Format options effect the audio samples that they immediately precede. If
they are placed before the input file name then they effect the input
data. If they are placed before the output file name then they will
effect the output data. By taking advantage of this, you can override
a input file's corrupted header or produce an output file that is totally
different style then the input file. It is also how SoX is informed about
the format of raw input data.
- avg [ -l | -r | -f | -b | n,n,...,n ]
-
Reduce the number of channels by averaging the samples,
or duplicate channels to increase the number of channels.
This effect is automatically used when the number of input
channels differ from the number of output channels. When reducing
the number of channels it is possible to manually specify the
avg effect and use the -l, -r, -f, or -b
options to select only
the left, right, front, or back channel(s) for the output instead of
averaging the channels.
The -f and -b options maintain left/right stereo
separation; use the avg effect twice to select a single channel.
The avg effect can also be invoked with up to 16 double-precision
numbers, which specify the proportion of each input channel that is
to be mixed into each output channel.
In two-channel mode, 4 numbers are given: l->l, l->r, r->l, and r->r,
respectively.
In four-channel mode, the first 4 numbers give the proportions for the
left-front output channel, as follows: lf->lf, rf->lf, lb->lf, and
rb->rf.
The next 4 give the right-front output in the same order, then
left-back and right-back.
It is also possible to use the 16 numbers to expand or reduce the
channel count; just specify 0 for unused channels.
Finally, if fewer than 4 numbers are given, certain special
abbreviations may be
invoked; see the source code for details.
- band [ -n ] center [ width ]
-
Apply a band-pass filter.
The frequency response drops logarithmically
around the
center
frequency.
The
width
gives the slope of the drop.
The frequencies at
center + width
and
center - width
will be half of their original amplitudes.
Band
defaults to a mode oriented to pitched signals,
i.e. voice, singing, or instrumental music.
The
-n
(for noise) option uses the alternate mode
for un-pitched signals.
Warning:
-n
introduces a power-gain of about 11dB in the filter, so beware
of output clipping.
Band
introduces noise in the shape of the filter,
i.e. peaking at the
center
frequency and settling around it.
See filter for a bandpass effect with steeper shoulders.
- bandpass frequency bandwidth
-
Butterworth bandpass filter. Description coming soon!
- bandreject
frequency bandwidth-
Butterworth bandreject filter. Description coming soon!
- chorus
gain-in gain-out delay decay speed depth -
-s -
Add a chorus to a sound sample. Each quadtuple
delay/decay/speed/depth gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz using depth in milliseconds.
The modulation is either sinusoidal (-s) or triangular
(-t). Gain-out is the volume of the output.
- compand attack1,decay1[,attack2,decay2...]
-
in-dB1,out-dB1[,in-dB2,out-dB2...]-
[gain [initial-volume [delay ] ] ]-
Compand (compress or expand) the dynamic range of a sample. The
attack and decay time specify the integration time over which the
absolute value of the input signal is integrated to determine its
volume; attacks refer to increases in volume and decays refer to
decreases. Where more than one pair of attack/decay parameters are
specified, each channel is treated separately and the number of pairs
must agree with the number of input channels. The second parameter is
a list of points on the compander's transfer function specified in dB
relative to the maximum possible signal amplitude. The input values
must be in a strictly increasing order but the transfer function does
not have to be monotonically rising. The special value -inf may
be used to indicate that the input volume should be associated output
volume. The points -inf,-inf and 0,0 are assumed; the
latter may be overridden, but the former may not.
The third
(optional) parameter is a post-processing gain in dB which is applied
after the compression has taken place; the fourth (optional) parameter
is an initial volume to be assumed for each channel when the effect
starts. This permits the user to supply a nominal level initially, so
that, for example, a very large gain is not applied to initial signal
levels before the companding action has begun to operate: it is quite
probable that in such an event, the output would be severely clipped
while the compander gain properly adjusts itself.
The fifth (optional) parameter is a delay in seconds.
The input signal is analyzed immediately to control the compander, but
it is delayed before being fed to the volume adjuster.
Specifying a delay approximately equal to the attack/decay times
allows the compander to effectively operate in a "predictive" rather than a
reactive mode.
- copy
-
Copy the input file to the output file.
This is the default effect if both files have the same
sampling rate.
- dcshift shift [ limitergain ]
-
DC Shift the audio data, with basic linear amplitude formula.
This is most useful if your audio data tends to not be centered around
a value of 0. Shifting it back will allow you to get the most volume
adjustments without clipping audio data.
The first option is the dcshift value. It is a floating point number that
indicates the amount to shift.
An option limtergain value can be specified as well. It should have a value much less then 1.0 and is used only on peaks to prevent clipping.
- deemph
-
Apply a treble attenuation shelving filter to samples in
audio cd format. The frequency response of pre-emphasized
recordings is rectified. The filtering is defined in the
standard document ISO 908.
- earwax
-
Makes sound easier to listen to on headphones.
Adds audio-cues to samples in audio cd format so that
when listened to on headphones the stereo image is
moved from inside
your head (standard for headphones) to outside and in front of the
listener (standard for speakers). See
www.geocities.com/beinges
for a full explanation.
- echo gain-in gain-out delay decay [ delay decay ... ]
-
Add echoing to a sound sample.
Each delay/decay part gives the delay in milliseconds
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
- echos gain-in gain-out delay decay [ delay decay ... ]
-
Add a sequence of echos to a sound sample.
Each delay/decay part gives the delay in milliseconds
and the decay (relative to gain-in) of that echo.
Gain-out is the volume of the output.
- fade [ type ] fade-in-length
-
[ stop-time [ fade-out-length ] ]-
Add a fade effect to the beginning, end, or both of the audio data.
For fade-ins, this starts from the first sample and ramps the volume of the audio from 0 to full volume over fade-in-length seconds. Specify 0 seconds if no fade-in is wanted.
For fade-outs, the audio data will be truncated at the stop-time and
the volume will be ramped from full volume down to 0 starting at
fade-out-length seconds before the stop-time. No fade-out
is performed if these options are not specified.
All times can be specified in either periods of time or sample counts.
To specify time periods use the format hh:mm:ss.frac format. To specify
using sample counts, specify the number of samples and append the letter 's'
to the sample count (for example 8000s).
An optional type can be specified to change the type of envelope. Choices are q for quarter of a sinewave, h for half a sinewave, t for linear slope, l for logarithmic, and p for inverted parabola. The default is a linear slope.
- filter [ low ]-[ high ] [ window-len [ beta ] ]
-
Apply a Sinc-windowed lowpass, highpass, or bandpass filter of given
window length to the signal.
low refers to the frequency of the lower 6dB corner of the filter.
high refers to the frequency of the upper 6dB corner of the filter.
A lowpass filter is obtained by leaving low unspecified, or 0.
A highpass filter is obtained by leaving high unspecified, or 0,
or greater than or equal to the Nyquist frequency.
The window-len, if unspecified, defaults to 128.
Longer windows give a sharper cutoff, smaller windows a more gradual cutoff.
The beta, if unspecified, defaults to 16. This selects a Kaiser window.
You can select a Nuttall window by specifying anything <= 2.0 here.
For more discussion of beta, look under the resample effect.
- flanger gain-in gain-out delay decay speed < -s | -t >
-
Add a flanger to a sound sample. Each triple
delay/decay/speed gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz.
The modulation is either sinodial (-s) or triangular
(-t). Gain-out is the volume of the output.
- highp frequency
-
Apply a single pole recursive high-pass filter.
The frequency response drops logarithmically with
I frequency
in the middle of the drop.
The slope of the filter is quite gentle.
See filter for a highpass effect with sharper cutoff.
- highpass frequency
-
Butterworth highpass filter. Description coming soon!
- lowp
frequency -
Apply a single pole recursive low-pass filter.
The frequency response drops logarithmically with
frequency
in the middle of the drop.
The slope of the filter is quite gentle.
See filter for a lowpass effect with sharper cutoff.
- lowpass frequency
-
Butterworth lowpass filter. Description coming soon!
- map
-
Display a list of loops in a sample,
and miscellaneous loop info.
- mask
-
Add "masking noise" to signal.
This effect deliberately adds white noise to a sound
in order to mask quantization effects,
created by the process of playing a sound digitally.
It tends to mask buzzing voices, for example.
It adds 1/2 bit of noise to the sound file at the
output bit depth.
- pan
direction-
Pan the sound of an audio file from one channel to another. This is done by
changing the volume of the input channels so that it fades out on one
channel and fades-in on another. If the number of input channels is
different then the number of output channels then this effect tries to
intelligently handle this. For instance, if the input contains 1 channel
and the output contains 2 channels, then it will create the missing channel
itself. The
direction
is a value from -1.0 to 1.0. -1.0 represents
far left and 1.0 represents far right. Numbers in between will start the
pan effect without totally muting the opposite channel.
- phaser gain-in gain-out delay decay speed < -s | -t >
-
Add a phaser to a sound sample. Each triple
delay/decay/speed gives the delay in milliseconds
and the decay (relative to gain-in) with a modulation
speed in Hz.
The modulation is either sinodial (-s) or triangular
(-t). The decay should be less than 0.5 to avoid
feedback. Gain-out is the volume of the output.
- pick [ -1 | -2 | -3 | -4 | -l | -r ]
-
Select the left or right channel of a stereo sample,
or one of four channels in a quadraphonic sample. The -l and -r
options represent either the left or right channel. It is required that
you use the -c 1 command line option in order to force the output file to
contain only 1 channel.
- pitch shift [ width interpole fade ]
-
Change the pitch of file without affecting its duration by cross-fading
shifted samples.
shift
is given in cents. Use a positive value to shift to treble, negative value to shift to bass.
Default shift is 0.
width
of window is in ms. Default width is 20ms. Try 30ms to lower pitch,
and 10ms to raise pitch.
interpole
option, can be "cubic" or "linear". Default is "cubic". The
fade
option, can be "cos", "hamming", "linear" or "trapezoid".
Default is "cos".
- polyphase [ -w < nut / ham > ]
-
[ -width < long / short / # > ] -
[ -cutoff # ]-
Translate input sampling rate to output sampling rate via polyphase
interpolation, a DSP algorithm. This method is slow and uses lots
of RAM, but gives much better results than
rate.
-w < nut / ham > : select either a Nuttal (~90 dB stopband) or Hamming
(~43 dB stopband) window. Default is
nut.
-width long / short / # : specify the (approximate) width of the filter.
long
is 1024 samples;
short
is 128 samples. Alternatively, an exact number can be used. Default is
long.
The
short
option is
not
recommended, as it produces poor quality results.
-cutoff # : specify the filter cutoff frequency in terms of fraction of
frequency bandwidth, also know as the Nyquist frequency. Please see
the resample effect for
further information on Nyquist frequency. If upsampling, then this is the
fraction of the original signal
that should go through. If downsampling, this is the fraction of the
signal left after downsampling. Default is 0.95. Remember that
this is a float.
- rate
-
Translate input sampling rate to output sampling rate
via linear interpolation to the Least Common Multiple
of the two sampling rates.
This is the default effect
if the two files have different sampling rates and the preview options
was specified.
This is fast but noisy:
the spectrum of the original sound will be shifted upwards
and duplicated faintly when up-translating by a multiple.
Lerp-ing is acceptable for cheap 8-bit sound hardware,
but for CD-quality sound you should instead use either
resample
or
polyphase.
If you are wondering which rate changing effects to use, you will want to read a
detailed analysis of all of them at http://eakaw2.et.tu-dresden.de/~wilde/resample/resample.html
- resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
-
Translate input sampling rate to output sampling rate
via simulated analog filtration.
This method is slower than
rate,
but gives much better results.
By default, linear interpolation is used,
with a window width about 45 samples at the lower of the two rate.
This gives an accuracy of about 16 bits, but insufficient stopband rejection
in the case that you want to have rolloff greater than about 0.80 of
the Nyquist frequency.
The -q* options will change the default values for rolloff and beta
as well as use quadratic interpolation of filter
coefficients, resulting in about 24 bits precision.
The -qs, -q, or -ql options specify increased accuracy
at the cost of lower execution speed. It is optional to specify
rolloff and beta parameters when using the -q* options.
Following is a table of the reasonable defaults which are built-in to SoX:
Option Window rolloff beta interpolation
------ ------ ------- ---- -------------
(none) 45 0.80 16 linear
-qs 45 0.80 16 quadratic
-q 75 0.875 16 quadratic
-ql 149 0.94 16 quadratic
------ ------ ------- ---- -------------
-qs, -q, or -ql use window lengths of 45, 75, or 149
samples, respectively, at the lower sample-rate of the two files.
This means progressively sharper stop-band rejection, at proportionally
slower execution times.
rolloff refers to the cut-off frequency of the
low pass filter and is given in terms of the
Nyquist frequency for the lower sample rate. rolloff therefore should
be something between 0.0 and 1.0, in practice 0.8-0.95. The defaults are
indicated above.
The Nyquist frequency is equal to (sample rate / 2). Logically,
this is because the A/D converter needs at least 2 samples to detect 1
cycle at the Nyquist frequency. Frequencies higher then the Nyquist
will actually appear as lower frequencies to the A/D converter and
is called aliasing. Normally, A/D converts run the signal through
a highpass filter first to avoid these problems.
Similar problems will happen in software when reducing the sample rate of
an audio file (frequencies above the new Nyquist frequency can be aliased
to lower frequencies). Therefore, a good resample effect
will remove all frequency information above the new Nyquist frequency.
The rolloff refers to how close to the Nyquist frequency this cutoff
is, with closer being better. When increasing the sample rate of an
audio file you would not expect to have any frequencies exist that are
past the original Nyquist frequency. Because of resampling properties, it
is common to have alaising data created that is above the old
Nyquist frequency. In that case the rolloff refers to how close
to the original Nyquist frequency to use a highpass filter to remove
this false data, with closer also being better.
The beta parameter
determines the type of filter window used. Any value greater than 2.0 is
the beta for a Kaiser window. Beta <= 2.0 selects a Nuttall window.
If unspecified, the default is a Kaiser window with beta 16.
In the case of Kaiser window (beta > 2.0), lower betas produce a somewhat
faster transition from passband to stopband, at the cost of noticeable artifacts.
A beta of 16 is the default, beta less than 10 is not recommended. If you want
a sharper cutoff, don't use low beta's, use a longer sample window.
A Nuttall window is selected by specifying any 'beta' <= 2, and the
Nuttall window has somewhat steeper cutoff than the default Kaiser window.
You will probably not need to use the beta parameter at all, unless you are
just curious about comparing the effects of Nuttall vs. Kaiser windows.
This is the default effect if the two files have different sampling rates.
Default parameters are, as indicated above, Kaiser window of length 45,
rolloff 0.80, beta 16, linear interpolation.
NOTE: -qs is only slightly slower, but more accurate for
16-bit or higher precision.
NOTE: In many cases of up-sampling, no interpolation is needed,
as exact filter coefficients can be computed in a reasonable amount of space.
To be precise, this is done when
input_rate < output_rate
&&
output_rate/gcd(input_rate,output_rate) <= 511
- reverb gain-out delay [ delay ... ]
-
Add reverberation to a sound sample. Each delay is given
in milliseconds and its feedback is depending on the
reverb-time in milliseconds. Each delay should be in
the range of half to quarter of reverb-time to get
a realistic reverberation. Gain-out is the volume of the
output.
- reverse
-
Reverse the sound sample completely.
Included for finding Satanic subliminals.
- silence above_periods [ duration threshold[ d | % ]
-
[ below_periods duration -
threshold-
Removes silence from the beginning or end of a sound file. Silence is anything below a specified threshold.
When trimming silence from the beginning of a sound file, you specify a duration of audio that is above a given silence threshold before audio data is processed. You can also specify the count of periods of none silence you want to detect before processing audio data. Specify a period of 0 if you do not want to trim data from the front of the sound file.
When optionally trimming silence form the end of a sound file, you specify the duration of audio that must be below a given threshold before stopping to process audio data. A count of periods that occur below the threshold may also be specified. If this options are not specified then data is not trimmed from the end of the audio file.
Duration counts may be in the format of time, hh:mm:ss.frac, or in the exact count of samples.
Threshold may be suffixed with d, or % to indicated the value is in decibels or a percentage of max value of the sample value. A value of '0%' will look for total silence.
- speed [ -c ] factor
-
Speed up or down the sound, as a magnetic tape with a speed control.
It affects both pitch and time. A factor of 1.0 means no change,
and is the default.
2.0 doubles speed, thus time length is cut by a half and pitch
is one octave higher.
0.5 halves speed thus time length doubles and pitch is one octave lower.
If the optional -c parameter is used then the factor is specified in "cents".
- split
-
Turn a mono sample into a stereo sample by copying
the input channel to the left and right channels.
- stat [
-s n ] [-rms ] [ -v ] [ -d ]-
Do a statistical check on the input file,
and print results on the standard error file. Audio data is passed
unmodified from input to output file unless used along with the
-e
option.
The "Volume Adjustment:" field in the statistics
gives you the argument to the
-v
number
which will make the sample as loud as possible without clipping.
The option
-v
will print out the "Volume Adjustment:" field's value only and
return. This could be of use in scripts to auto convert the
volume.
The
-s n
option is used to scale the input data by a given factor. The default value
of n is the max value of a signed long variable (0x7fffffff). Internal effects
always work with signed long PCM data and so the value should relate to this
fact.
The
-rms
option will convert all output average values to root mean square
format.
There is also an optional parameter
-d
that will print out a hex dump of the
sound file from the internal buffer that is in 32-bit signed PCM data.
This is mainly only of use in tracking down endian problems that
creep in to SoX on cross-platform versions.
- stretch factor [window fade shift fading]
-
Time stretch file by a given factor. Change duration without affecting the pitch.
factor
of stretching: >1.0 lengthen, <1.0 shorten duration.
window
size is in ms. Default is 20ms. The
fade
option, can be "lin".
shift
ratio, in [0.0 1.0]. Default depends on stretch factor. 1.0
to shorten, 0.8 to lengthen. The
fading
ratio, in [0.0 0.5]. The amount of a fade's default depends on factor
and shift.
- swap [ 1 2 | 1 2 3 4 ]
-
Swap channels in multi-channel sound files. Optionally, you may
specify the channel order you would like the output in. This defaults
to output channel 2 and then 1 for stereo and 2, 1, 4, 3 for quad-channels.
An interesting
feature is that you may duplicate a given channel by overwriting another.
This is done by repeating an output channel on the command line. For example,
swap 2 2 will overwrite channel 1 with channel 2's data; creating a stereo
file with both channels containing the same audio data.
- synth [
length ] type mix [ freq [ -freq2 ] -
[ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]-
The synth effect will generate various types of audio data. Although
this effect is used to generate audio data, an input file must be specified.
The length of the input audio file determines the length of the output
audio file.
<length> length in sec or hh:mm:ss.frac, 0=inputlength, default=0
<type> is sine, square, triangle, sawtooth, trapetz, exp,
whitenoise, pinknoise, brownnoise, default=sine
<mix> is create, mix, amod, default=create
<freq> frequency at beginning in Hz, not used for noise..
<freq2> frequency at end in Hz, not used for noise..
<freq/2> can be given as %%n, where 'n' is the number of
half notes in respect to A (440Hz)
<off> Bias (DC-offset) of signal in percent, default=0
<ph> phase shift 0..100 shift phase 0..2*Pi, not used for noise..
<p1> square: Ton/Toff, triangle+trapetz: rising slope time (0..100)
<p2> trapetz: ON time (0..100)
<p3> trapetz: falling slope position (0..100)
- trim start [ length ]
-
Trim can trim off unwanted audio data from the beginning and end of the
audio file. Audio samples are not sent to the output stream until
the start location is reached.
The optional length parameter tells the number of samples to output
after the start sample and is used to trim off the back side of the
audio data. Using a value of 0 for the start parameter will allow
trimming off the back side only.
Both options can be specified using either an amount of time and an exact count of samples. The format for specifying lengths in time is hh:mm:ss.frac. A start value of 1:30.5 will not start until 1 minute, thirty and 1/2 seconds into the audio data. The format for specifying sample counts is the number of samples with the letter 's' appended to it. A value of 8000s will wait until 8000 samples are read before starting to process audio data.
- vibro speed [ depth ]
-
Add the world-famous Fender Vibro-Champ sound
effect to a sound sample by using
a sine wave as the volume knob.
Speed
gives the Hertz value of the wave.
This must be under 30.
Depth
gives the amount the volume is cut into
by the sine wave,
ranging 0.0 to 1.0 and defaulting to 0.5.
- vol gain [ type [ limitergain ] ]
-
The vol effect is much like the command line option -v. It allows you to
adjust the volume of an input file and allows you to specify the adjustment
in relation to amplitude, power, or dB. If type is not specified then
it defaults to amplitude.
When type is
amplitude
then a linear change of the amplitude is performed based on the gain. Therefore,
a value of 1.0 will keep the volume the same, 0.0 to < 1.0 will cause the
volume to decrease and values of > 1.0 will cause the volume to increase.
Beware of clipping audio data when the gain is greater then 1.0. A negative
value performs the same adjustment while also changing the phase.
When type is
power
then a value of 1.0 also means no change in volume.
When type is
dB
the amplitude is changed logarithmically.
0.0 is constant while +6 doubles the amplitude.
An optional limitergain value can be specified and should be a
value much less
then 1.0 (ie 0.05 or 0.02) and is used only on peaks to prevent clipping.
Not specifying this parameter will cause no limiter to be used. In verbose
mode, this effect will display the percentage of audio data that needed to be
limited.
