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{id}=@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 optionally followed by "@@@var{id}".
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{FILTER_NAME} ::= @var{NAME}["@@"@var{NAME}]
216 @var{LINKLABEL} ::= "[" @var{NAME} "]"
217 @var{LINKLABELS} ::= @var{LINKLABEL} [@var{LINKLABELS}]
218 @var{FILTER_ARGUMENTS} ::= sequence of chars (possibly quoted)
219 @var{FILTER} ::= [@var{LINKLABELS}] @var{FILTER_NAME} ["=" @var{FILTER_ARGUMENTS}] [@var{LINKLABELS}]
220 @var{FILTERCHAIN} ::= @var{FILTER} [,@var{FILTERCHAIN}]
221 @var{FILTERGRAPH} ::= [sws_flags=@var{flags};] @var{FILTERCHAIN} [;@var{FILTERGRAPH}]
224 @anchor{filtergraph escaping}
225 @section Notes on filtergraph escaping
227 Filtergraph description composition entails several levels of
228 escaping. See @ref{quoting_and_escaping,,the "Quoting and escaping"
229 section in the ffmpeg-utils(1) manual,ffmpeg-utils} for more
230 information about the employed escaping procedure.
232 A first level escaping affects the content of each filter option
233 value, which may contain the special character @code{:} used to
234 separate values, or one of the escaping characters @code{\'}.
236 A second level escaping affects the whole filter description, which
237 may contain the escaping characters @code{\'} or the special
238 characters @code{[],;} used by the filtergraph description.
240 Finally, when you specify a filtergraph on a shell commandline, you
241 need to perform a third level escaping for the shell special
242 characters contained within it.
244 For example, consider the following string to be embedded in
245 the @ref{drawtext} filter description @option{text} value:
247 this is a 'string': may contain one, or more, special characters
250 This string contains the @code{'} special escaping character, and the
251 @code{:} special character, so it needs to be escaped in this way:
253 text=this is a \'string\'\: may contain one, or more, special characters
256 A second level of escaping is required when embedding the filter
257 description in a filtergraph description, in order to escape all the
258 filtergraph special characters. Thus the example above becomes:
260 drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters
262 (note that in addition to the @code{\'} escaping special characters,
263 also @code{,} needs to be escaped).
265 Finally an additional level of escaping is needed when writing the
266 filtergraph description in a shell command, which depends on the
267 escaping rules of the adopted shell. For example, assuming that
268 @code{\} is special and needs to be escaped with another @code{\}, the
269 previous string will finally result in:
271 -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"
274 @chapter Timeline editing
276 Some filters support a generic @option{enable} option. For the filters
277 supporting timeline editing, this option can be set to an expression which is
278 evaluated before sending a frame to the filter. If the evaluation is non-zero,
279 the filter will be enabled, otherwise the frame will be sent unchanged to the
280 next filter in the filtergraph.
282 The expression accepts the following values:
285 timestamp expressed in seconds, NAN if the input timestamp is unknown
288 sequential number of the input frame, starting from 0
291 the position in the file of the input frame, NAN if unknown
295 width and height of the input frame if video
298 Additionally, these filters support an @option{enable} command that can be used
299 to re-define the expression.
301 Like any other filtering option, the @option{enable} option follows the same
304 For example, to enable a blur filter (@ref{smartblur}) from 10 seconds to 3
305 minutes, and a @ref{curves} filter starting at 3 seconds:
307 smartblur = enable='between(t,10,3*60)',
308 curves = enable='gte(t,3)' : preset=cross_process
311 See @code{ffmpeg -filters} to view which filters have timeline support.
313 @c man end FILTERGRAPH DESCRIPTION
316 @chapter Options for filters with several inputs (framesync)
317 @c man begin OPTIONS FOR FILTERS WITH SEVERAL INPUTS
319 Some filters with several inputs support a common set of options.
320 These options can only be set by name, not with the short notation.
324 The action to take when EOF is encountered on the secondary input; it accepts
325 one of the following values:
329 Repeat the last frame (the default).
333 Pass the main input through.
337 If set to 1, force the output to terminate when the shortest input
338 terminates. Default value is 0.
341 If set to 1, force the filter to extend the last frame of secondary streams
342 until the end of the primary stream. A value of 0 disables this behavior.
346 @c man end OPTIONS FOR FILTERS WITH SEVERAL INPUTS
348 @chapter Audio Filters
349 @c man begin AUDIO FILTERS
351 When you configure your FFmpeg build, you can disable any of the
352 existing filters using @code{--disable-filters}.
353 The configure output will show the audio filters included in your
356 Below is a description of the currently available audio filters.
360 A compressor is mainly used to reduce the dynamic range of a signal.
361 Especially modern music is mostly compressed at a high ratio to
362 improve the overall loudness. It's done to get the highest attention
363 of a listener, "fatten" the sound and bring more "power" to the track.
364 If a signal is compressed too much it may sound dull or "dead"
365 afterwards or it may start to "pump" (which could be a powerful effect
366 but can also destroy a track completely).
367 The right compression is the key to reach a professional sound and is
368 the high art of mixing and mastering. Because of its complex settings
369 it may take a long time to get the right feeling for this kind of effect.
371 Compression is done by detecting the volume above a chosen level
372 @code{threshold} and dividing it by the factor set with @code{ratio}.
373 So if you set the threshold to -12dB and your signal reaches -6dB a ratio
374 of 2:1 will result in a signal at -9dB. Because an exact manipulation of
375 the signal would cause distortion of the waveform the reduction can be
376 levelled over the time. This is done by setting "Attack" and "Release".
377 @code{attack} determines how long the signal has to rise above the threshold
378 before any reduction will occur and @code{release} sets the time the signal
379 has to fall below the threshold to reduce the reduction again. Shorter signals
380 than the chosen attack time will be left untouched.
381 The overall reduction of the signal can be made up afterwards with the
382 @code{makeup} setting. So compressing the peaks of a signal about 6dB and
383 raising the makeup to this level results in a signal twice as loud than the
384 source. To gain a softer entry in the compression the @code{knee} flattens the
385 hard edge at the threshold in the range of the chosen decibels.
387 The filter accepts the following options:
391 Set input gain. Default is 1. Range is between 0.015625 and 64.
394 If a signal of stream rises above this level it will affect the gain
396 By default it is 0.125. Range is between 0.00097563 and 1.
399 Set a ratio by which the signal is reduced. 1:2 means that if the level
400 rose 4dB above the threshold, it will be only 2dB above after the reduction.
401 Default is 2. Range is between 1 and 20.
404 Amount of milliseconds the signal has to rise above the threshold before gain
405 reduction starts. Default is 20. Range is between 0.01 and 2000.
408 Amount of milliseconds the signal has to fall below the threshold before
409 reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
412 Set the amount by how much signal will be amplified after processing.
413 Default is 1. Range is from 1 to 64.
416 Curve the sharp knee around the threshold to enter gain reduction more softly.
417 Default is 2.82843. Range is between 1 and 8.
420 Choose if the @code{average} level between all channels of input stream
421 or the louder(@code{maximum}) channel of input stream affects the
422 reduction. Default is @code{average}.
425 Should the exact signal be taken in case of @code{peak} or an RMS one in case
426 of @code{rms}. Default is @code{rms} which is mostly smoother.
429 How much to use compressed signal in output. Default is 1.
430 Range is between 0 and 1.
434 Simple audio dynamic range commpression/expansion filter.
436 The filter accepts the following options:
440 Set contrast. Default is 33. Allowed range is between 0 and 100.
445 Copy the input audio source unchanged to the output. This is mainly useful for
450 Apply cross fade from one input audio stream to another input audio stream.
451 The cross fade is applied for specified duration near the end of first stream.
453 The filter accepts the following options:
457 Specify the number of samples for which the cross fade effect has to last.
458 At the end of the cross fade effect the first input audio will be completely
459 silent. Default is 44100.
462 Specify the duration of the cross fade effect. See
463 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
464 for the accepted syntax.
465 By default the duration is determined by @var{nb_samples}.
466 If set this option is used instead of @var{nb_samples}.
469 Should first stream end overlap with second stream start. Default is enabled.
472 Set curve for cross fade transition for first stream.
475 Set curve for cross fade transition for second stream.
477 For description of available curve types see @ref{afade} filter description.
484 Cross fade from one input to another:
486 ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
490 Cross fade from one input to another but without overlapping:
492 ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
497 Split audio stream into several bands.
499 This filter splits audio stream into two or more frequency ranges.
500 Summing all streams back will give flat output.
502 The filter accepts the following options:
506 Set split frequencies. Those must be positive and increasing.
509 Set filter order, can be @var{2nd}, @var{4th} or @var{8th}.
510 Default is @var{4th}.
515 Reduce audio bit resolution.
517 This filter is bit crusher with enhanced functionality. A bit crusher
518 is used to audibly reduce number of bits an audio signal is sampled
519 with. This doesn't change the bit depth at all, it just produces the
520 effect. Material reduced in bit depth sounds more harsh and "digital".
521 This filter is able to even round to continuous values instead of discrete
523 Additionally it has a D/C offset which results in different crushing of
524 the lower and the upper half of the signal.
525 An Anti-Aliasing setting is able to produce "softer" crushing sounds.
527 Another feature of this filter is the logarithmic mode.
528 This setting switches from linear distances between bits to logarithmic ones.
529 The result is a much more "natural" sounding crusher which doesn't gate low
530 signals for example. The human ear has a logarithmic perception,
531 so this kind of crushing is much more pleasant.
532 Logarithmic crushing is also able to get anti-aliased.
534 The filter accepts the following options:
550 Can be linear: @code{lin} or logarithmic: @code{log}.
559 Set sample reduction.
562 Enable LFO. By default disabled.
573 Delay audio filtering until a given wallclock timestamp. See the @ref{cue}
577 Remove impulsive noise from input audio.
579 Samples detected as impulsive noise are replaced by interpolated samples using
580 autoregressive modelling.
584 Set window size, in milliseconds. Allowed range is from @code{10} to
585 @code{100}. Default value is @code{55} milliseconds.
586 This sets size of window which will be processed at once.
589 Set window overlap, in percentage of window size. Allowed range is from
590 @code{50} to @code{95}. Default value is @code{75} percent.
591 Setting this to a very high value increases impulsive noise removal but makes
592 whole process much slower.
595 Set autoregression order, in percentage of window size. Allowed range is from
596 @code{0} to @code{25}. Default value is @code{2} percent. This option also
597 controls quality of interpolated samples using neighbour good samples.
600 Set threshold value. Allowed range is from @code{1} to @code{100}.
601 Default value is @code{2}.
602 This controls the strength of impulsive noise which is going to be removed.
603 The lower value, the more samples will be detected as impulsive noise.
606 Set burst fusion, in percentage of window size. Allowed range is @code{0} to
607 @code{10}. Default value is @code{2}.
608 If any two samples deteced as noise are spaced less than this value then any
609 sample inbetween those two samples will be also detected as noise.
614 It accepts the following values:
617 Select overlap-add method. Even not interpolated samples are slightly
618 changed with this method.
621 Select overlap-save method. Not interpolated samples remain unchanged.
624 Default value is @code{a}.
628 Remove clipped samples from input audio.
630 Samples detected as clipped are replaced by interpolated samples using
631 autoregressive modelling.
635 Set window size, in milliseconds. Allowed range is from @code{10} to @code{100}.
636 Default value is @code{55} milliseconds.
637 This sets size of window which will be processed at once.
640 Set window overlap, in percentage of window size. Allowed range is from @code{50}
641 to @code{95}. Default value is @code{75} percent.
644 Set autoregression order, in percentage of window size. Allowed range is from
645 @code{0} to @code{25}. Default value is @code{8} percent. This option also controls
646 quality of interpolated samples using neighbour good samples.
649 Set threshold value. Allowed range is from @code{1} to @code{100}.
650 Default value is @code{10}. Higher values make clip detection less aggressive.
653 Set size of histogram used to detect clips. Allowed range is from @code{100} to @code{9999}.
654 Default value is @code{1000}. Higher values make clip detection less aggressive.
659 It accepts the following values:
662 Select overlap-add method. Even not interpolated samples are slightly changed
666 Select overlap-save method. Not interpolated samples remain unchanged.
669 Default value is @code{a}.
674 Delay one or more audio channels.
676 Samples in delayed channel are filled with silence.
678 The filter accepts the following option:
682 Set list of delays in milliseconds for each channel separated by '|'.
683 Unused delays will be silently ignored. If number of given delays is
684 smaller than number of channels all remaining channels will not be delayed.
685 If you want to delay exact number of samples, append 'S' to number.
692 Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave
693 the second channel (and any other channels that may be present) unchanged.
699 Delay second channel by 500 samples, the third channel by 700 samples and leave
700 the first channel (and any other channels that may be present) unchanged.
706 @section aderivative, aintegral
708 Compute derivative/integral of audio stream.
710 Applying both filters one after another produces original audio.
714 Apply echoing to the input audio.
716 Echoes are reflected sound and can occur naturally amongst mountains
717 (and sometimes large buildings) when talking or shouting; digital echo
718 effects emulate this behaviour and are often used to help fill out the
719 sound of a single instrument or vocal. The time difference between the
720 original signal and the reflection is the @code{delay}, and the
721 loudness of the reflected signal is the @code{decay}.
722 Multiple echoes can have different delays and decays.
724 A description of the accepted parameters follows.
728 Set input gain of reflected signal. Default is @code{0.6}.
731 Set output gain of reflected signal. Default is @code{0.3}.
734 Set list of time intervals in milliseconds between original signal and reflections
735 separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}.
736 Default is @code{1000}.
739 Set list of loudness of reflected signals separated by '|'.
740 Allowed range for each @code{decay} is @code{(0 - 1.0]}.
741 Default is @code{0.5}.
748 Make it sound as if there are twice as many instruments as are actually playing:
750 aecho=0.8:0.88:60:0.4
754 If delay is very short, then it sound like a (metallic) robot playing music:
760 A longer delay will sound like an open air concert in the mountains:
762 aecho=0.8:0.9:1000:0.3
766 Same as above but with one more mountain:
768 aecho=0.8:0.9:1000|1800:0.3|0.25
773 Audio emphasis filter creates or restores material directly taken from LPs or
774 emphased CDs with different filter curves. E.g. to store music on vinyl the
775 signal has to be altered by a filter first to even out the disadvantages of
776 this recording medium.
777 Once the material is played back the inverse filter has to be applied to
778 restore the distortion of the frequency response.
780 The filter accepts the following options:
790 Set filter mode. For restoring material use @code{reproduction} mode, otherwise
791 use @code{production} mode. Default is @code{reproduction} mode.
794 Set filter type. Selects medium. Can be one of the following:
806 select Compact Disc (CD).
812 select 50µs (FM-KF).
814 select 75µs (FM-KF).
820 Modify an audio signal according to the specified expressions.
822 This filter accepts one or more expressions (one for each channel),
823 which are evaluated and used to modify a corresponding audio signal.
825 It accepts the following parameters:
829 Set the '|'-separated expressions list for each separate channel. If
830 the number of input channels is greater than the number of
831 expressions, the last specified expression is used for the remaining
834 @item channel_layout, c
835 Set output channel layout. If not specified, the channel layout is
836 specified by the number of expressions. If set to @samp{same}, it will
837 use by default the same input channel layout.
840 Each expression in @var{exprs} can contain the following constants and functions:
844 channel number of the current expression
847 number of the evaluated sample, starting from 0
853 time of the evaluated sample expressed in seconds
856 @item nb_out_channels
857 input and output number of channels
860 the value of input channel with number @var{CH}
863 Note: this filter is slow. For faster processing you should use a
872 aeval=val(ch)/2:c=same
876 Invert phase of the second channel:
885 Apply fade-in/out effect to input audio.
887 A description of the accepted parameters follows.
891 Specify the effect type, can be either @code{in} for fade-in, or
892 @code{out} for a fade-out effect. Default is @code{in}.
894 @item start_sample, ss
895 Specify the number of the start sample for starting to apply the fade
896 effect. Default is 0.
899 Specify the number of samples for which the fade effect has to last. At
900 the end of the fade-in effect the output audio will have the same
901 volume as the input audio, at the end of the fade-out transition
902 the output audio will be silence. Default is 44100.
905 Specify the start time of the fade effect. Default is 0.
906 The value must be specified as a time duration; see
907 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
908 for the accepted syntax.
909 If set this option is used instead of @var{start_sample}.
912 Specify the duration of the fade effect. See
913 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
914 for the accepted syntax.
915 At the end of the fade-in effect the output audio will have the same
916 volume as the input audio, at the end of the fade-out transition
917 the output audio will be silence.
918 By default the duration is determined by @var{nb_samples}.
919 If set this option is used instead of @var{nb_samples}.
922 Set curve for fade transition.
924 It accepts the following values:
927 select triangular, linear slope (default)
929 select quarter of sine wave
931 select half of sine wave
933 select exponential sine wave
937 select inverted parabola
951 select inverted quarter of sine wave
953 select inverted half of sine wave
955 select double-exponential seat
957 select double-exponential sigmoid
959 select logistic sigmoid
967 Fade in first 15 seconds of audio:
973 Fade out last 25 seconds of a 900 seconds audio:
975 afade=t=out:st=875:d=25
980 Denoise audio samples with FFT.
982 A description of the accepted parameters follows.
986 Set the noise reduction in dB, allowed range is 0.01 to 97.
987 Default value is 12 dB.
990 Set the noise floor in dB, allowed range is -80 to -20.
991 Default value is -50 dB.
996 It accepts the following values:
1005 Select shellac noise.
1008 Select custom noise, defined in @code{bn} option.
1010 Default value is white noise.
1014 Set custom band noise for every one of 15 bands.
1015 Bands are separated by ' ' or '|'.
1018 Set the residual floor in dB, allowed range is -80 to -20.
1019 Default value is -38 dB.
1022 Enable noise tracking. By default is disabled.
1023 With this enabled, noise floor is automatically adjusted.
1026 Enable residual tracking. By default is disabled.
1029 Set the output mode.
1031 It accepts the following values:
1034 Pass input unchanged.
1037 Pass noise filtered out.
1042 Default value is @var{o}.
1046 @subsection Commands
1048 This filter supports the following commands:
1050 @item sample_noise, sn
1051 Start or stop measuring noise profile.
1052 Syntax for the command is : "start" or "stop" string.
1053 After measuring noise profile is stopped it will be
1054 automatically applied in filtering.
1056 @item noise_reduction, nr
1057 Change noise reduction. Argument is single float number.
1058 Syntax for the command is : "@var{noise_reduction}"
1060 @item noise_floor, nf
1061 Change noise floor. Argument is single float number.
1062 Syntax for the command is : "@var{noise_floor}"
1064 @item output_mode, om
1065 Change output mode operation.
1066 Syntax for the command is : "i", "o" or "n" string.
1070 Apply arbitrary expressions to samples in frequency domain.
1074 Set frequency domain real expression for each separate channel separated
1075 by '|'. Default is "1".
1076 If the number of input channels is greater than the number of
1077 expressions, the last specified expression is used for the remaining
1081 Set frequency domain imaginary expression for each separate channel
1082 separated by '|'. If not set, @var{real} option is used.
1084 Each expression in @var{real} and @var{imag} can contain the following
1092 current frequency bin number
1095 number of available bins
1098 channel number of the current expression
1110 It accepts the following values:
1126 Default is @code{w4096}
1129 Set window function. Default is @code{hann}.
1132 Set window overlap. If set to 1, the recommended overlap for selected
1133 window function will be picked. Default is @code{0.75}.
1136 @subsection Examples
1140 Leave almost only low frequencies in audio:
1142 afftfilt="1-clip((b/nb)*b,0,1)"
1149 Apply an arbitrary Frequency Impulse Response filter.
1151 This filter is designed for applying long FIR filters,
1152 up to 60 seconds long.
1154 It can be used as component for digital crossover filters,
1155 room equalization, cross talk cancellation, wavefield synthesis,
1156 auralization, ambiophonics and ambisonics.
1158 This filter uses second stream as FIR coefficients.
1159 If second stream holds single channel, it will be used
1160 for all input channels in first stream, otherwise
1161 number of channels in second stream must be same as
1162 number of channels in first stream.
1164 It accepts the following parameters:
1168 Set dry gain. This sets input gain.
1171 Set wet gain. This sets final output gain.
1174 Set Impulse Response filter length. Default is 1, which means whole IR is processed.
1177 Enable applying gain measured from power of IR.
1179 Set which approach to use for auto gain measurement.
1183 Do not apply any gain.
1186 select peak gain, very conservative approach. This is default value.
1189 select DC gain, limited application.
1192 select gain to noise approach, this is most popular one.
1196 Set gain to be applied to IR coefficients before filtering.
1197 Allowed range is 0 to 1. This gain is applied after any gain applied with @var{gtype} option.
1200 Set format of IR stream. Can be @code{mono} or @code{input}.
1201 Default is @code{input}.
1204 Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds.
1205 Allowed range is 0.1 to 60 seconds.
1208 Show IR frequency reponse, magnitude(magenta) and phase(green) and group delay(yellow) in additional video stream.
1209 By default it is disabled.
1212 Set for which IR channel to display frequency response. By default is first channel
1213 displayed. This option is used only when @var{response} is enabled.
1216 Set video stream size. This option is used only when @var{response} is enabled.
1219 @subsection Examples
1223 Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:
1225 ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
1232 Set output format constraints for the input audio. The framework will
1233 negotiate the most appropriate format to minimize conversions.
1235 It accepts the following parameters:
1239 A '|'-separated list of requested sample formats.
1242 A '|'-separated list of requested sample rates.
1244 @item channel_layouts
1245 A '|'-separated list of requested channel layouts.
1247 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
1248 for the required syntax.
1251 If a parameter is omitted, all values are allowed.
1253 Force the output to either unsigned 8-bit or signed 16-bit stereo
1255 aformat=sample_fmts=u8|s16:channel_layouts=stereo
1260 A gate is mainly used to reduce lower parts of a signal. This kind of signal
1261 processing reduces disturbing noise between useful signals.
1263 Gating is done by detecting the volume below a chosen level @var{threshold}
1264 and dividing it by the factor set with @var{ratio}. The bottom of the noise
1265 floor is set via @var{range}. Because an exact manipulation of the signal
1266 would cause distortion of the waveform the reduction can be levelled over
1267 time. This is done by setting @var{attack} and @var{release}.
1269 @var{attack} determines how long the signal has to fall below the threshold
1270 before any reduction will occur and @var{release} sets the time the signal
1271 has to rise above the threshold to reduce the reduction again.
1272 Shorter signals than the chosen attack time will be left untouched.
1276 Set input level before filtering.
1277 Default is 1. Allowed range is from 0.015625 to 64.
1280 Set the level of gain reduction when the signal is below the threshold.
1281 Default is 0.06125. Allowed range is from 0 to 1.
1284 If a signal rises above this level the gain reduction is released.
1285 Default is 0.125. Allowed range is from 0 to 1.
1288 Set a ratio by which the signal is reduced.
1289 Default is 2. Allowed range is from 1 to 9000.
1292 Amount of milliseconds the signal has to rise above the threshold before gain
1294 Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
1297 Amount of milliseconds the signal has to fall below the threshold before the
1298 reduction is increased again. Default is 250 milliseconds.
1299 Allowed range is from 0.01 to 9000.
1302 Set amount of amplification of signal after processing.
1303 Default is 1. Allowed range is from 1 to 64.
1306 Curve the sharp knee around the threshold to enter gain reduction more softly.
1307 Default is 2.828427125. Allowed range is from 1 to 8.
1310 Choose if exact signal should be taken for detection or an RMS like one.
1311 Default is @code{rms}. Can be @code{peak} or @code{rms}.
1314 Choose if the average level between all channels or the louder channel affects
1316 Default is @code{average}. Can be @code{average} or @code{maximum}.
1321 Apply an arbitrary Infinite Impulse Response filter.
1323 It accepts the following parameters:
1327 Set numerator/zeros coefficients.
1330 Set denominator/poles coefficients.
1342 Set coefficients format.
1348 Z-plane zeros/poles, cartesian (default)
1350 Z-plane zeros/poles, polar radians
1352 Z-plane zeros/poles, polar degrees
1356 Set kind of processing.
1357 Can be @code{d} - direct or @code{s} - serial cascading. Defauls is @code{s}.
1360 Set filtering precision.
1364 double-precision floating-point (default)
1366 single-precision floating-point
1374 Show IR frequency reponse, magnitude and phase in additional video stream.
1375 By default it is disabled.
1378 Set for which IR channel to display frequency response. By default is first channel
1379 displayed. This option is used only when @var{response} is enabled.
1382 Set video stream size. This option is used only when @var{response} is enabled.
1385 Coefficients in @code{tf} format are separated by spaces and are in ascending
1388 Coefficients in @code{zp} format are separated by spaces and order of coefficients
1389 doesn't matter. Coefficients in @code{zp} format are complex numbers with @var{i}
1392 Different coefficients and gains can be provided for every channel, in such case
1393 use '|' to separate coefficients or gains. Last provided coefficients will be
1394 used for all remaining channels.
1396 @subsection Examples
1400 Apply 2 pole elliptic notch at arround 5000Hz for 48000 Hz sample rate:
1402 aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d
1406 Same as above but in @code{zp} format:
1408 aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s
1414 The limiter prevents an input signal from rising over a desired threshold.
1415 This limiter uses lookahead technology to prevent your signal from distorting.
1416 It means that there is a small delay after the signal is processed. Keep in mind
1417 that the delay it produces is the attack time you set.
1419 The filter accepts the following options:
1423 Set input gain. Default is 1.
1426 Set output gain. Default is 1.
1429 Don't let signals above this level pass the limiter. Default is 1.
1432 The limiter will reach its attenuation level in this amount of time in
1433 milliseconds. Default is 5 milliseconds.
1436 Come back from limiting to attenuation 1.0 in this amount of milliseconds.
1437 Default is 50 milliseconds.
1440 When gain reduction is always needed ASC takes care of releasing to an
1441 average reduction level rather than reaching a reduction of 0 in the release
1445 Select how much the release time is affected by ASC, 0 means nearly no changes
1446 in release time while 1 produces higher release times.
1449 Auto level output signal. Default is enabled.
1450 This normalizes audio back to 0dB if enabled.
1453 Depending on picked setting it is recommended to upsample input 2x or 4x times
1454 with @ref{aresample} before applying this filter.
1458 Apply a two-pole all-pass filter with central frequency (in Hz)
1459 @var{frequency}, and filter-width @var{width}.
1460 An all-pass filter changes the audio's frequency to phase relationship
1461 without changing its frequency to amplitude relationship.
1463 The filter accepts the following options:
1467 Set frequency in Hz.
1470 Set method to specify band-width of filter.
1485 Specify the band-width of a filter in width_type units.
1488 Specify which channels to filter, by default all available are filtered.
1491 @subsection Commands
1493 This filter supports the following commands:
1496 Change allpass frequency.
1497 Syntax for the command is : "@var{frequency}"
1500 Change allpass width_type.
1501 Syntax for the command is : "@var{width_type}"
1504 Change allpass width.
1505 Syntax for the command is : "@var{width}"
1512 The filter accepts the following options:
1516 Set the number of loops. Setting this value to -1 will result in infinite loops.
1520 Set maximal number of samples. Default is 0.
1523 Set first sample of loop. Default is 0.
1529 Merge two or more audio streams into a single multi-channel stream.
1531 The filter accepts the following options:
1536 Set the number of inputs. Default is 2.
1540 If the channel layouts of the inputs are disjoint, and therefore compatible,
1541 the channel layout of the output will be set accordingly and the channels
1542 will be reordered as necessary. If the channel layouts of the inputs are not
1543 disjoint, the output will have all the channels of the first input then all
1544 the channels of the second input, in that order, and the channel layout of
1545 the output will be the default value corresponding to the total number of
1548 For example, if the first input is in 2.1 (FL+FR+LF) and the second input
1549 is FC+BL+BR, then the output will be in 5.1, with the channels in the
1550 following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
1551 first input, b1 is the first channel of the second input).
1553 On the other hand, if both input are in stereo, the output channels will be
1554 in the default order: a1, a2, b1, b2, and the channel layout will be
1555 arbitrarily set to 4.0, which may or may not be the expected value.
1557 All inputs must have the same sample rate, and format.
1559 If inputs do not have the same duration, the output will stop with the
1562 @subsection Examples
1566 Merge two mono files into a stereo stream:
1568 amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
1572 Multiple merges assuming 1 video stream and 6 audio streams in @file{input.mkv}:
1574 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
1580 Mixes multiple audio inputs into a single output.
1582 Note that this filter only supports float samples (the @var{amerge}
1583 and @var{pan} audio filters support many formats). If the @var{amix}
1584 input has integer samples then @ref{aresample} will be automatically
1585 inserted to perform the conversion to float samples.
1589 ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
1591 will mix 3 input audio streams to a single output with the same duration as the
1592 first input and a dropout transition time of 3 seconds.
1594 It accepts the following parameters:
1598 The number of inputs. If unspecified, it defaults to 2.
1601 How to determine the end-of-stream.
1605 The duration of the longest input. (default)
1608 The duration of the shortest input.
1611 The duration of the first input.
1615 @item dropout_transition
1616 The transition time, in seconds, for volume renormalization when an input
1617 stream ends. The default value is 2 seconds.
1620 Specify weight of each input audio stream as sequence.
1621 Each weight is separated by space. By default all inputs have same weight.
1626 Multiply first audio stream with second audio stream and store result
1627 in output audio stream. Multiplication is done by multiplying each
1628 sample from first stream with sample at same position from second stream.
1630 With this element-wise multiplication one can create amplitude fades and
1631 amplitude modulations.
1633 @section anequalizer
1635 High-order parametric multiband equalizer for each channel.
1637 It accepts the following parameters:
1641 This option string is in format:
1642 "c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..."
1643 Each equalizer band is separated by '|'.
1647 Set channel number to which equalization will be applied.
1648 If input doesn't have that channel the entry is ignored.
1651 Set central frequency for band.
1652 If input doesn't have that frequency the entry is ignored.
1655 Set band width in hertz.
1658 Set band gain in dB.
1661 Set filter type for band, optional, can be:
1665 Butterworth, this is default.
1676 With this option activated frequency response of anequalizer is displayed
1680 Set video stream size. Only useful if curves option is activated.
1683 Set max gain that will be displayed. Only useful if curves option is activated.
1684 Setting this to a reasonable value makes it possible to display gain which is derived from
1685 neighbour bands which are too close to each other and thus produce higher gain
1686 when both are activated.
1689 Set frequency scale used to draw frequency response in video output.
1690 Can be linear or logarithmic. Default is logarithmic.
1693 Set color for each channel curve which is going to be displayed in video stream.
1694 This is list of color names separated by space or by '|'.
1695 Unrecognised or missing colors will be replaced by white color.
1698 @subsection Examples
1702 Lower gain by 10 of central frequency 200Hz and width 100 Hz
1703 for first 2 channels using Chebyshev type 1 filter:
1705 anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
1709 @subsection Commands
1711 This filter supports the following commands:
1714 Alter existing filter parameters.
1715 Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
1717 @var{fN} is existing filter number, starting from 0, if no such filter is available
1719 @var{freq} set new frequency parameter.
1720 @var{width} set new width parameter in herz.
1721 @var{gain} set new gain parameter in dB.
1723 Full filter invocation with asendcmd may look like this:
1724 asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...
1729 Pass the audio source unchanged to the output.
1733 Pad the end of an audio stream with silence.
1735 This can be used together with @command{ffmpeg} @option{-shortest} to
1736 extend audio streams to the same length as the video stream.
1738 A description of the accepted options follows.
1742 Set silence packet size. Default value is 4096.
1745 Set the number of samples of silence to add to the end. After the
1746 value is reached, the stream is terminated. This option is mutually
1747 exclusive with @option{whole_len}.
1750 Set the minimum total number of samples in the output audio stream. If
1751 the value is longer than the input audio length, silence is added to
1752 the end, until the value is reached. This option is mutually exclusive
1753 with @option{pad_len}.
1756 If neither the @option{pad_len} nor the @option{whole_len} option is
1757 set, the filter will add silence to the end of the input stream
1760 @subsection Examples
1764 Add 1024 samples of silence to the end of the input:
1770 Make sure the audio output will contain at least 10000 samples, pad
1771 the input with silence if required:
1773 apad=whole_len=10000
1777 Use @command{ffmpeg} to pad the audio input with silence, so that the
1778 video stream will always result the shortest and will be converted
1779 until the end in the output file when using the @option{shortest}
1782 ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
1787 Add a phasing effect to the input audio.
1789 A phaser filter creates series of peaks and troughs in the frequency spectrum.
1790 The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.
1792 A description of the accepted parameters follows.
1796 Set input gain. Default is 0.4.
1799 Set output gain. Default is 0.74
1802 Set delay in milliseconds. Default is 3.0.
1805 Set decay. Default is 0.4.
1808 Set modulation speed in Hz. Default is 0.5.
1811 Set modulation type. Default is triangular.
1813 It accepts the following values:
1822 Audio pulsator is something between an autopanner and a tremolo.
1823 But it can produce funny stereo effects as well. Pulsator changes the volume
1824 of the left and right channel based on a LFO (low frequency oscillator) with
1825 different waveforms and shifted phases.
1826 This filter have the ability to define an offset between left and right
1827 channel. An offset of 0 means that both LFO shapes match each other.
1828 The left and right channel are altered equally - a conventional tremolo.
1829 An offset of 50% means that the shape of the right channel is exactly shifted
1830 in phase (or moved backwards about half of the frequency) - pulsator acts as
1831 an autopanner. At 1 both curves match again. Every setting in between moves the
1832 phase shift gapless between all stages and produces some "bypassing" sounds with
1833 sine and triangle waveforms. The more you set the offset near 1 (starting from
1834 the 0.5) the faster the signal passes from the left to the right speaker.
1836 The filter accepts the following options:
1840 Set input gain. By default it is 1. Range is [0.015625 - 64].
1843 Set output gain. By default it is 1. Range is [0.015625 - 64].
1846 Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
1847 sawup or sawdown. Default is sine.
1850 Set modulation. Define how much of original signal is affected by the LFO.
1853 Set left channel offset. Default is 0. Allowed range is [0 - 1].
1856 Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
1859 Set pulse width. Default is 1. Allowed range is [0 - 2].
1862 Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
1865 Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing
1869 Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing
1873 Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used
1874 if timing is set to hz.
1880 Resample the input audio to the specified parameters, using the
1881 libswresample library. If none are specified then the filter will
1882 automatically convert between its input and output.
1884 This filter is also able to stretch/squeeze the audio data to make it match
1885 the timestamps or to inject silence / cut out audio to make it match the
1886 timestamps, do a combination of both or do neither.
1888 The filter accepts the syntax
1889 [@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate}
1890 expresses a sample rate and @var{resampler_options} is a list of
1891 @var{key}=@var{value} pairs, separated by ":". See the
1892 @ref{Resampler Options,,"Resampler Options" section in the
1893 ffmpeg-resampler(1) manual,ffmpeg-resampler}
1894 for the complete list of supported options.
1896 @subsection Examples
1900 Resample the input audio to 44100Hz:
1906 Stretch/squeeze samples to the given timestamps, with a maximum of 1000
1907 samples per second compensation:
1909 aresample=async=1000
1915 Reverse an audio clip.
1917 Warning: This filter requires memory to buffer the entire clip, so trimming
1920 @subsection Examples
1924 Take the first 5 seconds of a clip, and reverse it.
1926 atrim=end=5,areverse
1930 @section asetnsamples
1932 Set the number of samples per each output audio frame.
1934 The last output packet may contain a different number of samples, as
1935 the filter will flush all the remaining samples when the input audio
1938 The filter accepts the following options:
1942 @item nb_out_samples, n
1943 Set the number of frames per each output audio frame. The number is
1944 intended as the number of samples @emph{per each channel}.
1945 Default value is 1024.
1948 If set to 1, the filter will pad the last audio frame with zeroes, so
1949 that the last frame will contain the same number of samples as the
1950 previous ones. Default value is 1.
1953 For example, to set the number of per-frame samples to 1234 and
1954 disable padding for the last frame, use:
1956 asetnsamples=n=1234:p=0
1961 Set the sample rate without altering the PCM data.
1962 This will result in a change of speed and pitch.
1964 The filter accepts the following options:
1967 @item sample_rate, r
1968 Set the output sample rate. Default is 44100 Hz.
1973 Show a line containing various information for each input audio frame.
1974 The input audio is not modified.
1976 The shown line contains a sequence of key/value pairs of the form
1977 @var{key}:@var{value}.
1979 The following values are shown in the output:
1983 The (sequential) number of the input frame, starting from 0.
1986 The presentation timestamp of the input frame, in time base units; the time base
1987 depends on the filter input pad, and is usually 1/@var{sample_rate}.
1990 The presentation timestamp of the input frame in seconds.
1993 position of the frame in the input stream, -1 if this information in
1994 unavailable and/or meaningless (for example in case of synthetic audio)
2003 The sample rate for the audio frame.
2006 The number of samples (per channel) in the frame.
2009 The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar
2010 audio, the data is treated as if all the planes were concatenated.
2012 @item plane_checksums
2013 A list of Adler-32 checksums for each data plane.
2019 Display time domain statistical information about the audio channels.
2020 Statistics are calculated and displayed for each audio channel and,
2021 where applicable, an overall figure is also given.
2023 It accepts the following option:
2026 Short window length in seconds, used for peak and trough RMS measurement.
2027 Default is @code{0.05} (50 milliseconds). Allowed range is @code{[0.01 - 10]}.
2031 Set metadata injection. All the metadata keys are prefixed with @code{lavfi.astats.X},
2032 where @code{X} is channel number starting from 1 or string @code{Overall}. Default is
2035 Available keys for each channel are:
2071 For example full key look like this @code{lavfi.astats.1.DC_offset} or
2072 this @code{lavfi.astats.Overall.Peak_count}.
2074 For description what each key means read below.
2077 Set number of frame after which stats are going to be recalculated.
2078 Default is disabled.
2081 A description of each shown parameter follows:
2085 Mean amplitude displacement from zero.
2088 Minimal sample level.
2091 Maximal sample level.
2093 @item Min difference
2094 Minimal difference between two consecutive samples.
2096 @item Max difference
2097 Maximal difference between two consecutive samples.
2099 @item Mean difference
2100 Mean difference between two consecutive samples.
2101 The average of each difference between two consecutive samples.
2103 @item RMS difference
2104 Root Mean Square difference between two consecutive samples.
2108 Standard peak and RMS level measured in dBFS.
2112 Peak and trough values for RMS level measured over a short window.
2115 Standard ratio of peak to RMS level (note: not in dB).
2118 Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels
2119 (i.e. either @var{Min level} or @var{Max level}).
2122 Number of occasions (not the number of samples) that the signal attained either
2123 @var{Min level} or @var{Max level}.
2126 Overall bit depth of audio. Number of bits used for each sample.
2129 Measured dynamic range of audio in dB.
2131 @item Zero crossings
2132 Number of points where the waveform crosses the zero level axis.
2134 @item Zero crossings rate
2135 Rate of Zero crossings and number of audio samples.
2142 The filter accepts exactly one parameter, the audio tempo. If not
2143 specified then the filter will assume nominal 1.0 tempo. Tempo must
2144 be in the [0.5, 100.0] range.
2146 Note that tempo greater than 2 will skip some samples rather than
2147 blend them in. If for any reason this is a concern it is always
2148 possible to daisy-chain several instances of atempo to achieve the
2149 desired product tempo.
2151 @subsection Examples
2155 Slow down audio to 80% tempo:
2161 To speed up audio to 300% tempo:
2167 To speed up audio to 300% tempo by daisy-chaining two atempo instances:
2169 atempo=sqrt(3),atempo=sqrt(3)
2175 Trim the input so that the output contains one continuous subpart of the input.
2177 It accepts the following parameters:
2180 Timestamp (in seconds) of the start of the section to keep. I.e. the audio
2181 sample with the timestamp @var{start} will be the first sample in the output.
2184 Specify time of the first audio sample that will be dropped, i.e. the
2185 audio sample immediately preceding the one with the timestamp @var{end} will be
2186 the last sample in the output.
2189 Same as @var{start}, except this option sets the start timestamp in samples
2193 Same as @var{end}, except this option sets the end timestamp in samples instead
2197 The maximum duration of the output in seconds.
2200 The number of the first sample that should be output.
2203 The number of the first sample that should be dropped.
2206 @option{start}, @option{end}, and @option{duration} are expressed as time
2207 duration specifications; see
2208 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
2210 Note that the first two sets of the start/end options and the @option{duration}
2211 option look at the frame timestamp, while the _sample options simply count the
2212 samples that pass through the filter. So start/end_pts and start/end_sample will
2213 give different results when the timestamps are wrong, inexact or do not start at
2214 zero. Also note that this filter does not modify the timestamps. If you wish
2215 to have the output timestamps start at zero, insert the asetpts filter after the
2218 If multiple start or end options are set, this filter tries to be greedy and
2219 keep all samples that match at least one of the specified constraints. To keep
2220 only the part that matches all the constraints at once, chain multiple atrim
2223 The defaults are such that all the input is kept. So it is possible to set e.g.
2224 just the end values to keep everything before the specified time.
2229 Drop everything except the second minute of input:
2231 ffmpeg -i INPUT -af atrim=60:120
2235 Keep only the first 1000 samples:
2237 ffmpeg -i INPUT -af atrim=end_sample=1000
2244 Apply a two-pole Butterworth band-pass filter with central
2245 frequency @var{frequency}, and (3dB-point) band-width width.
2246 The @var{csg} option selects a constant skirt gain (peak gain = Q)
2247 instead of the default: constant 0dB peak gain.
2248 The filter roll off at 6dB per octave (20dB per decade).
2250 The filter accepts the following options:
2254 Set the filter's central frequency. Default is @code{3000}.
2257 Constant skirt gain if set to 1. Defaults to 0.
2260 Set method to specify band-width of filter.
2275 Specify the band-width of a filter in width_type units.
2278 Specify which channels to filter, by default all available are filtered.
2281 @subsection Commands
2283 This filter supports the following commands:
2286 Change bandpass frequency.
2287 Syntax for the command is : "@var{frequency}"
2290 Change bandpass width_type.
2291 Syntax for the command is : "@var{width_type}"
2294 Change bandpass width.
2295 Syntax for the command is : "@var{width}"
2300 Apply a two-pole Butterworth band-reject filter with central
2301 frequency @var{frequency}, and (3dB-point) band-width @var{width}.
2302 The filter roll off at 6dB per octave (20dB per decade).
2304 The filter accepts the following options:
2308 Set the filter's central frequency. Default is @code{3000}.
2311 Set method to specify band-width of filter.
2326 Specify the band-width of a filter in width_type units.
2329 Specify which channels to filter, by default all available are filtered.
2332 @subsection Commands
2334 This filter supports the following commands:
2337 Change bandreject frequency.
2338 Syntax for the command is : "@var{frequency}"
2341 Change bandreject width_type.
2342 Syntax for the command is : "@var{width_type}"
2345 Change bandreject width.
2346 Syntax for the command is : "@var{width}"
2349 @section bass, lowshelf
2351 Boost or cut the bass (lower) frequencies of the audio using a two-pole
2352 shelving filter with a response similar to that of a standard
2353 hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
2355 The filter accepts the following options:
2359 Give the gain at 0 Hz. Its useful range is about -20
2360 (for a large cut) to +20 (for a large boost).
2361 Beware of clipping when using a positive gain.
2364 Set the filter's central frequency and so can be used
2365 to extend or reduce the frequency range to be boosted or cut.
2366 The default value is @code{100} Hz.
2369 Set method to specify band-width of filter.
2384 Determine how steep is the filter's shelf transition.
2387 Specify which channels to filter, by default all available are filtered.
2390 @subsection Commands
2392 This filter supports the following commands:
2395 Change bass frequency.
2396 Syntax for the command is : "@var{frequency}"
2399 Change bass width_type.
2400 Syntax for the command is : "@var{width_type}"
2404 Syntax for the command is : "@var{width}"
2408 Syntax for the command is : "@var{gain}"
2413 Apply a biquad IIR filter with the given coefficients.
2414 Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
2415 are the numerator and denominator coefficients respectively.
2416 and @var{channels}, @var{c} specify which channels to filter, by default all
2417 available are filtered.
2419 @subsection Commands
2421 This filter supports the following commands:
2429 Change biquad parameter.
2430 Syntax for the command is : "@var{value}"
2434 Bauer stereo to binaural transformation, which improves headphone listening of
2435 stereo audio records.
2437 To enable compilation of this filter you need to configure FFmpeg with
2438 @code{--enable-libbs2b}.
2440 It accepts the following parameters:
2444 Pre-defined crossfeed level.
2448 Default level (fcut=700, feed=50).
2451 Chu Moy circuit (fcut=700, feed=60).
2454 Jan Meier circuit (fcut=650, feed=95).
2459 Cut frequency (in Hz).
2468 Remap input channels to new locations.
2470 It accepts the following parameters:
2473 Map channels from input to output. The argument is a '|'-separated list of
2474 mappings, each in the @code{@var{in_channel}-@var{out_channel}} or
2475 @var{in_channel} form. @var{in_channel} can be either the name of the input
2476 channel (e.g. FL for front left) or its index in the input channel layout.
2477 @var{out_channel} is the name of the output channel or its index in the output
2478 channel layout. If @var{out_channel} is not given then it is implicitly an
2479 index, starting with zero and increasing by one for each mapping.
2481 @item channel_layout
2482 The channel layout of the output stream.
2485 If no mapping is present, the filter will implicitly map input channels to
2486 output channels, preserving indices.
2488 @subsection Examples
2492 For example, assuming a 5.1+downmix input MOV file,
2494 ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav
2496 will create an output WAV file tagged as stereo from the downmix channels of
2500 To fix a 5.1 WAV improperly encoded in AAC's native channel order
2502 ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav
2506 @section channelsplit
2508 Split each channel from an input audio stream into a separate output stream.
2510 It accepts the following parameters:
2512 @item channel_layout
2513 The channel layout of the input stream. The default is "stereo".
2515 A channel layout describing the channels to be extracted as separate output streams
2516 or "all" to extract each input channel as a separate stream. The default is "all".
2518 Choosing channels not present in channel layout in the input will result in an error.
2521 @subsection Examples
2525 For example, assuming a stereo input MP3 file,
2527 ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv
2529 will create an output Matroska file with two audio streams, one containing only
2530 the left channel and the other the right channel.
2533 Split a 5.1 WAV file into per-channel files:
2535 ffmpeg -i in.wav -filter_complex
2536 'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
2537 -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
2538 front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
2543 Extract only LFE from a 5.1 WAV file:
2545 ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1:channels=LFE[LFE]'
2546 -map '[LFE]' lfe.wav
2551 Add a chorus effect to the audio.
2553 Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
2555 Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
2556 constant, with chorus, it is varied using using sinusoidal or triangular modulation.
2557 The modulation depth defines the range the modulated delay is played before or after
2558 the delay. Hence the delayed sound will sound slower or faster, that is the delayed
2559 sound tuned around the original one, like in a chorus where some vocals are slightly
2562 It accepts the following parameters:
2565 Set input gain. Default is 0.4.
2568 Set output gain. Default is 0.4.
2571 Set delays. A typical delay is around 40ms to 60ms.
2583 @subsection Examples
2589 chorus=0.7:0.9:55:0.4:0.25:2
2595 chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
2599 Fuller sounding chorus with three delays:
2601 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
2606 Compress or expand the audio's dynamic range.
2608 It accepts the following parameters:
2614 A list of times in seconds for each channel over which the instantaneous level
2615 of the input signal is averaged to determine its volume. @var{attacks} refers to
2616 increase of volume and @var{decays} refers to decrease of volume. For most
2617 situations, the attack time (response to the audio getting louder) should be
2618 shorter than the decay time, because the human ear is more sensitive to sudden
2619 loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and
2620 a typical value for decay is 0.8 seconds.
2621 If specified number of attacks & decays is lower than number of channels, the last
2622 set attack/decay will be used for all remaining channels.
2625 A list of points for the transfer function, specified in dB relative to the
2626 maximum possible signal amplitude. Each key points list must be defined using
2627 the following syntax: @code{x0/y0|x1/y1|x2/y2|....} or
2628 @code{x0/y0 x1/y1 x2/y2 ....}
2630 The input values must be in strictly increasing order but the transfer function
2631 does not have to be monotonically rising. The point @code{0/0} is assumed but
2632 may be overridden (by @code{0/out-dBn}). Typical values for the transfer
2633 function are @code{-70/-70|-60/-20|1/0}.
2636 Set the curve radius in dB for all joints. It defaults to 0.01.
2639 Set the additional gain in dB to be applied at all points on the transfer
2640 function. This allows for easy adjustment of the overall gain.
2644 Set an initial volume, in dB, to be assumed for each channel when filtering
2645 starts. This permits the user to supply a nominal level initially, so that, for
2646 example, a very large gain is not applied to initial signal levels before the
2647 companding has begun to operate. A typical value for audio which is initially
2648 quiet is -90 dB. It defaults to 0.
2651 Set a delay, in seconds. The input audio is analyzed immediately, but audio is
2652 delayed before being fed to the volume adjuster. Specifying a delay
2653 approximately equal to the attack/decay times allows the filter to effectively
2654 operate in predictive rather than reactive mode. It defaults to 0.
2658 @subsection Examples
2662 Make music with both quiet and loud passages suitable for listening to in a
2665 compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
2668 Another example for audio with whisper and explosion parts:
2670 compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
2674 A noise gate for when the noise is at a lower level than the signal:
2676 compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
2680 Here is another noise gate, this time for when the noise is at a higher level
2681 than the signal (making it, in some ways, similar to squelch):
2683 compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
2687 2:1 compression starting at -6dB:
2689 compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
2693 2:1 compression starting at -9dB:
2695 compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
2699 2:1 compression starting at -12dB:
2701 compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
2705 2:1 compression starting at -18dB:
2707 compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
2711 3:1 compression starting at -15dB:
2713 compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
2719 compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
2725 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
2729 Hard limiter at -6dB:
2731 compand=attacks=0:points=-80/-80|-6/-6|20/-6
2735 Hard limiter at -12dB:
2737 compand=attacks=0:points=-80/-80|-12/-12|20/-12
2741 Hard noise gate at -35 dB:
2743 compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
2749 compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
2753 @section compensationdelay
2755 Compensation Delay Line is a metric based delay to compensate differing
2756 positions of microphones or speakers.
2758 For example, you have recorded guitar with two microphones placed in
2759 different location. Because the front of sound wave has fixed speed in
2760 normal conditions, the phasing of microphones can vary and depends on
2761 their location and interposition. The best sound mix can be achieved when
2762 these microphones are in phase (synchronized). Note that distance of
2763 ~30 cm between microphones makes one microphone to capture signal in
2764 antiphase to another microphone. That makes the final mix sounding moody.
2765 This filter helps to solve phasing problems by adding different delays
2766 to each microphone track and make them synchronized.
2768 The best result can be reached when you take one track as base and
2769 synchronize other tracks one by one with it.
2770 Remember that synchronization/delay tolerance depends on sample rate, too.
2771 Higher sample rates will give more tolerance.
2773 It accepts the following parameters:
2777 Set millimeters distance. This is compensation distance for fine tuning.
2781 Set cm distance. This is compensation distance for tightening distance setup.
2785 Set meters distance. This is compensation distance for hard distance setup.
2789 Set dry amount. Amount of unprocessed (dry) signal.
2793 Set wet amount. Amount of processed (wet) signal.
2797 Set temperature degree in Celsius. This is the temperature of the environment.
2802 Apply headphone crossfeed filter.
2804 Crossfeed is the process of blending the left and right channels of stereo
2806 It is mainly used to reduce extreme stereo separation of low frequencies.
2808 The intent is to produce more speaker like sound to the listener.
2810 The filter accepts the following options:
2814 Set strength of crossfeed. Default is 0.2. Allowed range is from 0 to 1.
2815 This sets gain of low shelf filter for side part of stereo image.
2816 Default is -6dB. Max allowed is -30db when strength is set to 1.
2819 Set soundstage wideness. Default is 0.5. Allowed range is from 0 to 1.
2820 This sets cut off frequency of low shelf filter. Default is cut off near
2821 1550 Hz. With range set to 1 cut off frequency is set to 2100 Hz.
2824 Set input gain. Default is 0.9.
2827 Set output gain. Default is 1.
2830 @section crystalizer
2831 Simple algorithm to expand audio dynamic range.
2833 The filter accepts the following options:
2837 Sets the intensity of effect (default: 2.0). Must be in range between 0.0
2838 (unchanged sound) to 10.0 (maximum effect).
2841 Enable clipping. By default is enabled.
2845 Apply a DC shift to the audio.
2847 This can be useful to remove a DC offset (caused perhaps by a hardware problem
2848 in the recording chain) from the audio. The effect of a DC offset is reduced
2849 headroom and hence volume. The @ref{astats} filter can be used to determine if
2850 a signal has a DC offset.
2854 Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift
2858 Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
2859 used to prevent clipping.
2863 Measure audio dynamic range.
2865 DR values of 14 and higher is found in very dynamic material. DR of 8 to 13
2866 is found in transition material. And anything less that 8 have very poor dynamics
2867 and is very compressed.
2869 The filter accepts the following options:
2873 Set window length in seconds used to split audio into segments of equal length.
2874 Default is 3 seconds.
2878 Dynamic Audio Normalizer.
2880 This filter applies a certain amount of gain to the input audio in order
2881 to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
2882 contrast to more "simple" normalization algorithms, the Dynamic Audio
2883 Normalizer *dynamically* re-adjusts the gain factor to the input audio.
2884 This allows for applying extra gain to the "quiet" sections of the audio
2885 while avoiding distortions or clipping the "loud" sections. In other words:
2886 The Dynamic Audio Normalizer will "even out" the volume of quiet and loud
2887 sections, in the sense that the volume of each section is brought to the
2888 same target level. Note, however, that the Dynamic Audio Normalizer achieves
2889 this goal *without* applying "dynamic range compressing". It will retain 100%
2890 of the dynamic range *within* each section of the audio file.
2894 Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.
2895 Default is 500 milliseconds.
2896 The Dynamic Audio Normalizer processes the input audio in small chunks,
2897 referred to as frames. This is required, because a peak magnitude has no
2898 meaning for just a single sample value. Instead, we need to determine the
2899 peak magnitude for a contiguous sequence of sample values. While a "standard"
2900 normalizer would simply use the peak magnitude of the complete file, the
2901 Dynamic Audio Normalizer determines the peak magnitude individually for each
2902 frame. The length of a frame is specified in milliseconds. By default, the
2903 Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has
2904 been found to give good results with most files.
2905 Note that the exact frame length, in number of samples, will be determined
2906 automatically, based on the sampling rate of the individual input audio file.
2909 Set the Gaussian filter window size. In range from 3 to 301, must be odd
2910 number. Default is 31.
2911 Probably the most important parameter of the Dynamic Audio Normalizer is the
2912 @code{window size} of the Gaussian smoothing filter. The filter's window size
2913 is specified in frames, centered around the current frame. For the sake of
2914 simplicity, this must be an odd number. Consequently, the default value of 31
2915 takes into account the current frame, as well as the 15 preceding frames and
2916 the 15 subsequent frames. Using a larger window results in a stronger
2917 smoothing effect and thus in less gain variation, i.e. slower gain
2918 adaptation. Conversely, using a smaller window results in a weaker smoothing
2919 effect and thus in more gain variation, i.e. faster gain adaptation.
2920 In other words, the more you increase this value, the more the Dynamic Audio
2921 Normalizer will behave like a "traditional" normalization filter. On the
2922 contrary, the more you decrease this value, the more the Dynamic Audio
2923 Normalizer will behave like a dynamic range compressor.
2926 Set the target peak value. This specifies the highest permissible magnitude
2927 level for the normalized audio input. This filter will try to approach the
2928 target peak magnitude as closely as possible, but at the same time it also
2929 makes sure that the normalized signal will never exceed the peak magnitude.
2930 A frame's maximum local gain factor is imposed directly by the target peak
2931 magnitude. The default value is 0.95 and thus leaves a headroom of 5%*.
2932 It is not recommended to go above this value.
2935 Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.
2936 The Dynamic Audio Normalizer determines the maximum possible (local) gain
2937 factor for each input frame, i.e. the maximum gain factor that does not
2938 result in clipping or distortion. The maximum gain factor is determined by
2939 the frame's highest magnitude sample. However, the Dynamic Audio Normalizer
2940 additionally bounds the frame's maximum gain factor by a predetermined
2941 (global) maximum gain factor. This is done in order to avoid excessive gain
2942 factors in "silent" or almost silent frames. By default, the maximum gain
2943 factor is 10.0, For most inputs the default value should be sufficient and
2944 it usually is not recommended to increase this value. Though, for input
2945 with an extremely low overall volume level, it may be necessary to allow even
2946 higher gain factors. Note, however, that the Dynamic Audio Normalizer does
2947 not simply apply a "hard" threshold (i.e. cut off values above the threshold).
2948 Instead, a "sigmoid" threshold function will be applied. This way, the
2949 gain factors will smoothly approach the threshold value, but never exceed that
2953 Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.
2954 By default, the Dynamic Audio Normalizer performs "peak" normalization.
2955 This means that the maximum local gain factor for each frame is defined
2956 (only) by the frame's highest magnitude sample. This way, the samples can
2957 be amplified as much as possible without exceeding the maximum signal
2958 level, i.e. without clipping. Optionally, however, the Dynamic Audio
2959 Normalizer can also take into account the frame's root mean square,
2960 abbreviated RMS. In electrical engineering, the RMS is commonly used to
2961 determine the power of a time-varying signal. It is therefore considered
2962 that the RMS is a better approximation of the "perceived loudness" than
2963 just looking at the signal's peak magnitude. Consequently, by adjusting all
2964 frames to a constant RMS value, a uniform "perceived loudness" can be
2965 established. If a target RMS value has been specified, a frame's local gain
2966 factor is defined as the factor that would result in exactly that RMS value.
2967 Note, however, that the maximum local gain factor is still restricted by the
2968 frame's highest magnitude sample, in order to prevent clipping.
2971 Enable channels coupling. By default is enabled.
2972 By default, the Dynamic Audio Normalizer will amplify all channels by the same
2973 amount. This means the same gain factor will be applied to all channels, i.e.
2974 the maximum possible gain factor is determined by the "loudest" channel.
2975 However, in some recordings, it may happen that the volume of the different
2976 channels is uneven, e.g. one channel may be "quieter" than the other one(s).
2977 In this case, this option can be used to disable the channel coupling. This way,
2978 the gain factor will be determined independently for each channel, depending
2979 only on the individual channel's highest magnitude sample. This allows for
2980 harmonizing the volume of the different channels.
2983 Enable DC bias correction. By default is disabled.
2984 An audio signal (in the time domain) is a sequence of sample values.
2985 In the Dynamic Audio Normalizer these sample values are represented in the
2986 -1.0 to 1.0 range, regardless of the original input format. Normally, the
2987 audio signal, or "waveform", should be centered around the zero point.
2988 That means if we calculate the mean value of all samples in a file, or in a
2989 single frame, then the result should be 0.0 or at least very close to that
2990 value. If, however, there is a significant deviation of the mean value from
2991 0.0, in either positive or negative direction, this is referred to as a
2992 DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic
2993 Audio Normalizer provides optional DC bias correction.
2994 With DC bias correction enabled, the Dynamic Audio Normalizer will determine
2995 the mean value, or "DC correction" offset, of each input frame and subtract
2996 that value from all of the frame's sample values which ensures those samples
2997 are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
2998 boundaries, the DC correction offset values will be interpolated smoothly
2999 between neighbouring frames.
3002 Enable alternative boundary mode. By default is disabled.
3003 The Dynamic Audio Normalizer takes into account a certain neighbourhood
3004 around each frame. This includes the preceding frames as well as the
3005 subsequent frames. However, for the "boundary" frames, located at the very
3006 beginning and at the very end of the audio file, not all neighbouring
3007 frames are available. In particular, for the first few frames in the audio
3008 file, the preceding frames are not known. And, similarly, for the last few
3009 frames in the audio file, the subsequent frames are not known. Thus, the
3010 question arises which gain factors should be assumed for the missing frames
3011 in the "boundary" region. The Dynamic Audio Normalizer implements two modes
3012 to deal with this situation. The default boundary mode assumes a gain factor
3013 of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and
3014 "fade out" at the beginning and at the end of the input, respectively.
3017 Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
3018 By default, the Dynamic Audio Normalizer does not apply "traditional"
3019 compression. This means that signal peaks will not be pruned and thus the
3020 full dynamic range will be retained within each local neighbourhood. However,
3021 in some cases it may be desirable to combine the Dynamic Audio Normalizer's
3022 normalization algorithm with a more "traditional" compression.
3023 For this purpose, the Dynamic Audio Normalizer provides an optional compression
3024 (thresholding) function. If (and only if) the compression feature is enabled,
3025 all input frames will be processed by a soft knee thresholding function prior
3026 to the actual normalization process. Put simply, the thresholding function is
3027 going to prune all samples whose magnitude exceeds a certain threshold value.
3028 However, the Dynamic Audio Normalizer does not simply apply a fixed threshold
3029 value. Instead, the threshold value will be adjusted for each individual
3031 In general, smaller parameters result in stronger compression, and vice versa.
3032 Values below 3.0 are not recommended, because audible distortion may appear.
3037 Make audio easier to listen to on headphones.
3039 This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
3040 so that when listened to on headphones the stereo image is moved from
3041 inside your head (standard for headphones) to outside and in front of
3042 the listener (standard for speakers).
3048 Apply a two-pole peaking equalisation (EQ) filter. With this
3049 filter, the signal-level at and around a selected frequency can
3050 be increased or decreased, whilst (unlike bandpass and bandreject
3051 filters) that at all other frequencies is unchanged.
3053 In order to produce complex equalisation curves, this filter can
3054 be given several times, each with a different central frequency.
3056 The filter accepts the following options:
3060 Set the filter's central frequency in Hz.
3063 Set method to specify band-width of filter.
3078 Specify the band-width of a filter in width_type units.
3081 Set the required gain or attenuation in dB.
3082 Beware of clipping when using a positive gain.
3085 Specify which channels to filter, by default all available are filtered.
3088 @subsection Examples
3091 Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:
3093 equalizer=f=1000:t=h:width=200:g=-10
3097 Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:
3099 equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5
3103 @subsection Commands
3105 This filter supports the following commands:
3108 Change equalizer frequency.
3109 Syntax for the command is : "@var{frequency}"
3112 Change equalizer width_type.
3113 Syntax for the command is : "@var{width_type}"
3116 Change equalizer width.
3117 Syntax for the command is : "@var{width}"
3120 Change equalizer gain.
3121 Syntax for the command is : "@var{gain}"
3124 @section extrastereo
3126 Linearly increases the difference between left and right channels which
3127 adds some sort of "live" effect to playback.
3129 The filter accepts the following options:
3133 Sets the difference coefficient (default: 2.5). 0.0 means mono sound
3134 (average of both channels), with 1.0 sound will be unchanged, with
3135 -1.0 left and right channels will be swapped.
3138 Enable clipping. By default is enabled.
3141 @section firequalizer
3142 Apply FIR Equalization using arbitrary frequency response.
3144 The filter accepts the following option:
3148 Set gain curve equation (in dB). The expression can contain variables:
3151 the evaluated frequency
3155 channel number, set to 0 when multichannels evaluation is disabled
3157 channel id, see libavutil/channel_layout.h, set to the first channel id when
3158 multichannels evaluation is disabled
3162 channel_layout, see libavutil/channel_layout.h
3167 @item gain_interpolate(f)
3168 interpolate gain on frequency f based on gain_entry
3169 @item cubic_interpolate(f)
3170 same as gain_interpolate, but smoother
3172 This option is also available as command. Default is @code{gain_interpolate(f)}.
3175 Set gain entry for gain_interpolate function. The expression can
3179 store gain entry at frequency f with value g
3181 This option is also available as command.
3184 Set filter delay in seconds. Higher value means more accurate.
3185 Default is @code{0.01}.
3188 Set filter accuracy in Hz. Lower value means more accurate.
3189 Default is @code{5}.
3192 Set window function. Acceptable values are:
3195 rectangular window, useful when gain curve is already smooth
3197 hann window (default)
3203 3-terms continuous 1st derivative nuttall window
3205 minimum 3-terms discontinuous nuttall window
3207 4-terms continuous 1st derivative nuttall window
3209 minimum 4-terms discontinuous nuttall (blackman-nuttall) window
3211 blackman-harris window
3217 If enabled, use fixed number of audio samples. This improves speed when
3218 filtering with large delay. Default is disabled.
3221 Enable multichannels evaluation on gain. Default is disabled.
3224 Enable zero phase mode by subtracting timestamp to compensate delay.
3225 Default is disabled.
3228 Set scale used by gain. Acceptable values are:
3231 linear frequency, linear gain
3233 linear frequency, logarithmic (in dB) gain (default)
3235 logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain
3237 logarithmic frequency, logarithmic gain
3241 Set file for dumping, suitable for gnuplot.
3244 Set scale for dumpfile. Acceptable values are same with scale option.
3248 Enable 2-channel convolution using complex FFT. This improves speed significantly.
3249 Default is disabled.
3252 Enable minimum phase impulse response. Default is disabled.
3255 @subsection Examples
3260 firequalizer=gain='if(lt(f,1000), 0, -INF)'
3263 lowpass at 1000 Hz with gain_entry:
3265 firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
3268 custom equalization:
3270 firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
3273 higher delay with zero phase to compensate delay:
3275 firequalizer=delay=0.1:fixed=on:zero_phase=on
3278 lowpass on left channel, highpass on right channel:
3280 firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
3281 :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
3286 Apply a flanging effect to the audio.
3288 The filter accepts the following options:
3292 Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
3295 Set added sweep delay in milliseconds. Range from 0 to 10. Default value is 2.
3298 Set percentage regeneration (delayed signal feedback). Range from -95 to 95.
3302 Set percentage of delayed signal mixed with original. Range from 0 to 100.
3303 Default value is 71.
3306 Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
3309 Set swept wave shape, can be @var{triangular} or @var{sinusoidal}.
3310 Default value is @var{sinusoidal}.
3313 Set swept wave percentage-shift for multi channel. Range from 0 to 100.
3314 Default value is 25.
3317 Set delay-line interpolation, @var{linear} or @var{quadratic}.
3318 Default is @var{linear}.
3322 Apply Haas effect to audio.
3324 Note that this makes most sense to apply on mono signals.
3325 With this filter applied to mono signals it give some directionality and
3326 stretches its stereo image.
3328 The filter accepts the following options:
3332 Set input level. By default is @var{1}, or 0dB
3335 Set output level. By default is @var{1}, or 0dB.
3338 Set gain applied to side part of signal. By default is @var{1}.
3341 Set kind of middle source. Can be one of the following:
3351 Pick middle part signal of stereo image.
3354 Pick side part signal of stereo image.
3358 Change middle phase. By default is disabled.
3361 Set left channel delay. By default is @var{2.05} milliseconds.
3364 Set left channel balance. By default is @var{-1}.
3367 Set left channel gain. By default is @var{1}.
3370 Change left phase. By default is disabled.
3373 Set right channel delay. By defaults is @var{2.12} milliseconds.
3376 Set right channel balance. By default is @var{1}.
3379 Set right channel gain. By default is @var{1}.
3382 Change right phase. By default is enabled.
3387 Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with
3388 embedded HDCD codes is expanded into a 20-bit PCM stream.
3390 The filter supports the Peak Extend and Low-level Gain Adjustment features
3391 of HDCD, and detects the Transient Filter flag.
3394 ffmpeg -i HDCD16.flac -af hdcd OUT24.flac
3397 When using the filter with wav, note the default encoding for wav is 16-bit,
3398 so the resulting 20-bit stream will be truncated back to 16-bit. Use something
3399 like @command{-acodec pcm_s24le} after the filter to get 24-bit PCM output.
3401 ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
3402 ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav
3405 The filter accepts the following options:
3408 @item disable_autoconvert
3409 Disable any automatic format conversion or resampling in the filter graph.
3411 @item process_stereo
3412 Process the stereo channels together. If target_gain does not match between
3413 channels, consider it invalid and use the last valid target_gain.
3416 Set the code detect timer period in ms.
3419 Always extend peaks above -3dBFS even if PE isn't signaled.
3422 Replace audio with a solid tone and adjust the amplitude to signal some
3423 specific aspect of the decoding process. The output file can be loaded in
3424 an audio editor alongside the original to aid analysis.
3426 @code{analyze_mode=pe:force_pe=true} can be used to see all samples above the PE level.
3433 Gain adjustment level at each sample
3435 Samples where peak extend occurs
3437 Samples where the code detect timer is active
3439 Samples where the target gain does not match between channels
3445 Apply head-related transfer functions (HRTFs) to create virtual
3446 loudspeakers around the user for binaural listening via headphones.
3447 The HRIRs are provided via additional streams, for each channel
3448 one stereo input stream is needed.
3450 The filter accepts the following options:
3454 Set mapping of input streams for convolution.
3455 The argument is a '|'-separated list of channel names in order as they
3456 are given as additional stream inputs for filter.
3457 This also specify number of input streams. Number of input streams
3458 must be not less than number of channels in first stream plus one.
3461 Set gain applied to audio. Value is in dB. Default is 0.
3464 Set processing type. Can be @var{time} or @var{freq}. @var{time} is
3465 processing audio in time domain which is slow.
3466 @var{freq} is processing audio in frequency domain which is fast.
3467 Default is @var{freq}.
3470 Set custom gain for LFE channels. Value is in dB. Default is 0.
3473 Set size of frame in number of samples which will be processed at once.
3474 Default value is @var{1024}. Allowed range is from 1024 to 96000.
3477 Set format of hrir stream.
3478 Default value is @var{stereo}. Alternative value is @var{multich}.
3479 If value is set to @var{stereo}, number of additional streams should
3480 be greater or equal to number of input channels in first input stream.
3481 Also each additional stream should have stereo number of channels.
3482 If value is set to @var{multich}, number of additional streams should
3483 be exactly one. Also number of input channels of additional stream
3484 should be equal or greater than twice number of channels of first input
3488 @subsection Examples
3492 Full example using wav files as coefficients with amovie filters for 7.1 downmix,
3493 each amovie filter use stereo file with IR coefficients as input.
3494 The files give coefficients for each position of virtual loudspeaker:
3496 ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
3501 Full example using wav files as coefficients with amovie filters for 7.1 downmix,
3502 but now in @var{multich} @var{hrir} format.
3504 ffmpeg -i input.wav -lavfi-complex "amovie=minp.wav[hrirs],[a:0][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
3511 Apply a high-pass filter with 3dB point frequency.
3512 The filter can be either single-pole, or double-pole (the default).
3513 The filter roll off at 6dB per pole per octave (20dB per pole per decade).
3515 The filter accepts the following options:
3519 Set frequency in Hz. Default is 3000.
3522 Set number of poles. Default is 2.
3525 Set method to specify band-width of filter.
3540 Specify the band-width of a filter in width_type units.
3541 Applies only to double-pole filter.
3542 The default is 0.707q and gives a Butterworth response.
3545 Specify which channels to filter, by default all available are filtered.
3548 @subsection Commands
3550 This filter supports the following commands:
3553 Change highpass frequency.
3554 Syntax for the command is : "@var{frequency}"
3557 Change highpass width_type.
3558 Syntax for the command is : "@var{width_type}"
3561 Change highpass width.
3562 Syntax for the command is : "@var{width}"
3567 Join multiple input streams into one multi-channel stream.
3569 It accepts the following parameters:
3573 The number of input streams. It defaults to 2.
3575 @item channel_layout
3576 The desired output channel layout. It defaults to stereo.
3579 Map channels from inputs to output. The argument is a '|'-separated list of
3580 mappings, each in the @code{@var{input_idx}.@var{in_channel}-@var{out_channel}}
3581 form. @var{input_idx} is the 0-based index of the input stream. @var{in_channel}
3582 can be either the name of the input channel (e.g. FL for front left) or its
3583 index in the specified input stream. @var{out_channel} is the name of the output
3587 The filter will attempt to guess the mappings when they are not specified
3588 explicitly. It does so by first trying to find an unused matching input channel
3589 and if that fails it picks the first unused input channel.
3591 Join 3 inputs (with properly set channel layouts):
3593 ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
3596 Build a 5.1 output from 6 single-channel streams:
3598 ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
3599 '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'
3605 Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.
3607 To enable compilation of this filter you need to configure FFmpeg with
3608 @code{--enable-ladspa}.
3612 Specifies the name of LADSPA plugin library to load. If the environment
3613 variable @env{LADSPA_PATH} is defined, the LADSPA plugin is searched in
3614 each one of the directories specified by the colon separated list in
3615 @env{LADSPA_PATH}, otherwise in the standard LADSPA paths, which are in
3616 this order: @file{HOME/.ladspa/lib/}, @file{/usr/local/lib/ladspa/},
3617 @file{/usr/lib/ladspa/}.
3620 Specifies the plugin within the library. Some libraries contain only
3621 one plugin, but others contain many of them. If this is not set filter
3622 will list all available plugins within the specified library.
3625 Set the '|' separated list of controls which are zero or more floating point
3626 values that determine the behavior of the loaded plugin (for example delay,
3628 Controls need to be defined using the following syntax:
3629 c0=@var{value0}|c1=@var{value1}|c2=@var{value2}|..., where
3630 @var{valuei} is the value set on the @var{i}-th control.
3631 Alternatively they can be also defined using the following syntax:
3632 @var{value0}|@var{value1}|@var{value2}|..., where
3633 @var{valuei} is the value set on the @var{i}-th control.
3634 If @option{controls} is set to @code{help}, all available controls and
3635 their valid ranges are printed.
3637 @item sample_rate, s
3638 Specify the sample rate, default to 44100. Only used if plugin have
3642 Set the number of samples per channel per each output frame, default
3643 is 1024. Only used if plugin have zero inputs.
3646 Set the minimum duration of the sourced audio. See
3647 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
3648 for the accepted syntax.
3649 Note that the resulting duration may be greater than the specified duration,
3650 as the generated audio is always cut at the end of a complete frame.
3651 If not specified, or the expressed duration is negative, the audio is
3652 supposed to be generated forever.
3653 Only used if plugin have zero inputs.
3657 @subsection Examples
3661 List all available plugins within amp (LADSPA example plugin) library:
3667 List all available controls and their valid ranges for @code{vcf_notch}
3668 plugin from @code{VCF} library:
3670 ladspa=f=vcf:p=vcf_notch:c=help
3674 Simulate low quality audio equipment using @code{Computer Music Toolkit} (CMT)
3677 ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
3681 Add reverberation to the audio using TAP-plugins
3682 (Tom's Audio Processing plugins):
3684 ladspa=file=tap_reverb:tap_reverb
3688 Generate white noise, with 0.2 amplitude:
3690 ladspa=file=cmt:noise_source_white:c=c0=.2
3694 Generate 20 bpm clicks using plugin @code{C* Click - Metronome} from the
3695 @code{C* Audio Plugin Suite} (CAPS) library:
3697 ladspa=file=caps:Click:c=c1=20'
3701 Apply @code{C* Eq10X2 - Stereo 10-band equaliser} effect:
3703 ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
3707 Increase volume by 20dB using fast lookahead limiter from Steve Harris
3708 @code{SWH Plugins} collection:
3710 ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
3714 Attenuate low frequencies using Multiband EQ from Steve Harris
3715 @code{SWH Plugins} collection:
3717 ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
3721 Reduce stereo image using @code{Narrower} from the @code{C* Audio Plugin Suite}
3724 ladspa=caps:Narrower
3728 Another white noise, now using @code{C* Audio Plugin Suite} (CAPS) library:
3730 ladspa=caps:White:.2
3734 Some fractal noise, using @code{C* Audio Plugin Suite} (CAPS) library:
3736 ladspa=caps:Fractal:c=c1=1
3740 Dynamic volume normalization using @code{VLevel} plugin:
3742 ladspa=vlevel-ladspa:vlevel_mono
3746 @subsection Commands
3748 This filter supports the following commands:
3751 Modify the @var{N}-th control value.
3753 If the specified value is not valid, it is ignored and prior one is kept.
3758 EBU R128 loudness normalization. Includes both dynamic and linear normalization modes.
3759 Support for both single pass (livestreams, files) and double pass (files) modes.
3760 This algorithm can target IL, LRA, and maximum true peak. To accurately detect true peaks,
3761 the audio stream will be upsampled to 192 kHz unless the normalization mode is linear.
3762 Use the @code{-ar} option or @code{aresample} filter to explicitly set an output sample rate.
3764 The filter accepts the following options:
3768 Set integrated loudness target.
3769 Range is -70.0 - -5.0. Default value is -24.0.
3772 Set loudness range target.
3773 Range is 1.0 - 20.0. Default value is 7.0.
3776 Set maximum true peak.
3777 Range is -9.0 - +0.0. Default value is -2.0.
3779 @item measured_I, measured_i
3780 Measured IL of input file.
3781 Range is -99.0 - +0.0.
3783 @item measured_LRA, measured_lra
3784 Measured LRA of input file.
3785 Range is 0.0 - 99.0.
3787 @item measured_TP, measured_tp
3788 Measured true peak of input file.
3789 Range is -99.0 - +99.0.
3791 @item measured_thresh
3792 Measured threshold of input file.
3793 Range is -99.0 - +0.0.
3796 Set offset gain. Gain is applied before the true-peak limiter.
3797 Range is -99.0 - +99.0. Default is +0.0.
3800 Normalize linearly if possible.
3801 measured_I, measured_LRA, measured_TP, and measured_thresh must also
3802 to be specified in order to use this mode.
3803 Options are true or false. Default is true.
3806 Treat mono input files as "dual-mono". If a mono file is intended for playback
3807 on a stereo system, its EBU R128 measurement will be perceptually incorrect.
3808 If set to @code{true}, this option will compensate for this effect.
3809 Multi-channel input files are not affected by this option.
3810 Options are true or false. Default is false.
3813 Set print format for stats. Options are summary, json, or none.
3814 Default value is none.
3819 Apply a low-pass filter with 3dB point frequency.
3820 The filter can be either single-pole or double-pole (the default).
3821 The filter roll off at 6dB per pole per octave (20dB per pole per decade).
3823 The filter accepts the following options:
3827 Set frequency in Hz. Default is 500.
3830 Set number of poles. Default is 2.
3833 Set method to specify band-width of filter.
3848 Specify the band-width of a filter in width_type units.
3849 Applies only to double-pole filter.
3850 The default is 0.707q and gives a Butterworth response.
3853 Specify which channels to filter, by default all available are filtered.
3856 @subsection Examples
3859 Lowpass only LFE channel, it LFE is not present it does nothing:
3865 @subsection Commands
3867 This filter supports the following commands:
3870 Change lowpass frequency.
3871 Syntax for the command is : "@var{frequency}"
3874 Change lowpass width_type.
3875 Syntax for the command is : "@var{width_type}"
3878 Change lowpass width.
3879 Syntax for the command is : "@var{width}"
3884 Load a LV2 (LADSPA Version 2) plugin.
3886 To enable compilation of this filter you need to configure FFmpeg with
3887 @code{--enable-lv2}.
3891 Specifies the plugin URI. You may need to escape ':'.
3894 Set the '|' separated list of controls which are zero or more floating point
3895 values that determine the behavior of the loaded plugin (for example delay,
3897 If @option{controls} is set to @code{help}, all available controls and
3898 their valid ranges are printed.
3900 @item sample_rate, s
3901 Specify the sample rate, default to 44100. Only used if plugin have
3905 Set the number of samples per channel per each output frame, default
3906 is 1024. Only used if plugin have zero inputs.
3909 Set the minimum duration of the sourced audio. See
3910 @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
3911 for the accepted syntax.
3912 Note that the resulting duration may be greater than the specified duration,
3913 as the generated audio is always cut at the end of a complete frame.
3914 If not specified, or the expressed duration is negative, the audio is
3915 supposed to be generated forever.
3916 Only used if plugin have zero inputs.
3919 @subsection Examples
3923 Apply bass enhancer plugin from Calf:
3925 lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2
3929 Apply vinyl plugin from Calf:
3931 lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5
3935 Apply bit crusher plugin from ArtyFX:
3937 lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3
3942 Multiband Compress or expand the audio's dynamic range.
3944 The input audio is divided into bands using 4th order Linkwitz-Riley IIRs.
3945 This is akin to the crossover of a loudspeaker, and results in flat frequency
3946 response when absent compander action.
3948 It accepts the following parameters:
3952 This option syntax is:
3953 attack,decay,[attack,decay..] soft-knee points crossover_frequency [delay [initial_volume [gain]]] | attack,decay ...
3954 For explanation of each item refer to compand filter documentation.
3960 Mix channels with specific gain levels. The filter accepts the output
3961 channel layout followed by a set of channels definitions.
3963 This filter is also designed to efficiently remap the channels of an audio
3966 The filter accepts parameters of the form:
3967 "@var{l}|@var{outdef}|@var{outdef}|..."
3971 output channel layout or number of channels
3974 output channel specification, of the form:
3975 "@var{out_name}=[@var{gain}*]@var{in_name}[(+-)[@var{gain}*]@var{in_name}...]"
3978 output channel to define, either a channel name (FL, FR, etc.) or a channel
3979 number (c0, c1, etc.)
3982 multiplicative coefficient for the channel, 1 leaving the volume unchanged
3985 input channel to use, see out_name for details; it is not possible to mix
3986 named and numbered input channels
3989 If the `=' in a channel specification is replaced by `<', then the gains for
3990 that specification will be renormalized so that the total is 1, thus
3991 avoiding clipping noise.
3993 @subsection Mixing examples
3995 For example, if you want to down-mix from stereo to mono, but with a bigger
3996 factor for the left channel:
3998 pan=1c|c0=0.9*c0+0.1*c1
4001 A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
4002 7-channels surround:
4004 pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
4007 Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
4008 that should be preferred (see "-ac" option) unless you have very specific
4011 @subsection Remapping examples
4013 The channel remapping will be effective if, and only if:
4016 @item gain coefficients are zeroes or ones,
4017 @item only one input per channel output,
4020 If all these conditions are satisfied, the filter will notify the user ("Pure
4021 channel mapping detected"), and use an optimized and lossless method to do the
4024 For example, if you have a 5.1 source and want a stereo audio stream by
4025 dropping the extra channels:
4027 pan="stereo| c0=FL | c1=FR"
4030 Given the same source, you can also switch front left and front right channels
4031 and keep the input channel layout:
4033 pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"
4036 If the input is a stereo audio stream, you can mute the front left channel (and
4037 still keep the stereo channel layout) with:
4042 Still with a stereo audio stream input, you can copy the right channel in both
4043 front left and right:
4045 pan="stereo| c0=FR | c1=FR"
4050 ReplayGain scanner filter. This filter takes an audio stream as an input and
4051 outputs it unchanged.
4052 At end of filtering it displays @code{track_gain} and @code{track_peak}.
4056 Convert the audio sample format, sample rate and channel layout. It is
4057 not meant to be used directly.
4060 Apply time-stretching and pitch-shifting with librubberband.
4062 To enable compilation of this filter, you need to configure FFmpeg with
4063 @code{--enable-librubberband}.
4065 The filter accepts the following options:
4069 Set tempo scale factor.
4072 Set pitch scale factor.
4075 Set transients detector.
4076 Possible values are:
4085 Possible values are:
4094 Possible values are:
4101 Set processing window size.
4102 Possible values are:
4111 Possible values are:
4118 Enable formant preservation when shift pitching.
4119 Possible values are:
4127 Possible values are:
4136 Possible values are:
4143 @section sidechaincompress
4145 This filter acts like normal compressor but has the ability to compress
4146 detected signal using second input signal.
4147 It needs two input streams and returns one output stream.
4148 First input stream will be processed depending on second stream signal.
4149 The filtered signal then can be filtered with other filters in later stages of
4150 processing. See @ref{pan} and @ref{amerge} filter.
4152 The filter accepts the following options:
4156 Set input gain. Default is 1. Range is between 0.015625 and 64.
4159 If a signal of second stream raises above this level it will affect the gain
4160 reduction of first stream.
4161 By default is 0.125. Range is between 0.00097563 and 1.
4164 Set a ratio about which the signal is reduced. 1:2 means that if the level
4165 raised 4dB above the threshold, it will be only 2dB above after the reduction.
4166 Default is 2. Range is between 1 and 20.
4169 Amount of milliseconds the signal has to rise above the threshold before gain
4170 reduction starts. Default is 20. Range is between 0.01 and 2000.
4173 Amount of milliseconds the signal has to fall below the threshold before
4174 reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
4177 Set the amount by how much signal will be amplified after processing.
4178 Default is 1. Range is from 1 to 64.
4181 Curve the sharp knee around the threshold to enter gain reduction more softly.
4182 Default is 2.82843. Range is between 1 and 8.
4185 Choose if the @code{average} level between all channels of side-chain stream
4186 or the louder(@code{maximum}) channel of side-chain stream affects the
4187 reduction. Default is @code{average}.
4190 Should the exact signal be taken in case of @code{peak} or an RMS one in case
4191 of @code{rms}. Default is @code{rms} which is mainly smoother.
4194 Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
4197 How much to use compressed signal in output. Default is 1.
4198 Range is between 0 and 1.
4201 @subsection Examples
4205 Full ffmpeg example taking 2 audio inputs, 1st input to be compressed
4206 depending on the signal of 2nd input and later compressed signal to be
4207 merged with 2nd input:
4209 ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"
4213 @section sidechaingate
4215 A sidechain gate acts like a normal (wideband) gate but has the ability to
4216 filter the detected signal before sending it to the gain reduction stage.
4217 Normally a gate uses the full range signal to detect a level above the
4219 For example: If you cut all lower frequencies from your sidechain signal
4220 the gate will decrease the volume of your track only if not enough highs
4221 appear. With this technique you are able to reduce the resonation of a
4222 natural drum or remove "rumbling" of muted strokes from a heavily distorted
4224 It needs two input streams and returns one output stream.
4225 First input stream will be processed depending on second stream signal.
4227 The filter accepts the following options:
4231 Set input level before filtering.
4232 Default is 1. Allowed range is from 0.015625 to 64.
4235 Set the level of gain reduction when the signal is below the threshold.
4236 Default is 0.06125. Allowed range is from 0 to 1.
4239 If a signal rises above this level the gain reduction is released.
4240 Default is 0.125. Allowed range is from 0 to 1.
4243 Set a ratio about which the signal is reduced.
4244 Default is 2. Allowed range is from 1 to 9000.
4247 Amount of milliseconds the signal has to rise above the threshold before gain
4249 Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
4252 Amount of milliseconds the signal has to fall below the threshold before the
4253 reduction is increased again. Default is 250 milliseconds.
4254 Allowed range is from 0.01 to 9000.
4257 Set amount of amplification of signal after processing.
4258 Default is 1. Allowed range is from 1 to 64.
4261 Curve the sharp knee around the threshold to enter gain reduction more softly.
4262 Default is 2.828427125. Allowed range is from 1 to 8.
4265 Choose if exact signal should be taken for detection or an RMS like one.
4266 Default is rms. Can be peak or rms.
4269 Choose if the average level between all channels or the louder channel affects
4271 Default is average. Can be average or maximum.
4274 Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
4277 @section silencedetect
4279 Detect silence in an audio stream.
4281 This filter logs a message when it detects that the input audio volume is less
4282 or equal to a noise tolerance value for a duration greater or equal to the
4283 minimum detected noise duration.
4285 The printed times and duration are expressed in seconds.
4287 The filter accepts the following options:
4291 Set noise tolerance. Can be specified in dB (in case "dB" is appended to the
4292 specified value) or amplitude ratio. Default is -60dB, or 0.001.
4295 Set silence duration until notification (default is 2 seconds).
4298 Process each channel separately, instead of combined. By default is disabled.
4301 @subsection Examples
4305 Detect 5 seconds of silence with -50dB noise tolerance:
4307 silencedetect=n=-50dB:d=5
4311 Complete example with @command{ffmpeg} to detect silence with 0.0001 noise
4312 tolerance in @file{silence.mp3}:
4314 ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
4318 @section silenceremove
4320 Remove silence from the beginning, middle or end of the audio.
4322 The filter accepts the following options:
4326 This value is used to indicate if audio should be trimmed at beginning of
4327 the audio. A value of zero indicates no silence should be trimmed from the
4328 beginning. When specifying a non-zero value, it trims audio up until it
4329 finds non-silence. Normally, when trimming silence from beginning of audio
4330 the @var{start_periods} will be @code{1} but it can be increased to higher
4331 values to trim all audio up to specific count of non-silence periods.
4332 Default value is @code{0}.
4334 @item start_duration
4335 Specify the amount of time that non-silence must be detected before it stops
4336 trimming audio. By increasing the duration, bursts of noises can be treated
4337 as silence and trimmed off. Default value is @code{0}.
4339 @item start_threshold
4340 This indicates what sample value should be treated as silence. For digital
4341 audio, a value of @code{0} may be fine but for audio recorded from analog,
4342 you may wish to increase the value to account for background noise.
4343 Can be specified in dB (in case "dB" is appended to the specified value)
4344 or amplitude ratio. Default value is @code{0}.
4347 Specify max duration of silence at beginning that will be kept after
4348 trimming. Default is 0, which is equal to trimming all samples detected
4352 Specify mode of detection of silence end in start of multi-channel audio.
4353 Can be @var{any} or @var{all}. Default is @var{any}.
4354 With @var{any}, any sample that is detected as non-silence will cause
4355 stopped trimming of silence.
4356 With @var{all}, only if all channels are detected as non-silence will cause
4357 stopped trimming of silence.
4360 Set the count for trimming silence from the end of audio.
4361 To remove silence from the middle of a file, specify a @var{stop_periods}
4362 that is negative. This value is then treated as a positive value and is
4363 used to indicate the effect should restart processing as specified by
4364 @var{start_periods}, making it suitable for removing periods of silence
4365 in the middle of the audio.
4366 Default value is @code{0}.
4369 Specify a duration of silence that must exist before audio is not copied any
4370 more. By specifying a higher duration, silence that is wanted can be left in
4372 Default value is @code{0}.
4374 @item stop_threshold
4375 This is the same as @option{start_threshold} but for trimming silence from
4377 Can be specified in dB (in case "dB" is appended to the specified value)
4378 or amplitude ratio. Default value is @code{0}.
4381 Specify max duration of silence at end that will be kept after
4382 trimming. Default is 0, which is equal to trimming all samples detected
4386 Specify mode of detection of silence start in end of multi-channel audio.
4387 Can be @var{any} or @var{all}. Default is @var{any}.
4388 With @var{any}, any sample that is detected as non-silence will cause
4389 stopped trimming of silence.
4390 With @var{all}, only if all channels are detected as non-silence will cause
4391 stopped trimming of silence.
4394 Set how is silence detected. Can be @code{rms} or @code{peak}. Second is faster
4395 and works better with digital silence which is exactly 0.
4396 Default value is @code{rms}.
4399 Set duration in number of seconds used to calculate size of window in number
4400 of samples for detecting silence.
4401 Default value is @code{0.02}. Allowed range is from @code{0} to @code{10}.
4404 @subsection Examples
4408 The following example shows how this filter can be used to start a recording
4409 that does not contain the delay at the start which usually occurs between
4410 pressing the record button and the start of the performance:
4412 silenceremove=start_periods=1:start_duration=5:start_threshold=0.02
4416 Trim all silence encountered from beginning to end where there is more than 1
4417 second of silence in audio:
4419 silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-90dB
4425 SOFAlizer uses head-related transfer functions (HRTFs) to create virtual
4426 loudspeakers around the user for binaural listening via headphones (audio
4427 formats up to 9 channels supported).
4428 The HRTFs are stored in SOFA files (see @url{http://www.sofacoustics.org/} for a database).
4429 SOFAlizer is developed at the Acoustics Research Institute (ARI) of the
4430 Austrian Academy of Sciences.
4432 To enable compilation of this filter you need to configure FFmpeg with
4433 @code{--enable-libmysofa}.
4435 The filter accepts the following options:
4439 Set the SOFA file used for rendering.
4442 Set gain applied to audio. Value is in dB. Default is 0.
4445 Set rotation of virtual loudspeakers in deg. Default is 0.
4448 Set elevation of virtual speakers in deg. Default is 0.
4451 Set distance in meters between loudspeakers and the listener with near-field
4452 HRTFs. Default is 1.
4455 Set processing type. Can be @var{time} or @var{freq}. @var{time} is
4456 processing audio in time domain which is slow.
4457 @var{freq} is processing audio in frequency domain which is fast.
4458 Default is @var{freq}.
4461 Set custom positions of virtual loudspeakers. Syntax for this option is:
4462 <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...].
4463 Each virtual loudspeaker is described with short channel name following with
4464 azimuth and elevation in degrees.
4465 Each virtual loudspeaker description is separated by '|'.
4466 For example to override front left and front right channel positions use:
4467 'speakers=FL 45 15|FR 345 15'.
4468 Descriptions with unrecognised channel names are ignored.
4471 Set custom gain for LFE channels. Value is in dB. Default is 0.
4474 @subsection Examples
4478 Using ClubFritz6 sofa file:
4480 sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
4484 Using ClubFritz12 sofa file and bigger radius with small rotation:
4486 sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
4490 Similar as above but with custom speaker positions for front left, front right, back left and back right
4491 and also with custom gain:
4493 "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"
4497 @section stereotools
4499 This filter has some handy utilities to manage stereo signals, for converting
4500 M/S stereo recordings to L/R signal while having control over the parameters
4501 or spreading the stereo image of master track.
4503 The filter accepts the following options:
4507 Set input level before filtering for both channels. Defaults is 1.
4508 Allowed range is from 0.015625 to 64.
4511 Set output level after filtering for both channels. Defaults is 1.
4512 Allowed range is from 0.015625 to 64.
4515 Set input balance between both channels. Default is 0.
4516 Allowed range is from -1 to 1.
4519 Set output balance between both channels. Default is 0.
4520 Allowed range is from -1 to 1.
4523 Enable softclipping. Results in analog distortion instead of harsh digital 0dB
4524 clipping. Disabled by default.
4527 Mute the left channel. Disabled by default.
4530 Mute the right channel. Disabled by default.
4533 Change the phase of the left channel. Disabled by default.
4536 Change the phase of the right channel. Disabled by default.
4539 Set stereo mode. Available values are:
4543 Left/Right to Left/Right, this is default.
4546 Left/Right to Mid/Side.
4549 Mid/Side to Left/Right.
4552 Left/Right to Left/Left.
4555 Left/Right to Right/Right.
4558 Left/Right to Left + Right.
4561 Left/Right to Right/Left.
4564 Mid/Side to Left/Left.
4567 Mid/Side to Right/Right.
4571 Set level of side signal. Default is 1.
4572 Allowed range is from 0.015625 to 64.
4575 Set balance of side signal. Default is 0.
4576 Allowed range is from -1 to 1.
4579 Set level of the middle signal. Default is 1.
4580 Allowed range is from 0.015625 to 64.
4583 Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
4586 Set stereo base between mono and inversed channels. Default is 0.
4587 Allowed range is from -1 to 1.
4590 Set delay in milliseconds how much to delay left from right channel and
4591 vice versa. Default is 0. Allowed range is from -20 to 20.
4594 Set S/C level. Default is 1. Allowed range is from 1 to 100.
4597 Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.
4599 @item bmode_in, bmode_out
4600 Set balance mode for balance_in/balance_out option.
4602 Can be one of the following:
4606 Classic balance mode. Attenuate one channel at time.
4607 Gain is raised up to 1.
4610 Similar as classic mode above but gain is raised up to 2.
4613 Equal power distribution, from -6dB to +6dB range.
4617 @subsection Examples
4621 Apply karaoke like effect:
4623 stereotools=mlev=0.015625
4627 Convert M/S signal to L/R:
4629 "stereotools=mode=ms>lr"
4633 @section stereowiden
4635 This filter enhance the stereo effect by suppressing signal common to both
4636 channels and by delaying the signal of left into right and vice versa,
4637 thereby widening the stereo effect.
4639 The filter accepts the following options:
4643 Time in milliseconds of the delay of left signal into right and vice versa.
4644 Default is 20 milliseconds.
4647 Amount of gain in delayed signal into right and vice versa. Gives a delay
4648 effect of left signal in right output and vice versa which gives widening
4649 effect. Default is 0.3.
4652 Cross feed of left into right with inverted phase. This helps in suppressing
4653 the mono. If the value is 1 it will cancel all the signal common to both
4654 channels. Default is 0.3.
4657 Set level of input signal of original channel. Default is 0.8.
4660 @section superequalizer
4661 Apply 18 band equalizer.
4663 The filter accepts the following options:
4670 Set 131Hz band gain.
4672 Set 185Hz band gain.
4674 Set 262Hz band gain.
4676 Set 370Hz band gain.
4678 Set 523Hz band gain.
4680 Set 740Hz band gain.
4682 Set 1047Hz band gain.
4684 Set 1480Hz band gain.
4686 Set 2093Hz band gain.
4688 Set 2960Hz band gain.
4690 Set 4186Hz band gain.
4692 Set 5920Hz band gain.
4694 Set 8372Hz band gain.
4696 Set 11840Hz band gain.
4698 Set 16744Hz band gain.
4700 Set 20000Hz band gain.
4704 Apply audio surround upmix filter.
4706 This filter allows to produce multichannel output from audio stream.
4708 The filter accepts the following options:
4712 Set output channel layout. By default, this is @var{5.1}.
4714 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
4715 for the required syntax.
4718 Set input channel layout. By default, this is @var{stereo}.
4720 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
4721 for the required syntax.
4724 Set input volume level. By default, this is @var{1}.
4727 Set output volume level. By default, this is @var{1}.
4730 Enable LFE channel output if output channel layout has it. By default, this is enabled.
4733 Set LFE low cut off frequency. By default, this is @var{128} Hz.
4736 Set LFE high cut off frequency. By default, this is @var{256} Hz.
4739 Set front center input volume. By default, this is @var{1}.
4742 Set front center output volume. By default, this is @var{1}.
4745 Set LFE input volume. By default, this is @var{1}.
4748 Set LFE output volume. By default, this is @var{1}.
4751 @section treble, highshelf
4753 Boost or cut treble (upper) frequencies of the audio using a two-pole
4754 shelving filter with a response similar to that of a standard
4755 hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
4757 The filter accepts the following options:
4761 Give the gain at whichever is the lower of ~22 kHz and the
4762 Nyquist frequency. Its useful range is about -20 (for a large cut)
4763 to +20 (for a large boost). Beware of clipping when using a positive gain.
4766 Set the filter's central frequency and so can be used
4767 to extend or reduce the frequency range to be boosted or cut.
4768 The default value is @code{3000} Hz.
4771 Set method to specify band-width of filter.
4786 Determine how steep is the filter's shelf transition.
4789 Specify which channels to filter, by default all available are filtered.
4792 @subsection Commands
4794 This filter supports the following commands:
4797 Change treble frequency.
4798 Syntax for the command is : "@var{frequency}"
4801 Change treble width_type.
4802 Syntax for the command is : "@var{width_type}"
4805 Change treble width.
4806 Syntax for the command is : "@var{width}"
4810 Syntax for the command is : "@var{gain}"
4815 Sinusoidal amplitude modulation.
4817 The filter accepts the following options:
4821 Modulation frequency in Hertz. Modulation frequencies in the subharmonic range
4822 (20 Hz or lower) will result in a tremolo effect.
4823 This filter may also be used as a ring modulator by specifying
4824 a modulation frequency higher than 20 Hz.
4825 Range is 0.1 - 20000.0. Default value is 5.0 Hz.
4828 Depth of modulation as a percentage. Range is 0.0 - 1.0.
4829 Default value is 0.5.
4834 Sinusoidal phase modulation.
4836 The filter accepts the following options:
4840 Modulation frequency in Hertz.
4841 Range is 0.1 - 20000.0. Default value is 5.0 Hz.
4844 Depth of modulation as a percentage. Range is 0.0 - 1.0.
4845 Default value is 0.5.
4850 Adjust the input audio volume.
4852 It accepts the following parameters:
4856 Set audio volume expression.
4858 Output values are clipped to the maximum value.
4860 The output audio volume is given by the relation:
4862 @var{output_volume} = @var{volume} * @var{input_volume}
4865 The default value for @var{volume} is "1.0".
4868 This parameter represents the mathematical precision.
4870 It determines which input sample formats will be allowed, which affects the
4871 precision of the volume scaling.
4875 8-bit fixed-point; this limits input sample format to U8, S16, and S32.
4877 32-bit floating-point; this limits input sample format to FLT. (default)
4879 64-bit floating-point; this limits input sample format to DBL.
4883 Choose the behaviour on encountering ReplayGain side data in input frames.
4887 Remove ReplayGain side data, ignoring its contents (the default).
4890 Ignore ReplayGain side data, but leave it in the frame.
4893 Prefer the track gain, if present.
4896 Prefer the album gain, if present.
4899 @item replaygain_preamp
4900 Pre-amplification gain in dB to apply to the selected replaygain gain.
4902 Default value for @var{replaygain_preamp} is 0.0.
4905 Set when the volume expression is evaluated.
4907 It accepts the following values:
4910 only evaluate expression once during the filter initialization, or
4911 when the @samp{volume} command is sent
4914 evaluate expression for each incoming frame
4917 Default value is @samp{once}.
4920 The volume expression can contain the following parameters.
4924 frame number (starting at zero)
4927 @item nb_consumed_samples
4928 number of samples consumed by the filter
4930 number of samples in the current frame
4932 original frame position in the file
4938 PTS at start of stream
4940 time at start of stream
4946 last set volume value
4949 Note that when @option{eval} is set to @samp{once} only the
4950 @var{sample_rate} and @var{tb} variables are available, all other
4951 variables will evaluate to NAN.
4953 @subsection Commands
4955 This filter supports the following commands:
4958 Modify the volume expression.
4959 The command accepts the same syntax of the corresponding option.
4961 If the specified expression is not valid, it is kept at its current
4963 @item replaygain_noclip
4964 Prevent clipping by limiting the gain applied.
4966 Default value for @var{replaygain_noclip} is 1.
4970 @subsection Examples
4974 Halve the input audio volume:
4978 volume=volume=-6.0206dB
4981 In all the above example the named key for @option{volume} can be
4982 omitted, for example like in:
4988 Increase input audio power by 6 decibels using fixed-point precision:
4990 volume=volume=6dB:precision=fixed
4994 Fade volume after time 10 with an annihilation period of 5 seconds:
4996 volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame
5000 @section volumedetect
5002 Detect the volume of the input video.
5004 The filter has no parameters. The input is not modified. Statistics about
5005 the volume will be printed in the log when the input stream end is reached.
5007 In particular it will show the mean volume (root mean square), maximum
5008 volume (on a per-sample basis), and the beginning of a histogram of the
5009 registered volume values (from the maximum value to a cumulated 1/1000 of
5012 All volumes are in decibels relative to the maximum PCM value.
5014 @subsection Examples
5016 Here is an excerpt of the output:
5018 [Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB
5019 [Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB
5020 [Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6
5021 [Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62
5022 [Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286
5023 [Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042
5024 [Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551
5025 [Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609
5026 [Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409
5032 The mean square energy is approximately -27 dB, or 10^-2.7.
5034 The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
5036 There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
5039 In other words, raising the volume by +4 dB does not cause any clipping,
5040 raising it by +5 dB causes clipping for 6 samples, etc.
5042 @c man end AUDIO FILTERS
5044 @chapter Audio Sources
5045 @c man begin AUDIO SOURCES
5047 Below is a description of the currently available audio sources.
5051 Buffer audio frames, and make them available to the filter chain.
5053 This source is mainly intended for a programmatic use, in particular
5054 through the interface defined in @file{libavfilter/asrc_abuffer.h}.
5056 It accepts the following parameters:
5060 The timebase which will be used for timestamps of submitted frames. It must be
5061 either a floating-point number or in @var{numerator}/@var{denominator} form.
5064 The sample rate of the incoming audio buffers.
5067 The sample format of the incoming audio buffers.
5068 Either a sample format name or its corresponding integer representation from
5069 the enum AVSampleFormat in @file{libavutil/samplefmt.h}
5071 @item channel_layout
5072 The channel layout of the incoming audio buffers.
5073 Either a channel layout name from channel_layout_map in
5074 @file{libavutil/channel_layout.c} or its corresponding integer representation
5075 from the AV_CH_LAYOUT_* macros in @file{libavutil/channel_layout.h}
5078 The number of channels of the incoming audio buffers.
5079 If both @var{channels} and @var{channel_layout} are specified, then they
5084 @subsection Examples
5087 abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo
5090 will instruct the source to accept planar 16bit signed stereo at 44100Hz.
5091 Since the sample format with name "s16p" corresponds to the number
5092 6 and the "stereo" channel layout corresponds to the value 0x3, this is
5095 abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3
5100 Generate an audio signal specified by an expression.
5102 This source accepts in input one or more expressions (one for each
5103 channel), which are evaluated and used to generate a corresponding