mpv/DOCS/man/af.rst

252 lines
10 KiB
ReStructuredText

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-defaults``, ``--af-add``,
``--af-pre``, ``--af-del``, ``--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 (default).
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=<percent>``
Percentage 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 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
of ``scaletempo``, 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>``
Sets the pitch scaling factor. Frequencies are multiplied by this value.
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:
http://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``.)
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.