fixes the regression about losing the audio sample precision information The change in the fate test checksum un-does the change from the merge Previous version reviewed by: wm4 <nfxjfg@googlemail.com> Previous version reviewed by: Dominik 'Rathann' Mierzejewski <dominik@greysector.net> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* Merge commit '9200514ad8717c63f82101dc394f4378854325bf'Derek Buitenhuis2016-04-101-1/+1
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * commit '9200514ad8717c63f82101dc394f4378854325bf': lavf: replace AVStream.codec with AVStream.codecpar This has been a HUGE effort from: - Derek Buitenhuis <derek.buitenhuis@gmail.com> - Hendrik Leppkes <h.leppkes@gmail.com> - wm4 <nfxjfg@googlemail.com> - Clément Bœsch <clement@stupeflix.com> - James Almer <jamrial@gmail.com> - Michael Niedermayer <michael@niedermayer.cc> - Rostislav Pehlivanov <atomnuker@gmail.com> Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
| * Remove avserver.Anton Khirnov2014-06-181-3/+0
| | | | | | | | | | | | | | 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.
| * avconv: make -t insert trim/atrim filters.Anton Khirnov2013-04-301-2/+2
| | | | | | | | | | | | | | | | | | 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.
* | fate/ffm: Update test refDerek Buitenhuis2016-01-311-1/+1
| | | | | | | | | | | | | | Since timecode_frame)start is a private option now, it stays at the default, and is no longer written to the file. Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* | tests/fate/avformat: Fix fate-lavfMichael Niedermayer2015-11-101-1/+1
| | | | | | | | | | | | | | | | The CMP variable seems to have been inherited from fate-api-seek which set it to null the mxf reference needed a change due to c7e14a279fa7348db10ec824bb2d67858cb1c1ca Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* | lavf/ffm: use AVOption API to store/restore stream propertiesLukasz Marek2014-11-161-1/+1
| | | | | | | | | | | | | | | | | | | | | | This is a generic solution that will not reqiore modifications when new options are added. This also fixes problem with current implementation when qmin or qmax=-1. Only 8 bits was sent and read back as 255. Fixes #1275 Fixes #1461 Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
* | mpeg12enc: always set closed gop flag on the first gopMarton Balint2014-04-011-1/+1
| | | | | | | | | | | | | | | | Improves compatibility with XDCAM HD formats. It has been set for a long time in ffmbc. Reviewed-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Marton Balint <cus@passwd.hu>
* | ffmpeg: remove obsolete workaround in trim insertion.Nicolas George2013-08-071-2/+2
| | | | | | | | | | | | | | | | The bug it was working seems to have been fixed. This change causes ffmpeg to use the trim filter to implement the -t option. FATE tests are updated due to the more accurate handling of the last packets.
* | ffm: redesign header format to make it extensibleMichael Niedermayer2012-11-051-1/+1
| | | | | | | | | | | | | | | | | | Currently FFM files generated with one versions of ffmpeg generally cannot be read by another. By spliting data into chunks, more fields can saftely be appended to chunks as well as new chunks added. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-10-071-1/+1
|\ \ | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: wmaenc: use float planar sample format (e)ac3enc: use planar sample format aacenc: use planar sample format adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt adpcmenc: move 'ch' variable to higher scope adpcmenc: fix 3 instances of variable shadowing adpcm_ima_wav: simplify encoding libvorbis: use planar sample format libmp3lame: use planar sample formats vorbisenc: use float planar sample format ffm: do not write or read the audio sample format parseutils: fix parsing of invalid alpha values doc/RELEASE_NOTES: update for the 9 release. smoothstreamingenc: Add a more verbose error message smoothstreamingenc: Ignore the return value from mkdir smoothstreamingenc: Try writing a manifest when opening the muxer smoothstreamingenc: Move the output_chunk_list and write_manifest functions up smoothstreamingenc: Properly return errors from ism_flush to the caller smoothstreamingenc: Check the output UrlContext before accessing it Conflicts: doc/RELEASE_NOTES libavcodec/aacenc.c libavcodec/ac3enc_template.c libavcodec/wmaenc.c tests/ref/lavf/ffm Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * ffm: do not write or read the audio sample formatJustin Ruggles2012-10-061-1/+1
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* | mpegvideo_enc: reduce QMAT_SHIFT to avoid overflow in dnxhdMichael Niedermayer2012-09-271-2/+2
| | | | | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | ffmpeg: add support for audio filters.Anton Khirnov2012-05-171-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some of the FATE changes are due to off-by-one different rounding being used (lrintf vs av_rescale_q). Some fate changes are due to 1 audio frame less being encoded (the new variant seems matching what qatar does and according to ffprobe its closer to the requested duration) the mapchan feature sadly is lost in this commit because it depends on resampling being done in ffmpeg.c which is now moved completely into the av filter layer -async is broken after this commit, this will be fixed in subsequent commits the new filter reconfiguration system is flawed and will drop a frame on each parameter change which is why the nelly moser checksums need updating. Conflicts: ffmpeg.c tests/ref/fate/smjpeg
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-221-1/+1
|\ \ | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (26 commits) adxenc: use AVCodec.encode2() adxenc: Use the AVFrame in ADXContext for coded_frame indeo4: fix out-of-bounds function call. configure: Restructure help output. configure: Internal-only components should not be command-line selectable. vorbisenc: use AVCodec.encode2() libvorbis: use AVCodec.encode2() libopencore-amrnbenc: use AVCodec.encode2() ra144enc: use AVCodec.encode2() nellymoserenc: use AVCodec.encode2() roqaudioenc: use AVCodec.encode2() libspeex: use AVCodec.encode2() libvo_amrwbenc: use AVCodec.encode2() libvo_aacenc: use AVCodec.encode2() wmaenc: use AVCodec.encode2() mpegaudioenc: use AVCodec.encode2() libmp3lame: use AVCodec.encode2() libgsmenc: use AVCodec.encode2() libfaac: use AVCodec.encode2() g726enc: use AVCodec.encode2() ... Conflicts: configure libavcodec/Makefile libavcodec/ac3enc.c libavcodec/adxenc.c libavcodec/libgsm.c libavcodec/libvorbis.c libavcodec/vorbisenc.c libavcodec/wmaenc.c tests/ref/acodec/g722 tests/ref/lavf/asf tests/ref/lavf/ffm tests/ref/lavf/mkv tests/ref/lavf/mpg tests/ref/lavf/rm tests/ref/lavf/ts tests/ref/seek/lavf_asf tests/ref/seek/lavf_ffm tests/ref/seek/lavf_mkv tests/ref/seek/lavf_mpg tests/ref/seek/lavf_rm tests/ref/seek/lavf_ts Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * mpegaudioenc: use AVCodec.encode2()Justin Ruggles2012-03-201-2/+2
| | | | | | | | Update FATE references due to encoder delay.
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-011-1/+1
|\ \ | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (58 commits) amrnbdec: check frame size before decoding. cscd: use negative error values to indicate decode_init() failures. h264: prevent overreads in intra PCM decoding. FATE: do not decode audio in the nuv test. dxa: set audio stream time base using the sample rate psx-str: do not allow seeking by bytes asfdec: Do not set AVCodecContext.frame_size vqf: set packet parameters after av_new_packet() mpegaudiodec: use DSPUtil.butterflies_float(). FATE: add mp3 test for sample that exhibited false overreads fate: add cdxl test for bit line plane arrangement vmnc: return error on decode_init() failure. libvorbis: add/update error messages libvorbis: use AVFifoBuffer for output packet buffer libvorbis: remove unneeded e_o_s check libvorbis: check return values for functions that can return errors libvorbis: use float input instead of s16 libvorbis: do not flush libvorbis analysis if dsp state was not initialized libvorbis: use VBR by default, with default quality of 3 libvorbis: fix use of minrate/maxrate AVOptions ... Conflicts: Changelog doc/APIchanges libavcodec/avcodec.h libavcodec/dpxenc.c libavcodec/libvorbis.c libavcodec/vmnc.c libavformat/asfdec.c libavformat/id3v2enc.c libavformat/internal.h libavformat/mp3enc.c libavformat/utils.c libavformat/version.h libswscale/utils.c tests/fate/video.mak tests/ref/fate/nuv tests/ref/fate/prores-alpha tests/ref/lavf/ffm tests/ref/vsynth1/prores tests/ref/vsynth2/prores Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * mpegvideo_enc: add chroma/luma_elim_threshold private options.Anton Khirnov2012-02-291-1/+1
| | | | | | | | Deprecate corresponding AVCodecContext fields.
| * avconv: rework -t handling for encoding.Anton Khirnov2012-02-071-2/+2
|/ | | | | | | | | | | | | 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.
* 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.
* libx264: add 'direct-pred' private optionAnton Khirnov2011-09-071-1/+1
| | | | Deprecate AVCodecContext.directpred
* libx264: add 'partitions' private optionAnton Khirnov2011-09-071-1/+1
| | | | Deprecate AVCodecContext.partitions.
* lavf: deprecate AVStream.quality.Anton Khirnov2011-07-061-1/+1
| | | | AVStream is no place for it and it's unused outside of ffmpeg anyway.
* 10l, update ref value for ffm since default flags changed and are stored in ↵Baptiste Coudurier2010-03-251-1/+1
| | | | | | the file Originally committed as revision 22673 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Place regression test output files in subdirs per familyMåns Rullgård2010-03-021-3/+3
| | | | Originally committed as revision 22155 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Correct opts calulation in ffmpeg.c.Michael Niedermayer2010-02-031-2/+2
| | | | | | | | | | This correct the stop point for demuxing with -vcodec copy and -t as well as packet interleaving. (we already diddrop packets but kept demuxing them for too long due to opts being wrong) the change to ffm is due to 2 packets with timestamp 0 being stored in different order. Originally committed as revision 21626 to svn://svn.ffmpeg.org/ffmpeg/trunk
* regtest: split reference files allowing tests to run individuallyMåns Rullgård2010-01-161-0/+3
With this change, the output is checked immediately after each test has run. This means commands like "make regtest-mpeg2" can now be used to run a single test and get meaningful results. By default, make will abort if any test fails. To run all tests regardless, use make -k. Originally committed as revision 21254 to svn://svn.ffmpeg.org/ffmpeg/trunk
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