SoX(7)

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

DESCRIPTION

  SOX EFFECTS
Multiple effects may be applied to the audio by specifying them one after another at the end of the SoX command line. Note: Brackets [ ] are used to denote parameters that are optional, braces { } to denote those that are both optional and repeatable, and angle brackets < > to denote those that are repeatable but not optional.

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

band [-n] center [width[h|o|q]]
    Apply a band-pass filter. The frequency response drops logarithmically around the center frequency. The width in Hz (the default, or if appended with `h'), in octaves (if appended with `o'), or as a Q-factor (if appended with `q'), gives the slope of the drop. The frequencies at center + width and center - width will be half of their original amplitudes. band defaults to a mode oriented to pitched audio, i.e. voice, singing, or instrumental music. The -n (for noise) option uses the alternate mode for un-pitched audio (e.g. percussion). Warning: -n introduces a power-gain of about 11dB in the filter, so beware of output clipping. band introduces noise in the shape of the filter, i.e. peaking at the center frequency and settling around it. This effect supports the --plot global option. See also filter for a bandpass filter with steeper shoulders.

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

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

bass|treble gain [frequency [width[s|h|o|q]]]
    Boost or cut the bass (lower) or treble (upper) frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's (Baxandall) tone-controls. This is also known as shelving equalisation (EQ). gain gives the dB gain at 0 Hz (for bass), or whichever is the lower of ~22 kHz and the Nyquist frequency (for treble). Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of Clipping when using a positive gain. If desired, the filter can be fine-tuned using the following optional parameters: frequency sets the filter's central frequency and so can be used to extend or reduce the frequency range to be boosted or cut. The default value is 100 Hz (for bass) or 3 kHz (for treble). width determines how steep the filter's shelf transition is and can be expressed as: a `slope' (the default, or if appended with `s'), a Q-factor (if appended with `q'), the transition width in octaves (if appended with `o'), or the transition width in Hz (if appended with `h'). The useful range of `slope' is about 0.3, for a gentle slope, to 1 (the maximum), for a steep slope; the default value is 0.5. The filters are described in detail in [1]. These effects support the --plot global option. See also equalizer for a peaking equalisation effect.

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

compand attack1,decay1{,attack2,decay2}
    [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
    [gain [initial-volume-dB [delay]]] Compand (compress or expand) the dynamic range of the audio. The attack and decay parameters (in seconds) determine the time over which the instantaneous level of the input signal is averaged to determine its volume; attacks refer to increases in volume and decays refer to decreases. Where more than one pair of attack/decay parameters are specified, each input channel is companded separately and the number of pairs must agree with the number of input channels. Typical values are 0.3,0.8 seconds. The second parameter is a list of points on the compander's transfer function specified in dB relative to the maximum possible signal amplitude. The input values must be in a strictly increasing order but the transfer function does not have to be monotonically rising. If omitted, the value of out-dB1 defaults to the same value as in-dB1; levels below in-dB1 are not companded (but may have gain applied to them). The point 0,0 is assumed but may be overridden (by 0,out-dBn). If the list is preceded by a soft-knee-dB value, then the points at where adjacent line segments on the transfer function meet will be rounded by the amout given. Typical values for the transfer function are 6:-70,-60,-20. The third (optional) parameter is an additional gain in dB to be applied at all points on the transfer function and allows easy adjustment of the overall gain. The fourth (optional) parameter is an initial level to be assumed for each channel when companding starts. This permits the user to supply a nominal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding action has begun to operate: it is quite probable that in such an event, the output would be severely clipped while the compander gain properly adjusts itself. A typical value (for audio which is initially quiet) is -90 dB. The fifth (optional) parameter is a delay in seconds. The input signal is analysed immediately to control the compander, but it is delayed before being fed to the volume adjuster. Specifying a delay approximately equal to the attack/decay times allows the compander to effectively operate in a `predictive' rather than a reactive mode. A typical value is 0.2 seconds. This effect supports the --plot global option (for the transfer function). See also mcompand for a multiple-band companding effect.

dcshift shift [limitergain]
    DC Shift the audio, with basic linear amplitude formula. This is most useful if your audio tends to not be centered around a value of 0. Shifting it back will allow you to get the most volume adjustments without clipping. The first option is the dcshift value. It is a floating point number that indicates the amount to shift. An optional limitergain can be specified as well. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping.

deemph
    Apply a treble attenuation shelving filter to audio in audio-CD format. The frequency response of pre-emphasized recordings is rectified. The filter is defined in the standard document ISO 908. This effect supports the --plot global option. See also the bass and treble shelving equalisation effects.

dither [depth]
    Apply dithering to the audio. Dithering deliberately adds digital white noise to the signal in order to mask audible quantization effects that can occur if the output sample size is less than 24 bits. By default, the amount of noise added is ½ bit; the optional depth parameter is a (linear or voltage) multiplier of this amount. This effect should not be followed by any other effect that affects the audio.

earwax
    Makes audio easier to listen to on headphones. Adds `cues' to audio in audio-CD format so that when listened to on headphones the stereo image is moved from inside your head (standard for headphones) to outside and in front of the listener (standard for speakers). See http://www.geocities.com/beinges for a full explanation.

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

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

equalizer frequency width[q|o|h] gain
    Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at and around a selected frequency can be increased or decreased, whilst (unlike band-pass and band-reject filters) that at all other frequencies is unchanged. frequency gives the filter's central frequency in Hz, width, the band-width, as a Q-factor [2] (the default, or if appended with `q'), in octaves (if appended with `o'), or in Hz (if appended with `h'), and gain the required gain or attenuation in dB. Beware of Clipping when using a positive gain. In order to produce complex equalisation curves, this effect can be given several times, each with a different central frequency. The filter is described in detail in [1]. This effect supports the --plot global option. See also bass and treble for shelving equalisation effects.

fade [type] fade-in-length [stop-time [fade-out-length]]
    Add a fade effect to the beginning, end, or both of the audio. For fade-ins, this starts from the first sample and ramps the volume of the audio from 0 to full volume over fade-in-length seconds. Specify 0 seconds if no fade-in is wanted. For fade-outs, the audio will be truncated at stop-time and the volume will be ramped from full volume down to 0 starting at fade-out-length seconds before the stop-time. If fade-out-length is not specified, it defaults to the same value as fade-in-length. No fade-out is performed if stop-time is not specified. All times can be specified in either periods of time or sample counts. To specify time periods use the format hh:mm:ss.frac format. To specify using sample counts, specify the number of samples and append the letter `s' to the sample count (for example `8000s'). An optional type can be specified to change the type of envelope. Choices are q for quarter of a sine wave, h for half a sine wave, t for linear slope, l for logarithmic, and p for inverted parabola. The default is a linear slope.

filter [low]-[high] [window-len [beta]]
    Apply a sinc-windowed lowpass, highpass, or bandpass filter of given window length to the signal. low refers to the frequency of the lower 6dB corner of the filter. high refers to the frequency of the upper 6dB corner of the filter. A low-pass filter is obtained by leaving low unspecified, or 0. A high-pass filter is obtained by leaving high unspecified, or 0, or greater than or equal to the Nyquist frequency. The window-len, if unspecified, defaults to 128. Longer windows give a sharper cutoff, smaller windows a more gradual cutoff. The beta, if unspecified, defaults to 16. This selects a Kaiser window. You can select a Nuttall window by specifying anything ≤ 2 here. For more discussion of beta, look under the resample effect.

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

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

key [-q] shift [segment [search [overlap]]]
    Change the audio key (i.e. pitch but not tempo) using a WSOLA algorithm. shift gives the key shift as positive or negative `cents' (i.e. 100ths of a semitone). See the tempo effect for a description of the other parameters. See also pitch for a similar effect.

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

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

mcompand "attack1,decay1{,attack2,decay2}
    [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2}
    [gain [initial-volume-dB [delay]]]" {xover-freq "attack1,..."} The multi-band compander is similar to the single-band compander but the audio is first divided into bands using Butterworth cross-over filters and a separately specifiable compander run on each band. See the compand effect for the definition of its parameters. Compand parameters are specified between double quotes and the crossover frequency for that band is given by xover-freq; these can be repeated to create multiple bands. See also compand for a single-band companding effect.

mixer [ -l|-r|-f|-b|-1|-2|-3|-4|n{,n} ]
    Reduce the number of audio channels by mixing or selecting channels, or increase the number of channels by duplicating channels. Note: this effect operates on the audio channels within the SoX effects processing chain; it should not be confused with the -m global option (where multiple files are mix-combined before entering the effects chain). This effect is automatically used when the number of input channels differ from the number of output channels. When reducing the number of channels it is possible to manually specify the mixer effect and use the -l, -r, -f, -b, -1, -2, -3, -4, options to select only the left, right, front, back channel(s) or specific channel for the output instead of averaging the channels. The -l, and -r options will do averaging in quad-channel files so select the exact channel to prevent this. The mixer effect can also be invoked with up to 16 numbers, separated by commas, which specify the proportion (0 = 0% and 1 = 100%) of each input channel that is to be mixed into each output channel. In two-channel mode, 4 numbers are given: l → l, l → r, r → l, and r → r, respectively. In four-channel mode, the first 4 numbers give the proportions for the left-front output channel, as follows: lf → lf, rf → lf, lb → lf, and rb → rf. The next 4 give the right-front output in the same order, then left-back and right-back. It is also possible to use the 16 numbers to expand or reduce the channel count; just specify 0 for unused channels. Finally, certain reduced combination of numbers can be specified for certain input/output channel combinations.
    In Ch | Out Ch | Num | Mappings
    2 | 1 | 2 | l → l, r → l
    2 | 2 | 1 | adjust balance
    4 | 1 | 4 | lf → l, rf → l, lb → l, rb → l
    4 | 2 | 2 | lf → l&rf → r, lb → l&rb → r
    4 | 4 | 1 | adjust balance
    4 | 4 | 2 | front balance, back balance

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

noisered [profile-file [amount]]
    Reduce noise in the audio signal by profiling and filtering. This effect is moderately effective at removing consistent background noise such as hiss or hum. To use it, first run SoX with the noiseprof effect on a section of audio that ideally would contain silence but in fact contains noise - such sections are typically found at the beginning or the end of a recording. noiseprof will write out a noise profile to profile-file, or to stdout if no profile-file or if `-' is given. E.g.
        	sox speech.au -n trim 0 1.5 noiseprof speech.noise-profile
    To actually remove the noise, run SoX again, this time with the noisered effect; noisered will reduce noise according to a noise profile (which was generated by noiseprof), from profile-file, or from stdin if no profile-file or if `-' is given. E.g.
        	sox speech.au cleaned.au noisered speech.noise-profile 0.3
    How much noise should be removed is specified by amount - a number between 0 and 1 with a default of 0.5. Higher numbers will remove more noise but present a greater likelihood of removing wanted components of the audio signal. Before replacing an original recording with a noise-reduced version, experiment with different amount values to find the optimal one for your audio; use headphones to check that you are happy with the results, paying particular attention to quieter sections of the audio. On most systems, the two stages - profiling and reduction - can be combined using a pipe, e.g.
        	sox noisy.au -n trim 0 1 noiseprof | play noisy.au noisered

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

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

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

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

polyphase [-w nut|ham] [-width n] [-cutoff c]
    Change the sampling rate using `polyphase interpolation', a DSP algorithm. This method is relatively slow and memory intensive. If the -w parameter is nut, then a Nuttall (~90 dB stop-band) window will be used; ham selects a Hamming (~43 dB stop-band) window. The default is Nuttall. The -width parameter specifies the (approximate) width of the filter. The default is 1024 samples, which produces reasonable results. The -cutoff value (c) specifies the filter cutoff frequency in terms of fraction of frequency bandwidth, also know as the Nyquist frequency. See the resample effect for further information on Nyquist frequency. If up-sampling, then this is the fraction of the original signal that should go through. If down-sampling, this is the fraction of the signal left after down-sampling. The default is 0.95. See also rabbit and resample for other sample-rate changing effects.

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

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

resample [-qs|-q|-ql] [rolloff [beta]]
    Change the sampling rate using simulated analog filtration. Other rate changing effects available are polyphase and rabbit. There is a detailed analysis of resample and polyphase at http://leute.server.de/wilde/resample.html; see rabbit for a pointer to its own documentation. By default, linear interpolation is used, with a window width about 45 samples at the lower of the two rates. This gives an accuracy of about 16 bits, but insufficient stop-band rejection in the case that you want to have roll-off greater than about 0.8 of the Nyquist frequency. The -q* options will change the default values for roll-off and beta as well as use quadratic interpolation of filter coefficients, resulting in about 24 bits precision. The -qs, -q, or -ql options specify increased accuracy at the cost of lower execution speed. It is optional to specify roll-off and beta parameters when using the -q* options. Following is a table of the reasonable defaults which are built-in to SoX:
    Option | Window | Roll-off | Beta | Interpolation
    (none) | 45 | 0.80 | 16 | linear
    -qs | 45 | 0.80 | 16 | quadratic
    -q | 75 | 0.875 | 16 | quadratic
    -ql | 149 | 0.94 | 16 | quadratic
    -qs, -q, or -ql use window lengths of 45, 75, or 149 samples, respectively, at the lower sample-rate of the two files. This means progressively sharper stop-band rejection, at proportionally slower execution times. rolloff refers to the cut-off frequency of the low pass filter and is given in terms of the Nyquist frequency for the lower sample rate. rolloff therefore should be something between 0 and 1, in practise 0.8-0.95. The defaults are indicated above. The Nyquist frequency is equal to half the sample rate. Logically, this is because the A/D converter needs at least 2 samples to detect 1 cycle at the Nyquist frequency. Frequencies higher then the Nyquist will actually appear as lower frequencies to the A/D converter and is called aliasing. Normally, A/D converts run the signal through a lowpass filter first to avoid these problems. Similar problems will happen in software when reducing the sample rate of an audio file (frequencies above the new Nyquist frequency can be aliased to lower frequencies). Therefore, a good resample effect will remove all frequency information above the new Nyquist frequency. The rolloff refers to how close to the Nyquist frequency this cutoff is, with closer being better. When increasing the sample rate of an audio file you would not expect to have any frequencies exist that are past the original Nyquist frequency. Because of resampling properties, it is common to have aliasing artifacts created above the old Nyquist frequency. In that case the rolloff refers to how close to the original Nyquist frequency to use a highpass filter to remove these artifacts, with closer also being better. The beta parameter determines the type of filter window used. Any value greater than 2 is the beta for a Kaiser window. Beta ≤ 2 selects a Nuttall window. If unspecified, the default is a Kaiser window with beta 16. In the case of Kaiser window (beta > 2), lower betas produce a somewhat faster transition from pass-band to stop-band, at the cost of noticeable artifacts. A beta of 16 is the default, beta less than 10 is not recommended. If you want a sharper cutoff, don't use low beta's, use a longer sample window. A Nuttall window is selected by specifying any `beta' ≤ 2, and the Nuttall window has somewhat steeper cutoff than the default Kaiser window. You will probably not need to use the beta parameter at all, unless you are just curious about comparing the effects of Nuttall vs. Kaiser windows. This is the default effect if the two files have different sampling rates. Default parameters are, as indicated above, Kaiser window of length 45, roll-off 0.80, beta 16, linear interpolation. Note: -qs is only slightly slower, but more accurate for 16-bit or higher precision. Note: In many cases of up-sampling, no interpolation is needed, as exact filter coefficients can be computed in a reasonable amount of space. To be precise, this is done when both input-rate < output-rate, and output-rate ÷ gcd(input-rate, output-rate) ≤ 511.

reverb gain-out reverb-time <delay>
    Add reverberation to the audio. Each delay is given in milliseconds and its feedback is depending on the reverb-time in milliseconds. Each delay should be in the range of half to quarter of reverb-time to get a realistic reverberation. gain-out is the volume of the output.

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

silence [-l] above-periods [duration
    threshold[d|%] [below-periods duration threshold[d|%]] Removes silence from the beginning, middle, or end of the audio. Silence is anything below a specified threshold. The above-periods value is used to indicate if audio should be trimmed at the beginning of the audio. A value of zero indicates no silence should be trimmed from the beginning. When specifying an non-zero above-periods, it trims audio up until it finds non-silence. Normally, when trimming silence from beginning of audio the above-periods will be 1 but it can be increased to higher values to trim all audio up to a specific count of non-silence periods. For example, if you had an audio file with two songs that each contained 2 seconds of silence before the song, you could specify an above-period of 2 to strip out both silence periods and the first song. When above-periods is non-zero, you must also specify a duration and threshold. Duration indications the amount of time that non-silence must be detected before it stops trimming audio. By increasing the duration, burst of noise can be treated as silence and trimmed off. Threshold is used to indicate what sample value you should treat as silence. For digital audio, a value of 0 may be fine but for audio recorded from analog, you may wish to increase the value to account for background noise. When optionally trimming silence from the end of the audio, you specify a below-periods count. In this case, below-period means to remove all audio after silence is detected. Normally, this will be a value 1 of but it can be increased to skip over periods of silence that are wanted. For example, if you have a song with 2 seconds of silence in the middle and 2 second at the end, you could set below-period to a value of 2 to skip over the silence in the middle of the audio. For below-periods, duration specifies a period of silence that must exist before audio is not copied any more. By specifying a higher duration, silence that is wanted can be left in the audio. For example, if you have a song with an expected 1 second of silence in the middle and 2 seconds of silence at the end, a duration of 2 seconds could be used to skip over the middle silence. Unfortunately, you must know the length of the silence at the end of your audio file to trim off silence reliably. A work around is to use the silence effect in combination with the reverse effect. By first reversing the audio, you can use the above-periods to reliably trim all audio from what looks like the front of the file. Then reverse the file again to get back to normal. To remove silence from the middle of a file, specify a below-periods that is negative. This value is then treated as a positive value and is also used to indicate the effect should restart processing as specified by the above-periods, making it suitable for removing periods of silence in the middle of the audio. The option -l indicates that below-periods duration length of audio should be left intact at the beginning of each period of silence. For example, if you want to remove long pauses between words but do not want to remove the pauses completely. The period counts are in units of samples. Duration counts may be in the format of hh:mm:ss.frac, or the exact count of samples. Threshold numbers may be suffixed with d to indicate the value is in decibels, or % to indicate a percentage of maximum value of the sample value (0% specifies pure digital silence).

speed factor[c]
    Adjust the audio speed (pitch and tempo together). factor is either the ratio of the new speed to the old speed: greater than 1 speeds up, less than 1 slows down, or, if appended with the letter `c', the number of cents (i.e. 100ths of a semitone) by which the pitch (and tempo) should be adjusted: greater than 0 increases, less than 0 decreases. By default, the speed change is performed by the resample effect with its default parameters. For higher quality resampling, in addition to the speed effect, specify either the resample or the rabbit effect with appropriate parameters.

stat [-s n] [-rms] [-freq] [-v] [-d]
    Do a statistical check on the input file, and print results on the standard error file. Audio is passed unmodified through the SoX processing chain. The `Volume Adjustment:' field in the statistics gives you the parameter to the -v number which will make the audio as loud as possible without clipping. Note: See the discussion on Clipping above for reasons why it is rarely a good idea to actually do this. The option -v will print out the `Volume Adjustment:' field's value only and return. This could be of use in scripts to auto convert the volume. The -s option is used to scale the input data by a given factor. The default value of n is the max value of a signed long variable (0x7fffffff). Internal effects always work with signed long PCM data and so the value should relate to this fact. The -rms option will convert all output average values to `root mean square' format. The -freq option calculates the input's power spectrum and prints it to standard error. There is also an optional parameter -d that will print out a hex dump of the audio from the internal buffer that is in 32-bit signed PCM data. This is mainly only of use in tracking down endian problems that creep in to SoX on cross-platform versions.

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

synth [len] {[type] [combine] [freq[-freq2]] [off] [ph] [p1] [p2] [p3]}
    This effect can be used to generate fixed or swept frequency audio tones with various wave shapes, or to generate wide-band noise of various `colours'. Multiple synth effects can be cascaded to produce more complex waveforms; at each stage it is possible to choose whether the generated waveform will be mixed with, or modulated onto the output from the previous stage. Audio for each channel in a multi-channel audio file can be synthesised independently. Though this effect is used to generate audio, an input file must still be given, the characteristics of which will be used to set the synthesised audio length, the number of channels, and the sampling rate; however, since the input file's audio is not normally needed, a `null file' (with the special name -n) is often given instead (and the length specified as a parameter to synth or by another given effect that can has an associated length). For example, the following produces a 3 second, 44.1 kHz, audio file containing a sine-wave swept from 300 to 3300 Hz:
        	sox -n output.au synth 3 sine 300-3300
    and this produces an 8 kHz version:
        	sox -r 8000 -n output.au synth 3 sine 300-3300
    Multiple channels can be synthesised by specifying the set of parameters shown between braces multiple times; the following puts the swept tone in the left channel and adds `brown' noise in the right:
        	sox -n output.au synth 3 sine 300-3300 brownnoise
    The following example shows how two synth effects can be cascaded to create a more complex waveform:
        	sox -n output.au synth 0.5 sine 200-500 \
        		synth 0.5 sine fmod 700-100
    Frequencies can also be given as a number of musical semitones relative to `middle A' (440 Hz) by prefixing a `%' character; for example, the following could be used to help tune a guitar's `E' strings:
        	play -n synth sine %-17
    N.B. This effect generates audio at maximum volume, which means that there is a high chance of clipping when using the audio subsequently, so in most cases, you will want to follow this effect with the vol effect to prevent this from happening. (See also Clipping above.) A detailed description of each synth parameter follows: len is the length of audio to synthesise expressed as a time or as a number of samples; 0=inputlength, default=0. The format for specifying lengths in time is hh:mm:ss.frac. The format for specifying sample counts is the number of samples with the letter `s' appended to it. type is one of sine, square, triangle, sawtooth, trapezium, exp, [white]noise, pinknoise, brownnoise; default=sine combine is one of create, mix, amod (amplitude modulation), fmod (frequency modulation); default=create freq/freq2 are the frequencies at the beginning/end of synthesis in Hz or, if preceded with `%', semitones relative to A (440 Hz); for both, default=%0. If freq2 is given, then len must also have been given. Not used for noise. off is the bias (DC-offset) of the signal in percent; default=0. ph is the phase shift in percentage of 1 cycle; default=0. Not used for noise. p1 is the percentage of each cycle that is `on' (square), or `rising' (triangle, exp, trapezium); default=50 (square, triangle, exp), default=10 (trapezium). p2 (trapezium): the percentage through each cycle at which `falling' begins; default=50. exp: the amplitude in percent; default=100. p3 (trapezium): the percentage through each cycle at which `falling' ends; default=60.

tempo [-q] factor [segment [search [overlap]]]
    Change the audio tempo (but not its pitch) using a `WSOLA' algorithm. The audio is chopped up into segments which are then shifted in the time domain and overlapped (cross-faded) at points where their waveforms are most similar (as determined by measurement of `least squares'). By default, linear searches are used to find the best overlapping points; if the optional -q parameter is given, tree searches are used instead, giving a quicker, but possibly lower quality, result. factor gives the ratio of new tempo to the old tempo. The optional segment parameter selects the algorithm's segment size in milliseconds. The default value is 82 and is typically suited to making small changes to the tempo of music; for larger changes (e.g. a factor of 2), 50 ms may give a better result. When changing the tempo of speech, a segment size of around 30 ms often works well. The optional search parameter gives the audio length in milliseconds (default 14) over which the algorithm will search for overlapping points. Larger values use more processing time and do not necessarily produce better results. The optional overlap parameter gives the segment overlap length in milliseconds (default 12). See also stretch for a similar effect.

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

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

trim start [length]
    Trim can trim off unwanted audio from the beginning and end of the audio. Audio is not sent to the output stream until the start location is reached. The optional length parameter tells the number of samples to output after the start sample and is used to trim off the back side of the audio. Using a value of 0 for the start parameter will allow trimming off the back side only. Both options can be specified using either an amount of time or an exact count of samples. The format for specifying lengths in time is hh:mm:ss.frac. A start value of 1:30.5 will not start until 1 minute, thirty and ½ seconds into the audio. The format for specifying sample counts is the number of samples with the letter `s' appended to it. A value of 8000s will wait until 8000 samples are read before starting to process audio.

vol gain [type [limitergain]]
    Apply an amplification or an attenuation to the audio signal. Unlike the -v option (which is used for balancing multiple input files as they enter the SoX effects processing chain), vol is an effect like any other so can be applied anywhere, and several times if necessary, during the processing chain. The amount to change the volume is given by gain which is interpreted, according to the given type, as follows: if type is amplitude (or is omitted), then gain is an amplitude (i.e. voltage or linear) ratio, if power, then a power (i.e. wattage or voltage-squared) ratio, and if dB, then a power change in dB. When type is amplitude or power, a gain of 1 leaves the volume unchanged, less than 1 decreases it, and greater than 1 increases it; a negative gain inverts the audio signal in addition to adjusting its volume. When type is dB, a gain of 0 leaves the volume unchanged, less than 0 decreases it, and greater than 0 increases it. See [4] for a detailed discussion on electrical (and hence audio signal) voltage and power ratios. Beware of Clipping when the increasing the volume. The gain and the type parameters can be concatenated if desired, e.g. vol 10dB. An optional limitergain value can be specified and should be a value much less than 1 (e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping. Not specifying this parameter will cause no limiter to be used. In verbose mode, this effect will display the percentage of the audio that needed to be limited. See also compand for a dynamic-range compression/expansion/limiting effect.

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

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

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

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

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

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

pitch shift [width interpolate fade]
    Change the audio pitch (but not its duration). This effect is equivalent to the key effect with search set to zero, so its results are comparitively poor; it is retained for backwards compatibility only. Change by cross-fading shifted samples. shift is given in cents. Use a positive value to shift to treble, negative value to shift to bass. Default shift is 0. width of window is in ms. Default width is 20ms. Try 30ms to lower pitch, and 10ms to raise pitch. interpolate option, can be cubic or linear. Default is cubic. The fade option, can be cos, hamming, linear or trapezoid; the default is cos.

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

stretch factor [window fade shift fading]
    Change the audio duration (but not its pitch). This effect is equivalent to the tempo effect with (factor inverted and) search set to zero, so its results are comparitively poor; it is retained for backwards compatibility only. factor of stretching: >1 lengthen, <1 shorten duration. window size is in ms. Default is 20ms. The fade option, can be `lin'. shift ratio, in [0 1]. Default depends on stretch factor. 1 to shorten, 0.8 to lengthen. The fading ratio, in [0 0.5]. The amount of a fade's default depends on factor and shift.

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

SEE ALSO
sox(1), soxformat(7), libsox(3), soxexam(7), wget(1) The SoX web page at http://sox.sourceforge.net

  References

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

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

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

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

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

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

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

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