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author | Anton Khirnov <anton@khirnov.net> | 2012-05-08 16:33:50 +0200 |
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committer | Anton Khirnov <anton@khirnov.net> | 2012-05-14 21:36:11 +0200 |
commit | 9f26421b0be2af36b5405608f4e7429b4bd7fbdb (patch) | |
tree | d2ab8b7edc77fd7d0cb6be7e8956982d2ffb7e06 /doc | |
parent | fb604ae8500d4ee7de6af61387c11618b3dea25b (diff) | |
download | ffmpeg-9f26421b0be2af36b5405608f4e7429b4bd7fbdb.tar.gz |
lavfi: add asyncts filter.
Diffstat (limited to 'doc')
-rw-r--r-- | doc/filters.texi | 19 |
1 files changed, 19 insertions, 0 deletions
diff --git a/doc/filters.texi b/doc/filters.texi index f066657add..0e611d2793 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -137,6 +137,25 @@ aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo Pass the audio source unchanged to the output. +@section asyncts +Synchronize audio data with timestamps by squeezing/stretching it and/or +dropping samples/adding silence when needed. + +The filter accepts the following named parameters: +@table @option + +@item compensate +Enable stretching/squeezing the data to make it match the timestamps. + +@item min_delta +Minimum difference between timestamps and audio data (in seconds) to trigger +adding/dropping samples. + +@item max_comp +Maximum compensation in samples per second. + +@end table + @section resample Convert the audio sample format, sample rate and channel layout. This filter is not meant to be used directly, it is inserted automatically by libavfilter |