1 @chapter Filtering Introduction
2 @c man begin FILTERING INTRODUCTION
4 Filtering in FFmpeg is enabled through the libavfilter library.
6 In libavfilter, a filter can have multiple inputs and multiple
8 To illustrate the sorts of things that are possible, we consider the
13 input --> split ---------------------> overlay --> output
16 +-----> crop --> vflip -------+
19 This filtergraph splits the input stream in two streams, then sends one
20 stream through the crop filter and the vflip filter, before merging it
21 back with the other stream by overlaying it on top. You can use the
22 following command to achieve this:
25 ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT
28 The result will be that the top half of the video is mirrored
29 onto the bottom half of the output video.
31 Filters in the same linear chain are separated by commas, and distinct
32 linear chains of filters are separated by semicolons. In our example,
33 @var{crop,vflip} are in one linear chain, @var{split} and
34 @var{overlay} are separately in another. The points where the linear
35 chains join are labelled by names enclosed in square brackets. In the
36 example, the split filter generates two outputs that are associated to
37 the labels @var{[main]} and @var{[tmp]}.
39 The stream sent to the second output of @var{split}, labelled as
40 @var{[tmp]}, is processed through the @var{crop} filter, which crops
41 away the lower half part of the video, and then vertically flipped. The
42 @var{overlay} filter takes in input the first unchanged output of the
43 split filter (which was labelled as @var{[main]}), and overlay on its
44 lower half the output generated by the @var{crop,vflip} filterchain.
46 Some filters take in input a list of parameters: they are specified
47 after the filter name and an equal sign, and are separated from each other
50 There exist so-called @var{source filters} that do not have an
51 audio/video input, and @var{sink filters} that will not have audio/video
54 @c man end FILTERING INTRODUCTION
57 @c man begin GRAPH2DOT
59 The @file{graph2dot} program included in the FFmpeg @file{tools}
60 directory can be used to parse a filtergraph description and issue a
61 corresponding textual representation in the dot language.
68 to see how to use @file{graph2dot}.
70 You can then pass the dot description to the @file{dot} program (from
71 the graphviz suite of programs) and obtain a graphical representation
74 For example the sequence of commands:
76 echo @var{GRAPH_DESCRIPTION} | \
77 tools/graph2dot -o graph.tmp && \
78 dot -Tpng graph.tmp -o graph.png && \
82 can be used to create and display an image representing the graph
83 described by the @var{GRAPH_DESCRIPTION} string. Note that this string must be
84 a complete self-contained graph, with its inputs and outputs explicitly defined.
85 For example if your command line is of the form:
87 ffmpeg -i infile -vf scale=640:360 outfile
89 your @var{GRAPH_DESCRIPTION} string will need to be of the form:
91 nullsrc,scale=640:360,nullsink
93 you may also need to set the @var{nullsrc} parameters and add a @var{format}
94 filter in order to simulate a specific input file.
98 @chapter Filtergraph description
99 @c man begin FILTERGRAPH DESCRIPTION
101 A filtergraph is a directed graph of connected filters. It can contain
102 cycles, and there can be multiple links between a pair of
103 filters. Each link has one input pad on one side connecting it to one
104 filter from which it takes its input, and one output pad on the other
105 side connecting it to one filter accepting its output.
107 Each filter in a filtergraph is an instance of a filter class
108 registered in the application, which defines the features and the
109 number of input and output pads of the filter.
111 A filter with no input pads is called a "source", and a filter with no
112 output pads is called a "sink".
114 @anchor{Filtergraph syntax}
115 @section Filtergraph syntax
117 A filtergraph has a textual representation, which is recognized by the
118 @option{-filter}/@option{-vf}/@option{-af} and
119 @option{-filter_complex} options in @command{ffmpeg} and
120 @option{-vf}/@option{-af} in @command{ffplay}, and by the
121 @code{avfilter_graph_parse_ptr()} function defined in
122 @file{libavfilter/avfilter.h}.
124 A filterchain consists of a sequence of connected filters, each one
125 connected to the previous one in the sequence. A filterchain is
126 represented by a list of ","-separated filter descriptions.
128 A filtergraph consists of a sequence of filterchains. A sequence of
129 filterchains is represented by a list of ";"-separated filterchain
132 A filter is represented by a string of the form:
133 [@var{in_link_1}]...[@var{in_link_N}]@var{filter_name}=@var{arguments}[@var{out_link_1}]...[@var{out_link_M}]
135 @var{filter_name} is the name of the filter class of which the
136 described filter is an instance of, and has to be the name of one of
137 the filter classes registered in the program.
138 The name of the filter class is optionally followed by a string
141 @var{arguments} is a string which contains the parameters used to
142 initialize the filter instance. It may have one of two forms:
146 A ':'-separated list of @var{key=value} pairs.
149 A ':'-separated list of @var{value}. In this case, the keys are assumed to be
150 the option names in the order they are declared. E.g. the @code{fade} filter
151 declares three options in this order -- @option{type}, @option{start_frame} and
152 @option{nb_frames}. Then the parameter list @var{in:0:30} means that the value
153 @var{in} is assigned to the option @option{type}, @var{0} to
154 @option{start_frame} and @var{30} to @option{nb_frames}.
157 A ':'-separated list of mixed direct @var{value} and long @var{key=value}
158 pairs. The direct @var{value} must precede the @var{key=value} pairs, and
159 follow the same constraints order of the previous point. The following
160 @var{key=value} pairs can be set in any preferred order.
164 If the option value itself is a list of items (e.g. the @code{format} filter
165 takes a list of pixel formats), the items in the list are usually separated by
168 The list of arguments can be quoted using the character @samp{'} as initial
169 and ending mark, and the character @samp{\} for escaping the characters
170 within the quoted text; otherwise the argument string is considered
171 terminated when the next special character (belonging to the set
172 @samp{[]=;,}) is encountered.
174 The name and arguments of the filter are optionally preceded and
175 followed by a list of link labels.
176 A link label allows one to name a link and associate it to a filter output
177 or input pad. The preceding labels @var{in_link_1}
178 ... @var{in_link_N}, are associated to the filter input pads,
179 the following labels @var{out_link_1} ... @var{out_link_M}, are
180 associated to the output pads.
182 When two link labels with the same name are found in the
183 filtergraph, a link between the corresponding input and output pad is
186 If an output pad is not labelled, it is linked by default to the first
187 unlabelled input pad of the next filter in the filterchain.
188 For example in the filterchain
190 nullsrc, split[L1], [L2]overlay, nullsink
192 the split filter instance has two output pads, and the overlay filter
193 instance two input pads. The first output pad of split is labelled
194 "L1", the first input pad of overlay is labelled "L2", and the second
195 output pad of split is linked to the second input pad of overlay,
196 which are both unlabelled.
198 In a filter description, if the input label of the first filter is not
199 specified, "in" is assumed; if the output label of the last filter is not
200 specified, "out" is assumed.
202 In a complete filterchain all the unlabelled filter input and output
203 pads must be connected. A filtergraph is considered valid if all the
204 filter input and output pads of all the filterchains are connected.
206 Libavfilter will automatically insert @ref{scale} filters where format
207 conversion is required. It is possible to specify swscale flags
208 for those automatically inserted scalers by prepending
209 @code{sws_flags=@var{flags};}
210 to the filtergraph description.
212 Here is a BNF description of the filtergraph syntax:
214 @var{NAME} ::= sequence of alphanumeric characters and '_'
215 @var{LINKLABEL} ::= "[" @var{NAME} "]"
216 @var{LINKLABELS} ::= @var{LINKLABEL} [@var{LINKLABELS}]
217 @var{FILTER_ARGUMENTS} ::= sequence of chars (possibly quoted)
218 @var{FILTER} ::= [@var{LINKLABELS}] @var{NAME} ["=" @var{FILTER_ARGUMENTS}] [@var{LINKLABELS}]
219 @var{FILTERCHAIN} ::= @var{FILTER} [,@var{FILTERCHAIN}]
220 @var{FILTERGRAPH} ::= [sws_flags=@var{flags};] @var{FILTERCHAIN} [;@var{FILTERGRAPH}]
223 @section Notes on filtergraph escaping
225 Filtergraph description composition entails several levels of
226 escaping. See @ref{quoting_and_escaping,,the "Quoting and escaping"
227 section in the ffmpeg-utils(1) manual,ffmpeg-utils} for more
228 information about the employed escaping procedure.
230 A first level escaping affects the content of each filter option
231 value, which may contain the special character @code{:} used to
232 separate values, or one of the escaping characters @code{\'}.
234 A second level escaping affects the whole filter description, which
235 may contain the escaping characters @code{\'} or the special
236 characters @code{[],;} used by the filtergraph description.
238 Finally, when you specify a filtergraph on a shell commandline, you
239 need to perform a third level escaping for the shell special
240 characters contained within it.
242 For example, consider the following string to be embedded in
243 the @ref{drawtext} filter description @option{text} value:
245 this is a 'string': may contain one, or more, special characters
248 This string contains the @code{'} special escaping character, and the
249 @code{:} special character, so it needs to be escaped in this way:
251 text=this is a \'string\'\: may contain one, or more, special characters
254 A second level of escaping is required when embedding the filter
255 description in a filtergraph description, in order to escape all the
256 filtergraph special characters. Thus the example above becomes:
258 drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters
260 (note that in addition to the @code{\'} escaping special characters,
261 also @code{,} needs to be escaped).
263 Finally an additional level of escaping is needed when writing the
264 filtergraph description in a shell command, which depends on the
265 escaping rules of the adopted shell. For example, assuming that
266 @code{\} is special and needs to be escaped with another @code{\}, the
267 previous string will finally result in:
269 -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"
272 @chapter Timeline editing
274 Some filters support a generic @option{enable} option. For the filters
275 supporting timeline editing, this option can be set to an expression which is
276 evaluated before sending a frame to the filter. If the evaluation is non-zero,
277 the filter will be enabled, otherwise the frame will be sent unchanged to the
278 next filter in the filtergraph.
280 The expression accepts the following values:
283 timestamp expressed in seconds, NAN if the input timestamp is unknown
286 sequential number of the input frame, starting from 0
289 the position in the file of the input frame, NAN if unknown
293 width and height of the input frame if video
296 Additionally, these filters support an @option{enable} command that can be used
297 to re-define the expression.
299 Like any other filtering option, the @option{enable} option follows the same
302 For example, to enable a blur filter (@ref{smartblur}) from 10 seconds to 3
303 minutes, and a @ref{curves} filter starting at 3 seconds:
305 smartblur = enable='between(t,10,3*60)',
306 curves = enable='gte(t,3)' : preset=cross_process
309 @c man end FILTERGRAPH DESCRIPTION
311 @chapter Audio Filters
312 @c man begin AUDIO FILTERS
314 When you configure your FFmpeg build, you can disable any of the
315 existing filters using @code{--disable-filters}.
316 The configure output will show the audio filters included in your
319 Below is a description of the currently available audio filters.
323 A compressor is mainly used to reduce the dynamic range of a signal.
324 Especially modern music is mostly compressed at a high ratio to
325 improve the overall loudness. It's done to get the highest attention
326 of a listener, "fatten" the sound and bring more "power" to the track.
327 If a signal is compressed too much it may sound dull or "dead"
328 afterwards or it may start to "pump" (which could be a powerful effect
329 but can also destroy a track completely).
330 The right compression is the key to reach a professional sound and is
331 the high art of mixing and mastering. Because of its complex settings
332 it may take a long time to get the right feeling for this kind of effect.
334 Compression is done by detecting the volume above a chosen level
335 @code{threshold} and dividing it by the factor set with @code{ratio}.
336 So if you set the threshold to -12dB and your signal reaches -6dB a ratio
337 of 2:1 will result in a signal at -9dB. Because an exact manipulation of
338 the signal would cause distortion of the waveform the reduction can be
339 levelled over the time. This is done by setting "Attack" and "Release".
340 @code{attack} determines how long the signal has to rise above the threshold
341 before any reduction will occur and @code{release} sets the time the signal
342 has to fall below the threshold to reduce the reduction again. Shorter signals
343 than the chosen attack time will be left untouched.
344 The overall reduction of the signal can be made up afterwards with the
345 @code{makeup} setting. So compressing the peaks of a signal about 6dB and
346 raising the makeup to this level results in a signal twice as loud than the
347 source. To gain a softer entry in the compression the @code{knee} flattens the
348 hard edge at the threshold in the range of the chosen decibels.
350 The filter accepts the following options:
354 Set input gain. Default is 1. Range is between 0.015625 and 64.
357 If a signal of second stream rises above this level it will affect the gain
358 reduction of the first stream.
359 By default it is 0.125. Range is between 0.00097563 and 1.
362 Set a ratio by which the signal is reduced. 1:2 means that if the level
363 rose 4dB above the threshold, it will be only 2dB above after the reduction.
364 Default is 2. Range is between 1 and 20.
367 Amount of milliseconds the signal has to rise above the threshold before gain
368 reduction starts. Default is 20. Range is between 0.01 and 2000.
371 Amount of milliseconds the signal has to fall below the threshold before
372 reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
375 Set the amount by how much signal will be amplified after processing.
376 Default is 2. Range is from 1 and 64.
379 Curve the sharp knee around the threshold to enter gain reduction more softly.
380 Default is 2.82843. Range is between 1 and 8.
383 Choose if the @code{average} level between all channels of input stream
384 or the louder(@code{maximum}) channel of input stream affects the
385 reduction. Default is @code{average}.
388 Should the exact signal be taken in case of @code{peak} or an RMS one in case
389 of @code{rms}. Default is @code{rms} which is mostly smoother.
392 How much to use compressed signal in output. Default is 1.
393 Range is between 0 and 1.
398 Apply cross fade from one input audio stream to another input audio stream.
399 The cross fade is applied for specified duration near the end of first stream.
401 The filter accepts the following options:
405 Specify the number of samples for which the cross fade effect has to last.
406 At the end of the cross fade effect the first input audio will be completely
407 silent. Default is 44100.
410 Specify the duration of the cross fade effect. See
411 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
412 for the accepted syntax.
413 By default the duration is determined by @var{nb_samples}.
414 If set this option is used instead of @var{nb_samples}.
417 Should first stream end overlap with second stream start. Default is enabled.
420 Set curve for cross fade transition for first stream.
423 Set curve for cross fade transition for second stream.
425 For description of available curve types see @ref{afade} filter description.
432 Cross fade from one input to another:
434 ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
438 Cross fade from one input to another but without overlapping:
440 ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
446 Reduce audio bit resolution.
448 This filter is bit crusher with enhanced functionality. A bit crusher
449 is used to audibly reduce number of bits an audio signal is sampled
450 with. This doesn't change the bit depth at all, it just produces the
451 effect. Material reduced in bit depth sounds more harsh and "digital".
452 This filter is able to even round to continous values instead of discrete
454 Additionally it has a D/C offset which results in different crushing of
455 the lower and the upper half of the signal.
456 An Anti-Aliasing setting is able to produce "softer" crushing sounds.
458 Another feature of this filter is the logarithmic mode.
459 This setting switches from linear distances between bits to logarithmic ones.
460 The result is a much more "natural" sounding crusher which doesn't gate low
461 signals for example. The human ear has a logarithmic perception, too
462 so this kind of crushing is much more pleasant.
463 Logarithmic crushing is also able to get anti-aliased.
465 The filter accepts the following options:
481 Can be linear: @code{lin} or logarithmic: @code{log}.
490 Set sample reduction.
493 Enable LFO. By default disabled.
504 Delay one or more audio channels.
506 Samples in delayed channel are filled with silence.
508 The filter accepts the following option:
512 Set list of delays in milliseconds for each channel separated by '|'.
513 At least one delay greater than 0 should be provided.
514 Unused delays will be silently ignored. If number of given delays is
515 smaller than number of channels all remaining channels will not be delayed.
516 If you want to delay exact number of samples, append 'S' to number.
523 Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave
524 the second channel (and any other channels that may be present) unchanged.
530 Delay second channel by 500 samples, the third channel by 700 samples and leave
531 the first channel (and any other channels that may be present) unchanged.
539 Apply echoing to the input audio.
541 Echoes are reflected sound and can occur naturally amongst mountains
542 (and sometimes large buildings) when talking or shouting; digital echo
543 effects emulate this behaviour and are often used to help fill out the
544 sound of a single instrument or vocal. The time difference between the
545 original signal and the reflection is the @code{delay}, and the
546 loudness of the reflected signal is the @code{decay}.
547 Multiple echoes can have different delays and decays.
549 A description of the accepted parameters follows.
553 Set input gain of reflected signal. Default is @code{0.6}.
556 Set output gain of reflected signal. Default is @code{0.3}.
559 Set list of time intervals in milliseconds between original signal and reflections
560 separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}.
561 Default is @code{1000}.
564 Set list of loudnesses of reflected signals separated by '|'.
565 Allowed range for each @code{decay} is @code{(0 - 1.0]}.
566 Default is @code{0.5}.
573 Make it sound as if there are twice as many instruments as are actually playing:
575 aecho=0.8:0.88:60:0.4
579 If delay is very short, then it sound like a (metallic) robot playing music:
585 A longer delay will sound like an open air concert in the mountains:
587 aecho=0.8:0.9:1000:0.3
591 Same as above but with one more mountain:
593 aecho=0.8:0.9:1000|1800:0.3|0.25
598 Audio emphasis filter creates or restores material directly taken from LPs or
599 emphased CDs with different filter curves. E.g. to store music on vinyl the
600 signal has to be altered by a filter first to even out the disadvantages of
601 this recording medium.
602 Once the material is played back the inverse filter has to be applied to
603 restore the distortion of the frequency response.
605 The filter accepts the following options:
615 Set filter mode. For restoring material use @code{reproduction} mode, otherwise
616 use @code{production} mode. Default is @code{reproduction} mode.
619 Set filter type. Selects medium. Can be one of the following:
631 select Compact Disc (CD).
637 select 50µs (FM-KF).
639 select 75µs (FM-KF).
645 Modify an audio signal according to the specified expressions.
647 This filter accepts one or more expressions (one for each channel),
648 which are evaluated and used to modify a corresponding audio signal.
650 It accepts the following parameters:
654 Set the '|'-separated expressions list for each separate channel. If
655 the number of input channels is greater than the number of
656 expressions, the last specified expression is used for the remaining
659 @item channel_layout, c
660 Set output channel layout. If not specified, the channel layout is
661 specified by the number of expressions. If set to @samp{same}, it will
662 use by default the same input channel layout.
665 Each expression in @var{exprs} can contain the following constants and functions:
669 channel number of the current expression
672 number of the evaluated sample, starting from 0
678 time of the evaluated sample expressed in seconds
681 @item nb_out_channels
682 input and output number of channels
685 the value of input channel with number @var{CH}
688 Note: this filter is slow. For faster processing you should use a
697 aeval=val(ch)/2:c=same
701 Invert phase of the second channel:
710 Apply fade-in/out effect to input audio.
712 A description of the accepted parameters follows.
716 Specify the effect type, can be either @code{in} for fade-in, or
717 @code{out} for a fade-out effect. Default is @code{in}.
719 @item start_sample, ss
720 Specify the number of the start sample for starting to apply the fade
721 effect. Default is 0.
724 Specify the number of samples for which the fade effect has to last. At
725 the end of the fade-in effect the output audio will have the same
726 volume as the input audio, at the end of the fade-out transition
727 the output audio will be silence. Default is 44100.
730 Specify the start time of the fade effect. Default is 0.
731 The value must be specified as a time duration; see
732 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
733 for the accepted syntax.
734 If set this option is used instead of @var{start_sample}.
737 Specify the duration of the fade effect. See
738 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
739 for the accepted syntax.
740 At the end of the fade-in effect the output audio will have the same
741 volume as the input audio, at the end of the fade-out transition
742 the output audio will be silence.
743 By default the duration is determined by @var{nb_samples}.
744 If set this option is used instead of @var{nb_samples}.
747 Set curve for fade transition.
749 It accepts the following values:
752 select triangular, linear slope (default)
754 select quarter of sine wave
756 select half of sine wave
758 select exponential sine wave
762 select inverted parabola
776 select inverted quarter of sine wave
778 select inverted half of sine wave
780 select double-exponential seat
782 select double-exponential sigmoid
790 Fade in first 15 seconds of audio:
796 Fade out last 25 seconds of a 900 seconds audio:
798 afade=t=out:st=875:d=25
803 Apply arbitrary expressions to samples in frequency domain.
807 Set frequency domain real expression for each separate channel separated
808 by '|'. Default is "1".
809 If the number of input channels is greater than the number of
810 expressions, the last specified expression is used for the remaining
814 Set frequency domain imaginary expression for each separate channel
815 separated by '|'. If not set, @var{real} option is used.
817 Each expression in @var{real} and @var{imag} can contain the following
825 current frequency bin number
828 number of available bins
831 channel number of the current expression
843 It accepts the following values:
859 Default is @code{w4096}
862 Set window function. Default is @code{hann}.
865 Set window overlap. If set to 1, the recommended overlap for selected
866 window function will be picked. Default is @code{0.75}.
873 Leave almost only low frequencies in audio:
875 afftfilt="1-clip((b/nb)*b,0,1)"
882 Set output format constraints for the input audio. The framework will
883 negotiate the most appropriate format to minimize conversions.
885 It accepts the following parameters:
889 A '|'-separated list of requested sample formats.
892 A '|'-separated list of requested sample rates.
894 @item channel_layouts
895 A '|'-separated list of requested channel layouts.
897 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
898 for the required syntax.
901 If a parameter is omitted, all values are allowed.
903 Force the output to either unsigned 8-bit or signed 16-bit stereo
905 aformat=sample_fmts=u8|s16:channel_layouts=stereo
910 A gate is mainly used to reduce lower parts of a signal. This kind of signal
911 processing reduces disturbing noise between useful signals.
913 Gating is done by detecting the volume below a chosen level @var{threshold}
914 and divide it by the factor set with @var{ratio}. The bottom of the noise
915 floor is set via @var{range}. Because an exact manipulation of the signal
916 would cause distortion of the waveform the reduction can be levelled over
917 time. This is done by setting @var{attack} and @var{release}.
919 @var{attack} determines how long the signal has to fall below the threshold
920 before any reduction will occur and @var{release} sets the time the signal
921 has to raise above the threshold to reduce the reduction again.
922 Shorter signals than the chosen attack time will be left untouched.
926 Set input level before filtering.
927 Default is 1. Allowed range is from 0.015625 to 64.
930 Set the level of gain reduction when the signal is below the threshold.
931 Default is 0.06125. Allowed range is from 0 to 1.
934 If a signal rises above this level the gain reduction is released.
935 Default is 0.125. Allowed range is from 0 to 1.
938 Set a ratio about which the signal is reduced.
939 Default is 2. Allowed range is from 1 to 9000.
942 Amount of milliseconds the signal has to rise above the threshold before gain
944 Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
947 Amount of milliseconds the signal has to fall below the threshold before the
948 reduction is increased again. Default is 250 milliseconds.
949 Allowed range is from 0.01 to 9000.
952 Set amount of amplification of signal after processing.
953 Default is 1. Allowed range is from 1 to 64.
956 Curve the sharp knee around the threshold to enter gain reduction more softly.
957 Default is 2.828427125. Allowed range is from 1 to 8.
960 Choose if exact signal should be taken for detection or an RMS like one.
961 Default is rms. Can be peak or rms.
964 Choose if the average level between all channels or the louder channel affects
966 Default is average. Can be average or maximum.
971 The limiter prevents input signal from raising over a desired threshold.
972 This limiter uses lookahead technology to prevent your signal from distorting.
973 It means that there is a small delay after signal is processed. Keep in mind
974 that the delay it produces is the attack time you set.
976 The filter accepts the following options:
980 Set input gain. Default is 1.
983 Set output gain. Default is 1.
986 Don't let signals above this level pass the limiter. Default is 1.
989 The limiter will reach its attenuation level in this amount of time in
990 milliseconds. Default is 5 milliseconds.
993 Come back from limiting to attenuation 1.0 in this amount of milliseconds.
994 Default is 50 milliseconds.
997 When gain reduction is always needed ASC takes care of releasing to an
998 average reduction level rather than reaching a reduction of 0 in the release
1002 Select how much the release time is affected by ASC, 0 means nearly no changes
1003 in release time while 1 produces higher release times.
1006 Auto level output signal. Default is enabled.
1007 This normalizes audio back to 0dB if enabled.
1010 Depending on picked setting it is recommended to upsample input 2x or 4x times
1011 with @ref{aresample} before applying this filter.
1015 Apply a two-pole all-pass filter with central frequency (in Hz)
1016 @var{frequency}, and filter-width @var{width}.
1017 An all-pass filter changes the audio's frequency to phase relationship
1018 without changing its frequency to amplitude relationship.
1020 The filter accepts the following options:
1024 Set frequency in Hz.
1027 Set method to specify band-width of filter.
1040 Specify the band-width of a filter in width_type units.
1047 The filter accepts the following options:
1051 Set the number of loops.
1054 Set maximal number of samples.
1057 Set first sample of loop.
1063 Merge two or more audio streams into a single multi-channel stream.
1065 The filter accepts the following options:
1070 Set the number of inputs. Default is 2.
1074 If the channel layouts of the inputs are disjoint, and therefore compatible,
1075 the channel layout of the output will be set accordingly and the channels
1076 will be reordered as necessary. If the channel layouts of the inputs are not
1077 disjoint, the output will have all the channels of the first input then all
1078 the channels of the second input, in that order, and the channel layout of
1079 the output will be the default value corresponding to the total number of
1082 For example, if the first input is in 2.1 (FL+FR+LF) and the second input
1083 is FC+BL+BR, then the output will be in 5.1, with the channels in the
1084 following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
1085 first input, b1 is the first channel of the second input).
1087 On the other hand, if both input are in stereo, the output channels will be
1088 in the default order: a1, a2, b1, b2, and the channel layout will be
1089 arbitrarily set to 4.0, which may or may not be the expected value.
1091 All inputs must have the same sample rate, and format.
1093 If inputs do not have the same duration, the output will stop with the
1096 @subsection Examples
1100 Merge two mono files into a stereo stream:
1102 amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
1106 Multiple merges assuming 1 video stream and 6 audio streams in @file{input.mkv}:
1108 ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv
1114 Mixes multiple audio inputs into a single output.
1116 Note that this filter only supports float samples (the @var{amerge}
1117 and @var{pan} audio filters support many formats). If the @var{amix}
1118 input has integer samples then @ref{aresample} will be automatically
1119 inserted to perform the conversion to float samples.
1123 ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
1125 will mix 3 input audio streams to a single output with the same duration as the
1126 first input and a dropout transition time of 3 seconds.
1128 It accepts the following parameters:
1132 The number of inputs. If unspecified, it defaults to 2.
1135 How to determine the end-of-stream.
1139 The duration of the longest input. (default)
1142 The duration of the shortest input.
1145 The duration of the first input.
1149 @item dropout_transition
1150 The transition time, in seconds, for volume renormalization when an input
1151 stream ends. The default value is 2 seconds.
1155 @section anequalizer
1157 High-order parametric multiband equalizer for each channel.
1159 It accepts the following parameters:
1163 This option string is in format:
1164 "c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..."
1165 Each equalizer band is separated by '|'.
1169 Set channel number to which equalization will be applied.
1170 If input doesn't have that channel the entry is ignored.
1173 Set central frequency for band.
1174 If input doesn't have that frequency the entry is ignored.
1177 Set band width in hertz.
1180 Set band gain in dB.
1183 Set filter type for band, optional, can be:
1187 Butterworth, this is default.
1198 With this option activated frequency response of anequalizer is displayed
1202 Set video stream size. Only useful if curves option is activated.
1205 Set max gain that will be displayed. Only useful if curves option is activated.
1206 Setting this to reasonable value allows to display gain which is derived from
1207 neighbour bands which are too close to each other and thus produce higher gain
1208 when both are activated.
1211 Set frequency scale used to draw frequency response in video output.
1212 Can be linear or logarithmic. Default is logarithmic.
1215 Set color for each channel curve which is going to be displayed in video stream.
1216 This is list of color names separated by space or by '|'.
1217 Unrecognised or missing colors will be replaced by white color.
1220 @subsection Examples
1224 Lower gain by 10 of central frequency 200Hz and width 100 Hz
1225 for first 2 channels using Chebyshev type 1 filter:
1227 anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
1231 @subsection Commands
1233 This filter supports the following commands:
1236 Alter existing filter parameters.
1237 Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
1239 @var{fN} is existing filter number, starting from 0, if no such filter is available
1241 @var{freq} set new frequency parameter.
1242 @var{width} set new width parameter in herz.
1243 @var{gain} set new gain parameter in dB.
1245 Full filter invocation with asendcmd may look like this:
1246 asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...
1251 Pass the audio source unchanged to the output.
1255 Pad the end of an audio stream with silence.
1257 This can be used together with @command{ffmpeg} @option{-shortest} to
1258 extend audio streams to the same length as the video stream.
1260 A description of the accepted options follows.
1264 Set silence packet size. Default value is 4096.
1267 Set the number of samples of silence to add to the end. After the
1268 value is reached, the stream is terminated. This option is mutually
1269 exclusive with @option{whole_len}.
1272 Set the minimum total number of samples in the output audio stream. If
1273 the value is longer than the input audio length, silence is added to
1274 the end, until the value is reached. This option is mutually exclusive
1275 with @option{pad_len}.
1278 If neither the @option{pad_len} nor the @option{whole_len} option is
1279 set, the filter will add silence to the end of the input stream
1282 @subsection Examples
1286 Add 1024 samples of silence to the end of the input:
1292 Make sure the audio output will contain at least 10000 samples, pad
1293 the input with silence if required:
1295 apad=whole_len=10000
1299 Use @command{ffmpeg} to pad the audio input with silence, so that the
1300 video stream will always result the shortest and will be converted
1301 until the end in the output file when using the @option{shortest}
1304 ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
1309 Add a phasing effect to the input audio.
1311 A phaser filter creates series of peaks and troughs in the frequency spectrum.
1312 The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.
1314 A description of the accepted parameters follows.
1318 Set input gain. Default is 0.4.
1321 Set output gain. Default is 0.74
1324 Set delay in milliseconds. Default is 3.0.
1327 Set decay. Default is 0.4.
1330 Set modulation speed in Hz. Default is 0.5.
1333 Set modulation type. Default is triangular.
1335 It accepts the following values:
1344 Audio pulsator is something between an autopanner and a tremolo.
1345 But it can produce funny stereo effects as well. Pulsator changes the volume
1346 of the left and right channel based on a LFO (low frequency oscillator) with
1347 different waveforms and shifted phases.
1348 This filter have the ability to define an offset between left and right
1349 channel. An offset of 0 means that both LFO shapes match each other.
1350 The left and right channel are altered equally - a conventional tremolo.
1351 An offset of 50% means that the shape of the right channel is exactly shifted
1352 in phase (or moved backwards about half of the frequency) - pulsator acts as
1353 an autopanner. At 1 both curves match again. Every setting in between moves the
1354 phase shift gapless between all stages and produces some "bypassing" sounds with
1355 sine and triangle waveforms. The more you set the offset near 1 (starting from
1356 the 0.5) the faster the signal passes from the left to the right speaker.
1358 The filter accepts the following options:
1362 Set input gain. By default it is 1. Range is [0.015625 - 64].
1365 Set output gain. By default it is 1. Range is [0.015625 - 64].
1368 Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
1369 sawup or sawdown. Default is sine.
1372 Set modulation. Define how much of original signal is affected by the LFO.
1375 Set left channel offset. Default is 0. Allowed range is [0 - 1].
1378 Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
1381 Set pulse width. Default is 1. Allowed range is [0 - 2].
1384 Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
1387 Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing
1391 Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing
1395 Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used
1396 if timing is set to hz.
1402 Resample the input audio to the specified parameters, using the
1403 libswresample library. If none are specified then the filter will
1404 automatically convert between its input and output.
1406 This filter is also able to stretch/squeeze the audio data to make it match
1407 the timestamps or to inject silence / cut out audio to make it match the
1408 timestamps, do a combination of both or do neither.
1410 The filter accepts the syntax
1411 [@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate}
1412 expresses a sample rate and @var{resampler_options} is a list of
1413 @var{key}=@var{value} pairs, separated by ":". See the
1414 ffmpeg-resampler manual for the complete list of supported options.
1416 @subsection Examples
1420 Resample the input audio to 44100Hz:
1426 Stretch/squeeze samples to the given timestamps, with a maximum of 1000
1427 samples per second compensation:
1429 aresample=async=1000
1435 Reverse an audio clip.
1437 Warning: This filter requires memory to buffer the entire clip, so trimming
1440 @subsection Examples
1444 Take the first 5 seconds of a clip, and reverse it.
1446 atrim=end=5,areverse
1450 @section asetnsamples
1452 Set the number of samples per each output audio frame.
1454 The last output packet may contain a different number of samples, as
1455 the filter will flush all the remaining samples when the input audio
1458 The filter accepts the following options:
1462 @item nb_out_samples, n
1463 Set the number of frames per each output audio frame. The number is
1464 intended as the number of samples @emph{per each channel}.
1465 Default value is 1024.
1468 If set to 1, the filter will pad the last audio frame with zeroes, so
1469 that the last frame will contain the same number of samples as the
1470 previous ones. Default value is 1.
1473 For example, to set the number of per-frame samples to 1234 and
1474 disable padding for the last frame, use:
1476 asetnsamples=n=1234:p=0
1481 Set the sample rate without altering the PCM data.
1482 This will result in a change of speed and pitch.
1484 The filter accepts the following options:
1487 @item sample_rate, r
1488 Set the output sample rate. Default is 44100 Hz.
1493 Show a line containing various information for each input audio frame.
1494 The input audio is not modified.
1496 The shown line contains a sequence of key/value pairs of the form
1497 @var{key}:@var{value}.
1499 The following values are shown in the output:
1503 The (sequential) number of the input frame, starting from 0.
1506 The presentation timestamp of the input frame, in time base units; the time base
1507 depends on the filter input pad, and is usually 1/@var{sample_rate}.
1510 The presentation timestamp of the input frame in seconds.
1513 position of the frame in the input stream, -1 if this information in
1514 unavailable and/or meaningless (for example in case of synthetic audio)
1523 The sample rate for the audio frame.
1526 The number of samples (per channel) in the frame.
1529 The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar
1530 audio, the data is treated as if all the planes were concatenated.
1532 @item plane_checksums
1533 A list of Adler-32 checksums for each data plane.
1539 Display time domain statistical information about the audio channels.
1540 Statistics are calculated and displayed for each audio channel and,
1541 where applicable, an overall figure is also given.
1543 It accepts the following option:
1546 Short window length in seconds, used for peak and trough RMS measurement.
1547 Default is @code{0.05} (50 milliseconds). Allowed range is @code{[0.1 - 10]}.
1551 Set metadata injection. All the metadata keys are prefixed with @code{lavfi.astats.X},
1552 where @code{X} is channel number starting from 1 or string @code{Overall}. Default is
1555 Available keys for each channel are:
1586 For example full key look like this @code{lavfi.astats.1.DC_offset} or
1587 this @code{lavfi.astats.Overall.Peak_count}.
1589 For description what each key means read below.
1592 Set number of frame after which stats are going to be recalculated.
1593 Default is disabled.
1596 A description of each shown parameter follows:
1600 Mean amplitude displacement from zero.
1603 Minimal sample level.
1606 Maximal sample level.
1608 @item Min difference
1609 Minimal difference between two consecutive samples.
1611 @item Max difference
1612 Maximal difference between two consecutive samples.
1614 @item Mean difference
1615 Mean difference between two consecutive samples.
1616 The average of each difference between two consecutive samples.
1620 Standard peak and RMS level measured in dBFS.
1624 Peak and trough values for RMS level measured over a short window.
1627 Standard ratio of peak to RMS level (note: not in dB).
1630 Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels
1631 (i.e. either @var{Min level} or @var{Max level}).
1634 Number of occasions (not the number of samples) that the signal attained either
1635 @var{Min level} or @var{Max level}.
1638 Overall bit depth of audio. Number of bits used for each sample.
1643 Synchronize audio data with timestamps by squeezing/stretching it and/or
1644 dropping samples/adding silence when needed.
1646 This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
1648 It accepts the following parameters:
1652 Enable stretching/squeezing the data to make it match the timestamps. Disabled
1653 by default. When disabled, time gaps are covered with silence.
1656 The minimum difference between timestamps and audio data (in seconds) to trigger
1657 adding/dropping samples. The default value is 0.1. If you get an imperfect
1658 sync with this filter, try setting this parameter to 0.
1661 The maximum compensation in samples per second. Only relevant with compensate=1.
1662 The default value is 500.
1665 Assume that the first PTS should be this value. The time base is 1 / sample
1666 rate. This allows for padding/trimming at the start of the stream. By default,
1667 no assumption is made about the first frame's expected PTS, so no padding or
1668 trimming is done. For example, this could be set to 0 to pad the beginning with
1669 silence if an audio stream starts after the video stream or to trim any samples
1670 with a negative PTS due to encoder delay.
1678 The filter accepts exactly one parameter, the audio tempo. If not
1679 specified then the filter will assume nominal 1.0 tempo. Tempo must
1680 be in the [0.5, 2.0] range.
1682 @subsection Examples
1686 Slow down audio to 80% tempo:
1692 To speed up audio to 125% tempo:
1700 Trim the input so that the output contains one continuous subpart of the input.
1702 It accepts the following parameters:
1705 Timestamp (in seconds) of the start of the section to keep. I.e. the audio
1706 sample with the timestamp @var{start} will be the first sample in the output.
1709 Specify time of the first audio sample that will be dropped, i.e. the
1710 audio sample immediately preceding the one with the timestamp @var{end} will be
1711 the last sample in the output.
1714 Same as @var{start}, except this option sets the start timestamp in samples
1718 Same as @var{end}, except this option sets the end timestamp in samples instead
1722 The maximum duration of the output in seconds.
1725 The number of the first sample that should be output.
1728 The number of the first sample that should be dropped.
1731 @option{start}, @option{end}, and @option{duration} are expressed as time
1732 duration specifications; see
1733 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
1735 Note that the first two sets of the start/end options and the @option{duration}
1736 option look at the frame timestamp, while the _sample options simply count the
1737 samples that pass through the filter. So start/end_pts and start/end_sample will
1738 give different results when the timestamps are wrong, inexact or do not start at
1739 zero. Also note that this filter does not modify the timestamps. If you wish
1740 to have the output timestamps start at zero, insert the asetpts filter after the
1743 If multiple start or end options are set, this filter tries to be greedy and
1744 keep all samples that match at least one of the specified constraints. To keep
1745 only the part that matches all the constraints at once, chain multiple atrim
1748 The defaults are such that all the input is kept. So it is possible to set e.g.
1749 just the end values to keep everything before the specified time.
1754 Drop everything except the second minute of input:
1756 ffmpeg -i INPUT -af atrim=60:120
1760 Keep only the first 1000 samples:
1762 ffmpeg -i INPUT -af atrim=end_sample=1000
1769 Apply a two-pole Butterworth band-pass filter with central
1770 frequency @var{frequency}, and (3dB-point) band-width width.
1771 The @var{csg} option selects a constant skirt gain (peak gain = Q)
1772 instead of the default: constant 0dB peak gain.
1773 The filter roll off at 6dB per octave (20dB per decade).
1775 The filter accepts the following options:
1779 Set the filter's central frequency. Default is @code{3000}.
1782 Constant skirt gain if set to 1. Defaults to 0.
1785 Set method to specify band-width of filter.
1798 Specify the band-width of a filter in width_type units.
1803 Apply a two-pole Butterworth band-reject filter with central
1804 frequency @var{frequency}, and (3dB-point) band-width @var{width}.
1805 The filter roll off at 6dB per octave (20dB per decade).
1807 The filter accepts the following options:
1811 Set the filter's central frequency. Default is @code{3000}.
1814 Set method to specify band-width of filter.
1827 Specify the band-width of a filter in width_type units.
1832 Boost or cut the bass (lower) frequencies of the audio using a two-pole
1833 shelving filter with a response similar to that of a standard
1834 hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
1836 The filter accepts the following options:
1840 Give the gain at 0 Hz. Its useful range is about -20
1841 (for a large cut) to +20 (for a large boost).
1842 Beware of clipping when using a positive gain.
1845 Set the filter's central frequency and so can be used
1846 to extend or reduce the frequency range to be boosted or cut.
1847 The default value is @code{100} Hz.
1850 Set method to specify band-width of filter.
1863 Determine how steep is the filter's shelf transition.
1868 Apply a biquad IIR filter with the given coefficients.
1869 Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
1870 are the numerator and denominator coefficients respectively.
1873 Bauer stereo to binaural transformation, which improves headphone listening of
1874 stereo audio records.
1876 It accepts the following parameters:
1880 Pre-defined crossfeed level.
1884 Default level (fcut=700, feed=50).
1887 Chu Moy circuit (fcut=700, feed=60).
1890 Jan Meier circuit (fcut=650, feed=95).
1895 Cut frequency (in Hz).
1904 Remap input channels to new locations.
1906 It accepts the following parameters:
1908 @item channel_layout
1909 The channel layout of the output stream.
1912 Map channels from input to output. The argument is a '|'-separated list of
1913 mappings, each in the @code{@var{in_channel}-@var{out_channel}} or
1914 @var{in_channel} form. @var{in_channel} can be either the name of the input
1915 channel (e.g. FL for front left) or its index in the input channel layout.
1916 @var{out_channel} is the name of the output channel or its index in the output
1917 channel layout. If @var{out_channel} is not given then it is implicitly an
1918 index, starting with zero and increasing by one for each mapping.
1921 If no mapping is present, the filter will implicitly map input channels to
1922 output channels, preserving indices.
1924 For example, assuming a 5.1+downmix input MOV file,
1926 ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav
1928 will create an output WAV file tagged as stereo from the downmix channels of
1931 To fix a 5.1 WAV improperly encoded in AAC's native channel order
1933 ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav
1936 @section channelsplit
1938 Split each channel from an input audio stream into a separate output stream.
1940 It accepts the following parameters:
1942 @item channel_layout
1943 The channel layout of the input stream. The default is "stereo".
1946 For example, assuming a stereo input MP3 file,
1948 ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv
1950 will create an output Matroska file with two audio streams, one containing only
1951 the left channel and the other the right channel.
1953 Split a 5.1 WAV file into per-channel files:
1955 ffmpeg -i in.wav -filter_complex
1956 'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
1957 -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
1958 front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
1963 Add a chorus effect to the audio.
1965 Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
1967 Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
1968 constant, with chorus, it is varied using using sinusoidal or triangular modulation.
1969 The modulation depth defines the range the modulated delay is played before or after
1970 the delay. Hence the delayed sound will sound slower or faster, that is the delayed
1971 sound tuned around the original one, like in a chorus where some vocals are slightly
1974 It accepts the following parameters:
1977 Set input gain. Default is 0.4.
1980 Set output gain. Default is 0.4.
1983 Set delays. A typical delay is around 40ms to 60ms.
1995 @subsection Examples
2001 chorus=0.7:0.9:55:0.4:0.25:2
2007 chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
2011 Fuller sounding chorus with three delays:
2013 chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3
2018 Compress or expand the audio's dynamic range.
2020 It accepts the following parameters:
2026 A list of times in seconds for each channel over which the instantaneous level
2027 of the input signal is averaged to determine its volume. @var{attacks} refers to
2028 increase of volume and @var{decays} refers to decrease of volume. For most
2029 situations, the attack time (response to the audio getting louder) should be
2030 shorter than the decay time, because the human ear is more sensitive to sudden
2031 loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and
2032 a typical value for decay is 0.8 seconds.
2033 If specified number of attacks & decays is lower than number of channels, the last
2034 set attack/decay will be used for all remaining channels.
2037 A list of points for the transfer function, specified in dB relative to the
2038 maximum possible signal amplitude. Each key points list must be defined using
2039 the following syntax: @code{x0/y0|x1/y1|x2/y2|....} or
2040 @code{x0/y0 x1/y1 x2/y2 ....}
2042 The input values must be in strictly increasing order but the transfer function
2043 does not have to be monotonically rising. The point @code{0/0} is assumed but
2044 may be overridden (by @code{0/out-dBn}). Typical values for the transfer
2045 function are @code{-70/-70|-60/-20}.
2048 Set the curve radius in dB for all joints. It defaults to 0.01.
2051 Set the additional gain in dB to be applied at all points on the transfer
2052 function. This allows for easy adjustment of the overall gain.
2056 Set an initial volume, in dB, to be assumed for each channel when filtering
2057 starts. This permits the user to supply a nominal level initially, so that, for
2058 example, a very large gain is not applied to initial signal levels before the
2059 companding has begun to operate. A typical value for audio which is initially
2060 quiet is -90 dB. It defaults to 0.
2063 Set a delay, in seconds. The input audio is analyzed immediately, but audio is
2064 delayed before being fed to the volume adjuster. Specifying a delay
2065 approximately equal to the attack/decay times allows the filter to effectively
2066 operate in predictive rather than reactive mode. It defaults to 0.
2070 @subsection Examples
2074 Make music with both quiet and loud passages suitable for listening to in a
2077 compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
2080 Another example for audio with whisper and explosion parts:
2082 compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
2086 A noise gate for when the noise is at a lower level than the signal:
2088 compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
2092 Here is another noise gate, this time for when the noise is at a higher level
2093 than the signal (making it, in some ways, similar to squelch):
2095 compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
2099 2:1 compression starting at -6dB:
2101 compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
2105 2:1 compression starting at -9dB:
2107 compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
2111 2:1 compression starting at -12dB:
2113 compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
2117 2:1 compression starting at -18dB:
2119 compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
2123 3:1 compression starting at -15dB:
2125 compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
2131 compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
2137 compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3
2141 Hard limiter at -6dB:
2143 compand=attacks=0:points=-80/-80|-6/-6|20/-6
2147 Hard limiter at -12dB:
2149 compand=attacks=0:points=-80/-80|-12/-12|20/-12
2153 Hard noise gate at -35 dB:
2155 compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
2161 compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
2165 @section compensationdelay
2167 Compensation Delay Line is a metric based delay to compensate differing
2168 positions of microphones or speakers.
2170 For example, you have recorded guitar with two microphones placed in
2171 different location. Because the front of sound wave has fixed speed in
2172 normal conditions, the phasing of microphones can vary and depends on
2173 their location and interposition. The best sound mix can be achieved when
2174 these microphones are in phase (synchronized). Note that distance of
2175 ~30 cm between microphones makes one microphone to capture signal in
2176 antiphase to another microphone. That makes the final mix sounding moody.
2177 This filter helps to solve phasing problems by adding different delays
2178 to each microphone track and make them synchronized.
2180 The best result can be reached when you take one track as base and
2181 synchronize other tracks one by one with it.
2182 Remember that synchronization/delay tolerance depends on sample rate, too.
2183 Higher sample rates will give more tolerance.
2185 It accepts the following parameters:
2189 Set millimeters distance. This is compensation distance for fine tuning.
2193 Set cm distance. This is compensation distance for tightening distance setup.
2197 Set meters distance. This is compensation distance for hard distance setup.
2201 Set dry amount. Amount of unprocessed (dry) signal.
2205 Set wet amount. Amount of processed (wet) signal.
2209 Set temperature degree in Celsius. This is the temperature of the environment.
2213 @section crystalizer
2214 Simple algorithm to expand audio dynamic range.
2216 The filter accepts the following options:
2220 Sets the intensity of effect (default: 2.0). Must be in range between 0.0
2221 (unchanged sound) to 10.0 (maximum effect).
2224 Enable clipping. By default is enabled.
2228 Apply a DC shift to the audio.
2230 This can be useful to remove a DC offset (caused perhaps by a hardware problem
2231 in the recording chain) from the audio. The effect of a DC offset is reduced
2232 headroom and hence volume. The @ref{astats} filter can be used to determine if
2233 a signal has a DC offset.
2237 Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift
2241 Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
2242 used to prevent clipping.
2246 Dynamic Audio Normalizer.
2248 This filter applies a certain amount of gain to the input audio in order
2249 to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
2250 contrast to more "simple" normalization algorithms, the Dynamic Audio
2251 Normalizer *dynamically* re-adjusts the gain factor to the input audio.
2252 This allows for applying extra gain to the "quiet" sections of the audio
2253 while avoiding distortions or clipping the "loud" sections. In other words:
2254 The Dynamic Audio Normalizer will "even out" the volume of quiet and loud
2255 sections, in the sense that the volume of each section is brought to the
2256 same target level. Note, however, that the Dynamic Audio Normalizer achieves
2257 this goal *without* applying "dynamic range compressing". It will retain 100%
2258 of the dynamic range *within* each section of the audio file.
2262 Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.
2263 Default is 500 milliseconds.
2264 The Dynamic Audio Normalizer processes the input audio in small chunks,
2265 referred to as frames. This is required, because a peak magnitude has no
2266 meaning for just a single sample value. Instead, we need to determine the
2267 peak magnitude for a contiguous sequence of sample values. While a "standard"
2268 normalizer would simply use the peak magnitude of the complete file, the
2269 Dynamic Audio Normalizer determines the peak magnitude individually for each
2270 frame. The length of a frame is specified in milliseconds. By default, the
2271 Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has
2272 been found to give good results with most files.
2273 Note that the exact frame length, in number of samples, will be determined
2274 automatically, based on the sampling rate of the individual input audio file.
2277 Set the Gaussian filter window size. In range from 3 to 301, must be odd
2278 number. Default is 31.
2279 Probably the most important parameter of the Dynamic Audio Normalizer is the
2280 @code{window size} of the Gaussian smoothing filter. The filter's window size
2281 is specified in frames, centered around the current frame. For the sake of
2282 simplicity, this must be an odd number. Consequently, the default value of 31
2283 takes into account the current frame, as well as the 15 preceding frames and
2284 the 15 subsequent frames. Using a larger window results in a stronger
2285 smoothing effect and thus in less gain variation, i.e. slower gain
2286 adaptation. Conversely, using a smaller window results in a weaker smoothing
2287 effect and thus in more gain variation, i.e. faster gain adaptation.
2288 In other words, the more you increase this value, the more the Dynamic Audio
2289 Normalizer will behave like a "traditional" normalization filter. On the
2290 contrary, the more you decrease this value, the more the Dynamic Audio
2291 Normalizer will behave like a dynamic range compressor.
2294 Set the target peak value. This specifies the highest permissible magnitude
2295 level for the normalized audio input. This filter will try to approach the
2296 target peak magnitude as closely as possible, but at the same time it also
2297 makes sure that the normalized signal will never exceed the peak magnitude.
2298 A frame's maximum local gain factor is imposed directly by the target peak
2299 magnitude. The default value is 0.95 and thus leaves a headroom of 5%*.
2300 It is not recommended to go above this value.
2303 Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.
2304 The Dynamic Audio Normalizer determines the maximum possible (local) gain
2305 factor for each input frame, i.e. the maximum gain factor that does not
2306 result in clipping or distortion. The maximum gain factor is determined by
2307 the frame's highest magnitude sample. However, the Dynamic Audio Normalizer
2308 additionally bounds the frame's maximum gain factor by a predetermined
2309 (global) maximum gain factor. This is done in order to avoid excessive gain
2310 factors in "silent" or almost silent frames. By default, the maximum gain
2311 factor is 10.0, For most inputs the default value should be sufficient and
2312 it usually is not recommended to increase this value. Though, for input
2313 with an extremely low overall volume level, it may be necessary to allow even
2314 higher gain factors. Note, however, that the Dynamic Audio Normalizer does
2315 not simply apply a "hard" threshold (i.e. cut off values above the threshold).
2316 Instead, a "sigmoid" threshold function will be applied. This way, the
2317 gain factors will smoothly approach the threshold value, but never exceed that
2321 Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.
2322 By default, the Dynamic Audio Normalizer performs "peak" normalization.
2323 This means that the maximum local gain factor for each frame is defined
2324 (only) by the frame's highest magnitude sample. This way, the samples can
2325 be amplified as much as possible without exceeding the maximum signal
2326 level, i.e. without clipping. Optionally, however, the Dynamic Audio
2327 Normalizer can also take into account the frame's root mean square,
2328 abbreviated RMS. In electrical engineering, the RMS is commonly used to
2329 determine the power of a time-varying signal. It is therefore considered
2330 that the RMS is a better approximation of the "perceived loudness" than
2331 just looking at the signal's peak magnitude. Consequently, by adjusting all
2332 frames to a constant RMS value, a uniform "perceived loudness" can be
2333 established. If a target RMS value has been specified, a frame's local gain
2334 factor is defined as the factor that would result in exactly that RMS value.
2335 Note, however, that the maximum local gain factor is still restricted by the
2336 frame's highest magnitude sample, in order to prevent clipping.
2339 Enable channels coupling. By default is enabled.
2340 By default, the Dynamic Audio Normalizer will amplify all channels by the same
2341 amount. This means the same gain factor will be applied to all channels, i.e.
2342 the maximum possible gain factor is determined by the "loudest" channel.
2343 However, in some recordings, it may happen that the volume of the different
2344 channels is uneven, e.g. one channel may be "quieter" than the other one(s).
2345 In this case, this option can be used to disable the channel coupling. This way,
2346 the gain factor will be determined independently for each channel, depending
2347 only on the individual channel's highest magnitude sample. This allows for
2348 harmonizing the volume of the different channels.
2351 Enable DC bias correction. By default is disabled.
2352 An audio signal (in the time domain) is a sequence of sample values.
2353 In the Dynamic Audio Normalizer these sample values are represented in the
2354 -1.0 to 1.0 range, regardless of the original input format. Normally, the
2355 audio signal, or "waveform", should be centered around the zero point.
2356 That means if we calculate the mean value of all samples in a file, or in a
2357 single frame, then the result should be 0.0 or at least very close to that
2358 value. If, however, there is a significant deviation of the mean value from
2359 0.0, in either positive or negative direction, this is referred to as a
2360 DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic
2361 Audio Normalizer provides optional DC bias correction.
2362 With DC bias correction enabled, the Dynamic Audio Normalizer will determine
2363 the mean value, or "DC correction" offset, of each input frame and subtract
2364 that value from all of the frame's sample values which ensures those samples
2365 are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
2366 boundaries, the DC correction offset values will be interpolated smoothly
2367 between neighbouring frames.
2370 Enable alternative boundary mode. By default is disabled.
2371 The Dynamic Audio Normalizer takes into account a certain neighbourhood
2372 around each frame. This includes the preceding frames as well as the
2373 subsequent frames. However, for the "boundary" frames, located at the very
2374 beginning and at the very end of the audio file, not all neighbouring
2375 frames are available. In particular, for the first few frames in the audio
2376 file, the preceding frames are not known. And, similarly, for the last few
2377 frames in the audio file, the subsequent frames are not known. Thus, the
2378 question arises which gain factors should be assumed for the missing frames
2379 in the "boundary" region. The Dynamic Audio Normalizer implements two modes
2380 to deal with this situation. The default boundary mode assumes a gain factor
2381 of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and
2382 "fade out" at the beginning and at the end of the input, respectively.
2385 Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
2386 By default, the Dynamic Audio Normalizer does not apply "traditional"
2387 compression. This means that signal peaks will not be pruned and thus the
2388 full dynamic range will be retained within each local neighbourhood. However,
2389 in some cases it may be desirable to combine the Dynamic Audio Normalizer's
2390 normalization algorithm with a more "traditional" compression.
2391 For this purpose, the Dynamic Audio Normalizer provides an optional compression
2392 (thresholding) function. If (and only if) the compression feature is enabled,
2393 all input frames will be processed by a soft knee thresholding function prior
2394 to the actual normalization process. Put simply, the thresholding function is
2395 going to prune all samples whose magnitude exceeds a certain threshold value.
2396 However, the Dynamic Audio Normalizer does not simply apply a fixed threshold
2397 value. Instead, the threshold value will be adjusted for each individual
2399 In general, smaller parameters result in stronger compression, and vice versa.
2400 Values below 3.0 are not recommended, because audible distortion may appear.
2405 Make audio easier to listen to on headphones.
2407 This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
2408 so that when listened to on headphones the stereo image is moved from
2409 inside your head (standard for headphones) to outside and in front of
2410 the listener (standard for speakers).
2416 Apply a two-pole peaking equalisation (EQ) filter. With this
2417 filter, the signal-level at and around a selected frequency can
2418 be increased or decreased, whilst (unlike bandpass and bandreject
2419 filters) that at all other frequencies is unchanged.
2421 In order to produce complex equalisation curves, this filter can
2422 be given several times, each with a different central frequency.
2424 The filter accepts the following options:
2428 Set the filter's central frequency in Hz.
2431 Set method to specify band-width of filter.
2444 Specify the band-width of a filter in width_type units.
2447 Set the required gain or attenuation in dB.
2448 Beware of clipping when using a positive gain.
2451 @subsection Examples
2454 Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:
2456 equalizer=f=1000:width_type=h:width=200:g=-10
2460 Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:
2462 equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5
2466 @section extrastereo
2468 Linearly increases the difference between left and right channels which
2469 adds some sort of "live" effect to playback.
2471 The filter accepts the following options:
2475 Sets the difference coefficient (default: 2.5). 0.0 means mono sound
2476 (average of both channels), with 1.0 sound will be unchanged, with
2477 -1.0 left and right channels will be swapped.
2480 Enable clipping. By default is enabled.
2483 @section firequalizer
2484 Apply FIR Equalization using arbitrary frequency response.
2486 The filter accepts the following option:
2490 Set gain curve equation (in dB). The expression can contain variables:
2493 the evaluated frequency
2497 channel number, set to 0 when multichannels evaluation is disabled
2499 channel id, see libavutil/channel_layout.h, set to the first channel id when
2500 multichannels evaluation is disabled
2504 channel_layout, see libavutil/channel_layout.h
2509 @item gain_interpolate(f)
2510 interpolate gain on frequency f based on gain_entry
2512 This option is also available as command. Default is @code{gain_interpolate(f)}.
2515 Set gain entry for gain_interpolate function. The expression can
2519 store gain entry at frequency f with value g
2521 This option is also available as command.
2524 Set filter delay in seconds. Higher value means more accurate.
2525 Default is @code{0.01}.
2528 Set filter accuracy in Hz. Lower value means more accurate.
2529 Default is @code{5}.
2532 Set window function. Acceptable values are:
2535 rectangular window, useful when gain curve is already smooth
2537 hann window (default)
2543 3-terms continuous 1st derivative nuttall window
2545 minimum 3-terms discontinuous nuttall window
2547 4-terms continuous 1st derivative nuttall window
2549 minimum 4-terms discontinuous nuttall (blackman-nuttall) window
2551 blackman-harris window
2555 If enabled, use fixed number of audio samples. This improves speed when
2556 filtering with large delay. Default is disabled.
2559 Enable multichannels evaluation on gain. Default is disabled.
2562 Enable zero phase mode by substracting timestamp to compensate delay.
2563 Default is disabled.
2566 @subsection Examples
2571 firequalizer=gain='if(lt(f,1000), 0, -INF)'
2574 lowpass at 1000 Hz with gain_entry:
2576 firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
2579 custom equalization:
2581 firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
2584 higher delay with zero phase to compensate delay:
2586 firequalizer=delay=0.1:fixed=on:zero_phase=on
2589 lowpass on left channel, highpass on right channel:
2591 firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
2592 :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
2597 Apply a flanging effect to the audio.
2599 The filter accepts the following options:
2603 Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
2606 Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2.
2609 Set percentage regeneration (delayed signal feedback). Range from -95 to 95.
2613 Set percentage of delayed signal mixed with original. Range from 0 to 100.
2614 Default value is 71.
2617 Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
2620 Set swept wave shape, can be @var{triangular} or @var{sinusoidal}.
2621 Default value is @var{sinusoidal}.
2624 Set swept wave percentage-shift for multi channel. Range from 0 to 100.
2625 Default value is 25.
2628 Set delay-line interpolation, @var{linear} or @var{quadratic}.
2629 Default is @var{linear}.
2634 Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with
2635 embedded HDCD codes is expanded into a 20-bit PCM stream.
2637 The filter supports the Peak Extend and Low-level Gain Adjustment features
2638 of HDCD, and detects the Transient Filter flag.
2641 ffmpeg -i HDCD16.flac -af hdcd OUT24.flac
2644 When using the filter with wav, note the default encoding for wav is 16-bit,
2645 so the resulting 20-bit stream will be truncated back to 16-bit. Use something
2646 like @command{-acodec pcm_s24le} after the filter to get 24-bit PCM output.
2648 ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
2649 ffmpeg -i HDCD16.wav -af hdcd -acodec pcm_s24le OUT24.wav
2652 The filter accepts the following options:
2655 @item disable_autoconvert
2656 Disable any automatic format conversion or resampling in the filter graph.
2658 @item process_stereo
2659 Process the stereo channels together. If target_gain does not match between
2660 channels, consider it invalid and use the last valid target_gain.
2663 Set the code detect timer period in ms.
2666 Always extend peaks above -3dBFS even if PE isn't signaled.
2669 Replace audio with a solid tone and adjust the amplitude to signal some
2670 specific aspect of the decoding process. The output file can be loaded in
2671 an audio editor alongside the original to aid analysis.
2673 @code{analyze_mode=pe:force_pe=true} can be used to see all samples above the PE level.
2680 Gain adjustment level at each sample
2682 Samples where peak extend occurs
2684 Samples where the code detect timer is active
2686 Samples where the target gain does not match between channels
2692 Apply a high-pass filter with 3dB point frequency.
2693 The filter can be either single-pole, or double-pole (the default).
2694 The filter roll off at 6dB per pole per octave (20dB per pole per decade).
2696 The filter accepts the following options:
2700 Set frequency in Hz. Default is 3000.
2703 Set number of poles. Default is 2.
2706 Set method to specify band-width of filter.
2719 Specify the band-width of a filter in width_type units.
2720 Applies only to double-pole filter.
2721 The default is 0.707q and gives a Butterworth response.
2726 Join multiple input streams into one multi-channel stream.
2728 It accepts the following parameters:
2732 The number of input streams. It defaults to 2.
2734 @item channel_layout
2735 The desired output channel layout. It defaults to stereo.
2738 Map channels from inputs to output. The argument is a '|'-separated list of
2739 mappings, each in the @code{@var{input_idx}.@var{in_channel}-@var{out_channel}}
2740 form. @var{input_idx} is the 0-based index of the input stream. @var{in_channel}
2741 can be either the name of the input channel (e.g. FL for front left) or its
2742 index in the specified input stream. @var{out_channel} is the name of the output
2746 The filter will attempt to guess the mappings when they are not specified
2747 explicitly. It does so by first trying to find an unused matching input channel
2748 and if that fails it picks the first unused input channel.
2750 Join 3 inputs (with properly set channel layouts):
2752 ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
2755 Build a 5.1 output from 6 single-channel streams:
2757 ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
2758 'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
2764 Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.
2766 To enable compilation of this filter you need to configure FFmpeg with
2767 @code{--enable-ladspa}.
2771 Specifies the name of LADSPA plugin library to load. If the environment
2772 variable @env{LADSPA_PATH} is defined, the LADSPA plugin is searched in
2773 each one of the directories specified by the colon separated list in
2774 @env{LADSPA_PATH}, otherwise in the standard LADSPA paths, which are in
2775 this order: @file{HOME/.ladspa/lib/}, @file{/usr/local/lib/ladspa/},
2776 @file{/usr/lib/ladspa/}.
2779 Specifies the plugin within the library. Some libraries contain only
2780 one plugin, but others contain many of them. If this is not set filter
2781 will list all available plugins within the specified library.
2784 Set the '|' separated list of controls which are zero or more floating point
2785 values that determine the behavior of the loaded plugin (for example delay,
2787 Controls need to be defined using the following syntax:
2788 c0=@var{value0}|c1=@var{value1}|c2=@var{value2}|..., where
2789 @var{valuei} is the value set on the @var{i}-th control.
2790 Alternatively they can be also defined using the following syntax:
2791 @var{value0}|@var{value1}|@var{value2}|..., where
2792 @var{valuei} is the value set on the @var{i}-th control.
2793 If @option{controls} is set to @code{help}, all available controls and
2794 their valid ranges are printed.
2796 @item sample_rate, s
2797 Specify the sample rate, default to 44100. Only used if plugin have
2801 Set the number of samples per channel per each output frame, default
2802 is 1024. Only used if plugin have zero inputs.
2805 Set the minimum duration of the sourced audio. See
2806 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
2807 for the accepted syntax.
2808 Note that the resulting duration may be greater than the specified duration,
2809 as the generated audio is always cut at the end of a complete frame.
2810 If not specified, or the expressed duration is negative, the audio is
2811 supposed to be generated forever.
2812 Only used if plugin have zero inputs.
2816 @subsection Examples
2820 List all available plugins within amp (LADSPA example plugin) library:
2826 List all available controls and their valid ranges for @code{vcf_notch}
2827 plugin from @code{VCF} library:
2829 ladspa=f=vcf:p=vcf_notch:c=help
2833 Simulate low quality audio equipment using @code{Computer Music Toolkit} (CMT)
2836 ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
2840 Add reverberation to the audio using TAP-plugins
2841 (Tom's Audio Processing plugins):
2843 ladspa=file=tap_reverb:tap_reverb
2847 Generate white noise, with 0.2 amplitude:
2849 ladspa=file=cmt:noise_source_white:c=c0=.2
2853 Generate 20 bpm clicks using plugin @code{C* Click - Metronome} from the
2854 @code{C* Audio Plugin Suite} (CAPS) library:
2856 ladspa=file=caps:Click:c=c1=20'
2860 Apply @code{C* Eq10X2 - Stereo 10-band equaliser} effect:
2862 ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
2866 Increase volume by 20dB using fast lookahead limiter from Steve Harris
2867 @code{SWH Plugins} collection:
2869 ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
2873 Attenuate low frequencies using Multiband EQ from Steve Harris
2874 @code{SWH Plugins} collection:
2876 ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
2880 @subsection Commands
2882 This filter supports the following commands:
2885 Modify the @var{N}-th control value.
2887 If the specified value is not valid, it is ignored and prior one is kept.
2892 EBU R128 loudness normalization. Includes both dynamic and linear normalization modes.
2893 Support for both single pass (livestreams, files) and double pass (files) modes.
2894 This algorithm can target IL, LRA, and maximum true peak.
2896 To enable compilation of this filter you need to configure FFmpeg with
2897 @code{--enable-libebur128}.
2899 The filter accepts the following options:
2903 Set integrated loudness target.
2904 Range is -70.0 - -5.0. Default value is -24.0.
2907 Set loudness range target.
2908 Range is 1.0 - 20.0. Default value is 7.0.
2911 Set maximum true peak.
2912 Range is -9.0 - +0.0. Default value is -2.0.
2914 @item measured_I, measured_i
2915 Measured IL of input file.
2916 Range is -99.0 - +0.0.
2918 @item measured_LRA, measured_lra
2919 Measured LRA of input file.
2920 Range is 0.0 - 99.0.
2922 @item measured_TP, measured_tp
2923 Measured true peak of input file.
2924 Range is -99.0 - +99.0.
2926 @item measured_thresh
2927 Measured threshold of input file.
2928 Range is -99.0 - +0.0.
2931 Set offset gain. Gain is applied before the true-peak limiter.
2932 Range is -99.0 - +99.0. Default is +0.0.
2935 Normalize linearly if possible.
2936 measured_I, measured_LRA, measured_TP, and measured_thresh must also
2937 to be specified in order to use this mode.
2938 Options are true or false. Default is true.
2941 Treat mono input files as "dual-mono". If a mono file is intended for playback
2942 on a stereo system, its EBU R128 measurement will be perceptually incorrect.
2943 If set to @code{true}, this option will compensate for this effect.
2944 Multi-channel input files are not affected by this option.
2945 Options are true or false. Default is false.
2948 Set print format for stats. Options are summary, json, or none.
2949 Default value is none.
2954 Apply a low-pass filter with 3dB point frequency.
2955 The filter can be either single-pole or double-pole (the default).
2956 The filter roll off at 6dB per pole per octave (20dB per pole per decade).
2958 The filter accepts the following options:
2962 Set frequency in Hz. Default is 500.
2965 Set number of poles. Default is 2.
2968 Set method to specify band-width of filter.
2981 Specify the band-width of a filter in width_type units.
2982 Applies only to double-pole filter.
2983 The default is 0.707q and gives a Butterworth response.
2989 Mix channels with specific gain levels. The filter accepts the output
2990 channel layout followed by a set of channels definitions.
2992 This filter is also designed to efficiently remap the channels of an audio
2995 The filter accepts parameters of the form:
2996 "@var{l}|@var{outdef}|@var{outdef}|..."
3000 output channel layout or number of channels
3003 output channel specification, of the form:
3004 "@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]"
3007 output channel to define, either a channel name (FL, FR, etc.) or a channel
3008 number (c0, c1, etc.)
3011 multiplicative coefficient for the channel, 1 leaving the volume unchanged
3014 input channel to use, see out_name for details; it is not possible to mix
3015 named and numbered input channels
3018 If the `=' in a channel specification is replaced by `<', then the gains for
3019 that specification will be renormalized so that the total is 1, thus
3020 avoiding clipping noise.
3022 @subsection Mixing examples
3024 For example, if you want to down-mix from stereo to mono, but with a bigger
3025 factor for the left channel:
3027 pan=1c|c0=0.9*c0+0.1*c1
3030 A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
3031 7-channels surround:
3033 pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
3036 Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
3037 that should be preferred (see "-ac" option) unless you have very specific
3040 @subsection Remapping examples
3042 The channel remapping will be effective if, and only if:
3045 @item gain coefficients are zeroes or ones,
3046 @item only one input per channel output,
3049 If all these conditions are satisfied, the filter will notify the user ("Pure
3050 channel mapping detected"), and use an optimized and lossless method to do the
3053 For example, if you have a 5.1 source and want a stereo audio stream by
3054 dropping the extra channels:
3056 pan="stereo| c0=FL | c1=FR"
3059 Given the same source, you can also switch front left and front right channels
3060 and keep the input channel layout:
3062 pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"
3065 If the input is a stereo audio stream, you can mute the front left channel (and
3066 still keep the stereo channel layout) with:
3071 Still with a stereo audio stream input, you can copy the right channel in both
3072 front left and right:
3074 pan="stereo| c0=FR | c1=FR"
3079 ReplayGain scanner filter. This filter takes an audio stream as an input and
3080 outputs it unchanged.
3081 At end of filtering it displays @code{track_gain} and @code{track_peak}.
3085 Convert the audio sample format, sample rate and channel layout. It is
3086 not meant to be used directly.
3089 Apply time-stretching and pitch-shifting with librubberband.
3091 The filter accepts the following options:
3095 Set tempo scale factor.
3098 Set pitch scale factor.
3101 Set transients detector.
3102 Possible values are:
3111 Possible values are:
3120 Possible values are:
3127 Set processing window size.
3128 Possible values are:
3137 Possible values are:
3144 Enable formant preservation when shift pitching.
3145 Possible values are:
3153 Possible values are:
3162 Possible values are:
3169 @section sidechaincompress
3171 This filter acts like normal compressor but has the ability to compress
3172 detected signal using second input signal.
3173 It needs two input streams and returns one output stream.
3174 First input stream will be processed depending on second stream signal.
3175 The filtered signal then can be filtered with other filters in later stages of
3176 processing. See @ref{pan} and @ref{amerge} filter.
3178 The filter accepts the following options:
3182 Set input gain. Default is 1. Range is between 0.015625 and 64.
3185 If a signal of second stream raises above this level it will affect the gain
3186 reduction of first stream.
3187 By default is 0.125. Range is between 0.00097563 and 1.
3190 Set a ratio about which the signal is reduced. 1:2 means that if the level
3191 raised 4dB above the threshold, it will be only 2dB above after the reduction.
3192 Default is 2. Range is between 1 and 20.
3195 Amount of milliseconds the signal has to rise above the threshold before gain
3196 reduction starts. Default is 20. Range is between 0.01 and 2000.
3199 Amount of milliseconds the signal has to fall below the threshold before
3200 reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
3203 Set the amount by how much signal will be amplified after processing.
3204 Default is 2. Range is from 1 and 64.
3207 Curve the sharp knee around the threshold to enter gain reduction more softly.
3208 Default is 2.82843. Range is between 1 and 8.
3211 Choose if the @code{average} level between all channels of side-chain stream
3212 or the louder(@code{maximum}) channel of side-chain stream affects the
3213 reduction. Default is @code{average}.
3216 Should the exact signal be taken in case of @code{peak} or an RMS one in case
3217 of @code{rms}. Default is @code{rms} which is mainly smoother.
3220 Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
3223 How much to use compressed signal in output. Default is 1.
3224 Range is between 0 and 1.
3227 @subsection Examples
3231 Full ffmpeg example taking 2 audio inputs, 1st input to be compressed
3232 depending on the signal of 2nd input and later compressed signal to be
3233 merged with 2nd input:
3235 ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"
3239 @section sidechaingate
3241 A sidechain gate acts like a normal (wideband) gate but has the ability to
3242 filter the detected signal before sending it to the gain reduction stage.
3243 Normally a gate uses the full range signal to detect a level above the
3245 For example: If you cut all lower frequencies from your sidechain signal
3246 the gate will decrease the volume of your track only if not enough highs
3247 appear. With this technique you are able to reduce the resonation of a
3248 natural drum or remove "rumbling" of muted strokes from a heavily distorted
3250 It needs two input streams and returns one output stream.
3251 First input stream will be processed depending on second stream signal.
3253 The filter accepts the following options:
3257 Set input level before filtering.
3258 Default is 1. Allowed range is from 0.015625 to 64.
3261 Set the level of gain reduction when the signal is below the threshold.
3262 Default is 0.06125. Allowed range is from 0 to 1.
3265 If a signal rises above this level the gain reduction is released.
3266 Default is 0.125. Allowed range is from 0 to 1.
3269 Set a ratio about which the signal is reduced.
3270 Default is 2. Allowed range is from 1 to 9000.
3273 Amount of milliseconds the signal has to rise above the threshold before gain
3275 Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
3278 Amount of milliseconds the signal has to fall below the threshold before the
3279 reduction is increased again. Default is 250 milliseconds.
3280 Allowed range is from 0.01 to 9000.
3283 Set amount of amplification of signal after processing.
3284 Default is 1. Allowed range is from 1 to 64.
3287 Curve the sharp knee around the threshold to enter gain reduction more softly.
3288 Default is 2.828427125. Allowed range is from 1 to 8.
3291 Choose if exact signal should be taken for detection or an RMS like one.
3292 Default is rms. Can be peak or rms.
3295 Choose if the average level between all channels or the louder channel affects
3297 Default is average. Can be average or maximum.
3300 Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
3303 @section silencedetect
3305 Detect silence in an audio stream.
3307 This filter logs a message when it detects that the input audio volume is less
3308 or equal to a noise tolerance value for a duration greater or equal to the
3309 minimum detected noise duration.
3311 The printed times and duration are expressed in seconds.
3313 The filter accepts the following options:
3317 Set silence duration until notification (default is 2 seconds).
3320 Set noise tolerance. Can be specified in dB (in case "dB" is appended to the
3321 specified value) or amplitude ratio. Default is -60dB, or 0.001.
3324 @subsection Examples
3328 Detect 5 seconds of silence with -50dB noise tolerance:
3330 silencedetect=n=-50dB:d=5
3334 Complete example with @command{ffmpeg} to detect silence with 0.0001 noise
3335 tolerance in @file{silence.mp3}:
3337 ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
3341 @section silenceremove
3343 Remove silence from the beginning, middle or end of the audio.
3345 The filter accepts the following options:
3349 This value is used to indicate if audio should be trimmed at beginning of
3350 the audio. A value of zero indicates no silence should be trimmed from the
3351 beginning. When specifying a non-zero value, it trims audio up until it
3352 finds non-silence. Normally, when trimming silence from beginning of audio
3353 the @var{start_periods} will be @code{1} but it can be increased to higher
3354 values to trim all audio up to specific count of non-silence periods.
3355 Default value is @code{0}.
3357 @item start_duration
3358 Specify the amount of time that non-silence must be detected before it stops
3359 trimming audio. By increasing the duration, bursts of noises can be treated
3360 as silence and trimmed off. Default value is @code{0}.
3362 @item start_threshold
3363 This indicates what sample value should be treated as silence. For digital
3364 audio, a value of @code{0} may be fine but for audio recorded from analog,
3365 you may wish to increase the value to account for background noise.
3366 Can be specified in dB (in case "dB" is appended to the specified value)
3367 or amplitude ratio. Default value is @code{0}.
3370 Set the count for trimming silence from the end of audio.
3371 To remove silence from the middle of a file, specify a @var{stop_periods}
3372 that is negative. This value is then treated as a positive value and is
3373 used to indicate the effect should restart processing as specified by
3374 @var{start_periods}, making it suitable for removing periods of silence
3375 in the middle of the audio.
3376 Default value is @code{0}.
3379 Specify a duration of silence that must exist before audio is not copied any
3380 more. By specifying a higher duration, silence that is wanted can be left in
3382 Default value is @code{0}.
3384 @item stop_threshold
3385 This is the same as @option{start_threshold} but for trimming silence from
3387 Can be specified in dB (in case "dB" is appended to the specified value)
3388 or amplitude ratio. Default value is @code{0}.
3391 This indicate that @var{stop_duration} length of audio should be left intact
3392 at the beginning of each period of silence.
3393 For example, if you want to remove long pauses between words but do not want
3394 to remove the pauses completely. Default value is @code{0}.
3397 Set how is silence detected. Can be @code{rms} or @code{peak}. Second is faster
3398 and works better with digital silence which is exactly 0.
3399 Default value is @code{rms}.
3402 Set ratio used to calculate size of window for detecting silence.
3403 Default value is @code{0.02}. Allowed range is from @code{0} to @code{10}.
3406 @subsection Examples
3410 The following example shows how this filter can be used to start a recording
3411 that does not contain the delay at the start which usually occurs between
3412 pressing the record button and the start of the performance:
3414 silenceremove=1:5:0.02
3418 Trim all silence encountered from beginning to end where there is more than 1
3419 second of silence in audio:
3421 silenceremove=0:0:0:-1:1:-90dB
3427 SOFAlizer uses head-related transfer functions (HRTFs) to create virtual
3428 loudspeakers around the user for binaural listening via headphones (audio
3429 formats up to 9 channels supported).
3430 The HRTFs are stored in SOFA files (see @url{http://www.sofacoustics.org/} for a database).
3431 SOFAlizer is developed at the Acoustics Research Institute (ARI) of the
3432 Austrian Academy of Sciences.
3434 To enable compilation of this filter you need to configure FFmpeg with
3435 @code{--enable-netcdf}.
3437 The filter accepts the following options:
3441 Set the SOFA file used for rendering.
3444 Set gain applied to audio. Value is in dB. Default is 0.
3447 Set rotation of virtual loudspeakers in deg. Default is 0.
3450 Set elevation of virtual speakers in deg. Default is 0.
3453 Set distance in meters between loudspeakers and the listener with near-field
3454 HRTFs. Default is 1.
3457 Set processing type. Can be @var{time} or @var{freq}. @var{time} is
3458 processing audio in time domain which is slow.
3459 @var{freq} is processing audio in frequency domain which is fast.
3460 Default is @var{freq}.
3463 Set custom positions of virtual loudspeakers. Syntax for this option is:
3464 <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...].
3465 Each virtual loudspeaker is described with short channel name following with
3466 azimuth and elevation in degreees.
3467 Each virtual loudspeaker description is separated by '|'.
3468 For example to override front left and front right channel positions use:
3469 'speakers=FL 45 15|FR 345 15'.
3470 Descriptions with unrecognised channel names are ignored.
3473 @subsection Examples
3477 Using ClubFritz6 sofa file:
3479 sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
3483 Using ClubFritz12 sofa file and bigger radius with small rotation:
3485 sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
3489 Similar as above but with custom speaker positions for front left, front right, rear left and rear right
3490 and also with custom gain:
3492 "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|RL 135|RR 225:gain=28"
3496 @section stereotools
3498 This filter has some handy utilities to manage stereo signals, for converting
3499 M/S stereo recordings to L/R signal while having control over the parameters
3500 or spreading the stereo image of master track.
3502 The filter accepts the following options:
3506 Set input level before filtering for both channels. Defaults is 1.
3507 Allowed range is from 0.015625 to 64.
3510 Set output level after filtering for both channels. Defaults is 1.
3511 Allowed range is from 0.015625 to 64.
3514 Set input balance between both channels. Default is 0.
3515 Allowed range is from -1 to 1.
3518 Set output balance between both channels. Default is 0.
3519 Allowed range is from -1 to 1.
3522 Enable softclipping. Results in analog distortion instead of harsh digital 0dB
3523 clipping. Disabled by default.
3526 Mute the left channel. Disabled by default.
3529 Mute the right channel. Disabled by default.
3532 Change the phase of the left channel. Disabled by default.
3535 Change the phase of the right channel. Disabled by default.
3538 Set stereo mode. Available values are:
3542 Left/Right to Left/Right, this is default.
3545 Left/Right to Mid/Side.
3548 Mid/Side to Left/Right.
3551 Left/Right to Left/Left.
3554 Left/Right to Right/Right.
3557 Left/Right to Left + Right.
3560 Left/Right to Right/Left.
3564 Set level of side signal. Default is 1.
3565 Allowed range is from 0.015625 to 64.
3568 Set balance of side signal. Default is 0.
3569 Allowed range is from -1 to 1.
3572 Set level of the middle signal. Default is 1.
3573 Allowed range is from 0.015625 to 64.
3576 Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
3579 Set stereo base between mono and inversed channels. Default is 0.
3580 Allowed range is from -1 to 1.
3583 Set delay in milliseconds how much to delay left from right channel and
3584 vice versa. Default is 0. Allowed range is from -20 to 20.
3587 Set S/C level. Default is 1. Allowed range is from 1 to 100.
3590 Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.
3593 @subsection Examples
3597 Apply karaoke like effect:
3599 stereotools=mlev=0.015625
3603 Convert M/S signal to L/R:
3605 "stereotools=mode=ms>lr"
3609 @section stereowiden
3611 This filter enhance the stereo effect by suppressing signal common to both
3612 channels and by delaying the signal of left into right and vice versa,
3613 thereby widening the stereo effect.
3615 The filter accepts the following options:
3619 Time in milliseconds of the delay of left signal into right and vice versa.
3620 Default is 20 milliseconds.
3623 Amount of gain in delayed signal into right and vice versa. Gives a delay
3624 effect of left signal in right output and vice versa which gives widening
3625 effect. Default is 0.3.
3628 Cross feed of left into right with inverted phase. This helps in suppressing
3629 the mono. If the value is 1 it will cancel all the signal common to both
3630 channels. Default is 0.3.
3633 Set level of input signal of original channel. Default is 0.8.
3638 Boost or cut treble (upper) frequencies of the audio using a two-pole
3639 shelving filter with a response similar to that of a standard
3640 hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
3642 The filter accepts the following options:
3646 Give the gain at whichever is the lower of ~22 kHz and the
3647 Nyquist frequency. Its useful range is about -20 (for a large cut)
3648 to +20 (for a large boost). Beware of clipping when using a positive gain.
3651 Set the filter's central frequency and so can be used
3652 to extend or reduce the frequency range to be boosted or cut.
3653 The default value is @code{3000} Hz.
3656 Set method to specify band-width of filter.
3669 Determine how steep is the filter's shelf transition.
3674 Sinusoidal amplitude modulation.
3676 The filter accepts the following options:
3680 Modulation frequency in Hertz. Modulation frequencies in the subharmonic range
3681 (20 Hz or lower) will result in a tremolo effect.
3682 This filter may also be used as a ring modulator by specifying
3683 a modulation frequency higher than 20 Hz.
3684 Range is 0.1 - 20000.0. Default value is 5.0 Hz.
3687 Depth of modulation as a percentage. Range is 0.0 - 1.0.
3688 Default value is 0.5.
3693 Sinusoidal phase modulation.
3695 The filter accepts the following options:
3699 Modulation frequency in Hertz.
3700 Range is 0.1 - 20000.0. Default value is 5.0 Hz.
3703 Depth of modulation as a percentage. Range is 0.0 - 1.0.
3704 Default value is 0.5.
3709 Adjust the input audio volume.
3711 It accepts the following parameters:
3715 Set audio volume expression.
3717 Output values are clipped to the maximum value.
3719 The output audio volume is given by the relation:
3721 @var{output_volume} = @var{volume} * @var{input_volume}
3724 The default value for @var{volume} is "1.0".
3727 This parameter represents the mathematical precision.
3729 It determines which input sample formats will be allowed, which affects the
3730 precision of the volume scaling.
3734 8-bit fixed-point; this limits input sample format to U8, S16, and S32.
3736 32-bit floating-point; this limits input sample format to FLT. (default)
3738 64-bit floating-point; this limits input sample format to DBL.
3742 Choose the behaviour on encountering ReplayGain side data in input frames.
3746 Remove ReplayGain side data, ignoring its contents (the default).
3749 Ignore ReplayGain side data, but leave it in the frame.
3752 Prefer the track gain, if present.
3755 Prefer the album gain, if present.
3758 @item replaygain_preamp
3759 Pre-amplification gain in dB to apply to the selected replaygain gain.
3761 Default value for @var{replaygain_preamp} is 0.0.
3764 Set when the volume expression is evaluated.
3766 It accepts the following values:
3769 only evaluate expression once during the filter initialization, or
3770 when the @samp{volume} command is sent
3773 evaluate expression for each incoming frame
3776 Default value is @samp{once}.
3779 The volume expression can contain the following parameters.
3783 frame number (starting at zero)
3786 @item nb_consumed_samples
3787 number of samples consumed by the filter
3789 number of samples in the current frame
3791 original frame position in the file
3797 PTS at start of stream
3799 time at start of stream
3805 last set volume value
3808 Note that when @option{eval} is set to @samp{once} only the
3809 @var{sample_rate} and @var{tb} variables are available, all other
3810 variables will evaluate to NAN.
3812 @subsection Commands
3814 This filter supports the following commands:
3817 Modify the volume expression.
3818 The command accepts the same syntax of the corresponding option.
3820 If the specified expression is not valid, it is kept at its current
3822 @item replaygain_noclip
3823 Prevent clipping by limiting the gain applied.
3825 Default value for @var{replaygain_noclip} is 1.
3829 @subsection Examples
3833 Halve the input audio volume:
3837 volume=volume=-6.0206dB
3840 In all the above example the named key for @option{volume} can be
3841 omitted, for example like in:
3847 Increase input audio power by 6 decibels using fixed-point precision:
3849 volume=volume=6dB:precision=fixed
3853 Fade volume after time 10 with an annihilation period of 5 seconds:
3855 volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame
3859 @section volumedetect
3861 Detect the volume of the input video.
3863 The filter has no parameters. The input is not modified. Statistics about
3864 the volume will be printed in the log when the input stream end is reached.
3866 In particular it will show the mean volume (root mean square), maximum
3867 volume (on a per-sample basis), and the beginning of a histogram of the
3868 registered volume values (from the maximum value to a cumulated 1/1000 of
3871 All volumes are in decibels relative to the maximum PCM value.
3873 @subsection Examples
3875 Here is an excerpt of the output:
3877 [Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB
3878 [Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB
3879 [Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6
3880 [Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62
3881 [Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286
3882 [Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042
3883 [Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551
3884 [Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609
3885 [Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409
3891 The mean square energy is approximately -27 dB, or 10^-2.7.
3893 The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
3895 There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
3898 In other words, raising the volume by +4 dB does not cause any clipping,
3899 raising it by +5 dB causes clipping for 6 samples, etc.
3901 @c man end AUDIO FILTERS
3903 @chapter Audio Sources
3904 @c man begin AUDIO SOURCES
3906 Below is a description of the currently available audio sources.
3910 Buffer audio frames, and make them available to the filter chain.
3912 This source is mainly intended for a programmatic use, in particular
3913 through the interface defined in @file{libavfilter/asrc_abuffer.h}.
3915 It accepts the following parameters:
3919 The timebase which will be used for timestamps of submitted frames. It must be
3920 either a floating-point number or in @var{numerator}/@var{denominator} form.
3923 The sample rate of the incoming audio buffers.
3926 The sample format of the incoming audio buffers.
3927 Either a sample format name or its corresponding integer representation from
3928 the enum AVSampleFormat in @file{libavutil/samplefmt.h}
3930 @item channel_layout
3931 The channel layout of the incoming audio buffers.
3932 Either a channel layout name from channel_layout_map in
3933 @file{libavutil/channel_layout.c} or its corresponding integer representation
3934 from the AV_CH_LAYOUT_* macros in @file{libavutil/channel_layout.h}
3937 The number of channels of the incoming audio buffers.
3938 If both @var{channels} and @var{channel_layout} are specified, then they
3943 @subsection Examples
3946 abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo
3949 will instruct the source to accept planar 16bit signed stereo at 44100Hz.
3950 Since the sample format with name "s16p" corresponds to the number
3951 6 and the "stereo" channel layout corresponds to the value 0x3, this is
3954 abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3
3959 Generate an audio signal specified by an expression.
3961 This source accepts in input one or more expressions (one for each
3962 channel), which are evaluated and used to generate a corresponding
3965 This source accepts the following options:
3969 Set the '|'-separated expressions list for each separate channel. In case the
3970 @option{channel_layout} option is not specified, the selected channel layout
3971 depends on the number of provided expressions. Otherwise the last
3972 specified expression is applied to the remaining output channels.
3974 @item channel_layout, c
3975 Set the channel layout. The number of channels in the specified layout
3976 must be equal to the number of specified expressions.
3979 Set the minimum duration of the sourced audio. See
3980 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
3981 for the accepted syntax.
3982 Note that the resulting duration may be greater than the specified
3983 duration, as the generated audio is always cut at the end of a
3986 If not specified, or the expressed duration is negative, the audio is
3987 supposed to be generated forever.
3990 Set the number of samples per channel per each output frame,
3993 @item sample_rate, s
3994 Specify the sample rate, default to 44100.
3997 Each expression in @var{exprs} can contain the following constants:
4001 number of the evaluated sample, starting from 0
4004 time of the evaluated sample expressed in seconds, starting from 0
4011 @subsection Examples
4021 Generate a sin signal with frequency of 440 Hz, set sample rate to
4024 aevalsrc="sin(440*2*PI*t):s=8000"
4028 Generate a two channels signal, specify the channel layout (Front
4029 Center + Back Center) explicitly:
4031 aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"
4035 Generate white noise:
4037 aevalsrc="-2+random(0)"
4041 Generate an amplitude modulated signal:
4043 aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
4047 Generate 2.5 Hz binaural beats on a 360 Hz carrier:
4049 aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
4056 The null audio source, return unprocessed audio frames. It is mainly useful
4057 as a template and to be employed in analysis / debugging tools, or as
4058 the source for filters which ignore the input data (for example the sox
4061 This source accepts the following options:
4065 @item channel_layout, cl
4067 Specifies the channel layout, and can be either an integer or a string
4068 representing a channel layout. The default value of @var{channel_layout}
4071 Check the channel_layout_map definition in
4072 @file{libavutil/channel_layout.c} for the mapping between strings and
4073 channel layout values.
4075 @item sample_rate, r
4076 Specifies the sample rate, and defaults to 44100.
4079 Set the number of samples per requested frames.
4083 @subsection Examples
4087 Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.
4089 anullsrc=r=48000:cl=4
4093 Do the same operation with a more obvious syntax:
4095 anullsrc=r=48000:cl=mono
4099 All the parameters need to be explicitly defined.
4103 Synthesize a voice utterance using the libflite library.
4105 To enable compilation of this filter you need to configure FFmpeg with
4106 @code{--enable-libflite}.
4108 Note that the flite library is not thread-safe.
4110 The filter accepts the following options:
4115 If set to 1, list the names of the available voices and exit
4116 immediately. Default value is 0.
4119 Set the maximum number of samples per frame. Default value is 512.
4122 Set the filename containing the text to speak.
4125 Set the text to speak.
4128 Set the voice to use for the speech synthesis. Default value is
4129 @code{kal}. See also the @var{list_voices} option.
4132 @subsection Examples
4136 Read from file @file{speech.txt}, and synthesize the text using the
4137 standard flite voice:
4139 flite=textfile=speech.txt
4143 Read the specified text selecting the @code{slt} voice:
4145 flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
4149 Input text to ffmpeg:
4151 ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
4155 Make @file{ffplay} speak the specified text, using @code{flite} and
4156 the @code{lavfi} device:
4158 ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'
4162 For more information about libflite, check:
4163 @url{http://www.speech.cs.cmu.edu/flite/}
4167 Generate a noise audio signal.
4169 The filter accepts the following options:
4172 @item sample_rate, r
4173 Specify the sample rate. Default value is 48000 Hz.
4176 Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value
4180 Specify the duration of the generated audio stream. Not specifying this option
4181 results in noise with an infinite length.
4183 @item color, colour, c
4184 Specify the color of noise. Available noise colors are white, pink, and brown.