sonusai 0.20.3__py3-none-any.whl → 1.0.2__py3-none-any.whl

sonusai/mixture/sox_help.py

@@ -0,0 +1,650 @@
+def allpass() -> str:
+    return """
+allpass frequency[k] width[h|k|o|q]
+    Apply a two-pole all-pass filter with central frequency (in Hz) frequency,
+    and filter-width width. An all-pass filter changes the audio's frequency to
+    phase relationship without changing its frequency to amplitude relationship.
+"""
+
+
+def band() -> str:
+    return """
+band [-n] center[k] [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 center + width and center - width will be half of their original
+    amplitudes. band defaults to a mode oriented to pitched audio, i.e. voice,
+    singing, or instrumental music. The -n (for noise) option uses the alternate
+    mode for un-pitched audio (e.g. percussion). Warning: -n introduces a power-gain
+    of about 11 dB in the filter, so beware of output clipping. band introduces noise
+    in the shape of the filter, i.e. peaking at the center frequency and settling
+    around it.
+
+    See also sinc for a bandpass filter with steeper shoulders.
+"""
+
+
+def bandpass() -> str:
+    return """
+bandpass [-c] frequency[k] width[h|k|o|q]
+    Apply a two-pole Butterworth band-pass filter with central frequency frequency,
+    and (3 dB-point) band-width width. The -c option applies only to bandpass and
+    selects a constant skirt gain (peak gain = Q) instead of the default: constant
+    0 dB peak gain. The filters roll off at 6 dB per octave (20 dB per decade).
+
+    See also sinc for a bandpass filter with steeper shoulders.
+"""
+
+
+def bandreject() -> str:
+    return """
+bandreject [-c] frequency[k] width[h|k|o|q]
+    Apply a two-pole Butterworth band-reject filter with central frequency frequency,
+    and (3 dB-point) band-width width. The -c option applies only to bandpass and
+    selects a constant skirt gain (peak gain = Q) instead of the default: constant
+    0 dB peak gain. The filters roll off at 6 dB per octave (20 dB per decade).
+
+    See also sinc for a bandpass filter with steeper shoulders.
+"""
+
+
+def bass() -> str:
+    return """
+bass gain [frequency[k] [width[s|h|k|o|q]]]
+    Boost or cut the bass (lower) frequencies of the audio using a two-pole shelving
+    filter with a response similar to that of a standard hi-fi's tone-controls. This
+    is also known as shelving equalisation (EQ).
+
+    gain gives the gain at 0 Hz. Its useful range is about -20 (for a large cut) to
+    +20 (for a large boost). Beware of Clipping when using a positive gain.
+
+    If desired, the filter can be fine-tuned using the following optional parameters:
+
+    frequency sets the filter's central frequency and so can be used to extend or reduce
+    the frequency range to be boosted or cut. The default value is 100 Hz.
+
+    width determines how steep is the filter's shelf transition. In addition to the
+    common width specification methods described above, 'slope' (the default, or if
+    appended with 's') may be used. The useful range of 'slope' is about 0.3, for a
+    gentle slope, to 1 (the maximum), for a steep slope; the default value is 0.5.
+
+    See also equalizer for a peaking equalisation effect.
+"""
+
+
+def biquad() -> str:
+    return """
+biquad b0 b1 b2 a0 a1 a2
+    Apply a biquad IIR filter with the given coefficients. Where b* and a* are the numerator
+    and denominator coefficients respectively.
+
+    See https://en.wikipedia.org/wiki/Digital_biquad_filter (where a0 = 1).
+"""
+
+
+def chorus() -> str:
+    return """
+chorus gain-in gain-out <delay decay speed depth -s|-t>
+    Add a chorus effect to the audio. This can make a single vocal sound like a chorus,
+    but can also be applied to instrumentation.
+
+    Chorus resembles an echo effect with a short delay, but whereas with echo the delay
+    is constant, with chorus, it is varied using sinusoidal or triangular modulation. The
+    modulation depth defines the range the modulated delay is played before or after the
+    delay. Hence the delayed sound will sound slower or faster, that is the delayed sound
+    tuned around the original one, like in a chorus where some vocals are slightly off key.
+
+    Each four-tuple parameter delay/decay/speed/depth gives the delay in milliseconds and
+    the decay (relative to gain-in) with a modulation speed in Hz using depth in
+    milliseconds. The modulation is either sinusoidal (-s) or triangular (-t). Gain-out is
+    the volume of the output.
+
+    A typical delay is around 40 ms to 60 ms; the modulation speed is best near 0.25 Hz and
+    the modulation depth around 2 ms. For example, a single delay:
+
+    play guitar1.wav chorus 0.7 0.9 55 0.4 0.25 2 -t
+
+    Two delays of the original samples:
+
+    play guitar1.wav chorus 0.6 0.9 50 0.4 0.25 2 -t 60 0.32 0.4 1.3 -s
+
+    A fuller sounding chorus (with three additional delays):
+
+    play guitar1.wav chorus 0.5 0.9 50 0.4 0.25 2 -t 60 0.32 0.4 2.3 -t 40 0.3 0.3 1.3 -s
+"""
+
+
+def compand() -> str:
+    return """
+compand attack1,decay1{,attack2,decay2} [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2} [gain [initial-volume-dB [delay]]]
+    Compand (compress or expand) the dynamic range of the audio.
+
+    The attack and decay parameters (in seconds) determine the time over which the
+    instantaneous level of the input signal is averaged to determine its volume; attacks
+    refer to increases in volume and decays refer to decreases. For most situations, the
+    attack time (response to the music getting louder) should be shorter than the decay
+    time because the human ear is more sensitive to sudden loud music than sudden soft
+    music. Where more than one pair of attack/decay parameters are specified, each input
+    channel is companded separately and the number of pairs must agree with the number of
+    input channels. Typical values are 0.3,0.8 seconds.
+
+    The second parameter is a list of points on the compander's transfer function specified
+    in dB relative to the maximum possible signal amplitude. The input values must be in a
+    strictly increasing order but the transfer function does not have to be monotonically
+    rising. If omitted, the value of out-dB1 defaults to the same value as in-dB1; levels
+    below in-dB1 are not companded (but may have gain applied to them). The point 0,0 is
+    assumed but may be overridden (by 0,out-dBn). If the list is preceded by a soft-knee-dB
+    value, then the points at where adjacent line segments on the transfer function meet
+    will be rounded by the 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 permits the user to supply a nominal level initially, so that,
+    for example, a very large gain is not applied to initial signal levels before the
+    companding action has begun to operate: it is quite probable that in such an event,
+    the output would be severely clipped while the compander gain properly adjusts itself.
+    A typical value (for audio which is initially quiet) is -90 dB.
+
+    The fifth (optional) parameter is a delay in seconds. The input signal is analysed
+    immediately to control the compander, but it is delayed before being fed to the volume
+    adjuster. Specifying a delay approximately equal to the attack/decay times allows the
+    compander to effectively operate in a 'predictive' rather than a reactive mode. A
+    typical value is 0.2 seconds.
+
+    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 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 -70 dB) will
+    remain unchanged. This will stop the compander from boosting the volume on 'silent'
+    passages such as between movements. However, sounds in the range -60 dB to 0 dB (maximum
+    volume) will be boosted so that the 60 dB dynamic range of the original music will be
+    compressed 3-to-1 into a 20 dB range, which is wide enough to enjoy the music but narrow
+    enough to get around the road noise. The '6:' selects 6 dB 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) has the effect of causing the
+    compander to react a bit more quickly to sudden volume changes.
+
+    In the next example, compand is being used as a noise-gate for when the noise is at a
+    lower level than the signal:
+
+    play infile compand .1,.2 -inf,-50.1,-inf,-50,-50 0 -90 .1
+
+    Here is another noise-gate, this time for when the noise is at a higher level than the
+    signal (making it, in some ways, similar to squelch):
+
+    play infile compand .1,.1 -45.1,-45,-inf,0,-inf 45 -90 .1
+
+    See also mcompand for a multiple-band companding effect.
+"""
+
+
+def contrast() -> str:
+    return """
+contrast [enhancement-amount(75)]
+    Comparable with compression, this effect modifies an audio signal to make it sound louder.
+    enhancement-amount controls the amount of the enhancement and is a number in the range 0-100.
+    Note that enhancement-amount = 0 still gives a significant contrast enhancement.
+
+    See also the compand and mcompand effects.
+"""
+
+
+def dcshift() -> str:
+    return """
+dcshift shift [limitergain]
+    Apply a DC shift to the audio. This can be useful to remove a 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 limitergain can be specified as well. It should have a value much less than 1
+    (e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping.
+
+    An alternative approach to removing a DC offset (albeit with a short delay) is to use the
+    highpass filter effect at a frequency of say 10 Hz, as illustrated in the following example:
+
+    sox -n dc.wav synth 5 sin %0 50
+    sox dc.wav fixed.wav highpass 10
+"""
+
+
+def equalizer() -> str:
+    return """
+equalizer frequency[k] width[q|o|h|k] gain
+    Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at and
+    around a selected frequency can be increased or decreased, whilst (unlike band-pass and
+    band-reject filters) that at all other frequencies is unchanged.
+
+    frequency gives the filter's central frequency in Hz, width, the band-width, and gain the
+    required gain or attenuation in dB. Beware of Clipping when using a positive gain.
+
+    In order to produce complex equalisation curves, this effect can be given several times, each
+    with a different central frequency.
+
+    See also bass and treble for shelving equalisation effects.
+"""
+
+
+def flanger() -> str:
+    return """
+flanger [delay depth regen width speed shape phase interp]
+    Apply a flanging effect to the audio.
+
+    All parameters are optional (right to left).
+"""
+
+
+def gain() -> str:
+    return """
+gain [-e|-B|-b|-r] [-n] [-l|-h] [gain-dB]
+    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, or -n requires temporary file space to store
+    the audio to be processed, so may be unsuitable for use with 'streamed' audio.
+
+    Without other options, gain-dB is used to adjust the signal power level by the given number of
+    dB: positive amplifies (beware of Clipping), negative attenuates. With other options, the
+    gain-dB amplification or attenuation is (logically) applied after the processing due to those
+    options.
+
+    Given the -e option, the levels of the audio channels of a multi-channel file are 'equalised',
+    i.e. gain is applied to all channels other than that with the highest peak level, such that all
+    channels attain the same peak level (but, without also giving -n, the audio is not 'normalised').
+
+    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. Note, however, that in conjunction with
+    -n, -B and -b are synonymous.
+
+    The -r option is used in conjunction with a prior invocation of gain with the -h option - see below
+    for details.
+
+    The -n option normalises the audio to 0 dB FSD; it is often used in conjunction with a negative
+    gain-dB to the effect that the audio is normalised to a given level below 0 dB. For example,
+
+    sox infile outfile gain -n
+
+    normalises to 0 dB, and
+
+    sox infile outfile gain -n -3
+
+    normalises to -3 dB.
+
+    The -l option invokes a simple limiter, e.g.
+
+    sox infile outfile gain -l 6
+
+    will apply 6 dB of gain but never clip. 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 head-room for subsequent processing.
+    For example, with
+
+    sox infile outfile gain -h bass +6
+
+    6 dB of attenuation will be applied prior to the bass boosting effect thus ensuring that it will
+    not clip. Of course, with bass, it is obvious how much headroom will be needed, but with other
+    effects (e.g. rate, dither) it is not always as clear. 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 infile outfile 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 infile outfile gain -h bass +6 rate 44100 gain -rh dither
+
+    Here, the second gain invocation, reclaims as much of the headroom as it can from the preceding
+    effects, but retains as much headroom as is needed for subsequent processing.
+
+    See also the norm and vol effects.
+"""
+
+
+def highpass() -> str:
+    return """
+highpass [-1|-2] frequency[k] [width[q|o|h|k]]
+    Apply a high-pass filter with 3 dB 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 6 dB per pole per octave
+    (20 dB per pole per decade).
+
+    See also sinc for filters with a steeper roll-off.
+"""
+
+
+def hilbert() -> str:
+    return """
+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 bandpass characteristic, attenuating the lowest and
+    highest frequencies. Its bandwidth can be controlled by the number of filter taps, which can be
+    specified with -n. By default, the number of taps is chosen for a cutoff frequency of about 75 Hz.
+"""
+
+
+def loudness() -> str:
+    return """
+loudness [gain [reference]]
+    Loudness control - similar to the gain effect, but provides equalisation for the human auditory
+    system. See https://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 equalised
+    according to ISO 226 w.r.t. a reference level of 65 dB, though an alternative reference level
+    may be given if the original audio has been equalised for some other optimal level. A default
+    gain of -10 dB is used if a gain value is not given.
+
+    See also the gain effect.
+"""
+
+
+def lowpass() -> str:
+    return """
+lowpass [-1|-2] frequency[k] [width[q|o|h|k]]
+    Apply a low-pass filter with 3 dB 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 6 dB per pole per octave
+    (20 dB per pole per decade).
+
+    See also sinc for filters with a steeper roll-off.
+"""
+
+
+def mcompand() -> str:
+    return """
+mcompand "attack1,decay1{,attack2,decay2} [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2} [gain [initial-volume-dB [delay]]]" {crossover-freq[k] "attack1,..."}
+    The multi-band compander is similar to the single-band compander but the audio is first divided
+    into bands using Linkwitz-Riley cross-over filters and a separately specifiable compander run on
+    each band. See the compand effect for the definition of its parameters. Compand parameters are
+    specified between double quotes and the crossover frequency for that band is given by
+    crossover-freq; these can be repeated to create multiple bands.
+
+    For example, the following (one long) command shows how multi-band companding is typically used
+    in FM radio:
+
+    play track1.wav gain -3 sinc 8000- 29 100 mcompand \
+     "0.005,0.1 -47,-40,-34,-34,-17,-33" 100 \
+     "0.003,0.05 -47,-40,-34,-34,-17,-33" 400 \
+     "0.000625,0.0125 -47,-40,-34,-34,-15,-33" 1600 \
+     "0.0001,0.025 -47,-40,-34,-34,-31,-31,-0,-30" 6400 \
+     "0,0.025 -38,-31,-28,-28,-0,-25" \
+     gain 15 highpass 22 highpass 22 sinc -n 255 -b 16 -17500 \
+     gain 9 lowpass -1 17801
+
+    The audio file is played with a simulated FM radio sound (or broadcast signal condition if the
+    lowpass filter at the end is skipped). Note that the pipeline is set up with US-style 75 us
+    pre-emphasis.
+
+    See also compand for a single-band companding effect.
+"""
+
+
+def norm() -> str:
+    return """
+norm [dB-level]
+    Normalise the audio. norm is just an alias for gain -n; see the gain effect for details.
+"""
+
+
+def overdrive() -> str:
+    return """
+overdrive [gain(20) [colour(20)]]
+    Non linear distortion. The colour parameter controls the amount of even harmonic content in the
+    over-driven output.
+"""
+
+
+def phaser() -> str:
+    return """
+phaser gain-in gain-out delay decay speed [-s|-t]
+    Add a phasing effect to the audio.
+
+    delay/decay/speed gives the delay in milliseconds and the decay (relative to gain-in) with a
+    modulation speed in Hz. The modulation is either sinusoidal (-s) - preferable for multiple
+    instruments, or triangular (-t) - gives single instruments a sharper phasing effect. The decay
+    should be less than 0.5 to avoid feedback, and usually no less than 0.1. Gain-out is the
+    volume of the output.
+
+    For example:
+
+    play snare.flac phaser 0.8 0.74 3 0.4 0.5 -t
+
+    Gentler:
+
+    play snare.flac phaser 0.9 0.85 4 0.23 1.3 -s
+
+    A popular sound:
+
+    play snare.flac phaser 0.89 0.85 1 0.24 2 -t
+
+    More severe:
+
+    play snare.flac phaser 0.6 0.66 3 0.6 2 -t
+"""
+
+
+def pitch() -> str:
+    return """
+pitch [-q] shift [segment [search [overlap]]]
+    Change the audio pitch (but not tempo).
+
+    shift gives the pitch shift as positive or negative 'cents' (i.e. 100ths of a semitone). See
+    the tempo effect for a description of the other parameters.
+
+    See also the bend, speed, and tempo effects.
+"""
+
+
+def reverb() -> str:
+    return """
+reverb [-w|--wet-only] [reverberance (50%) [HF-damping (50%) [room-scale (100%) [stereo-depth (100%) [pre-delay (0 ms) [wet-gain (0 dB)]]]]]]
+    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 will usually make it sound more natural. See [3] for a detailed description of
+    reverberation.
+
+    Note that this effect increases both the volume and the length of the audio, so to prevent
+    clipping in these domains, a typical invocation might be:
+
+    play dry.wav gain -3 pad 0 3 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 -m voice.wav "|sox voice.wav -p reverse reverb -w reverse"
+
+    for a reverse reverb effect.
+"""
+
+
+def sinc() -> str:
+    return """
+sinc [-a att|-b beta] [-p phase|-M|-I|-L] [-t tbw|-n taps] [freqHP] [-freqLP [-t tbw|-n taps]]
+    Apply a sinc 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 6 dB points of a
+    high-pass and low-pass filter that may be invoked individually, or together. If both are
+    given, then freqHP less than freqLP creates a band-pass filter, 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 120 dB can be overridden with -a; alternatively, the
+    kaiser-window 'beta' parameter can be given directly with -b.
+
+    The default transition band-width 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.
+
+    If both freqHP and freqLP are given, then 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.
+"""
+
+
+def speed() -> str:
+    return """
+speed factor[c]
+    Adjust the audio speed (pitch and tempo together). factor is either the ratio of the new
+    speed to the old speed: greater than 1 speeds up, less than 1 slows down, or, if appended
+    with the letter 'c', the number of cents (i.e. 100ths of a semitone) by which the pitch
+    (and tempo) should be adjusted: greater than 0 increases, less than 0 decreases.
+
+    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 also the bend, pitch, and tempo effects.
+"""
+
+
+def tempo() -> str:
+    return """
+tempo [-q] [-m|-s|-l] factor [segment [search [overlap]]]
+    Change the audio playback speed but not its pitch. This effect uses the WSOLA algorithm. The
+    audio is chopped up into segments which are then shifted in the time domain and overlapped
+    (cross-faded) at points where their waveforms are most similar as determined by measurement
+    of 'least squares'.
+
+    By default, linear searches are used to find the best overlapping points. If the optional -q
+    parameter is given, tree searches are used instead. 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 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 will be calculated based on
+    factor, while default search and overlap values are based on segment. Any values you provide
+    still override these default values.
+
+    factor gives the ratio of new tempo to the old tempo, so e.g. 1.1 speeds up the tempo 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 and is typically suited to making small
+    changes to the tempo of music. For larger changes (e.g. a factor of 2), 41 ms may give a
+    better result. The -m, -s, and -l flags will cause the segment default to be automatically
+    adjusted based on factor. For example using -s (for speech) with a tempo of 1.25 will
+    calculate a default segment value of 32.
+
+    The optional search parameter gives the audio length in milliseconds over which the algorithm
+    will search 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 will cause the search
+    default to be automatically adjusted based on segment.
+
+    The optional overlap parameter gives the segment overlap length in milliseconds. Default
+    value is 12, but -m, -s, or -l flags automatically adjust overlap based on segment size.
+    Increasing overlap increases processing time and may increase quality. A practical maximum
+    for overlap is the value of search, with overlap typically being (at least) a little
+    smaller then search.
+
+    See also 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 tempo using a
+    different algorithm.
+"""
+
+
+def treble() -> str:
+    return """
+treble gain [frequency[k] [width[s|h|k|o|q]]]
+    Boost or cut the treble (upper) frequencies of the audio using a two-pole shelving
+    filter with a response similar to that of a standard hi-fi's tone-controls. This is
+    also known as shelving equalisation (EQ).
+
+    gain gives the gain at whichever is the lower of ~22 kHz 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.
+
+    If desired, the filter can be fine-tuned using the following optional parameters:
+
+    frequency sets the filter's central frequency and so can be used to extend or reduce
+    the frequency range to be boosted or cut. The default value is 3 kHz.
+
+    width determines how steep is the filter's shelf transition. In addition to the
+    common width specification methods described above, 'slope' (the default, or if
+    appended with 's') may be used. The useful range of 'slope' is about 0.3, for a
+    gentle slope, to 1 (the maximum), for a steep slope; the default value is 0.5.
+
+    See also equalizer for a peaking equalisation effect.
+"""
+
+
+def tremolo() -> str:
+    return """
+tremolo speed [depth]
+    Apply a tremolo (low frequency amplitude modulation) effect to the audio. The
+    tremolo frequency in Hz is given by speed, and the depth as a percentage by
+    depth (default 40).
+"""
+
+
+def vol() -> str:
+    return """
+vol gain [type [limitergain]]
+    Apply an amplification or an attenuation to the audio signal. Unlike the -v option
+    (which is used for balancing multiple input files as they enter the sox effects
+    processing chain), vol is an effect like any other so can be applied anywhere, and
+    several times if necessary, during the processing chain.
+
+    The amount to change the volume is given by gain which is interpreted, according
+    to the given type, as follows: if type is amplitude (or is omitted), then gain is
+    an amplitude (i.e. voltage or linear) ratio, if power, then a power (i.e. wattage
+    or voltage-squared) ratio, and if dB, then a power change in dB.
+
+    When type is amplitude or power, a gain of 1 leaves the volume unchanged, less
+    than 1 decreases it, and greater than 1 increases it; a negative gain inverts the
+    audio signal in addition to adjusting its volume.
+
+    When type is dB, a gain of 0 leaves the volume unchanged, less than 0 decreases
+    it, and greater than 0 increases it.
+
+    Beware of Clipping when the increasing the volume.
+
+    The gain and the type parameters can be concatenated if desired, e.g. vol 10 dB.
+
+    An optional limitergain value can be specified and should be a value much less
+    than 1 (e.g. 0.05 or 0.02) and is used only on peaks to prevent clipping. Not
+    specifying this parameter will cause no limiter to be used. In verbose mode,
+    this effect will display the percentage of the audio that needed to be limited.
+
+    See also gain for a volume-changing effect with different capabilities, and
+    compand for a dynamic-range compression/expansion/limiting effect.
+"""

sonusai/mixture/spectral_mask.py

@@ -1,5 +1,5 @@
-from sonusai.mixture.datatypes import AudioF
-from sonusai.mixture.datatypes import SpectralMask
+from ..datatypes import AudioF
+from ..datatypes import SpectralMask
 
 
 def apply_spectral_mask(audio_f: AudioF, spectral_mask: SpectralMask, seed: int | None = None) -> AudioF: