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Diffstat (limited to 'DOCS/man/af.rst')
| -rw-r--r-- | DOCS/man/af.rst | 69 |
1 files changed, 30 insertions, 39 deletions
diff --git a/DOCS/man/af.rst b/DOCS/man/af.rst index 4e806d482e..5ff7426382 100644 --- a/DOCS/man/af.rst +++ b/DOCS/man/af.rst @@ -24,44 +24,6 @@ See ``--vf`` group of options for info on how ``--af-defaults``, ``--af-add``, Available filters are: -``lavrresample[=option1:option2:...]`` - This filter uses libavresample (or libswresample, depending on the build) - to change sample rate, sample format, or channel layout of the audio stream. - This filter is automatically enabled if the audio output does not support - the audio configuration of the file being played. - - .. warning:: - - Deprecated. Either use the ``--audio-resample-...`` options to customize - resampling, or the libavfilter ``--af=aresample`` filter, which has its - own options. - - It supports only the following sample formats: u8, s16, s32, float. - - ``filter-size=<length>`` - Length of the filter with respect to the lower sampling rate. (default: - 16) - ``phase-shift=<count>`` - Log2 of the number of polyphase entries. (..., 10->1024, 11->2048, - 12->4096, ...) (default: 10->1024) - ``cutoff=<cutoff>`` - Cutoff frequency (0.0-1.0), default set depending upon filter length. - ``linear`` - If set then filters will be linearly interpolated between polyphase - entries. (default: no) - ``no-detach`` - Do not detach if input and output audio format/rate/channels match. - (If you just want to set defaults for this filter that will be used - even by automatically inserted lavrresample instances, you should - prefer setting them with the ``--audio-resample-...`` options.) This - does not do anything anymore and the filter will never detach. - ``normalize=<yes|no|auto>`` - Whether to normalize when remixing channel layouts (default: auto). - ``auto`` uses the value set by ``--audio-normalize-downmix``. - ``o=<string>`` - Set AVOptions on the SwrContext or AVAudioResampleContext. These should - be documented by FFmpeg or Libav. - ``lavcac3enc[=options]`` Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports 16-bit native-endian input format, maximum 6 channels. The output is @@ -199,6 +161,28 @@ Available filters are: Would play media at 1.2x normal speed, with audio at normal pitch. Changing playback speed would change pitch, leaving audio tempo at 1.2x. + +``scaletempo2[=option1:option2:...]`` + Scales audio tempo without altering pitch. + The algorithm is ported from chromium and uses the + Waveform Similarity Overlap-and-add (WSOLA) method. + It seems to achieve a higher audio quality than scaletempo and rubberband. + + By default, the ``search-interval`` and ``window-size`` parameters + have the same values as in chromium. + + ``min-speed=<speed>`` + Mute audio if the playback speed is below ``<speed>``. (default: 0.25) + + ``max-speed=<speed>`` + Mute audio if the playback speed is above ``<speed>`` + and ``<speed> != 0``. (default: 4.0) + + ``search-interval=<amount>`` + Length in milliseconds to search for best overlap position. (default: 30) + + ``window-size=<amount>`` + Length in milliseconds of the overlap-and-add window. (default: 20) ``rubberband`` High quality pitch correction with librubberband. This can be used in place @@ -214,7 +198,7 @@ Available filters are: for each option. The options are not documented here, because they are merely passed to librubberband. Look at the librubberband documentation to learn what each option does: - http://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html + https://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html (The mapping of the mpv rubberband filter sub-option names and values to those of librubberband follows a simple pattern: ``"Option" + Name + Value``.) @@ -258,3 +242,10 @@ Available filters are: broken filters. In practice, these broken filters will either cause slow A/V desync over time (with some files), or break playback completely if you seek or start playback from the middle of a file. + +``drop`` + This filter drops or repeats audio frames to adapt to playback speed. It + always operates on full audio frames, because it was made to handle SPDIF + (compressed audio passthrough). This is used automatically if the + ``--video-sync=display-adrop`` option is used. Do not use this filter (or + the given option); they are extremely low quality. |