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diff --git a/DOCS/man/af.rst b/DOCS/man/af.rst new file mode 100644 index 0000000..98f9a95 --- /dev/null +++ b/DOCS/man/af.rst @@ -0,0 +1,267 @@ +AUDIO FILTERS +============= + +Audio filters allow you to modify the audio stream and its properties. The +syntax is: + +``--af=...`` + Setup a chain of audio filters. See ``--vf`` (`VIDEO FILTERS`_) for the + full syntax. + +.. note:: + + To get a full list of available audio filters, see ``--af=help``. + + Also, keep in mind that most actual filters are available via the ``lavfi`` + wrapper, which gives you access to most of libavfilter's filters. This + includes all filters that have been ported from MPlayer to libavfilter. + + The ``--vf`` description describes how libavfilter can be used and how to + workaround deprecated mpv filters. + +See ``--vf`` group of options for info on how ``--af-add``, ``--af-pre``, +``--af-clr``, and possibly others work. + +Available filters are: + +``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 + big-endian when outputting a raw AC-3 stream, native-endian when + outputting to S/PDIF. If the input sample rate is not 48 kHz, 44.1 kHz or + 32 kHz, it will be resampled to 48 kHz. + + ``tospdif=<yes|no>`` + Output raw AC-3 stream if ``no``, output to S/PDIF for + pass-through if ``yes`` (default). + + ``bitrate=<rate>`` + The bitrate use for the AC-3 stream. Set it to 384 to get 384 kbps. + + The default is 640. Some receivers might not be able to handle this. + + Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128, + 160, 192, 224, 256, 320, 384, 448, 512, 576, 640. + + The special value ``auto`` selects a default bitrate based on the + input channel number: + + :1ch: 96 + :2ch: 192 + :3ch: 224 + :4ch: 384 + :5ch: 448 + :6ch: 448 + + ``minch=<n>`` + If the input channel number is less than ``<minch>``, the filter will + detach itself (default: 3). + + ``encoder=<name>`` + Select the libavcodec encoder used. Currently, this should be an AC-3 + encoder, and using another codec will fail horribly. + +``format=format:srate:channels:out-srate:out-channels`` + Does not do any format conversion itself. Rather, it may cause the + filter system to insert necessary conversion filters before or after this + filter if needed. It is primarily useful for controlling the audio format + going into other filters. To specify the format for audio output, see + ``--audio-format``, ``--audio-samplerate``, and ``--audio-channels``. This + filter is able to force a particular format, whereas ``--audio-*`` + may be overridden by the ao based on output compatibility. + + All parameters are optional. The first 3 parameters restrict what the filter + accepts as input. They will therefore cause conversion filters to be + inserted before this one. The ``out-`` parameters tell the filters or audio + outputs following this filter how to interpret the data without actually + doing a conversion. Setting these will probably just break things unless you + really know you want this for some reason, such as testing or dealing with + broken media. + + ``<format>`` + Force conversion to this format. Use ``--af=format=format=help`` to get + a list of valid formats. + + ``<srate>`` + Force conversion to a specific sample rate. The rate is an integer, + 48000 for example. + + ``<channels>`` + Force mixing to a specific channel layout. See ``--audio-channels`` option + for possible values. + + ``<out-srate>`` + + ``<out-channels>`` + + *NOTE*: this filter used to be named ``force``. The old ``format`` filter + used to do conversion itself, unlike this one which lets the filter system + handle the conversion. + +``scaletempo[=option1:option2:...]`` + Scales audio tempo without altering pitch, optionally synced to playback + speed. + + This works by playing 'stride' ms of audio at normal speed then consuming + 'stride*scale' ms of input audio. It pieces the strides together by + blending 'overlap'% of stride with audio following the previous stride. It + optionally performs a short statistical analysis on the next 'search' ms + of audio to determine the best overlap position. + + ``scale=<amount>`` + Nominal amount to scale tempo. Scales this amount in addition to + speed. (default: 1.0) + ``stride=<amount>`` + Length in milliseconds to output each stride. Too high of a value will + cause noticeable skips at high scale amounts and an echo at low scale + amounts. Very low values will alter pitch. Increasing improves + performance. (default: 60) + ``overlap=<factor>`` + Factor of stride to overlap. Decreasing improves performance. + (default: .20) + ``search=<amount>`` + Length in milliseconds to search for best overlap position. Decreasing + improves performance greatly. On slow systems, you will probably want + to set this very low. (default: 14) + ``speed=<tempo|pitch|both|none>`` + Set response to speed change. + + tempo + Scale tempo in sync with speed (default). + pitch + Reverses effect of filter. Scales pitch without altering tempo. + Add this to your ``input.conf`` to step by musical semi-tones:: + + [ multiply speed 0.9438743126816935 + ] multiply speed 1.059463094352953 + + .. warning:: + + Loses sync with video. + both + Scale both tempo and pitch. + none + Ignore speed changes. + + .. admonition:: Examples + + ``mpv --af=scaletempo --speed=1.2 media.ogg`` + Would play media at 1.2x normal speed, with audio at normal + pitch. Changing playback speed would change audio tempo to match. + + ``mpv --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg`` + Would play media at 1.2x normal speed, with audio at normal + pitch, but changing playback speed would have no effect on audio + tempo. + + ``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg`` + Would tweak the quality and performance parameters. + + ``mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg`` + 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 achieves higher audio quality than scaletempo, and rubberband R2 + engine, or ``engine=faster``. This filter is inserted automatically if + ``audio-pitch-correction`` option is used (on by default) when the playback + speed is changed. + + 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: 8.0) + + ``search-interval=<amount>`` + Length in milliseconds to search for best overlap position. (default: 40) + + ``window-size=<amount>`` + Length in milliseconds of the overlap-and-add window. (default: 12) + +``rubberband`` + High quality pitch correction with librubberband. This can be used in place + of ``scaletempo`` and ``scaletempo2``, and will be used to adjust audio pitch + when playing at speed different from normal. It can also be used to adjust + audio pitch without changing playback speed. + + ``pitch-scale=<amount>`` + Sets the pitch scaling factor. Frequencies are multiplied by this value. + (default: 1.0) + + ``engine=<faster|finer>`` + Select the core Rubberband engine to be used. There are two available: + + :Faster: This is the Rubberband R2 engine. It uses significantly less + CPU than the Finer (R3) engine. + :Finer: This is the Rubberband R3 engine. This engine is only available + with librubberband version 3 or newer. This produces significantly + higher quality output, at the cost of higher CPU usage. (Default + if available) + + This filter has a number of additional sub-options. You can list them with + ``mpv --af=rubberband=help``. This will also show the default values + 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: + https://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html + Do note that certain options are only applicable to one of R2 (faster) and + R3 (finer) engines. + (The mapping of the mpv rubberband filter sub-option names and values to + those of librubberband follows a simple pattern: ``"Option" + Name + Value``.) + + This filter supports the following ``af-command`` commands: + + ``set-pitch`` + Set the ``<pitch-scale>`` argument dynamically. This can be used to + change the playback pitch at runtime. Note that speed is controlled + using the standard ``speed`` property, not ``af-command``. + + ``multiply-pitch <factor>`` + Multiply the current value of ``<pitch-scale>`` dynamically. For + example: 0.5 to go down by an octave, 1.5 to go up by a perfect fifth. + If you want to go up or down by semi-tones, use 1.059463094352953 and + 0.9438743126816935 + +``lavfi=graph`` + Filter audio using FFmpeg's libavfilter. + + ``<graph>`` + Libavfilter graph. See ``lavfi`` video filter for details - the graph + syntax is the same. + + .. warning:: + + Don't forget to quote libavfilter graphs as described in the lavfi + video filter section. + + ``o=<string>`` + AVOptions. + + ``fix-pts=<yes|no>`` + Determine PTS based on sample count (default: no). If this is enabled, + the player won't rely on libavfilter passing through PTS accurately. + Instead, it pass a sample count as PTS to libavfilter, and compute the + PTS used by mpv based on that and the input PTS. This helps with filters + which output a recomputed PTS instead of the original PTS (including + filters which require the PTS to start at 0). mpv normally expects + filters to not touch the PTS (or only to the extent of changing frame + boundaries), so this is not the default, but it will be needed to use + 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. |