soxeffect_ng - Man Page

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.

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

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.

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.

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.

See also centercut.
    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.
    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
themselves, 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 B-a)
-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).

Note that the number of sample frames per pixel column is an integer,
so graphs whose number of samples is not a multiple of the number of columns
may render shorter than the file or with funny time legends.
-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).

By default, the lowest frequency is 0Hz for a linear graph or
1Hz for a logarithmic graph and the highest is the Nyquist frequency;
if a single frequency is given with no colon, it sets the low 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 behavior.

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.

[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

sox_ng(1).

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

libsoxeffect_ng(3), sox_ng(1).

The man page soxeffect(7) is an alias of soxeffect_ng(7).