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 See @code{ffmpeg -filters} to view which filters have timeline support.
311 @c man end FILTERGRAPH DESCRIPTION
313 @chapter Audio Filters
314 @c man begin AUDIO FILTERS
316 When you configure your FFmpeg build, you can disable any of the
317 existing filters using @code{--disable-filters}.
318 The configure output will show the audio filters included in your
321 Below is a description of the currently available audio filters.
325 A compressor is mainly used to reduce the dynamic range of a signal.
326 Especially modern music is mostly compressed at a high ratio to
327 improve the overall loudness. It's done to get the highest attention
328 of a listener, "fatten" the sound and bring more "power" to the track.
329 If a signal is compressed too much it may sound dull or "dead"
330 afterwards or it may start to "pump" (which could be a powerful effect
331 but can also destroy a track completely).
332 The right compression is the key to reach a professional sound and is
333 the high art of mixing and mastering. Because of its complex settings
334 it may take a long time to get the right feeling for this kind of effect.
336 Compression is done by detecting the volume above a chosen level
337 @code{threshold} and dividing it by the factor set with @code{ratio}.
338 So if you set the threshold to -12dB and your signal reaches -6dB a ratio
339 of 2:1 will result in a signal at -9dB. Because an exact manipulation of
340 the signal would cause distortion of the waveform the reduction can be
341 levelled over the time. This is done by setting "Attack" and "Release".
342 @code{attack} determines how long the signal has to rise above the threshold
343 before any reduction will occur and @code{release} sets the time the signal
344 has to fall below the threshold to reduce the reduction again. Shorter signals
345 than the chosen attack time will be left untouched.
346 The overall reduction of the signal can be made up afterwards with the
347 @code{makeup} setting. So compressing the peaks of a signal about 6dB and
348 raising the makeup to this level results in a signal twice as loud than the
349 source. To gain a softer entry in the compression the @code{knee} flattens the
350 hard edge at the threshold in the range of the chosen decibels.
352 The filter accepts the following options:
356 Set input gain. Default is 1. Range is between 0.015625 and 64.
359 If a signal of second stream rises above this level it will affect the gain
360 reduction of the first stream.
361 By default it is 0.125. Range is between 0.00097563 and 1.
364 Set a ratio by which the signal is reduced. 1:2 means that if the level
365 rose 4dB above the threshold, it will be only 2dB above after the reduction.
366 Default is 2. Range is between 1 and 20.
369 Amount of milliseconds the signal has to rise above the threshold before gain
370 reduction starts. Default is 20. Range is between 0.01 and 2000.
373 Amount of milliseconds the signal has to fall below the threshold before
374 reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
377 Set the amount by how much signal will be amplified after processing.
378 Default is 2. Range is from 1 and 64.
381 Curve the sharp knee around the threshold to enter gain reduction more softly.
382 Default is 2.82843. Range is between 1 and 8.
385 Choose if the @code{average} level between all channels of input stream
386 or the louder(@code{maximum}) channel of input stream affects the
387 reduction. Default is @code{average}.
390 Should the exact signal be taken in case of @code{peak} or an RMS one in case
391 of @code{rms}. Default is @code{rms} which is mostly smoother.
394 How much to use compressed signal in output. Default is 1.
395 Range is between 0 and 1.
400 Apply cross fade from one input audio stream to another input audio stream.
401 The cross fade is applied for specified duration near the end of first stream.
403 The filter accepts the following options:
407 Specify the number of samples for which the cross fade effect has to last.
408 At the end of the cross fade effect the first input audio will be completely
409 silent. Default is 44100.
412 Specify the duration of the cross fade effect. See
413 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
414 for the accepted syntax.
415 By default the duration is determined by @var{nb_samples}.
416 If set this option is used instead of @var{nb_samples}.
419 Should first stream end overlap with second stream start. Default is enabled.
422 Set curve for cross fade transition for first stream.
425 Set curve for cross fade transition for second stream.
427 For description of available curve types see @ref{afade} filter description.
434 Cross fade from one input to another:
436 ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
440 Cross fade from one input to another but without overlapping:
442 ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
448 Reduce audio bit resolution.
450 This filter is bit crusher with enhanced functionality. A bit crusher
451 is used to audibly reduce number of bits an audio signal is sampled
452 with. This doesn't change the bit depth at all, it just produces the
453 effect. Material reduced in bit depth sounds more harsh and "digital".
454 This filter is able to even round to continuous values instead of discrete
456 Additionally it has a D/C offset which results in different crushing of
457 the lower and the upper half of the signal.
458 An Anti-Aliasing setting is able to produce "softer" crushing sounds.
460 Another feature of this filter is the logarithmic mode.
461 This setting switches from linear distances between bits to logarithmic ones.
462 The result is a much more "natural" sounding crusher which doesn't gate low
463 signals for example. The human ear has a logarithmic perception, too
464 so this kind of crushing is much more pleasant.
465 Logarithmic crushing is also able to get anti-aliased.
467 The filter accepts the following options:
483 Can be linear: @code{lin} or logarithmic: @code{log}.
492 Set sample reduction.
495 Enable LFO. By default disabled.
506 Delay one or more audio channels.
508 Samples in delayed channel are filled with silence.
510 The filter accepts the following option:
514 Set list of delays in milliseconds for each channel separated by '|'.
515 At least one delay greater than 0 should be provided.
516 Unused delays will be silently ignored. If number of given delays is
517 smaller than number of channels all remaining channels will not be delayed.
518 If you want to delay exact number of samples, append 'S' to number.
525 Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave
526 the second channel (and any other channels that may be present) unchanged.
532 Delay second channel by 500 samples, the third channel by 700 samples and leave
533 the first channel (and any other channels that may be present) unchanged.
541 Apply echoing to the input audio.
543 Echoes are reflected sound and can occur naturally amongst mountains
544 (and sometimes large buildings) when talking or shouting; digital echo
545 effects emulate this behaviour and are often used to help fill out the
546 sound of a single instrument or vocal. The time difference between the
547 original signal and the reflection is the @code{delay}, and the
548 loudness of the reflected signal is the @code{decay}.
549 Multiple echoes can have different delays and decays.
551 A description of the accepted parameters follows.
555 Set input gain of reflected signal. Default is @code{0.6}.
558 Set output gain of reflected signal. Default is @code{0.3}.
561 Set list of time intervals in milliseconds between original signal and reflections
562 separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}.
563 Default is @code{1000}.
566 Set list of loudnesses of reflected signals separated by '|'.
567 Allowed range for each @code{decay} is @code{(0 - 1.0]}.
568 Default is @code{0.5}.
575 Make it sound as if there are twice as many instruments as are actually playing:
577 aecho=0.8:0.88:60:0.4
581 If delay is very short, then it sound like a (metallic) robot playing music:
587 A longer delay will sound like an open air concert in the mountains:
589 aecho=0.8:0.9:1000:0.3
593 Same as above but with one more mountain:
595 aecho=0.8:0.9:1000|1800:0.3|0.25
600 Audio emphasis filter creates or restores material directly taken from LPs or
601 emphased CDs with different filter curves. E.g. to store music on vinyl the
602 signal has to be altered by a filter first to even out the disadvantages of
603 this recording medium.
604 Once the material is played back the inverse filter has to be applied to
605 restore the distortion of the frequency response.
607 The filter accepts the following options:
617 Set filter mode. For restoring material use @code{reproduction} mode, otherwise
618 use @code{production} mode. Default is @code{reproduction} mode.
621 Set filter type. Selects medium. Can be one of the following:
633 select Compact Disc (CD).
639 select 50µs (FM-KF).
641 select 75µs (FM-KF).
647 Modify an audio signal according to the specified expressions.
649 This filter accepts one or more expressions (one for each channel),
650 which are evaluated and used to modify a corresponding audio signal.
652 It accepts the following parameters:
656 Set the '|'-separated expressions list for each separate channel. If
657 the number of input channels is greater than the number of
658 expressions, the last specified expression is used for the remaining
661 @item channel_layout, c
662 Set output channel layout. If not specified, the channel layout is
663 specified by the number of expressions. If set to @samp{same}, it will
664 use by default the same input channel layout.
667 Each expression in @var{exprs} can contain the following constants and functions:
671 channel number of the current expression
674 number of the evaluated sample, starting from 0
680 time of the evaluated sample expressed in seconds
683 @item nb_out_channels
684 input and output number of channels
687 the value of input channel with number @var{CH}
690 Note: this filter is slow. For faster processing you should use a
699 aeval=val(ch)/2:c=same
703 Invert phase of the second channel:
712 Apply fade-in/out effect to input audio.
714 A description of the accepted parameters follows.
718 Specify the effect type, can be either @code{in} for fade-in, or
719 @code{out} for a fade-out effect. Default is @code{in}.
721 @item start_sample, ss
722 Specify the number of the start sample for starting to apply the fade
723 effect. Default is 0.
726 Specify the number of samples for which the fade effect has to last. At
727 the end of the fade-in effect the output audio will have the same
728 volume as the input audio, at the end of the fade-out transition
729 the output audio will be silence. Default is 44100.
732 Specify the start time of the fade effect. Default is 0.
733 The value must be specified as a time duration; see
734 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
735 for the accepted syntax.
736 If set this option is used instead of @var{start_sample}.
739 Specify the duration of the fade effect. See
740 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
741 for the accepted syntax.
742 At the end of the fade-in effect the output audio will have the same
743 volume as the input audio, at the end of the fade-out transition
744 the output audio will be silence.
745 By default the duration is determined by @var{nb_samples}.
746 If set this option is used instead of @var{nb_samples}.
749 Set curve for fade transition.
751 It accepts the following values:
754 select triangular, linear slope (default)
756 select quarter of sine wave
758 select half of sine wave
760 select exponential sine wave
764 select inverted parabola
778 select inverted quarter of sine wave
780 select inverted half of sine wave
782 select double-exponential seat
784 select double-exponential sigmoid
792 Fade in first 15 seconds of audio:
798 Fade out last 25 seconds of a 900 seconds audio:
800 afade=t=out:st=875:d=25
805 Apply arbitrary expressions to samples in frequency domain.
809 Set frequency domain real expression for each separate channel separated
810 by '|'. Default is "1".
811 If the number of input channels is greater than the number of
812 expressions, the last specified expression is used for the remaining
816 Set frequency domain imaginary expression for each separate channel
817 separated by '|'. If not set, @var{real} option is used.
819 Each expression in @var{real} and @var{imag} can contain the following
827 current frequency bin number
830 number of available bins
833 channel number of the current expression
845 It accepts the following values:
861 Default is @code{w4096}
864 Set window function. Default is @code{hann}.
867 Set window overlap. If set to 1, the recommended overlap for selected
868 window function will be picked. Default is @code{0.75}.
875 Leave almost only low frequencies in audio:
877 afftfilt="1-clip((b/nb)*b,0,1)"
884 Set output format constraints for the input audio. The framework will
885 negotiate the most appropriate format to minimize conversions.
887 It accepts the following parameters:
891 A '|'-separated list of requested sample formats.
894 A '|'-separated list of requested sample rates.
896 @item channel_layouts
897 A '|'-separated list of requested channel layouts.
899 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
900 for the required syntax.
903 If a parameter is omitted, all values are allowed.
905 Force the output to either unsigned 8-bit or signed 16-bit stereo
907 aformat=sample_fmts=u8|s16:channel_layouts=stereo
912 A gate is mainly used to reduce lower parts of a signal. This kind of signal
913 processing reduces disturbing noise between useful signals.
915 Gating is done by detecting the volume below a chosen level @var{threshold}
916 and dividing it by the factor set with @var{ratio}. The bottom of the noise
917 floor is set via @var{range}. Because an exact manipulation of the signal
918 would cause distortion of the waveform the reduction can be levelled over
919 time. This is done by setting @var{attack} and @var{release}.
921 @var{attack} determines how long the signal has to fall below the threshold
922 before any reduction will occur and @var{release} sets the time the signal
923 has to rise above the threshold to reduce the reduction again.
924 Shorter signals than the chosen attack time will be left untouched.
928 Set input level before filtering.
929 Default is 1. Allowed range is from 0.015625 to 64.
932 Set the level of gain reduction when the signal is below the threshold.
933 Default is 0.06125. Allowed range is from 0 to 1.
936 If a signal rises above this level the gain reduction is released.
937 Default is 0.125. Allowed range is from 0 to 1.
940 Set a ratio by which the signal is reduced.
941 Default is 2. Allowed range is from 1 to 9000.
944 Amount of milliseconds the signal has to rise above the threshold before gain
946 Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
949 Amount of milliseconds the signal has to fall below the threshold before the
950 reduction is increased again. Default is 250 milliseconds.
951 Allowed range is from 0.01 to 9000.
954 Set amount of amplification of signal after processing.
955 Default is 1. Allowed range is from 1 to 64.
958 Curve the sharp knee around the threshold to enter gain reduction more softly.
959 Default is 2.828427125. Allowed range is from 1 to 8.
962 Choose if exact signal should be taken for detection or an RMS like one.
963 Default is @code{rms}. Can be @code{peak} or @code{rms}.
966 Choose if the average level between all channels or the louder channel affects
968 Default is @code{average}. Can be @code{average} or @code{maximum}.
973 The limiter prevents an input signal from rising over a desired threshold.
974 This limiter uses lookahead technology to prevent your signal from distorting.
975 It means that there is a small delay after the signal is processed. Keep in mind
976 that the delay it produces is the attack time you set.
978 The filter accepts the following options:
982 Set input gain. Default is 1.
985 Set output gain. Default is 1.
988 Don't let signals above this level pass the limiter. Default is 1.
991 The limiter will reach its attenuation level in this amount of time in
992 milliseconds. Default is 5 milliseconds.
995 Come back from limiting to attenuation 1.0 in this amount of milliseconds.
996 Default is 50 milliseconds.
999 When gain reduction is always needed ASC takes care of releasing to an
1000 average reduction level rather than reaching a reduction of 0 in the release
1004 Select how much the release time is affected by ASC, 0 means nearly no changes
1005 in release time while 1 produces higher release times.
1008 Auto level output signal. Default is enabled.
1009 This normalizes audio back to 0dB if enabled.
1012 Depending on picked setting it is recommended to upsample input 2x or 4x times
1013 with @ref{aresample} before applying this filter.
1017 Apply a two-pole all-pass filter with central frequency (in Hz)
1018 @var{frequency}, and filter-width @var{width}.
1019 An all-pass filter changes the audio's frequency to phase relationship
1020 without changing its frequency to amplitude relationship.
1022 The filter accepts the following options:
1026 Set frequency in Hz.
1029 Set method to specify band-width of filter.
1042 Specify the band-width of a filter in width_type units.
1045 Specify which channels to filter, by default all available are filtered.
1052 The filter accepts the following options:
1056 Set the number of loops.
1059 Set maximal number of samples.
1062 Set first sample of loop.
1068 Merge two or more audio streams into a single multi-channel stream.
1070 The filter accepts the following options:
1075 Set the number of inputs. Default is 2.
1079 If the channel layouts of the inputs are disjoint, and therefore compatible,
1080 the channel layout of the output will be set accordingly and the channels
1081 will be reordered as necessary. If the channel layouts of the inputs are not
1082 disjoint, the output will have all the channels of the first input then all
1083 the channels of the second input, in that order, and the channel layout of
1084 the output will be the default value corresponding to the total number of
1087 For example, if the first input is in 2.1 (FL+FR+LF) and the second input
1088 is FC+BL+BR, then the output will be in 5.1, with the channels in the
1089 following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
1090 first input, b1 is the first channel of the second input).
1092 On the other hand, if both input are in stereo, the output channels will be
1093 in the default order: a1, a2, b1, b2, and the channel layout will be
1094 arbitrarily set to 4.0, which may or may not be the expected value.
1096 All inputs must have the same sample rate, and format.
1098 If inputs do not have the same duration, the output will stop with the
1101 @subsection Examples
1105 Merge two mono files into a stereo stream:
1107 amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
1111 Multiple merges assuming 1 video stream and 6 audio streams in @file{input.mkv}:
1113 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
1119 Mixes multiple audio inputs into a single output.
1121 Note that this filter only supports float samples (the @var{amerge}
1122 and @var{pan} audio filters support many formats). If the @var{amix}
1123 input has integer samples then @ref{aresample} will be automatically
1124 inserted to perform the conversion to float samples.
1128 ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
1130 will mix 3 input audio streams to a single output with the same duration as the
1131 first input and a dropout transition time of 3 seconds.
1133 It accepts the following parameters:
1137 The number of inputs. If unspecified, it defaults to 2.
1140 How to determine the end-of-stream.
1144 The duration of the longest input. (default)
1147 The duration of the shortest input.
1150 The duration of the first input.
1154 @item dropout_transition
1155 The transition time, in seconds, for volume renormalization when an input
1156 stream ends. The default value is 2 seconds.
1160 @section anequalizer
1162 High-order parametric multiband equalizer for each channel.
1164 It accepts the following parameters:
1168 This option string is in format:
1169 "c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..."
1170 Each equalizer band is separated by '|'.
1174 Set channel number to which equalization will be applied.
1175 If input doesn't have that channel the entry is ignored.
1178 Set central frequency for band.
1179 If input doesn't have that frequency the entry is ignored.
1182 Set band width in hertz.
1185 Set band gain in dB.
1188 Set filter type for band, optional, can be:
1192 Butterworth, this is default.
1203 With this option activated frequency response of anequalizer is displayed
1207 Set video stream size. Only useful if curves option is activated.
1210 Set max gain that will be displayed. Only useful if curves option is activated.
1211 Setting this to a reasonable value makes it possible to display gain which is derived from
1212 neighbour bands which are too close to each other and thus produce higher gain
1213 when both are activated.
1216 Set frequency scale used to draw frequency response in video output.
1217 Can be linear or logarithmic. Default is logarithmic.
1220 Set color for each channel curve which is going to be displayed in video stream.
1221 This is list of color names separated by space or by '|'.
1222 Unrecognised or missing colors will be replaced by white color.
1225 @subsection Examples
1229 Lower gain by 10 of central frequency 200Hz and width 100 Hz
1230 for first 2 channels using Chebyshev type 1 filter:
1232 anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
1236 @subsection Commands
1238 This filter supports the following commands:
1241 Alter existing filter parameters.
1242 Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
1244 @var{fN} is existing filter number, starting from 0, if no such filter is available
1246 @var{freq} set new frequency parameter.
1247 @var{width} set new width parameter in herz.
1248 @var{gain} set new gain parameter in dB.
1250 Full filter invocation with asendcmd may look like this:
1251 asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...
1256 Pass the audio source unchanged to the output.
1260 Pad the end of an audio stream with silence.
1262 This can be used together with @command{ffmpeg} @option{-shortest} to
1263 extend audio streams to the same length as the video stream.
1265 A description of the accepted options follows.
1269 Set silence packet size. Default value is 4096.
1272 Set the number of samples of silence to add to the end. After the
1273 value is reached, the stream is terminated. This option is mutually
1274 exclusive with @option{whole_len}.
1277 Set the minimum total number of samples in the output audio stream. If
1278 the value is longer than the input audio length, silence is added to
1279 the end, until the value is reached. This option is mutually exclusive
1280 with @option{pad_len}.
1283 If neither the @option{pad_len} nor the @option{whole_len} option is
1284 set, the filter will add silence to the end of the input stream
1287 @subsection Examples
1291 Add 1024 samples of silence to the end of the input:
1297 Make sure the audio output will contain at least 10000 samples, pad
1298 the input with silence if required:
1300 apad=whole_len=10000
1304 Use @command{ffmpeg} to pad the audio input with silence, so that the
1305 video stream will always result the shortest and will be converted
1306 until the end in the output file when using the @option{shortest}
1309 ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
1314 Add a phasing effect to the input audio.
1316 A phaser filter creates series of peaks and troughs in the frequency spectrum.
1317 The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.
1319 A description of the accepted parameters follows.
1323 Set input gain. Default is 0.4.
1326 Set output gain. Default is 0.74
1329 Set delay in milliseconds. Default is 3.0.
1332 Set decay. Default is 0.4.
1335 Set modulation speed in Hz. Default is 0.5.
1338 Set modulation type. Default is triangular.
1340 It accepts the following values:
1349 Audio pulsator is something between an autopanner and a tremolo.
1350 But it can produce funny stereo effects as well. Pulsator changes the volume
1351 of the left and right channel based on a LFO (low frequency oscillator) with
1352 different waveforms and shifted phases.
1353 This filter have the ability to define an offset between left and right
1354 channel. An offset of 0 means that both LFO shapes match each other.
1355 The left and right channel are altered equally - a conventional tremolo.
1356 An offset of 50% means that the shape of the right channel is exactly shifted
1357 in phase (or moved backwards about half of the frequency) - pulsator acts as
1358 an autopanner. At 1 both curves match again. Every setting in between moves the
1359 phase shift gapless between all stages and produces some "bypassing" sounds with
1360 sine and triangle waveforms. The more you set the offset near 1 (starting from
1361 the 0.5) the faster the signal passes from the left to the right speaker.
1363 The filter accepts the following options:
1367 Set input gain. By default it is 1. Range is [0.015625 - 64].
1370 Set output gain. By default it is 1. Range is [0.015625 - 64].
1373 Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
1374 sawup or sawdown. Default is sine.
1377 Set modulation. Define how much of original signal is affected by the LFO.
1380 Set left channel offset. Default is 0. Allowed range is [0 - 1].
1383 Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
1386 Set pulse width. Default is 1. Allowed range is [0 - 2].
1389 Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
1392 Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing
1396 Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing
1400 Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used
1401 if timing is set to hz.
1407 Resample the input audio to the specified parameters, using the
1408 libswresample library. If none are specified then the filter will
1409 automatically convert between its input and output.
1411 This filter is also able to stretch/squeeze the audio data to make it match
1412 the timestamps or to inject silence / cut out audio to make it match the
1413 timestamps, do a combination of both or do neither.
1415 The filter accepts the syntax
1416 [@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate}
1417 expresses a sample rate and @var{resampler_options} is a list of
1418 @var{key}=@var{value} pairs, separated by ":". See the
1419 @ref{Resampler Options,,the "Resampler Options" section in the
1420 ffmpeg-resampler(1) manual,ffmpeg-resampler}
1421 for the complete list of supported options.
1423 @subsection Examples
1427 Resample the input audio to 44100Hz:
1433 Stretch/squeeze samples to the given timestamps, with a maximum of 1000
1434 samples per second compensation:
1436 aresample=async=1000
1442 Reverse an audio clip.
1444 Warning: This filter requires memory to buffer the entire clip, so trimming
1447 @subsection Examples
1451 Take the first 5 seconds of a clip, and reverse it.
1453 atrim=end=5,areverse
1457 @section asetnsamples
1459 Set the number of samples per each output audio frame.
1461 The last output packet may contain a different number of samples, as
1462 the filter will flush all the remaining samples when the input audio
1465 The filter accepts the following options:
1469 @item nb_out_samples, n
1470 Set the number of frames per each output audio frame. The number is
1471 intended as the number of samples @emph{per each channel}.
1472 Default value is 1024.
1475 If set to 1, the filter will pad the last audio frame with zeroes, so
1476 that the last frame will contain the same number of samples as the
1477 previous ones. Default value is 1.
1480 For example, to set the number of per-frame samples to 1234 and
1481 disable padding for the last frame, use:
1483 asetnsamples=n=1234:p=0
1488 Set the sample rate without altering the PCM data.
1489 This will result in a change of speed and pitch.
1491 The filter accepts the following options:
1494 @item sample_rate, r
1495 Set the output sample rate. Default is 44100 Hz.
1500 Show a line containing various information for each input audio frame.
1501 The input audio is not modified.
1503 The shown line contains a sequence of key/value pairs of the form
1504 @var{key}:@var{value}.
1506 The following values are shown in the output:
1510 The (sequential) number of the input frame, starting from 0.
1513 The presentation timestamp of the input frame, in time base units; the time base
1514 depends on the filter input pad, and is usually 1/@var{sample_rate}.
1517 The presentation timestamp of the input frame in seconds.
1520 position of the frame in the input stream, -1 if this information in
1521 unavailable and/or meaningless (for example in case of synthetic audio)
1530 The sample rate for the audio frame.
1533 The number of samples (per channel) in the frame.
1536 The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar
1537 audio, the data is treated as if all the planes were concatenated.
1539 @item plane_checksums
1540 A list of Adler-32 checksums for each data plane.
1546 Display time domain statistical information about the audio channels.
1547 Statistics are calculated and displayed for each audio channel and,
1548 where applicable, an overall figure is also given.
1550 It accepts the following option:
1553 Short window length in seconds, used for peak and trough RMS measurement.
1554 Default is @code{0.05} (50 milliseconds). Allowed range is @code{[0.1 - 10]}.
1558 Set metadata injection. All the metadata keys are prefixed with @code{lavfi.astats.X},
1559 where @code{X} is channel number starting from 1 or string @code{Overall}. Default is
1562 Available keys for each channel are:
1593 For example full key look like this @code{lavfi.astats.1.DC_offset} or
1594 this @code{lavfi.astats.Overall.Peak_count}.
1596 For description what each key means read below.
1599 Set number of frame after which stats are going to be recalculated.
1600 Default is disabled.
1603 A description of each shown parameter follows:
1607 Mean amplitude displacement from zero.
1610 Minimal sample level.
1613 Maximal sample level.
1615 @item Min difference
1616 Minimal difference between two consecutive samples.
1618 @item Max difference
1619 Maximal difference between two consecutive samples.
1621 @item Mean difference
1622 Mean difference between two consecutive samples.
1623 The average of each difference between two consecutive samples.
1627 Standard peak and RMS level measured in dBFS.
1631 Peak and trough values for RMS level measured over a short window.
1634 Standard ratio of peak to RMS level (note: not in dB).
1637 Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels
1638 (i.e. either @var{Min level} or @var{Max level}).
1641 Number of occasions (not the number of samples) that the signal attained either
1642 @var{Min level} or @var{Max level}.
1645 Overall bit depth of audio. Number of bits used for each sample.
1652 The filter accepts exactly one parameter, the audio tempo. If not
1653 specified then the filter will assume nominal 1.0 tempo. Tempo must
1654 be in the [0.5, 2.0] range.
1656 @subsection Examples
1660 Slow down audio to 80% tempo:
1666 To speed up audio to 125% tempo:
1674 Trim the input so that the output contains one continuous subpart of the input.
1676 It accepts the following parameters:
1679 Timestamp (in seconds) of the start of the section to keep. I.e. the audio
1680 sample with the timestamp @var{start} will be the first sample in the output.
1683 Specify time of the first audio sample that will be dropped, i.e. the
1684 audio sample immediately preceding the one with the timestamp @var{end} will be
1685 the last sample in the output.
1688 Same as @var{start}, except this option sets the start timestamp in samples
1692 Same as @var{end}, except this option sets the end timestamp in samples instead
1696 The maximum duration of the output in seconds.
1699 The number of the first sample that should be output.
1702 The number of the first sample that should be dropped.
1705 @option{start}, @option{end}, and @option{duration} are expressed as time
1706 duration specifications; see
1707 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
1709 Note that the first two sets of the start/end options and the @option{duration}
1710 option look at the frame timestamp, while the _sample options simply count the
1711 samples that pass through the filter. So start/end_pts and start/end_sample will
1712 give different results when the timestamps are wrong, inexact or do not start at
1713 zero. Also note that this filter does not modify the timestamps. If you wish
1714 to have the output timestamps start at zero, insert the asetpts filter after the
1717 If multiple start or end options are set, this filter tries to be greedy and
1718 keep all samples that match at least one of the specified constraints. To keep
1719 only the part that matches all the constraints at once, chain multiple atrim
1722 The defaults are such that all the input is kept. So it is possible to set e.g.
1723 just the end values to keep everything before the specified time.
1728 Drop everything except the second minute of input:
1730 ffmpeg -i INPUT -af atrim=60:120
1734 Keep only the first 1000 samples:
1736 ffmpeg -i INPUT -af atrim=end_sample=1000
1743 Apply a two-pole Butterworth band-pass filter with central
1744 frequency @var{frequency}, and (3dB-point) band-width width.
1745 The @var{csg} option selects a constant skirt gain (peak gain = Q)
1746 instead of the default: constant 0dB peak gain.
1747 The filter roll off at 6dB per octave (20dB per decade).
1749 The filter accepts the following options:
1753 Set the filter's central frequency. Default is @code{3000}.
1756 Constant skirt gain if set to 1. Defaults to 0.
1759 Set method to specify band-width of filter.
1772 Specify the band-width of a filter in width_type units.
1775 Specify which channels to filter, by default all available are filtered.
1780 Apply a two-pole Butterworth band-reject filter with central
1781 frequency @var{frequency}, and (3dB-point) band-width @var{width}.
1782 The filter roll off at 6dB per octave (20dB per decade).
1784 The filter accepts the following options:
1788 Set the filter's central frequency. Default is @code{3000}.
1791 Set method to specify band-width of filter.
1804 Specify the band-width of a filter in width_type units.
1807 Specify which channels to filter, by default all available are filtered.
1812 Boost or cut the bass (lower) frequencies of the audio using a two-pole
1813 shelving filter with a response similar to that of a standard
1814 hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
1816 The filter accepts the following options:
1820 Give the gain at 0 Hz. Its useful range is about -20
1821 (for a large cut) to +20 (for a large boost).
1822 Beware of clipping when using a positive gain.
1825 Set the filter's central frequency and so can be used
1826 to extend or reduce the frequency range to be boosted or cut.
1827 The default value is @code{100} Hz.
1830 Set method to specify band-width of filter.
1843 Determine how steep is the filter's shelf transition.
1846 Specify which channels to filter, by default all available are filtered.
1851 Apply a biquad IIR filter with the given coefficients.
1852 Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
1853 are the numerator and denominator coefficients respectively.
1854 and @var{channels}, @var{c} specify which channels to filter, by default all
1855 available are filtered.
1858 Bauer stereo to binaural transformation, which improves headphone listening of
1859 stereo audio records.
1861 It accepts the following parameters:
1865 Pre-defined crossfeed level.
1869 Default level (fcut=700, feed=50).
1872 Chu Moy circuit (fcut=700, feed=60).
1875 Jan Meier circuit (fcut=650, feed=95).
1880 Cut frequency (in Hz).
1889 Remap input channels to new locations.
1891 It accepts the following parameters:
1894 Map channels from input to output. The argument is a '|'-separated list of
1895 mappings, each in the @code{@var{in_channel}-@var{out_channel}} or
1896 @var{in_channel} form. @var{in_channel} can be either the name of the input
1897 channel (e.g. FL for front left) or its index in the input channel layout.
1898 @var{out_channel} is the name of the output channel or its index in the output
1899 channel layout. If @var{out_channel} is not given then it is implicitly an
1900 index, starting with zero and increasing by one for each mapping.
1902 @item channel_layout
1903 The channel layout of the output stream.
1906 If no mapping is present, the filter will implicitly map input channels to
1907 output channels, preserving indices.
1909 For example, assuming a 5.1+downmix input MOV file,
1911 ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav
1913 will create an output WAV file tagged as stereo from the downmix channels of
1916 To fix a 5.1 WAV improperly encoded in AAC's native channel order
1918 ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav
1921 @section channelsplit
1923 Split each channel from an input audio stream into a separate output stream.
1925 It accepts the following parameters:
1927 @item channel_layout
1928 The channel layout of the input stream. The default is "stereo".
1931 For example, assuming a stereo input MP3 file,
1933 ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv
1935 will create an output Matroska file with two audio streams, one containing only
1936 the left channel and the other the right channel.
1938 Split a 5.1 WAV file into per-channel files:
1940 ffmpeg -i in.wav -filter_complex
1941 'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
1942 -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
1943 front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
1948 Add a chorus effect to the audio.
1950 Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
1952 Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
1953 constant, with chorus, it is varied using using sinusoidal or triangular modulation.
1954 The modulation depth defines the range the modulated delay is played before or after
1955 the delay. Hence the delayed sound will sound slower or faster, that is the delayed
1956 sound tuned around the original one, like in a chorus where some vocals are slightly
1959 It accepts the following parameters:
1962 Set input gain. Default is 0.4.
1965 Set output gain. Default is 0.4.
1968 Set delays. A typical delay is around 40ms to 60ms.
1980 @subsection Examples
1986 chorus=0.7:0.9:55:0.4:0.25:2
1992 chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
1996 Fuller sounding chorus with three delays:
1998 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
2003 Compress or expand the audio's dynamic range.
2005 It accepts the following parameters:
2011 A list of times in seconds for each channel over which the instantaneous level
2012 of the input signal is averaged to determine its volume. @var{attacks} refers to
2013 increase of volume and @var{decays} refers to decrease of volume. For most
2014 situations, the attack time (response to the audio getting louder) should be
2015 shorter than the decay time, because the human ear is more sensitive to sudden
2016 loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and
2017 a typical value for decay is 0.8 seconds.
2018 If specified number of attacks & decays is lower than number of channels, the last
2019 set attack/decay will be used for all remaining channels.
2022 A list of points for the transfer function, specified in dB relative to the
2023 maximum possible signal amplitude. Each key points list must be defined using
2024 the following syntax: @code{x0/y0|x1/y1|x2/y2|....} or
2025 @code{x0/y0 x1/y1 x2/y2 ....}
2027 The input values must be in strictly increasing order but the transfer function
2028 does not have to be monotonically rising. The point @code{0/0} is assumed but
2029 may be overridden (by @code{0/out-dBn}). Typical values for the transfer
2030 function are @code{-70/-70|-60/-20}.
2033 Set the curve radius in dB for all joints. It defaults to 0.01.
2036 Set the additional gain in dB to be applied at all points on the transfer
2037 function. This allows for easy adjustment of the overall gain.
2041 Set an initial volume, in dB, to be assumed for each channel when filtering
2042 starts. This permits the user to supply a nominal level initially, so that, for
2043 example, a very large gain is not applied to initial signal levels before the
2044 companding has begun to operate. A typical value for audio which is initially
2045 quiet is -90 dB. It defaults to 0.
2048 Set a delay, in seconds. The input audio is analyzed immediately, but audio is
2049 delayed before being fed to the volume adjuster. Specifying a delay
2050 approximately equal to the attack/decay times allows the filter to effectively
2051 operate in predictive rather than reactive mode. It defaults to 0.
2055 @subsection Examples
2059 Make music with both quiet and loud passages suitable for listening to in a
2062 compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
2065 Another example for audio with whisper and explosion parts:
2067 compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
2071 A noise gate for when the noise is at a lower level than the signal:
2073 compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
2077 Here is another noise gate, this time for when the noise is at a higher level
2078 than the signal (making it, in some ways, similar to squelch):
2080 compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
2084 2:1 compression starting at -6dB:
2086 compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
2090 2:1 compression starting at -9dB:
2092 compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
2096 2:1 compression starting at -12dB:
2098 compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
2102 2:1 compression starting at -18dB:
2104 compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
2108 3:1 compression starting at -15dB:
2110 compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
2116 compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
2122 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
2126 Hard limiter at -6dB:
2128 compand=attacks=0:points=-80/-80|-6/-6|20/-6
2132 Hard limiter at -12dB:
2134 compand=attacks=0:points=-80/-80|-12/-12|20/-12
2138 Hard noise gate at -35 dB:
2140 compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
2146 compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
2150 @section compensationdelay
2152 Compensation Delay Line is a metric based delay to compensate differing
2153 positions of microphones or speakers.
2155 For example, you have recorded guitar with two microphones placed in
2156 different location. Because the front of sound wave has fixed speed in
2157 normal conditions, the phasing of microphones can vary and depends on
2158 their location and interposition. The best sound mix can be achieved when
2159 these microphones are in phase (synchronized). Note that distance of
2160 ~30 cm between microphones makes one microphone to capture signal in
2161 antiphase to another microphone. That makes the final mix sounding moody.
2162 This filter helps to solve phasing problems by adding different delays
2163 to each microphone track and make them synchronized.
2165 The best result can be reached when you take one track as base and
2166 synchronize other tracks one by one with it.
2167 Remember that synchronization/delay tolerance depends on sample rate, too.
2168 Higher sample rates will give more tolerance.
2170 It accepts the following parameters:
2174 Set millimeters distance. This is compensation distance for fine tuning.
2178 Set cm distance. This is compensation distance for tightening distance setup.
2182 Set meters distance. This is compensation distance for hard distance setup.
2186 Set dry amount. Amount of unprocessed (dry) signal.
2190 Set wet amount. Amount of processed (wet) signal.
2194 Set temperature degree in Celsius. This is the temperature of the environment.
2198 @section crystalizer
2199 Simple algorithm to expand audio dynamic range.
2201 The filter accepts the following options:
2205 Sets the intensity of effect (default: 2.0). Must be in range between 0.0
2206 (unchanged sound) to 10.0 (maximum effect).
2209 Enable clipping. By default is enabled.
2213 Apply a DC shift to the audio.
2215 This can be useful to remove a DC offset (caused perhaps by a hardware problem
2216 in the recording chain) from the audio. The effect of a DC offset is reduced
2217 headroom and hence volume. The @ref{astats} filter can be used to determine if
2218 a signal has a DC offset.
2222 Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift
2226 Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
2227 used to prevent clipping.
2231 Dynamic Audio Normalizer.
2233 This filter applies a certain amount of gain to the input audio in order
2234 to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
2235 contrast to more "simple" normalization algorithms, the Dynamic Audio
2236 Normalizer *dynamically* re-adjusts the gain factor to the input audio.
2237 This allows for applying extra gain to the "quiet" sections of the audio
2238 while avoiding distortions or clipping the "loud" sections. In other words:
2239 The Dynamic Audio Normalizer will "even out" the volume of quiet and loud
2240 sections, in the sense that the volume of each section is brought to the
2241 same target level. Note, however, that the Dynamic Audio Normalizer achieves
2242 this goal *without* applying "dynamic range compressing". It will retain 100%
2243 of the dynamic range *within* each section of the audio file.
2247 Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.
2248 Default is 500 milliseconds.
2249 The Dynamic Audio Normalizer processes the input audio in small chunks,
2250 referred to as frames. This is required, because a peak magnitude has no
2251 meaning for just a single sample value. Instead, we need to determine the
2252 peak magnitude for a contiguous sequence of sample values. While a "standard"
2253 normalizer would simply use the peak magnitude of the complete file, the
2254 Dynamic Audio Normalizer determines the peak magnitude individually for each
2255 frame. The length of a frame is specified in milliseconds. By default, the
2256 Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has
2257 been found to give good results with most files.
2258 Note that the exact frame length, in number of samples, will be determined
2259 automatically, based on the sampling rate of the individual input audio file.
2262 Set the Gaussian filter window size. In range from 3 to 301, must be odd
2263 number. Default is 31.
2264 Probably the most important parameter of the Dynamic Audio Normalizer is the
2265 @code{window size} of the Gaussian smoothing filter. The filter's window size
2266 is specified in frames, centered around the current frame. For the sake of
2267 simplicity, this must be an odd number. Consequently, the default value of 31
2268 takes into account the current frame, as well as the 15 preceding frames and
2269 the 15 subsequent frames. Using a larger window results in a stronger
2270 smoothing effect and thus in less gain variation, i.e. slower gain
2271 adaptation. Conversely, using a smaller window results in a weaker smoothing
2272 effect and thus in more gain variation, i.e. faster gain adaptation.
2273 In other words, the more you increase this value, the more the Dynamic Audio
2274 Normalizer will behave like a "traditional" normalization filter. On the
2275 contrary, the more you decrease this value, the more the Dynamic Audio
2276 Normalizer will behave like a dynamic range compressor.
2279 Set the target peak value. This specifies the highest permissible magnitude
2280 level for the normalized audio input. This filter will try to approach the
2281 target peak magnitude as closely as possible, but at the same time it also
2282 makes sure that the normalized signal will never exceed the peak magnitude.
2283 A frame's maximum local gain factor is imposed directly by the target peak
2284 magnitude. The default value is 0.95 and thus leaves a headroom of 5%*.
2285 It is not recommended to go above this value.
2288 Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.
2289 The Dynamic Audio Normalizer determines the maximum possible (local) gain
2290 factor for each input frame, i.e. the maximum gain factor that does not
2291 result in clipping or distortion. The maximum gain factor is determined by
2292 the frame's highest magnitude sample. However, the Dynamic Audio Normalizer
2293 additionally bounds the frame's maximum gain factor by a predetermined
2294 (global) maximum gain factor. This is done in order to avoid excessive gain
2295 factors in "silent" or almost silent frames. By default, the maximum gain
2296 factor is 10.0, For most inputs the default value should be sufficient and
2297 it usually is not recommended to increase this value. Though, for input
2298 with an extremely low overall volume level, it may be necessary to allow even
2299 higher gain factors. Note, however, that the Dynamic Audio Normalizer does
2300 not simply apply a "hard" threshold (i.e. cut off values above the threshold).
2301 Instead, a "sigmoid" threshold function will be applied. This way, the
2302 gain factors will smoothly approach the threshold value, but never exceed that
2306 Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.
2307 By default, the Dynamic Audio Normalizer performs "peak" normalization.
2308 This means that the maximum local gain factor for each frame is defined
2309 (only) by the frame's highest magnitude sample. This way, the samples can
2310 be amplified as much as possible without exceeding the maximum signal
2311 level, i.e. without clipping. Optionally, however, the Dynamic Audio
2312 Normalizer can also take into account the frame's root mean square,
2313 abbreviated RMS. In electrical engineering, the RMS is commonly used to
2314 determine the power of a time-varying signal. It is therefore considered
2315 that the RMS is a better approximation of the "perceived loudness" than
2316 just looking at the signal's peak magnitude. Consequently, by adjusting all
2317 frames to a constant RMS value, a uniform "perceived loudness" can be
2318 established. If a target RMS value has been specified, a frame's local gain
2319 factor is defined as the factor that would result in exactly that RMS value.
2320 Note, however, that the maximum local gain factor is still restricted by the
2321 frame's highest magnitude sample, in order to prevent clipping.
2324 Enable channels coupling. By default is enabled.
2325 By default, the Dynamic Audio Normalizer will amplify all channels by the same
2326 amount. This means the same gain factor will be applied to all channels, i.e.
2327 the maximum possible gain factor is determined by the "loudest" channel.
2328 However, in some recordings, it may happen that the volume of the different
2329 channels is uneven, e.g. one channel may be "quieter" than the other one(s).
2330 In this case, this option can be used to disable the channel coupling. This way,
2331 the gain factor will be determined independently for each channel, depending
2332 only on the individual channel's highest magnitude sample. This allows for
2333 harmonizing the volume of the different channels.
2336 Enable DC bias correction. By default is disabled.
2337 An audio signal (in the time domain) is a sequence of sample values.
2338 In the Dynamic Audio Normalizer these sample values are represented in the
2339 -1.0 to 1.0 range, regardless of the original input format. Normally, the
2340 audio signal, or "waveform", should be centered around the zero point.
2341 That means if we calculate the mean value of all samples in a file, or in a
2342 single frame, then the result should be 0.0 or at least very close to that
2343 value. If, however, there is a significant deviation of the mean value from
2344 0.0, in either positive or negative direction, this is referred to as a
2345 DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic
2346 Audio Normalizer provides optional DC bias correction.
2347 With DC bias correction enabled, the Dynamic Audio Normalizer will determine
2348 the mean value, or "DC correction" offset, of each input frame and subtract
2349 that value from all of the frame's sample values which ensures those samples
2350 are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
2351 boundaries, the DC correction offset values will be interpolated smoothly
2352 between neighbouring frames.
2355 Enable alternative boundary mode. By default is disabled.
2356 The Dynamic Audio Normalizer takes into account a certain neighbourhood
2357 around each frame. This includes the preceding frames as well as the
2358 subsequent frames. However, for the "boundary" frames, located at the very
2359 beginning and at the very end of the audio file, not all neighbouring
2360 frames are available. In particular, for the first few frames in the audio
2361 file, the preceding frames are not known. And, similarly, for the last few
2362 frames in the audio file, the subsequent frames are not known. Thus, the
2363 question arises which gain factors should be assumed for the missing frames
2364 in the "boundary" region. The Dynamic Audio Normalizer implements two modes
2365 to deal with this situation. The default boundary mode assumes a gain factor
2366 of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and
2367 "fade out" at the beginning and at the end of the input, respectively.
2370 Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
2371 By default, the Dynamic Audio Normalizer does not apply "traditional"
2372 compression. This means that signal peaks will not be pruned and thus the
2373 full dynamic range will be retained within each local neighbourhood. However,
2374 in some cases it may be desirable to combine the Dynamic Audio Normalizer's
2375 normalization algorithm with a more "traditional" compression.
2376 For this purpose, the Dynamic Audio Normalizer provides an optional compression
2377 (thresholding) function. If (and only if) the compression feature is enabled,
2378 all input frames will be processed by a soft knee thresholding function prior
2379 to the actual normalization process. Put simply, the thresholding function is
2380 going to prune all samples whose magnitude exceeds a certain threshold value.
2381 However, the Dynamic Audio Normalizer does not simply apply a fixed threshold
2382 value. Instead, the threshold value will be adjusted for each individual
2384 In general, smaller parameters result in stronger compression, and vice versa.
2385 Values below 3.0 are not recommended, because audible distortion may appear.
2390 Make audio easier to listen to on headphones.
2392 This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
2393 so that when listened to on headphones the stereo image is moved from
2394 inside your head (standard for headphones) to outside and in front of
2395 the listener (standard for speakers).
2401 Apply a two-pole peaking equalisation (EQ) filter. With this
2402 filter, the signal-level at and around a selected frequency can
2403 be increased or decreased, whilst (unlike bandpass and bandreject
2404 filters) that at all other frequencies is unchanged.
2406 In order to produce complex equalisation curves, this filter can
2407 be given several times, each with a different central frequency.
2409 The filter accepts the following options:
2413 Set the filter's central frequency in Hz.
2416 Set method to specify band-width of filter.
2429 Specify the band-width of a filter in width_type units.
2432 Set the required gain or attenuation in dB.
2433 Beware of clipping when using a positive gain.
2436 Specify which channels to filter, by default all available are filtered.
2439 @subsection Examples
2442 Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:
2444 equalizer=f=1000:width_type=h:width=200:g=-10
2448 Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:
2450 equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5
2454 @section extrastereo
2456 Linearly increases the difference between left and right channels which
2457 adds some sort of "live" effect to playback.
2459 The filter accepts the following options:
2463 Sets the difference coefficient (default: 2.5). 0.0 means mono sound
2464 (average of both channels), with 1.0 sound will be unchanged, with
2465 -1.0 left and right channels will be swapped.
2468 Enable clipping. By default is enabled.
2471 @section firequalizer
2472 Apply FIR Equalization using arbitrary frequency response.
2474 The filter accepts the following option:
2478 Set gain curve equation (in dB). The expression can contain variables:
2481 the evaluated frequency
2485 channel number, set to 0 when multichannels evaluation is disabled
2487 channel id, see libavutil/channel_layout.h, set to the first channel id when
2488 multichannels evaluation is disabled
2492 channel_layout, see libavutil/channel_layout.h
2497 @item gain_interpolate(f)
2498 interpolate gain on frequency f based on gain_entry
2499 @item cubic_interpolate(f)
2500 same as gain_interpolate, but smoother
2502 This option is also available as command. Default is @code{gain_interpolate(f)}.
2505 Set gain entry for gain_interpolate function. The expression can
2509 store gain entry at frequency f with value g
2511 This option is also available as command.
2514 Set filter delay in seconds. Higher value means more accurate.
2515 Default is @code{0.01}.
2518 Set filter accuracy in Hz. Lower value means more accurate.
2519 Default is @code{5}.
2522 Set window function. Acceptable values are:
2525 rectangular window, useful when gain curve is already smooth
2527 hann window (default)
2533 3-terms continuous 1st derivative nuttall window
2535 minimum 3-terms discontinuous nuttall window
2537 4-terms continuous 1st derivative nuttall window
2539 minimum 4-terms discontinuous nuttall (blackman-nuttall) window
2541 blackman-harris window
2547 If enabled, use fixed number of audio samples. This improves speed when
2548 filtering with large delay. Default is disabled.
2551 Enable multichannels evaluation on gain. Default is disabled.
2554 Enable zero phase mode by subtracting timestamp to compensate delay.
2555 Default is disabled.
2558 Set scale used by gain. Acceptable values are:
2561 linear frequency, linear gain
2563 linear frequency, logarithmic (in dB) gain (default)
2565 logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain
2567 logarithmic frequency, logarithmic gain
2571 Set file for dumping, suitable for gnuplot.
2574 Set scale for dumpfile. Acceptable values are same with scale option.
2578 Enable 2-channel convolution using complex FFT. This improves speed significantly.
2579 Default is disabled.
2582 @subsection Examples
2587 firequalizer=gain='if(lt(f,1000), 0, -INF)'
2590 lowpass at 1000 Hz with gain_entry:
2592 firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
2595 custom equalization:
2597 firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
2600 higher delay with zero phase to compensate delay:
2602 firequalizer=delay=0.1:fixed=on:zero_phase=on
2605 lowpass on left channel, highpass on right channel:
2607 firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
2608 :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
2613 Apply a flanging effect to the audio.
2615 The filter accepts the following options:
2619 Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
2622 Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2.
2625 Set percentage regeneration (delayed signal feedback). Range from -95 to 95.
2629 Set percentage of delayed signal mixed with original. Range from 0 to 100.
2630 Default value is 71.
2633 Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
2636 Set swept wave shape, can be @var{triangular} or @var{sinusoidal}.
2637 Default value is @var{sinusoidal}.
2640 Set swept wave percentage-shift for multi channel. Range from 0 to 100.
2641 Default value is 25.
2644 Set delay-line interpolation, @var{linear} or @var{quadratic}.
2645 Default is @var{linear}.
2650 Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with
2651 embedded HDCD codes is expanded into a 20-bit PCM stream.
2653 The filter supports the Peak Extend and Low-level Gain Adjustment features
2654 of HDCD, and detects the Transient Filter flag.
2657 ffmpeg -i HDCD16.flac -af hdcd OUT24.flac
2660 When using the filter with wav, note the default encoding for wav is 16-bit,
2661 so the resulting 20-bit stream will be truncated back to 16-bit. Use something
2662 like @command{-acodec pcm_s24le} after the filter to get 24-bit PCM output.
2664 ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
2665 ffmpeg -i HDCD16.wav -af hdcd -acodec pcm_s24le OUT24.wav
2668 The filter accepts the following options:
2671 @item disable_autoconvert
2672 Disable any automatic format conversion or resampling in the filter graph.
2674 @item process_stereo
2675 Process the stereo channels together. If target_gain does not match between
2676 channels, consider it invalid and use the last valid target_gain.
2679 Set the code detect timer period in ms.
2682 Always extend peaks above -3dBFS even if PE isn't signaled.
2685 Replace audio with a solid tone and adjust the amplitude to signal some
2686 specific aspect of the decoding process. The output file can be loaded in
2687 an audio editor alongside the original to aid analysis.
2689 @code{analyze_mode=pe:force_pe=true} can be used to see all samples above the PE level.
2696 Gain adjustment level at each sample
2698 Samples where peak extend occurs
2700 Samples where the code detect timer is active
2702 Samples where the target gain does not match between channels
2708 Apply a high-pass filter with 3dB point frequency.
2709 The filter can be either single-pole, or double-pole (the default).
2710 The filter roll off at 6dB per pole per octave (20dB per pole per decade).
2712 The filter accepts the following options:
2716 Set frequency in Hz. Default is 3000.
2719 Set number of poles. Default is 2.
2722 Set method to specify band-width of filter.
2735 Specify the band-width of a filter in width_type units.
2736 Applies only to double-pole filter.
2737 The default is 0.707q and gives a Butterworth response.
2740 Specify which channels to filter, by default all available are filtered.
2745 Join multiple input streams into one multi-channel stream.
2747 It accepts the following parameters:
2751 The number of input streams. It defaults to 2.
2753 @item channel_layout
2754 The desired output channel layout. It defaults to stereo.
2757 Map channels from inputs to output. The argument is a '|'-separated list of
2758 mappings, each in the @code{@var{input_idx}.@var{in_channel}-@var{out_channel}}
2759 form. @var{input_idx} is the 0-based index of the input stream. @var{in_channel}
2760 can be either the name of the input channel (e.g. FL for front left) or its
2761 index in the specified input stream. @var{out_channel} is the name of the output
2765 The filter will attempt to guess the mappings when they are not specified
2766 explicitly. It does so by first trying to find an unused matching input channel
2767 and if that fails it picks the first unused input channel.
2769 Join 3 inputs (with properly set channel layouts):
2771 ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
2774 Build a 5.1 output from 6 single-channel streams:
2776 ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
2777 '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'
2783 Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.
2785 To enable compilation of this filter you need to configure FFmpeg with
2786 @code{--enable-ladspa}.
2790 Specifies the name of LADSPA plugin library to load. If the environment
2791 variable @env{LADSPA_PATH} is defined, the LADSPA plugin is searched in
2792 each one of the directories specified by the colon separated list in
2793 @env{LADSPA_PATH}, otherwise in the standard LADSPA paths, which are in
2794 this order: @file{HOME/.ladspa/lib/}, @file{/usr/local/lib/ladspa/},
2795 @file{/usr/lib/ladspa/}.
2798 Specifies the plugin within the library. Some libraries contain only
2799 one plugin, but others contain many of them. If this is not set filter
2800 will list all available plugins within the specified library.
2803 Set the '|' separated list of controls which are zero or more floating point
2804 values that determine the behavior of the loaded plugin (for example delay,
2806 Controls need to be defined using the following syntax:
2807 c0=@var{value0}|c1=@var{value1}|c2=@var{value2}|..., where
2808 @var{valuei} is the value set on the @var{i}-th control.
2809 Alternatively they can be also defined using the following syntax:
2810 @var{value0}|@var{value1}|@var{value2}|..., where
2811 @var{valuei} is the value set on the @var{i}-th control.
2812 If @option{controls} is set to @code{help}, all available controls and
2813 their valid ranges are printed.
2815 @item sample_rate, s
2816 Specify the sample rate, default to 44100. Only used if plugin have
2820 Set the number of samples per channel per each output frame, default
2821 is 1024. Only used if plugin have zero inputs.
2824 Set the minimum duration of the sourced audio. See
2825 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
2826 for the accepted syntax.
2827 Note that the resulting duration may be greater than the specified duration,
2828 as the generated audio is always cut at the end of a complete frame.
2829 If not specified, or the expressed duration is negative, the audio is
2830 supposed to be generated forever.
2831 Only used if plugin have zero inputs.
2835 @subsection Examples
2839 List all available plugins within amp (LADSPA example plugin) library:
2845 List all available controls and their valid ranges for @code{vcf_notch}
2846 plugin from @code{VCF} library:
2848 ladspa=f=vcf:p=vcf_notch:c=help
2852 Simulate low quality audio equipment using @code{Computer Music Toolkit} (CMT)
2855 ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
2859 Add reverberation to the audio using TAP-plugins
2860 (Tom's Audio Processing plugins):
2862 ladspa=file=tap_reverb:tap_reverb
2866 Generate white noise, with 0.2 amplitude:
2868 ladspa=file=cmt:noise_source_white:c=c0=.2
2872 Generate 20 bpm clicks using plugin @code{C* Click - Metronome} from the
2873 @code{C* Audio Plugin Suite} (CAPS) library:
2875 ladspa=file=caps:Click:c=c1=20'
2879 Apply @code{C* Eq10X2 - Stereo 10-band equaliser} effect:
2881 ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
2885 Increase volume by 20dB using fast lookahead limiter from Steve Harris
2886 @code{SWH Plugins} collection:
2888 ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
2892 Attenuate low frequencies using Multiband EQ from Steve Harris
2893 @code{SWH Plugins} collection:
2895 ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
2899 @subsection Commands
2901 This filter supports the following commands:
2904 Modify the @var{N}-th control value.
2906 If the specified value is not valid, it is ignored and prior one is kept.
2911 EBU R128 loudness normalization. Includes both dynamic and linear normalization modes.
2912 Support for both single pass (livestreams, files) and double pass (files) modes.
2913 This algorithm can target IL, LRA, and maximum true peak.
2915 The filter accepts the following options:
2919 Set integrated loudness target.
2920 Range is -70.0 - -5.0. Default value is -24.0.
2923 Set loudness range target.
2924 Range is 1.0 - 20.0. Default value is 7.0.
2927 Set maximum true peak.
2928 Range is -9.0 - +0.0. Default value is -2.0.
2930 @item measured_I, measured_i
2931 Measured IL of input file.
2932 Range is -99.0 - +0.0.
2934 @item measured_LRA, measured_lra
2935 Measured LRA of input file.
2936 Range is 0.0 - 99.0.
2938 @item measured_TP, measured_tp
2939 Measured true peak of input file.
2940 Range is -99.0 - +99.0.
2942 @item measured_thresh
2943 Measured threshold of input file.
2944 Range is -99.0 - +0.0.
2947 Set offset gain. Gain is applied before the true-peak limiter.
2948 Range is -99.0 - +99.0. Default is +0.0.
2951 Normalize linearly if possible.
2952 measured_I, measured_LRA, measured_TP, and measured_thresh must also
2953 to be specified in order to use this mode.
2954 Options are true or false. Default is true.
2957 Treat mono input files as "dual-mono". If a mono file is intended for playback
2958 on a stereo system, its EBU R128 measurement will be perceptually incorrect.
2959 If set to @code{true}, this option will compensate for this effect.
2960 Multi-channel input files are not affected by this option.
2961 Options are true or false. Default is false.
2964 Set print format for stats. Options are summary, json, or none.
2965 Default value is none.
2970 Apply a low-pass filter with 3dB point frequency.
2971 The filter can be either single-pole or double-pole (the default).
2972 The filter roll off at 6dB per pole per octave (20dB per pole per decade).
2974 The filter accepts the following options:
2978 Set frequency in Hz. Default is 500.
2981 Set number of poles. Default is 2.
2984 Set method to specify band-width of filter.
2997 Specify the band-width of a filter in width_type units.
2998 Applies only to double-pole filter.
2999 The default is 0.707q and gives a Butterworth response.
3002 Specify which channels to filter, by default all available are filtered.
3005 @subsection Examples
3008 Lowpass only LFE channel, it LFE is not present it does nothing:
3017 Mix channels with specific gain levels. The filter accepts the output
3018 channel layout followed by a set of channels definitions.
3020 This filter is also designed to efficiently remap the channels of an audio
3023 The filter accepts parameters of the form:
3024 "@var{l}|@var{outdef}|@var{outdef}|..."
3028 output channel layout or number of channels
3031 output channel specification, of the form:
3032 "@var{out_name}=[@var{gain}*]@var{in_name}[(+-)[@var{gain}*]@var{in_name}...]"
3035 output channel to define, either a channel name (FL, FR, etc.) or a channel
3036 number (c0, c1, etc.)
3039 multiplicative coefficient for the channel, 1 leaving the volume unchanged
3042 input channel to use, see out_name for details; it is not possible to mix
3043 named and numbered input channels
3046 If the `=' in a channel specification is replaced by `<', then the gains for
3047 that specification will be renormalized so that the total is 1, thus
3048 avoiding clipping noise.
3050 @subsection Mixing examples
3052 For example, if you want to down-mix from stereo to mono, but with a bigger
3053 factor for the left channel:
3055 pan=1c|c0=0.9*c0+0.1*c1
3058 A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
3059 7-channels surround:
3061 pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
3064 Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
3065 that should be preferred (see "-ac" option) unless you have very specific
3068 @subsection Remapping examples
3070 The channel remapping will be effective if, and only if:
3073 @item gain coefficients are zeroes or ones,
3074 @item only one input per channel output,
3077 If all these conditions are satisfied, the filter will notify the user ("Pure
3078 channel mapping detected"), and use an optimized and lossless method to do the
3081 For example, if you have a 5.1 source and want a stereo audio stream by
3082 dropping the extra channels:
3084 pan="stereo| c0=FL | c1=FR"
3087 Given the same source, you can also switch front left and front right channels
3088 and keep the input channel layout:
3090 pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"
3093 If the input is a stereo audio stream, you can mute the front left channel (and
3094 still keep the stereo channel layout) with:
3099 Still with a stereo audio stream input, you can copy the right channel in both
3100 front left and right:
3102 pan="stereo| c0=FR | c1=FR"
3107 ReplayGain scanner filter. This filter takes an audio stream as an input and
3108 outputs it unchanged.
3109 At end of filtering it displays @code{track_gain} and @code{track_peak}.
3113 Convert the audio sample format, sample rate and channel layout. It is
3114 not meant to be used directly.
3117 Apply time-stretching and pitch-shifting with librubberband.
3119 The filter accepts the following options:
3123 Set tempo scale factor.
3126 Set pitch scale factor.
3129 Set transients detector.
3130 Possible values are:
3139 Possible values are:
3148 Possible values are:
3155 Set processing window size.
3156 Possible values are:
3165 Possible values are:
3172 Enable formant preservation when shift pitching.
3173 Possible values are:
3181 Possible values are:
3190 Possible values are:
3197 @section sidechaincompress
3199 This filter acts like normal compressor but has the ability to compress
3200 detected signal using second input signal.
3201 It needs two input streams and returns one output stream.
3202 First input stream will be processed depending on second stream signal.
3203 The filtered signal then can be filtered with other filters in later stages of
3204 processing. See @ref{pan} and @ref{amerge} filter.
3206 The filter accepts the following options:
3210 Set input gain. Default is 1. Range is between 0.015625 and 64.
3213 If a signal of second stream raises above this level it will affect the gain
3214 reduction of first stream.
3215 By default is 0.125. Range is between 0.00097563 and 1.
3218 Set a ratio about which the signal is reduced. 1:2 means that if the level
3219 raised 4dB above the threshold, it will be only 2dB above after the reduction.
3220 Default is 2. Range is between 1 and 20.
3223 Amount of milliseconds the signal has to rise above the threshold before gain
3224 reduction starts. Default is 20. Range is between 0.01 and 2000.
3227 Amount of milliseconds the signal has to fall below the threshold before
3228 reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
3231 Set the amount by how much signal will be amplified after processing.
3232 Default is 2. Range is from 1 and 64.
3235 Curve the sharp knee around the threshold to enter gain reduction more softly.
3236 Default is 2.82843. Range is between 1 and 8.
3239 Choose if the @code{average} level between all channels of side-chain stream
3240 or the louder(@code{maximum}) channel of side-chain stream affects the
3241 reduction. Default is @code{average}.
3244 Should the exact signal be taken in case of @code{peak} or an RMS one in case
3245 of @code{rms}. Default is @code{rms} which is mainly smoother.
3248 Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
3251 How much to use compressed signal in output. Default is 1.
3252 Range is between 0 and 1.
3255 @subsection Examples
3259 Full ffmpeg example taking 2 audio inputs, 1st input to be compressed
3260 depending on the signal of 2nd input and later compressed signal to be
3261 merged with 2nd input:
3263 ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"
3267 @section sidechaingate
3269 A sidechain gate acts like a normal (wideband) gate but has the ability to
3270 filter the detected signal before sending it to the gain reduction stage.
3271 Normally a gate uses the full range signal to detect a level above the
3273 For example: If you cut all lower frequencies from your sidechain signal
3274 the gate will decrease the volume of your track only if not enough highs
3275 appear. With this technique you are able to reduce the resonation of a
3276 natural drum or remove "rumbling" of muted strokes from a heavily distorted
3278 It needs two input streams and returns one output stream.
3279 First input stream will be processed depending on second stream signal.
3281 The filter accepts the following options:
3285 Set input level before filtering.
3286 Default is 1. Allowed range is from 0.015625 to 64.
3289 Set the level of gain reduction when the signal is below the threshold.
3290 Default is 0.06125. Allowed range is from 0 to 1.
3293 If a signal rises above this level the gain reduction is released.
3294 Default is 0.125. Allowed range is from 0 to 1.
3297 Set a ratio about which the signal is reduced.
3298 Default is 2. Allowed range is from 1 to 9000.
3301 Amount of milliseconds the signal has to rise above the threshold before gain
3303 Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
3306 Amount of milliseconds the signal has to fall below the threshold before the
3307 reduction is increased again. Default is 250 milliseconds.
3308 Allowed range is from 0.01 to 9000.
3311 Set amount of amplification of signal after processing.
3312 Default is 1. Allowed range is from 1 to 64.
3315 Curve the sharp knee around the threshold to enter gain reduction more softly.
3316 Default is 2.828427125. Allowed range is from 1 to 8.
3319 Choose if exact signal should be taken for detection or an RMS like one.
3320 Default is rms. Can be peak or rms.
3323 Choose if the average level between all channels or the louder channel affects
3325 Default is average. Can be average or maximum.
3328 Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
3331 @section silencedetect
3333 Detect silence in an audio stream.
3335 This filter logs a message when it detects that the input audio volume is less
3336 or equal to a noise tolerance value for a duration greater or equal to the
3337 minimum detected noise duration.
3339 The printed times and duration are expressed in seconds.
3341 The filter accepts the following options:
3345 Set silence duration until notification (default is 2 seconds).
3348 Set noise tolerance. Can be specified in dB (in case "dB" is appended to the
3349 specified value) or amplitude ratio. Default is -60dB, or 0.001.
3352 @subsection Examples
3356 Detect 5 seconds of silence with -50dB noise tolerance:
3358 silencedetect=n=-50dB:d=5
3362 Complete example with @command{ffmpeg} to detect silence with 0.0001 noise
3363 tolerance in @file{silence.mp3}:
3365 ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
3369 @section silenceremove
3371 Remove silence from the beginning, middle or end of the audio.
3373 The filter accepts the following options:
3377 This value is used to indicate if audio should be trimmed at beginning of
3378 the audio. A value of zero indicates no silence should be trimmed from the
3379 beginning. When specifying a non-zero value, it trims audio up until it
3380 finds non-silence. Normally, when trimming silence from beginning of audio
3381 the @var{start_periods} will be @code{1} but it can be increased to higher
3382 values to trim all audio up to specific count of non-silence periods.
3383 Default value is @code{0}.
3385 @item start_duration
3386 Specify the amount of time that non-silence must be detected before it stops
3387 trimming audio. By increasing the duration, bursts of noises can be treated
3388 as silence and trimmed off. Default value is @code{0}.
3390 @item start_threshold
3391 This indicates what sample value should be treated as silence. For digital
3392 audio, a value of @code{0} may be fine but for audio recorded from analog,
3393 you may wish to increase the value to account for background noise.
3394 Can be specified in dB (in case "dB" is appended to the specified value)
3395 or amplitude ratio. Default value is @code{0}.
3398 Set the count for trimming silence from the end of audio.
3399 To remove silence from the middle of a file, specify a @var{stop_periods}
3400 that is negative. This value is then treated as a positive value and is
3401 used to indicate the effect should restart processing as specified by
3402 @var{start_periods}, making it suitable for removing periods of silence
3403 in the middle of the audio.
3404 Default value is @code{0}.
3407 Specify a duration of silence that must exist before audio is not copied any
3408 more. By specifying a higher duration, silence that is wanted can be left in
3410 Default value is @code{0}.
3412 @item stop_threshold
3413 This is the same as @option{start_threshold} but for trimming silence from
3415 Can be specified in dB (in case "dB" is appended to the specified value)
3416 or amplitude ratio. Default value is @code{0}.
3419 This indicates that @var{stop_duration} length of audio should be left intact
3420 at the beginning of each period of silence.
3421 For example, if you want to remove long pauses between words but do not want
3422 to remove the pauses completely. Default value is @code{0}.
3425 Set how is silence detected. Can be @code{rms} or @code{peak}. Second is faster
3426 and works better with digital silence which is exactly 0.
3427 Default value is @code{rms}.
3430 Set ratio used to calculate size of window for detecting silence.
3431 Default value is @code{0.02}. Allowed range is from @code{0} to @code{10}.
3434 @subsection Examples
3438 The following example shows how this filter can be used to start a recording
3439 that does not contain the delay at the start which usually occurs between
3440 pressing the record button and the start of the performance:
3442 silenceremove=1:5:0.02
3446 Trim all silence encountered from beginning to end where there is more than 1
3447 second of silence in audio:
3449 silenceremove=0:0:0:-1:1:-90dB
3455 SOFAlizer uses head-related transfer functions (HRTFs) to create virtual
3456 loudspeakers around the user for binaural listening via headphones (audio
3457 formats up to 9 channels supported).
3458 The HRTFs are stored in SOFA files (see @url{http://www.sofacoustics.org/} for a database).
3459 SOFAlizer is developed at the Acoustics Research Institute (ARI) of the
3460 Austrian Academy of Sciences.
3462 To enable compilation of this filter you need to configure FFmpeg with
3463 @code{--enable-netcdf}.
3465 The filter accepts the following options:
3469 Set the SOFA file used for rendering.
3472 Set gain applied to audio. Value is in dB. Default is 0.
3475 Set rotation of virtual loudspeakers in deg. Default is 0.
3478 Set elevation of virtual speakers in deg. Default is 0.
3481 Set distance in meters between loudspeakers and the listener with near-field
3482 HRTFs. Default is 1.
3485 Set processing type. Can be @var{time} or @var{freq}. @var{time} is
3486 processing audio in time domain which is slow.
3487 @var{freq} is processing audio in frequency domain which is fast.
3488 Default is @var{freq}.
3491 Set custom positions of virtual loudspeakers. Syntax for this option is:
3492 <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...].
3493 Each virtual loudspeaker is described with short channel name following with
3494 azimuth and elevation in degreees.
3495 Each virtual loudspeaker description is separated by '|'.
3496 For example to override front left and front right channel positions use:
3497 'speakers=FL 45 15|FR 345 15'.
3498 Descriptions with unrecognised channel names are ignored.
3501 @subsection Examples
3505 Using ClubFritz6 sofa file:
3507 sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
3511 Using ClubFritz12 sofa file and bigger radius with small rotation:
3513 sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
3517 Similar as above but with custom speaker positions for front left, front right, back left and back right
3518 and also with custom gain:
3520 "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"
3524 @section stereotools
3526 This filter has some handy utilities to manage stereo signals, for converting
3527 M/S stereo recordings to L/R signal while having control over the parameters
3528 or spreading the stereo image of master track.
3530 The filter accepts the following options:
3534 Set input level before filtering for both channels. Defaults is 1.
3535 Allowed range is from 0.015625 to 64.
3538 Set output level after filtering for both channels. Defaults is 1.
3539 Allowed range is from 0.015625 to 64.
3542 Set input balance between both channels. Default is 0.
3543 Allowed range is from -1 to 1.
3546 Set output balance between both channels. Default is 0.
3547 Allowed range is from -1 to 1.
3550 Enable softclipping. Results in analog distortion instead of harsh digital 0dB
3551 clipping. Disabled by default.
3554 Mute the left channel. Disabled by default.
3557 Mute the right channel. Disabled by default.
3560 Change the phase of the left channel. Disabled by default.
3563 Change the phase of the right channel. Disabled by default.
3566 Set stereo mode. Available values are:
3570 Left/Right to Left/Right, this is default.
3573 Left/Right to Mid/Side.
3576 Mid/Side to Left/Right.
3579 Left/Right to Left/Left.
3582 Left/Right to Right/Right.
3585 Left/Right to Left + Right.
3588 Left/Right to Right/Left.
3592 Set level of side signal. Default is 1.
3593 Allowed range is from 0.015625 to 64.
3596 Set balance of side signal. Default is 0.
3597 Allowed range is from -1 to 1.
3600 Set level of the middle signal. Default is 1.
3601 Allowed range is from 0.015625 to 64.
3604 Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
3607 Set stereo base between mono and inversed channels. Default is 0.
3608 Allowed range is from -1 to 1.
3611 Set delay in milliseconds how much to delay left from right channel and
3612 vice versa. Default is 0. Allowed range is from -20 to 20.
3615 Set S/C level. Default is 1. Allowed range is from 1 to 100.
3618 Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.
3621 @subsection Examples
3625 Apply karaoke like effect:
3627 stereotools=mlev=0.015625
3631 Convert M/S signal to L/R:
3633 "stereotools=mode=ms>lr"
3637 @section stereowiden
3639 This filter enhance the stereo effect by suppressing signal common to both
3640 channels and by delaying the signal of left into right and vice versa,
3641 thereby widening the stereo effect.
3643 The filter accepts the following options:
3647 Time in milliseconds of the delay of left signal into right and vice versa.
3648 Default is 20 milliseconds.
3651 Amount of gain in delayed signal into right and vice versa. Gives a delay
3652 effect of left signal in right output and vice versa which gives widening
3653 effect. Default is 0.3.
3656 Cross feed of left into right with inverted phase. This helps in suppressing
3657 the mono. If the value is 1 it will cancel all the signal common to both
3658 channels. Default is 0.3.
3661 Set level of input signal of original channel. Default is 0.8.
3666 Boost or cut treble (upper) frequencies of the audio using a two-pole
3667 shelving filter with a response similar to that of a standard
3668 hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
3670 The filter accepts the following options:
3674 Give the gain at whichever is the lower of ~22 kHz and the
3675 Nyquist frequency. Its useful range is about -20 (for a large cut)
3676 to +20 (for a large boost). Beware of clipping when using a positive gain.
3679 Set the filter's central frequency and so can be used
3680 to extend or reduce the frequency range to be boosted or cut.
3681 The default value is @code{3000} Hz.
3684 Set method to specify band-width of filter.
3697 Determine how steep is the filter's shelf transition.
3700 Specify which channels to filter, by default all available are filtered.
3705 Sinusoidal amplitude modulation.
3707 The filter accepts the following options:
3711 Modulation frequency in Hertz. Modulation frequencies in the subharmonic range
3712 (20 Hz or lower) will result in a tremolo effect.
3713 This filter may also be used as a ring modulator by specifying
3714 a modulation frequency higher than 20 Hz.
3715 Range is 0.1 - 20000.0. Default value is 5.0 Hz.
3718 Depth of modulation as a percentage. Range is 0.0 - 1.0.
3719 Default value is 0.5.
3724 Sinusoidal phase modulation.
3726 The filter accepts the following options:
3730 Modulation frequency in Hertz.
3731 Range is 0.1 - 20000.0. Default value is 5.0 Hz.
3734 Depth of modulation as a percentage. Range is 0.0 - 1.0.
3735 Default value is 0.5.
3740 Adjust the input audio volume.
3742 It accepts the following parameters:
3746 Set audio volume expression.
3748 Output values are clipped to the maximum value.
3750 The output audio volume is given by the relation:
3752 @var{output_volume} = @var{volume} * @var{input_volume}
3755 The default value for @var{volume} is "1.0".
3758 This parameter represents the mathematical precision.
3760 It determines which input sample formats will be allowed, which affects the
3761 precision of the volume scaling.
3765 8-bit fixed-point; this limits input sample format to U8, S16, and S32.
3767 32-bit floating-point; this limits input sample format to FLT. (default)
3769 64-bit floating-point; this limits input sample format to DBL.
3773 Choose the behaviour on encountering ReplayGain side data in input frames.
3777 Remove ReplayGain side data, ignoring its contents (the default).
3780 Ignore ReplayGain side data, but leave it in the frame.
3783 Prefer the track gain, if present.
3786 Prefer the album gain, if present.
3789 @item replaygain_preamp
3790 Pre-amplification gain in dB to apply to the selected replaygain gain.
3792 Default value for @var{replaygain_preamp} is 0.0.
3795 Set when the volume expression is evaluated.
3797 It accepts the following values:
3800 only evaluate expression once during the filter initialization, or
3801 when the @samp{volume} command is sent
3804 evaluate expression for each incoming frame
3807 Default value is @samp{once}.
3810 The volume expression can contain the following parameters.
3814 frame number (starting at zero)
3817 @item nb_consumed_samples
3818 number of samples consumed by the filter
3820 number of samples in the current frame
3822 original frame position in the file
3828 PTS at start of stream
3830 time at start of stream
3836 last set volume value
3839 Note that when @option{eval} is set to @samp{once} only the
3840 @var{sample_rate} and @var{tb} variables are available, all other
3841 variables will evaluate to NAN.
3843 @subsection Commands
3845 This filter supports the following commands:
3848 Modify the volume expression.
3849 The command accepts the same syntax of the corresponding option.
3851 If the specified expression is not valid, it is kept at its current
3853 @item replaygain_noclip
3854 Prevent clipping by limiting the gain applied.
3856 Default value for @var{replaygain_noclip} is 1.
3860 @subsection Examples
3864 Halve the input audio volume:
3868 volume=volume=-6.0206dB
3871 In all the above example the named key for @option{volume} can be
3872 omitted, for example like in:
3878 Increase input audio power by 6 decibels using fixed-point precision:
3880 volume=volume=6dB:precision=fixed
3884 Fade volume after time 10 with an annihilation period of 5 seconds:
3886 volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame
3890 @section volumedetect
3892 Detect the volume of the input video.
3894 The filter has no parameters. The input is not modified. Statistics about
3895 the volume will be printed in the log when the input stream end is reached.
3897 In particular it will show the mean volume (root mean square), maximum
3898 volume (on a per-sample basis), and the beginning of a histogram of the
3899 registered volume values (from the maximum value to a cumulated 1/1000 of
3902 All volumes are in decibels relative to the maximum PCM value.
3904 @subsection Examples
3906 Here is an excerpt of the output:
3908 [Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB
3909 [Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB
3910 [Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6
3911 [Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62
3912 [Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286
3913 [Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042
3914 [Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551
3915 [Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609
3916 [Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409
3922 The mean square energy is approximately -27 dB, or 10^-2.7.
3924 The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
3926 There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
3929 In other words, raising the volume by +4 dB does not cause any clipping,
3930 raising it by +5 dB causes clipping for 6 samples, etc.
3932 @c man end AUDIO FILTERS
3934 @chapter Audio Sources
3935 @c man begin AUDIO SOURCES
3937 Below is a description of the currently available audio sources.
3941 Buffer audio frames, and make them available to the filter chain.
3943 This source is mainly intended for a programmatic use, in particular
3944 through the interface defined in @file{libavfilter/asrc_abuffer.h}.
3946 It accepts the following parameters:
3950 The timebase which will be used for timestamps of submitted frames. It must be
3951 either a floating-point number or in @var{numerator}/@var{denominator} form.
3954 The sample rate of the incoming audio buffers.
3957 The sample format of the incoming audio buffers.
3958 Either a sample format name or its corresponding integer representation from
3959 the enum AVSampleFormat in @file{libavutil/samplefmt.h}
3961 @item channel_layout
3962 The channel layout of the incoming audio buffers.
3963 Either a channel layout name from channel_layout_map in
3964 @file{libavutil/channel_layout.c} or its corresponding integer representation
3965 from the AV_CH_LAYOUT_* macros in @file{libavutil/channel_layout.h}
3968 The number of channels of the incoming audio buffers.
3969 If both @var{channels} and @var{channel_layout} are specified, then they
3974 @subsection Examples
3977 abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo
3980 will instruct the source to accept planar 16bit signed stereo at 44100Hz.
3981 Since the sample format with name "s16p" corresponds to the number
3982 6 and the "stereo" channel layout corresponds to the value 0x3, this is
3985 abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3
3990 Generate an audio signal specified by an expression.
3992 This source accepts in input one or more expressions (one for each
3993 channel), which are evaluated and used to generate a corresponding
3996 This source accepts the following options:
4000 Set the '|'-separated expressions list for each separate channel. In case the
4001 @option{channel_layout} option is not specified, the selected channel layout
4002 depends on the number of provided expressions. Otherwise the last
4003 specified expression is applied to the remaining output channels.
4005 @item channel_layout, c
4006 Set the channel layout. The number of channels in the specified layout
4007 must be equal to the number of specified expressions.
4010 Set the minimum duration of the sourced audio. See
4011 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
4012 for the accepted syntax.
4013 Note that the resulting duration may be greater than the specified
4014 duration, as the generated audio is always cut at the end of a
4017 If not specified, or the expressed duration is negative, the audio is
4018 supposed to be generated forever.
4021 Set the number of samples per channel per each output frame,
4024 @item sample_rate, s
4025 Specify the sample rate, default to 44100.
4028 Each expression in @var{exprs} can contain the following constants:
4032 number of the evaluated sample, starting from 0
4035 time of the evaluated sample expressed in seconds, starting from 0
4042 @subsection Examples
4052 Generate a sin signal with frequency of 440 Hz, set sample rate to
4055 aevalsrc="sin(440*2*PI*t):s=8000"
4059 Generate a two channels signal, specify the channel layout (Front
4060 Center + Back Center) explicitly:
4062 aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"
4066 Generate white noise:
4068 aevalsrc="-2+random(0)"
4072 Generate an amplitude modulated signal:
4074 aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
4078 Generate 2.5 Hz binaural beats on a 360 Hz carrier:
4080 aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
4087 The null audio source, return unprocessed audio frames. It is mainly useful
4088 as a template and to be employed in analysis / debugging tools, or as
4089 the source for filters which ignore the input data (for example the sox
4092 This source accepts the following options:
4096 @item channel_layout, cl
4098 Specifies the channel layout, and can be either an integer or a string
4099 representing a channel layout. The default value of @var{channel_layout}
4102 Check the channel_layout_map definition in
4103 @file{libavutil/channel_layout.c} for the mapping between strings and
4104 channel layout values.
4106 @item sample_rate, r
4107 Specifies the sample rate, and defaults to 44100.
4110 Set the number of samples per requested frames.
4114 @subsection Examples
4118 Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.
4120 anullsrc=r=48000:cl=4
4124 Do the same operation with a more obvious syntax:
4126 anullsrc=r=48000:cl=mono
4130 All the parameters need to be explicitly defined.
4134 Synthesize a voice utterance using the libflite library.
4136 To enable compilation of this filter you need to configure FFmpeg with
4137 @code{--enable-libflite}.
4139 Note that the flite library is not thread-safe.
4141 The filter accepts the following options:
4146 If set to 1, list the names of the available voices and exit
4147 immediately. Default value is 0.
4150 Set the maximum number of samples per frame. Default value is 512.
4153 Set the filename containing the text to speak.
4156 Set the text to speak.
4159 Set the voice to use for the speech synthesis. Default value is
4160 @code{kal}. See also the @var{list_voices} option.
4163 @subsection Examples
4167 Read from file @file{speech.txt}, and synthesize the text using the
4168 standard flite voice:
4170 flite=textfile=speech.txt
4174 Read the specified text selecting the @code{slt} voice:
4176 flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
4180 Input text to ffmpeg:
4182 ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt