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* dv: more precise weight table for 8x8Christophe Gisquet2014-10-251-3/+3
| | | | | | | | | It is derived from the actual equations of the specs. In particular, it is closer to the inverse of what the encoder uses. fate tests accordingly updated. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* ffmpeg: remove obsolete workaround in trim insertion.Nicolas George2013-08-071-3/+3
| | | | | | | | 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.
* fate: upate after 55d32eed8fPaul B Mahol2013-01-081-2/+2
| | | | Signed-off-by: Paul B Mahol <onemda@gmail.com>
* lswr: Improve default resampler's default parametersAlexander Strasser2013-01-041-4/+4
| | | | | | | | | | | | | | After making some blind tests on a small collection of music samples for home usage. It turned out that the default cutoff was too low. The impact of filter_size was not clearly distinguishable (the results were on the edge) with the music samples but turned out to be clearly audible in some synthetic samples. Thanks to Daniel for helping out with the listening tests. Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
* ffmpeg: rework checks for the -t option.Nicolas George2012-07-041-3/+3
| | | | | | | | | | | This commit is based on libav's implementation and makes sure to compare output timestamps together. It also reduces the differences with avconv. The changes to the test reference files are caused by an additional packet at the end, the timestamp of the frame encoded by this packet is always strictly below the limit stated by the -t option.
* af_aresample: fix rounding that led to sample accumulation in the buffers.Michael Niedermayer2012-05-181-3/+3
| | | | | | | This fixes a regression that apparently was missed when switching to the in af resampler Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* ffmpeg: add support for audio filters.Anton Khirnov2012-05-171-3/+3
| | | | | | | | | | | | | | | | | 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 Niedermayer2012-04-251-2/+2
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: FATE: use updated reference for aac-latm_stereo_to_51 avconv: use libavresample Add libavresample FATE: avoid channel mixing in lavf-dv_fmt Conflicts: Changelog Makefile cmdutils.c configure doc/APIchanges ffmpeg.c tests/lavf-regression.sh tests/ref/lavf/dv_fmt tests/ref/seek/lavf_dv Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * FATE: avoid channel mixing in lavf-dv_fmtJustin Ruggles2012-04-241-3/+3
| | | | | | | | | | | | | | This partially reverts acb1730218f1c614dc8ca3ba45d9de1e05059515 which would only have needed to change the checksums if channel mixing had been properly avoided. This changes the output file size reference and the seek test reference back to the previous values.
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-04-201-2/+2
|\ \ | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: avcodec: add a cook parser to get subpacket duration FATE: allow lavf tests to alter input parameters FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test FATE: replace the acodec-g726 test with 4 new encode/decode tests FATE: replace current g722 encoding tests with an encode/decode test FATE: add a pattern rule for generating asynth wav files FATE: optionally write a WAVE header in audiogen avutil: add audio fifo buffer Conflicts: doc/APIchanges libavcodec/version.h libavutil/avutil.h tests/Makefile tests/codec-regression.sh tests/fate/voice.mak tests/lavf-regression.sh tests/ref/acodec/g722 tests/ref/acodec/g726 tests/ref/acodec/pcm_s24daud tests/ref/lavf/dv_fmt tests/ref/lavf/gxf tests/ref/lavf/mxf tests/ref/lavf/mxf_d10 tests/ref/seek/lavf_dv Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * FATE: allow lavf tests to alter input parametersJustin Ruggles2012-04-201-3/+3
| | | | | | | | Change some lavf tests to avoid resampling and channel mixing.
* | timecode: add write regressions tests.Clément Bœsch2012-02-021-0/+6
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* | rematrix: dont use floats for int16 code.Michael Niedermayer2011-09-191-2/+2
| | | | | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Add libswresample.Michael Niedermayer2011-09-191-2/+2
| | | | | | | | | | | | | | | | | | Similar to libswscale this does resampling and format convertion, just for audio instead of video. changing sampling rate, sample formats, channel layouts and sample packing all in one with a very simple public interface. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | dv: honor timecode in DV muxer.Clément Bœsch2011-08-131-1/+1
|/ | | | This is based on the original work by Baptiste Coudurier.
* 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
* 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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