ffmpeg-filters - Man Page

A filtergraph has a textual representation, which is recognized by the -filter/-vf/-af and -filter_complex options in ffmpeg and -vf/-af in ffplay, and by the avfilter_graph_parse_ptr() function defined in libavfilter/avfilter.h.

A filterchain consists of a sequence of connected filters, each one connected to the previous one in the sequence. A filterchain is represented by a list of ","-separated filter descriptions.

A filtergraph consists of a sequence of filterchains. A sequence of filterchains is represented by a list of ";"-separated filterchain descriptions.

A filter is represented by a string of the form: [in_link_1]...[in_link_N]filter_name@id=arguments[out_link_1]...[out_link_M]

filter_name is the name of the filter class of which the described filter is an instance of, and has to be the name of one of the filter classes registered in the program optionally followed by "@id". The name of the filter class is optionally followed by a string "=arguments".

arguments is a string which contains the parameters used to initialize the filter instance. It may have one of two forms:

• A ':'-separated list of key=value pairs.
• A ':'-separated list of value. In this case, the keys are assumed to be the option names in the order they are declared. E.g. the fade filter declares three options in this order -- type, start_frame and nb_frames. Then the parameter list in:0:30 means that the value in is assigned to the option type, 0 to start_frame and 30 to nb_frames.
• A ':'-separated list of mixed direct value and long key=value pairs. The direct value must precede the key=value pairs, and follow the same constraints order of the previous point. The following key=value pairs can be set in any preferred order.

If the option value itself is a list of items (e.g. the format filter takes a list of pixel formats), the items in the list are usually separated by |.

The list of arguments can be quoted using the character ' as initial and ending mark, and the character \ for escaping the characters within the quoted text; otherwise the argument string is considered terminated when the next special character (belonging to the set []=;,) is encountered.

A special syntax implemented in the ffmpeg CLI tool allows loading option values from files. This is done be prepending a slash '/' to the option name, then the supplied value is interpreted as a path from which the actual value is loaded. E.g.

        ffmpeg -i <INPUT> -vf drawtext=/text=/tmp/some_text <OUTPUT>

will load the text to be drawn from /tmp/some_text. API users wishing to implement a similar feature should use the avfilter_graph_segment_*() functions together with custom IO code.

The name and arguments of the filter are optionally preceded and followed by a list of link labels. A link label allows one to name a link and associate it to a filter output or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... out_link_M, are associated to the output pads.

When two link labels with the same name are found in the filtergraph, a link between the corresponding input and output pad is created.

If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next filter in the filterchain. For example in the filterchain

        nullsrc, split[L1], [L2]overlay, nullsink

the split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled.

In a filter description, if the input label of the first filter is not specified, "in" is assumed; if the output label of the last filter is not specified, "out" is assumed.

In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected.

Leading and trailing whitespaces (space, tabs, or line feeds) separating tokens in the filtergraph specification are ignored. This means that the filtergraph can be expressed using empty lines and spaces to improve redability.

For example, the filtergraph:

        testsrc,split[L1],hflip[L2];[L1][L2] hstack

can be represented as:

        testsrc,
        split [L1], hflip [L2];
        
        [L1][L2] hstack

Libavfilter will automatically insert scale filters where format conversion is required. It is possible to specify swscale flags for those automatically inserted scalers by prepending sws_flags=flags; to the filtergraph description.

Here is a BNF description of the filtergraph syntax:

        <NAME>             ::= sequence of alphanumeric characters and '_'
        <FILTER_NAME>      ::= <NAME>["@"<NAME>]
        <LINKLABEL>        ::= "[" <NAME> "]"
        <LINKLABELS>       ::= <LINKLABEL> [<LINKLABELS>]
        <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
        <FILTER>           ::= [<LINKLABELS>] <FILTER_NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
        <FILTERCHAIN>      ::= <FILTER> [,<FILTERCHAIN>]
        <FILTERGRAPH>      ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

Filtergraph description composition entails several levels of escaping. See the "Quoting and escaping" section in the ffmpeg-utils(1) manual for more information about the employed escaping procedure.

A first level escaping affects the content of each filter option value, which may contain the special character : used to separate values, or one of the escaping characters \'.

A second level escaping affects the whole filter description, which may contain the escaping characters \' or the special characters [],; used by the filtergraph description.

Finally, when you specify a filtergraph on a shell commandline, you need to perform a third level escaping for the shell special characters contained within it.

For example, consider the following string to be embedded in the drawtext filter description text value:

        this is a 'string': may contain one, or more, special characters

This string contains the ' special escaping character, and the : special character, so it needs to be escaped in this way:

        text=this is a \'string\'\: may contain one, or more, special characters

A second level of escaping is required when embedding the filter description in a filtergraph description, in order to escape all the filtergraph special characters. Thus the example above becomes:

        drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

(note that in addition to the \' escaping special characters, also , needs to be escaped).

Finally an additional level of escaping is needed when writing the filtergraph description in a shell command, which depends on the escaping rules of the adopted shell. For example, assuming that \ is special and needs to be escaped with another \, the previous string will finally result in:

        -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

In order to avoid cumbersome escaping when using a commandline tool accepting a filter specification as input, it is advisable to avoid direct inclusion of the filter or options specification in the shell.

For example, in case of the drawtext filter, you might prefer to use the textfile option in place of text to specify the text to render.

Apply Affine Projection algorithm to the first audio stream using the second audio stream.

This adaptive filter is used to estimate unknown audio based on multiple input audio samples. Affine projection algorithm can make trade-offs between computation complexity with convergence speed.

A description of the accepted options follows.

order

Set the filter order.

projection

Set the projection order.

mu

Set the filter mu.

delta

Set the coefficient to initialize internal covariance matrix.

out_mode

Set the filter output samples. It accepts the following values:

i

Pass the 1st input.

d

Pass the 2nd input.

o

Pass difference between desired, 2nd input and error signal estimate.

n

Pass difference between input, 1st input and error signal estimate.

e

Pass error signal estimated samples.

Default value is o.

precision

Set which precision to use when processing samples.

auto

Auto pick internal sample format depending on other filters.

float

Always use single-floating point precision sample format.

double

Always use double-floating point precision sample format.

A compressor is mainly used to reduce the dynamic range of a signal. Especially modern music is mostly compressed at a high ratio to improve the overall loudness. It's done to get the highest attention of a listener, "fatten" the sound and bring more "power" to the track. If a signal is compressed too much it may sound dull or "dead" afterwards or it may start to "pump" (which could be a powerful effect but can also destroy a track completely). The right compression is the key to reach a professional sound and is the high art of mixing and mastering. Because of its complex settings it may take a long time to get the right feeling for this kind of effect.

Compression is done by detecting the volume above a chosen level threshold and dividing it by the factor set with ratio. So if you set the threshold to -12dB and your signal reaches -6dB a ratio of 2:1 will result in a signal at -9dB. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over the time. This is done by setting "Attack" and "Release". attack determines how long the signal has to rise above the threshold before any reduction will occur and release sets the time the signal has to fall below the threshold to reduce the reduction again. Shorter signals than the chosen attack time will be left untouched. The overall reduction of the signal can be made up afterwards with the makeup setting. So compressing the peaks of a signal about 6dB and raising the makeup to this level results in a signal twice as loud than the source. To gain a softer entry in the compression the knee flattens the hard edge at the threshold in the range of the chosen decibels.

The filter accepts the following options:

level_in

Set input gain. Default is 1. Range is between 0.015625 and 64.

mode

Set mode of compressor operation. Can be upward or downward. Default is downward.

threshold

If a signal of stream rises above this level it will affect the gain reduction. By default it is 0.125. Range is between 0.00097563 and 1.

ratio

Set a ratio by which the signal is reduced. 1:2 means that if the level rose 4dB above the threshold, it will be only 2dB above after the reduction. Default is 2. Range is between 1 and 20.

attack

Amount of milliseconds the signal has to rise above the threshold before gain reduction starts. Default is 20. Range is between 0.01 and 2000.

release

Amount of milliseconds the signal has to fall below the threshold before reduction is decreased again. Default is 250. Range is between 0.01 and 9000.

makeup

Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1 to 64.

knee

Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.

link

Choose if the average level between all channels of input stream or the louder(maximum) channel of input stream affects the reduction. Default is average.

detection

Should the exact signal be taken in case of peak or an RMS one in case of rms. Default is rms which is mostly smoother.

mix

How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Commands

This filter supports the all above options as commands.

Simple audio dynamic range compression/expansion filter.

The filter accepts the following options:

contrast

Set contrast. Default is 33. Allowed range is between 0 and 100.

Copy the input audio source unchanged to the output. This is mainly useful for testing purposes.

Apply cross fade from one input audio stream to another input audio stream. The cross fade is applied for specified duration near the end of first stream.

The filter accepts the following options:

nb_samples, ns

Specify the number of samples for which the cross fade effect has to last. At the end of the cross fade effect the first input audio will be completely silent. Default is 44100.

duration, d

Specify the duration of the cross fade effect. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. By default the duration is determined by nb_samples. If set this option is used instead of nb_samples.

overlap, o

Should first stream end overlap with second stream start. Default is enabled.

curve1

Set curve for cross fade transition for first stream.

curve2

Set curve for cross fade transition for second stream.

For description of available curve types see afade filter description.

Examples

• Cross fade from one input to another:

          ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac

• Cross fade from one input to another but without overlapping:

          ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac

Split audio stream into several bands.

This filter splits audio stream into two or more frequency ranges. Summing all streams back will give flat output.

The filter accepts the following options:

split

Set split frequencies. Those must be positive and increasing.

order

Set filter order for each band split. This controls filter roll-off or steepness of filter transfer function. Available values are:

2nd

12 dB per octave.

4th

24 dB per octave.

6th

36 dB per octave.

8th

48 dB per octave.

10th

60 dB per octave.

12th

72 dB per octave.

14th

84 dB per octave.

16th

96 dB per octave.

18th

108 dB per octave.

20th

120 dB per octave.

Default is 4th.

level

Set input gain level. Allowed range is from 0 to 1. Default value is 1.

gains

Set output gain for each band. Default value is 1 for all bands.

precision

Set which precision to use when processing samples.

auto

Auto pick internal sample format depending on other filters.

float

Always use single-floating point precision sample format.

double

Always use double-floating point precision sample format.

Default value is auto.

Examples

• Split input audio stream into two bands (low and high) with split frequency of 1500 Hz, each band will be in separate stream:

          ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

• Same as above, but with higher filter order:

          ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

• Same as above, but also with additional middle band (frequencies between 1500 and 8000):

          ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav

Reduce audio bit resolution.

This filter is bit crusher with enhanced functionality. A bit crusher is used to audibly reduce number of bits an audio signal is sampled with. This doesn't change the bit depth at all, it just produces the effect. Material reduced in bit depth sounds more harsh and "digital". This filter is able to even round to continuous values instead of discrete bit depths. Additionally it has a D/C offset which results in different crushing of the lower and the upper half of the signal. An Anti-Aliasing setting is able to produce "softer" crushing sounds.

Another feature of this filter is the logarithmic mode. This setting switches from linear distances between bits to logarithmic ones. The result is a much more "natural" sounding crusher which doesn't gate low signals for example. The human ear has a logarithmic perception, so this kind of crushing is much more pleasant. Logarithmic crushing is also able to get anti-aliased.

The filter accepts the following options:

level_in

Set level in.

level_out

Set level out.

bits

Set bit reduction.

mix

Set mixing amount.

mode

Can be linear: lin or logarithmic: log.

dc

Set DC.

aa

Set anti-aliasing.

samples

Set sample reduction.

lfo

Enable LFO. By default disabled.

lforange

Set LFO range.

lforate

Set LFO rate.

Commands

This filter supports the all above options as commands.

Delay audio filtering until a given wallclock timestamp. See the cue filter.

Remove impulsive noise from input audio.

Samples detected as impulsive noise are replaced by interpolated samples using autoregressive modelling.

window, w

Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.

overlap, o

Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent. Setting this to a very high value increases impulsive noise removal but makes whole process much slower.

arorder, a

Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 2 percent. This option also controls quality of interpolated samples using neighbour good samples.

threshold, t

Set threshold value. Allowed range is from 1 to 100. Default value is 2. This controls the strength of impulsive noise which is going to be removed. The lower value, the more samples will be detected as impulsive noise.

burst, b

Set burst fusion, in percentage of window size. Allowed range is 0 to 10. Default value is 2. If any two samples detected as noise are spaced less than this value then any sample between those two samples will be also detected as noise.

method, m

Set overlap method.

It accepts the following values:

add, a

Select overlap-add method. Even not interpolated samples are slightly changed with this method.

save, s

Select overlap-save method. Not interpolated samples remain unchanged.

Default value is a.

Remove clipped samples from input audio.

Samples detected as clipped are replaced by interpolated samples using autoregressive modelling.

window, w

Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.

overlap, o

Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent.

arorder, a

Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 8 percent. This option also controls quality of interpolated samples using neighbour good samples.

threshold, t

Set threshold value. Allowed range is from 1 to 100. Default value is 10. Higher values make clip detection less aggressive.

hsize, n

Set size of histogram used to detect clips. Allowed range is from 100 to 9999. Default value is 1000. Higher values make clip detection less aggressive.

method, m

Set overlap method.

It accepts the following values:

add, a

Select overlap-add method. Even not interpolated samples are slightly changed with this method.

save, s

Select overlap-save method. Not interpolated samples remain unchanged.

Default value is a.

Apply decorrelation to input audio stream.

The filter accepts the following options:

stages

Set decorrelation stages of filtering. Allowed range is from 1 to 16. Default value is 6.

seed

Set random seed used for setting delay in samples across channels.

Delay one or more audio channels.

Samples in delayed channel are filled with silence.

The filter accepts the following option:

delays

Set list of delays in milliseconds for each channel separated by '|'. Unused delays will be silently ignored. If number of given delays is smaller than number of channels all remaining channels will not be delayed. If you want to delay exact number of samples, append 'S' to number. If you want instead to delay in seconds, append 's' to number.

all

Use last set delay for all remaining channels. By default is disabled. This option if enabled changes how option delays is interpreted.

Examples

• Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave the second channel (and any other channels that may be present) unchanged.

          adelay=1500|0|500

• Delay second channel by 500 samples, the third channel by 700 samples and leave the first channel (and any other channels that may be present) unchanged.

          adelay=0|500S|700S

• Delay all channels by same number of samples:

          adelay=delays=64S:all=1

Remedy denormals in audio by adding extremely low-level noise.

This filter shall be placed before any filter that can produce denormals.

A description of the accepted parameters follows.

level

Set level of added noise in dB. Default is -351. Allowed range is from -451 to -90.

type

Set type of added noise.

dc

Add DC signal.

ac

Add AC signal.

square

Add square signal.

pulse

Add pulse signal.

Default is dc.

Commands

This filter supports the all above options as commands.

Compute derivative/integral of audio stream.

Applying both filters one after another produces original audio.

Apply spectral dynamic range controller filter to input audio stream.

A description of the accepted options follows.

transfer

Set the transfer expression.

The expression can contain the following constants:

ch

current channel number

sn

current sample number

nb_channels

number of channels

t

timestamp expressed in seconds

sr

sample rate

p

current frequency power value, in dB

f

current frequency in Hz

Default value is p.

attack

Set the attack in milliseconds. Default is 50 milliseconds. Allowed range is from 1 to 1000 milliseconds.

release

Set the release in milliseconds. Default is 100 milliseconds. Allowed range is from 5 to 2000 milliseconds.

channels

Set which channels to filter, by default all channels in audio stream are filtered.

Commands

This filter supports the all above options as commands.

Examples

• Apply spectral compression to all frequencies with threshold of -50 dB and 1:6 ratio:

          adrc=transfer='if(gt(p,-50),-50+(p-(-50))/6,p)':attack=50:release=100

• Similar to above but with 1:2 ratio and filtering only front center channel:

          adrc=transfer='if(gt(p,-50),-50+(p-(-50))/2,p)':attack=50:release=100:channels=FC

• Apply spectral noise gate to all frequencies with threshold of -85 dB and with short attack time and short release time:

          adrc=transfer='if(lte(p,-85),p-800,p)':attack=1:release=5

• Apply spectral expansion to all frequencies with threshold of -10 dB and 1:2 ratio:

          adrc=transfer='if(lt(p,-10),-10+(p-(-10))*2,p)':attack=50:release=100

• Apply limiter to max -60 dB to all frequencies, with attack of 2 ms and release of 10 ms:

          adrc=transfer='min(p,-60)':attack=2:release=10

Apply dynamic equalization to input audio stream.

A description of the accepted options follows.

threshold

Set the detection threshold used to trigger equalization. Threshold detection is using detection filter. Default value is 0. Allowed range is from 0 to 100.

dfrequency

Set the detection frequency in Hz used for detection filter used to trigger equalization. Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.

dqfactor

Set the detection resonance factor for detection filter used to trigger equalization. Default value is 1. Allowed range is from 0.001 to 1000.

tfrequency

Set the target frequency of equalization filter. Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.

tqfactor

Set the target resonance factor for target equalization filter. Default value is 1. Allowed range is from 0.001 to 1000.

attack

Set the amount of milliseconds the signal from detection has to rise above the detection threshold before equalization starts. Default is 20. Allowed range is between 1 and 2000.

release

Set the amount of milliseconds the signal from detection has to fall below the detection threshold before equalization ends. Default is 200. Allowed range is between 1 and 2000.

ratio

Set the ratio by which the equalization gain is raised. Default is 1. Allowed range is between 0 and 30.

makeup

Set the makeup offset by which the equalization gain is raised. Default is 0. Allowed range is between 0 and 100.

range

Set the max allowed cut/boost amount. Default is 50. Allowed range is from 1 to 200.

mode

Set the mode of filter operation, can be one of the following:

listen

Output only isolated detection signal.

cutbelow

Cut frequencies below detection threshold.

cutabove

Cut frequencies above detection threshold.

boostbelow

Boost frequencies below detection threshold.

boostabove

Boost frequencies above detection threshold.

Default mode is cutbelow.

dftype

Set the type of detection filter, can be one of the following:

bandpass

lowpass

highpass

peak
Default type is bandpass.

tftype

Set the type of target filter, can be one of the following:

bell

lowshelf

highshelf
Default type is bell.

auto

Automatically gather threshold from detection filter. By default is disabled. This option is useful to detect threshold in certain time frame of input audio stream, in such case option value is changed at runtime.

Available values are:

disabled

Disable using automatically gathered threshold value.

off

Stop picking threshold value.

on

Start picking threshold value.

adaptive

Adaptively pick threshold value, by calculating sliding window entropy.

precision

Set which precision to use when processing samples.

auto

Auto pick internal sample format depending on other filters.

float

Always use single-floating point precision sample format.

double

Always use double-floating point precision sample format.

Commands

This filter supports the all above options as commands.

Apply dynamic smoothing to input audio stream.

A description of the accepted options follows.

sensitivity

Set an amount of sensitivity to frequency fluctations. Default is 2. Allowed range is from 0 to 1e+06.

basefreq

Set a base frequency for smoothing. Default value is 22050. Allowed range is from 2 to 1e+06.

Commands

This filter supports the all above options as commands.

Apply echoing to the input audio.

Echoes are reflected sound and can occur naturally amongst mountains (and sometimes large buildings) when talking or shouting; digital echo effects emulate this behaviour and are often used to help fill out the sound of a single instrument or vocal. The time difference between the original signal and the reflection is the delay, and the loudness of the reflected signal is the decay. Multiple echoes can have different delays and decays.

A description of the accepted parameters follows.

in_gain

Set input gain of reflected signal. Default is 0.6.

out_gain

Set output gain of reflected signal. Default is 0.3.

delays

Set list of time intervals in milliseconds between original signal and reflections separated by '|'. Allowed range for each delay is (0 - 90000.0]. Default is 1000.

decays

Set list of loudness of reflected signals separated by '|'. Allowed range for each decay is (0 - 1.0]. Default is 0.5.

Examples

• Make it sound as if there are twice as many instruments as are actually playing:

          aecho=0.8:0.88:60:0.4

• If delay is very short, then it sounds like a (metallic) robot playing music:

          aecho=0.8:0.88:6:0.4

• A longer delay will sound like an open air concert in the mountains:

          aecho=0.8:0.9:1000:0.3

• Same as above but with one more mountain:

          aecho=0.8:0.9:1000|1800:0.3|0.25

Audio emphasis filter creates or restores material directly taken from LPs or emphased CDs with different filter curves. E.g. to store music on vinyl the signal has to be altered by a filter first to even out the disadvantages of this recording medium. Once the material is played back the inverse filter has to be applied to restore the distortion of the frequency response.

The filter accepts the following options:

level_in

Set input gain.

level_out

Set output gain.

mode

Set filter mode. For restoring material use reproduction mode, otherwise use production mode. Default is reproduction mode.

type

Set filter type. Selects medium. Can be one of the following:

col

select Columbia.

emi

select EMI.

bsi

select BSI (78RPM).

riaa

select RIAA.

cd

select Compact Disc (CD).

50fm

select 50µs (FM).

75fm

select 75µs (FM).

50kf

select 50µs (FM-KF).

75kf

select 75µs (FM-KF).

Commands

This filter supports the all above options as commands.

Modify an audio signal according to the specified expressions.

This filter accepts one or more expressions (one for each channel), which are evaluated and used to modify a corresponding audio signal.

It accepts the following parameters:

exprs

Set the '|'-separated expressions list for each separate channel. If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.

channel_layout, c

Set output channel layout. If not specified, the channel layout is specified by the number of expressions. If set to same, it will use by default the same input channel layout.

Each expression in exprs can contain the following constants and functions:

ch

channel number of the current expression

n

number of the evaluated sample, starting from 0

s

sample rate

t

time of the evaluated sample expressed in seconds

nb_in_channels

nb_out_channels

input and output number of channels

val(CH)

the value of input channel with number CH

Note: this filter is slow. For faster processing you should use a dedicated filter.

Examples

• Half volume:

          aeval=val(ch)/2:c=same

• Invert phase of the second channel:

          aeval=val(0)|-val(1)

An exciter is used to produce high sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. An Exciter raises the upper end of an audio signal without simply raising the higher frequencies like an equalizer would do to create a more "crisp" or "brilliant" sound.

The filter accepts the following options:

level_in

Set input level prior processing of signal. Allowed range is from 0 to 64. Default value is 1.

level_out

Set output level after processing of signal. Allowed range is from 0 to 64. Default value is 1.

amount

Set the amount of harmonics added to original signal. Allowed range is from 0 to 64. Default value is 1.

drive

Set the amount of newly created harmonics. Allowed range is from 0.1 to 10. Default value is 8.5.

blend

Set the octave of newly created harmonics. Allowed range is from -10 to 10. Default value is 0.

freq

Set the lower frequency limit of producing harmonics in Hz. Allowed range is from 2000 to 12000 Hz. Default is 7500 Hz.

ceil

Set the upper frequency limit of producing harmonics. Allowed range is from 9999 to 20000 Hz. If value is lower than 10000 Hz no limit is applied.

listen

Mute the original signal and output only added harmonics. By default is disabled.

Commands

This filter supports the all above options as commands.

Apply fade-in/out effect to input audio.

A description of the accepted parameters follows.

type, t

Specify the effect type, can be either in for fade-in, or out for a fade-out effect. Default is in.

start_sample, ss

Specify the number of the start sample for starting to apply the fade effect. Default is 0.

nb_samples, ns

Specify the number of samples for which the fade effect has to last. At the end of the fade-in effect the output audio will have the same volume as the input audio, at the end of the fade-out transition the output audio will be silence. Default is 44100.

start_time, st

Specify the start time of the fade effect. Default is 0. The value must be specified as a time duration; see the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. If set this option is used instead of start_sample.

duration, d

Specify the duration of the fade effect. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. At the end of the fade-in effect the output audio will have the same volume as the input audio, at the end of the fade-out transition the output audio will be silence. By default the duration is determined by nb_samples. If set this option is used instead of nb_samples.

curve

Set curve for fade transition.

It accepts the following values:

tri

select triangular, linear slope (default)

qsin

select quarter of sine wave

hsin

select half of sine wave

esin

select exponential sine wave

log

select logarithmic

ipar

select inverted parabola

qua

select quadratic

cub

select cubic

squ

select square root

cbr

select cubic root

par

select parabola

exp

select exponential

iqsin

select inverted quarter of sine wave

ihsin

select inverted half of sine wave

dese

select double-exponential seat

desi

select double-exponential sigmoid

losi

select logistic sigmoid

sinc

select sine cardinal function

isinc

select inverted sine cardinal function

quat

select quartic

quatr

select quartic root

qsin2

select squared quarter of sine wave

hsin2

select squared half of sine wave

nofade

no fade applied

silence

Set the initial gain for fade-in or final gain for fade-out. Default value is 0.0.

unity

Set the initial gain for fade-out or final gain for fade-in. Default value is 1.0.

Commands

This filter supports the all above options as commands.

Examples

• Fade in first 15 seconds of audio:

          afade=t=in:ss=0:d=15

• Fade out last 25 seconds of a 900 seconds audio:

          afade=t=out:st=875:d=25

Denoise audio samples with FFT.

A description of the accepted parameters follows.

noise_reduction, nr

Set the noise reduction in dB, allowed range is 0.01 to 97. Default value is 12 dB.

noise_floor, nf

Set the noise floor in dB, allowed range is -80 to -20. Default value is -50 dB.

noise_type, nt

Set the noise type.

It accepts the following values:

white, w

Select white noise.

vinyl, v

Select vinyl noise.

shellac, s

Select shellac noise.

custom, c

Select custom noise, defined in bn option.

Default value is white noise.

band_noise, bn

Set custom band noise profile for every one of 15 bands. Bands are separated by ' ' or '|'.

residual_floor, rf

Set the residual floor in dB, allowed range is -80 to -20. Default value is -38 dB.

track_noise, tn

Enable noise floor tracking. By default is disabled. With this enabled, noise floor is automatically adjusted.

track_residual, tr

Enable residual tracking. By default is disabled.

output_mode, om

Set the output mode.

It accepts the following values:

input, i

Pass input unchanged.

output, o

Pass noise filtered out.

noise, n

Pass only noise.

Default value is output.

adaptivity, ad

Set the adaptivity factor, used how fast to adapt gains adjustments per each frequency bin. Value 0 enables instant adaptation, while higher values react much slower. Allowed range is from 0 to 1. Default value is 0.5.

floor_offset, fo

Set the noise floor offset factor. This option is used to adjust offset applied to measured noise floor. It is only effective when noise floor tracking is enabled. Allowed range is from -2.0 to 2.0. Default value is 1.0.

noise_link, nl

Set the noise link used for multichannel audio.

It accepts the following values:

none

Use unchanged channel's noise floor.

min

Use measured min noise floor of all channels.

max

Use measured max noise floor of all channels.

average

Use measured average noise floor of all channels.

Default value is min.

band_multiplier, bm

Set the band multiplier factor, used how much to spread bands across frequency bins. Allowed range is from 0.2 to 5. Default value is 1.25.

sample_noise, sn

Toggle capturing and measurement of noise profile from input audio.

It accepts the following values:

start, begin

Start sample noise capture.

stop, end

Stop sample noise capture and measure new noise band profile.

Default value is none.

gain_smooth, gs

Set gain smooth spatial radius, used to smooth gains applied to each frequency bin. Useful to reduce random music noise artefacts. Higher values increases smoothing of gains. Allowed range is from 0 to 50. Default value is 0.

Commands

This filter supports the some above mentioned options as commands.

Examples

• Reduce white noise by 10dB, and use previously measured noise floor of -40dB:

          afftdn=nr=10:nf=-40

• Reduce white noise by 10dB, also set initial noise floor to -80dB and enable automatic tracking of noise floor so noise floor will gradually change during processing:

          afftdn=nr=10:nf=-80:tn=1

• Reduce noise by 20dB, using noise floor of -40dB and using commands to take noise profile of first 0.4 seconds of input audio:

          asendcmd=0.0 afftdn sn start,asendcmd=0.4 afftdn sn stop,afftdn=nr=20:nf=-40

Apply arbitrary expressions to samples in frequency domain.

real

Set frequency domain real expression for each separate channel separated by '|'. Default is "re". If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.

imag

Set frequency domain imaginary expression for each separate channel separated by '|'. Default is "im".

Each expression in real and imag can contain the following constants and functions:

sr

sample rate

b

current frequency bin number

nb

number of available bins

ch

channel number of the current expression

chs

number of channels

pts

current frame pts

re

current real part of frequency bin of current channel

im

current imaginary part of frequency bin of current channel

real(b, ch)

Return the value of real part of frequency bin at location (bin,channel)

imag(b, ch)

Return the value of imaginary part of frequency bin at location (bin,channel)

win_size

Set window size. Allowed range is from 16 to 131072. Default is 4096

win_func

Set window function.

It accepts the following values:

rect

bartlett

hann, hanning

hamming

blackman

welch

flattop

bharris

bnuttall

bhann

sine

nuttall

lanczos

gauss

tukey

dolph

cauchy

parzen

poisson

bohman

kaiser
Default is hann.

overlap

Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. Default is 0.75.

Examples

• Leave almost only low frequencies in audio:

          afftfilt="'real=re * (1-clip((b/nb)*b,0,1))':imag='im * (1-clip((b/nb)*b,0,1))'"

• Apply robotize effect:

          afftfilt="real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':win_size=512:overlap=0.75"

• Apply whisper effect:

          afftfilt="real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin((random(1)*2-1)*2*3.14)':win_size=128:overlap=0.8"

• Apply phase shift:

          afftfilt="real=re*cos(1)-im*sin(1):imag=re*sin(1)+im*cos(1)"

Apply an arbitrary Finite Impulse Response filter.

This filter is designed for applying long FIR filters, up to 60 seconds long.

It can be used as component for digital crossover filters, room equalization, cross talk cancellation, wavefield synthesis, auralization, ambiophonics, ambisonics and spatialization.

This filter uses the streams higher than first one as FIR coefficients. If the non-first stream holds a single channel, it will be used for all input channels in the first stream, otherwise the number of channels in the non-first stream must be same as the number of channels in the first stream.

It accepts the following parameters:

dry

Set dry gain. This sets input gain.

wet

Set wet gain. This sets final output gain.

length

Set Impulse Response filter length. Default is 1, which means whole IR is processed.

gtype

This option is deprecated, and does nothing.

irnorm

Set norm to be applied to IR coefficients before filtering. Allowed range is from -1 to 2. IR coefficients are normalized with calculated vector norm set by this option. For negative values, no norm is calculated, and IR coefficients are not modified at all. Default is 1.

irlink

For multichannel IR if this option is set to true, all IR channels will be normalized with maximal measured gain of all IR channels coefficients as set by irnorm option. When disabled, all IR coefficients in each IR channel will be normalized independently. Default is true.

irgain

Set gain to be applied to IR coefficients before filtering. Allowed range is 0 to 1. This gain is applied after any gain applied with irnorm option.

irfmt

Set format of IR stream. Can be mono or input. Default is input.

maxir

Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds. Allowed range is 0.1 to 60 seconds.

response

This option is deprecated, and does nothing.

channel

This option is deprecated, and does nothing.

size

This option is deprecated, and does nothing.

rate

This option is deprecated, and does nothing.

minp

Set minimal partition size used for convolution. Default is 8192. Allowed range is from 1 to 65536. Lower values decreases latency at cost of higher CPU usage.

maxp

Set maximal partition size used for convolution. Default is 8192. Allowed range is from 8 to 65536. Lower values may increase CPU usage.

nbirs

Set number of input impulse responses streams which will be switchable at runtime. Allowed range is from 1 to 32. Default is 1.

ir

Set IR stream which will be used for convolution, starting from 0, should always be lower than supplied value by nbirs option. Default is 0. This option can be changed at runtime via commands.

precision

Set which precision to use when processing samples.

auto

Auto pick internal sample format depending on other filters.

float

Always use single-floating point precision sample format.

double

Always use double-floating point precision sample format.

Default value is auto.

irload

Set when to load IR stream. Can be init or access. First one load and prepares all IRs on initialization, second one once on first access of specific IR. Default is init.

Examples

• Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:

          ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav

• Apply true stereo processing given input stereo stream, and two stereo impulse responses for left and right channel, the impulse response files are files with names l_ir.wav and r_ir.wav, and setting irnorm option value:

          "pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:irnorm=1.2,pan=stereo|FL<c0+c2|FR<c1+c3"

• Similar to above example, but with irgain explicitly set to estimated value and with irnorm disabled:

          "pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:irgain=-5dB:irnom=-1,pan=stereo|FL<c0+c2|FR<c1+c3"

Set output format constraints for the input audio. The framework will negotiate the most appropriate format to minimize conversions.

It accepts the following parameters:

sample_fmts, f

A '|'-separated list of requested sample formats.

sample_rates, r

A '|'-separated list of requested sample rates.

channel_layouts, cl

A '|'-separated list of requested channel layouts.

See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

If a parameter is omitted, all values are allowed.

Force the output to either unsigned 8-bit or signed 16-bit stereo

        aformat=sample_fmts=u8|s16:channel_layouts=stereo

Apply frequency shift to input audio samples.

The filter accepts the following options:

shift

Specify frequency shift. Allowed range is -INT_MAX to INT_MAX. Default value is 0.0.

level

Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.

order

Set filter order used for filtering. Allowed range is from 1 to 16. Default value is 8.

Commands

This filter supports the all above options as commands.

Reduce broadband noise from input samples using Wavelets.

A description of the accepted options follows.

sigma

Set the noise sigma, allowed range is from 0 to 1. Default value is 0. This option controls strength of denoising applied to input samples. Most useful way to set this option is via decibels, eg. -45dB.

levels

Set the number of wavelet levels of decomposition. Allowed range is from 1 to 12. Default value is 10. Setting this too low make denoising performance very poor.

wavet

Set wavelet type for decomposition of input frame. They are sorted by number of coefficients, from lowest to highest. More coefficients means worse filtering speed, but overall better quality. Available wavelets are:

sym2

sym4

rbior68

deb10

sym10

coif5

bl3

percent

Set percent of full denoising. Allowed range is from 0 to 100 percent. Default value is 85 percent or partial denoising.

profile

If enabled, first input frame will be used as noise profile. If first frame samples contain non-noise performance will be very poor.

adaptive

If enabled, input frames are analyzed for presence of noise. If noise is detected with high possibility then input frame profile will be used for processing following frames, until new noise frame is detected.

samples

Set size of single frame in number of samples. Allowed range is from 512 to 65536. Default frame size is 8192 samples.

softness

Set softness applied inside thresholding function. Allowed range is from 0 to 10. Default softness is 1.

Commands

This filter supports the all above options as commands.

A gate is mainly used to reduce lower parts of a signal. This kind of signal processing reduces disturbing noise between useful signals.

Gating is done by detecting the volume below a chosen level threshold and dividing it by the factor set with ratio. The bottom of the noise floor is set via range. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over time. This is done by setting attack and release.

attack determines how long the signal has to fall below the threshold before any reduction will occur and release sets the time the signal has to rise above the threshold to reduce the reduction again. Shorter signals than the chosen attack time will be left untouched.

level_in

Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.

mode

Set the mode of operation. Can be upward or downward. Default is downward. If set to upward mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of downward lower parts of signal will be reduced.

range

Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1. Setting this to 0 disables reduction and then filter behaves like expander.

threshold

If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.

ratio

Set a ratio by which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.

attack

Amount of milliseconds the signal has to rise above the threshold before gain reduction stops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.

release

Amount of milliseconds the signal has to fall below the threshold before the reduction is increased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.

makeup

Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.

knee

Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.

detection

Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.

link

Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.

Commands

This filter supports the all above options as commands.

Apply an arbitrary Infinite Impulse Response filter.

It accepts the following parameters:

zeros, z

Set B/numerator/zeros/reflection coefficients.

poles, p

Set A/denominator/poles/ladder coefficients.

gains, k

Set channels gains.

dry_gain

Set input gain.

wet_gain

Set output gain.

format, f

Set coefficients format.

ll

lattice-ladder function

sf

analog transfer function

tf

digital transfer function

zp

Z-plane zeros/poles, cartesian (default)

pr

Z-plane zeros/poles, polar radians

pd

Z-plane zeros/poles, polar degrees

sp

S-plane zeros/poles

process, r

Set type of processing.

d

direct processing

s

serial processing

p

parallel processing

precision, e

Set filtering precision.

dbl

double-precision floating-point (default)

flt

single-precision floating-point

i32

32-bit integers

i16

16-bit integers

normalize, n

Normalize filter coefficients, by default is enabled. Enabling it will normalize magnitude response at DC to 0dB.

mix

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

response

Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in additional video stream. By default it is disabled.

channel

Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled.

size

Set video stream size. This option is used only when response is enabled.

Coefficients in tf and sf format are separated by spaces and are in ascending order.

Coefficients in zp format are separated by spaces and order of coefficients doesn't matter. Coefficients in zp format are complex numbers with i imaginary unit.

Different coefficients and gains can be provided for every channel, in such case use '|' to separate coefficients or gains. Last provided coefficients will be used for all remaining channels.

Examples

• Apply 2 pole elliptic notch at around 5000Hz for 48000 Hz sample rate:

          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

• Same as above but in zp format:

          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

• Apply 3-rd order analog normalized Butterworth low-pass filter, using analog transfer function format:

          aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d

The limiter prevents an input signal from rising over a desired threshold. This limiter uses lookahead technology to prevent your signal from distorting. It means that there is a small delay after the signal is processed. Keep in mind that the delay it produces is the attack time you set.

The filter accepts the following options:

level_in

Set input gain. Default is 1.

level_out

Set output gain. Default is 1.

limit

Don't let signals above this level pass the limiter. Default is 1.

attack

The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5 milliseconds.

release

Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50 milliseconds.

asc

When gain reduction is always needed ASC takes care of releasing to an average reduction level rather than reaching a reduction of 0 in the release time.

asc_level

Select how much the release time is affected by ASC, 0 means nearly no changes in release time while 1 produces higher release times.

level

Auto level output signal. Default is enabled. This normalizes audio back to 0dB if enabled.

latency

Compensate the delay introduced by using the lookahead buffer set with attack parameter. Also flush the valid audio data in the lookahead buffer when the stream hits EOF.

Depending on picked setting it is recommended to upsample input 2x or 4x times with aresample before applying this filter.

Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width width. An all-pass filter changes the audio's frequency to phase relationship without changing its frequency to amplitude relationship.

The filter accepts the following options:

frequency, f

Set frequency in Hz.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Specify the band-width of a filter in width_type units.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

order, o

Set the filter order, can be 1 or 2. Default is 2.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

Commands

This filter supports the following commands:

frequency, f

Change allpass frequency. Syntax for the command is : "frequency"

width_type, t

Change allpass width_type. Syntax for the command is : "width_type"

width, w

Change allpass width. Syntax for the command is : "width"

mix, m

Change allpass mix. Syntax for the command is : "mix"

Loop audio samples.

The filter accepts the following options:

loop

Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.

size

Set maximal number of samples. Default is 0.

start

Set first sample of loop. Default is 0.

time

Set the time of loop start in seconds. Only used if option named start is set to -1.

Merge two or more audio streams into a single multi-channel stream.

The filter accepts the following options:

inputs

Set the number of inputs. Default is 2.

If the channel layouts of the inputs are disjoint, and therefore compatible, the channel layout of the output will be set accordingly and the channels will be reordered as necessary. If the channel layouts of the inputs are not disjoint, the output will have all the channels of the first input then all the channels of the second input, in that order, and the channel layout of the output will be the default value corresponding to the total number of channels.

For example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR, then the output will be in 5.1, with the channels in the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the first input, b1 is the first channel of the second input).

On the other hand, if both input are in stereo, the output channels will be in the default order: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or may not be the expected value.

All inputs must have the same sample rate, and format.

If inputs do not have the same duration, the output will stop with the shortest.

Examples

• Merge two mono files into a stereo stream:

          amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge

• Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:

          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

Mixes multiple audio inputs into a single output.

Note that this filter only supports float samples (the amerge and pan audio filters support many formats). If the amix input has integer samples then aresample will be automatically inserted to perform the conversion to float samples.

It accepts the following parameters:

inputs

The number of inputs. If unspecified, it defaults to 2.

duration

How to determine the end-of-stream.

longest

The duration of the longest input. (default)

shortest

The duration of the shortest input.

first

The duration of the first input.

dropout_transition

The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.

weights

Specify weight of each input audio stream as a sequence of numbers separated by a space. If fewer weights are specified compared to number of inputs, the last weight is assigned to the remaining inputs. Default weight for each input is 1.

normalize

Always scale inputs instead of only doing summation of samples. Beware of heavy clipping if inputs are not normalized prior or after filtering by this filter if this option is disabled. By default is enabled.

Examples

• This will mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds:

          ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

• This will mix one vocal and one music input audio stream to a single output with the same duration as the longest input. The music will have quarter the weight as the vocals, and the inputs are not normalized:

          ffmpeg -i VOCALS -i MUSIC -filter_complex amix=inputs=2:duration=longest:dropout_transition=0:weights="1 0.25":normalize=0 OUTPUT

Commands

This filter supports the following commands:

weights

normalize

Syntax is same as option with same name.

Multiply first audio stream with second audio stream and store result in output audio stream. Multiplication is done by multiplying each sample from first stream with sample at same position from second stream.

With this element-wise multiplication one can create amplitude fades and amplitude modulations.

High-order parametric multiband equalizer for each channel.

It accepts the following parameters:

params

This option string is in format: "cchn f=cf w=w g=g t=f | ..." Each equalizer band is separated by '|'.

chn

Set channel number to which equalization will be applied. If input doesn't have that channel the entry is ignored.

f

Set central frequency for band. If input doesn't have that frequency the entry is ignored.

w

Set band width in Hertz.

g

Set band gain in dB.

t

Set filter type for band, optional, can be:

0

Butterworth, this is default.

1

Chebyshev type 1.

2

Chebyshev type 2.

curves

With this option activated frequency response of anequalizer is displayed in video stream.

size

Set video stream size. Only useful if curves option is activated.

mgain

Set max gain that will be displayed. Only useful if curves option is activated. Setting this to a reasonable value makes it possible to display gain which is derived from neighbour bands which are too close to each other and thus produce higher gain when both are activated.

fscale

Set frequency scale used to draw frequency response in video output. Can be linear or logarithmic. Default is logarithmic.

colors

Set color for each channel curve which is going to be displayed in video stream. This is list of color names separated by space or by '|'. Unrecognised or missing colors will be replaced by white color.

Examples

• Lower gain by 10 of central frequency 200Hz and width 100 Hz for first 2 channels using Chebyshev type 1 filter:

          anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1

Commands

This filter supports the following commands:

change

Alter existing filter parameters. Syntax for the commands is : "fN|f=freq|w=width|g=gain"

fN is existing filter number, starting from 0, if no such filter is available error is returned. freq set new frequency parameter. width set new width parameter in Hertz. gain set new gain parameter in dB.

Full filter invocation with asendcmd may look like this: asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...

Reduce broadband noise in audio samples using Non-Local Means algorithm.

Each sample is adjusted by looking for other samples with similar contexts. This context similarity is defined by comparing their surrounding patches of size p. Patches are searched in an area of r around the sample.

The filter accepts the following options:

strength, s

Set denoising strength. Allowed range is from 0.00001 to 10000. Default value is 0.00001.

patch, p

Set patch radius duration. Allowed range is from 1 to 100 milliseconds. Default value is 2 milliseconds.

research, r

Set research radius duration. Allowed range is from 2 to 300 milliseconds. Default value is 6 milliseconds.

output, o

Set the output mode.

It accepts the following values:

i

Pass input unchanged.

o

Pass noise filtered out.

n

Pass only noise.

Default value is o.

smooth, m

Set smooth factor. Default value is 11. Allowed range is from 1 to 1000.

Commands

This filter supports the all above options as commands.

Apply Normalized Least-Mean-(Squares|Fourth) algorithm to the first audio stream using the second audio stream.

This adaptive filter is used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean square of the error signal (difference between the desired, 2nd input audio stream and the actual signal, the 1st input audio stream).

A description of the accepted options follows.

order

Set filter order.

mu

Set filter mu.

eps

Set the filter eps.

leakage

Set the filter leakage.

out_mode

It accepts the following values:

i

Pass the 1st input.

d

Pass the 2nd input.

o

Pass difference between desired, 2nd input and error signal estimate.

n

Pass difference between input, 1st input and error signal estimate.

e

Pass error signal estimated samples.

Default value is o.

precision

Set which precision to use when processing samples.

auto

Auto pick internal sample format depending on other filters.

float

Always use single-floating point precision sample format.

double

Always use double-floating point precision sample format.

Examples

• One of many usages of this filter is noise reduction, input audio is filtered with same samples that are delayed by fixed amount, one such example for stereo audio is:

          asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o

Commands

This filter supports the same commands as options, excluding option order.

Pass the audio source unchanged to the output.

Pad the end of an audio stream with silence.

This can be used together with ffmpeg -shortest to extend audio streams to the same length as the video stream.

A description of the accepted options follows.

packet_size

Set silence packet size. Default value is 4096.

pad_len

Set the number of samples of silence to add to the end. After the value is reached, the stream is terminated. This option is mutually exclusive with whole_len.

whole_len

Set the minimum total number of samples in the output audio stream. If the value is longer than the input audio length, silence is added to the end, until the value is reached. This option is mutually exclusive with pad_len.

pad_dur

Specify the duration of samples of silence to add. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-negative value.

whole_dur

Specify the minimum total duration in the output audio stream. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-negative value. If the value is longer than the input audio length, silence is added to the end, until the value is reached. This option is mutually exclusive with pad_dur

If neither the pad_len nor the whole_len nor pad_dur nor whole_dur option is set, the filter will add silence to the end of the input stream indefinitely.

Note that for ffmpeg 4.4 and earlier a zero pad_dur or whole_dur also caused the filter to add silence indefinitely.

Examples

• Add 1024 samples of silence to the end of the input:

          apad=pad_len=1024

• Make sure the audio output will contain at least 10000 samples, pad the input with silence if required:

          apad=whole_len=10000

• Use ffmpeg to pad the audio input with silence, so that the video stream will always result the shortest and will be converted until the end in the output file when using the shortest option:

          ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT

Add a phasing effect to the input audio.

A phaser filter creates series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.

A description of the accepted parameters follows.

in_gain

Set input gain. Default is 0.4.

out_gain

Set output gain. Default is 0.74

delay

Set delay in milliseconds. Default is 3.0.

decay

Set decay. Default is 0.4.

speed

Set modulation speed in Hz. Default is 0.5.

type

Set modulation type. Default is triangular.

It accepts the following values:

triangular, t

sinusoidal, s

Apply phase shift to input audio samples.

The filter accepts the following options:

shift

Specify phase shift. Allowed range is from -1.0 to 1.0. Default value is 0.0.

level

Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.

order

Set filter order used for filtering. Allowed range is from 1 to 16. Default value is 8.

Commands

This filter supports the all above options as commands.

Measure Audio Peak Signal-to-Noise Ratio.

This filter takes two audio streams for input, and outputs first audio stream. Results are in dB per channel at end of either input.

Apply Psychoacoustic clipper to input audio stream.

The filter accepts the following options:

level_in

Set input gain. By default it is 1. Range is [0.015625 - 64].

level_out

Set output gain. By default it is 1. Range is [0.015625 - 64].

clip

Set the clipping start value. Default value is 0dBFS or 1.

diff

Output only difference samples, useful to hear introduced distortions. By default is disabled.

adaptive

Set strength of adaptive distortion applied. Default value is 0.5. Allowed range is from 0 to 1.

iterations

Set number of iterations of psychoacoustic clipper. Allowed range is from 1 to 20. Default value is 10.

level

Auto level output signal. Default is disabled. This normalizes audio back to 0dBFS if enabled.

Commands

This filter supports the all above options as commands.

Audio pulsator is something between an autopanner and a tremolo. But it can produce funny stereo effects as well. Pulsator changes the volume of the left and right channel based on a LFO (low frequency oscillator) with different waveforms and shifted phases. This filter have the ability to define an offset between left and right channel. An offset of 0 means that both LFO shapes match each other. The left and right channel are altered equally - a conventional tremolo. An offset of 50% means that the shape of the right channel is exactly shifted in phase (or moved backwards about half of the frequency) - pulsator acts as an autopanner. At 1 both curves match again. Every setting in between moves the phase shift gapless between all stages and produces some "bypassing" sounds with sine and triangle waveforms. The more you set the offset near 1 (starting from the 0.5) the faster the signal passes from the left to the right speaker.

The filter accepts the following options:

level_in

Set input gain. By default it is 1. Range is [0.015625 - 64].

level_out

Set output gain. By default it is 1. Range is [0.015625 - 64].

mode

Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or sawdown. Default is sine.

amount

Set modulation. Define how much of original signal is affected by the LFO.

offset_l

Set left channel offset. Default is 0. Allowed range is [0 - 1].

offset_r

Set right channel offset. Default is 0.5. Allowed range is [0 - 1].

width

Set pulse width. Default is 1. Allowed range is [0 - 2].

timing

Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.

bpm

Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing is set to bpm.

ms

Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing is set to ms.

hz

Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used if timing is set to hz.

Resample the input audio to the specified parameters, using the libswresample library. If none are specified then the filter will automatically convert between its input and output.

This filter is also able to stretch/squeeze the audio data to make it match the timestamps or to inject silence / cut out audio to make it match the timestamps, do a combination of both or do neither.

The filter accepts the syntax [sample_rate:]resampler_options, where sample_rate expresses a sample rate and resampler_options is a list of key=value pairs, separated by ":". See the "Resampler Options" section in the ffmpeg-resampler(1) manual for the complete list of supported options.

Examples

• Resample the input audio to 44100Hz:

          aresample=44100

• Stretch/squeeze samples to the given timestamps, with a maximum of 1000 samples per second compensation:

          aresample=async=1000

Reverse an audio clip.

Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.

Examples

• Take the first 5 seconds of a clip, and reverse it.

          atrim=end=5,areverse

Apply Recursive Least Squares algorithm to the first audio stream using the second audio stream.

This adaptive filter is used to mimic a desired filter by recursively finding the filter coefficients that relate to producing the minimal weighted linear least squares cost function of the error signal (difference between the desired, 2nd input audio stream and the actual signal, the 1st input audio stream).

A description of the accepted options follows.

order

Set the filter order.

lambda

Set the forgetting factor.

delta

Set the coefficient to initialize internal covariance matrix.

out_mode

Set the filter output samples. It accepts the following values:

i

Pass the 1st input.

d

Pass the 2nd input.

o

Pass difference between desired, 2nd input and error signal estimate.

n

Pass difference between input, 1st input and error signal estimate.

e

Pass error signal estimated samples.

Default value is o.

precision

Set which precision to use when processing samples.

auto

Auto pick internal sample format depending on other filters.

float

Always use single-floating point precision sample format.

double

Always use double-floating point precision sample format.

Reduce noise from speech using Recurrent Neural Networks.

This filter accepts the following options:

model, m

Set train model file to load. This option is always required.

mix

Set how much to mix filtered samples into final output. Allowed range is from -1 to 1. Default value is 1. Negative values are special, they set how much to keep filtered noise in the final filter output. Set this option to -1 to hear actual noise removed from input signal.

Commands

This filter supports the all above options as commands.

Measure Audio Signal-to-Distortion Ratio.

This filter takes two audio streams for input, and outputs first audio stream. Results are in dB per channel at end of either input.

Set the number of samples per each output audio frame.

The last output packet may contain a different number of samples, as the filter will flush all the remaining samples when the input audio signals its end.

The filter accepts the following options:

nb_out_samples, n

Set the number of frames per each output audio frame. The number is intended as the number of samples per each channel. Default value is 1024.

pad, p

If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame will contain the same number of samples as the previous ones. Default value is 1.

For example, to set the number of per-frame samples to 1234 and disable padding for the last frame, use:

        asetnsamples=n=1234:p=0

Set the sample rate without altering the PCM data. This will result in a change of speed and pitch.

The filter accepts the following options:

sample_rate, r

Set the output sample rate. Default is 44100 Hz.

Show a line containing various information for each input audio frame. The input audio is not modified.

The shown line contains a sequence of key/value pairs of the form key:value.

The following values are shown in the output:

n

The (sequential) number of the input frame, starting from 0.

pts

The presentation timestamp of the input frame, in time base units; the time base depends on the filter input pad, and is usually 1/sample_rate.

pts_time

The presentation timestamp of the input frame in seconds.

fmt

The sample format.

chlayout

The channel layout.

rate

The sample rate for the audio frame.

nb_samples

The number of samples (per channel) in the frame.

checksum

The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio, the data is treated as if all the planes were concatenated.

plane_checksums

A list of Adler-32 checksums for each data plane.

Measure Audio Scaled-Invariant Signal-to-Distortion Ratio.

This filter takes two audio streams for input, and outputs first audio stream. Results are in dB per channel at end of either input.

Apply audio soft clipping.

Soft clipping is a type of distortion effect where the amplitude of a signal is saturated along a smooth curve, rather than the abrupt shape of hard-clipping.

This filter accepts the following options:

type

Set type of soft-clipping.

It accepts the following values:

hard

tanh

atan

cubic

exp

alg

quintic

sin

erf

threshold

Set threshold from where to start clipping. Default value is 0dB or 1.

output

Set gain applied to output. Default value is 0dB or 1.

param

Set additional parameter which controls sigmoid function.

oversample

Set oversampling factor.

Commands

This filter supports the all above options as commands.

Display frequency domain statistical information about the audio channels. Statistics are calculated and stored as metadata for each audio channel and for each audio frame.

It accepts the following option:

win_size

Set the window length in samples. Default value is 2048. Allowed range is from 32 to 65536.

win_func

Set window function.

It accepts the following values:

rect

bartlett

hann, hanning

hamming

blackman

welch

flattop

bharris

bnuttall

bhann

sine

nuttall

lanczos

gauss

tukey

dolph

cauchy

parzen

poisson

bohman

kaiser
Default is hann.

overlap

Set window overlap. Allowed range is from 0 to 1. Default value is 0.5.

measure

Select the parameters which are measured. The metadata keys can be used as flags, default is all which measures everything. none disables all measurement.

A list of each metadata key follows:

mean

variance

centroid

spread

skewness

kurtosis

entropy

flatness

crest

flux

slope

decrease

rolloff

Automatic Speech Recognition

This filter uses PocketSphinx for speech recognition. To enable compilation of this filter, you need to configure FFmpeg with --enable-pocketsphinx.

It accepts the following options:

rate

Set sampling rate of input audio. Defaults is 16000. This need to match speech models, otherwise one will get poor results.

hmm

Set dictionary containing acoustic model files.

dict

Set pronunciation dictionary.

lm

Set language model file.

lmctl

Set language model set.

lmname

Set which language model to use.

logfn

Set output for log messages.

The filter exports recognized speech as the frame metadata lavfi.asr.text.

Display time domain statistical information about the audio channels. Statistics are calculated and displayed for each audio channel and, where applicable, an overall figure is also given.

It accepts the following option:

length

Short window length in seconds, used for peak and trough RMS measurement. Default is 0.05 (50 milliseconds). Allowed range is [0 - 10].

metadata

Set metadata injection. All the metadata keys are prefixed with lavfi.astats.X, where X is channel number starting from 1 or string Overall. Default is disabled.

Available keys for each channel are: Bit_depth Crest_factor DC_offset Dynamic_range Entropy Flat_factor Max_difference Max_level Mean_difference Min_difference Min_level Noise_floor Noise_floor_count Number_of_Infs Number_of_NaNs Number_of_denormals Peak_count Abs_Peak_count Peak_level RMS_difference RMS_peak RMS_trough Zero_crossings Zero_crossings_rate

and for Overall: Bit_depth DC_offset Entropy Flat_factor Max_difference Max_level Mean_difference Min_difference Min_level Noise_floor Noise_floor_count Number_of_Infs Number_of_NaNs Number_of_denormals Number_of_samples Peak_count Abs_Peak_count Peak_level RMS_difference RMS_level RMS_peak RMS_trough

For example, a full key looks like lavfi.astats.1.DC_offset or lavfi.astats.Overall.Peak_count.

Read below for the description of the keys.

reset

Set the number of frames over which cumulative stats are calculated before being reset. Default is disabled.

measure_perchannel

Select the parameters which are measured per channel. The metadata keys can be used as flags, default is all which measures everything. none disables all per channel measurement.

measure_overall

Select the parameters which are measured overall. The metadata keys can be used as flags, default is all which measures everything. none disables all overall measurement.

A description of the measure keys follow:

none

no measures

all

all measures

Bit_depth

overall bit depth of audio, i.e. number of bits used for each sample

Crest_factor

standard ratio of peak to RMS level (note: not in dB)

DC_offset

mean amplitude displacement from zero

Dynamic_range

measured dynamic range of audio in dB

Entropy

entropy measured across whole audio, entropy of value near 1.0 is typically measured for white noise

Flat_factor

flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min_level or Max_level)

Max_difference

maximal difference between two consecutive samples

Max_level

maximal sample level

Mean_difference

mean difference between two consecutive samples, i.e. the average of each difference between two consecutive samples

Min_difference

minimal difference between two consecutive samples

Min_level

minimal sample level

Noise_floor

minimum local peak measured in dBFS over a short window

Noise_floor_count

number of occasions (not the number of samples) that the signal attained Noise floor

Number_of_Infs

number of samples with an infinite value

Number_of_NaNs

number of samples with a NaN (not a number) value

Number_of_denormals

number of samples with a subnormal value

Number_of_samples

number of samples

Peak_count

number of occasions (not the number of samples) that the signal attained either Min_level or Max_level

Abs_Peak_count

number of occasions that the absolute samples taken from the signal attained max absolute value of Min_level and Max_level

Peak_level

standard peak level measured in dBFS

RMS_difference

Root Mean Square difference between two consecutive samples

RMS_level

standard RMS level measured in dBFS

RMS_peak

RMS_trough

peak and trough values for RMS level measured over a short window, measured in dBFS.

Zero crossings

number of points where the waveform crosses the zero level axis

Zero crossings rate

rate of Zero crossings and number of audio samples

Boost subwoofer frequencies.

The filter accepts the following options:

dry

Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1. Default value is 1.0.

wet

Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1. Default value is 1.0.

boost

Set max boost factor. Allowed range is from 1 to 12. Default value is 2.

decay

Set delay line decay gain value. Allowed range is from 0 to 1. Default value is 0.0.

feedback

Set delay line feedback gain value. Allowed range is from 0 to 1. Default value is 0.9.

cutoff

Set cutoff frequency in Hertz. Allowed range is 50 to 900. Default value is 100.

slope

Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1. Default value is 0.5.

delay

Set delay. Allowed range is from 1 to 100. Default value is 20.

channels

Set the channels to process. Default value is all available.

Commands

This filter supports the all above options as commands.

Cut subwoofer frequencies.

This filter allows to set custom, steeper roll off than highpass filter, and thus is able to more attenuate frequency content in stop-band.

The filter accepts the following options:

cutoff

Set cutoff frequency in Hertz. Allowed range is 2 to 200. Default value is 20.

order

Set filter order. Available values are from 3 to 20. Default value is 10.

level

Set input gain level. Allowed range is from 0 to 1. Default value is 1.

Commands

This filter supports the all above options as commands.

Cut super frequencies.

The filter accepts the following options:

cutoff

Set cutoff frequency in Hertz. Allowed range is 20000 to 192000. Default value is 20000.

order

Set filter order. Available values are from 3 to 20. Default value is 10.

level

Set input gain level. Allowed range is from 0 to 1. Default value is 1.

Commands

This filter supports the all above options as commands.

Apply high order Butterworth band-pass filter.

The filter accepts the following options:

centerf

Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.

order

Set filter order. Available values are from 4 to 20. Default value is 4.

qfactor

Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.

level

Set input gain level. Allowed range is from 0 to 2. Default value is 1.

Commands

This filter supports the all above options as commands.

Apply high order Butterworth band-stop filter.

The filter accepts the following options:

centerf

Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.

order

Set filter order. Available values are from 4 to 20. Default value is 4.

qfactor

Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.

level

Set input gain level. Allowed range is from 0 to 2. Default value is 1.

Commands

This filter supports the all above options as commands.

Adjust audio tempo.

The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume nominal 1.0 tempo. Tempo must be in the [0.5, 100.0] range.

Note that tempo greater than 2 will skip some samples rather than blend them in.  If for any reason this is a concern it is always possible to daisy-chain several instances of atempo to achieve the desired product tempo.

Examples

• Slow down audio to 80% tempo:

          atempo=0.8

• To speed up audio to 300% tempo:

          atempo=3

• To speed up audio to 300% tempo by daisy-chaining two atempo instances:

          atempo=sqrt(3),atempo=sqrt(3)

Commands

This filter supports the following commands:

tempo

Change filter tempo scale factor. Syntax for the command is : "tempo"

Apply spectral tilt filter to audio stream.

This filter apply any spectral roll-off slope over any specified frequency band.

The filter accepts the following options:

freq

Set central frequency of tilt in Hz. Default is 10000 Hz.

slope

Set slope direction of tilt. Default is 0. Allowed range is from -1 to 1.

width

Set width of tilt. Default is 1000. Allowed range is from 100 to 10000.

order

Set order of tilt filter.

level

Set input volume level. Allowed range is from 0 to 4. Default is 1.

Commands

This filter supports the all above options as commands.

Trim the input so that the output contains one continuous subpart of the input.

It accepts the following parameters:

start

Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp start will be the first sample in the output.

end

Specify time of the first audio sample that will be dropped, i.e. the audio sample immediately preceding the one with the timestamp end will be the last sample in the output.

start_pts

Same as start, except this option sets the start timestamp in samples instead of seconds.

end_pts

Same as end, except this option sets the end timestamp in samples instead of seconds.

duration

The maximum duration of the output in seconds.

start_sample

The number of the first sample that should be output.

end_sample

The number of the first sample that should be dropped.

start, end, and duration are expressed as time duration specifications; see the Time duration section in the ffmpeg-utils(1) manual.

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _sample options simply count the samples that pass through the filter. So start/end_pts and start/end_sample will give different results when the timestamps are wrong, inexact or do not start at zero. Also note that this filter does not modify the timestamps. If you wish to have the output timestamps start at zero, insert the asetpts filter after the atrim filter.

If multiple start or end options are set, this filter tries to be greedy and keep all samples that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple atrim filters.

The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time.

Examples:

• Drop everything except the second minute of input:

          ffmpeg -i INPUT -af atrim=60:120

• Keep only the first 1000 samples:

          ffmpeg -i INPUT -af atrim=end_sample=1000

Calculate normalized windowed cross-correlation between two input audio streams.

Resulted samples are always between -1 and 1 inclusive. If result is 1 it means two input samples are highly correlated in that selected segment. Result 0 means they are not correlated at all. If result is -1 it means two input samples are out of phase, which means they cancel each other.

The filter accepts the following options:

size

Set size of segment over which cross-correlation is calculated. Default is 256. Allowed range is from 2 to 131072.

algo

Set algorithm for cross-correlation. Can be slow or fast or best. Default is best. Fast algorithm assumes mean values over any given segment are always zero and thus need much less calculations to make. This is generally not true, but is valid for typical audio streams.

Examples

• Calculate correlation between channels in stereo audio stream:

          ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav

Apply a two-pole Butterworth band-pass filter with central frequency frequency, and (3dB-point) band-width width. The csg option selects a constant skirt gain (peak gain = Q) instead of the default: constant 0dB peak gain. The filter roll off at 6dB per octave (20dB per decade).

The filter accepts the following options:

frequency, f

Set the filter's central frequency. Default is 3000.

csg

Constant skirt gain if set to 1. Defaults to 0.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Specify the band-width of a filter in width_type units.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Commands

This filter supports the following commands:

frequency, f

Change bandpass frequency. Syntax for the command is : "frequency"

width_type, t

Change bandpass width_type. Syntax for the command is : "width_type"

width, w

Change bandpass width. Syntax for the command is : "width"

mix, m

Change bandpass mix. Syntax for the command is : "mix"

Apply a two-pole Butterworth band-reject filter with central frequency frequency, and (3dB-point) band-width width. The filter roll off at 6dB per octave (20dB per decade).

The filter accepts the following options:

frequency, f

Set the filter's central frequency. Default is 3000.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Specify the band-width of a filter in width_type units.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Commands

This filter supports the following commands:

frequency, f

Change bandreject frequency. Syntax for the command is : "frequency"

width_type, t

Change bandreject width_type. Syntax for the command is : "width_type"

width, w

Change bandreject width. Syntax for the command is : "width"

mix, m

Change bandreject mix. Syntax for the command is : "mix"

Boost or cut the bass (lower) frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

gain, g

Give the gain at 0 Hz. Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of clipping when using a positive gain.

frequency, f

Set the filter's central frequency and so can be used to extend or reduce the frequency range to be boosted or cut. The default value is 100 Hz.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Determine how steep is the filter's shelf transition.

poles, p

Set number of poles. Default is 2.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Commands

This filter supports the following commands:

frequency, f

Change bass frequency. Syntax for the command is : "frequency"

width_type, t

Change bass width_type. Syntax for the command is : "width_type"

width, w

Change bass width. Syntax for the command is : "width"

gain, g

Change bass gain. Syntax for the command is : "gain"

mix, m

Change bass mix. Syntax for the command is : "mix"

Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are the numerator and denominator coefficients respectively. and channels, c specify which channels to filter, by default all available are filtered.

Commands

This filter supports the following commands:

a0

a1

a2

b0

b1

b2

Change biquad parameter. Syntax for the command is : "value"

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records.

To enable compilation of this filter you need to configure FFmpeg with --enable-libbs2b.

It accepts the following parameters:

profile

Pre-defined crossfeed level.

default

Default level (fcut=700, feed=50).

cmoy

Chu Moy circuit (fcut=700, feed=60).

jmeier

Jan Meier circuit (fcut=650, feed=95).

fcut

Cut frequency (in Hz).

feed

Feed level (in Hz).

Remap input channels to new locations.

It accepts the following parameters:

map

Map channels from input to output. The argument is a '|'-separated list of mappings, each in the in_channel-out_channel or in_channel form. in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the input channel layout. out_channel is the name of the output channel or its index in the output channel layout. If out_channel is not given then it is implicitly an index, starting with zero and increasing by one for each mapping. Mixing different types of mappings is not allowed and will result in a parse error.

channel_layout

The channel layout of the output stream. If not specified, then filter will guess it based on the out_channel names or the number of mappings. Guessed layouts will not necessarily contain channels in the order of the mappings.

If no mapping is present, the filter will implicitly map input channels to output channels, preserving indices.

Examples

• For example, assuming a 5.1+downmix input MOV file,

          ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav

  will create an output WAV file tagged as stereo from the downmix channels of the input.

• To fix a 5.1 WAV improperly encoded in AAC's native channel order

          ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav

Split each channel from an input audio stream into a separate output stream.

It accepts the following parameters:

channel_layout

The channel layout of the input stream. The default is "stereo".

channels

A channel layout describing the channels to be extracted as separate output streams or "all" to extract each input channel as a separate stream. The default is "all".

Choosing channels not present in channel layout in the input will result in an error.

Examples

• For example, assuming a stereo input MP3 file,

          ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

  will create an output Matroska file with two audio streams, one containing only the left channel and the other the right channel.

• Split a 5.1 WAV file into per-channel files:

          ffmpeg -i in.wav -filter_complex
          'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
          -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
          front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
          side_right.wav

• Extract only LFE from a 5.1 WAV file:

          ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1:channels=LFE[LFE]'
          -map '[LFE]' lfe.wav

Add a chorus effect to the audio.

Can make a single vocal sound like a chorus, but can also be applied to instrumentation.

Chorus resembles an echo effect with a short delay, but whereas with echo the delay is constant, with chorus, it is varied using using sinusoidal or triangular modulation. The modulation depth defines the range the modulated delay is played before or after the delay. Hence the delayed sound will sound slower or faster, that is the delayed sound tuned around the original one, like in a chorus where some vocals are slightly off key.

It accepts the following parameters:

in_gain

Set input gain. Default is 0.4.

out_gain

Set output gain. Default is 0.4.

delays

Set delays. A typical delay is around 40ms to 60ms.

decays

Set decays.

speeds

Set speeds.

depths

Set depths.

Examples

• A single delay:

          chorus=0.7:0.9:55:0.4:0.25:2

• Two delays:

          chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3

• Fuller sounding chorus with three delays:

          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

Compress or expand the audio's dynamic range.

It accepts the following parameters:

attacks

decays

A list of times in seconds for each channel over which the instantaneous level of the input signal is averaged to determine its volume. attacks refers to increase of volume and decays refers to decrease of volume. For most situations, the attack time (response to the audio getting louder) should be shorter than the decay time, because the human ear is more sensitive to sudden loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and a typical value for decay is 0.8 seconds. If specified number of attacks & decays is lower than number of channels, the last set attack/decay will be used for all remaining channels.

points

A list of points for the transfer function, specified in dB relative to the maximum possible signal amplitude. Each key points list must be defined using the following syntax: x0/y0|x1/y1|x2/y2|.... or x0/y0 x1/y1 x2/y2 ....

The input values must be in strictly increasing order but the transfer function does not have to be monotonically rising. The point 0/0 is assumed but may be overridden (by 0/out-dBn). Typical values for the transfer function are -70/-70|-60/-20|1/0.

soft-knee

Set the curve radius in dB for all joints. It defaults to 0.01.

gain

Set the additional gain in dB to be applied at all points on the transfer function. This allows for easy adjustment of the overall gain. It defaults to 0.

volume

Set an initial volume, in dB, to be assumed for each channel when filtering starts. This permits the user to supply a nominal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding has begun to operate. A typical value for audio which is initially quiet is -90 dB. It defaults to 0.

delay

Set a delay, in seconds. The input audio is analyzed immediately, but audio is delayed before being fed to the volume adjuster. Specifying a delay approximately equal to the attack/decay times allows the filter to effectively operate in predictive rather than reactive mode. It defaults to 0.

Examples

• Make music with both quiet and loud passages suitable for listening to in a noisy environment:

          compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2

  Another example for audio with whisper and explosion parts:

          compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

• A noise gate for when the noise is at a lower level than the signal:

          compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

• Here is another noise gate, this time for when the noise is at a higher level than the signal (making it, in some ways, similar to squelch):

          compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

• 2:1 compression starting at -6dB:

          compand=points=-80/-80|-6/-6|0/-3.8|20/3.5

• 2:1 compression starting at -9dB:

          compand=points=-80/-80|-9/-9|0/-5.3|20/2.9

• 2:1 compression starting at -12dB:

          compand=points=-80/-80|-12/-12|0/-6.8|20/1.9

• 2:1 compression starting at -18dB:

          compand=points=-80/-80|-18/-18|0/-9.8|20/0.7

• 3:1 compression starting at -15dB:

          compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2

• Compressor/Gate:

          compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6

• Expander:

          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

• Hard limiter at -6dB:

          compand=attacks=0:points=-80/-80|-6/-6|20/-6

• Hard limiter at -12dB:

          compand=attacks=0:points=-80/-80|-12/-12|20/-12

• Hard noise gate at -35 dB:

          compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20

• Soft limiter:

          compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8

Compensation Delay Line is a metric based delay to compensate differing positions of microphones or speakers.

For example, you have recorded guitar with two microphones placed in different locations. Because the front of sound wave has fixed speed in normal conditions, the phasing of microphones can vary and depends on their location and interposition. The best sound mix can be achieved when these microphones are in phase (synchronized). Note that a distance of ~30 cm between microphones makes one microphone capture the signal in antiphase to the other microphone. That makes the final mix sound moody. This filter helps to solve phasing problems by adding different delays to each microphone track and make them synchronized.

The best result can be reached when you take one track as base and synchronize other tracks one by one with it. Remember that synchronization/delay tolerance depends on sample rate, too. Higher sample rates will give more tolerance.

The filter accepts the following parameters:

mm

Set millimeters distance. This is compensation distance for fine tuning. Default is 0.

cm

Set cm distance. This is compensation distance for tightening distance setup. Default is 0.

m

Set meters distance. This is compensation distance for hard distance setup. Default is 0.

dry

Set dry amount. Amount of unprocessed (dry) signal. Default is 0.

wet

Set wet amount. Amount of processed (wet) signal. Default is 1.

temp

Set temperature in degrees Celsius. This is the temperature of the environment. Default is 20.

Commands

This filter supports the all above options as commands.

Apply headphone crossfeed filter.

Crossfeed is the process of blending the left and right channels of stereo audio recording. It is mainly used to reduce extreme stereo separation of low frequencies.

The intent is to produce more speaker like sound to the listener.

The filter accepts the following options:

strength

Set strength of crossfeed. Default is 0.2. Allowed range is from 0 to 1. This sets gain of low shelf filter for side part of stereo image. Default is -6dB. Max allowed is -30db when strength is set to 1.

range

Set soundstage wideness. Default is 0.5. Allowed range is from 0 to 1. This sets cut off frequency of low shelf filter. Default is cut off near 1550 Hz. With range set to 1 cut off frequency is set to 2100 Hz.

slope

Set curve slope of low shelf filter. Default is 0.5. Allowed range is from 0.01 to 1.

level_in

Set input gain. Default is 0.9.

level_out

Set output gain. Default is 1.

block_size

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Commands

This filter supports the all above options as commands.

Simple algorithm for audio noise sharpening.

This filter linearly increases differences between each audio sample.

The filter accepts the following options:

i

Sets the intensity of effect (default: 2.0). Must be in range between -10.0 to 0 (unchanged sound) to 10.0 (maximum effect). To inverse filtering use negative value.

c

Enable clipping. By default is enabled.

Commands

This filter supports the all above options as commands.

Apply a DC shift to the audio.

This can be useful to remove a DC offset (caused perhaps by a hardware problem in the recording chain) from the audio. The effect of a DC offset is reduced headroom and hence volume. The astats filter can be used to determine if a signal has a DC offset.

shift

Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the audio.

limitergain

Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent clipping.

Apply de-essing to the audio samples.

i

Set intensity for triggering de-essing. Allowed range is from 0 to 1. Default is 0.

m

Set amount of ducking on treble part of sound. Allowed range is from 0 to 1. Default is 0.5.

f

How much of original frequency content to keep when de-essing. Allowed range is from 0 to 1. Default is 0.5.

s

Set the output mode.

It accepts the following values:

i

Pass input unchanged.

o

Pass ess filtered out.

e

Pass only ess.

Default value is o.

Enhance dialogue in stereo audio.

This filter accepts stereo input and produce surround (3.0) channels output. The newly produced front center channel have enhanced speech dialogue originally available in both stereo channels. This filter outputs front left and front right channels same as available in stereo input.

The filter accepts the following options:

original

Set the original center factor to keep in front center channel output. Allowed range is from 0 to 1. Default value is 1.

enhance

Set the dialogue enhance factor to put in front center channel output. Allowed range is from 0 to 3. Default value is 1.

voice

Set the voice detection factor. Allowed range is from 2 to 32. Default value is 2.

Commands

This filter supports the all above options as commands.

Measure audio dynamic range.

DR values of 14 and higher is found in very dynamic material. DR of 8 to 13 is found in transition material. And anything less that 8 have very poor dynamics and is very compressed.

The filter accepts the following options:

length

Set window length in seconds used to split audio into segments of equal length. Default is 3 seconds.

Dynamic Audio Normalizer.

This filter applies a certain amount of gain to the input audio in order to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in contrast to more "simple" normalization algorithms, the Dynamic Audio Normalizer *dynamically* re-adjusts the gain factor to the input audio. This allows for applying extra gain to the "quiet" sections of the audio while avoiding distortions or clipping the "loud" sections. In other words: The Dynamic Audio Normalizer will "even out" the volume of quiet and loud sections, in the sense that the volume of each section is brought to the same target level. Note, however, that the Dynamic Audio Normalizer achieves this goal *without* applying "dynamic range compressing". It will retain 100% of the dynamic range *within* each section of the audio file.

framelen, f

Set the frame length in milliseconds. In range from 10 to 8000 milliseconds. Default is 500 milliseconds. The Dynamic Audio Normalizer processes the input audio in small chunks, referred to as frames. This is required, because a peak magnitude has no meaning for just a single sample value. Instead, we need to determine the peak magnitude for a contiguous sequence of sample values. While a "standard" normalizer would simply use the peak magnitude of the complete file, the Dynamic Audio Normalizer determines the peak magnitude individually for each frame. The length of a frame is specified in milliseconds. By default, the Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has been found to give good results with most files. Note that the exact frame length, in number of samples, will be determined automatically, based on the sampling rate of the individual input audio file.

gausssize, g

Set the Gaussian filter window size. In range from 3 to 301, must be odd number. Default is 31. Probably the most important parameter of the Dynamic Audio Normalizer is the window size of the Gaussian smoothing filter. The filter's window size is specified in frames, centered around the current frame. For the sake of simplicity, this must be an odd number. Consequently, the default value of 31 takes into account the current frame, as well as the 15 preceding frames and the 15 subsequent frames. Using a larger window results in a stronger smoothing effect and thus in less gain variation, i.e. slower gain adaptation. Conversely, using a smaller window results in a weaker smoothing effect and thus in more gain variation, i.e. faster gain adaptation. In other words, the more you increase this value, the more the Dynamic Audio Normalizer will behave like a "traditional" normalization filter. On the contrary, the more you decrease this value, the more the Dynamic Audio Normalizer will behave like a dynamic range compressor.

peak, p

Set the target peak value. This specifies the highest permissible magnitude level for the normalized audio input. This filter will try to approach the target peak magnitude as closely as possible, but at the same time it also makes sure that the normalized signal will never exceed the peak magnitude. A frame's maximum local gain factor is imposed directly by the target peak magnitude. The default value is 0.95 and thus leaves a headroom of 5%*. It is not recommended to go above this value.

maxgain, m

Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0. The Dynamic Audio Normalizer determines the maximum possible (local) gain factor for each input frame, i.e. the maximum gain factor that does not result in clipping or distortion. The maximum gain factor is determined by the frame's highest magnitude sample. However, the Dynamic Audio Normalizer additionally bounds the frame's maximum gain factor by a predetermined (global) maximum gain factor. This is done in order to avoid excessive gain factors in "silent" or almost silent frames. By default, the maximum gain factor is 10.0, For most inputs the default value should be sufficient and it usually is not recommended to increase this value. Though, for input with an extremely low overall volume level, it may be necessary to allow even higher gain factors. Note, however, that the Dynamic Audio Normalizer does not simply apply a "hard" threshold (i.e. cut off values above the threshold). Instead, a "sigmoid" threshold function will be applied. This way, the gain factors will smoothly approach the threshold value, but never exceed that value.

targetrms, r

Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled. By default, the Dynamic Audio Normalizer performs "peak" normalization. This means that the maximum local gain factor for each frame is defined (only) by the frame's highest magnitude sample. This way, the samples can be amplified as much as possible without exceeding the maximum signal level, i.e. without clipping. Optionally, however, the Dynamic Audio Normalizer can also take into account the frame's root mean square, abbreviated RMS. In electrical engineering, the RMS is commonly used to determine the power of a time-varying signal. It is therefore considered that the RMS is a better approximation of the "perceived loudness" than just looking at the signal's peak magnitude. Consequently, by adjusting all frames to a constant RMS value, a uniform "perceived loudness" can be established. If a target RMS value has been specified, a frame's local gain factor is defined as the factor that would result in exactly that RMS value. Note, however, that the maximum local gain factor is still restricted by the frame's highest magnitude sample, in order to prevent clipping.

coupling, n

Enable channels coupling. By default is enabled. By default, the Dynamic Audio Normalizer will amplify all channels by the same amount. This means the same gain factor will be applied to all channels, i.e. the maximum possible gain factor is determined by the "loudest" channel. However, in some recordings, it may happen that the volume of the different channels is uneven, e.g. one channel may be "quieter" than the other one(s). In this case, this option can be used to disable the channel coupling. This way, the gain factor will be determined independently for each channel, depending only on the individual channel's highest magnitude sample. This allows for harmonizing the volume of the different channels.

correctdc, c

Enable DC bias correction. By default is disabled. An audio signal (in the time domain) is a sequence of sample values. In the Dynamic Audio Normalizer these sample values are represented in the -1.0 to 1.0 range, regardless of the original input format. Normally, the audio signal, or "waveform", should be centered around the zero point. That means if we calculate the mean value of all samples in a file, or in a single frame, then the result should be 0.0 or at least very close to that value. If, however, there is a significant deviation of the mean value from 0.0, in either positive or negative direction, this is referred to as a DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic Audio Normalizer provides optional DC bias correction. With DC bias correction enabled, the Dynamic Audio Normalizer will determine the mean value, or "DC correction" offset, of each input frame and subtract that value from all of the frame's sample values which ensures those samples are centered around 0.0 again. Also, in order to avoid "gaps" at the frame boundaries, the DC correction offset values will be interpolated smoothly between neighbouring frames.

altboundary, b

Enable alternative boundary mode. By default is disabled. The Dynamic Audio Normalizer takes into account a certain neighbourhood around each frame. This includes the preceding frames as well as the subsequent frames. However, for the "boundary" frames, located at the very beginning and at the very end of the audio file, not all neighbouring frames are available. In particular, for the first few frames in the audio file, the preceding frames are not known. And, similarly, for the last few frames in the audio file, the subsequent frames are not known. Thus, the question arises which gain factors should be assumed for the missing frames in the "boundary" region. The Dynamic Audio Normalizer implements two modes to deal with this situation. The default boundary mode assumes a gain factor of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and "fade out" at the beginning and at the end of the input, respectively.

compress, s

Set the compress factor. In range from 0.0 to 30.0. Default is 0.0. By default, the Dynamic Audio Normalizer does not apply "traditional" compression. This means that signal peaks will not be pruned and thus the full dynamic range will be retained within each local neighbourhood. However, in some cases it may be desirable to combine the Dynamic Audio Normalizer's normalization algorithm with a more "traditional" compression. For this purpose, the Dynamic Audio Normalizer provides an optional compression (thresholding) function. If (and only if) the compression feature is enabled, all input frames will be processed by a soft knee thresholding function prior to the actual normalization process. Put simply, the thresholding function is going to prune all samples whose magnitude exceeds a certain threshold value. However, the Dynamic Audio Normalizer does not simply apply a fixed threshold value. Instead, the threshold value will be adjusted for each individual frame. In general, smaller parameters result in stronger compression, and vice versa. Values below 3.0 are not recommended, because audible distortion may appear.

threshold, t

Set the target threshold value. This specifies the lowest permissible magnitude level for the audio input which will be normalized. If input frame volume is above this value frame will be normalized. Otherwise frame may not be normalized at all. The default value is set to 0, which means all input frames will be normalized. This option is mostly useful if digital noise is not wanted to be amplified.

channels, h

Specify which channels to filter, by default all available channels are filtered.

overlap, o

Specify overlap for frames. If set to 0 (default) no frame overlapping is done. Using >0 and <1 values will make less conservative gain adjustments, like when framelen option is set to smaller value, if framelen option value is compensated for non-zero overlap then gain adjustments will be smoother across time compared to zero overlap case.

curve, v

Specify the peak mapping curve expression which is going to be used when calculating gain applied to frames. The max output frame gain will still be limited by other options mentioned previously for this filter.

The expression can contain the following constants:

ch

current channel number

sn

current sample number

nb_channels

number of channels

t

timestamp expressed in seconds

sr

sample rate

p

current frame peak value

Commands

This filter supports the all above options as commands.

Make audio easier to listen to on headphones.

This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when listened to on headphones the stereo image is moved from inside your head (standard for headphones) to outside and in front of the listener (standard for speakers).

Ported from SoX.

Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at and around a selected frequency can be increased or decreased, whilst (unlike bandpass and bandreject filters) that at all other frequencies is unchanged.

In order to produce complex equalisation curves, this filter can be given several times, each with a different central frequency.

The filter accepts the following options:

frequency, f

Set the filter's central frequency in Hz.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Specify the band-width of a filter in width_type units.

gain, g

Set the required gain or attenuation in dB. Beware of clipping when using a positive gain.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Examples

• Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

          equalizer=f=1000:t=h:width=200:g=-10

• Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:

          equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5

Commands

This filter supports the following commands:

frequency, f

Change equalizer frequency. Syntax for the command is : "frequency"

width_type, t

Change equalizer width_type. Syntax for the command is : "width_type"

width, w

Change equalizer width. Syntax for the command is : "width"

gain, g

Change equalizer gain. Syntax for the command is : "gain"

mix, m

Change equalizer mix. Syntax for the command is : "mix"

Linearly increases the difference between left and right channels which adds some sort of "live" effect to playback.

The filter accepts the following options:

m

Sets the difference coefficient (default: 2.5). 0.0 means mono sound (average of both channels), with 1.0 sound will be unchanged, with -1.0 left and right channels will be swapped.

c

Enable clipping. By default is enabled.

Commands

This filter supports the all above options as commands.

Apply FIR Equalization using arbitrary frequency response.

The filter accepts the following option:

gain

Set gain curve equation (in dB). The expression can contain variables:

f

the evaluated frequency

sr

sample rate

ch

channel number, set to 0 when multichannels evaluation is disabled

chid

channel id, see libavutil/channel_layout.h, set to the first channel id when multichannels evaluation is disabled

chs

number of channels

chlayout

channel_layout, see libavutil/channel_layout.h

and functions:

gain_interpolate(f)

interpolate gain on frequency f based on gain_entry

cubic_interpolate(f)

same as gain_interpolate, but smoother

This option is also available as command. Default is gain_interpolate(f).

gain_entry

Set gain entry for gain_interpolate function. The expression can contain functions:

entry(f, g)

store gain entry at frequency f with value g

This option is also available as command.

delay

Set filter delay in seconds. Higher value means more accurate. Default is 0.01.

accuracy

Set filter accuracy in Hz. Lower value means more accurate. Default is 5.

wfunc

Set window function. Acceptable values are:

rectangular

rectangular window, useful when gain curve is already smooth

hann

hann window (default)

hamming

hamming window

blackman

blackman window

nuttall3

3-terms continuous 1st derivative nuttall window

mnuttall3

minimum 3-terms discontinuous nuttall window

nuttall

4-terms continuous 1st derivative nuttall window

bnuttall

minimum 4-terms discontinuous nuttall (blackman-nuttall) window

bharris

blackman-harris window

tukey

tukey window

fixed

If enabled, use fixed number of audio samples. This improves speed when filtering with large delay. Default is disabled.

multi

Enable multichannels evaluation on gain. Default is disabled.

zero_phase

Enable zero phase mode by subtracting timestamp to compensate delay. Default is disabled.

scale

Set scale used by gain. Acceptable values are:

linlin

linear frequency, linear gain

linlog

linear frequency, logarithmic (in dB) gain (default)

loglin

logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain

loglog

logarithmic frequency, logarithmic gain

dumpfile

Set file for dumping, suitable for gnuplot.

dumpscale

Set scale for dumpfile. Acceptable values are same with scale option. Default is linlog.

fft2

Enable 2-channel convolution using complex FFT. This improves speed significantly. Default is disabled.

min_phase

Enable minimum phase impulse response. Default is disabled.

Examples

• lowpass at 1000 Hz:

          firequalizer=gain='if(lt(f,1000), 0, -INF)'

• lowpass at 1000 Hz with gain_entry:

          firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'

• custom equalization:

          firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'

• higher delay with zero phase to compensate delay:

          firequalizer=delay=0.1:fixed=on:zero_phase=on

• lowpass on left channel, highpass on right channel:

          firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
          :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on

Apply a flanging effect to the audio.

The filter accepts the following options:

delay

Set base delay in milliseconds. Range from 0 to 30. Default value is 0.

depth

Set added sweep delay in milliseconds. Range from 0 to 10. Default value is 2.

regen

Set percentage regeneration (delayed signal feedback). Range from -95 to 95. Default value is 0.

width

Set percentage of delayed signal mixed with original. Range from 0 to 100. Default value is 71.

speed

Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.

shape

Set swept wave shape, can be triangular or sinusoidal. Default value is sinusoidal.

phase

Set swept wave percentage-shift for multi channel. Range from 0 to 100. Default value is 25.

interp

Set delay-line interpolation, linear or quadratic. Default is linear.

Apply Haas effect to audio.

Note that this makes most sense to apply on mono signals. With this filter applied to mono signals it give some directionality and stretches its stereo image.

The filter accepts the following options:

level_in

Set input level. By default is 1, or 0dB

level_out

Set output level. By default is 1, or 0dB.

side_gain

Set gain applied to side part of signal. By default is 1.

middle_source

Set kind of middle source. Can be one of the following:

left

Pick left channel.

right

Pick right channel.

mid

Pick middle part signal of stereo image.

side

Pick side part signal of stereo image.

middle_phase

Change middle phase. By default is disabled.

left_delay

Set left channel delay. By default is 2.05 milliseconds.

left_balance

Set left channel balance. By default is -1.

left_gain

Set left channel gain. By default is 1.

left_phase

Change left phase. By default is disabled.

right_delay

Set right channel delay. By defaults is 2.12 milliseconds.

right_balance

Set right channel balance. By default is 1.

right_gain

Set right channel gain. By default is 1.

right_phase

Change right phase. By default is enabled.

Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with embedded HDCD codes is expanded into a 20-bit PCM stream.

The filter supports the Peak Extend and Low-level Gain Adjustment features of HDCD, and detects the Transient Filter flag.

        ffmpeg -i HDCD16.flac -af hdcd OUT24.flac

When using the filter with wav, note the default encoding for wav is 16-bit, so the resulting 20-bit stream will be truncated back to 16-bit. Use something like -acodec pcm_s24le after the filter to get 24-bit PCM output.

        ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
        ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav

The filter accepts the following options:

disable_autoconvert

Disable any automatic format conversion or resampling in the filter graph.

process_stereo

Process the stereo channels together. If target_gain does not match between channels, consider it invalid and use the last valid target_gain.

cdt_ms

Set the code detect timer period in ms.

force_pe

Always extend peaks above -3dBFS even if PE isn't signaled.

analyze_mode

Replace audio with a solid tone and adjust the amplitude to signal some specific aspect of the decoding process. The output file can be loaded in an audio editor alongside the original to aid analysis.

analyze_mode=pe:force_pe=true can be used to see all samples above the PE level.

Modes are:

0, off

Disabled

1, lle

Gain adjustment level at each sample

2, pe

Samples where peak extend occurs

3, cdt

Samples where the code detect timer is active

4, tgm

Samples where the target gain does not match between channels

Apply head-related transfer functions (HRTFs) to create virtual loudspeakers around the user for binaural listening via headphones. The HRIRs are provided via additional streams, for each channel one stereo input stream is needed.

The filter accepts the following options:

map

Set mapping of input streams for convolution. The argument is a '|'-separated list of channel names in order as they are given as additional stream inputs for filter. This also specify number of input streams. Number of input streams must be not less than number of channels in first stream plus one.

gain

Set gain applied to audio. Value is in dB. Default is 0.

type

Set processing type. Can be time or freq. time is processing audio in time domain which is slow. freq is processing audio in frequency domain which is fast. Default is freq.

lfe

Set custom gain for LFE channels. Value is in dB. Default is 0.

size

Set size of frame in number of samples which will be processed at once. Default value is 1024. Allowed range is from 1024 to 96000.

hrir

Set format of hrir stream. Default value is stereo. Alternative value is multich. If value is set to stereo, number of additional streams should be greater or equal to number of input channels in first input stream. Also each additional stream should have stereo number of channels. If value is set to multich, number of additional streams should be exactly one. Also number of input channels of additional stream should be equal or greater than twice number of channels of first input stream.

Examples

• Full example using wav files as coefficients with amovie filters for 7.1 downmix, each amovie filter use stereo file with IR coefficients as input. The files give coefficients for each position of virtual loudspeaker:

          ffmpeg -i input.wav
          -filter_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];[0:a][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
          output.wav

• Full example using wav files as coefficients with amovie filters for 7.1 downmix, but now in multich hrir format.

          ffmpeg -i input.wav -filter_complex "amovie=minp.wav[hrirs];[0:a][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
          output.wav

Apply a high-pass filter with 3dB point frequency. The filter can be either single-pole, or double-pole (the default). The filter roll off at 6dB per pole per octave (20dB per pole per decade).

The filter accepts the following options:

frequency, f

Set frequency in Hz. Default is 3000.

poles, p

Set number of poles. Default is 2.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Specify the band-width of a filter in width_type units. Applies only to double-pole filter. The default is 0.707q and gives a Butterworth response.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Commands

This filter supports the following commands:

frequency, f

Change highpass frequency. Syntax for the command is : "frequency"

width_type, t

Change highpass width_type. Syntax for the command is : "width_type"

width, w

Change highpass width. Syntax for the command is : "width"

mix, m

Change highpass mix. Syntax for the command is : "mix"

Join multiple input streams into one multi-channel stream.

It accepts the following parameters:

inputs

The number of input streams. It defaults to 2.

channel_layout

The desired output channel layout. It defaults to stereo.

map

Map channels from inputs to output. The argument is a '|'-separated list of mappings, each in the input_idx.in_channel-out_channel form. input_idx is the 0-based index of the input stream. in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the specified input stream. out_channel is the name of the output channel.

The filter will attempt to guess the mappings when they are not specified explicitly. It does so by first trying to find an unused matching input channel and if that fails it picks the first unused input channel.

Join 3 inputs (with properly set channel layouts):

        ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

Build a 5.1 output from 6 single-channel streams:

        ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
        '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'
        out

Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.

To enable compilation of this filter you need to configure FFmpeg with --enable-ladspa.

file, f

Specifies the name of LADSPA plugin library to load. If the environment variable LADSPA_PATH is defined, the LADSPA plugin is searched in each one of the directories specified by the colon separated list in LADSPA_PATH, otherwise in the standard LADSPA paths, which are in this order: HOME/.ladspa/lib/, /usr/local/lib/ladspa/, /usr/lib/ladspa/.

plugin, p

Specifies the plugin within the library. Some libraries contain only one plugin, but others contain many of them. If this is not set filter will list all available plugins within the specified library.

controls, c

Set the '|' separated list of controls which are zero or more floating point values that determine the behavior of the loaded plugin (for example delay, threshold or gain). Controls need to be defined using the following syntax: c0=value0|c1=value1|c2=value2|..., where valuei is the value set on the i-th control. Alternatively they can be also defined using the following syntax: value0|value1|value2|..., where valuei is the value set on the i-th control. If controls is set to help, all available controls and their valid ranges are printed.

sample_rate, s

Specify the sample rate, default to 44100. Only used if plugin have zero inputs.

nb_samples, n

Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.

duration, d

Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame. If not specified, or the expressed duration is negative, the audio is supposed to be generated forever. Only used if plugin have zero inputs.

latency, l

Enable latency compensation, by default is disabled. Only used if plugin have inputs.

Examples

• List all available plugins within amp (LADSPA example plugin) library:

          ladspa=file=amp

• List all available controls and their valid ranges for vcf_notch plugin from VCF library:

          ladspa=f=vcf:p=vcf_notch:c=help

• Simulate low quality audio equipment using Computer Music Toolkit (CMT) plugin library:

          ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12

• Add reverberation to the audio using TAP-plugins (Tom's Audio Processing plugins):

          ladspa=file=tap_reverb:tap_reverb

• Generate white noise, with 0.2 amplitude:

          ladspa=file=cmt:noise_source_white:c=c0=.2

• Generate 20 bpm clicks using plugin C* Click - Metronome from the C* Audio Plugin Suite (CAPS) library:

          ladspa=file=caps:Click:c=c1=20'

• Apply C* Eq10X2 - Stereo 10-band equaliser effect:

          ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2

• Increase volume by 20dB using fast lookahead limiter from Steve Harris SWH Plugins collection:

          ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2

• Attenuate low frequencies using Multiband EQ from Steve Harris SWH Plugins collection:

          ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0

• Reduce stereo image using Narrower from the C* Audio Plugin Suite (CAPS) library:

          ladspa=caps:Narrower

• Another white noise, now using C* Audio Plugin Suite (CAPS) library:

          ladspa=caps:White:.2

• Some fractal noise, using C* Audio Plugin Suite (CAPS) library:

          ladspa=caps:Fractal:c=c1=1

• Dynamic volume normalization using VLevel plugin:

          ladspa=vlevel-ladspa:vlevel_mono

Commands

This filter supports the following commands:

cN

Modify the N-th control value.

If the specified value is not valid, it is ignored and prior one is kept.

EBU R128 loudness normalization. Includes both dynamic and linear normalization modes. Support for both single pass (livestreams, files) and double pass (files) modes. This algorithm can target IL, LRA, and maximum true peak. In dynamic mode, to accurately detect true peaks, the audio stream will be upsampled to 192 kHz. Use the -ar option or aresample filter to explicitly set an output sample rate.

The filter accepts the following options:

I, i

Set integrated loudness target. Range is -70.0 - -5.0. Default value is -24.0.

LRA, lra

Set loudness range target. Range is 1.0 - 50.0. Default value is 7.0.

TP, tp

Set maximum true peak. Range is -9.0 - +0.0. Default value is -2.0.

measured_I, measured_i

Measured IL of input file. Range is -99.0 - +0.0.

measured_LRA, measured_lra

Measured LRA of input file. Range is  0.0 - 99.0.

measured_TP, measured_tp

Measured true peak of input file. Range is  -99.0 - +99.0.

measured_thresh

Measured threshold of input file. Range is -99.0 - +0.0.

offset

Set offset gain. Gain is applied before the true-peak limiter. Range is  -99.0 - +99.0. Default is +0.0.

linear

Normalize by linearly scaling the source audio. measured_I, measured_LRA, measured_TP, and measured_thresh must all be specified. Target LRA shouldn't be lower than source LRA and the change in integrated loudness shouldn't result in a true peak which exceeds the target TP. If any of these conditions aren't met, normalization mode will revert to dynamic. Options are true or false. Default is true.

dual_mono

Treat mono input files as "dual-mono". If a mono file is intended for playback on a stereo system, its EBU R128 measurement will be perceptually incorrect. If set to true, this option will compensate for this effect. Multi-channel input files are not affected by this option. Options are true or false. Default is false.

print_format

Set print format for stats. Options are summary, json, or none. Default value is none.

Apply a low-pass filter with 3dB point frequency. The filter can be either single-pole or double-pole (the default). The filter roll off at 6dB per pole per octave (20dB per pole per decade).

The filter accepts the following options:

frequency, f

Set frequency in Hz. Default is 500.

poles, p

Set number of poles. Default is 2.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Specify the band-width of a filter in width_type units. Applies only to double-pole filter. The default is 0.707q and gives a Butterworth response.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Examples

• Lowpass only LFE channel, it LFE is not present it does nothing:

          lowpass=c=LFE

Commands

This filter supports the following commands:

frequency, f

Change lowpass frequency. Syntax for the command is : "frequency"

width_type, t

Change lowpass width_type. Syntax for the command is : "width_type"

width, w

Change lowpass width. Syntax for the command is : "width"

mix, m

Change lowpass mix. Syntax for the command is : "mix"

Load a LV2 (LADSPA Version 2) plugin.

To enable compilation of this filter you need to configure FFmpeg with --enable-lv2.

plugin, p

Specifies the plugin URI. You may need to escape ':'.

controls, c

Set the '|' separated list of controls which are zero or more floating point values that determine the behavior of the loaded plugin (for example delay, threshold or gain). If controls is set to help, all available controls and their valid ranges are printed.

sample_rate, s

Specify the sample rate, default to 44100. Only used if plugin have zero inputs.

nb_samples, n

Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.

duration, d

Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame. If not specified, or the expressed duration is negative, the audio is supposed to be generated forever. Only used if plugin have zero inputs.

Examples

• Apply bass enhancer plugin from Calf:

          lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2

• Apply vinyl plugin from Calf:

          lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5

• Apply bit crusher plugin from ArtyFX:

          lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3

Commands

This filter supports all options that are exported by plugin as commands.

Multiband Compress or expand the audio's dynamic range.

The input audio is divided into bands using 4th order Linkwitz-Riley IIRs. This is akin to the crossover of a loudspeaker, and results in flat frequency response when absent compander action.

It accepts the following parameters:

args

This option syntax is: attack,decay,[attack,decay..] soft-knee points crossover_frequency [delay [initial_volume [gain]]] | attack,decay ... For explanation of each item refer to compand filter documentation.

Mix channels with specific gain levels. The filter accepts the output channel layout followed by a set of channels definitions.

This filter is also designed to efficiently remap the channels of an audio stream.

The filter accepts parameters of the form: "l|outdef|outdef|..."

l

output channel layout or number of channels

outdef

output channel specification, of the form: "out_name=[gain*]in_name[(+-)[gain*]in_name...]"

out_name

output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.)

gain

multiplicative coefficient for the channel, 1 leaving the volume unchanged

in_name

input channel to use, see out_name for details; it is not possible to mix named and numbered input channels

If the `=' in a channel specification is replaced by `<', then the gains for that specification will be renormalized so that the total is 1, thus avoiding clipping noise.

Mixing examples

For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel:

        pan=1c|c0=0.9*c0+0.1*c1

A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels surround:

        pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR

Note that ffmpeg integrates a default down-mix (and up-mix) system that should be preferred (see "-ac" option) unless you have very specific needs.

Remapping examples

The channel remapping will be effective if, and only if:

*<gain coefficients are zeroes or ones,>

*<only one input per channel output,>

If all these conditions are satisfied, the filter will notify the user ("Pure channel mapping detected"), and use an optimized and lossless method to do the remapping.

For example, if you have a 5.1 source and want a stereo audio stream by dropping the extra channels:

        pan="stereo| c0=FL | c1=FR"

Given the same source, you can also switch front left and front right channels and keep the input channel layout:

        pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"

If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo channel layout) with:

        pan="stereo|c1=c1"

Still with a stereo audio stream input, you can copy the right channel in both front left and right:

        pan="stereo| c0=FR | c1=FR"

ReplayGain scanner filter. This filter takes an audio stream as an input and outputs it unchanged. At end of filtering it displays track_gain and track_peak.

The filter accepts the following exported read-only options:

track_gain

Exported track gain in dB at end of stream.

track_peak

Exported track peak at end of stream.

Convert the audio sample format, sample rate and channel layout. It is not meant to be used directly.

Apply time-stretching and pitch-shifting with librubberband.

To enable compilation of this filter, you need to configure FFmpeg with --enable-librubberband.

The filter accepts the following options:

tempo

Set tempo scale factor.

pitch

Set pitch scale factor.

transients

Set transients detector. Possible values are:

crisp

mixed

smooth

detector

Set detector. Possible values are:

compound

percussive

soft

phase

Set phase. Possible values are:

laminar

independent

window

Set processing window size. Possible values are:

standard

short

long

smoothing

Set smoothing. Possible values are:

off

on

formant

Enable formant preservation when shift pitching. Possible values are:

shifted

preserved

pitchq

Set pitch quality. Possible values are:

quality

speed

consistency

channels

Set channels. Possible values are:

apart

together

Commands

This filter supports the following commands:

tempo

Change filter tempo scale factor. Syntax for the command is : "tempo"

pitch

Change filter pitch scale factor. Syntax for the command is : "pitch"

This filter acts like normal compressor but has the ability to compress detected signal using second input signal. It needs two input streams and returns one output stream. First input stream will be processed depending on second stream signal. The filtered signal then can be filtered with other filters in later stages of processing. See pan and amerge filter.

The filter accepts the following options:

level_in

Set input gain. Default is 1. Range is between 0.015625 and 64.

mode

Set mode of compressor operation. Can be upward or downward. Default is downward.

threshold

If a signal of second stream raises above this level it will affect the gain reduction of first stream. By default is 0.125. Range is between 0.00097563 and 1.

ratio

Set a ratio about which the signal is reduced. 1:2 means that if the level raised 4dB above the threshold, it will be only 2dB above after the reduction. Default is 2. Range is between 1 and 20.

attack

Amount of milliseconds the signal has to rise above the threshold before gain reduction starts. Default is 20. Range is between 0.01 and 2000.

release

Amount of milliseconds the signal has to fall below the threshold before reduction is decreased again. Default is 250. Range is between 0.01 and 9000.

makeup

Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1 to 64.

knee

Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.

link

Choose if the average level between all channels of side-chain stream or the louder(maximum) channel of side-chain stream affects the reduction. Default is average.

detection

Should the exact signal be taken in case of peak or an RMS one in case of rms. Default is rms which is mainly smoother.

level_sc

Set sidechain gain. Default is 1. Range is between 0.015625 and 64.

mix

How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Commands

This filter supports the all above options as commands.

Examples

• Full ffmpeg example taking 2 audio inputs, 1st input to be compressed depending on the signal of 2nd input and later compressed signal to be merged with 2nd input:

          ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

A sidechain gate acts like a normal (wideband) gate but has the ability to filter the detected signal before sending it to the gain reduction stage. Normally a gate uses the full range signal to detect a level above the threshold. For example: If you cut all lower frequencies from your sidechain signal the gate will decrease the volume of your track only if not enough highs appear. With this technique you are able to reduce the resonation of a natural drum or remove "rumbling" of muted strokes from a heavily distorted guitar. It needs two input streams and returns one output stream. First input stream will be processed depending on second stream signal.

The filter accepts the following options:

level_in

Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.

mode

Set the mode of operation. Can be upward or downward. Default is downward. If set to upward mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of downward lower parts of signal will be reduced.

range

Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1. Setting this to 0 disables reduction and then filter behaves like expander.

threshold

If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.

ratio

Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.

attack

Amount of milliseconds the signal has to rise above the threshold before gain reduction stops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.

release

Amount of milliseconds the signal has to fall below the threshold before the reduction is increased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.

makeup

Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.

knee

Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.

detection

Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.

link

Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.

level_sc

Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

Commands

This filter supports the all above options as commands.

Detect silence in an audio stream.

This filter logs a message when it detects that the input audio volume is less or equal to a noise tolerance value for a duration greater or equal to the minimum detected noise duration.

The printed times and duration are expressed in seconds. The lavfi.silence_start or lavfi.silence_start.X metadata key is set on the first frame whose timestamp equals or exceeds the detection duration and it contains the timestamp of the first frame of the silence.

The lavfi.silence_duration or lavfi.silence_duration.X and lavfi.silence_end or lavfi.silence_end.X metadata keys are set on the first frame after the silence. If mono is enabled, and each channel is evaluated separately, the .X suffixed keys are used, and X corresponds to the channel number.

The filter accepts the following options:

noise, n

Set noise tolerance. Can be specified in dB (in case "dB" is appended to the specified value) or amplitude ratio. Default is -60dB, or 0.001.

duration, d

Set silence duration until notification (default is 2 seconds). See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.

mono, m

Process each channel separately, instead of combined. By default is disabled.

Examples

• Detect 5 seconds of silence with -50dB noise tolerance:

          silencedetect=n=-50dB:d=5

• Complete example with ffmpeg to detect silence with 0.0001 noise tolerance in silence.mp3:

          ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

Remove silence from the beginning, middle or end of the audio.

The filter accepts the following options:

start_periods

This value is used to indicate if audio should be trimmed at beginning of the audio. A value of zero indicates no silence should be trimmed from the beginning. When specifying a non-zero value, it trims audio up until it finds non-silence. Normally, when trimming silence from beginning of audio the start_periods will be 1 but it can be increased to higher values to trim all audio up to specific count of non-silence periods. Default value is 0.

start_duration

Specify the amount of time that non-silence must be detected before it stops trimming audio. By increasing the duration, bursts of noises can be treated as silence and trimmed off. Default value is 0.

start_threshold

This indicates what sample value should be treated as silence. For digital audio, a value of 0 may be fine but for audio recorded from analog, you may wish to increase the value to account for background noise. Can be specified in dB (in case "dB" is appended to the specified value) or amplitude ratio. Default value is 0.

start_silence

Specify max duration of silence at beginning that will be kept after trimming. Default is 0, which is equal to trimming all samples detected as silence.

start_mode

Specify mode of detection of silence end at start of multi-channel audio. Can be any or all. Default is any. With any, any sample from any channel that is detected as non-silence will trigger end of silence trimming at start of audio stream. With all, only if every sample from every channel is detected as non-silence will trigger end of silence trimming at start of audio stream, limited usage.

stop_periods

Set the count for trimming silence from the end of audio. When specifying a positive value, it trims audio after it finds specified silence period. To remove silence from the middle of a file, specify a stop_periods that is negative. This value is then treated as a positive value and is used to indicate the effect should restart processing as specified by stop_periods, making it suitable for removing periods of silence in the middle of the audio. Default value is 0.

stop_duration

Specify a duration of silence that must exist before audio is not copied any more. By specifying a higher duration, silence that is wanted can be left in the audio. Default value is 0.

stop_threshold

This is the same as start_threshold but for trimming silence from the end of audio. Can be specified in dB (in case "dB" is appended to the specified value) or amplitude ratio. Default value is 0.

stop_silence

Specify max duration of silence at end that will be kept after trimming. Default is 0, which is equal to trimming all samples detected as silence.

stop_mode

Specify mode of detection of silence start after start of multi-channel audio. Can be any or all. Default is all. With any, any sample from any channel that is detected as silence will trigger start of silence trimming after start of audio stream, limited usage. With all, only if every sample from every channel is detected as silence will trigger start of silence trimming after start of audio stream.

detection

Set how is silence detected.

avg

Mean of absolute values of samples in moving window.

rms

Root squared mean of absolute values of samples in moving window.

peak

Maximum of absolute values of samples in moving window.

median

Median of absolute values of samples in moving window.

ptp

Absolute of max peak to min peak difference of samples in moving window.

dev

Standard deviation of values of samples in moving window.

Default value is rms.

window

Set duration in number of seconds used to calculate size of window in number of samples for detecting silence. Using 0 will effectively disable any windowing and use only single sample per channel for silence detection. In that case it may be needed to also set start_silence and/or stop_silence to nonzero values with also start_duration and/or stop_duration to nonzero values. Default value is 0.02. Allowed range is from 0 to 10.

timestamp

Set processing mode of every audio frame output timestamp.

write

Full timestamps rewrite, keep only the start time for the first output frame.

copy

Non-dropped frames are left with same timestamp as input audio frame.

Defaults value is write.

Examples

• The following example shows how this filter can be used to start a recording that does not contain the delay at the start which usually occurs between pressing the record button and the start of the performance:

          silenceremove=start_periods=1:start_duration=5:start_threshold=0.02

• Trim all silence encountered from beginning to end where there is more than 1 second of silence in audio:

          silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-90dB

• Trim all digital silence samples, using peak detection, from beginning to end where there is more than 0 samples of digital silence in audio and digital silence is detected in all channels at same positions in stream:

          silenceremove=window=0:detection=peak:stop_mode=all:start_mode=all:stop_periods=-1:stop_threshold=0

• Trim every 2nd encountered silence period from beginning to end where there is more than 1 second of silence per silence period in audio:

          silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB

• Similar as above, but keep maximum of 0.5 seconds of silence from each trimmed period:

          silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5

• Similar as above, but keep maximum of 1.5 seconds of silence from start of audio:

          silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5:start_periods=1:start_duration=1:start_silence=1.5:stop_threshold=-90dB

Commands

This filter supports some above options as commands.

SOFAlizer uses head-related transfer functions (HRTFs) to create virtual loudspeakers around the user for binaural listening via headphones (audio formats up to 9 channels supported). The HRTFs are stored in SOFA files (see <http://www.sofacoustics.org/> for a database). SOFAlizer is developed at the Acoustics Research Institute (ARI) of the Austrian Academy of Sciences.

To enable compilation of this filter you need to configure FFmpeg with --enable-libmysofa.

The filter accepts the following options:

sofa

Set the SOFA file used for rendering.

gain

Set gain applied to audio. Value is in dB. Default is 0.

rotation

Set rotation of virtual loudspeakers in deg. Default is 0.

elevation

Set elevation of virtual speakers in deg. Default is 0.

radius

Set distance in meters between loudspeakers and the listener with near-field HRTFs. Default is 1.

type

Set processing type. Can be time or freq. time is processing audio in time domain which is slow. freq is processing audio in frequency domain which is fast. Default is freq.

speakers

Set custom positions of virtual loudspeakers. Syntax for this option is: <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...]. Each virtual loudspeaker is described with short channel name following with azimuth and elevation in degrees. Each virtual loudspeaker description is separated by '|'. For example to override front left and front right channel positions use: 'speakers=FL 45 15|FR 345 15'. Descriptions with unrecognised channel names are ignored.

lfegain

Set custom gain for LFE channels. Value is in dB. Default is 0.

framesize

Set custom frame size in number of samples. Default is 1024. Allowed range is from 1024 to 96000. Only used if option type is set to freq.

normalize

Should all IRs be normalized upon importing SOFA file. By default is enabled.

interpolate

Should nearest IRs be interpolated with neighbor IRs if exact position does not match. By default is disabled.

minphase

Minphase all IRs upon loading of SOFA file. By default is disabled.

anglestep

Set neighbor search angle step. Only used if option interpolate is enabled.

radstep

Set neighbor search radius step. Only used if option interpolate is enabled.

Examples

• Using ClubFritz6 sofa file:

          sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1

• Using ClubFritz12 sofa file and bigger radius with small rotation:

          sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5

• Similar as above but with custom speaker positions for front left, front right, back left and back right and also with custom gain:

          "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"

Speech Normalizer.

This filter expands or compresses each half-cycle of audio samples (local set of samples all above or all below zero and between two nearest zero crossings) depending on threshold value, so audio reaches target peak value under conditions controlled by below options.

The filter accepts the following options:

peak, p

Set the expansion target peak value. This specifies the highest allowed absolute amplitude level for the normalized audio input. Default value is 0.95. Allowed range is from 0.0 to 1.0.

expansion, e

Set the maximum expansion factor. Allowed range is from 1.0 to 50.0. Default value is 2.0. This option controls maximum local half-cycle of samples expansion. The maximum expansion would be such that local peak value reaches target peak value but never to surpass it and that ratio between new and previous peak value does not surpass this option value.

compression, c

Set the maximum compression factor. Allowed range is from 1.0 to 50.0. Default value is 2.0. This option controls maximum local half-cycle of samples compression. This option is used only if threshold option is set to value greater than 0.0, then in such cases when local peak is lower or same as value set by threshold all samples belonging to that peak's half-cycle will be compressed by current compression factor.

threshold, t

Set the threshold value. Default value is 0.0. Allowed range is from 0.0 to 1.0. This option specifies which half-cycles of samples will be compressed and which will be expanded. Any half-cycle samples with their local peak value below or same as this option value will be compressed by current compression factor, otherwise, if greater than threshold value they will be expanded with expansion factor so that it could reach peak target value but never surpass it.

raise, r

Set the expansion raising amount per each half-cycle of samples. Default value is 0.001. Allowed range is from 0.0 to 1.0. This controls how fast expansion factor is raised per each new half-cycle until it reaches expansion value. Setting this options too high may lead to distortions.

fall, f

Set the compression raising amount per each half-cycle of samples. Default value is 0.001. Allowed range is from 0.0 to 1.0. This controls how fast compression factor is raised per each new half-cycle until it reaches compression value.

channels, h

Specify which channels to filter, by default all available channels are filtered.

invert, i

Enable inverted filtering, by default is disabled. This inverts interpretation of threshold option. When enabled any half-cycle of samples with their local peak value below or same as threshold option will be expanded otherwise it will be compressed.

link, l

Link channels when calculating gain applied to each filtered channel sample, by default is disabled. When disabled each filtered channel gain calculation is independent, otherwise when this option is enabled the minimum of all possible gains for each filtered channel is used.

rms, m

Set the expansion target RMS value. This specifies the highest allowed RMS level for the normalized audio input. Default value is 0.0, thus disabled. Allowed range is from 0.0 to 1.0.

Commands

This filter supports the all above options as commands.

Examples

• Weak and slow amplification:

          speechnorm=e=3:r=0.00001:l=1

• Moderate and slow amplification:

          speechnorm=e=6.25:r=0.00001:l=1

• Strong and fast amplification:

          speechnorm=e=12.5:r=0.0001:l=1

• Very strong and fast amplification:

          speechnorm=e=25:r=0.0001:l=1

• Extreme and fast amplification:

          speechnorm=e=50:r=0.0001:l=1

This filter has some handy utilities to manage stereo signals, for converting M/S stereo recordings to L/R signal while having control over the parameters or spreading the stereo image of master track.

The filter accepts the following options:

level_in

Set input level before filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.

level_out

Set output level after filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.

balance_in

Set input balance between both channels. Default is 0. Allowed range is from -1 to 1.

balance_out

Set output balance between both channels. Default is 0. Allowed range is from -1 to 1.

softclip

Enable softclipping. Results in analog distortion instead of harsh digital 0dB clipping. Disabled by default.

mutel

Mute the left channel. Disabled by default.

muter

Mute the right channel. Disabled by default.

phasel

Change the phase of the left channel. Disabled by default.

phaser

Change the phase of the right channel. Disabled by default.

mode

Set stereo mode. Available values are:

lr>lr

Left/Right to Left/Right, this is default.

lr>ms

Left/Right to Mid/Side.

ms>lr

Mid/Side to Left/Right.

lr>ll

Left/Right to Left/Left.

lr>rr

Left/Right to Right/Right.

lr>l+r

Left/Right to Left + Right.

lr>rl

Left/Right to Right/Left.

ms>ll

Mid/Side to Left/Left.

ms>rr

Mid/Side to Right/Right.

ms>rl

Mid/Side to Right/Left.

lr>l-r

Left/Right to Left - Right.

slev

Set level of side signal. Default is 1. Allowed range is from 0.015625 to 64.

sbal

Set balance of side signal. Default is 0. Allowed range is from -1 to 1.

mlev

Set level of the middle signal. Default is 1. Allowed range is from 0.015625 to 64.

mpan

Set middle signal pan. Default is 0. Allowed range is from -1 to 1.

base

Set stereo base between mono and inversed channels. Default is 0. Allowed range is from -1 to 1.

delay

Set delay in milliseconds how much to delay left from right channel and vice versa. Default is 0. Allowed range is from -20 to 20.

sclevel

Set S/C level. Default is 1. Allowed range is from 1 to 100.

phase

Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.

bmode_in, bmode_out

Set balance mode for balance_in/balance_out option.

Can be one of the following:

balance

Classic balance mode. Attenuate one channel at time. Gain is raised up to 1.

amplitude

Similar as classic mode above but gain is raised up to 2.

power

Equal power distribution, from -6dB to +6dB range.

Commands

This filter supports the all above options as commands.

Examples

• Apply karaoke like effect:

          stereotools=mlev=0.015625

• Convert M/S signal to L/R:

          "stereotools=mode=ms>lr"

This filter enhance the stereo effect by suppressing signal common to both channels and by delaying the signal of left into right and vice versa, thereby widening the stereo effect.

The filter accepts the following options:

delay

Time in milliseconds of the delay of left signal into right and vice versa. Default is 20 milliseconds.

feedback

Amount of gain in delayed signal into right and vice versa. Gives a delay effect of left signal in right output and vice versa which gives widening effect. Default is 0.3.

crossfeed

Cross feed of left into right with inverted phase. This helps in suppressing the mono. If the value is 1 it will cancel all the signal common to both channels. Default is 0.3.

drymix

Set level of input signal of original channel. Default is 0.8.

Commands

This filter supports the all above options except delay as commands.

Apply 18 band equalizer.

The filter accepts the following options:

1b

Set 65Hz band gain.

2b

Set 92Hz band gain.

3b

Set 131Hz band gain.

4b

Set 185Hz band gain.

5b

Set 262Hz band gain.

6b

Set 370Hz band gain.

7b

Set 523Hz band gain.

8b

Set 740Hz band gain.

9b

Set 1047Hz band gain.

10b

Set 1480Hz band gain.

11b

Set 2093Hz band gain.

12b

Set 2960Hz band gain.

13b

Set 4186Hz band gain.

14b

Set 5920Hz band gain.

15b

Set 8372Hz band gain.

16b

Set 11840Hz band gain.

17b

Set 16744Hz band gain.

18b

Set 20000Hz band gain.

Apply audio surround upmix filter.

This filter allows to produce multichannel output from audio stream.

The filter accepts the following options:

chl_out

Set output channel layout. By default, this is 5.1.

See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

chl_in

Set input channel layout. By default, this is stereo.

See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

level_in

Set input volume level. By default, this is 1.

level_out

Set output volume level. By default, this is 1.

lfe

Enable LFE channel output if output channel layout has it. By default, this is enabled.

lfe_low

Set LFE low cut off frequency. By default, this is 128 Hz.

lfe_high

Set LFE high cut off frequency. By default, this is 256 Hz.

lfe_mode

Set LFE mode, can be add or sub. Default is add. In add mode, LFE channel is created from input audio and added to output. In sub mode, LFE channel is created from input audio and added to output but also all non-LFE output channels are subtracted with output LFE channel.

smooth

Set temporal smoothness strength, used to gradually change factors when transforming stereo sound in time. Allowed range is from 0.0 to 1.0. Useful to improve output quality with focus option values greater than 0.0. Default is 0.0. Only values inside this range and without edges are effective.

angle

Set angle of stereo surround transform, Allowed range is from 0 to 360. Default is 90.

focus

Set focus of stereo surround transform, Allowed range is from -1 to 1. Default is 0.

fc_in

Set front center input volume. By default, this is 1.

fc_out

Set front center output volume. By default, this is 1.

fl_in

Set front left input volume. By default, this is 1.

fl_out

Set front left output volume. By default, this is 1.

fr_in

Set front right input volume. By default, this is 1.

fr_out

Set front right output volume. By default, this is 1.

sl_in

Set side left input volume. By default, this is 1.

sl_out

Set side left output volume. By default, this is 1.

sr_in

Set side right input volume. By default, this is 1.

sr_out

Set side right output volume. By default, this is 1.

bl_in

Set back left input volume. By default, this is 1.

bl_out

Set back left output volume. By default, this is 1.

br_in

Set back right input volume. By default, this is 1.

br_out

Set back right output volume. By default, this is 1.

bc_in

Set back center input volume. By default, this is 1.

bc_out

Set back center output volume. By default, this is 1.

lfe_in

Set LFE input volume. By default, this is 1.

lfe_out

Set LFE output volume. By default, this is 1.

allx

Set spread usage of stereo image across X axis for all channels. Allowed range is from -1 to 15. By default this value is negative -1, and thus unused.

ally

Set spread usage of stereo image across Y axis for all channels. Allowed range is from -1 to 15. By default this value is negative -1, and thus unused.

fcx, flx, frx, blx, brx, slx, srx, bcx

Set spread usage of stereo image across X axis for each channel. Allowed range is from 0.06 to 15. By default this value is 0.5.

fcy, fly, fry, bly, bry, sly, sry, bcy

Set spread usage of stereo image across Y axis for each channel. Allowed range is from 0.06 to 15. By default this value is 0.5.

win_size

Set window size. Allowed range is from 1024 to 65536. Default size is 4096.

win_func

Set window function.

It accepts the following values:

rect

bartlett

hann, hanning

hamming

blackman

welch

flattop

bharris

bnuttall

bhann

sine

nuttall

lanczos

gauss

tukey

dolph

cauchy

parzen

poisson

bohman

kaiser
Default is hann.

overlap

Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. Default is 0.5.

Boost or cut the lower frequencies and cut or boost higher frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

gain, g

Give the gain at 0 Hz. Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of clipping when using a positive gain.

frequency, f

Set the filter's central frequency and so can be used to extend or reduce the frequency range to be boosted or cut. The default value is 3000 Hz.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Determine how steep is the filter's shelf transition.

poles, p

Set number of poles. Default is 2.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Commands

This filter supports some options as commands.

Boost or cut treble (upper) frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

gain, g

Give the gain at whichever is the lower of ~22 kHz and the Nyquist frequency. Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of clipping when using a positive gain.

frequency, f

Set the filter's central frequency and so can be used to extend or reduce the frequency range to be boosted or cut. The default value is 3000 Hz.

width_type, t

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

k

kHz

width, w

Determine how steep is the filter's shelf transition.

poles, p

Set number of poles. Default is 2.

mix, m

How much to use filtered signal in output. Default is 1. Range is between 0 and 1.

channels, c

Specify which channels to filter, by default all available are filtered.

normalize, n

Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.

transform, a

Set transform type of IIR filter.

di

dii

tdi

tdii

latt

svf

zdf

precision, r

Set precision of filtering.

auto

Pick automatic sample format depending on surround filters.

s16

Always use signed 16-bit.

s32

Always use signed 32-bit.

f32

Always use float 32-bit.

f64

Always use float 64-bit.

block_size, b

Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.

Note that filter delay will be exactly this many samples when set to non-zero value.

Commands

This filter supports the following commands:

frequency, f

Change treble frequency. Syntax for the command is : "frequency"

width_type, t

Change treble width_type. Syntax for the command is : "width_type"

width, w

Change treble width. Syntax for the command is : "width"

gain, g

Change treble gain. Syntax for the command is : "gain"

mix, m

Change treble mix. Syntax for the command is : "mix"

Sinusoidal amplitude modulation.

The filter accepts the following options:

f

Modulation frequency in Hertz. Modulation frequencies in the subharmonic range (20 Hz or lower) will result in a tremolo effect. This filter may also be used as a ring modulator by specifying a modulation frequency higher than 20 Hz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.

d

Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

Sinusoidal phase modulation.

The filter accepts the following options:

f

Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.

d

Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

Apply audio Virtual Bass filter.

This filter accepts stereo input and produce stereo with LFE (2.1) channels output. The newly produced LFE channel have enhanced virtual bass originally obtained from both stereo channels. This filter outputs front left and front right channels unchanged as available in stereo input.

The filter accepts the following options:

cutoff

Set the virtual bass cutoff frequency. Default value is 250 Hz. Allowed range is from 100 to 500 Hz.

strength

Set the virtual bass strength. Allowed range is from 0.5 to 3. Default value is 3.

Adjust the input audio volume.

It accepts the following parameters:

volume

Set audio volume expression.

Output values are clipped to the maximum value.

The output audio volume is given by the relation:

        <output_volume> = <volume> * <input_volume>

The default value for volume is "1.0".

precision

This parameter represents the mathematical precision.

It determines which input sample formats will be allowed, which affects the precision of the volume scaling.

fixed

8-bit fixed-point; this limits input sample format to U8, S16, and S32.

float

32-bit floating-point; this limits input sample format to FLT. (default)

double

64-bit floating-point; this limits input sample format to DBL.

replaygain

Choose the behaviour on encountering ReplayGain side data in input frames.

drop

Remove ReplayGain side data, ignoring its contents (the default).

ignore

Ignore ReplayGain side data, but leave it in the frame.

track

Prefer the track gain, if present.

album

Prefer the album gain, if present.

replaygain_preamp

Pre-amplification gain in dB to apply to the selected replaygain gain.

Default value for replaygain_preamp is 0.0.

replaygain_noclip

Prevent clipping by limiting the gain applied.

Default value for replaygain_noclip is 1.

eval

Set when the volume expression is evaluated.

It accepts the following values:

once

only evaluate expression once during the filter initialization, or when the volume command is sent

frame

evaluate expression for each incoming frame

Default value is once.

The volume expression can contain the following parameters.

n

frame number (starting at zero)

nb_channels

number of channels

nb_consumed_samples

number of samples consumed by the filter

nb_samples

number of samples in the current frame

pos

original frame position in the file; deprecated, do not use

pts

frame PTS

sample_rate

sample rate

startpts

PTS at start of stream

startt

time at start of stream

t

frame time

tb

timestamp timebase

volume

last set volume value

Note that when eval is set to once only the sample_rate and tb variables are available, all other variables will evaluate to NAN.

Commands

This filter supports the following commands:

volume

Modify the volume expression. The command accepts the same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current value.

Examples

• Halve the input audio volume:

          volume=volume=0.5
          volume=volume=1/2
          volume=volume=-6.0206dB

  In all the above example the named key for volume can be omitted, for example like in:

          volume=0.5

• Increase input audio power by 6 decibels using fixed-point precision:

          volume=volume=6dB:precision=fixed

• Fade volume after time 10 with an annihilation period of 5 seconds:

          volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame

Detect the volume of the input video.

The filter has no parameters. It supports only 16-bit signed integer samples, so the input will be converted when needed. Statistics about the volume will be printed in the log when the input stream end is reached.

In particular it will show the mean volume (root mean square), maximum volume (on a per-sample basis), and the beginning of a histogram of the registered volume values (from the maximum value to a cumulated 1/1000 of the samples).

All volumes are in decibels relative to the maximum PCM value.

Examples

Here is an excerpt of the output:

        [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
        [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
        [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
        [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
        [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
        [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
        [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
        [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
        [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

It means that:

• The mean square energy is approximately -27 dB, or 10^-2.7.
• The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
• There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

In other words, raising the volume by +4 dB does not cause any clipping, raising it by +5 dB causes clipping for 6 samples, etc.

Buffer audio frames, and make them available to the filter chain.

This source is mainly intended for a programmatic use, in particular through the interface defined in libavfilter/buffersrc.h.

It accepts the following parameters:

time_base

The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.

sample_rate

The sample rate of the incoming audio buffers.

sample_fmt

The sample format of the incoming audio buffers. Either a sample format name or its corresponding integer representation from the enum AVSampleFormat in libavutil/samplefmt.h

channel_layout

The channel layout of the incoming audio buffers. Either a channel layout name from channel_layout_map in libavutil/channel_layout.c or its corresponding integer representation from the AV_CH_LAYOUT_* macros in libavutil/channel_layout.h

channels

The number of channels of the incoming audio buffers. If both channels and channel_layout are specified, then they must be consistent.

Examples

        abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

will instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample format with name "s16p" corresponds to the number 6 and the "stereo" channel layout corresponds to the value 0x3, this is equivalent to:

        abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

Generate an audio signal specified by an expression.

This source accepts in input one or more expressions (one for each channel), which are evaluated and used to generate a corresponding audio signal.

This source accepts the following options:

exprs

Set the '|'-separated expressions list for each separate channel. In case the channel_layout option is not specified, the selected channel layout depends on the number of provided expressions. Otherwise the last specified expression is applied to the remaining output channels.

channel_layout, c

Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.

duration, d

Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame.

If not specified, or the expressed duration is negative, the audio is supposed to be generated forever.

nb_samples, n

Set the number of samples per channel per each output frame, default to 1024.

sample_rate, s

Specify the sample rate, default to 44100.

Each expression in exprs can contain the following constants:

n

number of the evaluated sample, starting from 0

t

time of the evaluated sample expressed in seconds, starting from 0

s

sample rate

Examples

• Generate silence:

          aevalsrc=0

• Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:

          aevalsrc="sin(440*2*PI*t):s=8000"

• Generate a two channels signal, specify the channel layout (Front Center + Back Center) explicitly:

          aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

• Generate white noise:

          aevalsrc="-2+random(0)"

• Generate an amplitude modulated signal:

          aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

• Generate 2.5 Hz binaural beats on a 360 Hz carrier:

          aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

Generate a fractional delay FIR coefficients.

The resulting stream can be used with afir filter for filtering the audio signal.

The filter accepts the following options:

delay, d

Set the fractional delay. Default is 0.

sample_rate, r

Set the sample rate, default is 44100.

nb_samples, n

Set the number of samples per each frame. Default is 1024.

taps, t

Set the number of filter coefficients in output audio stream. Default value is 0.

channel_layout, c

Specifies the channel layout, and can be a string representing a channel layout. The default value of channel_layout is "stereo".

Generate a FIR equalizer coefficients.

The resulting stream can be used with afir filter for filtering the audio signal.

The filter accepts the following options:

preset, p

Set equalizer preset. Default preset is flat.

Available presets are:

custom

flat

acoustic

bass

beats

classic

clear

deep bass

dubstep

electronic

hard-style

hip-hop

jazz

metal

movie

pop

r&b

rock

vocal booster

gains, g

Set custom gains for each band. Only used if the preset option is set to custom. Gains are separated by white spaces and each gain is set in dBFS. Default is 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0.

bands, b

Set the custom bands from where custon equalizer gains are set. This must be in strictly increasing order. Only used if the preset option is set to custom. Bands are separated by white spaces and each band represent frequency in Hz. Default is 25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000.

taps, t

Set number of filter coefficients in output audio stream. Default value is 4096.

sample_rate, r

Set sample rate of output audio stream, default is 44100.

nb_samples, n

Set number of samples per each frame in output audio stream. Default is 1024.

interp, i

Set interpolation method for FIR equalizer coefficients. Can be linear or cubic.

phase, h

Set phase type of FIR filter. Can be linear or min: minimum-phase. Default is minimum-phase filter.

Generate a FIR coefficients using frequency sampling method.

The resulting stream can be used with afir filter for filtering the audio signal.

The filter accepts the following options:

taps, t

Set number of filter coefficients in output audio stream. Default value is 1025.

frequency, f

Set frequency points from where magnitude and phase are set. This must be in non decreasing order, and first element must be 0, while last element must be 1. Elements are separated by white spaces.

magnitude, m

Set magnitude value for every frequency point set by frequency. Number of values must be same as number of frequency points. Values are separated by white spaces.

phase, p

Set phase value for every frequency point set by frequency. Number of values must be same as number of frequency points. Values are separated by white spaces.

sample_rate, r

Set sample rate, default is 44100.

nb_samples, n

Set number of samples per each frame. Default is 1024.

win_func, w

Set window function. Default is blackman.

The null audio source, return unprocessed audio frames. It is mainly useful as a template and to be employed in analysis / debugging tools, or as the source for filters which ignore the input data (for example the sox synth filter).

This source accepts the following options:

channel_layout, cl

Specifies the channel layout, and can be either an integer or a string representing a channel layout. The default value of channel_layout is "stereo".

Check the channel_layout_map definition in libavutil/channel_layout.c for the mapping between strings and channel layout values.

sample_rate, r

Specifies the sample rate, and defaults to 44100.

nb_samples, n

Set the number of samples per requested frames.

duration, d

Set the duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.

If not specified, or the expressed duration is negative, the audio is supposed to be generated forever.

Examples

• Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.

          anullsrc=r=48000:cl=4

• Do the same operation with a more obvious syntax:

          anullsrc=r=48000:cl=mono

All the parameters need to be explicitly defined.

Synthesize a voice utterance using the libflite library.

To enable compilation of this filter you need to configure FFmpeg with --enable-libflite.

Note that versions of the flite library prior to 2.0 are not thread-safe.

The filter accepts the following options:

list_voices

If set to 1, list the names of the available voices and exit immediately. Default value is 0.

nb_samples, n

Set the maximum number of samples per frame. Default value is 512.

textfile

Set the filename containing the text to speak.

text

Set the text to speak.

voice, v

Set the voice to use for the speech synthesis. Default value is kal. See also the list_voices option.

Examples

• Read from file speech.txt, and synthesize the text using the standard flite voice:

          flite=textfile=speech.txt

• Read the specified text selecting the slt voice:

          flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

• Input text to ffmpeg:

          ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

• Make ffplay speak the specified text, using flite and the lavfi device:

          ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

For more information about libflite, check: <http://www.festvox.org/flite/>

Generate a noise audio signal.

The filter accepts the following options:

sample_rate, r

Specify the sample rate. Default value is 48000 Hz.

amplitude, a

Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.

duration, d

Specify the duration of the generated audio stream. Not specifying this option results in noise with an infinite length.

color, colour, c

Specify the color of noise. Available noise colors are white, pink, brown, blue, violet and velvet. Default color is white.

seed, s

Specify a value used to seed the PRNG.

nb_samples, n

Set the number of samples per each output frame, default is 1024.

density

Set the density (0.0 - 1.0) for the velvet noise generator, default is 0.05.

Examples

• Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:

          anoisesrc=d=60:c=pink:r=44100:a=0.5

Generate odd-tap Hilbert transform FIR coefficients.

The resulting stream can be used with afir filter for phase-shifting the signal by 90 degrees.

This is used in many matrix coding schemes and for analytic signal generation. The process is often written as a multiplication by i (or j), the imaginary unit.

The filter accepts the following options:

sample_rate, s

Set sample rate, default is 44100.

taps, t

Set length of FIR filter, default is 22051.

nb_samples, n

Set number of samples per each frame.

win_func, w

Set window function to be used when generating FIR coefficients.

Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients.

The resulting stream can be used with afir filter for filtering the audio signal.

The filter accepts the following options:

sample_rate, r

Set sample rate, default is 44100.

nb_samples, n

Set number of samples per each frame. Default is 1024.

hp

Set high-pass frequency. Default is 0.

lp

Set low-pass frequency. Default is 0. If high-pass frequency is lower than low-pass frequency and low-pass frequency is higher than 0 then filter will create band-pass filter coefficients, otherwise band-reject filter coefficients.

phase

Set filter phase response. Default is 50. Allowed range is from 0 to 100.

beta

Set Kaiser window beta.

att

Set stop-band attenuation. Default is 120dB, allowed range is from 40 to 180 dB.

round

Enable rounding, by default is disabled.

hptaps

Set number of taps for high-pass filter.

lptaps

Set number of taps for low-pass filter.

Generate an audio signal made of a sine wave with amplitude 1/8.

The audio signal is bit-exact.

The filter accepts the following options:

frequency, f

Set the carrier frequency. Default is 440 Hz.

beep_factor, b

Enable a periodic beep every second with frequency beep_factor times the carrier frequency. Default is 0, meaning the beep is disabled.

sample_rate, r

Specify the sample rate, default is 44100.

duration, d

Specify the duration of the generated audio stream.

samples_per_frame

Set the number of samples per output frame.

The expression can contain the following constants:

n

The (sequential) number of the output audio frame, starting from 0.

pts

The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.

t

The PTS of the output audio frame, expressed in seconds.

TB

The timebase of the output audio frames.

Default is 1024.