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Author pinterf v2.2

sh0dan v1.1 beta

Version v2.2
Download SoxFilter 2 releases
Category Audio filters
License GPLv2
Discussion Doom9 Forum
Doom9 Forum (v1.1 x64 version)


[edit] Description

This plugin will allow you to run SOX effects within AviSynth. Most effects are supported, and multiple effects can be stacked after each other.
Version 1.1 is based on SoX library 12.17.9
Version 2.0 is based on SoX library 14.4.2
SoX library works with 32 bit integer samples internally. All inputs will be automatically converted to this audio format.

[edit] Requirements

[edit] v2.x

[edit] v1.1

[edit] Syntax and Parameters

SoxFilter (clip, string effect1 [, string effect 2, string effect 3...])

clip   =
Input clip.

string   =
Any number of effects can be entered, and they will be executed in the order they are specified. Effect syntax is just like SOX. See Effect Overview and Filter Reference below.

[edit] Usage

A simple filter could look like this:

SoxFilter("bandpass 500 100")

This will keep a 100Hz band around 500Hz. SoxFilter converts audio to 32 bit integers. This allows to keep the additional dynamic range of float point samples, but it requires a convertion to 16 bit audio before output, since most codecs only support 16 bit.

Multiple effects can be stacked like this:

SoxFilter("bandpass 2000 1000", "vol 2.0", "reverb 1.0 600.0 180.0 200.0 220.0 240.0")

Which is a faster version of:

SoxFilter("bandpass 2000 1000")
SoxFilter("vol 2.0")
SoxFilter("reverb 1.0 600.0 180.0 200.0 220.0 240.0")

[edit] AviSynth Specifics

[edit] Remarks

Since v1.1 release SoX library internal API has been changed. v2.x was created from scratch. Also, some effects were removed, and others added in the past ~15 years. The parameters of some effects had been changed as well.

E.g. reverb parameters were modified in 2008; "filter" was removed, use "sinc" instead (with a different syntax).

Check SoxFilter_usages.txt in v2.x release pack, or visit for effect parameters and their actual description.

[edit] Known Issues

  • Linear access is heavily recommended. SOX filters doesn't support seeking, so stream is restarted every time a samples previous to the last one is requested.
  • Do not use Avisynth+ versions before 3.7.3 if filter result is used later in the script multiple times. Audio cache (which was reintroduced in Avisynth+ 3.7.3) is a must have in those cases.
  • (v2.1-) none of filters were removed from 14.4.2 SoX library, still, some may be incompatible. Filters can even change sample rate or alter the number of channels. In v1.1 these filters were not allowed.
  • (v1.1 only) The AviSynth version is VERY picky about spaces. If you make double spaces or a space before or after the quotes the command will not be recognized.
  • (v1.1 only) If one filter doesn't support multichannel audio the entire chain is converted to mono filters, this might affect some filters like earwax. Use multiple filter instances instead.
  • (v1.1 only) Some filters have been removed since they were incompatible with the AviSynth version. All these filters have internal Avisynth equivalents.
  • (v1.1 only) compand is very unstable in the current interface, and doesn't support restarts (distorted audio). Don't use this together with other effetcs.
  • (v1.1 only) The length of audio cannot be modified.

[edit] Version 1.1

  • Solved issue with earwax refusing to play.
  • Solved problem with time alternating effects.
  • Solved issues with some filters producing clicks.
  • Made sure compand and mcompand doesn't crash. Still quite buggy though.

[edit] Effect Overview

This section contains effects supported in SoxFilter v1.1 (12.17.9 SoX Library version).

For effects supported by SoxFilter v2.x (SoX Library 14.4.2) visit for parameters and their actual description.

           band [ -n ] center [ width ]
           bandpass frequency bandwidth

           bandreject frequency bandwidth
           chorus gain-in gain out delay decay speed depth

                  -s | -t [ delay decay speed depth -s | -t ]
           compand attack1,decay1[,attack2,decay2...]
                   [ gain [ initial-volume [ delay ] ] ]

           dcshift shift [ limitergain ]
           echo gain-in gain-out delay decay [ delay decay ... ]
           echos gain-in gain-out delay decay [ delay decay ... ]
           fade [ type ] fade-in-length

                [ stop-time [ fade-out-length ] ]
           filter [ low ]-[ high ] [ window-len [ beta ]]
           flanger gain-in gain-out delay decay speed < -s | -t >

           highp frequency
           highpass frequency
           lowp frequency
           lowpass frequency

           mcompand "attack1,decay1[,attack2,decay2...]
              [ gain [ initial-volume [ delay ] ] ]" xover_freq
           noiseprof [profile-file]
           noisered profile-file [threshold]
           phaser gain-in gain-out delay decay speed < -s | -t >

           pitch shift [ width interpole fade ]
           reverb gain-out reverb-time delay [ delay ... ]
           silence above_periods [ duration threshold[ d | % ]
              [ below_periods duration
              threshold[ d | % ]]
           speed [ -c ] factor
           stretch [ factor [ window fade shift fading ]
           swap [ 1 2 | 1 2 3 4 ]
           synth [ length ] type mix [ freq [ -freq2 ]
                 [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
           vibro speed [ depth ]
           vol gain [ type [ limitergain ] ]

[edit] Filter Reference

This section contains effects supported in SoxFilter v1.1 (12.17.9 SoX Library version).

For effects supported by SoxFilter v2.x (SoX Library 14.4.2) visit for parameters and their actual description.

       band [ -n ] center [ width ]
                 Apply a band-pass filter.  The frequency response drops loga-
                 rithmically around the center frequency.  The width gives the
                 slope of the drop.  The frequencies at  center  +  width  and
                 center  -  width  will  be half of their original amplitudes.
                 Band defaults to a mode oriented  to  pitched  signals,  i.e.
                 voice,  singing,  or  instrumental music.  The -n (for noise)
                 option uses the alternate mode for un-pitched signals.  Warn-
                 ing:  -n introduces a power-gain of about 11dB in the filter,
                 so beware of output clipping.  Band introduces noise  in  the
                 shape of the filter, i.e. peaking at the center frequency and
                 settling around it.  See filter for a  bandpass  effect  with
                 steeper shoulders.

       bandpass frequency bandwidth
                 Butterworth bandpass filter. Description coming soon!

       bandreject frequency bandwidth
                 Butterworth bandreject filter.  Description coming soon!

       chorus gain-in gain-out delay decay speed depth

              -s | -t [ delay decay speed depth -s | -t ... ]
                 which the absolute value of the input signal is integrated to
                 determine  its  volume;  attacks refer to increases in volume
                 and decays refer to decreases.  Where more than one  pair  of
                 attack/decay   parameters  are  specified,  each  channel  is
                 treated separately and the number of pairs  must  agree  with
                 the number of input channels.  The second parameter is a list
                 of points on the compander's transfer function  specified  in
                 dB  relative  to  the maximum possible signal amplitude.  The
                 input values must be in a strictly increasing order  but  the
                 transfer  function  does not have to be monotonically rising.
                 The special value -inf may be used to indicate that the input
                 volume  should  be  associated  output  volume.   The  points
                 -inf,-inf and 0,0 are assumed; the latter may be  overridden,
                 but the former may not.

                 The  third  (optional) parameter is a post-processing gain in
                 dB which is applied after the compression  has  taken  place;
                 the  fourth  (optional)  parameter is an initial volume to be
                 assumed for each channel when the effect starts.   This  per-
                 mits  the  user to supply a nominal level initially, so that,
                 for example, a very large gain is not applied to initial sig-
                 nal levels before the companding action has begun to operate:
                 it is quite probable that in such an event, the output  would
                 be severely clipped while the compander gain properly adjusts

                 The fifth (optional) parameter is a delay  in  seconds.   The
                 input  signal  is analyzed immediately to control the compan-
                 der, but it  is  delayed  before  being  fed  to  the  volume
                 adjuster.   Specifying  a  delay  approximately  equal to the
                 attack/decay times allows the compander to effectively  oper-
                 ate in a "predictive" rather than a reactive mode.

       dcshift shift [ limitergain ]
                 DC Shift the audio data, with basic linear amplitude formula.
                 This  is  most useful if your audio data tends to not be cen-
                 tered around a value of 0.  Shifting it back will  allow  you
                 to  get  the  most  volume adjustments without clipping audio
                 The first option is the dcshift  value.   It  is  a  floating
                 point number that indicates the amount to shift.
                 An  option  limtergain  value  can  be specified as well.  It
                 should have a value much less then 1.0 and is  used  only  on
                 peaks to prevent clipping.

       deemph    Apply  a  treble  attenuation  shelving  filter to samples in
                 audio cd format.  The frequency  response  of  pre-emphasized
                 recordings  is  rectified.   The  filtering is defined in the
                 standard document ISO 908.
                 Add a sequence of echos to a sound sample.  Each  delay/decay
                 part  gives the delay in milliseconds and the decay (relative
                 to gain-in) of that echo.  Gain-out is the volume of the out-
       earwax      Makes    sound  easier to listen to on headphones.  Adds audio-
                 cues to samples in audio cd format 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 lis-
                 tener (standard for speakers). See
        for a full explanation.

       echo gain-in gain-out delay decay [ delay decay ... ]
                 Add  echoing  to a sound sample.  Each delay/decay part gives
                 the delay in milliseconds and the decay (relative to gain-in)
                 of that echo.  Gain-out is the volume of the output.

       echos gain-in gain-out delay decay [ delay decay ... ]
                 Add  a sequence of echos to a sound sample.  Each delay/decay
                 part gives the delay in milliseconds and the decay  (relative
                 to gain-in) of that echo.  Gain-out is the volume of the out-

       fade [ type ] fade-in-length
            [ stop-time [ fade-out-length ] ]
                 Add a fade effect to the beginning, end, or both of the audio

                 For fade-ins, this starts from the first sample and ramps the
                 volume of the audio from 0 to full volume over fade-in-length
                 seconds.  Specify 0 seconds if no fade-in is wanted.

                 For fade-outs, the audio data will be truncated at the  stop-
                 time and the volume will be ramped from full volume down to 0
                 starting at fade-out-length seconds before the stop-time.  No
                 fade-out is performed if these options are not specified.
                 All  times can be specified in either periods of time or sam-
                 ple  counts.   To  specify  time  periods  use   the   format
                 hh:mm:ss.frac  format.  To specify using sample counts, spec-
                 ify the number of samples and append the letter  's'  to  the
                 sample count (for example 8000s).
                 An optional type can be specified to change the type of enve-
                 lope.  Choices are q for quarter of a sinewave, h for half  a
                 sinewave,  t  for  linear slope, l for logarithmic, and p for
                 inverted parabola.  The default is a linear slope.

       filter [ low ]-[ high ] [ window-len [ beta ] ]
                 Apply a Sinc-windowed lowpass, highpass, or  bandpass  filter
                 of given window length to the signal.  low refers to the fre-
                 quency of the lower 6dB corner of the filter.  high refers to
                 the frequency of the upper 6dB corner of the filter.

                 A  lowpass  filter is obtained by leaving low unspecified, or
                 0.  A highpass filter is obtained by  leaving  high  unspeci-
                 fied,  or  0,  or  greater  than or equal to the Nyquist fre-

                 The window-len, if unspecified, defaults to 128.  Longer win-
                 dows  give  a  sharper cutoff, smaller windows a more gradual

                 The beta, if unspecified, defaults to  16.   This  selects  a
                 Kaiser window.  You can select a Nuttall window by specifying
                 anything <= 2.0 here.  For  more  discussion  of  beta,  look
                 under the resample effect.

       flanger gain-in gain-out delay decay speed < -s | -t >
                 Add   a   flanger   to   a   sound   sample.    Each   triple
                 delay/decay/speed  gives  the  delay  in milliseconds and the
                 decay (relative to gain-in) with a modulation  speed  in  Hz.
                 The  modulation  is  either sinodial (-s) or triangular (-t).
                 Gain-out is the volume of the output.
       highp frequency
                 Apply a single pole recursive  high-pass  filter.   The  fre-
                 quency response drops logarithmically with I frequency in the
                 middle of the drop.  The slope of the filter is quite gentle.
                 See filter for a highpass effect with sharper cutoff.

       highpass frequency
                 Butterworth highpass filter.  Description coming soon!

        lowp frequency
                 Apply a single pole recursive low-pass filter.  The frequency
                 response  drops  logarithmically with frequency in the middle
                 of the drop.  The slope of the filter is quite  gentle.   See
                 filter for a lowpass effect with sharper cutoff.

       lowpass frequency
                 Butterworth lowpass filter.  Description coming soon!

       mask      Add "masking noise" to signal.  This effect deliberately adds
                 white noise to a sound in order to mask quantization effects,
                 created by the process of  playing  a  sound  digitally.   It
                 tends  to  mask buzzing voices, for example.  It adds 1/2 bit
                 of noise to the sound file at the output bit depth.

       mcompand "attack1,decay1[,attack2,decay2...]
                 [gain [initial-volume [delay ] ] ]" xover_freq

                 Multi-band compander is similar to the single band  compander
                 but  the  audio  file is first divided up into bands and then
                 the compander is ran on each band.  See  the  compand  effect
                 for definition of its options.  Compand options are specified
                 between double quotes and the crossover  frequency  for  that
                 band  is  specefied  seperately  with xover_fre.  This can be
                 repeated multiple times to create multiple bands.

       noiseprof [profile-file]

       noisered profile-file [threshold]
                Noise reduction filter with profiling. This filter is   moder-
                ately   effective at removing consistent background noise such
                as hiss or hum. To use it, first run the noiseprof effect  on
                a section of silence (that is, a section which contains noth-
                ing but noise). The noiseprof effect will print a noise  pro-
                file  to  profile-file,  or  to  stdout if no profile-file is
                specified.  If there is sound output on stdout then the  pro-
                file will instead be directed to stderr.

                To actually remove the noise, run SoX again with the noisered
                filter. The filter needs one  argument,  profile-file,   which
                contains  the   noise profile from noiseprof. thershold speci-
                fies how much noise should be removed, and may be  between  0
                and  1   with a default of 0.5. Higher values will remove more
                noise but present a greater  possibility  of  distorting  the
                desired  audio   signal.   Experiment with different threshold
                values to find the optimal one for your sample.

       phaser gain-in gain-out delay decay speed < -s | -t >
                 Add   a   phaser   to   a   sound   sample.    Each    triple
                 delay/decay/speed  gives  the  delay  in milliseconds and the
                 decay (relative to gain-in) with a modulation  speed  in  Hz.
                 The  modulation  is  either sinodial (-s) or triangular (-t).
                 The decay should be less than 0.5 to avoid  feedback.   Gain-
                 out is the volume of the output.

       pitch shift [ width interpole fade ]
                 Change  the  pitch  of file without affecting its duration by
                 cross-fading shifted samples.  shift is given in cents. Use a
                 positive value to shift to treble, negative value to shift to
                 bass.  Default shift is 0.  width of window is in ms. Default
                 width  is  20ms.  Try  30ms to lower pitch, and 10ms to raise
                 -w  <  nut / ham > : select either a Nuttal (~90 dB stopband)
                 or Hamming (~43 dB stopband) window.  Default is nut.

                 -width long / short / # : specify the (approximate) width  of
                 the  filter.   long  is  1024  samples; short is 128 samples.
                 Alternatively, an exact number can be used.  Default is long.
                 The  short  option  is  not  recommended, as it produces poor
                 quality results.

                 -cutoff # : specify the filter cutoff frequency in  terms  of
                 fraction  of  frequency  bandwidth,  also know as the Nyquist
                 frequency.  Please see the resample effect for further infor-
                 mation on Nyquist frequency.  If upsampling, then this is the
                 fraction of the original signal that should go  through.   If
                 downsampling,  this  is the fraction of the signal left after
                 downsampling.  Default is 0.95.   Remember  that  this  is  a

       reverb gain-out reverbe-time delay [ delay ... ]
                 Add  reverberation to a sound sample.  Each delay is given in
                 milliseconds and its feedback is depending on the reverb-time
                 in  milliseconds.   Each delay should be in the range of half
                 to quarter of reverb-time to get a  realistic  reverberation.
                 Gain-out is the volume of the output.

       silence above_periods [ duration threshold[ d | % ]
               [ below_periods duration threshold[ d | % ]]
                 Removes silence from the beginning or end of  a  sound  file.
                 Silence is anything below a specified threshold.
                 When trimming silence from the beginning of a sound file, you
                 specify a duration of audio that is  above  a  given  silence
                 threshold before audio data is processed.  You can also spec-
                 ify the count of periods of none silence you want  to  detect
                 before  processing  audio data.  Specify a period of 0 if you
                 do not want to trim data from the front of the sound file.
                 When optionally trimming silence form  the  end  of  a  sound
                 file,  you specify the duration of audio that must be below a
                 given threshold before stopping to  process  audio  data.   A
                 count  of  periods that occur below the threshold may also be
                 specified.  If this options are not specified  then  data  is
                 not trimmed from the end of the audio file.
                 Duration  counts may be in the format of time, hh:mm:ss.frac,
                 or in the exact count of samples.
                 Threshold may be suffixed with d, or % to indicated the value
                 is  in  decibels  or  a percentage of max value of the sample
                 value.  A value of '0%' will look for total silence.

       speed [ -c ] factor
                 Speed up or down the sound, as a magnetic tape with  a  speed
                 control.   It  affects  both  pitch and time. A factor of 1.0
                 means no change, and is the default.  2.0 doubles speed, thus
                 time  length is cut by a half and pitch is one octave higher.
                 0.5 halves speed thus time length doubles and  pitch  is  one
                 octave  lower.  If the optional -c parameter is used then the
                 factor is specified in "cents".

       stretch factor [window fade shift fading]
                 Time  stretch file by a given factor. Change duration without
                 affecting the pitch.  factor of  stretching:  >1.0  lengthen,
                 <1.0  shorten  duration.   window  size  is in ms. Default is
                 20ms. The fade option, can be "lin".  shift  ratio,  in  [0.0
                 1.0].  Default depends on stretch factor. 1.0 to shorten, 0.8
                 to lengthen.  The fading ratio, in [0.0 0.5]. The amount of a
                 fade's default depends on factor and shift.

       synth [ length ] type mix [ freq [ -freq2 ]
             [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
                 The  synth  effect will generate various types of audio data.
                 Although this effect is used to generate audio data, an input
                 file  must  be specified.  The length of the input audio file
                 determines the length of the output audio file.
                 <length>  length  in  sec  or  hh:mm:ss.frac,  0=inputlength,
                 <type>  is  sine,  square,  triangle, sawtooth, trapetz, exp,
                 whitenoise, pinknoise, brownnoise, default=sine
                 <mix> is create, mix, amod, default=create
                 <freq> frequency at beginning in Hz, not used  for noise..
                 <freq2>  frequency  at  end  in  Hz,  not  used  for  noise..
                 <freq/2> can be given as %%n, where 'n' is the number of half
                 notes in respect to A (440Hz)
                 <off> Bias (DC-offset)  of signal in percent, default=0
                 <ph> phase shift 0..100 shift phase  0..2*Pi,  not  used  for
                 <p1>  square:  Ton/Toff,  triangle+trapetz: rising slope time
                 <p2> trapetz: ON time (0..100)
                 the  audio  data.  The format for specifying sample counts is
                 the number of samples with the letter 's' appended to it.   A
                 value  of  8000s will wait until 8000 samples are read before
                 starting to process audio data.

       vibro speed  [ depth ]
                 Add the world-famous Fender Vibro-Champ  sound  effect  to  a
                 sound  sample by using a sine wave as the volume knob.  Speed
                 gives the Hertz value of the wave.  This must  be  under  30.
                 Depth  gives  the  amount  the volume is cut into by the sine
                 wave, ranging 0.0 to 1.0 and defaulting to 0.5.

       vol gain [ type [ limitergain ] ]
                 The vol effect is much like the command line option  -v.   It
                 allows  you  to adjust the volume of an input file and allows
                 you to specify  the  adjustment  in  relation  to  amplitude,
                 power,  or  dB.  If type is not specified then it defaults to
                 When type is amplitude then a linear change of the  amplitude
                 is  performed  based  on the gain.  Therefore, a value of 1.0
                 will keep the volume the same, 0.0 to < 1.0  will  cause  the
                 volume  to decrease and values of > 1.0 will cause the volume
                 to increase.  Beware of clipping audio data when the gain  is
                 greater then 1.0.  A negative value performs the same adjust-
                 ment while also changing the phase.
                 When type is power then a value of 1.0 also means  no  change
                 in volume.
                 When  type  is  dB  the amplitude is changed logarithmically.
                 0.0 is constant while +6 doubles the amplitude.
                 An optional limitergain value can be specified and should  be
                 a value much less then 1.0 (ie 0.05 or 0.02) and is used only
                 on peaks to prevent clipping.  Not specifying this  parameter
                 will  cause  no  limiter  to  be used.  In verbose mode, this
                 effect will display the percentage of audio data that  needed
                 to be limited.

[edit] Examples


       An echo(1,3x,1 builtins) effect can be naturally found in(1,8) the mountains, standing  some-
       where  on  a  mountain and shouting a single word will result in(1,8) one or
       more repetitions of the word (if(3,n) not, turn a bit around and try  again,
       or climb to the next mountain).

       However,  the  time(1,2,n)  difference  between  shouting and repeating is the
       delay (time(1,2,n)), its loudness is the decay. Multiple echos can  have  dif-
       ferent delays and decays.

       It  is  very  popular  to  use  echos to play an instrument with itself
       together, like some guitar players (Brain May from Queen) or  vocalists
       are  doing.  For music samples of more than one instrument, echo(1,3x,1 builtins) can be
       used to add a second sample shortly after the original one.

       This will sound as if(3,n) you are doubling the number of instruments  play-
       ing in(1,8) the same sample:

             SoxFilter("echo 0.8 0.88 60.0 0.4")

       If the delay is very short, then it sound like a (metallic) robot play-
       ing music:

             SoxFilter("echo 0.8 0.88 6.0 0.4")

       Longer delay will sound like an open(2,3,n) air concert in(1,8) the mountains:

             SoxFilter("echo 0.8 0.9 1000.0 0.3")

       One mountain more, and:

             SoxFilter("echo 0.8 0.9 1000.0 0.3 1800.0 0.25")


       Like the echo effect, echos stand for "ECHO in  Sequel",  that  is  the
       first  echos takes the input, the second the input and the first echos,
       the third the input and the first and the second echos, ... and so  on.
       Care  should  be  taken  using  many echos (see introduction); a single
       echos has the same effect as a single echo.

       The sample will be bounced twice in(1,8) symmetric echos:

             SoxFilter("echos 0.8 0.7 700.0 0.25 700.0 0.3")

       The sample will be bounced twice in(1,8) asymmetric echos:

             SoxFilter("echos 0.8 0.7 700.0 0.25 900.0 0.3")

       The sample will sound as if(3,n) played in(1,8) a garage:

             SoxFilter("echos 0.8 0.7 40.0 0.25 63.0 0.3")


       The chorus effect has its name because it will often be used to make  a
       single  vocal  sound  like  a  chorus.  But  it can be applied to other
       instrument samples too.

       It works like the echo(1,3x,1 builtins) effect with a short delay, but the  delay  isn't
       constant.  The delay is varied using a sinusoidal or triangular modula-
       tion. 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(1,8) a chorus where some vocals are a bit out of tune.

       The  typical  delay is around 40ms to 60ms, the speed of the modulation
       is best near 0.25Hz and the modulation depth around 2ms.

       A single delay will make the sample more overloaded:

             SoxFilter("chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t")

       Two delays of the original samples sound like this:

             SoxFilter("chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 1.3 -s")

       A big chorus of the sample is (three additional samples):

              SoxFilter("chorus 0.5 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 0.4 2.3 -t 40.0 0.3 0.3 1.3 -s")


       The flanger effect is like the chorus  effect,  but  the  delay  varies
       between  0ms  and  maximal  5ms.  It sound like wind blowing, sometimes
       faster or slower including changes of the speed.

       The flanger effect is widely used in(1,8) funk and  soul  music,  where  the
       guitar sound varies frequently slow or a bit faster.

       The  typical delay is around 3ms to 5ms, the speed of the modulation is
       best near 0.5Hz.

       Now, let's groove the sample:

             SoxFilter("flanger 0.6 0.87 3.0 0.9 0.5 -s")

       listen carefully between the difference of  sinusoidal  and  triangular

             SoxFilter("flanger 0.6 0.87 3.0 0.9 0.5 -t")

       If the decay is a bit lower, than the effect sounds more popular:

             SoxFilter("flanger 0.8 0.88 3.0 0.4 0.5 -t")

       The drunken loudspeaker system:

             SoxFilter("flanger 0.9 0.9 4.0 0.23 1.3 -s")


       The  reverb effect is often used in(1,8) audience hall which are to small or
       contain too many many visitors which disturb (dampen) the reflection of
       sound  at  the walls.  Reverb will make the sound be perceived as if(3,n) it
       were in(1,8) a large hall.  You can try the reverb effect in(1,8)  your  bathroom
       or  garage  or sport halls by shouting loud some words. You'll hear the
       words reflected from the walls.

       The biggest problem in(1,8) using the reverb effect is the  correct  setting
       of the (wall) delays such that the sound is realistic and doesn't sound
       like music playing in(1,8) a  tin(1,5)  can  or  has  overloaded  feedback  which
       destroys  any  illusion  of  playing in(1,8) a big hall.  To help you obtain
       realistic reverb effects, you should decide first how long  the  reverb
       should  take place until it is not loud enough to be registered by your
       ears. This is be done by varying the  reverb  time(1,2,n)  "t".   To  simulate
       small halls, use 200ms.  To simulate large halls, use 1000ms.  Clearly,
       the walls of such a hall aren't far away, so you should define its set-
       ting  be  given  every wall its delay time.  However, if(3,n) the wall is to
       far away for the reverb time(1,2,n), you won't hear the reverb, so the nearest
       wall will be best at "t/4" delay and the farthest at "t/2". You can try
       other distances as well, but it won't sound very realistic.  The  walls
       shouldn't  stand  to  close(2,7,n) to each other and not in(1,8) a multiple integer
       distance to each other ( so avoid wall like: 200.0 and 202.0, or  some-
       thing like 100.0 and 200.0 ).

       Since  audience  halls  do have a lot of walls, we will start designing
       one beginning with one wall:

             SoxFilter("reverb 1.0 600.0 180.0")

       One wall more:

             SoxFilter("reverb 1.0 600.0 180.0 200.0")

       Next two walls:

             SoxFilter("reverb 1.0 600.0 180.0 200.0 220.0 240.0")

       Now, why not a futuristic hall with six walls:

             SoxFilter("reverb 1.0 600.0 180.0 200.0 220.0 240.0 280.0 300.0")

       If  you  run out of machine power or memory, then stop as many applica-
       tions as possible (every interrupt will consume a lot of CPU time(1,2,n) which
       for bigger halls is absolutely necessary).


       The  phaser  effect  is  like  the flanger effect, but it uses a reverb
       instead of an echo(1,3x,1 builtins) and does phase shifting. You'll hear the  difference
       in(1,8) the examples comparing both effects (simply change the effect name).
       The delay modulation can be sinusoidal or triangular, preferable is the
       later for multiple instruments. For single instrument sounds, the sinu-
       soidal phaser effect will give a sharper  phasing  effect.   The  decay
       shouldn't  be  to  close(2,7,n)  to 1.0 which will cause dramatic feedback.  A
       good range is about 0.5 to 0.1 for the decay.

       We will take a parameter setting as for the flanger before (gain-out is
       lower since feedback can raise(3,n) the output dramatically):

             SoxFilter("phaser 0.8 0.74 3.0 0.4 0.5 -t")

       The drunken loudspeaker system (now less(1,3) alcohol):

             SoxFilter("phaser 0.9 0.85 4.0 0.23 1.3 -s")

       A popular sound of the sample is as follows:

             SoxFilter("phaser 0.89 0.85 1.0 0.24 2.0 -t")

       The sample sounds if(3,n) ten springs are in(1,8) your ears:

             SoxFilter("phaser 0.6 0.66 3.0 0.6 2.0 -t")


       The  compander  effect  allows the dynamic range of a signal(2,7) to be com-
       pressed or expanded.  For most situations, the attack time(1,2,n) (response to
       the music getting louder) should be shorter than the decay time(1,2,n) because
       our ears are more sensitive to suddenly loud  music  than  to  suddenly
       soft music.

       For  example,  suppose  you  are  listening  to  Strauss'  "Also Sprach
       Zarathustra" in(1,8) a noisy environment such as a car.  If you turn up  the
       volume  enough  to hear the soft passages over the road noise, the loud
       sections will be too loud.  You could try this:

             SoxFilter("compand 0.3,1 -90,-90,-70,-70,-60,-20,0,0 -5 0 0.2")

       The transfer function ("-90,...") says that very  soft  sounds  between
       -90  and  -70 decibels (-90 is about the limit of 16-bit encoding(3,n)) will
       remain unchanged.  That keeps the compander from boosting the volume on
       "silent"  passages  such  as between movements.  However, sounds in(1,8) the
       range -60 decibels to 0 decibels (maximum volume) will  be  boosted  so
       that  the  60-dB dynamic range of the original music will be compressed
       3-to-1 into a 20-dB range, which is wide enough to enjoy the music  but
       narrow  enough  to get around the road noise.  The -5 dB output gain is
       needed to avoid clipping (the number is inexact,  and  was  derived  by
       experimentation).   The  0  for the initial volume will work fine for a
       clip that starts with a bit of silence, and the delay of  0.2  has  the
       effect  of  causing the compander to react a bit more quickly to sudden
       volume changes.

[edit] Changelog

Version        Date            Changes
v2.2 2024/01/04 - Change the way how the effect chain is reinitialized: Fix the "Normalize" after SoxFilter("compand") case. v2.1 2023/12/03 - Allow effects which can alter sampling rate - Allow effects which can change the number of channels - Add distinct XP build (v141_xp toolset) needed in 2023 v2.0 2023/12/02 - Full rewrite, using 14.4.2 libsox library version, moved to github v1.1 beta 2006/01/02 - release (12.17.9 libsox) v1.0 2005/12/29 - initial release

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