Steve Harris LADSPA Plugin Docs

[email protected]
2007-01-31

Contents
1 Preamble
1.1 What plugins? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.2 Where can I get them . . . . . . . . . . . . . . . . . . . . . . . . . .

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2 The plugins
2.1 Aliasing (alias, 1407) . . . . . . . . . . . . . . . . . . . . . .
2.2 Allpass delay line, noninterpolating (allpass_n, 1895) . . . . .
2.3 Allpass delay line, linear interpolation (allpass_l, 1896) . . . .
2.4 Allpass delay line, cubic spline interpolation (allpass_c, 1897)
2.5 AM pitchshifter (amPitchshift, 1433) . . . . . . . . . . . . . .
2.6 Simple amplifier (amp, 1181) . . . . . . . . . . . . . . . . . .
2.7 Analogue Oscillator (analogueOsc, 1416) . . . . . . . . . . .
2.8 Glame Bandpass Analog Filter (bandpass_a_iir, 1893) . . . .
2.9 Glame Bandpass Filter (bandpass_iir, 1892) . . . . . . . . . .
2.10 Bode frequency shifter (bodeShifter, 1431) . . . . . . . . . . .
2.11 Bode frequency shifter (CV) (bodeShifterCV, 1432) . . . . . .
2.12 Glame Butterworth X-over Filter (bwxover_iir, 1902) . . . . .
2.13 GLAME Butterworth Lowpass (buttlow_iir, 1903) . . . . . . .
2.14 GLAME Butterworth Highpass (butthigh_iir, 1904) . . . . . .
2.15 Chebyshev distortion (chebstortion, 1430) . . . . . . . . . . .
2.16 Comb Filter (comb, 1190) . . . . . . . . . . . . . . . . . . .
2.17 Comb delay line, noninterpolating (comb_n, 1889) . . . . . .
2.18 Comb delay line, linear interpolation (comb_l, 1887) . . . . .
2.19 Comb delay line, cubic spline interpolation (comb_c, 1888) . .
2.20 Comb Splitter (combSplitter, 1411) . . . . . . . . . . . . . . .
2.21 Constant Signal Generator (const, 1909) . . . . . . . . . . . .
2.22 Crossover distortion (crossoverDist, 1404) . . . . . . . . . . .
2.23 DC Offset Remover (dcRemove, 1207) . . . . . . . . . . . . .
2.24 Debug Plugin (debug, 1184) . . . . . . . . . . . . . . . . . .
2.25 Exponential signal decay (decay, 1886) . . . . . . . . . . . .
2.26 Decimator (decimator, 1202) . . . . . . . . . . . . . . . . . .
2.27 Declipper (declip, 1195) . . . . . . . . . . . . . . . . . . . .
2.28 Simple delay line, noninterpolating (delay_n, 1898) . . . . . .
2.29 Simple delay line, linear interpolation (delay_l, 1899) . . . . .

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2.30
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2.74
2.75

Simple delay line, cubic spline interpolation (delay_c, 1900) . . . . .

Delayorama (delayorama, 1402) . . . . . . . . . . . . . . . . . . . .
Diode Processor (diode, 1185) . . . . . . . . . . . . . . . . . . . . .
Audio Divider (Suboctave Generator) (divider, 1186) . . . . . . . . .
DJ EQ (mono) (dj_eq_mono, 1907) . . . . . . . . . . . . . . . . . .
DJ EQ (dj_eq, 1901) . . . . . . . . . . . . . . . . . . . . . . . . . .
DJ flanger (djFlanger, 1438) . . . . . . . . . . . . . . . . . . . . . .
Dyson compressor (dysonCompress, 1403) . . . . . . . . . . . . . .
Fractionally Addressed Delay Line (fadDelay, 1192) . . . . . . . . .
Fast Lookahead limiter (fastLookaheadLimiter, 1913) . . . . . . . . .
Flanger (flanger, 1191) . . . . . . . . . . . . . . . . . . . . . . . . .
FM Oscillator (fmOsc, 1415) . . . . . . . . . . . . . . . . . . . . . .
Foldover distortion (foldover, 1213) . . . . . . . . . . . . . . . . . .
Fast overdrive (foverdrive, 1196) . . . . . . . . . . . . . . . . . . . .
Frequency tracker (freqTracker, 1418) . . . . . . . . . . . . . . . . .
Gate (gate, 1410) . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Gate (gate, 1921) . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Stereo Gate (stereo_gate, 1922) . . . . . . . . . . . . . . . . . . . . .
Giant flange (giantFlange, 1437) . . . . . . . . . . . . . . . . . . . .
Gong model (gong, 1424) . . . . . . . . . . . . . . . . . . . . . . . .
Gong beater (gongBeater, 1439) . . . . . . . . . . . . . . . . . . . .
GSM simulator (gsm, 1215) . . . . . . . . . . . . . . . . . . . . . .
GVerb (gverb, 1216) . . . . . . . . . . . . . . . . . . . . . . . . . .
Hard Limiter (hardLimiter, 1413) . . . . . . . . . . . . . . . . . . . .
Harmonic generator (harmonicGen, 1220) . . . . . . . . . . . . . . .
Hermes Filter (hermesFilter, 1200) . . . . . . . . . . . . . . . . . . .
Glame Highpass Filter (highpass_iir, 1890) . . . . . . . . . . . . . .
Hilbert transformer (hilbert, 1440) . . . . . . . . . . . . . . . . . . .
Impulse convolver (imp, 1199) . . . . . . . . . . . . . . . . . . . . .
Nonbandlimited single-sample impulses (Frequency: Control) (impulse_fc, 1885) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Inverter (inv, 1429) . . . . . . . . . . . . . . . . . . . . . . . . . . .
Karaoke (karaoke, 1409) . . . . . . . . . . . . . . . . . . . . . . . .
Artificial latency (artificialLatency, 1914) . . . . . . . . . . . . . . .
L/C/R Delay (lcrDelay, 1436) . . . . . . . . . . . . . . . . . . . . . .
Lookahead limiter (lookaheadLimiter, 1435) . . . . . . . . . . . . . .
Lookahead limiter (fixed latency) (lookaheadLimiterConst, 1906) . .
Glame Lowpass Filter (lowpass_iir, 1891) . . . . . . . . . . . . . . .
LS Filter (lsFilter, 1908) . . . . . . . . . . . . . . . . . . . . . . . .
Matrix: MS to Stereo (matrixMSSt, 1421) . . . . . . . . . . . . . . .
Matrix Spatialiser (matrixSpatialiser, 1422) . . . . . . . . . . . . . .
Matrix: Stereo to MS (matrixStMS, 1420) . . . . . . . . . . . . . . .
Multiband EQ (mbeq, 1197) . . . . . . . . . . . . . . . . . . . . . .
Modulatable delay (modDelay, 1419) . . . . . . . . . . . . . . . . .
Multivoice Chorus (multivoiceChorus, 1201) . . . . . . . . . . . . .
Mags Notch Filter (notch_iir, 1894) . . . . . . . . . . . . . . . . . .
LFO Phaser (lfoPhaser, 1217) . . . . . . . . . . . . . . . . . . . . .
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2.76 4 x 4 pole allpass (fourByFourPole, 1218) . . . . . .

2.77 Auto phaser (autoPhaser, 1219) . . . . . . . . . . . .
2.78 Pitch Scaler (pitchScale, 1193) . . . . . . . . . . . .
2.79 Higher Quality Pitch Scaler (pitchScaleHQ, 1194) . .
2.80 Plate reverb (plate, 1423) . . . . . . . . . . . . . . .
2.81 Pointer cast distortion (pointerCastDistortion, 1910) .
2.82 Rate shifter (rateShifter, 1417) . . . . . . . . . . . .
2.83 Retro Flanger (retroFlange, 1208) . . . . . . . . . .
2.84 Reverse Delay (5s max) (revdelay, 1605) . . . . . . .
2.85 Ringmod with two inputs (ringmod_2i1o, 1188) . . .
2.86 Ringmod with LFO (ringmod_1i1o1l, 1189) . . . . .
2.87 Barrys Satan Maximiser (satanMaximiser, 1408) . .
2.88 SC1 (sc1, 1425) . . . . . . . . . . . . . . . . . . . .
2.89 SC2 (sc2, 1426) . . . . . . . . . . . . . . . . . . . .
2.90 SC3 (sc3, 1427) . . . . . . . . . . . . . . . . . . . .
2.91 SC4 (sc4, 1882) . . . . . . . . . . . . . . . . . . . .
2.92 SC4 mono (sc4m, 1916) . . . . . . . . . . . . . . .
2.93 SE4 (se4, 1883) . . . . . . . . . . . . . . . . . . . .
2.94 Wave shaper (shaper, 1187) . . . . . . . . . . . . . .
2.95 Signal sifter (sifter, 1210) . . . . . . . . . . . . . . .
2.96 Sine + cosine oscillator (sinCos, 1881) . . . . . . . .
2.97 Single band parametric (singlePara, 1203) . . . . . .
2.98 Sinus wavewrapper (sinusWavewrapper, 1198) . . . .
2.99 Smooth Decimator (smoothDecimate, 1414) . . . . .
2.100Mono to Stereo splitter (split, 1406) . . . . . . . . .
2.101Step Demuxer (stepMuxer, 1212) . . . . . . . . . . .
2.102Surround matrix encoder (surroundEncoder, 1401) .
2.103State Variable Filter (svf, 1214) . . . . . . . . . . . .
2.104Tape Delay Simulation (tapeDelay, 1211) . . . . . .
2.105Transient mangler (transient, 1206) . . . . . . . . . .
2.106Triple band parametric with shelves (triplePara, 1204)
2.107Valve saturation (valve, 1209) . . . . . . . . . . . .
2.108Valve rectifier (valveRect, 1405) . . . . . . . . . . .
2.109VyNil (Vinyl Effect) (vynil, 1905) . . . . . . . . . .
2.110Wave Terrain Oscillator (waveTerrain, 1412) . . . . .
2.111Crossfade (xfade, 1915) . . . . . . . . . . . . . . . .
2.112Crossfade (4 outs) (xfade4, 1917) . . . . . . . . . . .
2.113z-1 (zm1, 1428) . . . . . . . . . . . . . . . . . . . .
3 Licensing

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1 Preamble
1.1 What plugins?
This is the documentation for some plugins that I have written for the Linux Audio
Developers Simple Plugin Architecture. It is a nice audio plugin architecture with a
very easy learning curve.

1.2 Where can I get them

From the website at http://plugin.org.uk/.

2 The plugins
2.1 Aliasing (alias, 1407)
Simulates aliasing using nyquist frequency modulation. Produces wacky results if the
blocks arent even numbers of samples long.
Aliasing level
Controls the amount of simulated aliasing in the output.

2.2 Allpass delay line, noninterpolating (allpass_n, 1895)

Based on work by James McCartney in SuperCollider.
Max Delay (s)
Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.
Decay Time (s)
Time for the echoes to decay by 60 decibels. If this time is negative then the feedback
coefficient will be negative, thus emphasizing only odd harmonics at an octave lower.

2.3 Allpass delay line, linear interpolation (allpass_l, 1896)

Based on work by James McCartney in SuperCollider.

Max Delay (s)

Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.
Decay Time (s)
Time for the echoes to decay by 60 decibels. If this time is negative then the feedback
coefficient will be negative, thus emphasizing only odd harmonics at an octave lower.

2.4 Allpass delay line, cubic spline interpolation (allpass_c, 1897)

Based on work by James McCartney in SuperCollider.
Max Delay (s)
Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.
Decay Time (s)
Time for the echoes to decay by 60 decibels. If this time is negative then the feedback
coefficient will be negative, thus emphasizing only odd harmonics at an octave lower.

2.5 AM pitchshifter (amPitchshift, 1433)

This plugin works by running a single write pointer (monotonic) and two read pointers (pitchscaled) over a ringbuffer.The output is faded between the two readpointers
according to the sine of the distance from the write pointer. The design is based on the
mechanism of a mechanical pitchshifter I saw in the Gemeentemuseum in Den Haag,
though Im sure it is a common enough algorithm.
Pitch shift
The multiple of the output pitch, eg. 2.0 will increase the pitch by one octave.
Buffer size
The order of magnitude of the buffer size. Small buffers will sound fluttery, large
buffers will have flangy sounding echos.I recommend a buffer size of 3 for a reasonable
compromise, with wideband material at around 48KHz. For drums you might have to
lower it, and for voiced background noises it can go higher.
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2.6 Simple amplifier (amp, 1181)

Amps gain (dB)
Controls the gain of the input signal in dBs.

2.7 Analogue Oscillator (analogueOsc, 1416)

This plugin simulates the output you get from an analogue synths osciallators.You can
get a reasonable emualtion of a 303s square (for exmaple) if you set the warmth to
about 0.4 and the instability to about 0.05.The frequency is currently a control input,
and there is no interpolation, so if your host is using large block sieze it will sound
steppy.Im unsure whether to convert this to an audio input or inpolate the control in.
Waveform (1=sin, 2=tri, 3=squ, 4=saw)
The basic shape of the waveform is selected using this control:
Value
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4

Waveform
Sine
Triangle
Square
Saw

Frequency (Hz)
The frequency of the output (Hz).
Warmth
The degree of softening that is applied to the generted waveform, reduces the number
of harmonics in the output.
Instability
The degree of pitch instability of the output. Turning this too high with square and saw
waves will produce an anoying jittery sound, I want to fix this but it is tricky.

2.8 Glame Bandpass Analog Filter (bandpass_a_iir, 1893)

IIR bandpass filter modeled after an analog circuit. This filter was ported from the
glame multitrack editor to ladspa.

2.9 Glame Bandpass Filter (bandpass_iir, 1892)

IIR bandpass filter based using chebishev coefficients. The filter allows you to tweak
the number of stages used for filtering. Every stage adds two more poles, which leads
to a steeper dropoff. More stages need more CPU power. This filter was ported from
the glame multitrack editor to ladspa.

2.10 Bode frequency shifter (bodeShifter, 1431)

A Bode/Moog Frequency Shifter is a popular analogue synth module, it works by shifting all the frequencies of an input signal up or down by a specified frequency. This version shifts in noth directions at the same time as its almost no extra work and its often
useful to have both directions.It doesnt actually work in the same way as an analogue
Bode/Moog, which use Dome filters as the core, it uses a Hilbert Transformer, which is
much simpler to implemtent in digital systems. The output is very similar though, and
people are familiar with the name Bode.The theory of operation is pretty simple, and
uses some clever maths to cancel out the upper or lower sidebands of a ringmodulator
applied to the input signal. Read the source if you want more information. The Hilbert
Transformet coefficents came from mkfilter, the excellent filter calculator, available at
http://www-users.cs.york.ac.uk/~fisher/mkfilter/.
Frequency shift
Controls the frequency shift applied to the input signal, in Hz. Note, this is not a pitch
shift, so you not get natural sounding results out, it is an audio effect popular with
modular synthesists.

2.11 Bode frequency shifter (CV) (bodeShifterCV, 1432)

See the non CV version for information.This is more or less a copy of the Doepfer
A126, http://www.doepfer.de/a126.htm.
Shift CV
Controls the frequency shift applied to the input signal, in KHz.

2.12 Glame Butterworth X-over Filter (bwxover_iir, 1902)

Butterworth X-over filter

2.13 GLAME Butterworth Lowpass (buttlow_iir, 1903)

Butterworth lowpass filter
7

2.14 GLAME Butterworth Highpass (butthigh_iir, 1904)

Butterworth highpass filter

2.15 Chebyshev distortion (chebstortion, 1430)

This is an interesting distortion effect that is seeded from incoming signal envelope.
As the level of the signal rises more and more harmonics will for added to the output signal.The distortion control sets the sensitivity of the input.The effect eveolved
from some experiments between Tim Goetze and myself, apptempting to emulate valve
based guitar amp distortion. This was one of the failures, but it still makes an interesting noise.

2.16 Comb Filter (comb, 1190)

Band separation (Hz)
Controls the distance between the filters peaks.
Feedback
Feedback level, increases the distinctive wooshy phaser sound.

2.17 Comb delay line, noninterpolating (comb_n, 1889)

Based on work by James McCartney in SuperCollider.
Max Delay (s)
Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.
Decay Time (s)
Time for the echoes to decay by 60 decibels. If this time is negative then the feedback
coefficient will be negative, thus emphasizing only odd harmonics at an octave lower.

2.18 Comb delay line, linear interpolation (comb_l, 1887)

Based on work by James McCartney in SuperCollider.

Max Delay (s)

Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.
Decay Time (s)
Time for the echoes to decay by 60 decibels. If this time is negative then the feedback
coefficient will be negative, thus emphasizing only odd harmonics at an octave lower.

2.19 Comb delay line, cubic spline interpolation (comb_c, 1888)

Based on work by James McCartney in SuperCollider.
Max Delay (s)
Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.
Decay Time (s)
Time for the echoes to decay by 60 decibels. If this time is negative then the feedback
coefficient will be negative, thus emphasizing only odd harmonics at an octave lower.

2.20 Comb Splitter (combSplitter, 1411)

Divides the input up into two parts with frequency peaks at f Hz intervals, skewed by
f/2 Hz between the two outputs. Mixing the two outputs will get you exactly the input
signal.I generally use this trick to divide up an input signal, process the two halves
differently, then mix them again. It sounds pretty funky.
Band separation (Hz)
The distance between the frequency peaks.
Output 1
The sum output.

Output 2
The difference output.

2.21 Constant Signal Generator (const, 1909)

This plugin add an output DC offset at the given amplitude to the input signal. It has
no real use other than for debugging and in modular synths.
Signal amplitude
Controls the amplitude of the output signal.

2.22 Crossover distortion (crossoverDist, 1404)

This is a simulation of the distortion that happens in class B and AB power amps when
the signal crosses 0.For class B simulations the smooth value should be set to about
0.3 +/- 0.2 and for AB it should be set to near 1.0.
Crossover amplitude
Controls the point at which the output signal becomes linear.
Smoothing
Controls degree of smoothing of the crossover point.

2.23 DC Offset Remover (dcRemove, 1207)

Simply removes the DC (0 Hz) component from an audio signal, uses a high pass filter,
so has some side effects, but they should be minimal.

2.24 Debug Plugin (debug, 1184)

Prints some stats about the input stream to stdout. Not intended for general use.

2.25 Exponential signal decay (decay, 1886)

Based on work by James McCartney in SuperCollider.
Decay Time (s)
Time for the echoes to decay by 60 decibels.
10

2.26 Decimator (decimator, 1202)

Decimates (reduces the effective sample rate), and reduces the bit depth of the input
signal, allows non integer values for smooth transitions between clean and lofi signals.
Bit depth
The bit depth that the signal will be reduced to.
Sample rate (Hz)
The sample rate that the signal will be resampled at.

2.27 Declipper (declip, 1195)

Removes nasty clicks from input signals, not very kind to them though.This code came
from the music-dsp mailing list, but it was unattributed, if its yours, please drop me a
line and Ill credit you.

2.28 Simple delay line, noninterpolating (delay_n, 1898)

Based on work by James McCartney in SuperCollider.
Max Delay (s)
Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.

2.29 Simple delay line, linear interpolation (delay_l, 1899)

Based on work by James McCartney in SuperCollider.
Max Delay (s)
Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.

2.30 Simple delay line, cubic spline interpolation (delay_c, 1900)

Based on work by James McCartney in SuperCollider.
11

Max Delay (s)

Maximum delay. Used to set the delay buffer size upon activation. Cannot be modulated. Note that if you do not connect to this port before activation, it will default to 1
second.

2.31 Delayorama (delayorama, 1402)

Random seed
Controls the random numbers that will be used to stagger the delays and amplitudes if
random is turned up on them. Changing this forces the random values to be recalulated.
Input gain (dB)
Controls the gain of the input signal in dBs.
Feedback (%)
Controls the amount of output signal fed back into the input.
Number of taps
Controls the number of taps in the delay.
First delay (s)
The time of the first delay.
Delay range (s)
The time difference between the first and last delay.
Delay change
The scaling factor between one delay and the next.
Delay random (%)
The random factor applied to the delay.

12

Amplitude change
The scaling factor between one amplitude and the next.
Amplitude random (%)
The random factor applied to the amplitude.
Dry/wet mix
The level of delayed sound mixed into the output.

2.32 Diode Processor (diode, 1185)

Mangles the signal as if it had been passed through a diode rectifier network.You
should probably follow this with a DC offset remover, unless you want the offset.
Mode (0 for none, 1 for half wave, 2 for full wave)
The mode parameter is continuously variable from thru to half-wave rectification to
full-wave to silence.

2.33 Audio Divider (Suboctave Generator) (divider, 1186)

Reduces the period of the signal by the factor given, and makes it a square wave in
the process. Has some amplitude tracking capability, but not really useful on complex
signals.
Denominator
The factor the incoming frequency will be divided by.

2.34 DJ EQ (mono) (dj_eq_mono, 1907)

The design for this plugin is taken from the Allen & Heath Xone 32 DJ mixer. It was
suggested by Patrick Shirkey. Mono version requested by Adam King
Lo gain (dB)
Controls the gain of the low (100Hz) peak/dip band

13

Mid gain (dB)

Controls the gain of the mid (1000Hz) peak/dip band
Hi gain (dB)
Controls the gain of the high (10000Hz) shelf band

2.35 DJ EQ (dj_eq, 1901)

The design for this plugin is taken from the Allen & Heath Xone 32 DJ mixer. It was
suggested by Patrick Shirkey.
Lo gain (dB)
Controls the gain of the low (100Hz) peak/dip band
Mid gain (dB)
Controls the gain of the mid (1000Hz) peak/dip band
Hi gain (dB)
Controls the gain of the high (10000Hz) shelf band

2.36 DJ flanger (djFlanger, 1438)

This is a flanger which is more or less typical of DJ mising desks. Requested by Patrick
Shirkey.
LFO sync
When turned from off to on it resets the phase of the LFO back to the start of the cycle.
Used to sync the LFO to the track.
LFO period (s)
The cycle period of the LFO in seconds.
LFO depth (ms)
The maximum delay the LFO will use to flange, in milliseconds.
14

Feedback (%)
The amount of the delays output that is mixed back into the delay.

2.37 Dyson compressor (dysonCompress, 1403)

Peak limit (dB)
Controls the desired limit of the output signal in dBs.
Release time (s)
Controls the time taken for the compressor to relax its gain control over the input
signal.
Fast compression ratio
I have no clear idea what this controls.
Compression ratio
I have no clear idea what this controls.

2.38 Fractionally Addressed Delay Line (fadDelay, 1192)

A fixed ring buffer delay implementation. Has different dynamics to a normal delay,
more suitable for certain things.Changes in delay length are generally more pleasing,
but delays >2s long have reduced sound quality.
Delay (seconds)
The neutral delay time is 2 seconds. Times above 2 seconds will have reduced quality
and times below will have increased CPU usage.

2.39 Fast Lookahead limiter (fastLookaheadLimiter, 1913)

This is a limiter with an attack time of 5ms. It adds just over 5ms of lantecy to the
input signal, but it guatantees that there will be no signals over the limit, and tries to
get the minimum ammount of distortion.
Input gain (dB)
Gain that is applied to the input stage. Can be used to trim gain to bring it roughly
under the limit or to push the signal against the limit.
15

Limit (dB)
The maximum output amplitude. Peaks over this level will be attenuated as smoothly
as possible to bring them as close as possible to this level.
Release time (s)
The time taken for the limiters attenuation to return to 0 dBs
Attenuation (dB)
The current attenuation of the signal coming out of the delay buffer.

2.40 Flanger (flanger, 1191)

A digital flanger implementation. Uses a novel zero excursion, controlled bandwidth
modulation function, which should make the modulation less repetitive and noticable.This effect is similar in character to a phaser (see section 2.75). The main difference is that a phaser sounds more regular and stable.
Delay base (ms)
This is the offset from the input time that the detune delay moves around.10 is probably
a good starting value.
Max slowdown (ms)
This is the maximum delay that will be applied to the delayed signal, relative to the
dry signal.
LFO frequency (Hz)
This is the core frequency that the LFO will move at. The LFO isnt actually an
oscillator, but it does vary periodically.
Feedback
Feedback applied from the output to the input, increases the depth of the effect, but
makes it sound less like a real flanger.

16

2.41 FM Oscillator (fmOsc, 1415)

Waveform (1=sin, 2=tri, 3=squ, 4=saw)
Value
1
The shape of the waveform.
2
3
4

Waveform
Sine
Triangle
Square
Saw

Frequency (Hz)
The frequency of the output (in Hertz).

2.42 Foldover distortion (foldover, 1213)

Uses a sinwave approximation to simulate valve style foldover distortion.Probably
should have a DC offset remover on the output, but its not always necessary.
Drive
Controls the degree of distortion.
Skew
Controls the asymmetry of the waveform.

2.43 Fast overdrive (foverdrive, 1196)

A simple overdrive. Compresses the extreme peaks to make a sound similar to an
overdriven amplifier.
Drive level
Controls the point at which the signal starts to distort, and the degree of distortion.

2.44 Frequency tracker (freqTracker, 1418)

Tracking speed
This controls the level of damping applied to the output.High values will make the
frequency output jump around, low values will make it a bit slow to respond.

17

2.45 Gate (gate, 1410)

The parameters are copied from the Drawmer DS-201, but Ive never used one, so if
someone out there has one, please tell me if it behaves differently.
LF key filter (Hz)
Controls the cutoff of the low frequency filter (highpass).
HF key filter (Hz)
Controls the cutoff of the high frequency filter (lowpass).
Threshold (dB)
Controls the level at which the gate will open.
Attack (ms)
Controls the time the gate will take to open fully.
Hold (ms)
Controls the minimum time the gate will stay open for.
Decay (ms)
Controls the time the gate will take to close fully.
Range (dB)
Controls the difference between the gates open and closed state.
Output select (-1 = key listen, 0 = gate, 1 = bypass)
Controls output monitor. -1 is the output of the key filters (so you can check what is
being gated on). 0 is the normal, gated output. 1 is bypass mode.

2.46 Gate (gate, 1921)

The parameters are copied from the Drawmer DS-201, but Ive never used one, so if
someone out there has one, please tell me if it behaves differently.
18

LF key filter (Hz)

Controls the cutoff of the low frequency filter (highpass).
HF key filter (Hz)
Controls the cutoff of the high frequency filter (lowpass).
Key level (dB)
Shows the current level of the key.
Threshold (dB)
Controls the level at which the gate will open.
Attack (ms)
Controls the time the gate will take to open fully.
Hold (ms)
Controls the minimum time the gate will stay open for.
Decay (ms)
Controls the time the gate will take to close fully.
Range (dB)
Controls the difference between the gates open and closed state.
Output select (-1 = key listen, 0 = gate, 1 = bypass)
Controls output monitor. -1 is the output of the key filters (so you can check what is
being gated on). 0 is the normal, gated output. 1 is bypass mode.

2.47 Stereo Gate (stereo_gate, 1922)

Derived from Steve Harris gate plugin

19

LF key filter (Hz)

Controls the cutoff of the low frequency filter (highpass).
HF key filter (Hz)
Controls the cutoff of the high frequency filter (lowpass).
Key level (dB)
Shows the current level of the key.
Threshold (dB)
Controls the level at which the gate will open.
Attack (ms)
Controls the time the gate will take to open fully.
Hold (ms)
Controls the minimum time the gate will stay open for.
Decay (ms)
Controls the time the gate will take to close fully.
Range (dB)
Controls the difference between the gates open and closed state.
Output select (-1 = key listen, 0 = gate, 1 = bypass)
Controls output monitor. -1 is the output of the key filters (so you can check what is
being gated on). 0 is the normal, gated output. 1 is bypass mode.

2.48 Giant flange (giantFlange, 1437)

This is a fairly normal flanger but with excessivly long delay times. Requested by
Patrick Shirkey.To cut down the memory requirements the internal delay buffer noly
has 15bits or resolution, so there is no headroom, if you feed in signals over 0dB it will
clip the output. There is code to soften the effect of the clipping, but beware of it.
20

Double delay
doubles the length of the delays, this will reduce the sound quality.
LFO frequency 1 (Hz)
The delay of the first LFO in seconds.
Delay 1 range (s)
The delay range of the first LFO in seconds.
LFO frequency 2 (Hz)
The delay of the second LFO in seconds.
Delay 2 range (s)
The delay range of the second LFO in seconds.
Feedback
The amount of the delays output that is mixed back into the delay.
Dry/Wet level
The ammounts of the input and effect mixed to produce the output.

2.49 Gong model (gong, 1424)

A physical model of a metal gong.Based on Josep Comajuncosas gong explorer,
which was built in Sync Modular, it uses 8 linear waveguides with nonlinear filters
to model the gong surface.
Inner damping
Controls the degree of damping in the centre of the gong.
Outer damping
Controls the degree of damping on the edge of the gong.

21

Mic position
Controls the vertical position of the "microphone", 0 is the centre and 1 is the edge.
Inner size 1
The size of the upper, inner waveguide.
Inner stiffness 1 +
The stiffness of the gong against deflections in the positive direction.
Inner stiffness 1 The stiffness of the gong against deflections in the negative direction.
Inner size 2
The size of the right, inner waveguide.
Inner stiffness 2 +
The stiffness of the gong against deflections in the positive direction.
Inner stiffness 2 The stiffness of the gong against deflections in the negative direction.
Inner size 3
The size of the lower, inner waveguide.
Inner stiffness 3 +
The stiffness of the gong against deflections in the positive direction.
Inner stiffness 3 The stiffness of the gong against deflections in the negative direction.

22

Inner size 4
The size of the left, inner waveguide.
Inner stiffness 4 +
The stiffness of the gong against deflections in the positive direction.
Inner stiffness 4 The stiffness of the gong against deflections in the negative direction.
Outer size 1
The size of the upper right, outer waveguide.
Outer stiffness 1 +
The stiffness of the gong against deflections in the positive direction.
Outer stiffness 1 The stiffness of the gong against deflections in the negative direction.
Outer size 2
The size of the lower right, outer waveguide.
Outer stiffness 2 +
The stiffness of the gong against deflections in the positive direction.
Outer stiffness 2 The stiffness of the gong against deflections in the negative direction.
Outer size 3
The size of the lower left, outer waveguide.

23

Outer stiffness 3 +
The stiffness of the gong against deflections in the positive direction.
Outer stiffness 3 The stiffness of the gong against deflections in the negative direction.
Outer size 4
The size of the upper left, outer waveguide.
Outer stiffness 4 +
The stiffness of the gong against deflections in the positive direction.
Outer stiffness 4 The stiffness of the gong against deflections in the negative direction.

2.50 Gong beater (gongBeater, 1439)

A plugin to simulator the action of a beator on a gong surface, used to trigger the gong
physical model.It is triggered by an impulse on the input, eg. from a mic or piezo
placed near a solid surface, for an event sequencer.
Impulse gain (dB)
The gain of the input impulse mixed into the output, bringing this up allows you to
make the outputted strike more impulsive, but may reduce the gongyness of the resulting output sound.
Strike gain (dB)
The gain of the simulated pressure wave mixed into the output, bringing this up allows
you to make the outputted strike more pure. The final output level is also proprtional
to the amplitude of the trigger.
Strike duration (s)
The duration of the pressure wave used to simulate the action of the beater on the gong
surface. The logner the duration the more sonorus the resulting gong sound.
24

2.51 GSM simulator (gsm, 1215)

Encodes and decodes a signal using the GSM voice compression system. Has the
effect of making the signal sound like it is being sent over a European mobile phone
network.
Dry/wet mix
Controls the dry/wet mix, 0 will give you the dry signal (but with the appropriate
amount of delay), 1 will give you a totally wet signal.
Number of passes
The number of times the signal is sent through the encode/decode process. Increases
the CPU consumption almost linearly, and it will become more peaky so less friendly
to realtime processing.
Error rate (bits/block)
The number of simulated bits that get changed during the transmission process.I really
wanted to reduce the bandwidth to get that "shouting down a drainpipe" effect, but Im
not sure how the reduced bandwidth is dealt with by real phones. I suspect its heavily
patented technology.

2.52 GVerb (gverb, 1216)

A mono in, stereo out reverb implementation by Juhana Sadeharju (kouhia at nic.funet.fi).
I ported it to LADSPA and did some testing.Please contact Juhana directly regarding
any bugs you find. Paul Winkler recommends a good starting point for a "large hall"
as follows:
Parameter
Roomsize
Reverb time
Damping
Input bandwidth
Dry signal level
Early reflection level
Tail level

amp; Value
amp; 200
amp; 1.3
amp; 0.4
amp; 0.5
amp; 0
amp; -12
amp; 0

Roomsize (m)
The size of the room, in meters. Excessivly large, and excessivly small values will
make it sound a bit unrealistic.Values of around 30 sound good.

25

Reverb time (s)

Reverb decay time, in seconds. 7 is a good place to start.
Damping
This controls the high frequency damping (a lowpass filter), values near 1 will make it
sound very bright, values near 0 will make it sound very dark.
Input bandwidth
This is like a damping control for the input, it has a similar effect to the damping
control, but is subtly different.
Dry signal level (dB)
The amount of dry signal to be mixed with the reverberated signal.
Early reflection level (dB)
The quantity of early reflections (scatter reflections directly from the source). Think
of Lexicons ambiance patches.
Tail level (dB)
The level of the classic reverb tail reflections.

2.53 Hard Limiter (hardLimiter, 1413)

Brick hard limiter with residue mixer.
Wet level
Output level for limited signal.
Residue level
Output level for residue signal.

2.54 Harmonic generator (harmonicGen, 1220)

What does it do?
Allows you to add harmonics and remove the fundamental from any audio signal.
26

Known bugs
There is no bandwith limiting filter on the output, so it is easy to create excessively
high frequency harmonics that could cause aliasing problems. In practive this doesnt
seem to be a serious problem however.
Examples
There are many interesting effects you can achieve with sinewaves, one example is
producing bandlimited squarewaves from sinewaves. To do this set the parameters to
1, 0, -0.3333, 0, 0.2, 0, -0.14285, 0, 0.11111.To get a triangle like signal use 1, 0,
-0.3333, 0, -0.2, 0, -0.14285, 0, -0.11111.
Fundamental magnitude
The amplitude of the fundamental of the signal, reduce it to 0 to remove the base signal
altogether, or -1 to phase invert it.
2nd harmonic magnitude
The 2nd harmonic, its frequency is twice the frequency of the harmonic.Even harmonics add a distorted feel to the sound, valve (tube) amplifiers introduce distortions at all
the harmonics.
3rd harmonic magnitude
The 3rd harmonic, its frequency is three time the frequency of the fundamental.Transistor
amplifiers only introduce distortion into the odd harmonics.

2.55 Hermes Filter (hermesFilter, 1200)

This plugin is a simulation of a modern analogue synth called a Pro Tone, with some
extra features bolted on, like a crossover. I tried to make it as comprehensive as possible, without requiring ludicrous amounts of CPU juice.N.B. as far as I know, noone
has tried to use this (I certainly havent), so it may be full of bugs and what not. The
parameters are all undocumented, but there is a diagram of the routing on the website.
Without a custom interface however it would be very hard to use.Historical note: the
name is a bad pun, it comes from the name Hermes Trimegistus given to the Egyptian
god Thoth by the greeks, it means Thrice Blessed, or something similar.

27

2.56 Glame Highpass Filter (highpass_iir, 1890)

IIR highpass filter based using chebishev coefficients. The filter allows you to tweak
the number of stages used for filtering. Every stage adds two more poles, which leads
to a steeper dropoff. More stages need more CPU power. This filter was ported from
the glame multitrack editor to ladspa.

2.57 Hilbert transformer (hilbert, 1440)

A Hilbert Transformer phase shifts the input signal by 90degrees. It outputs the 90
degree phase shifted signal and the unshifted signal, both delayed by an equivlaent
ammountThis plugin was written for a demo at the LAD Meet in 2003.

2.58 Impulse convolver (imp, 1199)

This is a convolver for a set of fairly short impulses.
The set of impulses has to be compiled in, they are:
Id
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21

Impulse
Unit impulse (identity)
My flat (light natural reverb)
Yamaha Marshall stack simulator
Fender 68 Vibrolux (SM57 on axis)
Fender 68 Vibrolux (SM57 off axis)
Fender 68 Vibrolux (Audio-technica AT4050)
Fender 68 Vibrolux (Neumann U87)
Fender Bassman (SM57 on axis)
Fender Bassman (SM57 off axis)
Fender Bassman (Audio-technica AT4050)
Fender Bassman (Neumann U87)
Fender Superchamp (SM57 on axis)
Fender Superchamp (SM57 off axis)
Fender Superchamp (Audio-technica AT4050)
Fender Superchamp (Neumann U87)
Marshall JCM2000 (SM57 on axis)
Marshall JCM2000 (SM57 off axis)
Marshall Plexi (SM57 on axis)
Marshall Plexi (SM57 off axis)
Matchless Chieftan (SM57 on axis)
Matchless Chieftan (SM57 off axis)

The first three were quickly grabbed by me using jack_impulse_grabber, and the others
we collected by someone else, but unfortunately Ive lost his email address and cant
find him on the web :(

28

Impulse ID
Selects the impulse to convolve with. New impulses have to be compiled in.
High latency mode
If you are running with blocks that are not whole powers of two long, or you are
hearing distortion, try changing this to 1.
Gain (dB)
Controls the gain of the output signal in dBs.

2.59 Nonbandlimited single-sample impulses (Frequency: Control) (impulse_fc, 1885)

Based on work by James McCartney in SuperCollider.
Frequency (Hz)
Frequency for the impulses.

2.60 Inverter (inv, 1429)

A utility plugin that inverts the signal, also (wrongly) known as a 180 degree phase
shift.

2.61 Karaoke (karaoke, 1409)

Attempts to strip the vocals from a stereo signal.
Vocal volume (dB)
Controls the attenuation of the vocal (centre channel) in dBs.The greater the attenuation the greater the loss of stereo field.

2.62 Artificial latency (artificialLatency, 1914)

Reports its delay value as systemic latency. Does nothing else, *this is not a delay*.Can
be used to correct for latency between channels.

29

2.63 L/C/R Delay (lcrDelay, 1436)

This is a left/centre/right delay with feedback, based on the one in the Korg Trinity.
Requested by Marek Peteraj.
L delay (ms)
The delay of the left output in milliseconds.
L level
The level of the left output.
C delay (ms)
The delay of the centre output in milliseconds.
C level
The level of the centre output.
R delay (ms)
The delay of the right output in milliseconds.
R level
The level of the right output.
Feedback
The amount of the C delay output that is mixed back into the delay.
High damp (%)
The damping of the high frequencies in the feedback path.
Low damp (%)
The damping of the low frequencies in the feedback path.

30

Spread
The width of the stereo image.
Dry/Wet level
The ammounts of the input and effect mixed to produce the output.

2.64 Lookahead limiter (lookaheadLimiter, 1435)

Limit (dB)
The maximum output amplitude. Peaks over this level will be attenuated as smoothly
as possible to bring them as close as possible to this level.
Lookahead delay
The delay used by the lookahead predictor. The longer the delay the smoother the
limiting will be, but higher the latency.
Attenuation (dB)
The current attenuation of the signal coming out of the delay buffer.

2.65 Lookahead limiter (fixed latency) (lookaheadLimiterConst, 1906)

A lookahead limiter - similar to the original Lookahead Limiter, but with a constant
latency of around 150ms and a reduacued maximum lookahead time.
Limit (dB)
The maximum output amplitude. Peaks over this level will be attenuated as smoothly
as possible to bring them as close as possible to this level.
Lookahead time (s)
The lookahead time used by the lookahead predictor. The longer the time the smoother
the limiting will be, but will tend to make the changes in dynamic range more obvious.
Attenuation (dB)
The current limiting attenuation of the signal coming out of the delay buffer.
31

2.66 Glame Lowpass Filter (lowpass_iir, 1891)

IIR lowpass filter based using chebishev coefficients. The filter allows you to tweak
the number of stages used for filtering. Every stage adds two more poles, which leads
to a steeper dropoff. More stages need more CPU power. This filter was ported from
the glame multitrack editor to ladspa.

2.67 LS Filter (lsFilter, 1908)

This is a filter created for the LinkSampler project - its designed to closly follow the
filter used in giga sampler.
Filter type (0=LP, 1=BP, 2=HP)
The type of the filter, 0 for low pass, 1 for band pass, 2 for high pass.
Cutoff frequency (Hz)
Controls the frequency at which the filter starts to effect the audio signal.eg. a lowpass filter with a cutoff frequency of 1000 Hz will only let frequencies below 100 Hz
through.
Resonance
Creates a peak at the cutoff frequency, for the classic overdriven fileter sound. At high
vlaues the peak at the cutoff will overwhelm the filtered signal.

2.68 Matrix: MS to Stereo (matrixMSSt, 1421)

Width
The width of the dematrixed stereo field. 1 will give you normal width, 0 will make it
completely mono, < 1 will make it narrower and > 1 will make it wider.

2.69 Matrix Spatialiser (matrixSpatialiser, 1422)

A simple spatializer that lets you control the width of a stereo signal.
We convert it into a MS (mid/side) signal, manipulate the gain coefficients with a
constant-power panning function, and reconvert to left/right stereo.
mid = (i_lef t + i_right)/2
side = (i_lef t i_right)/2
width = (pi/4)..0..(pi/4)
o_lef t = mid cos(width + pi/4) o_right = side sin(width + pi/4)
shifted by pi/4, so that 0 is neutral.

32

Width
0 is neutral (unmodified signal) + 128 is side only (=very wide) - 128 is mid only
(=mono)

2.70 Matrix: Stereo to MS (matrixStMS, 1420)

2.71 Multiband EQ (mbeq, 1197)
This is a fairly typical multiband graphical equalizer. Its implemented using a FFT, so
it takes quite a lot of CPU power, but should have less phase effects than an equivalent
filter implementation.If the input signal is at too low a sample rate then the top bands
will be ignored, the highest useful band will always be a high shelf.

2.72 Modulatable delay (modDelay, 1419)

A delay whose tap can be modulated at audio rate.Requested by Matthias Nagorni at
LinuxTag 2002, in order to make a Leslie simulator.

2.73 Multivoice Chorus (multivoiceChorus, 1201)

This is an implementation of a Multivoice (as opposed to Multiscale) chorus algorithm.
Its uses a novel, sinc based noise interpolation method to produce a subtle modulation
law which makes it possible to get away with larger numbers of voices without the
metallic, artificial sound common in chorus effects.
Voice separation (ms)
The individual voices can either be running at the same base delay (set this to zero)
or staggered.Setting this to non-zero values can make the output sound richer, but will
make it sound grainy with some type of signal.
Detune (%)
The maximum amount that a voice will be detuned by. I recommend a value of 1, but
you may be able to get away with higher values if the signal is less harmonic.
LFO frequency (Hz)
The frequency that the detune effect will be modulated at. A matter of taste, for most
types of input lower will be more subtle.

33

Output attenuation (dB)

With large numbers of voices the output can become too high, so use this to trim the
amplitude to a more helpful level.

2.74 Mags Notch Filter (notch_iir, 1894)

IIR notch filter based using chebishev coefficients. The filter allows you to tweak the
number of stages used for filtering. Every stage adds two more poles, which leads to a
steeper dropoff. More stages need more CPU power.

2.75 LFO Phaser (lfoPhaser, 1217)

2.76 4 x 4 pole allpass (fourByFourPole, 1218)
2.77 Auto phaser (autoPhaser, 1219)
2.78 Pitch Scaler (pitchScale, 1193)
A pitch shifter implementation that scales the harmonics appropriately with the base
frequencies. It is an implementation of Stephen M. Sprenglers pitch scaler design. It
gives reasonable, general purpose results for small changes, but wont give Antares or
Eventide anything to worry about.The FFT block size and oversampling has been kept
at a minimum to keep the CPU usage low.

2.79 Higher Quality Pitch Scaler (pitchScaleHQ, 1194)

A pitch shifter implementation that scales the harmonics appropriately with the base
frequencies. It is an implementation of Stephen M. Sprenglers pitch scaler design. It
gives reasonable, general purpose results for small changes, but wont give Antares or
Eventide anything to worry about.The FFT block size and oversampling has been kept
at reasonable levels to keep the CPU usage low, but it is smoother than the other Pitch
Scaler.
Pitch co-efficient
The pitch scaling factor, a value of 2.0 will increase the pitch by one octave, etc.

2.80 Plate reverb (plate, 1423)

A physical model of a steel plate reverb.Based on Josep Comajuncosas gong model,
it uses 8 linear waveguides to model the plate.

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Reverb time
Controls the RT60 time of the reverb. Actually controls the size of the plate. The
mapping betwwen plate size and RT60 time is just a heuristic, so its not very accurate.
Damping
Controls the degree that the surface of the plate is damped.
Dry/wet mix
Controls the balance between the dry and wet signals.

2.81 Pointer cast distortion (pointerCastDistortion, 1910)

This distortion is created by treating the floating point repesentation of the input signal as a 0.32 1s complement fixedpoint integer. Its very unmusical but supprisingly
recognisable. Im not sure that its useful for anything, but it can make interesting
noises.
Effect cutoff freq (Hz)
Controls the frequencies that will be passed to the effect.
Dry/wet mix
Controls the ammount of distioning mixed into the output.

2.82 Rate shifter (rateShifter, 1417)

Stretches or compresses the input with a ringbuffer.Because of the ringbuffer you will
get stretches of silence or clicks when the read pointer passes the write pointer.The
ringbuffer is about 2.7-3.0s long, depending on the sample rate.Versions with variable
buffer sizes or declicking code would be easy (but a bit less efficient); shout if you
would find them useful.
Rate
The rate of the output signal; eg. 2.0 will double the speed. Negative numbers will
play backwards.Pretty much any value will work, but the ranges give what most people
are going to want to use. You can get some interesting sounds with very high numbers
(e.g. 2000).

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2.83 Retro Flanger (retroFlange, 1208)

A model of someone flanging the input.Models the tape saturation effects, and frequency smear of a manual flanger. The results are a slightly distorted, but more subtle
flanger sound that you get from a normal digial flanger.
Average stall (ms)
The average time difference between the two tapes, per stall
Flange frequency (Hz)
The rate the tape is stalled at.

2.84 Reverse Delay (5s max) (revdelay, 1605)

A reverse delay not really modelled on any existing one. Youll want to set the Crossfade Samples parameter to something reasonably small (but more than 20) for most
applications, but you can try larger values if you start getting clicking.
Dry Level (dB)
Controls the level of the dry input signal in dBs.
Wet Level (dB)
Controls the level of the delayed signal in dBs.

2.85 Ringmod with two inputs (ringmod_2i1o, 1188)

This is a simple 2 input ring modulator.It is important that the modulator input is
bounded to (-1, +1), otherwise you will get rubbish on the output.
Input
This is the audio input.
Modulator
This is the modulator input.

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2.86 Ringmod with LFO (ringmod_1i1o1l, 1189)

This is a simple ring modulator and LFO.

2.87 Barrys Satan Maximiser (satanMaximiser, 1408)

Formerly Stupid Compressor. Thanks to Matt Yee-King for the name.Compresses
signals with a stupidly short attack and decay, infinite ratio and hard knee. Not really
as a compressor, but good harsh (non-musical) distortion.
Decay time (samples)
Controls the envelope decay time.
Knee point (dB)
Controls the knee roll-off point, ie. the point above which the compression kicks in. 0
will have no effect, -90 will remove virtually all dynamic range.

2.88 SC1 (sc1, 1425)

An high quality, reasonably low CPU cost RMS compressor designed for musical
work.It has controls for the compression point, compression ratio and knee softness.
Attack time (ms)
The attack time in milliseconds.
Release time (ms)
The release time in milliseconds.
Threshold level (dB)
The point at which the compressor will start to kick in.
Ratio (1:n)
The gain reduction ratio used when the signal level exceeds the threshold.
Knee radius (dB)
The distance from the threshold where the knee curve starts.
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Makeup gain (dB)

Controls the gain of the makeup input signal in dBs.

2.89 SC2 (sc2, 1426)

A compressor with sidechain. Based on the code for SC1.
Attack time (ms)
The attack time in milliseconds.
Release time (ms)
The release time in milliseconds.
Threshold level (dB)
The point at which the compressor will start to kick in.
Ratio (1:n)
The gain reduction ratio used when the signal level exceeds the threshold.
Knee radius (dB)
The distance from the threshold where the knee curve starts.
Makeup gain (dB)
Controls the gain of the makeup input signal in dBs.

2.90 SC3 (sc3, 1427)

A stereo compressor with sidechain input. Based on the code for SC1.
Attack time (ms)
The attack time in milliseconds.

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Release time (ms)

The release time in milliseconds.
Threshold level (dB)
The point at which the compressor will start to kick in.
Ratio (1:n)
The gain reduction ratio used when the signal level exceeds the threshold.
Knee radius (dB)
The distance from the threshold where the knee curve starts.
Makeup gain (dB)
Controls the gain of the makeup input signal in dBs.
Chain balance
Controls the chain signal used, 0 = Left+right in, 1 = Sidechain.

2.91 SC4 (sc4, 1882)

A stereo compressor with variable envelope follower for RMS / peak behaviour. Based
on the code for SC1.
RMS/peak
The blanace between the RMS and peak envelope followers.RMS is generally better
for subtle, musical compression and peak is better for heavier, fast compression and
percussion.
Attack time (ms)
The attack time in milliseconds.
Release time (ms)
The release time in milliseconds.
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Threshold level (dB)

The point at which the compressor will start to kick in.
Ratio (1:n)
The gain reduction ratio used when the signal level exceeds the threshold.
Knee radius (dB)
The distance from the threshold where the knee curve starts.
Makeup gain (dB)
Controls the gain of the makeup input signal in dBs.
Amplitude (dB)
The level of the input signal, in decibels.
Gain reduction (dB)
The degree of gain reduction applied to the input signal, in decibels.

2.92 SC4 mono (sc4m, 1916)

A mono compressor with variable envelope follower for RMS / peak behaviour. Based
on the code for SC4.
RMS/peak
The blanace between the RMS and peak envelope followers.RMS is generally better
for subtle, musical compression and peak is better for heavier, fast compression and
percussion.
Attack time (ms)
The attack time in milliseconds.
Release time (ms)
The release time in milliseconds.
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Threshold level (dB)

The point at which the compressor will start to kick in.
Ratio (1:n)
The gain reduction ratio used when the signal level exceeds the threshold.
Knee radius (dB)
The distance from the threshold where the knee curve starts.
Makeup gain (dB)
Controls the gain of the makeup input signal in dBs.
Amplitude (dB)
The level of the input signal, in decibels.
Gain reduction (dB)
The degree of gain reduction applied to the input signal, in decibels.

2.93 SE4 (se4, 1883)

A stereo expander with variable envelope follower for RMS / peak behaviour. Based
on the code for SC1.
RMS/peak
The blanace between the RMS and peak envelope followers.
Attack time (ms)
The attack time in milliseconds.
Release time (ms)
The release time in milliseconds.

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Threshold level (dB)

The point at which the expander will start to kick in.
Ratio (1:n)
The gain expansion ratio used when the signal level exceeds the threshold.
Knee radius (dB)
The distance from the threshold where the knee curve starts.
Attenuation (dB)
Controls the gain of the output signal in dBs. Used to correct for excessive amplitude
caused by the extra dynamic range.
Amplitude (dB)
The level of the input signal, in decibels.
Gain expansion (dB)
The degree of gain expansion applied to the input signal, in decibels.

2.94 Wave shaper (shaper, 1187)

This plugin reshapes the wave by an exponential function, inspiration was taken from
the Nord module of the same name.If you are getting rubbish out then its probably
because the host isnt using the input/output range hints, which are very important for
this plugin.
Waveshape
Positive values have an expanding effect, and negative values have a compressing effect.

2.95 Signal sifter (sifter, 1210)

Sorts and mixes blocks of the input signal to give a "bumpy ramp" effect.Certain types
of input will produce silence on the output (mostly ones with only low frequency
components).This is a very odd effect, and doesnt really have any music applications,
but can produce some interesting noises.
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2.96 Sine + cosine oscillator (sinCos, 1881)

This is a simple oscillator that outputs sinewaves with a 90 degree phase shift between
them.The current implementation is very inefficient, but I will improve it later.
Base frequency (Hz)
The base frequency of the output waves.
Pitch offset
The pitch offset of the output waves. Final oscillator frequency is base + 2p itch.

2.97 Single band parametric (singlePara, 1203)

A single band of a parametric filter.
Gain (dB)
The attenuation/gain of the eq.
Frequency (Hz)
The centre frequency (ie. point of most/least attenuation).Beware of high values for
Frequency and Bandwidth, if the high pitch (Frequency * 2Bandwidth ) goes over half
the sample rate you will get aliasing.Note: if your host offers you a frequency range
between 0 and 0.4 then its not rendering the input parameter correctly, the input frequency will actually be that number multiplied by the sample rate (e.g. 44.1kHz).
Bandwidth (octaves)
The pitch difference from the centre before the attenuation has reached half the gain.

2.98 Sinus wavewrapper (sinusWavewrapper, 1198)

Produces an unusual distortion effect, for a more amp like tone, see the valve saturation
plugin (section 2.107).

2.99 Smooth Decimator (smoothDecimate, 1414)

Resample rate
The rate at which the output signal will be resampled
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Smoothing
The amount of smoothing on the output signal.

2.100 Mono to Stereo splitter (split, 1406)

Takes a mono input signal, and outputs it to both left and right channel, thus "stereophizing" it.

2.101 Step Demuxer (stepMuxer, 1212)

Inputs up to 8 signals and switches between them on the output when then signal on
the clock input goes high.This plugin is untested, and may not work.

2.102 Surround matrix encoder (surroundEncoder, 1401)

I havent been able to test this plugin, so there may be bugs. I have successfully tested
the algorithm, but the implementation is suspect.
What does it do?
It allows you to encode four channels of sound into a stereo compatible stream that
will be decoded by a Dolby1 Surround/Pro-Logic decoder into Left, Right, Center and
Surround signals.This is not a proper implementation of Pro-logic, there is no Dolby
B processing done on the surround channel, which would help, but would be using
Dolby intellectual property.
Caveats
Obviously you cant wedge four channels into two without loss, so something has to
give. You will probably notice significant cross-talk between the channels, but the
decoder should do cross-talk correction, which will help a lot. A side effect of this
is that it will make left-right panning unusual, the sources will dwell near the left
and right speakers and zip across the centre channel. Because of this it is only really
possible to master surround recordings through a pre-logic decoder. Do not attempt to
use a conventional 5 point surround multichannel setup, it behaves very differently.In
addition to this, output from this process is not entirly mono compatible, in mono
output the L, C and R will be preserved as per a stereo recording (centre will be mixed
equally), but the surround channel will be totally lost.Careful gain control on the output
is required, as the level of the output will be greater than the L and R inputs, but
different to the sum amplitude of the input signals.Widely panned reverb fed to the L
and R channels will often leak into the S channel, if this is not desired (often it is)
1

"Dolby" is a trademark of Dolby Laboratories.

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reduce the width of the stereo image.The encoding will survive some processes (eg.
copying to CD, MD etc.), but may not survive conversion to MP3 or recoding to tape
with azimuth errors.
Legalese
This is a four channel matrix encoder, it happens to be compatible with Dolby Surround Pro-Logic.This implementation does not convey a license nor imply a right under any patent, or any other industrial or intellectual property right of Dolby Laboratories.
L
Left channel input. Can be treated as per normal stereo recoding, except that the
speaker should be at -22.5 , rather than the normal stereo -30 .
R
Right channel input. As per left channel.
C
Center channel input. Will be directly in front of the listener, stereo and mono compatible.
S
Surround channel. Should sound from the rear speakers, may also leak into the left
and right. Has slight delay and bandwidth reduction (cut below 100 Hz, and above 7
KHz) for leakage and noise reduction and enhanced psychoacoustic effects.Not mono
compatible.

2.103 State Variable Filter (svf, 1214)

An oversampled state variable filter with a few tweaksQuite a nice State Variable Filter,
tends to be unstable with high resonance and Q values, but good when kept under
control.
Filter type (0=none, 1=LP, 2=HP, 3=BP, 4=BR, 5=AP)
Select between no filtering, low-pass, high-pass, band-pass, band-reject and all-pass.

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Filter freq
Cutoff frequency, beware of high values with low sample rates.
Filter Q
The filters Q, or cutoff slope.
Filter resonance
The filters resonance, sort of separate from Q but very related (implemented with
feedback).Do not use with the bandpass mode.

2.104 Tape Delay Simulation (tapeDelay, 1211)

Correctly models the tape motion and some of the smear effect, there is no simulation
fo the head saturation yet, as I dont have a good model of it. When I get one I will
add it.The way the tape accelerates and decelerates gives a nicer delay effect for many
purposes.

2.105 Transient mangler (transient, 1206)

2.106 Triple band parametric with shelves (triplePara, 1204)
Actually five bands of eq, but the first and last are locked to shelving filters.For details
see the single band parametric (section 2.97).

2.107 Valve saturation (valve, 1209)

A model of valve (tube) distortion, lacking some of the harmonics you would get in
a real tube amp, but sounds good nonetheless.Taken from Ragnar Bendiksens thesis:
http://www.notam02.no/~rbendiks/Diplom/Innhold.html.
Distortion level
How hard the signal is driven against the limit of the amplifier.
Distortion character
The hardness of the sound, low for soft, high for hard.

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2.108 Valve rectifier (valveRect, 1405)

Sag level
The level of power supply sag that will be caused by attacks.
Distortion
How harsh the distortion caused by the sag will be.

2.109 VyNil (Vinyl Effect) (vynil, 1905)

Year
The date of the recording/playback equipment to be simulated.
RPM
The rotational speed of the platter.
Surface warping
The degree of variation in height of the record surface.
Crackle
The number of scratches on the record surface.
Wear
The ammount of wear on the grooves.

2.110 Wave Terrain Oscillator (waveTerrain, 1412)

A Wave Terrain oscillator, taken from Curtis Roads example in The Computer Music
Tutorial.Inputs x and y move the cursor around on a 2D landscape "wavetable" that is
used to generate the output. The function used is z = (x - y) * (x - 1) * (x + 1) * (y - 1)
* (y + 1).

2.111 Crossfade (xfade, 1915)

Crossfade
Controls the degree to which the inputs are mixed into the output. A value of -1 means
that the output is just the A input, and a value of 1.0 means that it is just the B input.
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2.112 Crossfade (4 outs) (xfade4, 1917)

Crossfade
Controls the degree to which the inputs are mixed into the output. A value of -1 means
that the output is just the A input, and a value of 1.0 means that it is just the B input.

2.113 z-1 (zm1, 1428)

A plugin that implements the z 1 function (a single sample delay).

3 Licensing
All this code is available under the GNU Public Licence, see the file COPYING
included with the source for more detials.

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