SOXEFFECT_NG(7)

NAME
soxeffect_ng - Effects supported by sox_ng and libsox_ng

SYNOPSIS
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 the other at the end of the SoX command line, forming an `effects chain'. Note that applying multiple effects in real time (i.e. when playing audio) may require a high performance computer. Some of the SoX effects are primarily intended to be applied to a single instrument or `voice'. To facilitate this, the remix effect and the global SoX option -M can be used to isolate then recombine tracks from a multitrack recording.

MULTIPLE EFFECTS CHAINS
A single 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 terminates the chain. SoX supports running multiple effects chains over the input audio. In this case, when one chain indicates that it is done processing audio, the audio data is 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. Effects chains can be separated by placing a : (colon) after an effect; any following effects are a part of a new effects chain. It is important to place the effect that stops the chain as the first effect in the chain 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 the sample rate, if the terminating effect cannot be first. Further information on stopping effects can be found in the Stopping SoX section. There are a few pseudo-effects that can help when using multiple effects chains. These include newfile, which starts writing to a new output file before moving to the next effects chain, and restart, which moves 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 (i.e. they must have a : before and after them). Here is an example of multiple effects chains. It splits the input file into multiple files, each of 30 seconds in length and each output filename will have unique number in its name, as documented in the Output Files section.
   sox_ng in.au out.au trim 0 30 : newfile : restart

COMMON NOTATION AND PARAMETERS
In the descriptions that follow, [square brackets] are used to denote parameters that are optional, {braces} to denote those that are both optional and repeatable, <angle brackets> to denote those that are repeatable but not optional and pipe characters `|' separate options from which to choose one of several alternatives. Where applicable, default values for optional parameters are shown (in parentheses). The following parameters are used with, and have the same meaning for, several effects:

frequency
    A frequency in Hz or, if followed by k, in kHz or, if preceded by %, in semitones relative to A (440Hz); alternatively, scientific note names (e.g. E2) may be used.

gain
    A power gain in dB. Zero gives no gain, less than zero gives an attenuation and greater than zero amplifies.

duration
    See Time Specifications below.

position
    A position within the audio stream; the syntax is [=|-|+]timespec, where timespec is a time specification (see below). The optional first character indicates whether the timespec is to be interpreted relative to the start (=) or end (-) of the audio or relative to the previous position (+) if the effect accepts multiple positional arguments. The audio length must be known for end-relative locations to work, though some effects do accept -0 for the end of the audio even if the length is unknown. Which of =, - and + is the default depends on the effect and is shown in its syntax as, e.g., position(+). Examples: `=2:00' is two minutes into the audio stream, `-100s' is one hundred samples before the end of the audio, `+0:12+10s' is twelve seconds and ten samples after the previous position and `-0.5+1s' is one sample less than half a second before the end of the audio.

width[h|k|o|q]
    Used to specify the bandwidth 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 |
    b | Hz | Old non-frequency-warped response
    o | octaves |
    q | Q-factor | See [2]
    s | slope |
    For each effect that uses this parameter, the default method (if no character is appended) is the one that is listed first in the first line of the effect's description.

TIME SPECIFICATIONS
A timespec can be given in one the following two forms:

[[hours:]minutes:]seconds[.frac][t]
    For example, a time specification of `1:30.5' corresponds to one minute, thirty and ½ seconds. The component values do not have to be normalized; e.g. `1:23:45', `83:45', `79:0285', `1:0:1425', `1::1425' and `5025' are all equivalent.

sampless
    Specifies the number of samples directly, as in `8000s'. For large sample counts, e notation is supported: `1.7e6s' is the same as `1700000s'.

Time specifications can also be chained with + or - into a new time specification where the right part is added to or subtracted from the total so far. For example, `3:00-200s' means two hundred samples less than three minutes.

If a time specification is a plain whole number with no t or s suffix, whether it is taken as a number of seconds or a number of samples depends on the effect in question. At present, it always means seconds except for the duration parameters of the silence effect.

SUPPORTED EFFECTS
To see whether SoX has support for an optional effect, enter sox_ng -h and look for its name in the EFFECTS list; a categorized list of the effects can be found in the accompanying README file.

allpass [-1|2] frequency [width[h|k|o|q]]
    Apply a two-pole all-pass filter with central frequency frequency and filter width width. An all-pass filter changes the audio's frequency to phase relationship without changing its frequency to amplitude relationship. The filter is described in detail in [1]. -1 or -2 use an experimental 1-pole or 2-pole filter, in which case width does not apply. This effect supports the --plot global option.

band [-n] frequency [width[h|k|o|q]]
    Apply a band-pass filter. The frequency response drops logarithmically around the center frequency. The width parameter gives the slope of the drop: the frequencies at frequency + width and frequency - width will have half their original amplitudes. Its default value is half of the center frequency. 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 unpitched audio (e.g. percussion), though -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, peaking at the center frequency and settling around it. This effect supports the --plot global option. See sinc for a band-pass filter with steeper shoulders.

bandpass|bandreject [-c] frequency width[h|k|o|q|b]
    Apply a two-pole Butterworth band-pass or band-reject filter with central frequency frequency, and (3dB-point) bandwidth width. The -c option applies only to bandpass and selects a constant skirt gain (peak gain = Q) instead of the default, a 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 sinc for a band-pass filter with steeper shoulders.

bass|treble gain [frequency [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 hifi's tone controls. This is also known as shelving equalization. gain gives the gain at 0Hz for bass or, for treble, whichever is the lower of ~22kHz and the Nyquist frequency. Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of Clipping when using a positive gain. 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 values are 100Hz for bass and 3kHz for treble. width determines how steep the filter's shelf transition is. In addition to the common width specification methods, `slope' (the default) may be used. Its useful range is about 0.3 for a gentle slope to 1 (the maximum) for a steep slope; and its default value is 0.5. The filters are described in detail in [1]. These effects support the --plot global option. See equalizer for a peaking equalization effect.

bend [-f frame-rate(25)] [-o oversampling(16)]
    {start-position(+),cents,end-position(+)} Changes the pitch by specified amounts at specified times without changing the duration. Each given triple: 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. During each bend, the frequency changes logarithmically, i.e. by the same number of cents per second. The pitch bending algorithm uses the Discrete Fourier Transform (DFT) at a particular frame rate and oversampling rate. The -f (from 10 to 80) and -o (from 4 to 32) parameters may be used to adjust these parameters and thus control the smoothness of the changes in pitch. For example, an initial tone is generated, then bent three times, yielding four different notes in total:
       play_ng -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 seconds 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 pitch.

biquad b0 b1 b2 a0 a1 a2
    Apply a biquad Infinite Impulse Response filter with the given coefficients, 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.

centercut [-a gain] [-b] [-w size]
    Remove the center from a stereo file leaving the far left and right parts of the stereo file intact and the center in a third channel. The -a option is gain-out for all channels, default 1.0.. The -b option moves the bass (below 200Hz) out of the center into the sides for those who want karaoke. The -w option changes the window size from its default of 8192 sample frames to a power of two from 4 to 32768. A stereo equivalent of the oops effect is
        sox_ng in.wav centercut remix 1 2 The keymap for the -a option is centercut.gain. See also oops.

channels channels
    Invoke a simple algorithm to change the number of channels in the audio signal to the given number: mixing if decreasing the number of channels or duplicating if increasing the number of channels. The channels effect is invoked automatically if SoX's -c option specifies a number of channels that is different to that of the input file(s). Alternatively, if this effect is given explicitly, SoX's -c option need not be given. For example, the following two commands are equivalent:
       sox_ng input.wav -c 1 output.wav bass -b 24
       sox_ng 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. For example, when making a stereo file quadraphonic, the left and right channels are copied into the third and fourth and when mixing a four-channel file down to stereo, the left channel is the mix of the first and third and the right of the second and fourth. See remix for an effect that allows channels to be mixed and selected arbitrarily.

chorus [-n|l|q] [-s|t] [gain-in [gain-out {delay [decay [speed [depth [-s|t]]]]}]]
    Add a chorus effect to the audio. This can make a single voice sound like a chorus but can also be applied to instrumentation. Chorus resembles an echo effect with a short delay but, while echo's delay is constant, chorus' delay varies by a sinusoidal or triangular modulation. See [3] for further discussion of the chorus effect. The -l flag makes chorus do linear interpolation between samples when the offset into the delay line is not a whole number, which is about 15% slower but makes it considerably less noisy and -q asks for quadratic interpolation which is about 40% slower but makes it even less noisy. -n explicitly asks for no interpolation, the default, fast and fuzzy. -s or -t before the stages change the default wave type for all of them. All parameters are all optional and, if missing, assume the following values:
    Parameter | Range | Default | Description
    gain-in | -1-1 | 0.5 | Proportion of input delivered clean to the adder
    gain-out | -1-1 | 1 | Final volume adjustment
    delay | 0-1000 | 40-60 | Fixed delay in milliseconds
    decay | -1-1 | 0.5 | Volume of delayed output
    speed | 0-192k | 0.25 | Modulation frequency
    depth | 0-1000 | 2 | Extra delay in milliseconds
    wave | -s|-t | -s | Sinusoidal/triangular modulation
    There are keymaps for gain_in and gain_out. Each delay ranges from the fixed delay to delay + depth. Gain-out is then applied to the sum of the input scaled by gain-in and the outputs from the delays scaled by their decays. For internal reasons regarding the speed of chorus, there is a limit of 256 chorus stages. A typical delay is around 40ms to 60ms; the modulation speed is best near 0.25Hz and the modulation depth around 2ms. For example, a single delay:
       play_ng guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t Two delays of the original samples:
       play_ng guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t \
         60 0.32 0.4 1.3 -s A fuller-sounding chorus (with three additional delays):
       play_ng 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 flanger can do everything that chorus does except multiple stages but works in floating point internally instead of integers so is slower without a floating point processor:
       chorus -l gain-in gain-out delay decay speed depth -wave is equivalent to
       flanger delay depth 0 100×decay÷gain-in speed wave 0 \
          vol gain-out÷(gain-in+decay) For a flow diagram of how chorus works, say sox_ng --help-effect chorus.

compand attack1,decay1{,attack,decay}
    [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB,out-dB}
    [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. For most situations, the attack time (its 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. When more than one pair of attack/decay parameters is 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 are rounded by the amount given. Typical values for the transfer function are `6:-70,-60,-20'. The third (optional) parameter is an additional gain in dB to be applied at all points on the transfer function and allows easy adjustment of the overall gain. The fourth (optional) parameter is an initial level to be assumed for each channel when companding starts. This lets you 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 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 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 operate in a predictive rather than a reactive mode. A typical value is 0.2 seconds.
    * | * | *
    The following example might be used to make a piece of music with both quiet and loud passages suitable for listening to in a noisy environment such as a moving vehicle:
       sox_ng 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) remain unchanged. This stops the compander from boosting the volume on `silent' passages such as between movements. However, sounds in the range -60dB to 0dB (maximum volume) are boosted so that the 60dB dynamic range of the original music is compressed 3-to-1 into a 20dB range, which is wide enough to enjoy the music but narrow enough to get around the road noise. The `6:' selects 6dB soft-knee companding. The -5 dB output gain is needed to avoid clipping (the number is inexact and was derived by experimentation). The -90 dB for the initial volume will work fine for a clip that starts with near silence and the delay of 0.2 seconds makes the compander react more quickly to sudden volume changes. In the next example, compand is used as a noise-gate for when the noise is at a lower level than the signal:
       play_ng in.au 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 than the signal (making it, in some ways, similar to a squelch effect):
       play_ng in.au compand .1,.1 -45.1,-45,-inf,0,-inf 45 -90 .1 This effect supports the --plot global option (for the transfer function). For a flow diagram of how compand works, say sox_ng --help-effect compand. See mcompand for a multiple-band companding effect.

contrast [amount(75)]
    Comparable with compression, this effect modifies an audio signal to make it sound louder. amount controls the amount of the enhancement and is a number in the range 0-100. Note that amount = 0 still gives a significant contrast enhancement. There is a keymap for amount. See the compand and mcompand effects.

dcshift shift [limiter-gain]
    Apply a DC shift to the audio. This can be useful to remove a known DC offset (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 that indicates the amount to shift the audio (which is in the range of ±1). An optional limiter-gain can be specified as well. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping. An alternative approach to removing a DC offset (albeit with a short delay) is to use the highpass filter effect at a frequency of say 10Hz, as illustrated in the following example:
       sox_ng -n dc.wav synth 5 sin %0 50
       sox_ng dc.wav fixed.wav highpass 10

deemph
    Apply Compact Disc (IEC 60908) de-emphasis with a treble attenuation shelving filter. Pre-emphasis was applied in the mastering of some CDs issued in the early 1980s. These included many classical music albums, as well as now sought-after issues of albums by The Beatles, Pink Floyd and others. Pre-emphasis should be removed at playback time by a de-emphasis filter in the playback device. However, not all modern CD players have this filter and very few PC CD drives have it; playing pre-emphasized 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-emphasized CD and then either burn the de-emphasized audio to a new CD (which will then play correctly on any CD player) or simply play the correctly de-emphasized audio files on the PC. For example:
       sox_ng track1.wav track1-deemph.wav deemph and then burn track1-deemph.wav to CD, or
       play_ng track1-deemph.wav or simply
       play_ng 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. Its maximum deviation from the ideal response is only 0.06dB (up to 20kHz). This effect supports the --plot global option.

delay {position(=)}
    Delay zero or more audio channels such that they start at the given position. For example, delay 1.5 +1 3000s delays the first channel by 1.5 seconds, the second channel by 2.5 seconds (one second more than the previous channel), the third channel by 3000 samples and leaves other channels undelayed. The following (one long) command plays a chime sound:
       play_ng -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 an arpeggiated guitar chord:
       play_ng -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 With no parameters it does nothing. To delay all channels by the same amount, use the pad effect.

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 adds TPDF white noise. The -S option selects a slightly `sloped' TPDF, biased towards higher frequencies. It can be used at any sampling rate but, below ≈22kHz, plain TPDF is probably better and, above ≈ 37kHz, noise-shaping (if available) is probably better. The -s option enables noise-shaping with the shibata filter (the same as -f shibata) and with the -f option it is possible to select a particular noise-shaping filter from the following list: lipshitz, f-weighted, modified-e-weighted, improved-e-weighted, gesemann, shibata, low-shibata, high-shibata, shibata-(A|B)(0|1|2|3|4|5|6) and shibata-A-saturated. The latter shibata- ones use the new shaper coefficients from Naoki Shibata's SSRC package, described at https://shibatch.org/ssrc The filter types are distinguished by the following properties: audibility of noise, level of (inaudible, but in some circumstances problematic) shaped high frequency noise and processing speed and they are available for the following sample rates:
    Filter | Sample rates
    lipshitz | 44100
    e- and f-weighted | 48000
    gesemann | 44100, 48000
    shibata | 8000, 11025, 16000, 22050
     | 32000, 37800, 44100, 48000
    low-shibata | 44100, 48000
    high-shibata | 44100
    shibaba-A0 and A1 | 8000, 11025, 22050, 44100,
     | 48000, 88200, 96000, 192000
    shibata-A2 | 44100, 48000, 88200, 96000, 192000
    shibata-A3 to A6 | 44100, 48000
    shibata-B0 to B6 | 44100, 48000
    shibata-A-saturated | 8000, 11025, 22050
    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 foolproof, so the fades should be checked carefully for any noise modulation; if this occurs, then either redither the whole file or use trim and fade and concatenate the results. The -p option overrides the target precision in bits and can be from 1 to 24. If the SoX global option -R option is not given, the pseudo-random number generator used to generate the white noise is reseeded, i.e. the generated noise will be different on every invocation. If the target precision is 1-bit, the sdm effect is applied automatically with default settings. Invoke it manually to control its options. See the above section on Dithering.

dolbyb [-e|d] [-u upsamp] [-h] [-t gain(1.0)] [-a prec(-5.0)] [-f {1|2|3|4}]
    dolbyb is a Dolby B decoder/encoder based on dolbybcsoftwaredecode which simulates the operation of a Dolby B en/decoder's electronic circuit. By default, dolbyb applies Dolby B decoding to its input signal; with -e it does Dolby B encoding. -d is also accepted but only for symmetry, as it is the default mode of operation. -u sets the upsampling ratio to use in the sliding filter. Digital filtering only works well if the sample rate is well above the cutoff frequency of the filter. For Dolby B's sliding filter, that frequency can be as high as 34kHz and this does not work well if the sample rate is only 44.1Khz. To get around this, it upsamples the audio to a higher rate when it passes through this filter. By default, -u0, the upsampling rate is set automatically so that the upper sample rate is at least 200Khz; upsampling can be switched off with -u1. If -h is given, upsampling is used throughout the effect from when the audio enters to when it leaves, not just in the sliding filter. As dolbyb's up/downsampling algorithm is simple (repeating and averaging samples) you may obtain higher quality results by upsampling with rate before dolbyb -u1 and downsampling it afterwards. -t ("threshold") adjusts the gain when the audio is fed to the Dolby gain control circuits. When a tape deck is encoding or decoding a magnetic tape, it knows the signal level at the tape heads but with audio files the maximum signal level may not accurately represent the tape's maximum flux density (200nWb/m for cassette tapes), giving erroneous results. The -t option adjusts the volume level at which the sliding filter reacts to overcome this. Its default value is 1.0, which assumes that the maximum amplitude of the signal represents the maximum recording level on tape; higher values assume that it was recorded too quietly and values below 1.0 are for when it was recorded too loud. To begin with, when you have little idea of what level to use, try a wide range of levels like 5, 10, 15 and 20. If the result sounds muffled, the threshold is too low and if it seems to have too much treble, the threshold is too high. Once you know the approximate level, you can try more closely-spaced levels and listen carefully to find the best level possible. Logic would suggest listening to where tracks fade out, to see if the treble increases, but this method doesn't seem to work well and the best way seems to be to see how low the level can be set before the results sound dull and muffled, then choose a level a bit higher than this; you can just about hear the difference between results that differ in threshold setting by about 2. In decode mode, the program has to use trial and error to get the right output sample values. -a sets how accurate it needs to be before it is considered OK. A figure of 0.0 dB would mean an accuracy of about 1 sample value. The default is -5.0 dB, which is accurate to less than one sample value. The keymap for gain is dolbyb.gain. For example:
       play_ng -V -k D:dolbyb.gain+2 -k d:dolbyb.gain-2 in.wav lets you adjust the Threshold Gain in +/- 2dB steps; to see what the new value is as you proceed, use `-V`. -f selects one of four types of filter to use. The program originally simulated an analog circuit for a Dolby B noise reducer. However, too much filtering in the side path was altering the phase of the side path audio, which caused problems when the side path was recombined with the main signal. Basically signals don't add together very well if there is too much difference in the phase. To fix this, there are now 4 filter modes with hopefully less of a phase change:

    -f1 is the original method.

    -f2 is a newer method that seems to work better than 1.

    -f3 is another rearrangement which in practice doesn't seem to be any better than 1.

    -f4 seems to work best, hence it is the default mode.

    For further detail on these parameters and advice on digitizing and processing Dolby B-encoded tapes, consult the wiki pages at https://codeberg.org/sox_ng/libdolbyb

dop DSD over PCM. 1-bit DSD data is packed into 24-bit samples for transport over non-DSD-aware links.

downsample [factor(2)]
    Downsample the signal by an integer factor: Only the first of each factor samples is retained, the others are discarded. No decimation filter is applied. If the input is not a properly band-limited baseband signal, aliasing will occur. This may be desirable, e.g., for frequency translation. The new lower sample rate propagates forward in the effects chain but, unless you specify the new sample rate with -r before the output filename or with a final (no-op) rate effect, it will be resampled back up to the original sample rate. For a general resampling effect with antialiasing, see rate. See upsample.

earwax
    This effect takes a 44.1kHz stereo signal and adds audio cues that, when listened to on headphones, move the sound stage from inside your head to outside and in front of you, as if listening to loudspeakers. To see how earwax works, say sox_ng --help-effect earwax.

echo gain-in gain-out <delay decay>
    Add echoes to the audio. In nature, echoes are reflected sound and digital echo effects emulate this and are often used to help fill out the sound of a single instrument or vocal. Gain-in controls how much of the input signal is delivered clean to the output, delay is the time difference in milliseconds between the original signal and its reflection, decay is the loudness of the reflected signal and gain-out is a final volume adjustment of the result. There are keymaps for gain_in and gain_out. There is no limit to the number of delay/decay pairs you can use and gains and decays can be negative or greater than 1 if you wish. echo extends the length of the signal by the maximum delay time. For example, this makes it sound as if there are twice as many instruments as are actually playing:
       play_ng lead.aiff echo 0.8 0.88 60 0.4 If the delay is very short, it sound like a metallic robot: music:
       play_ng lead.aiff echo 0.8 0.88 6 0.4 A longer delay sounds like an open air concert in the mountains:
       play_ng lead.aiff echo 0.8 0.9 1000 0.3 One mountain more, and:
       play_ng lead.aiff echo 0.8 0.9 1000 0.3 1800 0.25 For a flow diagram of how echo works, say sox_ng --help-effect echo.

echos gain-in gain-out <delay decay>
    Echos stands for `Echo in Sequel' and adds a sequence of echoes to the audio. That is, the first echo takes the input, the second the input and the first echo, the third the input and the output of the second echo and so on. A single echos has the same effect as a single echo. Each delay decay pair gives the delay in milliseconds (with a minimum of one sample) and the decay of that echo. Gain-out is a final volume multiplier applied to the sum of the input × gain-in and the delays' outputs × their respective decays, and there are keymaps for both. There are keymaps for gain_in and gain_out. echos extends the length of the signal by the maximum delay time. For example: The sample is bounced twice in symmetric echos:
       play_ng lead.aiff echos 0.8 0.7 700 0.25 700 0.3 The sample is bounced twice in asymmetric echos:
       play_ng lead.aiff echos 0.8 0.7 700 0.25 900 0.3 The sample sounds as if it were played in a garage:
       play_ng lead.aiff echos 0.8 0.7 40 0.25 63 0.3 For a flow diagram of how echos works, say sox_ng --help-effect echos.

equalizer frequency width[q|o|h|k] gain
    Apply a two-pole peaking equalization filter. With this filter, the signal level at and around a selected frequency can be increased or decreased while, unlike band-pass and band-reject filters, the level at all other frequencies is unchanged. frequency gives the filter's central frequency in Hz, width gives its bandwidth and gain the required gain or attenuation in dB. Beware of Clipping when using a positive gain. In order to produce complex equalization 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.

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 wave, t for linear (`triangular') slope, l for logarithmic, p for inverted parabola and s for square-law. The default is logarithmic. A fade-in starts from the first sample and ramps the signal level from 0 to full volume over the time given as fade-in-length. Specify 0 if no fade-in is wanted. For a fade-out, the audio is truncated at stop-position and the signal level is ramped from full volume down to 0 over an interval of fade-out-length before the stop-position. If fade-out-length is not specified, it defaults to the same value as fade-in-length. No fade-out is performed if stop-position is not specified. If the audio length can be determined from the input file header and any previous effects, then `-0' (or, for historical reasons, `0') may be specified for stop-position to indicate the usual case of a fade out that ends at the end of the input audio stream. See the splice effect.

fir [coefs-file|coef <coef>]
    Use SoX's FFT convolution engine with given Finite Impulse Response filter coefficients. If a single argument is given, it is the name of a file containing the filter coefficients (white space separated; may contain `#' comments). If the filename is `-' or if no argument is given, the coefficients are read from the `standard input' (stdin); otherwise, coefficients may be given on the command line. Examples:
       sox_ng in.au out.au fir .0195 -.082 .234 .891 -.145 .043
       sox_ng in.au out.au fir coefs.txt with coefs.txt containing
       # HP filter: freq=10000
         1.2311233052619888e-01
        -4.4777096106211783e-01
         5.1031563346705155e-01
        -6.6502926320995331e-02 This effect supports the --plot global option.

firfit [knots-file|<freq gain>]
    Use SoX's FFT convolution engine to make a filter whose frequency response approximates a spline passing through a series of frequency/gain pairs. If a single argument is given, it is the name of a file containing the knots (white space separated; may contain `#' comments). If the given filename is `-' or if no argument is given, the knots are read from the `standard input' (stdin); otherwise, knots may be given on the command line. Gains are in dB and the knot frequencies must be in increasing order. Examples:
       sox_ng in.au out.au firfit 20 0 10000 -3 gives a gentle low-pass filter and
       sox_ng in.au out.au firfit knots.txt with knots.txt containing
       # Approximate telephone response
       300 -100
       400 -10
       480 0
       2800 0
       3000 -10
       3400 -100 approximates the response of a carbon microphone telephone. This effect supports the --plot global option.

flanger [-n|l|q] [-s|t] [delay(0) [depth(2) [regen(0) [width(71) [speed(0.5) [shape(sine)] [phase(25) [interp(linear)]]]]]]]
    Apply a flanging effect to the audio. See [3] for a detailed description of flanging. The parameters give the base delay and the added swept delay in milliseconds, the percentage of regeneration (the delayed signal feedback), width the percentage of delayed signal that is mixed with the original, speed the number of sweeps per second, the shape of the swept wave (sine or triangle), the percentage of phase shift of the swept wave in multichannel flanges (0 = 100 = the same phase on each channel) and the type of digital delay line interpolation (none, linear or quadratic). sine, triangle, none, linear and quadratic can be abbreviated. The input and the delay's output are mixed and balanced so they don't clip, so a width of 100 gives 50:50 mixing; to obtain only the delayed output and none of the input, specify width as inf. Despite containing a delay, flanger does not extend the length of the signal so, if you also want the last dregs of the delayed output and feedback, pad the signal beforehand. sine, triangle, none, linear and quadratic can be abbreviated and -s, -t, -n, -l and -q are alternative ways to set the waveshape and the interpolation type without having to specify the rest of the parameters. For a flow diagram of how flanger works, say sox_ng --help-effect flanger.

gain [-e|B|b|r] [-n] [-l|h] [gain-dB(0)]
    Apply amplification or attenuation to the audio signal or, in some cases, to some of its channels. Note that use of any of -e, -B, -b, -r and -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 adjusts the signal power level by the given number of dB: positive amplifies (beware of clipping), negative attenuates. With other options, the gain-dB amplification or attenuation is applied after the processing due to those options. With the -e option, the levels of the audio channels of a multichannel file are equalized, i.e. gain is applied to all channels other than that with the highest peak level so that all channels attain the same peak level (but, without also giving -n, the audio is not normalized). The -B (balance) option is similar to -e, but with -B, the RMS level is used instead of the peak level. -B might be used to correct stereo imbalance caused by an imperfect record turntable cartridge. Note that, unlike -e, -B might cause some clipping. -b is similar to -B but has clipping protection, i.e. if necessary to prevent clipping whilst balancing, attenuation is applied to all channels. 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 for details. The -n option normalizes the audio to 0dB FSD. It is often used in conjunction with a negative gain-dB so that the audio is normalized to a given level below 0dB. For example,
       sox_ng in.au out.au gain -n normalizes to 0dB, and
       sox_ng in.au out.au gain -n -3 normalizes to -3dB. The -l option invokes a simple limiter. For example,
       sox_ng in.au out.au gain -l 6 applies 6dB of gain but never clips. Note that limiting more than a few dBs more than occasionally 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 headroom for subsequent processing. For example, with
       sox_ng in.au out.au gain -h bass +6 6dB of attenuation is applied prior to the bass boosting effect, ensuring that it does not clip. Of course, with bass, it is obvious how much headroom is needed but, with other effects (e.g. rate, dither), it is not always as clear. Another 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, for example:
       sox_ng in.au out.au gain -h bass +6 rate 44100 gain -r The above effects chain guarantees never to clip 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_ng in.au out.au 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 automatically invoke gain -h and gain -r. Note that synth without the -n option incorporates the functionality of gain -h. See the norm and vol effects.

highpass [-1|2] frequency [width[q|o|h|k]]
    Apply a high-pass filter with 3dB point frequency. The filter can be either single-pole (with -1), or double-pole (the default, or with -2). width applies only to double-pole filters; the default is Q = 0.707 and gives a Butterworth response. The filters roll off at 6dB per pole per octave (20dB per pole per decade). The double-pole filters are described in detail in [1]. This effect supports the --plot global option. See sinc for filters with a steeper roll-off.

hilbert [-n taps]
    Apply an odd-tap Hilbert transform filter, phase shifting the signal by 90 degrees. This is used in many matrix 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 band-pass characteristic, attenuating the lowest and highest frequencies. Its bandwidth can be controlled by the number of filter taps which, by default, is chosen for a cutoff frequency of about 75 Hz. For a cutoff frequency of about N Hz, give the -n option with the sample rate divided by N. The number of taps can be from 3 to 1,073,741,823 but the maximum value requires 56GB of physical RAM to complete within minutes rather than days and 100,000,001 requires 18GB. This effect supports the --plot global option.

ladspa [-l] [-r] 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. Normally, the 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 multichannel input. Some plugins introduce latency which SoX may optionally compensate for. The -l (latency compensation) option automatically compensates for latency as reported by the plugin via an output control port named "latency". If it is set, the environment variable LADSPA_PATH is used as the search path for plugins. See LADSPA_PATH in the section ENVIRONMENT.

loudness [gain [reference]]
    Loudness control is similar to the gain effect but provides equalization 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) and the signal equalized according to ISO 226 w.r.t. a reference level of 65dB, though an alternative reference level may be given if the original audio has been equalized at some other level. A default gain of -10dB is used if a gain value is not given. See the gain effect.

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

mcompand "compand-args" {frequency "compand-args"}
    The quoted compand-args are as for the compand effect:
    attack1,decay1{,attack,decay}
    [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB,out-dB}
    [gain [initial-volume-dB [delay]]] The multi-band compander is similar to the single-band compander but the audio is first divided into bands using Linkwitz-Riley crossover filters and a separately specifiable compander is 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 crossover-freq; these can be repeated to create multiple bands. The following examples approximate Dolby A compression and decompression, as used for tape noise reduction in professional recording studios:
       # Dolby A compressor
       sox_ng in.au dolbyA.au mcompand \
          ".1,.1 4:-56,-46,-36,-26,-26,-20,-17,-15,-9,-9" 80 \
          ".1,.1 4:-56,-46,-36,-26,-26,-20,-17,-15,-9,-9" 3k \
          ".1,.1 4:-56,-46,-36,-26,-26,-20,-17,-15,-9,-9" 9k \
          ".1,.1 4:-56,-42,-36,-23,-26,-18,-17,-14,-9,-9"
       # Dolby A decompressor
       sox_ng dolbyA.au out.au mcompand \
          ".1,.1 4:-46,-56,-26,-36,-20,-26,-15,-17,-9,-9" 80 \
          ".1,.1 4:-46,-56,-26,-36,-20,-26,-15,-17,-9,-9" 3k \
          ".1,.1 4:-46,-56,-26,-36,-20,-26,-15,-17,-9,-9" 9k \
          ".1,.1 4:-42,-56,-23,-36,-18,-26,-14,-17,-9,-9" Real Dolby A probably compands each channel separately but that is left as an exercise to interested readers. See 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 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 writes a noise profile to profile-file or to stdout if no profile-file or if `-' is given. E.g.
       sox_ng speech.wav -n trim 0 1.5 noiseprof speech.noise-profile To actually remove the noise, run SoX again, this time with the noisered effect; noisered reduces noise according to a noise profile generated by noiseprof, from profile-file if it is given or from stdin if no profile-file or if `-' is given. E.g.
       sox_ng speech.wav cleaned.wav 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 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_ng noisy.wav -n trim 0 1 noiseprof | \
          play_ng noisy.wav noisered

norm [dB-level(0)]
    Normalize the audio. norm is just an alias for gain -n; see the gain effect for details.

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 equivalent to remix 1,2i 1,2i.

overdrive [gain(20) [color(20)]]
    Non-linear distortion. The color parameter controls the amount of even harmonic content in the overdriven output. Both parameters range from 0 to 100. There are keymaps for both gain and color. See also centercut.

pad { [%]length[@position(=)] }
    Pad the audio with silence at the beginning, at the end or at any specified points throughout 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 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 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, either specify the end position or specify a zero-length pad at the start. If a pad specification starts with with a % sign, the output is padded to a multiple of length at the specified position. For example, pad 0 %10 adds silence at the end of the audio up to the next multiple of 10 seconds. See delay for an effect that can add silence at the beginning of the audio on a channel-by-channel basis.

phaser [-n|l|q] [-s|t] [gain-in(.4) gain-out(.74) delay(3) regen(.4) speed(.5) [-s|t]
    Add a phasing effect to the audio. See [3] for a detailed description of phasing. delay gives the maximum delay in milliseconds from 0 to 1000, regen the amount of feedback from the delay from -1 to +1 and speed the frequency of delay-time modulation wave in Hz. The modulation is either sinusoidal (-s, the default), which is preferable for multiple instruments, or triangular (-t) which gives single instruments a sharper phasing effect. regen can be from -1 to +1 but should usually be less than 0.5 to avoid clipping and gain-out is the final volume adjustment from -1 to +1. The -l flag makes phaser do linear interpolation between samples when the offset into the delay line is not a whole number, which is about 15% slower but much less noisy and -q does quadratic interpolation, which is about 50% slower but even less noisy. -n explicitly asks for no interpolation, the default, fast and fuzzy. In sox_ng, -s or -t can be given at the start or the end; to be compatible with earlier versions of SoX, supply all the parameters with one of these at the end, use gain-in and gain-out from 0 to 1, delay from 0 to 5, speed from 0.1 to 2 and don't use interpolation. There are keymaps for gain_in, gain_out and regen. Technically, the SoX phaser is not a phaser; it is a flanger. A flanger does comb filtering with equidistant spacing (e.g. 100Hz, 200Hz, 300Hz, 400Hz, ...), while a real phaser does comb filtering with factored spacing (e.g. 100Hz, 200Hz, 400Hz, 800Hz, ...) that sounds more harmonic. For example:
       play_ng snare.flac phaser 0.8 0.74 3 0.4 0.5 -t Gentler:
       play_ng snare.flac phaser 0.9 0.85 4 0.23 1.3 -s A popular sound:
       play_ng snare.flac phaser 0.89 0.85 1 0.24 2 -t More severe:
       play_ng snare.flac phaser 0.6 0.66 3 0.6 2 -t For a flow diagram of how phaser works, say sox_ng --help-effect phaser.

pitch [-q] shift [segment [search [overlap]]]
    Change the audio pitch but not the tempo. shift gives the pitch shift as positive or negative `cents' (i.e. 100ths of a semitone). Note that raising the pitch increases the sample rate and this can make following effects slower, in particular pitch or tempo thenselves, whose running times are proportional to the sample rate times the overlap, all squared. This can be compensated for by following pitch with a fast rate effect. Pitch and tempo share the same fundamental algorithm; see the tempo effect for a description of the other parameters. See the bend, speed and tempo effects.

rate [-q|l|m|g|h|e|v|u] [override-options] [frequency]
    Change the audio sampling rate (i.e. resample the audio) to any given frequency (even non-integer if this is supported by the output file format) using a quality level defined as follows:
     | Quality | B/W | Rej dB | Typical Use
    -q | quick | n/a | ≈30@Fs/4 | playback on ancient hardware
    -l | low | 80% | 100 | playback on old hardware
    -m | medium | 95% | 100 | audio playback
    -g | generic | 95% | 100 | 16-bit
    -h | high | 95% | 125 | 20-bit for 16-bit mastering
    -e | extreme | 95% | 150 | 24-bit
    -v | very high | 95% | 175 | 28-bit for 24-bit mastering
    -u | ultra | 95% | 200 | 32-bit
    These can also be selected with -Q n with n from 0 to 7. B/W (bandwidth) is the percentage of the audio frequency band that is preserved and Rej dB is the level of noise rejection. Increasing levels of resampling quality come at the expense of increasing amounts of time to process the audio. If no quality option is given, the quality level used is `high' when processing audio and `low' when playing it. See Playing & Recording Audio above. The `quick' algorithm uses cubic interpolation; all others use band-limited interpolation. By default, all algorithms have a linear phase response; for `medium' and above, the phase response is configurable (see below). The rate effect is invoked automatically if SoX's -r option specifies a rate that is different to that of the input file(s). Alternatively, if this effect is given explicitly, then SoX's -r option need not be given. For example, the following two commands are equivalent:
       sox_ng input.wav -r 48k output.wav bass -b 24
       sox_ng input.wav output.wav bass -b 24 rate 48k though the second command is more flexible as it allows rate options to be given, and allows the effects to be ordered arbitrarily. A user notes that resampling tracks and then concatenating them is more likely to create clicks at the joints than joining them first and resampling the result, due to edge effects. Override Options The simple quality selection described above provides settings that satisfy the needs of the vast majority of resampling tasks. Occasionally, however, it may be desirable to fine-tune the resampler's filter response; for qualities `medium' and above, this can be achieved using the override options in the following table:
    -M/-I/-L | Phase response=minimum/intermediate/linear
    -s | Steep filter (bandwidth=99%)
    -a | Allow aliasing/imaging above the pass band
    -b width | Any bandwidth % (74-99.7 or 85-99.7 with fR)
    -p phase | Any phase response (0=minimum, 25=intermediate, 50=linear, 100=maximum)
    All resamplers use filters that can sometimes create `echo' (a.k.a. `ringing') artefacts with transient signals such as those that occur with `finger snaps' or other highly percussive sounds. Such artefacts are much more noticeable to the human ear if they occur before the transient (`pre-echo') than if they occur after it (`post-echo'). Note that the frequency of any such artefacts is related to the smaller of the original and new sampling rates but if this is at least 44.1kHz, the artefacts will lie outside the range of human hearing. A phase response setting may be used to control the distribution of any transient echo between `pre' and `post': with minimum phase, there is no pre-echo but the longest post-echo; with linear phase, pre- and post-echo are in equal amounts (in signal terms, but not in audibility); the intermediate phase setting attempts to find the best compromise by selecting a small length (and level) of pre-echo and a medium-length of post-echo. A minimum, intermediate or linear phase response is selected using the -M, -I and -L options; a custom phase response can be created with the -p option. Note that phase responses between `linear' and `maximum' (greater than 50) are rarely useful. A resampler's bandwidth setting determines how much of the frequency content of the original signal (w.r.t. the original sample rate when upsampling or the new sample rate when downsampling) is preserved during conversion. The term `pass band' is used to refer to all frequencies up to the bandwidth point (e.g. for a 44.1kHz sampling rate and a resampling bandwidth of 95%, the pass band represents frequencies from 0Hz (DC) to circa 21kHz). Increasing the resampler's bandwidth results in a slower conversion and can increase transient echo artefacts (and vice versa). The -s `steep filter' option changes the resampling bandwidth from the default of 95% (based on the 3dB point) to 99%. The -b option allows the bandwidth to be set to any value in the range 74-99.7% but bandwidth values greater than 99% are not recommended for normal use as they can cause excessive transient echo. If the -a option is given, aliasing/imaging above the pass band is allowed. For example, with 44.1kHz sampling rate and a resampling bandwidth of 95%, this means that frequency content above 21kHz can be distorted. However, since this is above the pass band (i.e. above the highest frequency of interest/audibility), this may not be a problem. The benefits of allowing aliasing/imaging are reduced processing time and reduced (by almost half) transient echo artefacts. The -d option sets the bit-accuracy in the range 15 to 33, or -R sets the bit-accuracy to obtain rejection of a specified number of dB. Examples:
       sox_ng input.wav -b 16 output.wav rate -s -a 44100 dither -s is default (high) quality resampling with overrides for a steep filter, to allow aliasing, at a 44.1kHz sample rate and noise-shaped dithering to a 16-bit WAV file.
       sox_ng input.wav -b 24 output.aiff rate -v -I -b 90 48k is very high quality resampling with overrides for an intermediate phase, a bandwidth of 90%, at a 48k sampling rate and storing the output to a 24-bit AIFF file. Advanced Options The -i option forces the use of a particular interpolator coefficient from -1 to 2. The -c option tries to limit the number of coefficients to a number of kilobytes; its argument can be from 100 up. The -B option sets the percentage of the pass-band to preserve, from 53 to 95. The -A option sets the percentage of the bandwidth without aliasing, from 85 to 100. -f sets zero pass-band roll-off instead of 0.01dB for -Q 0-2. -n disables internal small-integer optimizations and -t increases the irrational ratio accuracy.

remix [-a|m] [-p] <out-spec>
    out-spec    = 0 | in-spec{,in-spec}
    in-spec    = [in-chan][-[in-chan2]][vol-spec]
    vol-spec    = p|i|v[volume]
    Select and mix input audio channels into output audio channels. Each output channel is specified in turn by a given out-spec which is a list of contributing input channels and volume specifications. Note that this effect operates on the audio channels within the SoX effects processing chain; it should not be confused with the -m global option, where multiple files are mix-combined before entering the effects chain. An out-spec contains comma-separated input channel numbers and hyphen-delimited channel number ranges; alternatively, 0 may be given to create a silent output channel. For example,
       sox_ng input.wav output.wav remix 6 7 8 0 creates an output file with four channels, where channels 1, 2, and 3 are copies of channels 6, 7, and 8 in the input file, and channel 4 is silent. Whereas
       sox_ng input.wav output.wav remix 1-3,7 3 creates a (somewhat bizarre) stereo output file where the left channel is a mix-down of input channels 1, 2, 3 and 7 and the right channel is a copy of input channel 3. Where a range of channels is specified, the channel numbers to the left and right of the hyphen are optional and default to 1 and to the number of input channels respectively. Thus
       sox_ng input.wav output.wav remix - performs a mix-down of all input channels to mono. By default, where an output channel is mixed from multiple input channels, each input channel is scaled by a factor of ¹/n. Custom mixing volumes can be set by following a given input channel or range of input channels with a vol-spec (volume specification) which is one of the letters p, i, or v, followed by a volume number, the meaning of which depends on the given letter:
    Letter | Volume number | Notes
    p | power adjust in dB | 0 = no change
    i | power adjust in dB | As for p but invert the audio
    v | voltage multiplier | 1 = no change; 0.5 ≈ 6dB attenuation; 2 ≈ 6dB gain; -1 = invert
    If an out-spec includes at least one vol-spec then, by default, ¹/n scaling is not applied to any other channels in the same out-spec (though maybe in other out-specs) though the -a (automatic) option can be given to retain the automatic scaling in this case. For example,
       sox_ng input.wav output.wav remix 1,2 3,4v0.8 results in channel level multipliers of 0.5,0.5 and 1,0.8, whereas
       sox_ng input.wav output.wav remix -a 1,2 3,4v0.8 results in channel level multipliers of 0.5,0.5 and 0.5,0.8. The -m (manual) option disables all automatic volume adjustments, so
       sox_ng input.wav output.wav remix -m 1,2 3,4v0.8 results in channel level multipliers of 1,1 and 1,0.8. The volume number is optional and omitting it corresponds to no volume change; however, the only case in which this is useful is in conjunction with i. For example, if input.wav is stereo, then
       sox_ng input.wav output.wav remix 1,2i is a mono equivalent of the oops effect and
       play file.mp3 remix 1,2 1i,2i lets you get twice as much power from a mono speaker connected between the left and right poles than you would from two connected the usual way, (but mind it doesn't blow the amplifier, as it draws twice as much current). If the -p option is given, any automatic ¹/n scaling is replaced by ¹/√n (`power') scaling; this gives a louder mix but one that may occasionally clip. One use of the remix effect is to split an audio file into a set of files, each containing one of the constituent channels in order to perform subsequent processing on individual audio channels. When more than a few channels are involved, a script such as the following is useful: #! /bin/sh chans=`soxi_ng -c "$1"` while [ $chans -ge 1 ]; do
       chans0=`printf %02i $chans` # 2 digits hence up to 99 chans
       out=`echo "$1" | sed "s/\(.*\)\.\(.*\)/\1-$chans0.\2/"`
       sox_ng "$1" "$out" remix $chans
       chans=`expr $chans - 1` done If a file input.wav containing six audio channels were given, the script would produce six output files: input-01.wav, input-02.wav, ..., input-06.wav. See the swap effect.

repeat [count(1)|-]
    Repeat the entire audio count times, or once if count is not given. The special value - requests infinite repetition. It requires temporary file space to store the audio to be repeated. Note that repeating once yields two copies: the original audio and the repeated audio.

reverb [-w] [reverberance(50%) [HF-damping(50%) [room-scale(100%)
    [stereo-depth(100%) [pre-delay(0ms) [wet-gain(0dB)]]]]]] Add reverberation to the audio using the `freeverb' algorithm. A reverberation effect is sometimes desirable for concert halls that are too small or contain so many people that the hall's natural reverberance is diminished. Applying a small amount of stereo reverb to a dry mono signal usually makes it sound more natural. See [3] for a detailed description of reverberation. This effect increases the volume of the audio and continues to reverberate after the input finishes so, to prevent clipping and keep the audible part of the final reverberation, a typical invocation might be:
       play_ng dry.au gain -3 pad 0 1 reverb The -w option can be given to select only the `wet' signal, thus allowing it to be processed further, independently of the `dry' signal. E.g.
       play_ng -m in.au "|sox_ng in.au -p reverse reverb -w reverse" for a reverse reverb effect.

reverse
    Reverse the audio completely. Requires temporary file space to store the audio to be reversed.

riaa
    Apply RIAA vinyl playback equalization. The sampling rate must be 44.1, 48, 88.2, 96 or 192kHz. This effect supports the --plot global option.

saturation [type [blend [offset [drive|color|threshold]]]]
    Add saturation, which can produce effects ranging from subtle warmth to crunchy fuzz. The type parameter selects the saturation type: tanh (the default), sqrt or diode. For all types, the blend parameter (default 1) controls the mixture of wet and dry signals in the output, with 1 being fully wet. The offset parameter (default 0) adds a DC offset to the input to produce asymmetric distortion. The offset is removed from the output, so that a zero input level produces a zero output level, but when the input is non-zero the output waveform is likely to be asymmetric. The tanh saturation type uses the hyperbolic tangent function to apply soft clipping. The drive parameter (default 1) controls the input gain and thus the amount of distortion. The sqrt saturation type uses a mixture of two functions: x*sqrt(|x|) and sgn(x)*sqrt(|x|), which give different tonal qualities to the output. The color parameter (default 0.5) controls the mixture of these functions, with 0 being purely x*sqrt(|x|) and 1 being purely sgn(x)*sqrt(|x|). The diode saturation type models the effect of using a pair of diodes to clip the signal when it exceeds a threshold (default 0.5). The blend parameter, by mixing the wet and dry signals, effectively controls the amount of attenuation that occurs above the threshold, from no attenuation when blend is 0 to complete attenuation (hard clipping) when blend is 1. There are keymaps for blend, offset, drive, color and threshold. See the overdrive effect for another kind of non-linear distortion. When the offset parameter is used to produce asymmetric distortion, the highpass effect can be used to rebalance the waveform's positive and negative amplitude.

sdm [-f filter] [-t order] [-n num] [-l latency]
    Apply a 1-bit sigma-delta modulator producing DSD output. The input should be previously upsampled, e.g. with the rate effect, to a high rate, 2.8224MHz for DSD64. The -f option selects the noise-shaping filter from the following list where the number indicates the order of the filter:
    clans-4 | sdm-4
    clans-5 | sdm-5
    clans-6 | sdm-6
    clans-7 | sdm-7
    clans-8 | sdm-8

    The noise filter may be combined with a partial trellis/viterbi search by supplying the following options:

    -t order
        Trellis order, max 32.

    -n num
        Number of paths to consider, max 32.

    -l latency
        Output latency, max 2048.

    The result of using these parameters is hard to predict and can include high noise levels or instability. Caution is advised.

silence [-l] above-periods [duration threshold[d|%]]
    [below-periods duration threshold[d|%]] Removes silence from the beginning, middle or end of the audio, where `silence' is determined by a specified threshold. The above-periods value is used to indicate whether audio should be trimmed at the beginning of the audio. A value of zero indicates that no silence should be trimmed from the beginning in which case duration and threshold are omitted. When a non-zero above-periods is specified, you must also specify a duration and threshold and it trims audio until it finds non-silence. It will normally be 1 when trimming silence from the beginning of the audio, but it can be increased to higher values to trim all audio up to the Nth non-silence period. For example, if you have an audio file with two songs that each contains 2 seconds of silence before the song, you could specify an above-period of 2 to strip out both silences and the first song. duration indicates the amount of time for which non-silence must be detected before it stops trimming the silence before it. By increasing duration, short bursts of quiet noise can be treated as silence and trimmed off. duration has the peculiarity that a bare number is interpreted as a sample count, not as a number of seconds. To specify seconds, either use the t suffix (as in 2t), a decimal point (as in 2.0) or specify minutes too (as in 0:02). threshold indicates the maximum sample value in any channel is considered 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 include background noise. threshold numbers may be suffixed with d to indicate that the value is in decibels or % to indicate a percentage of the maximum possible sample value. By default, it is in percent. To trim silence from the end of the audio, specify a below-periods count, which means to remove all audio after the last onset of silence is detected. Normally, this will be 1 but it can be increased to leave shorter periods of silence and the audio that follows them intact. For example, if you have a track with 1 second of silence in the middle and 1 second at the end, you could set below-period to 2 to leave the middle silence and what follows it and remove from the final silence on. When below-periods is given, its duration specifies the length of silence that must exist before audio is not copied any more. By specifying a higher duration, shorter silences that are wanted can be left in the audio. For example, if you have a song with 1 second of silence in the middle and 2 seconds of silence at the end, a duration of 2 could be used to skip over the middle silence and trim the end instead of starting trimming from half way through. Unfortunately, the length of the silence at the end has to be longer than any preceding silence for this to work so you must know the length of the silence at the end. A more reliable way to trim silence from the end is to use the silence effect in combination with the reverse effect. By first reversing the audio, you can use the above-periods to trim from what looks like the front of the file, then reverse it again to get back to normal. To remove silence from the middle of a file, give a negative below-periods. This value is then treated as positive value and is also used to indicate that 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 -l option indicates that below-periods' duration of `silent' 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 following example shows how this effect can be used to make a recording that does not contain the silence that usually occurs between pressing the record button and the start of the performance:
       rec_ng parameters filename other-effects silence 1 5 2% This example should remove the start of the recording until there's a period of non-silence longer than 0.2s and louder than 0.1%, then start searching for a silence that's longer than 1s and quieter than 3% and remove it if found, leaving the first 1s of it in place, then start copying again until a silence is found that's longer than 1s and quieter than 3%, trim that to 1s and so on.
       sox_ng in.au out.au silence -l 1 0.2 0.1% -1 1.0 3%

sinc [-a att|-b beta] [-p phase|-M|-I|-L] [-t tbw|-n taps]
    [freqHP][-freqLP [-t tbw|-n taps]] [-r]] [-d]] Apply a kaiser-windowed low-pass, high-pass, band-pass or band-reject filter to the signal. The freqHP and freqLP parameters give the frequencies of the 6dB points of a high-pass and low-pass filter that may be invoked individually or together. If both are given, freqHP less than freqLP creates a band-pass filter and freqHP greater than freqLP creates a band-reject filter. For example, the invocations
       sinc 3k
       sinc -4k
       sinc 3k-4k
       sinc 4k-3k create a high-pass, low-pass, band-pass and band-reject filter respectively. The default stop band attenuation of 120dB can be overridden with -a; alternatively, the kaiser window's `beta' parameter can be given directly with -b. The default transition bandwidth of 5% of the total band can be overridden with -t (and tbw in Hertz); alternatively, the number of filter taps can be given directly with -n and is limited to the range of 11-1,073,741,823 though the maximum requires 56GB of physical RAM if it is to complete in minutes rather than days, while 100,000,000 requires 8GB. If both freqHP and freqLP are given, a -t or -n option given to the left of the frequencies applies to both frequencies; one of these options given to the right of the frequencies applies only to freqLP. The -p, -M, -I and -L options control the filter's phase response; see the rate effect for details. The -r option controls whether the filter should round the number of taps to the closest integer instead of truncating it. The -d option specifies that, if a low-pass filter is being created and the cutoff frequency is at or above the Nyquist frequency, the sinc effect should be deleted from the effects chain instead of failing. This effect supports the --plot global option.

softvol [volume(1.0) [double-time(0) [headroom(0)]]]
    The soft volume effect applies a simple multiplier to the audio ensuring that it does not clip. When a sample would have clipped the volume multiplier is automatically reduced to compensate. It is a simple compander with the advantages of running fast, having no pre- or post-echo and reacting on the crests of the wave, so its volume-reduction glitches don't add audible noise. volume sets the initial volume multiplier; the default of 1.0 means no change. double-time says that the volume should slowly increase at a rate that makes it double every double-time seconds. A good value for usual music is 10 and the default value of 0 says that the volume should not increase automatically. headroom is in dB and limits the loudest amplitude to less than the 32-bit maximum. This may be necessary when the final bit-depth reduction and/or dithering make it clip. A value of 0.1 is sufficient to protect down to a bit-depth of 8 with dithering. There are keymaps for volume, double_time and headroom. When playing sound in interactive mode, the `v' and `V' keys reduce and increase the volume if there is a softvol in the effects chain. If there are more than one, which one it adjusts is probably random.

spectrogram [options]
    Create a spectrogram of the audio. The audio is passed unmodified through the SoX processing chain. This effect is optional - type sox_ng --help and check the list of supported effects to see if it has been included. The spectrogram is rendered in a Portable Network Graphic (PNG) file and shows time in the X axis, frequency in the Y axis and audio signal magnitude in the Z axis, represented by the color (or optionally the intensity) of the pixels in the X-Y plane. If the audio signal contains multiple channels, these are shown from top to bottom starting from channel 1, which is the left channel for stereo audio. For example, if `my.wav' is a stereo file, then
       sox_ng my.wav -n spectrogram creates a spectrogram of the entire file in the file `spectrogram.png'. More often though, analysis of a smaller portion of the audio is required; e.g. with
       sox_ng my.wav -n remix 2 trim 20 30 spectrogram the spectrogram shows information only from the second (right) channel of thirty seconds of audio starting from twenty seconds in. To analyze a small portion of the frequency domain, the rate effect may be used, e.g.
       sox_ng my.wav -n rate 6k spectrogram allows detailed analysis of frequencies up to 3kHz (half the sampling rate) i.e. where the human auditory system is most sensitive. See also the -R option below. With
       sox_ng my.wav -n trim 0 10 spectrogram -x 600 -y 200 -z 100 the given options control the size of the spectrogram's X, Y & Z axes (in this case, the spectrogram area of the produced image will be 600 by 200 pixels in size and the Z axis range will be 100 dB). Note that the produced image includes axes, legends etc. and will be larger than the specified spectrogram size unless the -r option is given: if each spectrogram is x × y and there are c channels, the image will be x + 144 by (y × c) + 78, plus c - 1 if -a was not given, and 20 pixels higher than this if you gave -t Title. A raw spectrogram will be x by y × c. In this example
       sox_ng -n -n synth 6 tri 10k:14k spectrogram -z 100 -w kaiser an analysis window with high dynamic range is selected to best display the spectrogram of a swept triangular wave. For a similar example, append the following to the `chime' command in the description of the delay effect (above):
       rate 2k spectrogram -X 200 -Z -10 -w kaiser Options are also available to control the appearance (color set, brightness, contrast etc.) and filename of the spectrogram; e.g. with
       sox_ng my.wav -n spectrogram -m -l -o print.png a spectrogram is created suitable for printing on a black and white printer. Options

    -x num
        Change the (maximum) width (X axis) of the spectrogram from its default value of 800 pixels to a given number between 100 and a million. See -X and -d.

    -X num
        X axis pixels per second; the default is auto-calculated to fit the audio to the X axis size if its duration is known or given with -d, or 100 otherwise. If given without a -x option when the length of the audio is known, this option determines the width of the spectrogram; otherwise, it affects the duration of the spectrogram. num can be from 1 (low time resolution) to 5000 (high time resolution) and need not be an integer. SoX may make a slight adjustment to the given number for processing quantization reasons; if so, SoX reports the actual number used (viewable when the SoX global option -V is in effect).

    -y num
        Sets the size of the Y axis per channel in pixels; this is the number of frequency `bins' used in the Fourier analysis that produces the spectrogram. By default the Y axis size is chosen automatically, depending on the -Y height and the number of channels, with a minimum of 64. The DFT size is set to 2 × (num - 1) and if SoX was compiled with FFTW, sizes of 2^a × 3^b × 5^c × 7^d × 11^e × 13^f where e + f < 2 are said to be fastest. If it wasn't, anything other than powers of two is a hundred of times slower in which case heights of a power of two plus one will be faster.

    -Y num
        Sets the total height of the spectrogram(s). The default value is 550 pixels and the maximum is a million. If num is not an exact multiple of the number of channels with -r, the actual total height of the spectrogram area will be a few pixel rows less. For non-raw spectrograms instead, the height of the graph area will be slightly less for the same reason, slightly more for the single-pixel row between adjacent channel graphs if -a wasn't given, and the overall height of the image will be greater by the time axes (28) and by the title (20) if present.

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

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

    -n  Normalizes the upper limit of the Z axis so that the loudest pixels are shown using the brightest color in the palette - a kind of automatic -Z flag.

    -q num
        Sets the Z axis quantization, i.e. the number of different colors (or intensities) in which to render Z axis values. A small number (e.g. 4) gives a poster-like effect making it easier to discern magnitude bands of similar level and results in a smaller PNG file. The number given specifies the number of colors to use in the Z axis range; two colors are reserved to represent out-of-range values.

    -w name
        Select a window function: Hann (the default), Hamming, Bartlett, Rectangular, Kaiser or Dolph. The spectrogram is produced using the Discrete Fourier Transform (DFT) algorithm and a significant parameter of this algorithm is the choice of window function. By default, SoX uses the Hann window, which has good all-round properties for frequency resolution and dynamic range. For better frequency resolution but lower dynamic range, select a Hamming window; for higher dynamic range but poorer frequency resolution, select a Dolph window.

    -W num
        Window adjustment parameter. This can be used to make small adjustments to the Kaiser and Dolph windows. A positive number (up to ten) increases its dynamic range, a negative number decreases it.

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

    -a  Suppress the display of the axis lines. This is sometimes useful in helping to discern artefacts at the spectrogram edges.

    -r  Raw spectrogram: suppress the display of axes and legends.

    -g  When there are many seconds, show them as MM:SS or HH:MM:SS.

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

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

    -p num
        Permute the colors in a color or hybrid palette. The num parameter, from 1 (the default) to 6, selects the permutation.

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

    -A  Selects an alternative, fixed color set. This is provided only for compatibility with spectrograms produced by another package. It should not normally be used as it has some problems, not least, a lack of differentiation at the bottom end which results in masking of low-level artefacts.

    -t text
        Set the image title, the text to display above the spectrogram. If you need it to be `chorus' or some other effect's name, surround it by spaces inside double quotes.

    -c text
        Set (or clear) the image comment, the text to display below and to the left of the spectrogram.

    -o file
        The name of the spectrogram output PNG file, default `spectrogram.png'. If `-' is given, the spectrogram is sent to the `standard output' (stdout).

    -L  Plot the frequency on a logarithmic axis.

    -R L:H
        Specify the frequency range (from L to H). The frequencies can have an optional suffix
           sox_ng mymusic.mp3 -n spectrogram -L -R 100:8k By default, the lowest frequency is 0Hz for a linear graph or 1Hz for a logarithmic graph and the highest is the Nyquist frequency.

-i  Interpolate vertically: where there are more output pixels than frequency bins, use a weighted average of the bins above and below the pixel's frequency and where there are more frequency bins than pixels, average the bins that fall in this pixel row.

    Advanced Options
    In order to process a smaller section of audio without affecting other effects or the output signal (unlike when the trim effect is used), the following options may be used:

    -d duration
        This option sets the X axis resolution such that audio with the given duration (a time specification) fits the selected (or default) X axis width. It defaults, if the audio length is known, to the audio length minus the start time. For example,
           sox_ng input.mp3 output.wav -n spectrogram -d 1:00 stats creates a spectrogram showing the first minute of the audio, while the stats effect is applied to the entire audio signal. See -X for an alternative way of setting the X axis resolution.

    -S position(=)
        Start the spectrogram at the given point in the audio stream. For example
           sox_ng input.aiff output.wav spectrogram -S 1:00 creates a spectrogram showing all but the first minute of the audio (the output file, however, receives the entire audio stream).

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

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 it up, less than 1 slows it down) or, if the letter c is appended, it's the number of cents (100ths of a semitone) by which the pitch (and tempo) should be adjusted: greater than 0 increases, less than 0 decreases. Technically, the speed effect only changes the sample rate information, leaving the samples themselves untouched. The rate effect is invoked automatically to resample to the output sample rate, using its default quality/speed. For higher quality or higher speed resampling, in addition to the speed effect, specify the rate effect with the desired quality option. See the bend, pitch and tempo effects.

speexdsp [-agc [target_level(100)]] [-denoise [max_db(15)]] [-dereverb]
    [-fps frames_per_second(20)] [-spf samples_per_frame] Use the Speex DSP library to improve perceived sound quality. If no options are specified, the -agc and -denoise features are enabled.

    -agc [target_level]
        Enable automatic gain control and optionally specify a target volume level from 1 to 100.

    -denoise [max_db]
        Enable noise reduction and optionally specify the maximum attenuation from 1 to 100.

    -dereverb
        Enable reverb reduction.

    -fps frames_per_second
        Specify the number of frames per second from 1-100.

    -spf samples_per_frame
        Specify the number of samples per frame. The default is derived from the -fps setting so that frames abut but do not overlap.

splice  [-h|t|q] {position(=)[,excess[,leeway]]}
    Splice audio sections together. This effect provides two things over simple audio concatenation: a (usually short) cross-fade is applied at the join and a wave similarity comparison is made to help determine the best place at which to make the join. One of the options -h, -t, or -q may be given to select the fade envelope as half cosine wave (the default), triangular (a.k.a. linear), or quarter cosine wave (e.g. for a cross-fade of correlated audio).
     | Audio | Fade level | Transitions
    -h | correlated | constant gain | smooth
    -t | correlated | constant gain | abrupt
    -q | uncorrelated | constant power | smooth
    To perform a splice, first use the trim effect to select the audio sections to be joined together. As when performing a tape splice, the end of the section to be spliced onto should be trimmed with a small excess (default 0.005 seconds) after the ideal joining point. The beginning of the audio section to splice on should be trimmed with the same excess before the ideal joining point plus an additional leeway (default 0.005 seconds). SoX should then be invoked with the two audio sections as input files and the splice effect given with the position at which to perform the splice - this is length of the first audio section (including the excess). The following diagram uses the tape analogy to illustrate the splice operation. The effect simulates the diagonal cuts and joins the two pieces:
         length1 excess
       -----------><--->
       _________ : : _________________
                \ : : :\ `
                 \ : : : \ `
                  \: : : \ `
                   * : : * - - *
                    \ : : :\ `
                     \ : : : \ `
       _______________\: : : \_____`____
                         : : : :
                         <---> <----->
                         excess leeway where * indicates the joining points. For example, a long song begins with two verses which start (as determined e.g. by using the play_ng command with the trim (start) effect) at times 0:30.125 and 1:03.432. The following commands cut out the first verse:
       sox_ng too-long.wav part1.wav trim 0 30.130 (5 ms excess, after the first verse starts)
       sox_ng too-long.wav part2.wav trim 1:03.422 (5 ms excess plus 5 ms leeway, before the second verse starts)
       sox_ng part1.wav part2.wav just-right.wav splice 30.130 For another example, the SoX command
       play_ng "|sox_ng -n -p synth 1 sin %1" "|sox_ng -n -p synth 1 sin %3" generates and plays two notes, but there is a nasty click at the transition; the click can be removed by splicing instead of concatenating the audio, i.e. by appending splice 1 to the command. Clicks at the beginning and end of the audio can be removed by preceding the splice effect with fade q .01 2 .01. Provided your arithmetic is good enough, multiple splices can be performed with a single splice invocation. For example, with a Bourne shell script `acpo':
       #! /bin/sh
       # Audio Copy and Paste Over
       # acpo infile copy-start copy-stop paste-over-start outfile
       # No chained time specifications allowed for the parameters
       # (i.e. such that contain +/-).
       e=0.005 # Using default excess
       l=$e # and leeway.
       sox_ng "$1" piece.wav trim $2-$e-$l =$3+$e
       sox_ng "$1" part1.wav trim 0 $4+$e
       sox_ng "$1" part2.wav trim $4+$3-$2-$e-$l
       sox_ng part1.wav piece.wav part2.wav "$5" \
          splice $4+$e +$3-$2+$e+$l+$e two splices are used to `copy and paste' audio. It is also possible to use this effect to perform general cross-fades, e.g. to join two songs. In this case, excess would typically be a number of seconds, the -q option would typically be given to select an `equal power' cross-fade and leeway should be zero (which is the default if -q is given). For example, if f1.wav and f2.wav are audio files to be cross-faded, then
       sox_ng f1.wav f2.wav out.wav splice -q $(soxi_ng -D f1.wav),3 cross-fades the files where the point of equal loudness is 3 seconds before the end of f1.wav, i.e. the total length of the cross-fade is 2 × 3 = 6 seconds ($(...) is POSIX shell notation that is replaced by the output of the enclosed command).

stat [-s scale] [-rms] [-freq] [-v] [-d] [-a] [-h]
    Display time and frequency domain statistical information about the audio. Audio is passed unmodified through the SoX processing chain. The information is output to the `standard error' (stderr) stream and is calculated (where n is the duration of the audio in samples, c is the number of audio channels, r is the audio sample rate and xk represents the value (in the range -1 to +1) of each successive sample in the audio), as follows:
    Samples read | n×c
    Length (seconds) | n÷r
    Scaled by | See -s below.
    Maximum amplitude | max(xk) The maximum sample value in the audio; usually this will be a positive number.
    Minimum amplitude | min(xk) The minimum sample value in the audio; usually this will be a negative number.
    Midline amplitude | ½min(xk)+½max(xk)
    Mean norm | ¹/nΣ|xk| The average of the absolute value of each sample in the audio.
    Mean amplitude | ¹/nΣxk The average of each sample in the audio. If this figure is non-zero, then it indicates the presence of a DC offset which could be removed using the dcshift effect.
    RMS amplitude | √(¹/nΣxk²) The level of a DC signal that would have the same power as the audio's average power.
    Maximum delta | max(|xk-xk-1|)
    Minimum delta | min(|xk-xk-1|)
    Mean delta | ¹/n-1Σ|xk-xk-1|
    RMS delta | √(¹/n-1Σ(xk-xk-1)²)
    EBUR128 Momentary | The maximum momentary loudness over 400ms
    EBUR128 Short Term | The maximum short term loudness over 3 seconds
    EBUR128 Integrated | The integrated loudness over the whole file
    EBUR128 True Peak | The maximum of the True Peak of each channel
    Rough frequency | In Hz.
    Volume Adjustment | The parameter to the vol effect which would make the audio as loud as possible without clipping. See the discussion on Clipping above for reasons why it is rarely a good idea actually to do this.
    Note that the delta measurements are not applicable to multichannel audio and EBU R 128 (=ITU-R BS.1770) measurements are in Loudness Units referenced to Full Scale (LUFS), The -s option can be used to scale the input data by a given factor. The default value of scale is 2147483647 (the maximum value of a 32-bit signed integer) as internal effects always work with those. A lower value means that a different sample value should be considered as the full-scale amplitude. The -rms option converts all average values to `root mean square' format. The -freq option outputs the input's power spectrum (a 4096-point DFT) instead of the statistics listed above. This should only be used with a single-channel audio file. The -v option displays only the `Volume Adjustment' value. The -d option displays a hex dump of the 32-bit signed PCM data audio in SoX's internal buffer. This is mainly used to help track down endian problems that sometimes occur in cross-platform versions of SoX. The -a option outputs the average power spectrum instead of the power spectrum for each 4096-point DFT. The -h option uses the "histogram algorithm" to calculate the integrated EBU R-128 loudness, which requires less memory but is less accurate. The -j option outputs the statistics in JSON format, e.g.: {
      "samples_read": 22699008,
      "length": 236.448,
      "scaled_by": 2.14748e+09,
      "maximum_amplitude": 0.818604,
      "minimum_amplitude": -0.532471,
      "midline_amplitude": 0.143066,
      "mean_norm": 0.0352694,
      "mean_amplitude": 0.00180676,
      "rms_amplitude": 0.056726,
      "maximum_delta": 0.367126,
      "minimum_delta": 0,
      "mean_delta": 0.0177341,
      "rms_delta": 0.0268538,
      "rough_frequency": 3616,
      "volume_adjustment": 1.22159 } If -rms was given, "scaled_by" will be "scaled_by_rms" and if -e was given, you also get
      "ebur128_momentary": -30.3408,
      "ebur128_short_term": -35.4501,
      "ebur128_integrated": -21.3583, Some fields may be absent if their values are incalculable (EBUR128 figures) or would be infinite (like the RMS of silence). As JSON uses scientific notation, it can shows the values of very small numbers that the usual output shows as zero. The most common use of stat is to measure the characteristics of a single audio file, for which the syntax is:
       sox_ng file.wav -n stat where -n means "No audio output is required."

stats [-b bits|-x bits|-s scale] [-w time] [-j]
    Display time domain statistical information about the audio channels; audio is passed unmodified through the SoX processing chain. Statistics are calculated and displayed for each audio channel and, where applicable, an overall figure is also given. For example, for a typical well-mastered stereo music file:
                    Overall Left Right
       DC offset 0.000803 -0.000391 0.000803
       Min level -0.750977 -0.750977 -0.653412
       Max level 0.708801 0.708801 0.653534
       Pk lev dB -2.49 -2.49 -3.69
       RMS lev dB -19.41 -19.13 -19.71
       RMS Pk dB -13.82 -13.82 -14.38
       RMS Tr dB -85.25 -85.25 -82.66
       Crest factor - 6.79 6.32
       Flat factor 0.00 0.00 0.00
       Pk count 2 2 2
       Bit-depth 16/16 16/16 16/16
       Num samples 7.72M
       Length s 174.973
       Scale max 1.000000
       Window s 0.050 DC offset, Min level, and Max level are shown, by default, in the range ±1. If the -b (bits) options is given, these three measurements are scaled to a signed integer with the given number of bits from 2 to 32. For example, for 16 bits, the scale would be -32768 to +32767. The -x option behaves the same way as -b except that the signed integer values are displayed in hexadecimal. The -s option scales the three measurements by a given floating point number. Pk lev dB and RMS lev dB are the standard peak and RMS levels measured in dBFS. RMS Pk dB and RMS Tr dB are peak and trough values of the RMS level measured over a short window (default: 50ms). That can be changed with the -w option in seconds from 0.01 to 10. Crest factor is the ratio of peak to RMS level (note: not in dB). Flat factor is a measure of the flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min level or Max level). Pk count is the number of occasions (not the number of samples) that the signal attained either Min level, or Max level. The primary goal of the Peak Count value is to answer the question "has this audio been clipped?", quite possibly as a result of the frowned-upon-by-some but common practice of 'brick wall limiting' in modern mastering. The closer the "Peak Count" is to 1, the higher the confidence that the audio has not been clipped. The right-hand Bit-depth figure is the standard definition of bit-depth, i.e. that all bits other than this number of the most significant bits are always zero. The left-hand figure is the number of bits at the least significant end of those most significant bits that would be sufficient to represent all sample values accurately (including the sign bit). In mathematical terms, the right-hand figure is the ordinal, counting from the most significant bit, of the least significant bit that is set to one in at least one sample. The left-hand figure is the ordinal, counting from the least significant repeated sign bit across all samples, of the least significant bit that is set to one in at least one sample. Bit-depths are not intended to be properties of the signal per se but properties of its 2's-complement PCM encoding. The primary use case of bit-depth measurement concerns manipulation of PCM audio by simple bit shifting, to answer questions such as: "Is it likely that this 24-bit PCM file was created by simply converting a 16-bit PCM file to 24-bit?" or "Can I losslessly shift all the samples in this PCM audio file m-bits left or n-bits right?" For multichannel audio, an overall figure for each of the above measurements is given and derived from the channel figures as follows: DC offset: maximum magnitude; Max level, Pk lev dB, RMS Pk dB, Bit-depth: maximum; Min level, RMS Tr dB: minimum; RMS lev dB, Flat factor, Pk count: average; Crest factor: not applicable. Length s is the duration in seconds of the audio and, unlike stat, Num samples is equal to the sample rate multiplied by Length. Scale max is the scaling applied to the first three measurements; specifically, it is the maximum value that could apply to Max level. Window s is the length of the window used for the peak and trough RMS measurements. The -j option outputs JSON with three fields:

    "channel_count"
        An integer.

    "overall"
        An object with a member for each row of the first column of the usual output, which are all numbers except for "bit_depth", which is an array of two numbers.

    "channels"
        An array of objects with the per-channel values.
To know the overall and the channels' member names, have a look at the output. Like stat, the usual way to measure the characteristics of a single audio file is:
   sox_ng file.wav -n stats

stretch [factor [window [fade [shift [fading]]]]]
    Change the audio duration but not its pitch by cross-fading between short windows of samples. This effect is broadly equivalent to the tempo effect with factor inverted and search set to zero so, in general, its results are comparatively poor; it is retained as it can sometimes outperform tempo for small factors. factor determines the change in length: >1 lengthens and <1 shortens. By default, it is 1 (no change) window is the length of the cross-fading window in milliseconds with a default of 20. The fade option chooses the type of crossfading: linear and half-cosine give equal-gain crossfading and cannot clip; sqrt and quarter-cosine give two kinds of equal-power crossfading. The shift ratio can be from 0 to 1 and its default depends on the stretch factor: 1 when speeding up, 0.8 when slowing down. The fading ratio, from 0 to 0.5, seems to be how much of each window is cross-faded with the adjacent ones. The default value depends on factor and shift: 1.0 - (factor × shift) if speeding up, 1.0 - shift if slowing down, with a maximum of 0.5. The duration of stretch's output is slightly longer than the duration of the input multiplied by factor as it has to empty the delay line it uses; tempo is more precise.

swap
    Swap stereo channels. If the input is not stereo, pairs of channels are swapped and a possible odd last channel is passed through. E.g., for seven channels, the output order will be 2, 1, 4, 3, 6, 5, 7. See remix for an effect that allows arbitrary channel selection, ordering and mixing.

synth [-j|p key] [-n] [length [offset [phase [p1 [p2 [p3]]]]]] {type [combine [fixed[,extra[,mix]]]] [freq[:|+|/|-freq2] [offset [phase [p1 [p2 [p3]]]]]]}
    synth generates fixed or swept frequency audio tones with various wave shapes and wide-band noise of various colors. Multiple synth effects can be cascaded to produce more complex waveforms and at each stage it is possible to choose whether the generated waveform is mixed with or modulated onto the output of the previous stage, and the audio for each channel in a multichannel audio file can be synthesized independently. It generates audio at maximum volume (0dBFS), which means that there is a high chance of clipping so, in many cases, you will want to follow it with the gain effect to prevent this from happening. (See Clipping above.) Though this effect is used to generate audio, an input file must still be given, the characteristics of which are used to set the synthesized 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 input filename -n) is often given instead and the length specified as a parameter to synth or by some other effect that has an associated length. By default, the tuning used with note notations is equal temperament; the -j key option selects just intonation, where key is a whole number of semitones relative to A (so for example, -9 or 3 selects the key of C) or a note in scientific notation and -p selects Pythagorean tuning. By default, the synth effect incorporates the functionality of gain -h (see the gain effect for details); synth's -n option may be given to disable this behaviour. length is the length of audio to synthesize. A value of 0 indicated to use the input length, which is also the default. Note that, if the input is -n and the length is 0 or absent, it continues generating audio until it is stopped in some other way. type is one of

    sine
        A sinusoidal wave is the default type and ignores all the p parameters.

    square
        A square wave. p1 sets the percentage of each cycle that is `on' with a default of 50.
          |_______ | +1
          | | |
          |_______|_______| 0
          | | |
          | |_______| -1
          | |
          0 p1 1

    triangle
        p1 sets the percentage of each cycle that is `rising' with a default of 50.
          | . | +1
          | / \ |
          |__/___\__| 0
          | / \ |
          |/ \| -1
          | |
          0 p1 1

    sawtooth
        A sawtooth wave. With a phase of 0 it starts at -1 and rises to 1, and of 10 it starts at -0.9. The offset makes no difference.
          | /| +1
          | / |
          |__/__| 0
          | / |
          |/ | -1
          0 1

    trapezium
        The trapezoidal wave starts at -1, rises linearly to 1, stays there, falls linearly to -1, stays there and repeats. p1 sets the percentage of the cycle in which the wave is rising with a default of 10, p2 sets the percentage through each cycle at which falling begins with a default of 50 and p3 sets the percentage through each cycle at which falling ends with a default of 60.
          | ______ |+1
          | / \ |
          |__/________\___________| 0
          | / \ |
          |/ \_________|-1
          | |
          0 p1 p2 p3 1

    exp The exponential wave rises from -1 to 1 where it peaks and immediately begins an exponential fall. p1 sets the position of the maximum with a default of 50. p2 sets the minimum amplitude in multiples of 2dB down from the maximum with a default of 50 (100dB); values below 50 raise the shoulders of the wave and values above 50 lower the shoulders, increasing the pointedness of the spike.
          | | +1
          | /\ |
          | _' `_ | 0
          | _- -_ |
          |____---' `---____ | f(p2)
          | |
          0 p1 1

    whitenoise
        Random noise with equal power at every frequency. All noise generators ignore the frequency and phase parameters but if a DC offset is given, the signal's amplitude is automatically adjusted to prevent clipping so, for noise in the range 0 to 1, an offset of 0.5 would give a signal ranging from 0.0 to 1.0 and -0.9 from -1.0 to -0.8 noise is a handy alias for whitenoise

    tpdfnoise
        Noise with a Triangular Probability Density Function.

    pinknoise
        Random noise with the power at each frequency inversely proportional to the frequency.

    brownnoise
        Random noise with the power at each frequency inversely proportional to the frequency squared.

    pluck
        A plucked string simulation in which an array of sample values representing a taut string is set in motion with a burst of noise and decayed over time. A plucked note's frequency can be from 27.5 to 4220Hz and the sampling rate must be between 44100 and 48000Hz. If a DC offset is used, the amplitude is automatically adjusted to prevent clipping. p1 affects the sustain with a default of 40 (2dB per second); higher values give a slower decay and lower values a faster one. p2 and p3 are tone controls for the initial excitation, with default values of 20 and 90 and a special case when p3 is exactly 100. If the phase is non-zero, it uses a different kind of random numbers.

    If the offset, phase and p parameters are given before the first type, they set the default values for all the following stages. combine is one of

    create
        Puts each stage's output in a new output channel and is the default:

    mix Mixes the generated audio 50:50 with the input signal.

    amod
        Amplitude-modulates (multiplies) the input signal by the synthesized one considered as a value from 0 (for the most negative value) to 1 (for the most positive value).

    fmod
        Multiplies the input signal with the synthesized one (ring modulation).

    vdelay
        Mixes the input signal with a delayed version of it using the synthesized signal to modulate the depth of the delay. The following three-part option fixed[,extra[,mix]] specifies the fixed and additional parts of the delay in milliseconds and what percentage of the output consists of the delayed signal from 0 for all input signal to 100 for all delayed signal with a default of 50 (half and half). The synthesized signal's value from -1 to +1 varies the delay from fixed seconds to fixed + (0 to extra) seconds. It interpolates linearly between the input samples and can be used to make precision phaser, flanger and chorus-like effects, vibrato and frequency modulation (FM) synthesis (actually phase modulation, as used in the Yamaha DX7). A chorus-like effect:
           sox_ng solo.au -d synth sine vdelay 50,2,50 .25 0 75 A flanger:
           sox_ng solo.au -d synth triangle vdelay 0,2,41.52 0.5 0 0

    freq and freq2 are the frequencies at the beginning and end of the synthesis and the default frequency is 440Hz. If freq2 is given, length must also have been given and the generated tone is swept between the given frequencies. The two given frequencies must be separated by one of the characters `:', `+', `/' and `-', which specify the sweep function as follows:

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

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

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

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

    The frequency or frequency range is not used for the noise types. offset is the bias (DC offset) of the signal in percent; default=0. phase is the phase shift as a percentage of 1 cycle with a default of 0 (not used for noise). For example, the following produces a 3-second 48kHz audio file containing a sine wave swept from 300 to 3300Hz:
       sox_ng -n output.wav synth 3 sine 300-3300 Multiple channels can be synthesized by specifying the set of parameters shown between curly braces multiple times; the following puts the swept tone in the left channel and brown noise in the right:
       sox_ng -n output.wav synth 3 sine 300-3300 brownnoise The following example shows how two synth effects can be cascaded to create a more complex waveform:
       play_ng -n synth 0.5 sine 200-500 synth 0.5 sine fmod 700-100 The following could be used to help tune a guitar:
       for n in E2 A2 D3 G3 B3 E4; do
        play_ng -n synth 4 pluck $n repeat 2; done

tempo [-q] [-m|s|l] factor [segment(82) [search(14.68) [overlap(12)]]]
    Change the audio playback speed but not its pitch. This effect uses the WSOLA (Waveform Similarity OverLap and Add) 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 the 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. This makes the effect work more quickly, but the result may not sound as good. However, if you must improve the processing speed, this generally reduces the sound quality less than reducing the search or overlap values. The -m option is used to optimize the default values of segment, search and overlap for music processing. The -s option is used to optimize default values of segment, search and overlap for speech processing. The -l option is used to optimize default values of segment, search and overlap for `linear' processing that tends to cause more noticeable distortion but may be useful when factor is close to 1. If -m, -s or -l is specified, the default value of segment is based on factor, while default search and overlap values are based on segment. Any values you provide override these default values. factor gives the ratio of new tempo to the old tempo, so 1.1 speeds the tempo up by 10% and 0.9 slows it down by 10%. The optional segment parameter selects the algorithm's segment size in milliseconds. If no other flags are specified, the default value is 82, which is suited to small changes in the tempo of music. For larger changes (e.g. a factor of 2), 41 may give a better result. The -m, -s, and -l flags cause segment's default value to be adjusted automatically based on factor. The optional search parameter gives the audio length in milliseconds over which the algorithm searches for overlapping points. If no other flags are specified, the default value is 14.68. Larger values use more processing time and may or may not produce better results. A practical maximum is half the value of segment. Search can be reduced to cut processing time at the risk of degrading output quality. The -m, -s and -l flags cause the search default to be adjusted automatically based on segment. The optional overlap parameter gives the segment overlap length in milliseconds. Its default value is 12 but the -m, -s and -l flags automatically adjust it based on the segment size. Increasing overlap increases processing time but may increase quality. A practical maximum for overlap is a little less then search. Note that lowering the tempo increases the sample rate and this can make following effects slower, in particular tempo or pitch themselves, whose running times are proportional to the sample rate times the overlap, all squared. This can be compensated for by following tempo with a fast rate effect. See speed for an effect that changes tempo and pitch together, pitch and bend for effects that change pitch only and stretch for an effect that changes the tempo using a different algorithm.

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

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

trim {position(+)}
    Cuts out portions of the audio. Any number of positions may be given; audio is not sent to the output until the first position is reached. The effect then alternates between copying and discarding audio at each position. Using a value of 0 for the first position parameter allows copying from the beginning of the audio. For example,
       sox_ng in.au out.au trim 0 10 copies the first ten seconds, while
       play_ng in.au trim 12:34 =15:00 -2:00 and
       play_ng in.au trim 12:34 2:26 -2:00 both play from 12 minutes 34 seconds into the audio up to 15 minutes in (i.e. 2 minutes and 26 seconds long) then resume playing two minutes before the end. SoX has an internal speed hack which, when trim is the first effect and removes audio from the beginning, seeks in the audio file instead of decoding it and throwing the data away but this is only used when the input is a single file. To achieve fast gapless playing with multiple files and trimming the first, you can use something like:
       FMT='-t s32 -r 44100 -c2'
       (sox file1.mp3 $FMT - trim 30 && sox file2.mp3 $FMT -) | play $FMT -

upsample [factor(2)]
    Upsample the signal by an integer factor: factor-1 zero-valued samples are inserted between each pair of input samples. As a result, the original spectrum is replicated into the new frequency space and attenuated. This attenuation can be compensated for by adding vol factor. The upsample effect is typically used in combination with filtering effects. For a general resampling effect with antialiasing, see rate. See downsample.

vad [options]
    The Voice Activity Detector attempts to trim silence and quiet background sounds from the ends of (fairly high resolution i.e. 16-bit, 44-48kHz) recordings of speech. The algorithm currently uses a simple cepstral power measurement to detect voice, so may be fooled by other things, especially music. The effect can trim only from the front of the audio, so in order to trim from the back, the reverse effect must also be used. E.g.
       play_ng speech.wav norm vad to trim from the front,
       play_ng speech.wav norm reverse vad reverse to trim from the back and
       play_ng speech.wav norm vad reverse vad reverse to trim from both ends. The use of the norm effect is recommended, but remember that neither reverse nor norm is suitable for use with streamed audio.

    Options
    Default values are shown in parentheses, the allowed range in square brackets.

    -t num (7) [0 - 20]
        The measurement level used to trigger activity detection. This might need to be changed depending on the noise level, signal level and other characteristics of the input audio.

    -T num (0.25) [0.01 - 1]
        The time constant (in seconds) used to help ignore short bursts of sound.

    -s num (1) [0.1 - 4]
        The amount of audio (in seconds) to search for quieter/shorter bursts of audio to include prior to the detected trigger point.

    -g num (0.25) [0.1 - 1]
        Allowed gap (in seconds) between quieter/shorter bursts of audio to include prior to the detected trigger point.

    -p num (0) [0 - 4]
        The amount of audio (in seconds) to preserve before the trigger point and any found quieter/shorter bursts.
There are keymaps on trigger_level, trigger_time and gap.

    Advanced Options
    These allow fine tuning of the algorithm's internal parameters.

    -b num (0.35) [0.1 - 10]
        The algorithm uses adaptive noise estimation/reduction in order to detect the start of the wanted audio. This option sets the time in seconds for the initial noise estimate.

    -N num (0.1) [0.1 - 10]
        Time constant used by the adaptive noise estimator when the noise level is increasing.

    -n num (0.01) [0.001 - 0.1]
        Time constant used by the adaptive noise estimator when the noise level is decreasing.

    -r num (1.35) [0 - 2]
        Amount of noise reduction to use in the detection algorithm.

    -f num (20) [5 - 50]
        Frequency of the algorithm's processing/measurements.

    -m num (0.1) [0.01 - 1]
        Measurement duration. By default, it is twice the measurement period; i.e. with 50% overlap, but if you set -f, you also need to change -m to 2 divided by its value to keep a 50% overlap.

    -M num (0.4) [0-1 - 1]
        Time constant used to smooth spectral measurements.

    -h freq (50) [10 -]
        `Brick-wall' frequency of the high-pass filter applied at the detector algorithm's input.

    -l freq (6000) [1000 -]
        `Brick-wall' frequency of the low-pass filter applied at the detector algorithm's input.

    -H freq (150) [10 -]
        `Brick-wall' frequency of the high-pass lifter used in the detector algorithm.

    -L freq (2000) [1000 -]
        `Brick-wall' frequency of the low-pass lifter used in the detector algorithm.

    See the silence effect.

vol gain [type [limiter-gain]]
    Apply amplification or attenuation to the audio signal. Unlike -v, 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 in the processing chain and several times if necessary. 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), gain is an amplitude ratio (voltage or linear), if power, a power ratio (wattage or voltage squared) and if dB, 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 limiter-gain 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 causes no limiter to be used. In verbose mode, this effect displays the percentage of the audio that needed to be limited. There is a keymap on vol.gain, which is adjusted in the units that were specified (amplitude, dB or power). See gain for a volume-changing effect with different capabilities and compand for a dynamic range compression/expansion/limiting effect.

REFERENCES

[1] R. Bristow-Johnson, Cookbook formulae for audio EQ biquad filter coefficients,
    https://www.w3.org/TR/audio-eq-cookbook

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

[3] Scott Lehman, Effects Explained,
    https://codeberg.org/sox_ng/Effects-Explained

[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/overview.html

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

SEE ALSO
sox_ng(1).

AUTHORS
Lance Norskog, Chris Bagwell and many other authors and contributors listed in the README file that is distributed with the source code.