Provided by: sox_14.0.0-5_i386 bug

NAME

       SoX - Sound eXchange, the Swiss Army knife of audio manipulation

DESCRIPTION

   SOX EFFECTS
       Multiple  effects  may  be  applied to the audio by specifying them one
       after another at the end of the SoX command line.

       Note: Brackets [ ] are used to denote  parameters  that  are  optional,
       braces  {  } to denote those that are both optional and repeatable, and
       angle brackets <  >  to  denote  those  that  are  repeatable  but  not
       optional.

       allpass frequency width[h|o|q]
              Apply  a two-pole all-pass filter with central frequency (in Hz)
              frequency, and filter-width width: in Hz  (the  default,  or  if
              appended  with  ‘h’), in octaves (if appended with ‘o’), or as a
              Q-factor (if appended with ‘q’).  An all-pass filter changes the
              audio’s  frequency  to  phase  relationship without changing its
              frequency to amplitude relationship.  The filter is described in
              detail in [1].

              This effect supports the --plot global option.

       band [-n] center [width[h|o|q]]
              Apply   a   band-pass  filter.   The  frequency  response  drops
              logarithmically around the center frequency.  The  width  in  Hz
              (the  default, or if appended with ‘h’), in octaves (if appended
              with ‘o’), or as a Q-factor (if appended with  ‘q’),  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  audio, i.e. voice,
              singing, or instrumental music.  The -n (for noise) option  uses
              the  alternate  mode  for  un-pitched  audio  (e.g. percussion).
              Warning: -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.

              This effect supports the --plot global option.

              See also filter for a bandpass filter with steeper shoulders.

       bandpass|bandreject [-c] frequency width[h|o|q]
              Apply  a  two-pole  Butterworth  band-pass or band-reject filter
              with central frequency (in Hz) frequency, and (3dB-point)  band-
              width  width:  in  Hz (the default, or if appended with ‘h’), in
              octaves (if appended with ‘o’), or as a  Q-factor  (if  appended
              with ‘q’).  The -c option applies only to bandpass and selects a
              constant skirt gain (peak gain =  Q)  instead  of  the  default:
              constant  0dB peak gain.  The filters roll off at 6dB per octave
              (20dB per decade) and are described in detail in [1].

              These effects support the --plot global option.

              See also filter for a bandpass filter with steeper shoulders.

       bandreject frequency width[h|o|q]
              Apply a band-reject filter.  See the description of the bandpass
              effect for details.

       bass|treble gain [frequency [width[s|h|o|q]]]
              Boost  or  cut the bass (lower) or treble (upper) frequencies of
              the audio using a  two-pole  shelving  filter  with  a  response
              similar to that of a standard hi-fi’s (Baxandall) tone-controls.
              This is also known as shelving equalisation (EQ).

              gain gives the dB gain at 0 Hz (for bass), or whichever  is  the
              lower  of  ∼22 kHz  and the Nyquist frequency (for treble).  Its
              useful range is about -20 (for a large cut) to +20 (for a  large
              boost).  Beware of Clipping when using a positive gain.

              If  desired,  the  filter  can be fine-tuned using the following
              optional parameters:

              frequency sets 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 (for bass) or 3 kHz (for treble).

              width determines how steep the filter’s shelf transition is  and
              can be expressed as: a ‘slope’ (the default, or if appended with
              ‘s’), a Q-factor (if appended with ‘q’), the transition width in
              octaves  (if  appended  with ‘o’), or the transition width in Hz
              (if appended with ‘h’).  The useful range of  ‘slope’  is  about
              0.3,  for a gentle slope, to 1 (the maximum), for a steep slope;
              the default value is 0.5.

              The filters are described in detail in [1].

              These effects support the --plot global option.

              See also equalizer for a peaking equalisation effect.

       chorus gain-in gain-out <delay decay speed depth -s|-t>
              Add  a  chorus   effect   to   the   audio.    Each   four-tuple
              delay/decay/speed/depth  gives the delay in milliseconds and the
              decay (relative to gain-in) with a modulation speed in Hz  using
              depth in milliseconds.  The modulation is either sinusoidal (-s)
              or triangular (-t).  Gain-out is the volume of the output.

       compand attack1,decay1{,attack2,decay2}
              [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
              [gain [initial-volume-dB [delay]]]

              Compand (compress or expand) the dynamic range of the audio.

              The attack and decay parameters (in seconds) determine the  time
              over  which  the  instantaneous  level  of  the  input signal is
              averaged 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 input channel  is
              companded separately and the number of pairs must agree with the
              number of input channels.  Typical values are 0.3,0.8 seconds.

              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.  If omitted, the value of  out-
              dB1  defaults  to  the same value as in-dB1; levels below in-dB1
              are not companded (but may have  gain  applied  to  them).   The
              point  0,0  is assumed but may be overridden (by 0,out-dBn).  If
              the list is preceded by a soft-knee-dB value, then the points at
              where  adjacent line segments on the transfer function meet will
              be rounded by the amout given.  Typical values for the  transfer
              function are 6:-70,-60,-20.

              The third (optional) parameter is an additional gain in dB to be
              applied at all points on the transfer function and  allows  easy
              adjustment of the overall gain.

              The  fourth  (optional)  parameter  is  an  initial  level to be
              assumed for each channel when companding 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 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
              itself.  A typical value (for audio which is initially quiet) is
              -90 dB.

              The fifth (optional) parameter is a delay in seconds.  The input
              signal is analysed immediately to control the compander, 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 operate in a ‘predictive’ rather than a
              reactive mode.  A typical value is 0.2 seconds.

              This effect supports the --plot global option (for the  transfer
              function).

              See also mcompand for a multiple-band companding effect.

       dcshift shift [limitergain]
              DC  Shift  the audio, with basic linear amplitude formula.  This
              is most useful if your audio tends to not be centered  around  a
              value  of  0.   Shifting  it back will allow you to get the most
              volume adjustments without clipping.

              The first option is the dcshift value.  It is a  floating  point
              number that indicates the amount to shift.

              An  optional  limitergain  can  be specified as well.  It should
              have a value much less than 1 (e.g. 0.05 or 0.02)  and  is  used
              only on peaks to prevent clipping.

       deemph Apply  a treble attenuation shelving filter to audio in audio-CD
              format.  The frequency response of pre-emphasized recordings  is
              rectified.   The  filter is defined in the standard document ISO
              908.

              This effect supports the --plot global option.

              See also the bass and treble shelving equalisation effects.

       dither [depth]
              Apply dithering  to  the  audio.   Dithering  deliberately  adds
              digital  white  noise  to  the  signal  in order to mask audible
              quantization effects that can occur if the output sample size is
              less  than  24 bits.  By default, the amount of noise added is ½
              bit; the optional depth  parameter  is  a  (linear  or  voltage)
              multiplier of this amount.

              This  effect  should  not  be  followed by any other effect that
              affects the audio.

       earwax Makes audio easier to listen to on headphones.  Adds  ‘cues’  to
              audio  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 listener (standard
              for speakers).  See http://www.geocities.com/beinges for a  full
              explanation.

       echo gain-in gain-out <delay decay>
              Add echoing to the audio.  Each delay decay pair 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>
              Add  a  sequence  of  echos to the audio.  Each delay decay pair
              gives the delay in milliseconds and the decay (relative to gain-
              in) of that echo.  Gain-out is the volume of the output.

       equalizer frequency width[q|o|h] gain
              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 band-pass and band-
              reject filters) that at all other frequencies is unchanged.

              frequency gives the filter’s central frequency in Hz, width, the
              band-width,  as a Q-factor [2] (the default, or if appended with
              ‘q’), in octaves (if appended with ‘o’), or in Hz  (if  appended
              with  ‘h’),  and  gain  the  required gain or attenuation in dB.
              Beware of Clipping when using a positive gain.

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

              The filter is described in detail in [1].

              This effect supports the --plot global option.

              See also bass and treble for shelving equalisation effects.

       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 will be truncated at stop-time and  the
              volume  will  be  ramped  from full volume down to 0 starting at
              fade-out-length seconds  before  the  stop-time.   If  fade-out-
              length  is not specified, it defaults to the same value as fade-
              in-length.   No  fade-out  is  performed  if  stop-time  is  not
              specified.

              All  times  can be specified in either periods of time or sample
              counts.  To specify time periods use  the  format  hh:mm:ss.frac
              format.   To  specify using sample counts, specify 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
              envelope.  Choices are q for quarter of a sine wave, h for  half
              a  sine  wave,  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 frequency
              of the lower 6dB corner of  the  filter.   high  refers  to  the
              frequency of the upper 6dB corner of the filter.

              A  low-pass filter is obtained by leaving low unspecified, or 0.
              A high-pass filter is obtained by leaving high  unspecified,  or
              0, or greater than or equal to the Nyquist frequency.

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

              The beta, if unspecified, defaults to 16.  This selects a Kaiser
              window.   You can select a Nuttall window by specifying anything
              ≤ 2 here.  For more discussion of beta, look under the  resample
              effect.

       flanger [delay depth regen width speed shape phase interp]
              Apply  a  flanging  effect  to  the  audio.   All parameters are
              optional (right to left).

             +-----------------------------------------------------------------+
             |          Range     Default   Description                        |
             |delay     0 - 10       0      Base delay in milliseconds.        |
             |depth     0 - 10       2      Added swept delay in milliseconds. |
             |regen    -95 - 95      0      Percentage regeneration (delayed   |
             |                              signal feedback).                  |
             |width    0 - 100      71      Percentage of delayed signal mixed |
             |                              with original.                     |
             |speed    0.1 - 10     0.5     Sweeps per second (Hz).            |
             |shape                 sin     Swept wave shape: sine|triangle.   |
             |phase    0 - 100      25      Swept wave percentage phase-shift  |
             |                              for multi-channel (e.g. stereo)    |
             |                              flange; 0 = 100 = same phase on    |
             |                              each channel.                      |
             |interp                lin     Digital delay-line interpolation:  |
             |                              linear|quadratic.                  |
             +-----------------------------------------------------------------+
              See [3] for a detailed description of flanging.

       highpass|lowpass [-1|-2] frequency [width[q|o|h]]
              Apply a high-pass or low-pass filter with 3dB  point  frequency.
              The  filter  can be either single-pole (with -1), or double-pole
              (the default, or with -2).  width applies  only  to  double-pole
              filters  and is the filter-width: as a Q-factor (the default, or
              if appended with ‘q’), in octaves (if appended with ‘o’), or  in
              Hz  (if  appended  with ‘h’); the default Q is 0.707 and gives a
              Butterworth response.  The filters roll off at 6dB per pole  per
              octave  (20dB per pole per decade).  The double-pole filters are
              described in detail in [1].

              These effects support the --plot global option.

              See also filter for filters with a steeper roll-off.

       key [-q] shift [segment [search [overlap]]]
              Change the audio key (i.e. pitch but not tempo)  using  a  WSOLA
              algorithm.

              shift  gives the key shift as positive or negative ‘cents’ (i.e.
              100ths of a semitone).  See the tempo effect for  a  description
              of the other parameters.

              See also pitch for a similar effect.

       ladspa module [plugin] [argument...]
              Apply  a  LADSPA [5] (Linux Audio Developer’s Simple Plugin API)
              plugin.  Despite the name, LADSPA is not Linux-specific,  and  a
              wide  range  of  effects is available as LADSPA plugins, such as
              cmt [6] (the Computer Music Toolkit) and Steve  Harris’s  plugin
              collection  [7].  The  first  argument is the plugin module, the
              second the name of the plugin (a module can  contain  more  than
              one plugin) and any other arguments are for the control ports of
              the plugin. Missing arguments are supplied by default values  if
              possible.  Only  plugins  with  at  most one audio input and one
              audio output port can be used.

       lowpass [-1|-2] frequency [width[q|o|h]]
              Apply a low-pass filter.  See the description  of  the  highpass
              effect for details.

       mcompand "attack1,decay1{,attack2,decay2}
              [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
              [gain [initial-volume-dB [delay]]]" {xover-freq "attack1,..."}

              The multi-band compander is similar to the single-band compander
              but the audio is first  divided  into  bands  using  Butterworth
              cross-over filters and a separately specifiable compander run on
              each band.  See the compand effect for  the  definition  of  its
              parameters.   Compand  parameters  are  specified between double
              quotes and the crossover frequency for that  band  is  given  by
              xover-freq; these can be repeated to create multiple bands.

              See also compand for a single-band companding effect.

       mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
              Reduce  the  number  of  audio  channels  by mixing or selecting
              channels, or increase the  number  of  channels  by  duplicating
              channels.   Note:  this  effect  operates  on the audio channels
              within the SoX  effects  processing  chain;  it  should  not  be
              confused  with  the  -m  global option (where multiple files are
              mix-combined before entering the effects chain).

              This effect is automatically  used  when  the  number  of  input
              channels  differ  from  the  number  of  output  channels.  When
              reducing the number of  channels  it  is  possible  to  manually
              specify the mixer effect and use the -l, -r, -f, -b, -1, -2, -3,
              -4,  options  to  select  only  the  left,  right,  front,  back
              channel(s)  or  specific  channel  for  the  output  instead  of
              averaging  the  channels.   The  -l,  and  -r  options  will  do
              averaging  in  quad-channel files so select the exact channel to
              prevent this.

              The mixer effect can also be invoked  with  up  to  16  numbers,
              separated  by commas, which specify the proportion (0 = 0% and 1
              = 100%) of each input channel that is  to  be  mixed  into  each
              output  channel.   In two-channel mode, 4 numbers are given: l →
              l, l → r, r → l, and r → r, respectively.  In four-channel mode,
              the  first  4  numbers  give  the proportions for the left-front
              output channel, as follows: lf → lf, rf → lf, lb → lf, and rb  →
              rf.   The  next 4 give the right-front output in the same order,
              then left-back and right-back.

              It is also possible to use the 16 numbers to  expand  or  reduce
              the channel count; just specify 0 for unused channels.

              Finally, certain reduced combination of numbers can be specified
              for certain input/output channel combinations.

                  +------------------------------------------------------+
                  |In Ch   Out Ch   Num   Mappings                       |
                  |  2       1       2    l → l, r → l                   |
                  |  2       2       1    adjust balance                 |
                  |  4       1       4    lf → l, rf → l, lb → l, rb → l |
                  |  4       2       2    lf → l&rf → r, lb → l&rb → r   |
                  |  4       4       1    adjust balance                 |
                  |  4       4       2    front balance, back balance    |
                  +------------------------------------------------------+

       noiseprof [profile-file]
              Calculate a profile of the audio for  use  in  noise  reduction.
              See the description of the noisered effect for details.

       noisered [profile-file [amount]]
              Reduce  noise  in  the  audio signal by profiling and filtering.
              This effect  is  moderately  effective  at  removing  consistent
              background  noise such as hiss or hum.  To use it, first run SoX
              with the noiseprof effect on a section  of  audio  that  ideally
              would contain silence but in fact contains noise - such sections
              are typically found at the beginning or the end of a  recording.
              noiseprof  will write out a noise profile to profile-file, or to
              stdout if no profile-file or if ‘-’ is given.  E.g.

                   sox speech.au -n trim 0 1.5 noiseprof speech.noise-profile

              To actually remove the noise, run SoX again, this time with  the
              noisered effect; noisered will reduce noise according to a noise
              profile (which was generated by noiseprof),  from  profile-file,
              or from stdin if no profile-file or if ‘-’ is given.  E.g.

                   sox speech.au cleaned.au noisered speech.noise-profile 0.3

              How much noise should be removed is specified by amount-a number
              between 0 and 1 with a default  of  0.5.   Higher  numbers  will
              remove  more  noise but present a greater likelihood of removing
              wanted components of the  audio  signal.   Before  replacing  an
              original recording with a noise-reduced version, experiment with
              different amount values to find the optimal one for your  audio;
              use  headphones  to  check  that you are happy with the results,
              paying particular attention to quieter sections of the audio.

              On most systems, the two stages - profiling and reduction -  can
              be combined using a pipe, e.g.

                   sox noisy.au -n trim 0 1 noiseprof | play noisy.au noisered

       oops   Out  Of  Phase  Stereo  effect.  Mixes stereo to twin-mono where
              each mono channel contains the difference between the  left  and
              right  stereo  channels.  This is sometimes known as the karaoke
              effect as it often has the effect of removing most or all of the
              vocals from a recording.

       pad { length[@position] }
              Pad  the  audio  with silence, at the beginning, the end, or any
              specified points through the audio.  Both  length  and  position
              can  specify  a  time  or,  if appended with an ‘s’, a number of
              samples.  length is the amount of silence to insert and position
              the  position  in  the input audio stream at which to insert it.
              Any number of lengths and positions may be  specified,  provided
              that  a  specified  position  is not less that the previous one.
              position is optional for the first and  last  lengths  specified
              and  if  omitted  correspond to the beginning and the end of the
              audio respectively.  For example: pad 1.5 1.5 adds  1.5  seconds
              of  silence  padding  at  each  end  of  the  audio,  whilst pad
              4000s@3:00 inserts 4000 samples of silence 3  minutes  into  the
              audio.   If  silence  is  wanted  only  at the end of the audio,
              specify either the end position or specify a zero-length pad  at
              the start.

       pan direction
              Pan  the  audio  from  one  channel to another.  This is done by
              changing the volume of the input channels so that it  fades  out
              on  one channel and fades-in on another.  If the number of input
              channels is different then the number of  output  channels  then
              this  effect  tries to intelligently handle this.  For instance,
              if the input contains  1  channel  and  the  output  contains  2
              channels,  then  it will create the missing channel itself.  The
              direction is a value from -1 to 1.  -1 represents far left and 1
              represents  far  right.   Numbers  in between will start the pan
              effect without totally muting the opposite channel.

       phaser gain-in gain-out delay decay speed [-s|-t]
              Add a phasing effect to the audio.  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  sinusoidal
              (-s)  or  triangular (-t).  The decay should be less than 0.5 to
              avoid feedback.  Gain-out is the volume of the output.

       polyphase [-w nut|ham] [-width n] [-cutoff c]
              Change the sampling rate using ‘polyphase interpolation’, a  DSP
              algorithm.  This method is relatively slow and memory intensive.

              If the -w parameter is nut, then a Nuttall  (~90  dB  stop-band)
              window  will  be  used; ham selects a Hamming (~43 dB stop-band)
              window.  The default is Nuttall.

              The -width parameter specifies the (approximate)  width  of  the
              filter.  The  default is 1024 samples, which produces reasonable
              results.

              The -cutoff value (c) specifies the filter cutoff  frequency  in
              terms  of  fraction  of  frequency  bandwidth,  also know as the
              Nyquist  frequency.   See  the  resample  effect   for   further
              information  on Nyquist frequency.  If up-sampling, then this is
              the fraction of the original signal that should go through.   If
              down-sampling,  this  is  the  fraction of the signal left after
              down-sampling.  The default is 0.95.

              See also rabbit and  resample  for  other  sample-rate  changing
              effects.

       rabbit [-c0|-c1|-c2|-c3|-c4]
              Change  the  sampling  rate  using  libsamplerate, also known as
              ‘Secret Rabbit Code’.  This effect is  optional  and  must  have
              been  selected  at  compile  time  of SoX.  See http://www.mega-
              nerd.com/SRC  for  details  of  the  algorithms.   Algorithms  0
              through 2 are progressively faster and lower quality versions of
              the sinc algorithm; the default is -c0, which  is  probably  the
              best  quality  algorithm  for general use currently available in
              SoX.   Algorithm  3  is  zero-order  hold,  and  4   is   linear
              interpolation.

              See  also  polyphase and resample for other sample-rate changing
              effects, and see resample for more discussion of resampling.

       repeat count
              Repeat the entire audio count times.   Requires  disk  space  to
              store  the data to be repeated.  Note that repeating once yields
              two copies: the original audio and the repeated audio.

       resample [-qs|-q|-ql] [rolloff [beta]]
              Change the sampling  rate  using  simulated  analog  filtration.
              Other  rate changing effects available are polyphase and rabbit.
              There is a  detailed  analysis  of  resample  and  polyphase  at
              http://leute.server.de/wilde/resample.html;  see  rabbit  for  a
              pointer to its own documentation.

              By default, linear interpolation is used, with  a  window  width
              about  45  samples at the lower of the two rates.  This gives an
              accuracy of about 16 bits, but insufficient stop-band  rejection
              in  the  case  that you want to have roll-off greater than about
              0.8 of the Nyquist frequency.

              The -q* options will change the default values for roll-off  and
              beta   as   well   as  use  quadratic  interpolation  of  filter
              coefficients, resulting in about 24 bits  precision.   The  -qs,
              -q,  or  -ql  options  specify increased accuracy at the cost of
              lower execution speed.  It is optional to specify  roll-off  and
              beta parameters when using the -q* options.

              Following is a table of the reasonable defaults which are built-
              in to SoX:

                    +--------------------------------------------------+
                    |Option   Window   Roll-off   Beta   Interpolation |
                    |(none)     45       0.80      16       linear     |
                    | -qs       45       0.80      16      quadratic   |
                    |  -q       75      0.875      16      quadratic   |
                    | -ql      149       0.94      16      quadratic   |
                    +--------------------------------------------------+
              -qs, -q, or -ql use window lengths of 45, 75,  or  149  samples,
              respectively,  at  the lower sample-rate of the two files.  This
              means   progressively   sharper    stop-band    rejection,    at
              proportionally slower execution times.

              rolloff  refers  to the cut-off frequency of the low pass filter
              and is given in terms of the Nyquist  frequency  for  the  lower
              sample  rate.   rolloff  therefore should be something between 0
              and 1, in practise 0.8-0.95.  The defaults are indicated  above.

              The  Nyquist  frequency  is  equal  to  half  the  sample  rate.
              Logically, this is because the A/D converter needs  at  least  2
              samples to detect 1 cycle at the Nyquist frequency.  Frequencies
              higher  then  the  Nyquist  will  actually   appear   as   lower
              frequencies  to  the  A/D  converter  and  is  called  aliasing.
              Normally, A/D converts run the signal through a  lowpass  filter
              first to avoid these problems.

              Similar  problems  will  happen  in  software  when reducing the
              sample rate of an audio file (frequencies above the new  Nyquist
              frequency  can  be  aliased to lower frequencies).  Therefore, a
              good resample effect will remove all frequency information above
              the new Nyquist frequency.

              The  rolloff  refers  to how close to the Nyquist frequency this
              cutoff is, with closer being better.  When increasing the sample
              rate  of  an  audio  file  you  would  not  expect  to  have any
              frequencies exist that are past the original Nyquist  frequency.
              Because  of resampling properties, it is common to have aliasing
              artifacts created above the old Nyquist frequency.  In that case
              the  rolloff  refers  to  how  close  to  the  original  Nyquist
              frequency to use a highpass filter to  remove  these  artifacts,
              with closer also being better.

              The  beta  parameter  determines the type of filter window used.
              Any value greater than 2 is the beta for a Kaiser window.   Beta
              ≤  2 selects a Nuttall window.  If unspecified, the default is a
              Kaiser window with beta 16.

              In the case of Kaiser window (beta > 2), lower betas  produce  a
              somewhat  faster  transition from pass-band to stop-band, at the
              cost of noticeable artifacts. A beta of 16 is the default,  beta
              less  than  10 is not recommended. If you want a sharper cutoff,
              don’t use low beta’s, use a  longer  sample  window.  A  Nuttall
              window is selected by specifying any ‘beta’ ≤ 2, and the Nuttall
              window has somewhat  steeper  cutoff  than  the  default  Kaiser
              window.  You will probably not need to use the beta parameter at
              all, unless you are just curious about comparing the effects  of
              Nuttall vs. Kaiser windows.

              This  is  the  default  effect  if  the two files have different
              sampling rates.  Default parameters  are,  as  indicated  above,
              Kaiser  window  of  length  45,  roll-off  0.80, beta 16, linear
              interpolation.

              Note: -qs is only slightly slower, but more accurate for  16-bit
              or higher precision.

              Note:  In many cases of up-sampling, no interpolation is needed,
              as exact filter coefficients can be  computed  in  a  reasonable
              amount  of  space.  To be precise, this is done when both input-
              rate < output-rate, and output-rate  ÷  gcd(input-rate,  output-
              rate) ≤ 511.

       reverb gain-out reverb-time <delay>
              Add  reverberation  to  the  audio.   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.

       reverse
              Reverse  the audio completely.  Requires disk space to store the
              data to be reversed.

       silence [-l] above-periods [duration
              threshold[d|%] [below-periods duration threshold[d|%]]

              Removes silence from the beginning, middle, or end of the audio.
              Silence is anything below a specified threshold.

              The  above-periods  value is used to indicate if audio should be
              trimmed at the beginning of the audio. A value of zero indicates
              no silence should be trimmed from the beginning. When specifying
              an non-zero above-periods, it trims audio up until it finds non-
              silence. Normally, when trimming silence from beginning of audio
              the above-periods will be 1 but it can be  increased  to  higher
              values  to  trim all audio up to a specific count of non-silence
              periods. For example, if you had an audio file  with  two  songs
              that  each  contained  2 seconds of silence before the song, you
              could specify an above-period of 2 to  strip  out  both  silence
              periods and the first song.

              When above-periods is non-zero, you must also specify a duration
              and threshold. Duration indications the amount of time that non-
              silence  must  be  detected  before  it stops trimming audio. By
              increasing the duration,  burst  of  noise  can  be  treated  as
              silence and trimmed off.

              Threshold is used to indicate what sample value you should treat
              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.

              When optionally trimming silence from the end of the audio,  you
              specify a below-periods count.  In this case, below-period means
              to remove all audio after silence is detected.   Normally,  this
              will  be  a  value  1  of  but  it can be increased to skip over
              periods of silence that are wanted.  For example, if you have  a
              song with 2 seconds of silence in the middle and 2 second at the
              end, you could set below-period to a value of 2 to skip over the
              silence in the middle of the audio.

              For  below-periods,  duration specifies a period 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.  For example, if you have  a  song  with  an  expected  1
              second  of silence in the middle and 2 seconds of silence at the
              end, a duration of 2 seconds could be  used  to  skip  over  the
              middle silence.

              Unfortunately,  you  must  know the length of the silence at the
              end of your audio file to trim off  silence  reliably.   A  work
              around  is  to  use  the  silence effect in combination with the
              reverse effect.  By first reversing the audio, you can  use  the
              above-periods  to  reliably  trim all audio from what looks like
              the front of the file.  Then reverse the file again to get  back
              to normal.

              To  remove  silence  from the middle of a file, specify a below-
              periods that is negative.  This  value  is  then  treated  as  a
              positive  value  and  is also used to indicate the effect should
              restart processing as specified by the above-periods, making  it
              suitable  for  removing  periods of silence in the middle of the
              audio.

              The option -l indicates that below-periods  duration  length  of
              audio  should  be left intact at the beginning of each period of
              silence.  For example, if you want to remove long pauses between
              words but do not want to remove the pauses completely.

              The  period  counts are in units of samples. Duration counts may
              be in the  format  of  hh:mm:ss.frac,  or  the  exact  count  of
              samples.   Threshold  numbers may be suffixed with d to indicate
              the value is in decibels, or  %  to  indicate  a  percentage  of
              maximum  value  of  the  sample value (0% specifies pure digital
              silence).

       speed factor[c]
              Adjust the audio speed (pitch and tempo  together).   factor  is
              either the ratio of the new speed to the old speed: greater than
              1 speeds up, less than 1 slows down, or, if  appended  with  the
              letter  ‘c’,  the number of cents (i.e. 100ths of a semitone) by
              which the pitch (and tempo) should be adjusted: greater  than  0
              increases, less than 0 decreases.

              By default, the speed change is performed by the resample effect
              with its default parameters.  For higher quality resampling,  in
              addition to the speed effect, specify either the resample or the
              rabbit effect with appropriate parameters.

       stat [-s n] [-rms] [-freq] [-v] [-d]
              Do a statistical check on the input file, and print  results  on
              the standard error file.  Audio is passed unmodified through the
              SoX processing chain.

              The ‘Volume Adjustment:’ field in the statistics gives  you  the
              parameter  to the -v number which will make the audio as loud as
              possible without clipping.  Note: See the discussion on Clipping
              above  for  reasons  why it is rarely a good idea to actually do
              this.

              The option -v will print out the  ‘Volume  Adjustment:’  field’s
              value  only and return.  This could be of use in scripts to auto
              convert the volume.

              The -s option is used to scale the input data by a given factor.
              The  default  value  of  n  is  the  max  value of a signed long
              variable (0x7fffffff).  Internal effects always work with signed
              long PCM data and so the value should relate to this fact.

              The  -rms option will convert all output average values to ‘root
              mean square’ format.

              The -freq option  calculates  the  input’s  power  spectrum  and
              prints it to standard error.

              There is also an optional parameter -d that will print out a hex
              dump of the audio from the internal buffer  that  is  in  32-bit
              signed  PCM  data.   This is mainly only of use in tracking down
              endian problems that creep in to SoX on cross-platform versions.

       swap [1 2 | 1 2 3 4]
              Swap channels in multi-channel audio files.  Optionally, you may
              specify the channel order you would like the  output  in.   This
              defaults  to output channel 2 and then 1 for stereo and 2, 1, 4,
              3 for quad-channels.  An interesting feature  is  that  you  may
              duplicate  a given channel by overwriting another.  This is done
              by  repeating  an  output  channel  on  the  command-line.   For
              example,  swap  2  2  will  overwrite  channel 1 with channel 2;
              creating a stereo file with both channels  containing  the  same
              audio.

       synth [len] {[type] [combine] [freq[-freq2]] [off] [ph] [p1] [p2] [p3]}
              This  effect  can  be  used to generate fixed or swept frequency
              audio tones with various wave shapes, or to  generate  wide-band
              noise  of  various  ‘colours’.   Multiple  synth  effects can be
              cascaded to produce more complex waveforms; at each stage it  is
              possible  to choose whether the generated waveform will be mixed
              with, or modulated onto the  output  from  the  previous  stage.
              Audio  for  each  channel  in  a multi-channel audio file can be
              synthesised independently.

              Though this effect is used to generate audio, an input file must
              still be given, the characteristics of which will be used to set
              the synthesised audio length, the number of  channels,  and  the
              sampling  rate;  however,  since  the  input file’s audio is not
              normally needed, a ‘null file’ (with the  special  name  -n)  is
              often  given instead (and the length specified as a parameter to
              synth or by another given effect  that  can  has  an  associated
              length).

              For  example, the following produces a 3 second, 44.1 kHz, audio
              file containing a sine-wave swept from 300 to 3300 Hz:

                   sox -n output.au synth 3 sine 300-3300

              and this produces an 8 kHz version:

                   sox -r 8000 -n output.au synth 3 sine 300-3300

              Multiple channels can be synthesised by specifying  the  set  of
              parameters  shown  between  braces multiple times; the following
              puts the swept tone in the left channel and adds  ‘brown’  noise
              in the right:

                   sox -n output.au synth 3 sine 300-3300 brownnoise

              The  following  example  shows  how  two  synth  effects  can be
              cascaded to create a more complex waveform:

                   sox -n output.au synth 0.5 sine 200-500 \
                        synth 0.5 sine fmod 700-100

              Frequencies can also be given as a number of  musical  semitones
              relative  to  ‘middle  A’ (440 Hz) by prefixing a ‘%’ character;
              for example, the following could be used to help tune a guitar’s
              ‘E’ strings:

                   play -n synth sine %-17

              N.B.  This effect generates audio at maximum volume, which means
              that there is a high chance of clipping  when  using  the  audio
              subsequently,  so  in  most  cases, you will want to follow this
              effect with the vol effect to prevent this from happening.  (See
              also Clipping above.)

              A detailed description of each synth parameter follows:

              len  is the length of audio to synthesise expressed as a time or
              as a number of samples; 0=inputlength, default=0.

              The format for specifying lengths in time is hh:mm:ss.frac.  The
              format  for  specifying  sample  counts is the number of samples
              with the letter ‘s’ appended to it.

              type is one of sine, square, triangle, sawtooth, trapezium, exp,
              [white]noise, pinknoise, brownnoise; default=sine

              combine is one of create, mix, amod (amplitude modulation), fmod
              (frequency modulation); default=create

              freq/freq2 are the frequencies at the beginning/end of synthesis
              in  Hz  or,  if  preceded  with  ‘%’,  semitones  relative  to A
              (440 Hz); for both, default=%0.  If freq2  is  given,  then  len
              must also have been given.  Not used for noise.

              off is the bias (DC-offset) of the signal in percent; default=0.

              ph is the phase shift in percentage of 1 cycle; default=0.   Not
              used for noise.

              p1  is  the  percentage  of each cycle that is ‘on’ (square), or
              ‘rising’  (triangle,  exp,   trapezium);   default=50   (square,
              triangle, exp), default=10 (trapezium).

              p2  (trapezium):  the  percentage  through  each  cycle at which
              ‘falling’ begins; default=50. exp:  the  amplitude  in  percent;
              default=100.

              p3  (trapezium):  the  percentage  through  each  cycle at which
              ‘falling’ ends; default=60.

       tempo [-q] factor [segment [search [overlap]]]
              Change the audio tempo (but  not  its  pitch)  using  a  ‘WSOLA’
              algorithm.  The audio is chopped up into segments which are then
              shifted in the  time  domain  and  overlapped  (cross-faded)  at
              points  where their waveforms are most similar (as determined by
              measurement of ‘least squares’).

              By  default,  linear  searches  are  used  to  find   the   best
              overlapping  points; if the optional -q parameter is given, tree
              searches are used instead, giving a quicker, but possibly  lower
              quality, result.

              factor gives the ratio of new tempo to the old tempo.

              The  optional  segment parameter selects the algorithm’s segment
              size in milliseconds.  The default value is 82 and is  typically
              suited to making small changes to the tempo of music; for larger
              changes (e.g. a factor of 2), 50 ms may give  a  better  result.
              When  changing  the  tempo  of  speech, a segment size of around
              30 ms often works well.

              The  optional  search  parameter  gives  the  audio  length   in
              milliseconds  (default  14) over which the algorithm will search
              for overlapping points.  Larger values use more processing  time
              and do not necessarily produce better results.

              The  optional overlap parameter gives the segment overlap length
              in milliseconds (default 12).

              See also stretch for a similar effect.

       treble gain [frequency [width[s|h|o|q]]]
              Apply a treble tone-control effect.  See the description of  the
              bass effect for details.

       tremolo speed [depth]
              Apply  a  tremolo (low frequency amplitude modulation) effect to
              the audio.  The tremolo frequency in Hz is given by  speed,  and
              the depth as a percentage by depth (default 40).

              Note: This effect is a special case of the synth effect.

       trim start [length]
              Trim  can  trim off unwanted audio from the beginning and end of
              the audio.  Audio is not sent to the  output  stream  until  the
              start location is reached.

              The  optional  length  parameter  tells the number of samples to
              output after the start sample and is used to trim off  the  back
              side  of  the audio.  Using a value of 0 for the start parameter
              will allow trimming off the back side only.

              Both options can be specified using either an amount of time  or
              an exact count of samples.  The format for specifying lengths in
              time is hh:mm:ss.frac.  A start value of 1:30.5 will  not  start
              until 1 minute, thirty and ½ seconds into the audio.  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.

       vol gain [type [limitergain]]
              Apply an amplification or an attenuation to  the  audio  signal.
              Unlike the -v option (which is used for balancing multiple input
              files as they enter the SoX effects processing chain), vol is an
              effect  like  any  other so can be applied anywhere, and several
              times if necessary, during the processing chain.

              The amount to change the  volume  is  given  by  gain  which  is
              interpreted, according to the given type, as follows: if type is
              amplitude (or is omitted),  then  gain  is  an  amplitude  (i.e.
              voltage  or  linear) ratio, if power, then a power (i.e. wattage
              or voltage-squared) ratio, and if dB, then a power change in dB.

              When  type  is amplitude or power, a gain of 1 leaves the volume
              unchanged,  less  than  1  decreases  it,  and  greater  than  1
              increases  it;  a  negative  gain  inverts  the  audio signal in
              addition to adjusting its volume.

              When type is dB, a gain of 0 leaves the volume  unchanged,  less
              than 0 decreases it, and greater than 0 increases it.

              See [4] for a detailed discussion on electrical (and hence audio
              signal) voltage and power ratios.

              Beware of Clipping when the increasing the volume.

              The gain and the type parameters can be concatenated if desired,
              e.g.  vol 10dB.

              An  optional  limitergain value can be specified and should be a
              value much less than 1 (e.g. 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 the audio that needed to be limited.

              See       also       compand       for      a      dynamic-range
              compression/expansion/limiting effect.

   Deprecated Effects
       The following effects have been renamed  or  have  their  functionality
       included  in  another effect.  They continue to work in this version of
       SoX but may be removed in future.

       avg [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
              Reduce the number of  audio  channels  by  mixing  or  selecting
              channels,  or  duplicate  channels  to  increase  the  number of
              channels.  This effect is just an alias of the mixer effect  and
              is retained for backwards compatibility only.

       highp frequency
              Apply  a high-pass filter.  This effect is just an alias for the
              highpass effect used with its -1  option;  it  is  retained  for
              backwards compatibility only.

       lowp frequency
              Apply  a  low-pass filter.  This effect is just an alias for the
              lowpass effect used with its  -1  option;  it  is  retained  for
              backwards compatibility only.

       mask [depth]
              This  effect  is  just a deprecated alias for the dither effect,
              left for historical reasons.

       pick [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
              Pick a subset of channels to be copied  into  the  output  file.
              This effect is just an alias of the mixer effect and is retained
              for backwards compatibility only.

       pitch shift [width interpolate fade]
              Change the audio pitch (but not its duration).  This  effect  is
              equivalent  to  the  key  effect with search set to zero, so its
              results are comparitively poor; it  is  retained  for  backwards
              compatibility only.

              Change  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  pitch.   interpolate  option, can be cubic or linear.
              Default is cubic.  The fade option, can be cos, hamming,  linear
              or trapezoid; the default is cos.

       rate   Does  the  same  as  resample  with no parameters; it exists for
              backwards compatibility.

       stretch factor [window fade shift fading]
              Change the audio duration (but not its pitch).  This  effect  is
              equivalent to the tempo effect with (factor inverted and) search
              set to zero, so  its  results  are  comparitively  poor;  it  is
              retained for backwards compatibility only.

              factor  of stretching: >1 lengthen, <1 shorten duration.  window
              size is in ms.  Default is 20ms.  The fade option, can be ‘lin’.
              shift  ratio, in [0 1].  Default depends on stretch factor. 1 to
              shorten, 0.8 to lengthen.  The fading ratio, in  [0  0.5].   The
              amount of a fade’s default depends on factor and shift.

       vibro speed [depth]
              This  is  a deprecated alias for the tremolo effect.  It differs
              in that the depth parameter ranges from 0 to 1 and  defaults  to
              0.5.

SEE ALSO

       sox(1), soxformat(7), libsox(3), soxexam(7), wget(1)

       The SoX web page at http://sox.sourceforge.net

   References
       [1]    R. Bristow-Johnson, Cookbook formulae for audio EQ biquad filter
              coefficients, http://musicdsp.org/files/Audio-EQ-Cookbook.txt

       [2]    Wikipedia, Q-factor, http://en.wikipedia.org/wiki/Q_factor

       [3]    Scott         Lehman,         Flanging,          http://harmony-
              central.com/Effects/Articles/Flanging

       [4]    Wikipedia, Decibel, http://en.wikipedia.org/wiki/Decibel

       [5]    Richard  Furse,  Linux  Audio  Developers  Simple  Plugin  API,
              http://www.ladspa.org

       [6]    Richard Furse, Computer Music Toolkit, http://www.ladspa.org/cmt

       [7]    Steve Harris, LADSPA plugins, http://plugin.org.uk

AUTHORS

       Chris  Bagwell  (cbagwell@users.sourceforge.net).   Other  authors  and
       contributors are listed in the AUTHORS file that  is  distributed  with
       the source code.