- 25 Oct, 2014 1 commit
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Christophe Gisquet authored
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>
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- 07 Aug, 2013 1 commit
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Nicolas George authored
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.
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- 08 Jan, 2013 1 commit
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Paul B Mahol authored
Signed-off-by:
Paul B Mahol <onemda@gmail.com>
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- 04 Jan, 2013 1 commit
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Alexander Strasser authored
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>
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- 04 Jul, 2012 1 commit
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Nicolas George authored
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.
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- 18 May, 2012 1 commit
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Michael Niedermayer authored
This fixes a regression that apparently was missed when switching to the in af resampler Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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- 17 May, 2012 1 commit
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Anton Khirnov authored
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
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- 24 Apr, 2012 1 commit
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Justin Ruggles authored
This partially reverts acb17302 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.
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- 20 Apr, 2012 1 commit
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Justin Ruggles authored
Change some lavf tests to avoid resampling and channel mixing.
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- 02 Feb, 2012 1 commit
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Clément Bœsch authored
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- 19 Sep, 2011 2 commits
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Michael Niedermayer authored
Signed-off-by:
Michael Niedermayer <michaelni@gmx.at>
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Michael Niedermayer authored
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>
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- 13 Aug, 2011 1 commit
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Clément Bœsch authored
This is based on the original work by Baptiste Coudurier.
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- 02 Mar, 2010 1 commit
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Måns Rullgård authored
Originally committed as revision 22155 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 16 Jan, 2010 1 commit
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Måns Rullgård authored
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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