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===channels CHANNELS ===
==== channels CHANNELS ====
Invoke a simple algorithm to change the number of channels in the audio signal to the given n um-
Invoke a simple algorithm to change the number of channels in the audio signal to the given number CHANNELS: mixing if  decreasing  the  number  of  channels  or  duplicating  if  increasing  the number of channels.
ber CHANNELS: m ixing if  decreasing  the  number  of  channels  or  duplicating  if  increasing  the
The channels effect  is  invoked a utomatically  if  SoX’s −c option  specifies  a  number  of  channels that  is  different  to  that  of  the  input  file(s). Alternatively, i f t his  effect  is  given e xplicitly, t hen
number of channels.
The channels effect  is  invoked a utomatically  if  SoX’s −c option  specifies  a  number  of  channels
that  is  different  to  that  of  the  input  file(s). Alternatively, i f t his  effect  is  given e xplicitly, t hen
SoX’s −c option need not be given.  For example, the following two c ommands are equivalent:
SoX’s −c option need not be given.  For example, the following two c ommands are equivalent:
sox input.wav −c 1 output.wav bass −b 24
sox input.wav −c 1 output.wav bass −b 24
sox input.wav output.wav bass −b 24 channels 1
sox input.wav output.wav bass −b 24 channels 1
though the second form is more flexible as it allows the effects to be ordered arbitrarily.
though the second form is more flexible as it allows the effects to be ordered arbitrarily.
See also remix for an effect that allows channels to be mixed/selected arbitrarily.
See also remix for an effect that allows channels to be mixed/selected arbitrarily.
chorus gain-in gain-out <delay decay speed depth −s | −t>
chorus gain-in gain-out <delay decay speed depth −s | −t>
Add a chorus effect to the audio. This can make a s ingle vocal sound like a c horus, but can also be
Add a chorus effect to the audio. This can make a single vocal sound like a chorus, but can also be applied to instrumentation.
applied to instrumentation.
Chorus  resembles  an  echo  effect  with  a  short  delay, b ut  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. H ence  the  delayed  sound  will sound  slower  or  faster, t hat  is  the  delayed  sound  tuned  around  the  original  one,  like in a c horus where some vocals are slightly off k ey. S ee [3] for more discussion of the chorus effect.
Chorus  resembles  an  echo  effect  with  a  short  delay, b ut  whereas  with  echo  the  delay  is  constant,
Each four-tuple parameter delay/decay/speed/depth gives t he delay in milliseconds and the decay (relative to g ain-in) with a modulation speed in Hz using depth in milliseconds. The modulation is either sinusoidal (−s) or t riangular (−t).  Gain-out is the volume of the output.
with chorus, it is varied using sinusoidal or triangular modulation. The modulation depth defines
A t ypical delay is around 40ms to 60ms; the modulation speed is best near 0.25Hz and the modu-lation depth around 2ms. For e xample, a single delay:
the  range  the  modulated  delay  is  played  before  or  after  the  delay. H ence  the  delayed  sound  will
sound  slower  or  faster, t hat  is  the  delayed  sound  tuned  around  the  original  one,  like in a c horus
where some vocals are slightly off k ey. S ee [3] for more discussion of the chorus effect.
Each four-tuple parameter delay/decay/speed/depth gives t he delay in milliseconds and the decay
(relative to g ain-in) with a modulation speed in Hz using depth in milliseconds. The modulation is
either sinusoidal (−s) or t riangular (−t).  Gain-out is the volume of the output.
A t ypical delay is around 40ms to 60ms; the modulation speed is best near 0.25Hz and the modu-
lation depth around 2ms. For e xample, a single delay:
play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 −t
play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 −t
Tw o delays of the original samples:
Two delays of the original samples:
play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 −t \
play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 −t \
60 0.32 0.4 1.3 −s
60 0.32 0.4 1.3 −s
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Compand (compress or expand) the dynamic range of the audio.
Compand (compress or expand) the dynamic range of the audio.
The attack and decay parameters  (in  seconds)  determine  the  time  over w hich  the  instantaneous
The attack and decay parameters  (in  seconds)  determine  the  time  over w hich  the  instantaneous
level of t he input signal is averaged to determine its volume; attacks refer to increases in volume
level of t he input signal is averaged to determine its volume; attacks refer to increases in volume and decays refer to decreases. For m ost situations, the attack time (response to the music getting louder) should be shorter than the decay time because the human ear is more sensitive  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. Ty pical values are 0.3,0.8 seconds.
and decays refer to decreases. For m ost situations, the attack time (response to the music getting
The second parameter is a list of points on the compander’s t ransfer function specified in dB relative to t he maximum possible signal amplitude. The input values must be in a strictly increasing order  but  the  transfer  function  does  not  have  to be m onotonically  rising. If  omitted,  the  value  of
louder) should be shorter than the decay time because the human ear is more sensitive  to s udden
out-dB1 defaults  to  the  same  value  as in-dB1; l ev els  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
loud  music  than  sudden  soft  music. Where  more  than  one  pair  of  attack/decay  parameters  are
transfer function meet will be rounded by the amount given.  Typical values for the transfer func- tion are 6:−70,−60,−20
specified, each input channel is companded separately and the number of pairs must agree with the
number of input channels. Ty pical values are 0.3,0.8 seconds.
The second parameter is a list of points on the compander’s t ransfer function specified in dB rela-
tive to t he maximum possible signal amplitude. The input values must be in a strictly increasing
order  but  the  transfer  function  does  not  have  to be m onotonically  rising. If  omitted,  the  value  of
out-dB1 defaults  to  the  same  value  as in-dB1; l ev els  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 amount given.  Typical values for the transfer func-
tion are 6:−70,−60,−20


   
   
The third (optional) parameter is an additional gain in dB to be applied at all points on the transfer
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.
function and allows easy adjustment of the overall gain.
The fourth (optional) parameter is an initial level to be a ssumed for each channel when compand- ing  starts. This  permits  the  user  to  supply  a  nominal  level i nitially, s o t hat,  for  example,  a  very
The fourth (optional) parameter is an initial level to be a ssumed for each channel when compand-
ing  starts. This  permits  the  user  to  supply  a  nominal  level i nitially, s o t hat,  for  example,  a  very
large gain is not applied to initial signal levels before the companding action has begun to operate:
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 compan-
it is quite probable that in such an event, the output would be severely clipped while the compan-
der gain properly adjusts itself. A t ypical value (for audio which is initially quiet) is −90 dB.
der gain properly adjusts itself. A t ypical 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
The fifth (optional) parameter is a delay in seconds. The input signal is analysed immediately to control  the  compander, b ut  it  is  delayed  before  being  fed  to  the  volume  adjuster. S pecifying  a delay approximately equal to the attack/decay times allows the compander to effectively operate in a ‘ predictive’ r ather than a reactive mode.  A typical value is 0.2 seconds.
control  the  compander, b ut  it  is  delayed  before  being  fed  to  the  volume  adjuster. S pecifying  a
delay approximately equal to the attack/decay times allows the compander to effectively operate in
a ‘ predictive’ r ather than a reactive mode.  A typical value is 0.2 seconds.
* * *
The following example might be used to make a p iece of music with both quiet and loud passages
The following example might be used to make a p iece of music with both quiet and loud passages
suitable for listening to in a noisy environment such as a moving vehicle:
suitable for listening to in a noisy environment such as a moving vehicle:
sox asz.wav asz-car.wav compand 0.3,1 6:−70,−60,−20 −5 −90 0.2
sox asz.wav asz-car.wav 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
The transfer function (‘6:−70,...’) says that very soft sounds (below − 70dB) will remain unchanged.  This will stop the compander 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
unchanged.  This will stop the compander from boosting the volume on ‘silent’ passages such as
20dB  range,  which  is  wide  enough  to  enjoy t he  music  but  narrow e nough  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 e xperimentation).  The −90 (dB) for the initial
between  movements. However,  sounds  in  the  range  −60dB  to  0dB  (maximum  volume)  will  be
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.
boosted  so  that  the  60dB  dynamic  range  of  the  original  music  will  be  compressed  3-to-1  into  a
In the next example, compand is being used as a noise-gate for when the noise is at a lower level than the signal:
20dB  range,  which  is  wide  enough  to  enjoy t he  music  but  narrow e nough  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 e xperimentation).  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.
In the next example, compand is being used as a noise-gate for when the noise is at a lower level
than the signal:
play infile compand .1,.2 −inf,−50.1,−inf,−50,−50 0 −90 .1
play infile compand .1,.2 −inf,−50.1,−inf,−50,−50 0 −90 .1
Here is another noise-gate, this time for when the noise is at a higher level t han the signal (making
Here is another noise-gate, this time for when the noise is at a higher level t han the signal (making it, in some ways, similar to squelch):
it, in some ways, similar to squelch):
play infile compand .1,.1 −45.1,−45,−inf,0,−inf 45 −90 .1
play infile compand .1,.1 −45.1,−45,−inf,0,−inf 45 −90 .1
This effect supports the −−plot global option (for the transfer function).
This effect supports the −−plot global option (for the transfer function).
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contrast [enhancement-amount(75)]
contrast [enhancement-amount(75)]
Comparable  with  compression,  this  effect  modifies  an  audio  signal  to  make it s ound  louder.
Comparable  with  compression,  this  effect  modifies  an  audio  signal  to  make it s ound  louder.
enhancement-amount controls  the  amount  of  the  enhancement  and  is  a  number  in  the  range
enhancement-amount controls  the  amount  of  the  enhancement  and  is  a  number  in  the  range 0−100.  Note that enhancement-amount = 0 s till gives a s ignificant contrast enhancement.
0−100.  Note that enhancement-amount = 0 s till gives a s ignificant contrast enhancement.
See also the compand and mcompand effects.
See also the compand and mcompand effects.
dcshift shift [limitergain]
dcshift shift [limitergain]
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headroom  and  hence  volume. The stat or stats effect  can  be  used  to  determine  if  a  signal  has  a
headroom  and  hence  volume. The stat or stats effect  can  be  used  to  determine  if  a  signal  has  a
DC offset.
DC offset.
The given dcshift value is a floating point number in the range of ±2 t hat indicates the amount to
The given dcshift value is a floating point number in the range of ±2 t hat indicates the amount to
shift the audio (which is in the range of ±1).
shift the audio (which is in the range of ±1).
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deemph
deemph
Apply Compact Disc (IEC 60908) de-emphasis (a treble attenuation shelving filter).
Apply Compact Disc (IEC 60908) de-emphasis (a treble attenuation shelving filter).
Pre-emphasis was applied in the mastering of some CDs issued in the early 1980s. These included
 
many c lassical  music  albums,  as  well  as  now s ought-after  issues  of  albums  by  The  Beatles,  Pink
 
Floyd and others. Pre-emphasis should be removed at p layback time by a de-emphasis filter in the
Pre-emphasis was applied in the mastering of some CDs issued in the early 1980s. These included many c lassical  music  albums,  as  well  as  now s ought-after  issues  of  albums  by  The  Beatles,  Pink
playback device.  However, not all modern CD players have this filter, a nd very few PC CD d rives
Floyd and others. Pre-emphasis should be removed at p layback time by a de-emphasis filter in the playback device.  However, not all modern CD players have this filter, a nd very few PC CD d rives have  it;  playing  pre-emphasised  audio  without  the  correct  de-emphasis  filter  results  in  audio  that
have  it;  playing  pre-emphasised  audio  without  the  correct  de-emphasis  filter  results  in  audio  that
sounds harsh and is far from what its creators intended.
sounds harsh and is far from what its creators intended.
With the deemph effect, it is possible to apply the necessary de-emphasis to audio that has been
With the deemph effect, it is possible to apply the necessary de-emphasis to audio that has been extracted from a pre-emphasised CD, and then either burn the de-emphasised audio to a new C D (which  will  then  play  correctly  on  any CD p layer),  or  simply  play  the  correctly  de-emphasised
extracted from a pre-emphasised CD, and then either burn the de-emphasised audio to a new C D
(which  will  then  play  correctly  on  any CD p layer),  or  simply  play  the  correctly  de-emphasised
audio files on the PC. For e xample:
audio files on the PC. For e xample:
sox track1.wav track1−deemph.wav deemph
sox track1.wav track1−deemph.wav deemph
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or simply
or simply
play track1.wav deemph
play track1.wav deemph
The de-emphasis filter is implemented as a biquad and requires the input audio sample rate to be
 
either  44.1kHz  or  48kHz. Maximum  deviation  from  the  ideal  response  is  only  0.06dB  (up  to
 
20kHz).
The de-emphasis filter is implemented as a biquad and requires the input audio sample rate to be either  44.1kHz  or  48kHz. Maximum  deviation  from  the  ideal  response  is  only  0.06dB  (up  to 20kHz).
This effect supports the −−plot global option.
This effect supports the −−plot global option.
See also the bass and treble shelving equalisation effects.
See also the bass and treble shelving equalisation effects.
delay {position(=)}
delay {position(=)}
Delay  one  or  more  audio  channels  such  that  they s tart  at  the  given position. F or  example, delay
Delay  one  or  more  audio  channels  such  that  they s tart  at  the  given position. F or  example, delay 1.5 +1 3 000s delays the first channel by 1.5 s econds, the second channel by 2.5 s econds (one sec-
1.5 +1 3 000s delays the first channel by 1.5 s econds, the second channel by 2.5 s econds (one sec-
ond  more  than  the  previous  channel),  the  third  channel  by  3000  samples,  and  leaves a ny  other channels  that  may  be  present  un-delayed. The  following  (one  long)  command  plays  a  chime sound:
ond  more  than  the  previous  channel),  the  third  channel  by  3000  samples,  and  leaves a ny  other
channels  that  may  be  present  un-delayed. The  following  (one  long)  command  plays  a  chime
sound:
play −n synth −j 3 sin %3 sin %−2 sin %−5 sin %−9 \
play −n synth −j 3 sin %3 sin %−2 sin %−5 sin %−9 \
sin %−14 sin %−21 fade h .01 2 1.5 delay \
sin %−14 sin %−21 fade h .01 2 1.5 delay \
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delay 0 .05 .1 .15 .2 .25 remix − fade 0 4 .1 norm −1
delay 0 .05 .1 .15 .2 .25 remix − fade 0 4 .1 norm −1
dither [−S | −s | −f filter] [ −a] [ −p precision]
dither [−S | −s | −f filter] [ −a] [ −p precision]
Apply dithering to the audio. Dithering deliberately adds a small amount of noise to the signal in
 
order to mask audible quantization effects that can occur if the output sample size is less than 24
 
bits.  With no options, this effect will add triangular (TPDF) white noise. Noise-shaping (only for
Apply dithering to the audio. Dithering deliberately adds a small amount of noise to the signal in order to mask audible quantization effects that can occur if the output sample size is less than 24 bits.  With no options, this effect will add triangular (TPDF) white noise. Noise-shaping (only for certain sample rates) can be selected with −s. W ith the −f option, it is possible to select a particu- lar noise-shaping filter from the following list: lipshitz, f-weighted, modified-e-weighted, improved-e-weighted,  gesemann,  shibata,  low-shibata,  high-shibata. Note  that  most  filter  types are  available  only  with  44100Hz  sample  rate. The  filter  types  are  distinguished  by  the  following
certain sample rates) can be selected with −s. W ith the −f option, it is possible to select a particu-
properties: audibility of noise, level of ( inaudible, but in some circumstances, otherwise problem- atic) shaped high frequency n oise, and processing speed.
lar noise-shaping filter from the following list: lipshitz, f-weighted, modified-e-weighted,
improved-e-weighted,  gesemann,  shibata,  low-shibata,  high-shibata. Note  that  most  filter  types
are  available  only  with  44100Hz  sample  rate. The  filter  types  are  distinguished  by  the  following
properties: audibility of noise, level of ( inaudible, but in some circumstances, otherwise problem-
atic) shaped high frequency n oise, and processing speed.
See http://sox.sourceforge.net/SoX/NoiseShaping for graphs of the different noise-shaping curves.
See http://sox.sourceforge.net/SoX/NoiseShaping for graphs of the different noise-shaping curves.
The −S option selects a slightly ‘sloped’ TPDF, b iased towards higher frequencies. It can be used
 
at any s ampling rate but below ≈22k, plain TPDF is probably better, a nd above ≈ 37k, noise-shap-
 
The −S option selects a slightly ‘sloped’ TPDF, b iased towards higher frequencies. It can be used at any s ampling rate but below ≈22k, plain TPDF is probably better, a nd above ≈ 37k, noise-shap-
ing (if available) is probably better.
ing (if available) is probably better.
The −a option enables a mode where dithering (and noise-shaping if applicable) are automatically
The −a option enables a mode where dithering (and noise-shaping if applicable) are automatically enabled  only  when  needed. The  most  likely  use  for  this  is  when  applying  fade  in  or  out  to  an already  dithered  file,  so  that  the  redithering  applies  only  to  the  faded  portions. However,  auto
enabled  only  when  needed. The  most  likely  use  for  this  is  when  applying  fade  in  or  out  to  an
dithering  is  not  fool-proof,  so  the  fades  should  be  carefully  checked  for  any n oise  modulation;  if this occurs, then either re-dither the whole file, or use trim, fade, a nd concatencate.
already  dithered  file,  so  that  the  redithering  applies  only  to  the  faded  portions. However,  auto
dithering  is  not  fool-proof,  so  the  fades  should  be  carefully  checked  for  any n oise  modulation;  if
this occurs, then either re-dither the whole file, or use trim, fade, a nd concatencate.
The −p option allows overriding the target precision.
The −p option allows overriding the target precision.
If the SoX global option −R option is not given, then the pseudo-random number generator used to
If the SoX global option −R option is not given, then the pseudo-random number generator used to
generate the white noise will be ‘reseeded’, i.e. the generated noise will be different between invo-
generate the white noise will be ‘reseeded’, i.e. the generated noise will be different between invo- cations.
cations.
This effect should not be followed by any o ther effect that affects the audio.
This effect should not be followed by any o ther effect that affects the audio.
See also the ‘Dithering’ section above.
See also the ‘Dithering’ section above.
downsample [factor(2)]
downsample [factor(2)]
Downsample the signal by an integer factor: Only the first out of each factor samples is retained,
 
the others are discarded.
 
No decimation filter is applied. If the input is not a properly bandlimited baseband signal, aliasing
Downsample the signal by an integer factor: Only the first out of each factor samples is retained, the others are discarded.
will occur. T his may be desirable, e.g., for frequency t ranslation.
No decimation filter is applied. If the input is not a properly bandlimited baseband signal, aliasing will occur. T his may be desirable, e.g., for frequency t ranslation.
For a g eneral resampling effect with anti-aliasing, see rate. S ee also upsample.
For a g eneral resampling effect with anti-aliasing, see rate. S ee also upsample.
earwax
earwax
Makes audio easier to listen to on headphones. Adds ‘cues’ to 44.1kHz stereo (i.e. audio CD for-
Makes audio easier to listen to on headphones. Adds ‘cues’ to 44.1kHz stereo (i.e. audio CD for-
mat)  audio  so  that  when  listened  to  on  headphones  the  stereo  image  is  moved f rom  inside  your
mat)  audio  so  that  when  listened  to  on  headphones  the  stereo  image  is  moved f rom  inside  your head (standard for headphones) to outside and in front of the listener (standard for speakers).
head (standard for headphones) to outside and in front of the listener (standard for speakers).
echo gain-in gain-out <delay decay>
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 buildings) when talking or shouting; digital echo effects emulate this behav-
 
iour and are often used to help fill out the sound of a single instrument or vocal.  The time differ-
Add echoing to the audio. Echoes are reflected sound and can occur naturally amongst mountains (and sometimes large buildings) when talking or shouting; digital echo effects emulate this behav- iour and are often used to help fill out the sound of a single instrument or vocal.  The time differ- ence  between  the  original  signal  and  the  reflection  is  the  ‘delay’  (time),  and  the  loudness  of  the reflected signal is the ‘decay’. Multiple echoes can have different delays and decays.
ence  between  the  original  signal  and  the  reflection  is  the  ‘delay’  (time),  and  the  loudness  of  the
reflected signal is the ‘decay’. Multiple echoes can have different delays and decays.
Each given delay decay pair gives t he delay in milliseconds and the decay (relative to g ain-in) of
Each given delay decay pair gives t he delay in milliseconds and the decay (relative to g ain-in) of
that echo. Gain-out is the volume of the output. For e xample: This will make it s ound as if there
that echo. Gain-out is the volume of the output. For e xample: This will make it s ound as if there are twice as many i nstruments as are actually playing:
are twice as many i nstruments as are actually playing:
play lead.aiff echo 0.8 0.88 60 0.4
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:
If the delay is very short, then it sound like a ( metallic) robot playing music:
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play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25
play lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25
echos gain-in gain-out <delay decay>
echos gain-in gain-out <delay decay>
Add a sequence of echoes to the audio. Each delay decay pair gives t he delay in milliseconds and
Add a sequence of echoes to the audio. Each delay decay pair gives t he delay in milliseconds and the decay (relative to g ain-in) of that echo. Gain-out is the volume of the output.
the decay (relative to g ain-in) of that echo. Gain-out is the volume of the output.
 
Like t he echo effect, echos stand for ‘ECHO in Sequel’, that is the first echos takes the input, the
Like t he 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 e chos; a single echos has the same effect as a single echo.
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 e chos; a single echos has the same effect as a single echo.
The sample will be bounced twice in symmetric echos:
The sample will be bounced twice in symmetric echos:
play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
play lead.aiff echos 0.8 0.7 700 0.25 700 0.3
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play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
play lead.aiff echos 0.8 0.7 40 0.25 63 0.3
equalizer frequency[k] width[q | o | h | k] gain
equalizer frequency[k] width[q | o | h | k] gain
Apply a two-pole peaking equalisation (EQ) filter. W ith this filter, t he signal-level at a nd around a
 
selected frequency c an be increased or decreased, whilst (unlike b and-pass and band-reject filters)
 
Apply a two-pole peaking equalisation (EQ) filter. W ith this filter, t he signal-level at a nd around a selected frequency c an be increased or decreased, whilst (unlike b and-pass and band-reject filters)
that at all other frequencies is unchanged.
that at all other frequencies is unchanged.
frequency gives t he  filter’s c entral  frequency i n H z, width, t he  band-width,  and gain the  required
frequency gives t he  filter’s c entral  frequency i n H z, width, t he  band-width,  and gain the  required
Line 1,099: Line 1,050:
In order to produce complex e qualisation curves, this effect can be given s ev eral times, each with a
In order to produce complex e qualisation curves, this effect can be given s ev eral times, each with a
different central frequency.
different central frequency.
The filter is described in detail in [1].
The filter is described in detail in [1]. This effect supports the −−plot global option. See also bass and treble for shelving equalisation effects.
This effect supports the −−plot global option.
See also bass and treble for shelving equalisation effects.
fade [type] fade-in-length [stop-position(=) [fade-out-length]]
fade [type] fade-in-length [stop-position(=) [fade-out-length]]
Apply a fade effect to the beginning, end, or both of the audio.
Apply a fade effect to the beginning, end, or both of the audio.
An  optional type can  be  specified  to  select  the  shape  of  the  fade  curve: q for  quarter  of  a  sine
An  optional type can  be  specified  to  select  the  shape  of  the  fade  curve: q for  quarter  of  a  sine wave, h for half a sine wav e, t for linear (‘triangular’) slope, l for logarithmic, and p for inverted parabola.  The default is logarithmic.
wave, h for half a sine wav e, t for linear (‘triangular’) slope, l for logarithmic, and p for inverted
 
parabola.  The default is logarithmic.
A f ade-in  starts  from  the  first  sample  and  ramps  the  signal  level f rom  0  to  full  volume  over t he time given a s fade-in-length. S pecify 0 if no fade-in is wanted. For f ade-outs, the audio will be truncated at stop-position and the signal level w ill be ramped from full  volume  down  to  0  over an i nterval  of fade-out-length before  the stop-position. I f fade-out-
A f ade-in  starts  from  the  first  sample  and  ramps  the  signal  level f rom  0  to  full  volume  over t he
length is not specified, it defaults to the same value as fade-in-length. N o f ade-out is performed if stop-position is not specified. If the audio length can be determined from the input file header and any p revious  effects,  then −0 (or, f or  historical  reasons, 0) m ay  be  specified  for stop-position to
time given a s fade-in-length. S pecify 0 if no fade-in is wanted.
For f ade-outs, the audio will be truncated at stop-position and the signal level w ill be ramped from
full  volume  down  to  0  over an i nterval  of fade-out-length before  the stop-position. I f fade-out-
length is not specified, it defaults to the same value as fade-in-length. N o f ade-out is performed if
stop-position is not specified. If the audio length can be determined from the input file header and
any p revious  effects,  then −0 (or, f or  historical  reasons, 0) m ay  be  specified  for stop-position to
indicate the usual case of a fade-out that ends at the end of the input audio stream.
indicate the usual case of a fade-out that ends at the end of the input audio stream.
Any t ime specification may be used for fade-in-length and fade-out-length.
 
Any time specification may be used for fade-in-length and fade-out-length.
See also the splice effect.
See also the splice effect.
fir [coefs-file | coefs]
fir [coefs-file | coefs]
Use SoX’s F FT convolution engine with given F IR filter coefficients.  If a s ingle argument is given
 
then  this  is  treated  as  the  name  of  a  file  containing  the  filter  coefficients  (white-space  separated;
 
may contain ‘#’ comments). If the given fi lename is ‘−’, or if no argument is given, then the coef-
Use SoX’s F FT convolution engine with given F IR filter coefficients.  If a s ingle argument is given then  this  is  treated  as  the  name  of  a  file  containing  the  filter  coefficients  (white-space  separated;
ficients are read from the ‘standard input’ (stdin); otherwise, coefficients may be given on t he com-
may contain ‘#’ comments). If the given fi lename is ‘−’, or if no argument is given, then the coef-ficients are read from the ‘standard input’ (stdin); otherwise, coefficients may be given on the com-mand line. Examples:
mand line. Examples:
sox infile outfile fir 0.0195 −0.082 0.234 0.891 −0.145 0.043
sox infile outfile fir 0.0195 −0.082 0.234 0.891 −0.145 0.043
sox infile outfile fir coefs.txt
sox infile outfile fir coefs.txt
Line 1,150: Line 1,093:
phase 0 − 1 00 25 Swept wav e percentage phase-shift
phase 0 − 1 00 25 Swept wav e percentage phase-shift
for multi-channel (e.g. stereo)
for multi-channel (e.g. stereo)
flange; 0 = 100 = same phase on
flange; 0 = 100 = same phase on each channel.
each channel.
interp lin Digital delay-line interpolation:
interp lin Digital delay-line interpolation:
linear | quadratic.
linear | quadratic.
gain [−e | −B | −b | −r] [ −n] [ −l | −h] [ gain-dB]
 
gain [−e | −B | −b | −r] [ −n] [ −l | −h] [ gain-dB]
 
Apply amplification or attenuation to the audio signal, or, in s ome cases, to some of its channels.
Apply amplification or attenuation to the audio signal, or, in s ome cases, to some of its channels.
Note that use of any o f −e, −B, −b, −r, o r −n requires temporary file space to store the audio to be
Note that use of any o f −e, −B, −b, −r, o r −n requires temporary file space to store the audio to be
processed, so may be unsuitable for use with ‘streamed’ audio.
processed, so may be unsuitable for use with ‘streamed’ audio.
Without other options, gain-dB is used to adjust the signal power level b y t he given n umber of dB:
Without other options, gain-dB is used to adjust the signal power level b y t he given n umber of dB:
positive  amplifies  (beware  of  Clipping),  negative  attenuates. With  other  options,  the gain-dB
 
amplification or attenuation is (logically) applied after the processing due to those options.
positive  amplifies  (beware  of  Clipping),  negative  attenuates. With  other  options,  the gain-dB amplification or attenuation is (logically) applied after the processing due to those options.
 
Given t he −e option,  the  levels  of  the  audio  channels  of  a  multi-channel  file  are  ‘equalised’,  i.e.
Given t he −e option,  the  levels  of  the  audio  channels  of  a  multi-channel  file  are  ‘equalised’,  i.e.
gain  is  applied  to  all  channels  other  than  that  with  the  highest  peak  level,  such  that  all  channels
 
attain the same peak level ( but, without also giving −n, t he audio is not ‘normalised’).
gain  is  applied  to  all  channels  other  than  that  with  the  highest  peak  level,  such  that  all  channels attain the same peak level ( but, without also giving −n, t he audio is not ‘normalised’).
The −B (balance) option is similar to −e, b ut with −B, the RMS level is u sed instead of the peak
The −B (balance) option is similar to −e, b ut with −B, the RMS level is u sed instead of the peak
level. −B might be used to correct stereo imbalance caused by an imperfect record turntable car-
level. −B might be used to correct stereo imbalance caused by an imperfect record turntable car- tridge. Note that unlike −e, −B might cause some clipping.
tridge. Note that unlike −e, −B might cause some clipping.
−b is similar to −B but h as clipping protection, i.e. if necessary to prevent clipping whilst balanc- ing, attenuation is applied to all channels. Note, however, that in conjunction with −n, −B and −b
−b is similar to −B but h as clipping protection, i.e. if necessary to prevent clipping whilst balanc-
ing, attenuation is applied to all channels. Note, however, that in conjunction with −n, −B and −b
are synonymous.
are synonymous.
The −r option  is  used  in  conjunction  with  a  prior  invocation  of gain with  the −h option—see
The −r option  is  used  in  conjunction  with  a  prior  invocation  of gain with  the −h option—see
below f or details.
below f or details.
Line 1,180: Line 1,124:
The −l option invokes a s imple limiter, e .g.
The −l option invokes a s imple limiter, e .g.
sox infile outfile gain −l 6
sox infile outfile gain −l 6
will apply 6dB of gain but never c lip.  Note that limiting more than a few d Bs more than occasion-
 
ally (in a piece of audio) is not recommended as it can cause audible distortion. See the compand
will apply 6dB of gain but never c lip.  Note that limiting more than a few d Bs more than occasion- ally (in a piece of audio) is not recommended as it can cause audible distortion. See the compand effect for a more capable limiter.
effect for a more capable limiter.
The −h option is used to apply gain to provide head-room for subsequent processing. For e xam- ple, withsox infile outfile gain −h bass +6
The −h option is used to apply gain to provide head-room for subsequent processing. For e xam-
6dB  of  attenuation  will  be  applied  prior  to  the  bass  boosting  effect  thus  ensuring  that  it  will  not clip.  Of course, with bass, it is obvious how much headroom will be needed, but with other effects
ple, with
sox infile outfile gain −h bass +6
6dB  of  attenuation  will  be  applied  prior  to  the  bass  boosting  effect  thus  ensuring  that  it  will  not
clip.  Of course, with bass, it is obvious how m uch headroom will be needed, but with other effects


   
   
(e.g. rate, dither)  it  is  not  always  as  clear. A nother  advantage  of  using gain  −h rather  than  an
(e.g. rate, dither)  it  is  not  always  as  clear. A nother  advantage  of  using gain  −h rather  than  an explicit attenuation, is that if the headroom is not used by subsequent effects, it can be reclaimed with gain −r, f or example:
explicit attenuation, is that if the headroom is not used by subsequent effects, it can be reclaimed
 
with gain −r, f or example:
sox infile outfile gain −h bass +6 rate 44100 gain −r
sox infile outfile gain −h bass +6 rate 44100 gain −r
 
 
The above effects chain guarantees never to c lip nor amplify; it attenuates if necessary to prevent
The above effects chain guarantees never to c lip nor amplify; it attenuates if necessary to prevent
clipping, but by only as much as is needed to do so.
clipping, but by only as much as is needed to do so.
Output  formatting  (dithering  and  bit-depth  reduction)  also  requires  headroom  (which  cannot  be
 
‘reclaimed’), e.g.
Output  formatting  (dithering  and  bit-depth  reduction)  also  requires  headroom  (which  cannot  be ‘reclaimed’), e.g.
sox infile outfile gain −h bass +6 rate 44100 gain −rh dither
sox infile outfile gain −h bass +6 rate 44100 gain −rh dither
Here, the second gain invocation, reclaims as much of the headroom as it can from the preceding
Here, the second gain invocation, reclaims as much of the headroom as it can from the preceding effects,  but  retains  as  much  headroom  as  is  needed  for  subsequent  processing. The  SoX  global option −G can be given to a utomatically invoke gain −h and gain −r.
effects,  but  retains  as  much  headroom  as  is  needed  for  subsequent  processing. The  SoX  global
 
option −G can be given to a utomatically invoke gain −h and gain −r.
See also the norm and vol effects.
See also the norm and vol effects.
highpass | lowpass [−1|−2] frequency[k] [ width[q | o | h | k]]
highpass | lowpass [−1|−2] frequency[k] [ width[q | o | h | k]]
Apply a high-pass or low-pass filter with 3dB point frequency. T he filter can be either single-pole
Apply a high-pass or low-pass filter with 3dB point frequency. T he 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 B utterworth  response. The  filters  roll  off at 6 dB  per  pole  per
(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 B utterworth  response. The  filters  roll  off at 6 dB  per  pole  per
octave (20dB per pole per decade). The double-pole filters are described in detail in [1].
octave (20dB per pole per decade). The double-pole filters are described in detail in [1].
These effects support the −−plot global option.
These effects support the −−plot global option.
See also sinc for filters with a steeper roll-off.
See also sinc for filters with a steeper roll-off.
hilbert [−n taps]
hilbert [−n taps]
Apply an odd-tap Hilbert transform filter, p hase-shifting the signal by 90 degrees.
Apply an odd-tap Hilbert transform filter, p hase-shifting the signal by 90 degrees.
This  is  used  in  many m atrix  coding  schemes  and  for  analytic  signal  generation. The  process  is
This  is  used  in  many m atrix  coding  schemes  and  for  analytic  signal  generation. The  process  is often written as a multiplication by i (or j), the imaginary unit.
often written as a multiplication by i (or j), the imaginary unit.
 
An odd-tap Hilbert transform filter has a bandpass characteristic, attenuating the lowest and high-
An odd-tap Hilbert transform filter has a bandpass characteristic, attenuating the lowest and high- est frequencies. Its bandwidth can be controlled by the number of filter taps, which can be speci- fied with −n. B y d efault, the number of taps is chosen for a cutoff f requency of a bout 75 Hz.
est frequencies. Its bandwidth can be controlled by the number of filter taps, which can be speci-
fied with −n. B y d efault, the number of taps is chosen for a cutoff f requency of a bout 75 Hz.
This effect supports the −−plot global option.
This effect supports the −−plot global option.
ladspa [-l | -r] module [plugin] [ argument ...]
ladspa [-l | -r] module [plugin] [ argument ...]
Apply  a  LADSPA [ 5]  (Linux  Audio  Developer’s S imple  Plugin  API)  plugin. Despite  the  name,
Apply  a  LADSPA [ 5]  (Linux  Audio  Developer’s S imple  Plugin  API)  plugin. Despite  the  name,
LADSPA is n ot Linux-specific, and a wide range of effects is available as LADSPA p lugins, such
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 p lugin  collection  [7].  The  first  argu-
as  cmt  [6]  (the  Computer  Music  Toolkit)  and  Steve  Harris’s p lugin  collection  [7].  The  first  argument is the plugin module, the second the name of the plugin (a module can contain more than one
ment is the plugin module, the second the name of the plugin (a module can contain more than one
plugin),  and  any o ther  arguments  are  for  the  control  ports  of  the  plugin.  Missing  arguments  are supplied by default values if possible.
plugin),  and  any o ther  arguments  are  for  the  control  ports  of  the  plugin.  Missing  arguments  are
Normally, t he number of input ports of the plugin must match the number of input channels, and the  number  of  output  ports  determines  the  output  channel  count. However,  the −r (replicate) option allows cloning a mono plugin to handle multi-channel input.
supplied by default values if possible.
Some plugins introduce latency w hich SoX may optionally compensate for. T he −l (latency c om- pensation)  option  automatically  compensates  for  latency as r eported  by  the  plugin  via  an  output control port named "latency".
Normally, t he number of input ports of the plugin must match the number of input channels, and
the  number  of  output  ports  determines  the  output  channel  count. However,  the −r (replicate)
option allows cloning a mono plugin to handle multi-channel input.
Some plugins introduce latency w hich SoX may optionally compensate for. T he −l (latency c om-
pensation)  option  automatically  compensates  for  latency as r eported  by  the  plugin  via  an  output
control port named "latency".
If found, the environment variable LADSPA_PAT H w ill be used as search path for plugins.
If found, the environment variable LADSPA_PAT H w ill be used as search path for plugins.
loudness [gain [reference]]
loudness [gain [reference]]
Loudness  control—similar  to  the gain effect,  but  provides  equalisation  for  the  human  auditory
Loudness  control—similar  to  the gain effect,  but  provides  equalisation  for  the  human  auditory
system. See http://en.wikipedia.org/wiki/Loudness  for  a  detailed  description  of  loudness. The
system. See http://en.wikipedia.org/wiki/Loudness  for  a  detailed  description  of  loudness. The gain is adjusted by the given gain parameter (usually negative) a nd the signal equalised according to ISO 226 w.r.t. a reference level of 6 5dB, though an alternative reference level m ay be given i f the  original  audio  has  been  equalised  for  some  other  optimal  level. A default  gain  of  −10dB  is
gain is adjusted by the given gain parameter (usually negative) a nd the signal equalised according
to ISO 226 w.r.t. a reference level of 6 5dB, though an alternative reference level m ay be given i f
the  original  audio  has  been  equalised  for  some  other  optimal  level. A default  gain  of  −10dB  is
sox December 31, 2
 
SoX(1) Sound eXchange SoX(1)
used if a gain value is not given.
used if a gain value is not given.
See also the gain effect.
See also the gain effect.
Line 1,249: Line 1,177:
[gain [initial-volume-dB [delay]]]" {crossover-freq[k] " attack1,..."}
[gain [initial-volume-dB [delay]]]" {crossover-freq[k] " attack1,..."}
The  multi-band  compander  is  similar  to  the  single-band  compander  but  the  audio  is  first  divided
The  multi-band  compander  is  similar  to  the  single-band  compander  but  the  audio  is  first  divided
into  bands  using  Linkwitz-Riley c ross-over fi lters  and  a  separately  specifiable  compander  run  on
into  bands  using  Linkwitz-Riley c ross-over fi lters  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 f requency f or that band is given b y crossover-
each band. See the compand effect for the definition of its parameters. Compand parameters are
specified between double quotes and the crossover f requency f or that band is given b y crossover-
freq; t hese can be repeated to create multiple bands.
freq; t hese can be repeated to create multiple bands.
For e xample,  the  following  (one  long)  command  shows  how m ulti-band  companding  is  typically
For e xample,  the  following  (one  long)  command  shows  how m ulti-band  companding  is  typically
Line 1,263: Line 1,189:
gain 15 highpass 22 highpass 22 sinc −n 255 −b 16 −17500 \
gain 15 highpass 22 highpass 22 sinc −n 255 −b 16 −17500 \
gain 9 lowpass −1 17801
gain 9 lowpass −1 17801
The audio file is played with a simulated FM radio sound (or broadcast signal condition if the low-
The audio file is played with a simulated FM radio sound (or broadcast signal condition if the low- pass filter at the end is skipped). Note that the pipeline is set up with US-style 75us pre-emphasis.
pass filter at the end is skipped). Note that the pipeline is set up with US-style 75us pre-emphasis.
See also compand for a single-band companding effect.
See also compand for a single-band companding effect.
noiseprof [profile-file]
noiseprof [profile-file]
Calculate  a  profile  of  the  audio  for  use  in  noise  reduction. See  the  description  of  the noisered
Calculate  a  profile  of  the  audio  for  use  in  noise  reduction. See  the  description  of  the noisered effect for details.
effect for details.
noisered [profile-file [amount]]
noisered [profile-file [amount]]
Reduce  noise  in  the  audio  signal  by  profiling  and  filtering. This  effect  is  moderately  effective  at
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  u se it, first run SoX with the noise- prof 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 s tdout if no profile-file or if ‘−’ is given.  E.g.
removing consistent background noise such as hiss or hum. To  u se it, first run SoX with the noise-
prof 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 s tdout if no profile-file or if ‘−’ is given.  E.g.
sox speech.wav −n trim 0 1.5 noiseprof speech.noise-profile
sox speech.wav −n trim 0 1.5 noiseprof speech.noise-profile
To  a ctually  remove  the  noise,  run  SoX  again,  this  time  with  the noisered effect; noisered will
To  a ctually  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, o r from stdin if no profile-file or if ‘−’ is given.  E.g. sox speech.wav cleaned.wav noisered speech.noise-profile 0.3
reduce noise according to a noise profile (which was generated by noiseprof), from profile-file, o r
How much noise  should  be  removed is s pecified  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 remov- ing 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;
from stdin if no profile-file or if ‘−’ is given.  E.g.
use headphones to check that you are happy w ith the results, paying particular attention to quieter sections of the audio.
sox speech.wav cleaned.wav noisered speech.noise-profile 0.3
On most systems, the two s tages—profiling and reduction—can be combined using a pipe, e.g.sox noisy.wav −n trim 0 1 noiseprof | play noisy.wav noisered
How m uch noise  should  be  removed is s pecified  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 remov-
ing 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 w ith the results, paying particular attention to quieter
sections of the audio.
On most systems, the two s tages—profiling and reduction—can be combined using a pipe, e.g.
sox noisy.wav −n trim 0 1 noiseprof | play noisy.wav noisered
norm [dB-level]
norm [dB-level]
Normalise the audio. norm is just an alias for gain −n; s ee the gain effect for detail
Normalise the audio. norm is just an alias for gain −n; s ee the gain effect for detail
SoX(1) Sound eXchange SoX(1)
SoX(1) Sound eXchange SoX(1)
oops Out  Of  Phase  Stereo  effect. Mixes  stereo  to  twin-mono  where  each  mono  channel  contains  the
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. It is equiva-
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. It is equiva-
lent to remix 1,2i 1,2i.
lent to remix 1,2i 1,2i.
overdrive [gain(20) [colour(20)]]
overdrive [gain(20) [colour(20)]]
Non linear distortion. The colour parameter controls the amount of even h armonic content in the
Non linear distortion. The colour parameter controls the amount of even h armonic content in the over-driven output.
over-driven o utput.
pad { length[@position(=)] }  
pad { length[@position(=)] }  
Pad t he  audio  with  silence,  at  the  beginning,  the  end,  or  any s pecified points  through  the  audio.
Pad t he  audio  with  silence,  at  the  beginning,  the  end,  or  any specified points  through  the  audio.
length is  the  amount  of  silence  to  insert  and position the  position  in  the  input  audio  stream  at
length is  the  amount  of  silence  to  insert  and position the  position  in  the  input  audio  stream  at which to insert it. Any n umber of lengths and positions may be specified, provided that a specified position is not less that the previous one, and any t ime specification may be used for them. posi-
which to insert it. Any n umber of lengths and positions may be specified, provided that a specified
tion is  optional  for  the  first  and  last  lengths  specified  and  if  omitted  correspond  to  the  beginning and the end of the audio respectively. F or example, pad 1.5 1 .5 adds 1.5 s econds of silence pad- ding at each end of the audio, whilst pad 4000s@3:00 inserts 4000 samples of silence 3 minutes
position is not less that the previous one, and any t ime specification may be used for them. posi-
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.
tion is  optional  for  the  first  and  last  lengths  specified  and  if  omitted  correspond  to  the  beginning
See  also delay for  an  effect  that  can  add  silence  at  the  beginning  of  the  audio  on  a  channel-by- channel basis.
and the end of the audio respectively. F or example, pad 1.5 1 .5 adds 1.5 s econds of silence pad-
ding 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.
See  also delay for  an  effect  that  can  add  silence  at  the  beginning  of  the  audio  on  a  channel-by-
channel basis.
phaser gain-in gain-out delay decay speed [−s | −t]
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.
Add a phasing effect to the audio. See [3] for a detailed description of phasing.
delay/decay/speed gives t he delay in milliseconds and the decay (relative to g ain-in) with a modu-
delay/decay/speed gives t he delay in milliseconds and the decay (relative to g ain-in) with a modu-lation speed in Hz. The modulation is either sinusoidal (−s) — preferable for multiple instruments,or  triangular  (−t) — gives s ingle  instruments  a  sharper  phasing  effect. The decay  should  be  less than 0.5 to a void feedback, and usually no less than 0.1.  Gain-out is the volume of the output.
lation speed in Hz. The modulation is either sinusoidal (−s) — preferable for multiple instruments,
or  triangular  (−t) — gives s ingle  instruments  a  sharper  phasing  effect. The decay  should  be  less
than 0.5 to a void feedback, and usually no less than 0.1.  Gain-out is the volume of the output.
For e xample:
For e xample:
play snare.flac phaser 0.8 0.74 3 0.4 0.5 −t
play snare.flac phaser 0.8 0.74 3 0.4 0.5 −t
Line 1,326: Line 1,226:
pitch [−q] shift [segment [search [overlap]]]
pitch [−q] shift [segment [search [overlap]]]
Change the audio pitch (but not tempo).
Change the audio pitch (but not tempo).
shift gives t he pitch shift as positive or n eg ative ‘cents’ (i.e. 100ths of a semitone). See the tempo
shift gives t he pitch shift as positive or n eg ative ‘cents’ (i.e. 100ths of a semitone). See the tempo effect for a description of the other parameters.
effect for a description of the other parameters.
See also the bend, speed, a nd tempo effects.
See also the bend, speed, a nd tempo effects.
rate [−q | −l | −m | −h | −v] [ override-options] RATE[k]
rate [−q | −l | −m | −h | −v] [ override-options] RATE[k]
Change the audio sampling rate (i.e. resample the audio) to any g iv en RATE (even n on-integer if
Change the audio sampling rate (i.e. resample the audio) to any g iv en RATE (even n on-integer if

Revision as of 14:19, 2 September 2021

SoX − Sound eXchange, the Swiss Army knife of audio manipulation[edit]

SYNOPSIS[edit]

  sox [global-options] [ format-options] infile1
      [[format-options] infile2] . .. [format-options] outfile
      [effect [effect-options]] ...
 play [global-options] [ format-options] infile1
      [[format-options] infile2] . .. [format-options]
      [effect [effect-options]] ...
  rec [global-options] [ format-options] outfile
      [effect [effect-options]] ...

DESCRIPTION[edit]

Introduction[edit]

SoX reads and writes audio files in most popular formats and can optionally apply effects to them. It can combine multiple input sources, synthesise audio, and, on many s ystems, act as a general purpose audio player or a multi-track audio recorder. It a lso has limited ability to split the input into multiple output files.

All SoX functionality is available using just the sox command. To s implify playing and recording audio, if SoX is invoked a s play, t he output file is automatically set to be the default sound device, and if invoked a s rec, t he default sound device is used as an input source. Additionally, t he soxi(1) command provides a convenient way to just query audio file header information.

The heart of SoX is a library called libSoX. Those interested in extending SoX or using it in other programs should refer to the libSoX manual page: libsox(3).

SoX is a command-line audio processing tool, particularly suited to making quick, simple edits and to batch processing. If you need an interactive, g raphical audio editor, u se audacity(1).


The overall SoX processing chain can be summarised as follows: 

Input(s) → Combiner → Effects → Output(s)

Note however, that on the SoX command line, the positions of the Output(s) and the Effects are swapped w.r.t. the logical flow j ust shown. Note also that whilst options pertaining to files are placed before their respective file name, the opposite is true for effects. To s how h ow this works in practice, here is a selection of examples of how S oX might be used. The simple 

sox recital.au recital.wav

translates an audio file in Sun AU f ormat to a Microsoft WAV fi le, whilst sox recital.au −b 16 recital.wav channels 1 rate 16k fade 3 norm performs the same format translation, but also applies four effects (down-mix to one channel, sample rate change, fade-in, nomalize), and stores the result at a bit-depth of 16. 

sox −r 16k −e signed −b 8 −c 1 voice-memo.raw voice-memo.wav

converts ‘raw’ (a.k.a. ‘headerless’) audio to a self-describing file format, 

sox slow.aiff fixed.aiff speed 1.027

adjusts audio speed, 

sox short.wav long.wav longer.wav

concatenates two a udio files, and 

sox −m music.mp3 voice.wav mixed.flac

mixes together two a udio files. 

play "The Moonbeams/Greatest/*.ogg" bass +3

plays a collection of audio files whilst applying a bass boosting effect, 

play −n −c1 synth sin %−12 sin %−9 sin %−5 sin %−2 fade h 0.1 1 0.1

plays a synthesised ‘A m inor seventh’ chord with a pipe-organ s ound, 

rec −c 2 radio.aiff trim 0 30:00

records half an hour of stereo audio, and 

play −q take1.aiff & rec −M take1.aiff take1−dub.aiff

(with POSIX shell and where supported by hardware) records a new t rack in a multi-track recording. Finally, 

rec −r 44100 −b 16 −e signed-integer −p \
silence 1 0.50 0.1% 1 10:00 0.1% | \
sox −p song.ogg silence 1 0.50 0.1% 1 2.0 0.1% : \
newfile : restart

records a stream of audio such as LP/cassette and splits in to multiple audio files at points with 2 seconds of silence. Also, it does not start recording until it detects audio is playing and stops after it sees 10 minutes of silence.

N.B. The above is just an overview of SoX’s c apabilities; detailed explanations of how to u se all SoX parameters, file formats, and effects can be found below in t his manual, in soxformat(7), and in soxi(1).

File Format Types[edit]

SoX can work with ‘self-describing’ and ‘raw’ audio files. ‘self-describing’ formats (e.g. WAV , FLAC, MP3) have a header that completely describes the signal and encoding attributes of the audio data that fol- lows. ‘raw’ or ‘headerless’ formats do not contain this information, so the audio characteristics of these must be described on the SoX command line or inferred from those of the input file. The following four characteristics are used to describe the format of audio data such that it can be processed with SoX:

sample rate[edit]

The sample rate in samples per second (‘Hertz’ or ‘Hz’). Digital telephony traditionally uses a sample rate of 8000 Hz (8 kHz), though these days, 16 and even 3 2 k Hz are becoming more common. Audio Compact Discs use 44100 Hz (44.1 k Hz). Digital Audio Tape and many computer systems use 48 kHz. Professional audio systems often use 96 kHz.

sample size[edit]

The number of bits used to store each sample. To day, 1 6-bit is commonly used. 8-bit was popular in the early days of computer audio. 24-bit is used in the professional audio arena. Other sizes are also used.

data encoding[edit]

The way in which each audio sample is represented (or ‘encoded’). Some encodings have variants with different byte-orderings or bit-orderings. Some compress the audio data so that the stored audio data takes up less space (i.e. disk space or transmission bandwidth) than the other format parameters and the number of samples would imply. Commonly-used encoding types include floating-point, μ-law, ADPCM, signed-integer PCM, MP3, and FLAC.

channels[edit]

The number of audio channels contained in the file. One (‘mono’) and two ( ‘stereo’) are widely used. ‘Surround sound’ audio typically contains six or more channels.

The term ‘bit-rate’ is a measure of the amount of storage occupied by an encoded audio signal over a u nit of time. It can depend on all of the above and is typically denoted as a number of kilo-bits per second (kbps). An A-law telephony signal has a bit-rate of 64 kbps. MP3-encoded stereo music typically has a bit-rate of 128−196 kbps. FLAC-encoded stereo music typically has a bit-rate of 550−760 kbps.

Most self-describing formats also allow textual ‘comments’ to be embedded in the file that can be used to describe the audio in some way, e .g. for music, the title, the author, etc.

One important use of audio file comments is to convey ‘Replay Gain’ information. SoX supports applying Replay Gain information (for certain input file formats only; currently, at l east FLAC and Ogg Vorbis), but not generating it. Note that by default, SoX copies input file comments to output files that support comments, so output files may contain Replay Gain information if some was present in the input file. In this case, if anything other than a simple format conversion was performed then the output file Replay Gain information is likely to be incorrect and so should be recalculated using a tool that supports this (not SoX

The soxi(1) command can be used to display information from audio file headers.

Determining & Setting The File Format[edit]

There are several mechanisms available for SoX to use to determine or set the format characteristics of an audio file. Depending on the circumstances, individual characteristics may be determined or set using dif- ferent mechanisms.

To d etermine the format of an input file, SoX will use, in order of precedence and as given or a vailable: 1. Command-line format options. 2. The contents of the file header. 3. The filename extension. To s et the output file format, SoX will use, in order of precedence and as given or a vailable: 1. Command-line format options. 2. The filename extension. 3. The input file format characteristics, or the closest that is supported by the output file type.

For a ll files, SoX will exit with an error if the file type cannot be determined. Command-line format options may need to be added or changed to resolve t he problem.

Playing & Recording Audio[edit]

The play and rec commands are provided so that basic playing and recording is as simple as 

play existing-file.wav

and 

rec new-file.wav

These two c ommands are functionally equivalent to 

sox existing-file.wav −d

and 

sox −d new-file.wav

Of course, further options and effects (as described below) can be added to the commands in either form.

  • * *

Some systems provide more than one type of (SoX-compatible) audio driver, e.g. ALSA & OSS, or SUNAU & AO. S ystems can also have more than one audio device (a.k.a. ‘sound card’). If more than one audio driver h as been built-in to SoX, and the default selected by SoX when recording or playing is not the one that is wanted, then the AUDIODRIVER environment variable can be used to override the default. For e xample (on many s ystems): 

set AUDIODRIVER=oss
play ...

The AUDIODEV environment variable can be used to override the default audio device, e.g. 

set AUDIODEV=/dev/dsp2
play ...
sox ... −t oss

or 

set AUDIODEV=hw:soundwave,1,2
play ...
sox ... −t alsa

Note that the way of setting environment variables varies from system to system—for some specific exam- ples, see ‘SOX_OPTS’ below.

When playing a file with a sample rate that is not supported by the audio output device, SoX will automati- cally invoke the rate effect to perform the necessary sample rate conversion. For compatibility with old hardware, the default rate quality level i s s et to ‘low’. This can be changed by explicitly specifying the rate effect with a different quality level, e.g. 

play ... rate −m

or by using the −−play−rate−arg option (see below).

  • * *

On some systems, SoX allows audio playback volume to be adjusted whilst using play. W here supported, this is achieved by t apping the ‘v’ & ‘V’ keys d uring playback.

To h elp with setting a suitable recording level, SoX includes a peak-level m eter which can be invoked (before making the actual recording) as follows: 

rec −n

The recording level s hould be adjusted (using the system-provided mixer program, not SoX) so that the meter is at most occasionally full scale, and never ‘ in the red’ (an exclamation mark is shown). See also −S below.

Accuracy[edit]

Many fi le formats that compress audio discard some of the audio signal information whilst doing so. Con- verting to such a format and then converting back again will not produce an exact copy of t he original audio. This is the case for many f ormats used in telephony ( e.g. A-law, GSM) where low s ignal bandwidth is more important than high audio fidelity, a nd for many f ormats used in portable music players (e.g. MP3, Vo rbis) where adequate fidelity can be retained even w ith the large compression ratios that are needed to make p ortable players practical.

Formats that discard audio signal information are called ‘lossy’. Formats that do not are called ‘lossless’. The term ‘quality’ is used as a measure of how c losely the original audio signal can be reproduced when using a lossy format.

Audio file conversion with SoX is lossless when it can be, i.e. when not using lossy compression, when not reducing the sampling rate or number of channels, and when the number of bits used in the destination for- mat is not less than in the source format. E.g. converting from an 8-bit PCM format to a 16-bit PCM for- mat is lossless but converting from an 8-bit PCM format to (8-bit) A-law i sn’t. N.B. SoX converts all audio files to an internal uncompressed format before performing any a udio process- ing. This means that manipulating a file that is stored in a lossy format can cause further losses in audio fidelity. E .g. with 

sox long.mp3 short.mp3 trim 10

SoX first decompresses the input MP3 file, then applies the trim effect, and finally creates the output MP3 file by re-compressing the audio—with a possible reduction in fidelity above that which occurred when the input file was created. Hence, if what is ultimately desired is lossily compressed audio, it is highly recom- mended to perform all audio processing using lossless file formats and then convert to the lossy format only at the final stage.

N.B. Applying multiple effects with a single SoX invocation will, in general, produce more accurate results than those produced using multiple SoX invocations.

Dithering[edit]

Dithering is a technique used to maximise the dynamic range of audio stored at a particular bit-depth. Any distortion introduced by quantisation is decorrelated by adding a small amount of white noise to the signal. In most cases, SoX can determine whether the selected processing requires dither and will add it during output formatting if appropriate.

Specifically, by d efault, SoX automatically adds TPDF dither when the output bit-depth is less than 24 and any of t he following are true:

• bit-depth reduction has been specified explicitly using a command-line option • the output file format supports only bit-depths lower than that of the input file format • an effect has increased effective bit-depth within the internal processing chain

For e xample, adjusting volume with vol 0 .25 requires two a dditional bits in which to losslessly store its results (since 0.25 decimal equals 0.01 binary). So if the input file bit-depth is 16, then SoX’s i nternal rep- resentation will utilise 18 bits after processing this volume change. In order to store the output at the same depth as the input, dithering is used to remove the additional bits. 

Use  the −V option  to  see  what  processing  SoX  has  automatically  added.  The −D option  may  be  given t o

override automatic dithering. To i nv oke dithering manually (e.g. to select a noise-shaping curve), see the dither effect.

Clipping[edit]

Clipping is distortion that occurs when an audio signal level ( or ‘volume’) exceeds the range of the chosen representation. In most cases, clipping is undesirable and so should be corrected by adjusting the level prior to the point (in the processing chain) at which it occurs.

In SoX, clipping could occur, as y ou might expect, when using the vol or gain effects to increase the audio volume. Clipping could also occur with many o ther effects, when converting one format to another, a nd ev en w hen simply playing the audio.

Playing an audio file often involves resampling, and processing by analogue components can introduce a small DC offset and/or amplification, all of which can produce distortion if the audio signal level w as ini- tially too close to the clipping point.

For t hese reasons, it is usual to make s ure that an audio file’s s ignal level h as some ‘headroom’, i.e. it does not exceed a particular level b elow t he maximum possible level f or the given r epresentation. Some stan- dards bodies recommend as much as 9dB headroom, but in most cases, 3dB (≈ 70% linear) is enough. Note that this wisdom seems to have been lost in modern music production; in fact, many C Ds, MP3s, etc. are now m astered at levels above 0dBFS i.e. the audio is clipped as delivered. SoX’s stat and stats effects can assist in determining the signal level in an a udio file. The gain or vol effect can be used to prevent clipping, e.g.

sox dull.wav bright.wav gain −6 treble +6

guarantees that the treble boost will not clip.

If clipping occurs at any p oint during processing, SoX will display a warning message to that effect. See also −G and the gain and norm effects.

Input File Combining[edit]

SoX’s input combiner can be configured (see OPTIONS below) to combine multiple files using any of the following methods: ‘concatenate’, ‘sequence’, ‘mix’, ‘mix-power’, ‘merge’, or ‘multiply’. The default method is ‘sequence’ for play, a nd ‘concatenate’ for rec and sox. For a ll methods other than ‘sequence’, multiple input files must have the same sampling rate. If necessary, separate SoX invocations can be used to make s ampling rate adjustments prior to combining. If the ‘concatenate’ combining method is selected (usually, t his will be by default) then the input files must also have the same number of channels. The audio from each input will be concatenated in the order given to form the output file.

The ‘sequence’ combining method is selected automatically for play. I t i s s imilar to ‘concatenate’ in that the audio from each input file is sent serially to the output file. However, here the output file may be closed and reopened at the corresponding transition between input files. This may be just what is needed when sending different types of audio to an output device, but is not generally useful when the output is a normal file.

If either the ‘mix’ or ‘mix-power’ combining method is selected then two or m ore input files must be given and will be mixed together to form the output file. The number of channels in each input file need not be the same, but SoX will issue a warning if they a re not and some channels in the output file will not contain audio from every input file. A m ixed audio file cannot be un-mixed without reference to the original input files.

If the ‘merge’ combining method is selected then two or m ore input files must be given a nd will be merged together to form the output file. The number of channels in each input file need not be the same. A merged audio file comprises all of the channels from all of the input files. Un-merging is possible using multiple invocations of SoX with the remix effect. For example, two m ono files could be merged to form one stereo file. The first and second mono files would become the left and right channels of the stereo file.

The ‘multiply’ combining method multiplies the sample values of corresponding channels (treated as numbers in the interval −1 to +1). If the number of channels in the input files is not the same, the missing channels are considered to contain all zero.

When combining input files, SoX applies any s pecified effects (including, for example, the vol volume adjustment effect) after the audio has been combined. However, it is o ften useful to be able to set the vol- ume of (i.e. ‘balance’) the inputs individually, b efore combining takes place. For a ll combining methods, input file volume adjustments can be made manually using the −v option (below) which can be given f or one or more input files. If it is given f or only some of the input files then the others receive no v olume adjustment. In some circumstances, automatic volume adjustments may be applied (see below).

The −V option (below) can be used to show t he input file volume adjustments that have been selected (either manually or automatically).

There are some special considerations that need to made when mixing input files:

Unlike t he other methods, ‘mix’ combining has the potential to cause clipping in the combiner if no balancing is performed. In this case, if manual volume adjustments are not given, SoX will try to ensure that clipping does not occur by automatically adjusting the volume (amplitude) of each input signal by a factor of ¹/n, w here n is the number of input files. If this results in audio that is too quiet or otherwise unbalanced then the input file volumes can be set manually as described above. U sing the norm effect on the mix is another alternative.

If mixed audio seems loud enough at some points but too quiet in others then dynamic range compression should be applied to correct this—see the comp and effect.

With the ‘mix-power’ combine method, the mixed volume is approximately equal to that of one of the input signals. This is achieved b y balancing using a factor of ¹/√n instead of ¹/n. Note that this balancing factor does not guarantee that clipping will not occur, but the number of clips will usually be low a nd the resultant distortion is generally imperceptible.

Output Files[edit]

SoX’s d efault behaviour is to take o ne or more input files and write them to a single output file. This behaviour can be changed by specifying the pseudo-effect ‘newfile’ within the effects list. SoX will then enter multiple output mode.

In multiple output mode, a new fi le is created when the effects prior to the ‘newfile’ indicate they a re done. The effects chain listed after ‘newfile’ is then started up and its output is saved to t he new fi le. In multiple output mode, a unique number will automatically be appended to the end of all filenames. If the filename has an extension then the number is inserted before the extension. This behaviour can be custom- ized by placing a %n anywhere in the filename where the number should be substituted. An optional num- ber can be placed after the % to indicate a minimum fixed width for the number.

Multiple output mode is not very useful unless an effect that will stop the effects chain early is specified before the ‘newfile’. If end of file is reached before the effects chain stops itself then no new fi le will be cre- ated as it would be empty.

The following is an example of splitting the first 60 seconds of an input file into two 30 s econd files and ignoring the rest. 

sox song.wav ringtone%1n.wav trim 0 30 : newfile : trim 0 30

Stopping SoX[edit]

Usually SoX will complete its processing and exit automatically once it has read all available audio data from the input files.

If desired, it can be terminated earlier by sending an interrupt signal to the process (usually by pressing the

keyboard interrupt key which is normally Ctrl-C). This is a natural requirement in some circumstances, e.g. when using SoX to make a r ecording. Note that when using SoX to play multiple files, Ctrl-C behaves slightly differently: pressing it once causes SoX to skip to the next file; pressing it twice in quick succession causes SoX to exit. Another option to stop processing early is to use an effect that has a time period or sample count to determine the stopping point. The trim effect is an example of this. Once all effects chains have stopped then SoX will also stop.

FILENAMES[edit]

Filenames can be simple file names, absolute or relative path names, or URLs (input files only). Note that URL support requires that wget(1) is available. Note: Giving SoX an input or output filename that is the same as a SoX effect-name will not work since SoX will treat it as an effect specification. The only work-around to this is to avoid such filenames. This is generally not difficult since most audio filenames have a filename ‘extension’, whilst effect-names do not.

Special Filenames[edit]

The following special filenames may be used in certain circumstances in place of a normal filename on the command line:

− SoX can be used in simple pipeline operations by using the special filename ‘−’ which, if used as an input filename, will cause SoX will read audio data from ‘standard input’ (stdin), and which, if used as the output filename, will cause SoX will send audio data to ‘standard output’ (stdout). Note that when using this option for the output file, and sometimes when using it for an input file, the file-type (see −t below) must also be given.

" | program [options] . .."[edit]

This can be used in place of an input filename to specify the the given p rogram’s s tandard output(stdout) be used as an input file. Unlike − (above), this can be used for several inputs to one SoX command. For example, if ‘genw’ generates mono WAV formatted signals to its standard output, then the following command makes a stereo file from two g enerated signals: sox −M "|genw −−imd −" "|genw −−thd −" out.wav For h eaderless (raw) audio, −t (and perhaps other format options) will need to be given, preceding the input command.

"wildcard-filename"[edit]

Specifies that filename ‘globbing’ (wild-card matching) should be performed by SoX instead of by the shell. This allows a single set of file options to be applied to a group of files. For e xample, if the current directory contains three ‘vox’ files, file1.vox, file2.vox, and file3.vox, then 

play −−rate 6k *.vox

will be expanded by the ‘shell’ (in most environments) to 

play −−rate 6k file1.vox file2.vox file3.vox

which will treat only the first vox file as having a sample rate of 6k. With 

play −−rate 6k "*.vox"

the given s ample rate option will be applied to all three vox files.

−p, −−sox−pipe[edit]

This can be used in place of an output filename to specify that the SoX command should be used as in input pipe to another SoX command. For e xample, the command: 

play "|sox −n −p synth 2" "|sox −n −p synth 2 tremolo 10" stat

plays two ‘ files’ in succession, each with different effects.

−p is in fact an alias for ‘−t sox −’.

−d, −−default−device[edit]

This can be used in place of an input or output filename to specify that the default audio device (if one has been built into SoX) is to be used. This is akin to invoking rec or play (as described above).


-n, −−null[edit]

This can be used in place of an input or output filename to specify that a ‘null file’ is to be used. Note that here, ‘null file’ refers to a SoX-specific mechanism and is not related to any o perating-system mechanism with a similar name. Using a null file to input audio is equivalent to using a normal audio file that contains an infinite amount of silence, and as such is not generally useful unless used with an effect that specifies a finite time length (such as trim or synth).

Using a null file to output audio amounts to discarding the audio and is useful mainly with effects that produce information about the audio instead of affecting it (such as noiseprof or stat). The sampling rate associated with a null file is by default 48 kHz, but, as with a normal file, this can be overridden if desired using command-line format options (see below).

Supported File & Audio Device Types[edit]

See soxformat(7) for a list and description of the supported file formats and audio device drivers.

OPTIONS[edit]

Global Options[edit]

These options can be specified on the command line at any p oint before the first effect name.

The SOX_OPTS environment variable can be used to provide alternative default values for SoX’s global options. For example: 

SOX_OPTS="−−buffer 20000 −−play−rate−arg −hs −−temp /mnt/temp"

Note that setting SOX_OPTS can potentially create unwanted changes in the behaviour of scripts or other programs that invoke SoX. SOX_OPTS might best be used for things (such as in the given example) that reflect the environment in which SoX is being run. Enabling options such as −−no−clobber as default might be handled better using a shell alias since a shell alias will not affect operation in scripts etc.

One way to ensure that a script cannot be affected by SOX_OPTS is to clear SOX_OPTS at the start of thescript, but this of course loses the benefit of SOX_OPTS carrying some system-wide default options. An alternative approach is to explicitly invoke SoX with default option values, e.g. 

SOX_OPTS="−V −−no-clobber"
...
sox −V2 −−clobber $input $output ...

Note that the way to set environment variables varies from system to system. Here are some examples: Unix bash: export SOX_OPTS="−V −−no-clobber" Unix csh: 

setenv SOX_OPTS "−V −−no-clobber"

MS-DOS/MS-Windows: 

set SOX_OPTS=−V −−no-clobber

MS-Windows GUI: via Control Panel : System : Advanced : Environment Variables Mac OS X GUI: Refer to Apple’s Technical Q&A QA1067 document.

−−buffer BYTES, −−input−buffer BYTES[edit]

Set the size in bytes of the buffers used for processing audio (default 8192). −−buffer applies to input, effects, and output processing; −−input−buffer applies only to input processing (for which it overrides −−buffer if both are given). Be aware that large values for −−buffer will cause SoX to be become slow to r espond to requests to terminate or to skip the current input file.

−−clobber[edit]

Don’t prompt before overwriting an existing file with the same name as that given f or the output file. This is the default behaviour.


−−combine concatenate | merge | mix | mix−power | multiply | sequence[edit]

Select the input file combining method; for some of these, short options are available: −m selects ‘mix’, −M selects ‘merge’, and −T selects ‘multiply’.

See Input File Combining above for a description of the different combining methods.

−D, −−no−dither[edit]

Disable automatic dither—see ‘Dithering’ above. A n e xample of why t his might occasionally be useful is if a file has been converted from 16 to 24 bit with the intention of doing some processing on it, but in fact no processing is needed after all and the original 16 bit file has been lost, then, strictly speaking, no dither is needed if converting the file back to 16 bit. See also the stats effect for how to determine the actual bit depth of the audio within a file.

−−effects−file FILENAME[edit]

Use FILENAME to obtain all effects and their arguments. The file is parsed as if the values were specified on the command line. A new line can be used in place of the special : marker to separate effect chains. For convenience, such markers at the end of the file are normally ignored; if you want to specify an empty last effects chain, use an explicit : by itself on the last line of the file. This option causes any e ffects specified on the command line to be discarded. −G, −−guard

Automatically invoke the gain effect to guard against clipping. E.g. sox −G infile −b 16 outfile rate 44100 dither −s is shorthand for 

sox infile −b 16 outfile gain −h rate 44100 gain −rh dither −s

See also −V, − −norm, and the gain effect.

−h, −−help[edit]

Show version number and usage information.

−−help−effect NAME[edit]

Show usage information on the specified effect. The name all can be used to show u sage on all effects.

−−help−format NAME[edit]

Show i nformation about the specified file format. The name all can be used to show i nformation on all formats.

−−i, −−info[edit]

Only if given as t he first parameter to sox, behave as soxi(1).

−m | −M[edit]

Equivalent to −−combine mix and −−combine merge, respectively.

−−magic[edit]

If SoX has been built with the optional ‘libmagic’ library then this option can be given to enable its use in helping to detect audio file types.

−−multi−threaded | −−single−threaded[edit]

By default, SoX is ‘single threaded’. If the −−multi−threaded option is given however then SoX will process audio channels for most multi-channel effects in parallel on hyper-threading/multi- core architectures. This may reduce processing time, though sometimes it may be necessary to use this option in conjunction with a larger buffer size than is the default to gain any benefit from multi-threaded processing (e.g. 131072; see −−buffer above).

−−no−clobber[edit]

Prompt before overwriting an existing file with the same name as that given f or the output file. N.B. Unintentionally overwriting a file is easier than you might think, for example, if you accidentally enter 

sox file1 file2 effect1 effect2 ...

when what you really meant was

play file1 file2 effect1 effect2 ...


then, without this option, file2 will be overwritten. Hence, using this option is recommended.

SOX_OPTS (above), a ‘shell’ alias, script, or batch file may be an appropriate way of permanently enabling it. −−norm[=dB-level] Automatically invoke the gain effect to guard against clipping and to normalise the audio. E.g. sox −−norm infile −b 16 outfile rate 44100 dither −s is shorthand for sox infile −b 16 outfile gain −h rate 44100 gain −nh dither −s Optionally, t he audio can be normalized to a given l ev el ( usually) below 0 d BFS: sox −−norm=−3 infile outfile See also −V, − G, and the gain effect. −−play−rate−arg A RG Selects a quality option to be used when the ‘rate’ effect is automatically invoked w hilst playing audio. This option is typically set via the SOX_OPTS environment variable (see above). −−plot gnuplot | octave | off If not set to off (the default if −−plot is not given), run in a mode that can be used, in conjunction with the gnuplot program or the GNU Octave program, to assist with the selection and configura- tion of many of t he transfer-function based effects. For the first given e ffect that supports the selected plotting program, SoX will output commands to plot the effect’s t ransfer function, and then exit without actually processing any a udio. E.g. sox −−plot octave input-file −n highpass 1320 > highpass.plt octave highpass.plt −q, −−no−show−progress Run in quiet mode when SoX wouldn’t o therwise do so. This is the opposite of the −S option. −R Run in ‘repeatable’ mode. When this option is given, where applicable, SoX will embed a fixed time-stamp in the output file (e.g. AIFF) a nd will ‘seed’ pseudo random number generators (e.g. dither) w ith a fixed number, t hus ensuring that successive SoX invocations with the same inputs and the same parameters yield the same output. −−replay−gain track | album | off Select whether or not to apply replay-gain adjustment to input files. The default is off for sox and rec, album for play where (at least) the first two i nput files are tagged with the same Artist and Album names, and track for play otherwise. −S, −−show−progress Display input file format/header information, and processing progress as input file(s) percentage complete, elapsed time, and remaining time (if known; shown in brackets), and the number of samples written to the output file. Also shown is a peak-level m eter, a nd an indication if clipping has occurred. The peak-level m eter shows up to two c hannels and is calibrated for digital audio as follows (right channel shown): dB FSD Display dB FSD Display −25 − −11 ==== −23 −9 ====−= −21 =− −7 ===== −19 == −5 =====− −17 ==− −3 ====== −15 === −1 =====! −13 ===− A t hree-second peak-held value of headroom in dBs will be shown to the right of the meter if this is below 6 dB. This option is enabled by default when using SoX to play or record audio

SoX(1) Sound eXchange SoX(1) −T Equivalent to −−combine multiply. −−temp DIRECTORY Specify that any t emporary files should be created in the given DIRECTORY . T his can be useful if there are permission or free-space problems with the default location. In this case, using ‘−−temp .’ ( to use the current directory) is often a good solution. 

−−version

Show SoX’s v ersion number and exit. 

−V[level]

Set verbosity. T his is particularly useful for seeing how a ny automatic effects have been invoked by SoX. SoX displays messages on the console (stderr) according to the following verbosity levels: 0 No messages are shown at all; use the exit status to determine if an error has occurred. 1 Only error messages are shown. These are generated if SoX cannot complete the requested commands. 2 Warning messages are also shown. These are generated if SoX can complete the requested commands, but not exactly according to the requested command parameters, or if clipping occurs. 3 Descriptions of SoX’s p rocessing phases are also shown. Useful for seeing exactly how SoX is processing your audio. 4 and above Messages to help with debugging SoX are also shown. By default, the verbosity level i s s et to 2 (shows errors and warnings). Each occurrence of the −V option increases the verbosity level by 1 . Alternatively, t he verbosity level c an be set to an absolute number by specifying it immediately after the −V, e .g. −V0 sets it to 0.

Input File Options[edit]

These options apply only to input files and may precede only input filenames on the command line.

−−ignore−length[edit]

Override an (incorrect) audio length given in an a udio file’s h eader. If t his option is given then SoX will keep reading audio until it reaches the end of the input file. 

−v, −−volume FACTOR

Intended for use when combining multiple input files, this option adjusts the volume of the file that follows it on the command line by a factor of FA C TOR. T his allows it to be ‘balanced’ w.r.t. the other input files. This is a linear (amplitude) adjustment, so a number less than 1 decreases the volume and a number greater than 1 increases it. If a negative number is given t hen in addition to the volume adjustment, the audio signal will be inverted. See also the norm, vol, a nd gain effects, and see Input File Balancing above. Input & Output File Format Options These options apply to the input or output file whose name they i mmediately precede on the command line and are used mainly when working with headerless file formats or when specifying a format for the output file that is different to that of the input file.

-b BITS, −−bits BITS[edit]

The number of bits (a.k.a. bit-depth or sometimes word-length) in each encoded sample. Not applicable to complex e ncodings such as MP3 or GSM. Not necessary with encodings that have a fixed number of bits, e.g. A/μ-law, ADPCM. For an i nput file, the most common use for this option is to inform SoX of the number of bits per sample in a ‘raw’ (‘headerless’) audio file. For e xample sox −r 16k −e signed −b 8 input.raw output.wa

converts a particular ‘raw’ file to a self-describing ‘WAV ’ fi le. For an o utput file, this option can be used (perhaps along with −e) to s et the output encoding size. By default (i.e. if this option is not given), the output encoding size will (providing it is supported by the output file type) be set to the input encoding size. For e xample sox input.cdda −b 24 output.wav converts raw CD d igital audio (16-bit, signed-integer) to a 24-bit (signed-integer) ‘WAV ’ fi le.

−c CHANNELS, −−channels CHANNELS[edit]

The number of audio channels in the audio file. This can be any n umber greater than zero. For an i nput file, the most common use for this option is to inform SoX of the number of channels in a ‘raw’ (‘headerless’) audio file. Occasionally, it m ay be useful to use this option with a ‘head- ered’ file, in order to override the (presumably incorrect) value in the header—note that this is only supported with certain file types. Examples: sox −r 48k −e float −b 32 −c 2 input.raw output.wav converts a particular ‘raw’ file to a self-describing ‘WAV ’ fi le. play −c 1 music.wav interprets the file data as belonging to a single channel regardless of what is indicated in the file header. N ote that if the file does in fact have two channels, this will result in the file playing at half speed. For an o utput file, this option provides a shorthand for specifying that the channels effect should be invoked in o rder to change (if necessary) the number of channels in the audio signal to the num- ber given. For example, the following two c ommands are equivalent: sox input.wav −c 1 output.wav bass −b 24 sox input.wav output.wav bass −b 24 channels 1 though the second form is more flexible as it allows the effects to be ordered arbitrarily.

−e ENCODING, −−encoding ENCODING[edit]

The audio encoding type. Sometimes needed with file-types that support more than one encoding type. For example, with raw, WAV , or AU ( but n ot, for example, with MP3 or FLAC). The avail- able encoding types are as follows: signed-integer PCM data stored as signed (‘two’s c omplement’) integers. Commonly used with a 16 or 24 −bit encoding size. A v alue of 0 represents minimum signal power. unsigned-integer PCM data stored as unsigned integers. Commonly used with an 8-bit encoding size. A value of 0 represents maximum signal power. floating-point PCM data stored as IEEE 753 single precision (32-bit) or double precision (64-bit) float- ing-point (‘real’) numbers. A v alue of 0 represents minimum signal power. a-law International telephony s tandard for logarithmic encoding to 8 bits per sample. It has a precision equivalent to roughly 13-bit PCM and is sometimes encoded with reversed bit- ordering (see the −X option).

u-law, m u-law[edit]

North American telephony s tandard for logarithmic encoding to 8 bits per sample. A.k.a. μ-law. It h as a precision equivalent to roughly 14-bit PCM and is sometimes encoded with reversed bit-ordering (see the −X option).

oki-adpcm[edit]

OKI (a.k.a. VOX, D ialogic, or Intel) 4-bit ADPCM; it has a precision equivalent to roughly 12-bit PCM. ADPCM is a form of audio compression that has a good compro- mise between audio quality and encoding/decoding speed SoX(1) Sound eXchange SoX(1) ima-adpcm IMA (a.k.a. DVI) 4-bit ADPCM; it has a precision equivalent to roughly 13-bit PCM. ms-adpcm Microsoft 4-bit ADPCM; it has a precision equivalent to roughly 14-bit PCM. gsm-full-rate GSM is currently used for the vast majority of the world’s d igital wireless telephone calls. It utilises several audio formats with different bit-rates and associated speech quality. SoX has support for GSM’s o riginal 13kbps ‘Full Rate’ audio format. It is usually CPU- intensive to w ork with GSM audio. Encoding names can be abbreviated where this would not be ambiguous; e.g. ‘unsigned-integer’ can be given as ‘ un’, but not ‘u’ (ambiguous with ‘u-law’). For an i nput file, the most common use for this option is to inform SoX of the encoding of a ‘raw’ (‘headerless’) audio file (see the examples in −b and −c above). For an o utput file, this option can be used (perhaps along with −b) to s et the output encoding type For e xample sox input.cdda −e float output1.wav sox input.cdda −b 64 −e float output2.wav convert raw CD d igital audio (16-bit, signed-integer) to floating-point ‘WAV ’ fi les (single & dou- ble precision respectively). By default (i.e. if this option is not given), the output encoding type will (providing it is supported by the output file type) be set to the input encoding type. −−no−glob Specifies that filename ‘globbing’ (wild-card matching) should not be performed by SoX on the following filename. For e xample, if the current directory contains the two fi les ‘five-seconds.wav’ and ‘five*.wav’, then play −−no−glob "five*.wav" can be used to play just the single file ‘five*.wav’. −r, − −rate RATE[k] Gives t he sample rate in Hz (or kHz if appended with ‘k’) of the file. For an i nput file, the most common use for this option is to inform SoX of the sample rate of a ‘raw’ (‘headerless’) audio file (see the examples in −b and −c above). Occasionally it may be use- ful to use this option with a ‘headered’ file, in order to override the (presumably incorrect) value in the header—note that this is only supported with certain file types. For e xample, if audio was recorded with a sample-rate of say 48k from a source that played back a little, say 1.5%, too slowly, t hen sox −r 48720 input.wav output.wav effectively corrects the speed by changing only the file header (but see also the speed effect for the more usual solution to this problem). For an o utput file, this option provides a shorthand for specifying that the rate effect should be invoked in o rder to change (if necessary) the sample rate of the audio signal to the given v alue. For e xample, the following two c ommands are equivalent: sox input.wav −r 48k output.wav bass −b 24 sox input.wav output.wav bass −b 24 rate 48k though the second form is more flexible as it allows rate options to be given, and allows the effects to be ordered arbitrarily. −t, −−type FILE-TYPE Gives t he type of the audio file. For b oth input and output files, this option is commonly used to inform SoX of the type a ‘headerless’ audio file (e.g. raw, mp3) where the actual/desired type can- not be determined from a given fi lename extension. For example:

another-command | sox −t mp3 − output.wav sox input.wav −t raw output.bin It can also be used to override the type implied by an input filename extension, but if overriding with a type that has a header, S oX will exit with an appropriate error message if such a header is not actually present. See soxformat(7) for a list of supported file types. −L, −−endian little −B, −−endian big −x, −−endian swap These options specify whether the byte-order of the audio data is, respectively, ‘ little endian’, ‘big endian’, or the opposite to that of the system on which SoX is being used. Endianness applies only to data encoded as floating-point, or as signed or unsigned integers of 16 or more bits. It is often necessary to specify one of these options for headerless files, and sometimes necessary for (otherwise) self-describing files. A g iv en e ndian-setting option may be ignored for an input file whose header contains a specific endianness identifier, or f or an output file that is actually an audio device. N.B. Unlike o ther format characteristics, the endianness (byte, nibble, & bit ordering) of the input file is not automatically used for the output file; so, for example, when the following is run on a lit- tle-endian system: sox −B audio.s16 trimmed.s16 trim 2 trimmed.s16 will be created as little-endian; sox −B audio.s16 −B trimmed.s16 trim 2 must be used to preserve b ig-endianness in the output file. The −V option can be used to check the selected orderings. −N, −−rev erse−nibbles Specifies that the nibble ordering (i.e. the 2 halves of a byte) of the samples should be reversed; sometimes useful with ADPCM-based formats. N.B. See also N.B. in section on −x above. −X, −−rev erse−bits Specifies that the bit ordering of the samples should be reversed; sometimes useful with a few (mostly headerless) formats. N.B. See also N.B. in section on −x above. Output File Format Options These options apply only to the output file and may precede only the output filename on the command line. −−add−comment TEXT Append a comment in the output file header (where applicable). −−comment TEXT Specify the comment text to store in the output file header (where applicable). SoX will provide a default comment if this option (or −−comment−file) is n ot given. To s pecify that no comment should be stored in the output file, use −−comment "" . −−comment−file FILENAME Specify a file containing the comment text to store in the output file header (where applicable). −C, −−compression FA C TOR The compression factor for variably compressing output file formats. If this option is not given then a default compression factor will apply. T he compression factor is interpreted differently for different compressing file formats. See the description of the file formats that use this option in soxformat(7) for more information

EFFECTS[edit]

In addition to converting, playing and recording 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 com- mand line, forming an ‘effects chain’. Note that applying multiple effects in real-time (i.e. when playing audio) is likely to require a high performance computer. S topping other applications may alleviate perfor- mance issues should they o ccur. Some of the SoX effects are primarily intended to be applied to a single instrument or ‘voice’. To f acilitate this, the remix effect and the global SoX option −M can be used to isolate then recombine tracks from a multi-track recording.

Multiple Effects Chains[edit]

A s ingle effects chain is made up of one or more effects. Audio from the input runs through the chain until either the end of the input file is reached or an effect in the chain requests to terminate the chain. SoX supports running multiple effects chains over t he input audio. In this case, when one chain indicates it is done processing audio, the audio data is then sent through the next effects chain. This continues until either no more effects chains exist or the input has reached the end of the file. An effects chain is terminated by placing a : (colon) after an effect. Any f ollowing effects are a part of a new e ffects chain. It is important to place the effect that will stop the chain as the first effect in the chain. This is because any samples that are buffered by effects to the left of the terminating effect will be discarded. The amount of samples discarded is related to the −−buffer option and it should be kept small, relative to t he sample rate, if the terminating effect cannot be first. Further information on stopping effects can be found in the Stop- ping SoX section. There are a few p seudo-effects that aid using multiple effects chains. These include newfile which will start writing to a new o utput file before moving to the next effects chain and restart which will move back to the first effects chain. Pseudo-effects must be specified as the first effect in a chain and as the only effect in a chain (they m ust have a : before and after they a re specified). The following is an example of multiple effects chains. It will split the input file into multiple files of 30 seconds in length. Each output filename will have unique number in its name as documented in the Output Files section. sox infile.wav output.wav trim 0 30 : newfile : restart

Common Notation And Parameters[edit]

In the descriptions that follow, 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 repeat- able 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, s ev eral effects: center[k] See frequency. frequency[k] A f requency in H z, or, if a ppended with ‘k’, kHz. gain A p ower gain in dB. Zero gives no g ain; less than zero gives an a ttenuation. position A p osition within the audio stream; the syntax is [= | + | −]timespec, w here timespec is a time speci- fication (see below). The optional first character indicates whether the timespec is to be inter- preted relative to t he start (=) or e nd (−) of a udio, or to the previous position if the effect accepts multiple position arguments (+). The audio length must be known for end-relative locations to work; some effects do accept −0 for end-of-audio, though, even if t he length is unknown. Which of =, +, − is the default depends on the effect and is shown in its syntax as, e.g., position(+). Examples: =2:00 (two m inutes into the audio stream), −100s (one hundred samples before the end of audio), +0:12+10s (twelve s econds and ten samples after the previous position), −0.5+1s (one sample less than half a second before the end of audio). width[h | k | o | q] Used to specify the band-width of a filter. A number of different 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 e ach effect that uses this parameter, t he default method (i.e. if no character is appended) is the one that it listed first in the first line of the effect’s d escription. Most effects that expect an audio position or duration in a parameter, i .e. a time specification, a ccept either of the following two f orms: [[hours:]minutes:]seconds[.frac][t] A s pecification of ‘1:30.5’ corresponds to one minute, thirty and ½ seconds. The t suffix is entirely optional (however, see the silence effect for an exception). Note that the component val- ues do not have to be n ormalized; e.g., ‘1:23:45’, ‘83:45’, ‘79:0285’, ‘1:0:1425’, ‘1::1425’ and ‘5025’ all are legal a nd equivalent to each other. sampless Specifies the number of samples directly, as in ‘ 8000s’. For large sample counts, e n otation is sup- ported: ‘1.7e6s’ is the same as ‘1700000s’. Time specifications can also be chained with + or − into a new t ime specification where the right part is added to or subtracted from the left, respectively: ‘3:00−200s’ means two h undred samples less than three minutes. To s ee if SoX has support for an optional effect, enter sox −h and look for its name under the list: ‘EFFECTS’.

Supported Effects[edit]

Note: a categorised list of the effects can be found in the accompanying ‘README’ file. allpass frequency[k] width[h | k | o | q] Apply a two-pole all-pass filter with central frequency ( in Hz) frequency, a nd filter-width width. An all-pass filter changes the audio’s f requency to p hase relationship without changing its fre- quency to a mplitude 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. T he frequency r esponse drops logarithmically around the center fre- quency. The width parameter gives t he 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 alter- nate mode for un-pitched audio (e.g. percussion). Warning: −n introduces a power-gain of about 11dB in the filter, so b ew are of output clipping. band introduces noise in the shape of the filter, i.e. peaking at the center frequency a nd settling around it. This effect supports the −−plot global option. See also sinc 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. T he −c option applies only to bandpass and selects a constant SoX(1) Sound eXchange SoX(1) skirt gain (peak gain = Q) instead of the default: constant 0dB peak gain. The filters roll off at 6 dB per octave (20dB per decade) and are described in detail in [1]. These effects support the −−plot global option. See also sinc for a bandpass filter with steeper shoulders. bandreject frequency[k] width[h | k | o | q] Apply a band-reject filter. S ee 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 similar to that of a standard hi-fi’s t one-controls. This is also known as shelving equalisation (EQ). gain gives t he gain at 0 Hz (for bass), or whichever is t he lower of ∼22 kHz and the Nyquist fre- quency ( 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 c entral frequency a nd 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 k Hz (for treble). width determines how s teep is the filter’s s helf transition. In addition to the common width speci- fication 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. bend [−f frame-rate(25)] [−o over-sample(16)] { start-position(+),cents,end-position(+) } Changes pitch by specified amounts at specified times. Each given t riple: start-position,cents,end- position specifies one bend. cents is the number of cents (100 cents = 1 semitone) by which to bend the pitch. The other values specify the points in time at which to start and end bending the pitch, respectively. The pitch-bending algorithm utilises the Discrete Fourier Transform (DFT) at a particular frame rate and over-sampling rate. The −f and −o parameters may be used to adjust these parameters and thus control the smoothness of the changes in pitch. For e xample, an initial tone is generated, then bent three times, yielding four different notes in total: play −n synth 2.5 sin 667 gain 1 \ bend .35,180,.25 .15,740,.53 0,−520,.3 Here, the first bend runs from 0.35 to 0.6, and the second one from 0.75 to 1.28 seconds. Note that the clipping that is produced in this example is deliberate; to remove it, use gain −5 in place of gain 1. See also pitch. biquad b0 b1 b2 a0 a1 a2 Apply a biquad IIR filter with the given c oefficients. Where b* and a* are the numerator and denominator coefficients respectively. See http://en.wikipedia.org/wiki/Digital_biquad_filter (where a0 = 1). This effect supports the −−plot global option


channels CHANNELS[edit]

Invoke a simple algorithm to change the number of channels in the audio signal to the given number CHANNELS: mixing if decreasing the number of channels or duplicating if increasing the number of channels. The channels effect is invoked a utomatically if SoX’s −c option specifies a number of channels that is different to that of the input file(s). Alternatively, i f t his effect is given e xplicitly, t hen SoX’s −c option need not be given. For example, the following two c ommands are equivalent: 

sox input.wav −c 1 output.wav bass −b 24
sox input.wav output.wav bass −b 24 channels 1

though the second form is more flexible as it allows the effects to be ordered arbitrarily. See also remix for an effect that allows channels to be mixed/selected arbitrarily. 

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, b ut 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. H ence the delayed sound will sound slower or faster, t hat is the delayed sound tuned around the original one, like in a c horus where some vocals are slightly off k ey. S ee [3] for more discussion of the chorus effect. Each four-tuple parameter delay/decay/speed/depth gives t he delay in milliseconds and the decay (relative to g ain-in) with a modulation speed in Hz using depth in milliseconds. The modulation is either sinusoidal (−s) or t riangular (−t). Gain-out is the volume of the output. A t ypical delay is around 40ms to 60ms; the modulation speed is best near 0.25Hz and the modu-lation depth around 2ms. For e xample, 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 f uller 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 w hich the instantaneous level of t he input signal is averaged to determine its volume; attacks refer to increases in volume and decays refer to decreases. For m ost situations, the attack time (response to the music getting louder) should be shorter than the decay time because the human ear is more sensitive 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. Ty pical values are 0.3,0.8 seconds. The second parameter is a list of points on the compander’s t ransfer function specified in dB relative to t he maximum possible signal amplitude. The input values must be in a strictly increasing order but the transfer function does not have to be m onotonically rising. If omitted, the value of out-dB1 defaults to the same value as in-dB1; l ev els 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 amount given. Typical values for the transfer func- tion 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 a ssumed for each channel when compand- ing starts. This permits the user to supply a nominal level i nitially, s o t hat, 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 compan- der gain properly adjusts itself. A t ypical 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, b ut it is delayed before being fed to the volume adjuster. S pecifying a delay approximately equal to the attack/decay times allows the compander to effectively operate in a ‘ predictive’ r ather than a reactive mode. A typical value is 0.2 seconds.

The following example might be used to make a p iece of music with both quiet and loud passages suitable for listening to in a noisy environment such as a moving vehicle: sox asz.wav asz-car.wav 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 compander 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 t he music but narrow e nough 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 e xperimentation). 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. In the next example, compand is being used as a noise-gate for when the noise is at a lower level than the signal:

play infile compand .1,.2 −inf,−50.1,−inf,−50,−50 0 −90 .1 Here is another noise-gate, this time for when the noise is at a higher level t han the signal (making it, in some ways, similar to squelch): play infile compand .1,.1 −45.1,−45,−inf,0,−inf 45 −90 .1 This effect supports the −−plot global option (for the transfer function). See also mcompand for a multiple-band companding effect. contrast [enhancement-amount(75)] Comparable with compression, this effect modifies an audio signal to make it s ound louder. enhancement-amount controls the amount of the enhancement and is a number in the range 0−100. Note that enhancement-amount = 0 s till gives a s ignificant contrast enhancement. See also the compand and mcompand effects. dcshift shift [limitergain] Apply a DC shift to the audio. This can be useful to remove a DC o ffset (caused perhaps by a hardware problem in the recording chain) from the audio. The effect of a DC offset is reduced headroom and hence volume. The stat or stats effect can be used to determine if a signal has a DC offset.


The given dcshift value is a floating point number in the range of ±2 t hat indicates the amount to shift the audio (which is in the range of ±1). 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 s ay 10Hz, as illustrated in the following example: sox −n dc.wav synth 5 sin %0 50 sox dc.wav fixed.wav highpass 10 deemph Apply Compact Disc (IEC 60908) de-emphasis (a treble attenuation shelving filter).


Pre-emphasis was applied in the mastering of some CDs issued in the early 1980s. These included many c lassical music albums, as well as now s ought-after issues of albums by The Beatles, Pink Floyd and others. Pre-emphasis should be removed at p layback time by a de-emphasis filter in the playback device. However, not all modern CD players have this filter, a nd very few PC CD d rives have it; playing pre-emphasised audio without the correct de-emphasis filter results in audio that sounds harsh and is far from what its creators intended. With the deemph effect, it is possible to apply the necessary de-emphasis to audio that has been extracted from a pre-emphasised CD, and then either burn the de-emphasised audio to a new C D (which will then play correctly on any CD p layer), or simply play the correctly de-emphasised audio files on the PC. For e xample: sox track1.wav track1−deemph.wav deemph and then burn track1-deemph.wav to C D, or play track1−deemph.wav or simply play track1.wav deemph


The de-emphasis filter is implemented as a biquad and requires the input audio sample rate to be either 44.1kHz or 48kHz. Maximum deviation from the ideal response is only 0.06dB (up to 20kHz). This effect supports the −−plot global option. See also the bass and treble shelving equalisation effects. delay {position(=)} Delay one or more audio channels such that they s tart at the given position. F or example, delay 1.5 +1 3 000s delays the first channel by 1.5 s econds, the second channel by 2.5 s econds (one sec- ond more than the previous channel), the third channel by 3000 samples, and leaves a ny other channels that may be present un-delayed. The following (one long) command plays a chime sound: play −n synth −j 3 sin %3 sin %−2 sin %−5 sin %−9 \ sin %−14 sin %−21 fade h .01 2 1.5 delay \ 1.3 1 .76 .54 .27 remix − fade h 0 2.7 2.5 norm −1 and this plays a guitar chord: play −n synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 \ delay 0 .05 .1 .15 .2 .25 remix − fade 0 4 .1 norm −1 dither [−S | −s | −f filter] [ −a] [ −p precision]


Apply dithering to the audio. Dithering deliberately adds a small amount of noise to the signal in order to mask audible quantization effects that can occur if the output sample size is less than 24 bits. With no options, this effect will add triangular (TPDF) white noise. Noise-shaping (only for certain sample rates) can be selected with −s. W ith the −f option, it is possible to select a particu- lar noise-shaping filter from the following list: lipshitz, f-weighted, modified-e-weighted, improved-e-weighted, gesemann, shibata, low-shibata, high-shibata. Note that most filter types are available only with 44100Hz sample rate. The filter types are distinguished by the following properties: audibility of noise, level of ( inaudible, but in some circumstances, otherwise problem- atic) shaped high frequency n oise, and processing speed. See http://sox.sourceforge.net/SoX/NoiseShaping for graphs of the different noise-shaping curves.


The −S option selects a slightly ‘sloped’ TPDF, b iased towards higher frequencies. It can be used at any s ampling rate but below ≈22k, plain TPDF is probably better, a nd above ≈ 37k, noise-shap- ing (if available) is probably better. The −a option enables a mode where dithering (and noise-shaping if applicable) are automatically enabled only when needed. The most likely use for this is when applying fade in or out to an already dithered file, so that the redithering applies only to the faded portions. However, auto dithering is not fool-proof, so the fades should be carefully checked for any n oise modulation; if this occurs, then either re-dither the whole file, or use trim, fade, a nd concatencate. The −p option allows overriding the target precision. If the SoX global option −R option is not given, then the pseudo-random number generator used to generate the white noise will be ‘reseeded’, i.e. the generated noise will be different between invo- cations. This effect should not be followed by any o ther effect that affects the audio. See also the ‘Dithering’ section above. downsample [factor(2)]


Downsample the signal by an integer factor: Only the first out of each factor samples is retained, the others are discarded. No decimation filter is applied. If the input is not a properly bandlimited baseband signal, aliasing will occur. T his may be desirable, e.g., for frequency t ranslation. For a g eneral resampling effect with anti-aliasing, see rate. S ee also upsample. earwax


Makes audio easier to listen to on headphones. Adds ‘cues’ to 44.1kHz stereo (i.e. audio CD for- mat) audio so that when listened to on headphones the stereo image is moved f rom inside your head (standard for headphones) to outside and in front of the listener (standard for speakers). 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 buildings) when talking or shouting; digital echo effects emulate this behav- iour and are often used to help fill out the sound of a single instrument or vocal. The time differ- ence between the original signal and the reflection is the ‘delay’ (time), and the loudness of the reflected signal is the ‘decay’. Multiple echoes can have different delays and decays. Each given delay decay pair gives t he delay in milliseconds and the decay (relative to g ain-in) of that echo. Gain-out is the volume of the output. For e xample: This will make it s ound as if there are twice as many i nstruments 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 l onger delay will sound like an o pen air concert in the mountains: 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 t he delay in milliseconds and the decay (relative to g ain-in) of that echo. Gain-out is the volume of the output.

Like t he 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 e chos; 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: SoX(1) Sound eXchange SoX(1) 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. W ith this filter, t he signal-level at a nd around a selected frequency c an be increased or decreased, whilst (unlike b and-pass and band-reject filters) that at all other frequencies is unchanged. frequency gives t he filter’s c entral frequency i n H z, width, t he band-width, and gain the required gain or attenuation in dB. Beware of Clipping when using a positive gain. In order to produce complex e qualisation curves, this effect can be given s ev eral 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-position(=) [fade-out-length]] Apply a fade effect to the beginning, end, or both of the audio. An optional type can be specified to select the shape of the fade curve: q for quarter of a sine wave, h for half a sine wav e, t for linear (‘triangular’) slope, l for logarithmic, and p for inverted parabola. The default is logarithmic.

A f ade-in starts from the first sample and ramps the signal level f rom 0 to full volume over t he time given a s fade-in-length. S pecify 0 if no fade-in is wanted. For f ade-outs, the audio will be truncated at stop-position and the signal level w ill be ramped from full volume down to 0 over an i nterval of fade-out-length before the stop-position. I f fade-out- length is not specified, it defaults to the same value as fade-in-length. N o f ade-out is performed if stop-position is not specified. If the audio length can be determined from the input file header and any p revious effects, then −0 (or, f or historical reasons, 0) m ay be specified for stop-position to indicate the usual case of a fade-out that ends at the end of the input audio stream.

Any time specification may be used for fade-in-length and fade-out-length. See also the splice effect. fir [coefs-file | coefs]


Use SoX’s F FT convolution engine with given F IR filter coefficients. If a s ingle argument is given then this is treated as the name of a file containing the filter coefficients (white-space separated; may contain ‘#’ comments). If the given fi lename is ‘−’, or if no argument is given, then the coef-ficients are read from the ‘standard input’ (stdin); otherwise, coefficients may be given on the com-mand line. Examples: sox infile outfile fir 0.0195 −0.082 0.234 0.891 −0.145 0.043 sox infile outfile fir coefs.txt with coefs.txt containing

  1. HP f ilter
  2. f req=10000

1.2311233052619888e−01 −4.4777096106211783e−01 5.1031563346705155e−01 −6.6502926320995331e−02 ... This effect supports the −−plot global option. flanger [delay depth reg en w idth 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 D efault Description delay 0 − 3 0 0 Base delay in milliseconds. depth 0 − 1 0 2 Added swept delay in milliseconds. regen −95 − 95 0 Percentage regeneration (delayed signal feedback). width 0 − 1 00 71 Percentage of delayed signal mixed with original. speed 0.1 − 1 0 0 .5 S weeps per second (Hz). shape sin Swept wave shape: sine | triangle. phase 0 − 1 00 25 Swept wav e 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 [−e | −B | −b | −r] [ −n] [ −l | −h] [ gain-dB]

Apply amplification or attenuation to the audio signal, or, in s ome cases, to some of its channels. Note that use of any o f −e, −B, −b, −r, o r −n requires temporary file space to store the audio to be processed, so may be unsuitable for use with ‘streamed’ audio. Without other options, gain-dB is used to adjust the signal power level b y t he given n umber of dB:

positive amplifies (beware of Clipping), negative attenuates. With other options, the gain-dB amplification or attenuation is (logically) applied after the processing due to those options.

Given t he −e option, the levels of the audio channels of a multi-channel file are ‘equalised’, i.e.

gain is applied to all channels other than that with the highest peak level, such that all channels attain the same peak level ( but, without also giving −n, t he audio is not ‘normalised’). The −B (balance) option is similar to −e, b ut with −B, the RMS level is u sed instead of the peak level. −B might be used to correct stereo imbalance caused by an imperfect record turntable car- tridge. Note that unlike −e, −B might cause some clipping. −b is similar to −B but h as clipping protection, i.e. if necessary to prevent clipping whilst balanc- ing, attenuation is applied to all channels. Note, however, that in conjunction with −n, −B and −b are synonymous.

The −r option is used in conjunction with a prior invocation of gain with the −h option—see below f or details. The −n option normalises the audio to 0dB FSD; it is often used in conjunction with a negative gain-dB to the effect that the audio is normalised to a given l ev el b elow 0 dB. For example, sox infile outfile gain −n normalises to 0dB, and sox infile outfile gain −n −3 normalises to −3dB. The −l option invokes a s imple limiter, e .g. sox infile outfile gain −l 6

will apply 6dB of gain but never c lip. Note that limiting more than a few d Bs more than occasion- ally (in a piece of audio) is not recommended as it can cause audible distortion. See the compand effect for a more capable limiter. The −h option is used to apply gain to provide head-room for subsequent processing. For e xam- ple, withsox infile outfile gain −h bass +6 6dB of attenuation will be applied prior to the bass boosting effect thus ensuring that it will not clip. Of course, with bass, it is obvious how much headroom will be needed, but with other effects


(e.g. rate, dither) it is not always as clear. A nother advantage of using gain −h rather than an explicit attenuation, is that if the headroom is not used by subsequent effects, it can be reclaimed with gain −r, f or example: 

sox infile outfile gain −h bass +6 rate 44100 gain −r


The above effects chain guarantees never to c lip nor amplify; it attenuates if necessary to prevent clipping, but by only as much as is needed to do so.

Output formatting (dithering and bit-depth reduction) also requires headroom (which cannot be ‘reclaimed’), e.g. sox infile outfile gain −h bass +6 rate 44100 gain −rh dither Here, the second gain invocation, reclaims as much of the headroom as it can from the preceding effects, but retains as much headroom as is needed for subsequent processing. The SoX global option −G can be given to a utomatically invoke gain −h and gain −r.

See also the norm and vol effects.

highpass | lowpass [−1|−2] frequency[k] [ width[q | o | h | k]] Apply a high-pass or low-pass filter with 3dB point frequency. T he 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 B utterworth response. The filters roll off at 6 dB 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 sinc for filters with a steeper roll-off. hilbert [−n taps]

Apply an odd-tap Hilbert transform filter, p hase-shifting the signal by 90 degrees. This is used in many m atrix coding schemes and for analytic signal generation. The process is often written as a multiplication by i (or j), the imaginary unit.

An odd-tap Hilbert transform filter has a bandpass characteristic, attenuating the lowest and high- est frequencies. Its bandwidth can be controlled by the number of filter taps, which can be speci- fied with −n. B y d efault, the number of taps is chosen for a cutoff f requency of a bout 75 Hz. This effect supports the −−plot global option. ladspa [-l | -r] module [plugin] [ argument ...] Apply a LADSPA [ 5] (Linux Audio Developer’s S imple 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 p lugin 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 o ther arguments are for the control ports of the plugin. Missing arguments are supplied by default values if possible. Normally, t he number of input ports of the plugin must match the number of input channels, and the number of output ports determines the output channel count. However, the −r (replicate) option allows cloning a mono plugin to handle multi-channel input. Some plugins introduce latency w hich SoX may optionally compensate for. T he −l (latency c om- pensation) option automatically compensates for latency as r eported by the plugin via an output control port named "latency". If found, the environment variable LADSPA_PAT H w ill be used as search path for plugins. loudness [gain [reference]] Loudness control—similar to the gain effect, but provides equalisation for the human auditory system. See http://en.wikipedia.org/wiki/Loudness for a detailed description of loudness. The gain is adjusted by the given gain parameter (usually negative) a nd the signal equalised according to ISO 226 w.r.t. a reference level of 6 5dB, though an alternative reference level m ay be given i f the original audio has been equalised for some other optimal level. A default gain of −10dB is

used if a gain value is not given. See also the gain effect. lowpass [−1|−2] frequency[k] [ width[q | o | h | k]] Apply a low-pass filter. S ee 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]]]" {crossover-freq[k] " attack1,..."} The multi-band compander is similar to the single-band compander but the audio is first divided into bands using Linkwitz-Riley c ross-over fi lters 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 f requency f or that band is given b y crossover- freq; t hese can be repeated to create multiple bands. For e xample, the following (one long) command shows how m ulti-band companding is typically used in FM radio: play track1.wav gain −3 sinc 8000− 29 100 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 sinc −n 255 −b 16 −17500 \ gain 9 lowpass −1 17801 The audio file is played with a simulated FM radio sound (or broadcast signal condition if the low- pass filter at the end is skipped). Note that the pipeline is set up with US-style 75us pre-emphasis. See also compand for a single-band companding effect. 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 u se it, first run SoX with the noise- prof 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 s tdout if no profile-file or if ‘−’ is given. E.g. sox speech.wav −n trim 0 1.5 noiseprof speech.noise-profile To a ctually 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, o r from stdin if no profile-file or if ‘−’ is given. E.g. sox speech.wav cleaned.wav noisered speech.noise-profile 0.3 How much noise should be removed is s pecified 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 remov- ing 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 w ith the results, paying particular attention to quieter sections of the audio. On most systems, the two s tages—profiling and reduction—can be combined using a pipe, e.g.sox noisy.wav −n trim 0 1 noiseprof | play noisy.wav noisered norm [dB-level] Normalise the audio. norm is just an alias for gain −n; s ee the gain effect for detail SoX(1) Sound eXchange 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. It is equiva- lent to remix 1,2i 1,2i. overdrive [gain(20) [colour(20)]] Non linear distortion. The colour parameter controls the amount of even h armonic content in the over-driven output. pad { length[@position(=)] } Pad t he audio with silence, at the beginning, the end, or any specified points through the audio. length is the amount of silence to insert and position the position in the input audio stream at which to insert it. Any n umber of lengths and positions may be specified, provided that a specified position is not less that the previous one, and any t ime specification may be used for them. posi- tion is optional for the first and last lengths specified and if omitted correspond to the beginning and the end of the audio respectively. F or example, pad 1.5 1 .5 adds 1.5 s econds of silence pad- ding 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. See also delay for an effect that can add silence at the beginning of the audio on a channel-by- channel basis. 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 t he delay in milliseconds and the decay (relative to g ain-in) with a modu-lation speed in Hz. The modulation is either sinusoidal (−s) — preferable for multiple instruments,or triangular (−t) — gives s ingle instruments a sharper phasing effect. The decay should be less than 0.5 to a void feedback, and usually no less than 0.1. Gain-out is the volume of the output. For e xample: 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 p opular 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 pitch [−q] shift [segment [search [overlap]]] Change the audio pitch (but not tempo). shift gives t he pitch shift as positive or n eg ative ‘cents’ (i.e. 100ths of a semitone). See the tempo effect for a description of the other parameters. See also the bend, speed, a nd tempo effects. rate [−q | −l | −m | −h | −v] [ override-options] RATE[k] Change the audio sampling rate (i.e. resample the audio) to any g iv en RATE (even n on-integer if

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