1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
|
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`` for the 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:
``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 ``--af-defaults=lavrresample:...``.)
``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
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-format: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-format>``
``<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.
``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.
|
