SoX(7)				Sound eXchange				SoX(7)



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

DESCRIPTION
       This manual describes SoX audio effects; the SoX manual set starts with
       sox(1).

       In addition to converting and playing audio files, SoX can be  used  to
       invoke a number of audio 'effects'.  Multiple effects may be applied by
       specifying them one after another at the end of the SoX	command	 line.
       Note  that  applying  multiple  effects in real-time (i.e. when playing
       audio) is likely to need a high performance  computer;  stopping	 other
       applications may alleviate performance issues should they occur.

       Some  of the SoX effects are primarily intended to be applied to a sin-
       gle instrument or 'voice'.  To facilitate this, the  remix  effect  and
       the  global  SoX option -M can be used to isolate then recombine tracks
       from a multi-track recording.

       In the descriptions that follow, brackets [ ] are used to denote param-
       eters  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.  Where applicable, default values for
       optional parameters are shown in parenthesis ( ).

       The following parameters are used with, and have the same meaning  for,
       several effects:

       centre[k]
	      See frequency.

       frequency[k]
	      A frequency in Hz, or, if appended with 'k', kHz.

       gain   A power gain in dB.  Zero gives no gain; less than zero gives an
	      attenuation.

       width[h|k|o|q]
	      Used to specify the band-width of a filter.  A number of differ-
	      ent  methods  to specify the width are available (though not all
	      for every effect); one of the characters shown may  be  appended
	      to select the desired method as follows:

				  +-----------------------+
				  |	Method	  Notes	  |
				  |h	  Hz		  |
				  |k	 kHz		  |
				  |o   Octaves		  |
				  |q   Q-factor	  See [2] |
				  +-----------------------+
	      For  each	 effect	 that  uses this parameter, the default method
	      (i.e. if no character is appended) is the	 one  that  it	listed
	      first in the effect's first line of description.

       To see if SoX has support for an optional effect, enter sox -h and look
       for its name under the list: 'EFFECTS'.

   SOX EFFECTS
       allpass frequency[k] width[h|k|o|q]
	      Apply a two-pole all-pass filter with central frequency (in  Hz)
	      frequency,  and  filter-width width.  An all-pass filter changes
	      the audio's frequency to phase relationship without changing its
	      frequency to amplitude relationship.  The filter is described in
	      detail in [1].

	      This effect supports the --plot global option.

       band [-n] center[k] [width[h|k|o|q]]
	      Apply a band-pass filter.	 The frequency	response  drops	 loga-
	      rithmically  around  the	center frequency.  The width parameter
	      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[k] width[h|k|o|q]
	      Apply a two-pole Butterworth  band-pass  or  band-reject	filter
	      with  central  frequency	frequency,  and (3dB-point) band-width
	      width.  The -c option applies only to  bandpass  and  selects  a
	      constant skirt gain (peak gain = Q) instead of the default: con-
	      stant 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[k] width[h|k|o|q]
	      Apply a band-reject filter.  See the description of the bandpass
	      effect for details.

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

	      gain  gives  the	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 is the filter's shelf transition.  In
	      addition to the common  width  specification  methods  described
	      above,  'slope'  (the  default,  or if appended with 's') may be
	      used.  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.	 This can make a single	 vocal
	      sound like a chorus, but can also be applied to instrumentation.

	      Chorus resembles an echo effect with a short delay, but  whereas
	      with echo the delay is constant, with chorus, it is varied using
	      sinusoidal  or  triangular  modulation.	The  modulation	 depth
	      defines  the range the modulated delay is played before or after
	      the delay. Hence the delayed sound will sound slower or  faster,
	      that is the delayed sound tuned around the original one, like in
	      a chorus where some vocals are slightly off key.	 See  [3]  for
	      more discussion of the chorus effect.

	      Each  four-tuple	parameter  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 modula-
	      tion is either sinusoidal (-s) or triangular (-t).  Gain-out  is
	      the volume of the output.

	      A	 typical delay is around 40ms to 60ms; the modulation speed is
	      best near 0.25Hz and the modulation depth around 2ms.  For exam-
	      ple, a single delay:

		   play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t

	      Two delays of the original samples:

		   play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
			 60 0.32 0.4 1.3 -s

	      A fuller sounding chorus (with three additional delays):

		   play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t \
			 60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s


       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	 aver-
	      aged to determine its volume; attacks refer to increases in vol-
	      ume and decays refer to decreases.   For	most  situations,  the
	      attack  time  (response  to  the music getting louder) should be
	      shorter than the decay time because the human ear is more sensi-
	      tive  to	sudden	loud music than sudden soft music.  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 possi-
	      ble  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  com-
	      panded  (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	amount 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 exam-
	      ple,  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).

	      The  following  example  might  be used to make a piece of music
	      with both quiet and loud passages suitable for listening to in a
	      noisy environment such as a moving vehicle:

		   sox asz.au asz-car.au compand 0.3,1 6:-70,-60,-20 -5 -90 0.2

	      The  transfer  function ('6:-70,...') says that very soft sounds
	      (below -70dB) will remain unchanged.  This will stop the compan-
	      der  from	 boosting  the	volume	on  'silent'  passages such as
	      between movements.  However, sounds in the range	-60dB  to  0dB
	      (maximum	volume) will be boosted so that the 60dB dynamic range
	      of the original music will be  compressed	 3-to-1	 into  a  20dB
	      range, which is wide enough to enjoy the music but narrow enough
	      to get around the road noise.  The '6:'  selects	6dB  soft-knee
	      companding.  The -5 (dB) output gain is needed to avoid clipping
	      (the number is inexact, and  was	derived	 by  experimentation).
	      The  -90	(dB)  for the initial volume will work fine for a clip
	      that starts with near silence, and the delay  of	0.2  (seconds)
	      has  the	effect	of  causing  the compander to react a bit more
	      quickly to sudden volume changes.

	      See also mcompand for a multiple-band companding effect.

       contrast [enhancement-amount (75)]
	      Comparable with compression, this effect modifies an audio  sig-
	      nal  to  make  it sound louder.  enhancement-amount controls the
	      amount of the enhancement and is a number in  the	 range	0-100.
	      Note  that enhancement-amount = 0 still gives a significant con-
	      trast enhancement.  contrast is often used in  conjunction  with
	      the norm effect as follows:

		   sox infile outfile norm -i contrast


       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.

	      An  alternative  approach to removing a DC offset (albeit with a
	      short delay) is to use the highpass filter effect at a frequency
	      of say 10Hz; i.e.

		   sox -n out.au synth 5 sin %0 50 highpass 10


       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.

       delay {length}
	      Delay one or more audio channels.	 length can specify a time or,
	      if  appended  with  an  's',  a number of samples.  For example,
	      delay 1.5 0 0.5 delays the first channel	by  1.5	 seconds,  the
	      third channel by 0.5 seconds, and leaves the second channel (and
	      any other channels that may be present) un-delayed.  The follow-
	      ing (one long) command plays a chime sound:

		   play -n synth sin %-21.5 sin %-14.5 sin %-9.5 sin %-5.5 \
		     sin %-2.5 sin %2.5 gain -5.4 fade h 0.008 2 1.5 \
		     delay 0 .27 .54 .76 1.01 1.3 remix - fade h 0.1 2.72 2.5


       dither [depth]
	      Apply dithering to the audio.  Dithering deliberately adds digi-
	      tal white noise to the signal in order to mask audible quantiza-
	      tion  effects  that  can occur if the output sample size is less
	      than 24 bits.  By default, the amount of noise added is    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
	      44.1kHz stereo (i.e. audio CD format) audio so  that  when  lis-
	      tened  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.geoci-
	      ties.com/beinges for a full explanation.

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

	      Each  given 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.  For example: This will make it sound as
	      if there are twice as many instruments as are actually playing:

		   play lead.aiff echo 0.8 0.88 60 0.4

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

		   play lead.aiff echo 0.8 0.88 6 0.4

	      A	 longer delay will sound like an open air concert in the moun-
	      tains:

		   play lead.aiff echo 0.8 0.9 1000 0.3

	      One mountain more, and:

		   play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25


       echos gain-in gain-out <delay decay>
	      Add a sequence of echoes 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.

	      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; a
	      single echos has the same effect as a single echo.

	      The sample will be bounced twice in symmetric echos:

		   play lead.aiff echos 0.8 0.7 700 0.25 700 0.3

	      The sample will be bounced twice in asymmetric echos:

		   play lead.aiff echos 0.8 0.7 700 0.25 900 0.3

	      The sample will sound as if played in a garage:

		   play lead.aiff echos 0.8 0.7 40 0.25 63 0.3


       equalizer frequency[k] width[q|o|h|k] 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, 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 speci-
	      fied.  If the file length can be determined from the input  file
	      header and length-changing effects are not in effect, then 0 may
	      be specified for stop-time to indicate the usual case of a fade-
	      out that ends at the end of the input audio stream.

	      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 exam-
	      ple '8000s').

	      An  optional  type  can be specified to change the type of enve-
	      lope.  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 logarithmic.

       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  fre-
	      quency 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 cut-off, smaller windows a more gradual  cut-off.

	      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 cut-off,
	      don't use low beta's, use a longer sample window. A Nuttall win-
	      dow  is  selected	 by specifying any 'beta' <= 2, and the Nuttall
	      window has somewhat steeper cut-off than the default Kaiser win-
	      dow.  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.

       flanger [delay depth regen width speed shape phase interp]
	      Apply  a	flanging  effect to the audio.	See [3] for a detailed
	      description of flanging.

	      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.		       |
	     +-----------------------------------------------------------------+
       gain dB-gain
	      Apply an amplification or an attenuation to  the	audio  signal.
	      This is an alias for the vol effect - handy for those who prefer
	      to work in dBs by default.

       highpass|lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
	      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;	the  default  is  Q  =	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.  If found, the environment	 vari-
	      ble LADSPA_PATH will be used as search path for plugins.

       lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
	      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[k]
	      "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.

	      For  example,  the following (one long) command shows how multi-
	      band companding is typically used in FM radio:

		   play track1.wav gain -3 filter 8000- 32 100 mcompand \
		   "0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
		   "0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
		   "0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
		   "0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
		   "0,0.025 -38,-31,-28,-28,-0,-25" \
		   gain 15 highpass 22 highpass 22 filter -17500 256 \
		   gain 9 lowpass -1 17801

	      The audio file is played with a simulated	 FM  radio  sound  (or
	      broadcast	 signal	 condition if the lowpass filter at the end is
	      skipped).	 Note that the pipeline is set up with	US-style  75us
	      preemphasis.

	      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 chan-
	      nels, or increase the number of channels	by  duplicating	 chan-
	      nels.   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 chan-
	      nels  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  spe-
	      cific  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, sep-
	      arated 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	 |
		  +------------------------------------------------------+
	      See  also	 remix	for a mixing effect that handles any number of
	      channels.

       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 back-
	      ground 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


       norm [-i] [level]
	      Normalise	 audio to 0dB FSD or to a given level relative to 0dB.
	      Requires	temporary  file	 space	to  store  the	audio  to   be
	      normalised.

	      To create a normalised copy of an audio file,

		   sox infile outfile norm

	      can  be used, though note that if 'infile' has a simple encoding
	      (e.g.  PCM), then

		   sox infile outfile vol `sox infile -n stat -v 2>&1`

	      (on systems that support this construct)	might  be  quicker  to
	      execute  (though	perhaps	 not to type!) as it doesn't require a
	      temporary file.

	      For a more complex example, suppose that 'effect1' performs some
	      unknown or unpredictable attenuation and that 'effect2' requires
	      up to 10dB of headroom, then

		   sox infile outfile effect1 norm -10 effect2 norm

	      gives both effect2 and the output file the highest possible sig-
	      nal levels.

	      Normally,	 audio is normalised based on the level of the channel
	      with the highest peak level, which means that whilst  all	 chan-
	      nels  are	 adjusted,  only  one  channel	attains the normalised
	      level.  If the -i option is given, then each channel is  treated
	      individually and will attain the normalised level.

	      In  most	cases, norm -3 should be the maximum level used at the
	      output file (to leave headroom for  playback-resampling,	etc.).
	      See  also the discussions of clipping and Replay Gain in sox(1).

       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 sam-
	      ples.   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, spec-
	      ify  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 chan-
	      nels, 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.	See [3] for a detailed
	      description of phasing.

	      delay/decay/speed gives the delay in milliseconds and the	 decay
	      (relative	 to gain-in) with a modulation speed in Hz.  The modu-
	      lation is either sinusoidal  (-s)	  -  preferable	 for  multiple
	      instruments,  or	triangular  (-t)  - gives single instruments a
	      sharper phasing effect.  The decay should be less	 than  0.5  to
	      avoid  feedback,	and usually no less than 0.1.  Gain-out is the
	      volume of the output.

	      For example:

		   play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t

	      Gentler:

		   play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s

	      A popular sound:

		   play snare.flac phaser 0.89 0.85 1 0.24 2 -t

	      More severe:

		   play snare.flac phaser 0.6 0.66 3 0.6 2 -t


       rate [-q|-l|-m|-h|-v] [RATE[k]]
	      Change the audio sampling rate (i.e. resample the audio) using a
	      quality level as follows:

		+-----------------------------------------------------------+
		|	 Quality      BW %     Rej dB	  Typical Use	    |
		|-q   quick & dirty   n/a    ?30 @ Fs/4	  playback on	    |
		|					  ancient hardware  |
		|-l	   low	       80	100	  playback on old   |
		|					  hardware	    |
		|-m	 medium	       99	100	  audio playback    |
		|-h	  high	       99	125	  16-bit mastering  |
		|					  (use with dither) |
		|-v	very high      99	175	  24-bit mastering  |
		+-----------------------------------------------------------+
	      where BW % is the percentage of the audio band that is preserved
	      (based  on the 3dB point) during sample rate conversion, and Rej
	      dB is the level of noise rejection.  The default	quality	 level
	      is  'high' (-h).	The -q algorithm uses cubic interpolation; the
	      others use linear-phase bandwidth-limited interpolation.

	      This effect is invoked automatically if SoX's -r	option	speci-
	      fies  a  rate  that  is  different to that of the input file(s).
	      Alternatively, this effect may be invoked with the  output  rate
	      parameter RATE and SoX's -r option need not be given.  For exam-
	      ple, the following two commands are equivalent:

		   sox input.au -r 48k output.au bass -3
		   sox input.au output.au bass -3 rate 48k

	      though the second command is more flexible as it allows  a  rate
	      quality  option  to  be  given,  and it allows the effects to be
	      ordered arbitrarily.

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

       remix [-a|-m|-p] <out-spec>
	      out-spec	= in-spec{,in-spec} | 0
	      in-spec	= [in-chan][-[in-chan2]][vol-spec]
	      vol-spec	= p|i|v[volume]

	      Select  and mix input audio channels into output audio channels.
	      Each output channel is specified, in turn, by a given  out-spec:
	      a list of contributing input channels and volume specifications.

	      Note that 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).

	      An  out-spec  contains comma-separated input channel-numbers and
	      hyphen-delimited channel-number ranges; alternatively, 0 may  be
	      given to create a silent output channel.	For example,

		   sox input.au output.au remix 6 7 8 0

	      creates  an output file with four channels, where channels 1, 2,
	      and 3 are copies of channels 6, 7, and 8 in the input file,  and
	      channel 4 is silent.  Whereas

		   sox input.au output.au remix 1-3,7 3

	      creates  a  stereo  output file where the left channel is a mix-
	      down of input channels 1, 2, 3, and 7, and the right channel  is
	      a copy of input channel 3.

	      Where  a	range of channels is specified, the channel numbers to
	      the left and right of the hyphen are optional and default	 to  1
	      and to the number of input channels respectively. Thus

		   sox input.au output.au remix -

	      performs a mix-down of all input channels to mono.

	      By  default,  where an output channel is mixed from multiple (n)
	      input channels, each input channel will be scaled by a factor of
	      /n.   Custom  mixing  volumes  can  be set by following a given
	      input channel or range of input channels with a vol-spec (volume
	      specification).  This is one of the letters p, i, or v, followed
	      by a volume number, the meaning of which depends	on  the	 given
	      letter and is defined as follows:

		      Letter   Volume number	    Notes
			p      power adjust in dB   0 = no change
			i      power adjust in dB   As 'p', but invert
						    the audio
			v      voltage multiplier   1 = no change, 0.5
						    ? 6dB attenuation,
						    2 ? 6dB gain, -1 =
						    invert

	      If  an out-spec includes at least one vol-spec then, by default,
	      /n scaling is not applied to any other  channels	 in  the  same
	      out-spec (though may be in other out-specs).  The -a (automatic)
	      option however, can be given to retain the automatic scaling  in
	      this case.  For example,

		   sox input.au output.au remix 1,2 3,4v0.8

	      results in channel level multipliers of 0.5,0.5 1,0.8, whereas

		   sox input.au output.au remix -a 1,2 3,4v0.8

	      results in channel level multipliers of 0.5,0.5 0.5,0.8.

	      The  -m  (manual)	 option	 disables all automatic volume adjust-
	      ments, so

		   sox input.au output.au remix -m 1,2 3,4v0.8

	      results in channel level multipliers of 1,1 1,0.8.

	      The volume number is optional and omitting it corresponds to  no
	      volume change; however, the only case in which this is useful is
	      in conjunction with i.  For example, if input.au is stereo, then

		   sox input.au output.au remix 1,2i

	      is a mono equivalent of the oops effect.

	      If  the  -p  option  is given, then any automatic /n scaling is
	      replaced by /?n ('power') scaling; this gives a louder mix  but
	      one that might occasionally clip.

				    *	     *	      *

	      One  typical  use	 of the remix effect is to split an audio file
	      into a set of files, each	 containing  one  of  the  constituent
	      channels	(in order to perform subsequent processing on individ-
	      ual audio	 channels).   Where  more  than	 a  few	 channels  are
	      involved, a script such as the following is useful:

	      #!/bin/sh			       # This is a Bourne shell script
	      chans=`soxi -c "$1"`
	      while [ $chans -ge 1 ]; do
		chans0=`printf %02i $chans`   # 2 digits hence up to 99 chans
		out=`echo "$1"|sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
		sox "$1" "$out" remix $chans
		chans=`expr $chans - 1`
	      done

	      If a file input.au containing six audio channels were given, the
	      script would produce six output files: input-01.au, input-02.au,
	      ..., input-06.au.

	      See also mixer and swap for similar effects.

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

       reverb [-w|--wet-only] [reverberance (50%) [HF-damping (50%)
	      [room-scale (100%) [stereo-depth (100%)
	      [pre-delay (0ms) [wet-gain (0dB)]]]]]]

	      Add reverberation to the audio using the	'freeverb'  algorithm.
	      A	 reverberation effect is sometimes desirable for concert halls
	      that are too small or contain so many  people  that  the	hall's
	      natural  reverberance is diminished.  Applying a small amount of
	      stereo reverb to a (dry) mono signal will usually make it	 sound
	      more  natural.  See [3] for a detailed description of reverbera-
	      tion.

	      Note that this effect increases both the volume and  the	length
	      of the audio, so to prevent clipping in these domains, a typical
	      invocation might be:

		   play dry.au gain -3 pad 0 3 reverb


       reverse
	      Reverse the audio completely.  Requires temporary file space  to
	      store the audio 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	 peri-
	      ods 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 sec-
	      ond 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 mid-
	      dle 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 posi-
	      tive 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 sam-
	      ples.  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).

	      The following example shows how this effect can be used to start
	      a	 recording  that does not contain the delay at the start which
	      usually occurs between 'pressing	the  record  button'  and  the
	      start of the performance:

		   rec parameters filename other-effects silence 1 5 2%


       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 resampling with the
	      rate effect using its default quality/speed.  For higher quality
	      or higher speed resampling, in addition  to  the	speed  effect,
	      specify the rate effect with the desired quality option.

       spectrogram [options]
	      Create  a	 spectrogram  of  the audio.  This effect is optional;
	      type sox --help and check the list of supported effects  to  see
	      if it has been included.

	      The  spectrogram is rendered in a Portable Network Graphic (PNG)
	      file, and shows time in the X-axis, frequency in the Y-axis, and
	      audio  signal magnitude in the Z-axis.  Z-axis values are repre-
	      sented by the colour (or intensity) of the  pixels  in  the  X-Y
	      plane.

	      This  effect  supports only one channel; for multi-channel input
	      files, use either SoX's -c 1 option with	the  output  file  (to
	      obtain  a spectrogram on the mix-down), or the remix n effect to
	      select a particular channel.  Be	aware  though,	that  both  of
	      these methods affect the audio in the effects chain.

	      -x num X-axis  pixels/second,  default  100.   This controls the
		     width of the spectrogram; num can be  from	 1  (low  time
		     resolution)  to  5000 (high time resolution) and need not
		     be an integer.  SoX may make a slight adjustment  to  the
		     given  number for processing quantisation reasons; if so,
		     SoX will report the actual	 number	 used  (viewable  when
		     --verbose is in effect).

		     The  maximum  width  of the spectrogram is 999 pixels; if
		     the audio length and the given -x number  are  such  that
		     this  would  be  exceeded,	 then the spectrogram (and the
		     effects chain) will be truncated.	To move	 the  spectro-
		     gram  to  a point later in the audio stream, first invoke
		     the trim effect; e.g.

		       sox audio.ogg -n trim 1:00 spectrogram

		     starts the spectrogram at 1 minute through the audio.

	      -y num Y-axis resolution (1 - 4), default 2.  This controls  the
		     height  of	 the  spectrogram; num can be from 1 (low fre-
		     quency resolution) to 4 (high frequency resolution).  For
		     values  greater  than  2,	the resulting image may be too
		     tall to display on the screen; if so, a graphic manipula-
		     tion  package (such as ImageMagick(1)) can be used to re-
		     size the image.

		     To increase the frequency resolution  without  increasing
		     the  height  of  the  spectrogram, the rate effect may be
		     invoked to reduce the sampling rate of the signal	before
		     invoking spectrogram; e.g.

		       sox audio.ogg -r 4k -n rate spectrogram

		     allows  detailed analysis of frequencies up to 2kHz (half
		     the sampling rate).

	      -z num Z-axis (colour) range in dB, default 120.	This sets  the
		     dynamic-range  of	the  spectrogram  to  be  -num dBFS to
		     0 dBFS.  Num  may	range  from  20	 to  180.   Decreasing
		     dynamic-range effectively increases the 'contrast' of the
		     spectrogram display, and vice versa.

	      -Z num Sets the upper limit of the Z-axis in dBFS.   A  negative
		     num  effectively  increases the 'brightness' of the spec-
		     trogram display, and vice versa.

	      -q num Sets the Z-axis quantisation, i.e. the number of  differ-
		     ent  colours  (or	intensities) in which to render Z-axis
		     values.   A  small	 number	  (e.g.	  4)   will   give   a
		     'poster'-like  effect  making it easier to discern magni-
		     tude bands of similar level.  Smaller numbers  also  usu-
		     ally result in smaller PNG files.	The number given spec-
		     ifies the number of colours  to  use  inside  the	Z-axis
		     range; two colours are reserved to represent out-of-range
		     values.

	      -w name
		     Window: Hann (default), Hamming, Bartlett, Rectangular or
		     Kaiser.   The  spectrogram is produced using the Discrete
		     Fourier Transform (DFT) algorithm.	 A significant parame-
		     ter to this algorithm is the choice of 'window function'.
		     By default, SoX uses the Hann window which has good  all-
		     round  frequency-resolution and dynamic-range properties.
		     For  better  frequency  resolution	 (but  lower  dynamic-
		     range), select a Hamming window; for higher dynamic-range
		     (but poorer frequency-resolution), select a  Kaiser  win-
		     dow.   Bartlett  and  Rectangular windows are also avail-
		     able.  Selecting a window other than  Hann	 will  usually
		     require a corresponding -z setting.

	      -s     Allow  slack  overlapping	of  DFT windows.  This can, in
		     some cases, increase image	 sharpness  and	 give  greater
		     adherence to the -x value, but at the expense of a little
		     spectral loss.

	      -m     Creates a monochrome spectrogram (the default is colour).

	      -h     Selects  a	 high-colour  palette - less visually pleasing
		     than the default colour palette, but it may make it  eas-
		     ier to differentiate different levels.  If this option is
		     used in conjunction with -m, the result will be a	hybrid
		     monochrome/colour palette.

	      -p num Permute  the  colours in a colour or hybrid palette.  The
		     num parameter (from 1 to 6) selects the permutation.

	      -l     Creates a 'printer friendly'  spectrogram	with  a	 light
		     background (the default has a dark background).

	      -a     Suppress  the  display  of the axis lines.	 This is some-
		     times useful in helping to discern artefacts at the spec-
		     trogram edges.

	      -t text
		     Set  the image title - text to display above the spectro-
		     gram.

	      -c text
		     Set the image comment - text to display below and to  the
		     left of the spectrogram.

	      -o text
		     Name  of  the spectrogram output PNG file, default 'spec-
		     trogram.png'.

	      For example, let's see what the spectrogram of a swept  triangu-
	      lar wave looks like:

		   sox -n -n synth 6 tri 10k:14k spectrogram -z 100 -w k

	      For the ability to perform off-line processing of spectral data,
	      see the stat effect.

       splice  { position[,excess[,leeway]] }
	      Splice together audio sections.  This effect provides two things
	      over simple audio concatenation: a (usually short) cross-fade is
	      applied at the join, and a wave similarity comparison is made to
	      help determine the best place at which to make the join.

	      To  perform  a  splice,  first use the trim effect to select the
	      audio sections to be joined together.  As when performing a tape
	      splice,  the  end	 of  the  section to be spliced onto should be
	      trimmed with a small excess (default  0.005  seconds)  of	 audio
	      after  the ideal joining point.  The beginning of the audio sec-
	      tion to splice on should be trimmed with the same excess (before
	      the  ideal  joining  point),  plus an additional leeway (default
	      0.005 seconds).  SoX should then be invoked with the  two	 audio
	      sections	as  input  files  and the splice effect given with the
	      position at which to perform the splice - this is length of  the
	      first audio section (including the excess).

	      For  example, a long song begins with two verses which start (as
	      determined e.g. by using the play command with the trim  (start)
	      effect)  at times 0:30.125 and 1:03.432.	The following commands
	      cut out the first verse:

		   sox too-long.au part1.au trim 0 30.130

	      (5 ms excess, after the first verse starts)

		   sox long.au part2.au trim 1:03.422

	      (5 ms excess plus 5 ms leeway, before the second verse starts)

		   sox part1.au part2.au just-right.au splice 30.130

	      Provided your arithmetic is good enough, multiple splices can be
	      performed with a single splice invocation.  For example:

	      #!/bin/sh
	      # Audio Copy and Paste Over
	      # acpo infile copy-start copy-stop paste-over-start outfile
	      # All times measured in samples.
	      rate=`soxi -r "$1"`
	      e=`expr $rate '*' 5 / 1000`  # Using default excess
	      l=$e			   # and leeway.
	      sox "$1" piece.au trim `expr $2 - $e - $l`s \
		   `expr $3 - $2 + $e + $l + $e`s
	      sox "$1" part1.au trim 0 `expr $4 + $e`s
	      sox "$1" part2.au trim `expr $4 + $3 - $2 - $e - $l`s
	      sox part1.au piece.au part2.au "$5" splice \
		   `expr $4 + $e`s \
		   `expr $4 + $e + $3 - $2 + $e + $l + $e`s

	      In  the above Bourne shell script, two splices are used to 'copy
	      and paste' audio.

				    *	     *	      *

	      It is also possible to use this effect to perform general cross-
	      fades, e.g. to join two songs.  In this case, excess would typi-
	      cally be an number of seconds, and leeway should be set to zero.

       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
	      in sox(1) 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 maximum value  of  a  signed  long
	      integer (7fffffff in hexadecimal).  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	 exam-
	      ple,  swap 2 2 will overwrite channel 1 with channel 2; creating
	      a stereo file with both channels containing the same audio.

	      See also the remix effect.

       synth [len] {[type]  [combine]  [[%]freq[k][:|+|/|-[%]freq2[k]]]	 [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  cas-
	      caded  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 syn-
	      thesised 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 nor-
	      mally  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, 48kHz, 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 cas-
	      caded 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 (0dBFS),
	      which means that there is a high chance of clipping  when	 using
	      the  audio subsequently, so in most cases, you will want to fol-
	      low this effect with the gain effect to prevent this  from  hap-
	      pening. (See also Clipping in sox(1).)

	      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 and the generated tone will be swept
	      between the given frequencies.  The two given  frequencies  must
	      be  separated  by	 one  of the characters ':', '+', '/', or '-'.
	      This character is used to specify the sweep function as follows:

	      :	     Linear:  the  tone will change by a fixed number of hertz
		     per second.

	      +	     Square: a second-order function is	 used  to  change  the
		     tone.

	      /	     Exponential:  the	tone  will change by a fixed number of
		     semitones per second.

	      -	     Exponential: as '/', but initial phase always  zero,  and
		     stepped (less smooth) frequency changes.

	      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,	trian-
	      gle, 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' algo-
	      rithm.   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  overlap-
	      ping  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, so e.g.
	      1.1 speeds up the tempo by 10%, and 0.9 slows it down by 10%.

	      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 mil-
	      liseconds (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, speed for an effect that
	      changes tempo and key together,  and  key	 for  an  effect  that
	      changes key without changing tempo.

       treble gain [frequency[k] [width[s|h|k|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  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 inter-
	      preted, according to the given type,  as	follows:  if  type  is
	      amplitude (or is omitted), then gain is an amplitude (i.e. volt-
	      age 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 addi-
	      tion 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/lim-
	      iting 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.

       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 comparatively 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.

       polyphase [-w nut|ham] [-width n] [-cut-off c]
	      Change  the sampling rate using 'polyphase interpolation', a DSP
	      algorithm.  polyphase copes with only certain rational  fraction
	      resampling ratios, and, compared with the rate effect, is gener-
	      ally slow, memory intensive, and has poorer stop-band rejection.

	      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 -cut-off value (c) specifies the filter cut-off frequency in
	      terms of fraction of  frequency  bandwidth,  also	 know  as  the
	      Nyquist frequency.  See the resample effect for further informa-
	      tion 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 rate, rabbit and resample for other sample-rate chang-
	      ing effects.

       rabbit [-c0|-c1|-c2|-c3|-c4]
	      Change the sampling rate	using  libsamplerate,  also  known  as
	      'Secret  Rabbit  Code'.	This  effect  is  optional and, due to
	      licence issues, is not included in all versions  of  SoX.	  Com-
	      pared with the rate effect, rabbit is very slow.

	      See  http://www.mega-nerd.com/SRC for details of the algorithms.
	      Algorithms 0 through 2 are progressively faster and lower	 qual-
	      ity  versions  of the sinc algorithm; the default is -c0.	 Algo-
	      rithm 3 is zero-order hold, and 4 is linear interpolation.

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

       resample [-qs|-q|-ql] [rolloff [beta]]
	      Change the sampling  rate	 using	simulated  analog  filtration.
	      Compared	with the rate effect, resample is slow, and has poorer
	      stop-band rejection.  Only the low quality option works with all
	      resampling ratios.

	      By  default,  linear interpolation of the filter coefficients 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 insuf-
	      ficient 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  coeffi-
	      cients,  resulting  in about 24 bits precision.  The -qs, -q, or
	      -ql options specify increased accuracy at the cost of lower exe-
	      cution  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 proportion-
	      ally 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.	 Logi-
	      cally,  this  is because the A/D converter needs at least 2 sam-
	      ples to detect 1 cycle at the  Nyquist  frequency.   Frequencies
	      higher  then  the Nyquist will actually appear as lower frequen-
	      cies 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  sam-
	      ple  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
	      cut-off  is, with closer being better.  When increasing the sam-
	      ple rate of an audio file you would not expect to have any  fre-
	      quencies	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  fre-
	      quency  to use a highpass filter to remove these artifacts, with
	      closer also being better.

	      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	 filter
	      effect.

	      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.

	      See also rate, polyphase and rabbit for other sample-rate chang-
	      ing effects.  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.

       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  comparatively	 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.

SEE ALSO
       sox(1), soxi(1), soxformat(7), libsox(3),

       The SoX web page at http://sox.sourceforge.net
       SoX scripting examples at http://sox.sourceforge.net/Docs/Scripts

   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,	  Effects    Explained,	   http://harmony-cen-
	      tral.com/Effects/effects-explained.html

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

       [5]    Richard  Furse,  Linux  Audio  Developer's  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 con-
       tributors are listed in the AUTHORS file that is distributed  with  the
       source code.



soxeffect			 July 27, 2008				SoX(7)
