-- | Advanced Signal Processing module Csound.Opcode.Advanced ( ----------------------------------------------------- -- * Modulation and Distortion -- ** Frequency Modulation foscil, foscili, crossfm, crossfmi, crosspm, crosspmi, crossfmpm, crossfmpmi, -- ** Distortion and Wave Shaping distort, distort1, powershape, polynomial, chebyshevpoly, -- ** Flanging, Phasing, Phase Shaping flanger, harmon, phaser1, phaser2, pdclip, pdhalf, pdhalfy, -- ** Doppler Shift doppler, ----------------------------------------------------- -- * Convolution pconvolve, convolve, ftconv, dconv, ----------------------------------------------------- -- * FFT and Spectral Processing -- ** Realtime Analysis And Resynthesis pvsanal, pvstanal, pvsynth, pvsadsyn, -- ** Writing FFT Data To A File And Reading From It pvswrite, pvsfread, pvsdiskin, -- ** FFT Info pvsinfo, pvsbin, pvscent, -- ** Manipulating FFT Signals pvscale, pvshift, pvsbandp, pvsbandr, pvsmix, pvscross, pvsfilter, pvsvoc, pvsmorph, pvsfreeze, pvsmaska, pvsblur, pvstencil, pvsarp, pvsmooth, ----------------------------------------------------- -- * Physical Models and FM Instruments -- ** Waveguide Physical Modelling streson, pluck, repluck, wgbow, wgbowedbar, wgbrass, wgclar, wgflute, wgpluck, wgpluck2, wguide1, wguide2, -- ** FM Instrument Models fmb3, fmbell, fmmetal, fmpercfl, fmrhode, fmvoice, fmwurlie, -- ** PhISEM opcodes -- | PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. bamboo, cabasa, crunch, dripwater, guiro, sandpaper, sekere, sleighbells, stix, tambourine, -- ** Some Perry Cook's instruments -- | The method is a physically inspired model developed from Perry Cook, but re-coded for Csound. gogobel, marimba, shaker, vibes, -- ** Other Models barmodel, mandol, moog, voice ) where import Csound.Exp import Csound.Exp.Wrapper import Csound.LowLevel ----------------------------------------------------- -- * Modulation and Distortion ----------------------------------------------------- -- ** Frequency Modulation -- | A basic frequency modulated oscillator. -- -- > ares foscil xamp, kcps, xcar, xmod, kndx, ifn [, iphs] -- -- doc: <http://www.csounds.com/manual/html/foscil.html> foscil :: Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Sig foscil = opc6 "foscil" [(a, [x, k, x, x, k, i, i])] -- | Basic frequency modulated oscillator with linear interpolation. -- -- > ares foscili xamp, kcps, xcar, xmod, kndx, ifn [, iphs] -- -- doc: <http://www.csounds.com/manual/html/foscili.html> foscili :: Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Sig foscili = opc6 "foscili" [(a, [x, k, x, x, k, i, i])] -- | Two oscillators, mutually frequency and/or phase modulated by each other. -- -- > a1, a2 crossfm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] -- > a1, a2 crossfmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] -- > a1, a2 crosspm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] -- > a1, a2 crosspmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] -- > a1, a2 crossfmpm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] -- > a1, a2 crossfmpmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] -- -- doc: <http://www.csounds.com/manual/html/crossfm.html> crossfmGen :: Name -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> (Sig, Sig) crossfmGen name = mopc7 name ([a, a], [x, x, x, x, k, i, i, i, i]) crossfm, crossfmi, crosspm, crosspmi, crossfmpm, crossfmpmi :: Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> (Sig, Sig) crossfm = crossfmGen "crossfm" crossfmi = crossfmGen "crossfmi" crosspm = crossfmGen "crosspm" crosspmi = crossfmGen "crosspmi" crossfmpm = crossfmGen "crossfmpm" crossfmpmi = crossfmGen "crossfmpmi" ----------------------------------------------------- -- ** Distortion and Wave Shaping -- | Distort an audio signal via waveshaping and optional clipping. -- -- > ar distort asig, kdist, ifn[, ihp, istor] -- -- doc: <http://www.csounds.com/manual/html/distort.html> distort :: Sig -> Sig -> Tab -> Sig distort = opc3 "distort" [(a, [a, k, i, i, i])] -- | Implementation of modified hyperbolic tangent distortion. distort1 can be used to generate wave shaping distortion based on a modification of the tanh function. -- -- > exp(asig * (shape1 + pregain)) - exp(asig * (shape2 - pregain)) -- > aout = --------------------------------------------------------------- -- > exp(asig * pregain) + exp(-asig * pregain) -- -- > ares distort1 asig, kpregain, kpostgain, kshape1, kshape2[, imode] -- -- doc: <http://www.csounds.com/manual/html/distort1.html> distort1 :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig distort1 = opc5 "distort1" [(a, [a, k, k, k, k, i])] -- | The powershape opcode raises an input signal to a power with pre- and post-scaling of the signal so that -- the output will be in a predictable range. It also processes negative inputs in a symmetrical way to positive inputs, calculating a dynamic transfer function that is useful for waveshaping. -- -- > aout powershape ain, kShapeAmount [, ifullscale] -- -- doc: <http://www.csounds.com/manual/html/powershape.html> powershape :: Sig -> Sig -> Sig powershape = opc2 "powershape" [(a, [a, k, i])] -- | The polynomial opcode calculates a polynomial with a single a-rate input variable. The polynomial is a sum -- of any number of terms in the form kn*x^n where kn is the nth coefficient of the expression. These coefficients are k-rate values. -- -- > aout polynomial ain, k0 [, k1 [, k2 [...]]] -- -- doc: <http://www.csounds.com/manual/html/polynomial.html> polynomial :: Sig -> [Sig] -> Sig polynomial a1 a2 = opcs "polynomial" [(a, a:repeat k)] (a1:a2) -- | The chebyshevpoly opcode calculates the value of a polynomial expression with a single a-rate input variable that is -- made up of a linear combination of the first N Chebyshev polynomials of the first kind. Each Chebyshev polynomial, Tn(x), -- is weighted by a k-rate coefficient, kn, so that the opcode is calculating a sum of any number of terms in the form kn*Tn(x). -- Thus, the chebyshevpoly opcode allows for the waveshaping of an audio signal with a dynamic transfer function that gives -- precise control over the harmonic content of the output. -- -- > aout chebyshevpoly ain, k0 [, k1 [, k2 [...]]] -- -- doc: <http://www.csounds.com/manual/html/chebyshevpoly.html> chebyshevpoly :: Sig -> [Sig] -> Sig chebyshevpoly a1 a2 = opcs "chebyshevpoly" [(a, a:repeat k)] (a1:a2) ----------------------------------------------------- -- ** Flanging, Phasing, Phase Shaping -- | A user controlled flanger. -- -- > ares flanger asig, adel, kfeedback [, imaxd] -- -- doc: <http://www.csounds.com/manual/html/flanger.html> flanger :: Sig -> Sig -> Sig -> Sig flanger = opc3 "flanger" [(a, [a, a, k, i])] -- | Analyze an audio input and generate harmonizing voices in synchrony. -- -- > ares harmon asig, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, imode, \ -- > iminfrq, iprd -- -- doc: <http://www.csounds.com/manual/html/harmon.html> harmon :: Sig -> Sig -> Sig -> Sig -> Sig -> D -> D -> D -> Sig harmon = opc8 "harmon" [(a, [a, k, k, k, k, i, i, i])] -- | An implementation of iord number of first-order allpass filters in series. -- -- > ares phaser1 asig, kfreq, kord, kfeedback [, iskip] -- -- doc: <http://www.csounds.com/manual/html/phaser1.html> phaser1 :: Sig -> Sig -> Sig -> Sig -> Sig phaser1 = opc4 "phaser1" [(a, [a, k, k, k, i])] -- | An implementation of iord number of second-order allpass filters in series. -- -- > ares phaser2 asig, kfreq, kq, kord, kmode, ksep, kfeedback -- -- doc: <http://www.csounds.com/manual/html/phaser2.html> phaser2 :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Sig phaser2 = opc7 "phaser2" [(a, [a, k, k, k, k, k, k])] -- | The pdclip opcode allows a percentage of the input range of a signal to be clipped to fullscale. -- It is similar to simply multiplying the signal and limiting the range of the result, but pdclip -- allows you to think about how much of the signal range is being distorted instead of the scalar -- factor and has a offset parameter for assymetric clipping of the signal range. pdclip is also useful -- for remapping phasors for phase distortion synthesis. -- -- > aout pdclip ain, kWidth, kCenter [, ibipolar [, ifullscale]] -- -- doc: <http://www.csounds.com/manual/html/pdclip.html> pdclip :: Sig -> Sig -> Sig -> Sig pdclip = opc3 "pdclip" [(a, [a, k, k, i, i])] -- | The pdhalf opcode is designed to emulate the "classic" phase distortion synthesis method of the -- Casio CZ-series of synthesizers from the mid-1980's. This technique reads the first and second halves -- of a function table at different rates in order to warp the waveform. For example, pdhalf can smoothly -- transform a sine wave into something approximating the shape of a saw wave. -- -- > aout pdhalf ain, kShapeAmount [, ibipolar [, ifullscale]] -- -- doc: <http://www.csounds.com/manual/html/pdhalf.html> pdhalf :: Sig -> Sig -> Sig -> Sig pdhalf = opc3 "pdhalf" [(a, [a, k, k, i, i])] -- | The pdhalfy opcode is a variation on the phase distortion synthesis method of the pdhalf opcode. -- It is useful for distorting a phasor in order to read two unequal portions of a table in the same number of samples. -- -- > aout pdhalfy ain, kShapeAmount [, ibipolar [, ifullscale]] -- -- doc: <http://www.csounds.com/manual/html/pdhalfy.html> pdhalfy :: Sig -> Sig -> Sig -> Sig pdhalfy = opc3 "pdhalfy" [(a, [a, k, k, i, i])] ----------------------------------------------------- -- ** Doppler Shift -- | A fast and robust method for approximating sound propagation, achieving convincing -- Doppler shifts without having to solve equations. The method computes frequency shifts -- based on reading an input delay line at a delay time computed from the distance between -- source and mic and the speed of sound. One instance of the opcode is required for each -- dimension of space through which the sound source moves. If the source sound moves at -- a constant speed from in front of the microphone, through the microphone, to behind -- the microphone, then the output will be frequency shifted above the source frequency -- at a constant frequency while the source approaches, then discontinuously will be -- shifted below the source frequency at a constant frequency as the source recedes from -- the microphone. If the source sound moves at a constant speed through a point to one -- side of the microphone, then the rate of change of position will not be constant, and -- the familiar Doppler frequency shift typical of a siren or engine approaching and -- receding along a road beside a listener will be heard. -- -- > ashifted doppler asource, ksourceposition, kmicposition [, isoundspeed, ifiltercutoff] -- -- doc: <http://www.csounds.com/manual/html/doppler.html> doppler :: Sig -> Sig -> Sig -> Sig doppler = opc3 "doppler" [(a, [a, k, k, i, i])] ----------------------------------------------------- -- * Granular Synthesis ----------------------------------------------------- -- * Convolution -- | Convolution based on a uniformly partitioned overlap-save algorithm. Compared to the convolve opcode, pconvolve has these benefits: -- -- * small delay -- -- * possible to run in real-time for shorter impulse files -- -- * no pre-process analysis pass -- -- * can often render faster than convolve -- -- -- > ar1 [, ar2] [, ar3] [, ar4] pconvolve ain, ifilcod [, ipartitionsize, ichannel] -- -- doc: <http://www.csounds.com/manual/html/pconvolve.html> pconvolve :: CsdTuple a => Sig -> Str -> a pconvolve = mopc2 "pconvolve" ([a,a,a,a], [a,s,i,i]) -- | Output is the convolution of signal ain and the impulse response contained in ifilcod. If more than one -- output signal is supplied, each will be convolved with the same impulse response. Note that it is -- considerably more efficient to use one instance of the operator when processing a mono input to create stereo, or quad, outputs. -- -- > ar1 [, ar2] [, ar3] [, ar4] convolve ain, ifilcod [, ichannel] -- -- doc: <http://www.csounds.com/manual/html/convolve.html> convolve :: CsdTuple a => Sig -> D -> a convolve = mopc2 "convolve" ([a, a, a, a], [a, i, i]) -- | Low latency multichannel convolution, using a function table as impulse response source. The algorithm is to split the -- impulse response to partitions of length determined by the iplen parameter, and delay and mix partitions so that the original, -- full length impulse response is reconstructed without gaps. The output delay (latency) is iplen samples, and does not depend -- on the control rate, unlike in the case of other convolve opcodes. -- -- > a1[, a2[, a3[, ... a8]]] ftconv ain, ift, iplen[, iskipsamples \ -- > [, iirlen[, iskipinit]]] -- -- doc: <http://www.csounds.com/manual/html/ftconv.html> ftconv :: CsdTuple a => Sig -> Tab -> D -> a ftconv = mopc3 "ftconv" (as 8, [a,i,i,i,i,i]) -- | A direct convolution opcode. -- -- > ares dconv asig, isize, ifn -- -- doc: <http://www.csounds.com/manual/html/dconv.html> dconv :: Sig -> I -> Tab -> Sig dconv = opc3 "dconv" [(a, [a,i,i])] ----------------------------------------------------- -- * FFT and Spectral Processing ----------------------------------------------------- -- ** Realtime Analysis And Resynthesis -- | Generate an fsig from a mono audio source ain, using phase vocoder overlap-add analysis. -- -- > fsig pvsanal ain, ifftsize, ioverlap, iwinsize, iwintype [, iformat] [, iinit] -- -- doc: <http://www.csounds.com/manual/html/pvsanal.html> pvsanal :: Sig -> I -> I -> I -> I -> Spec pvsanal = opc5 "pvsanal" [(f, a:is 6)] -- | pvstanal implements phase vocoder analysis by reading function tables containing sampled-sound sources, with GEN01, and pvstanal will accept deferred allocation tables. -- -- This opcode allows for time and frequency-independent scaling. Time is advanced internally, but controlled by a tempo scaling parameter; when an onset is detected, -- timescaling is momentarily stopped to avoid smearing of attacks. The quality of the effect is generally improved with phase locking switched on. -- -- pvstanal will also scale pitch, independently of frequency, using a transposition factor (k-rate). -- -- > fsig pvstanal ktimescal, kamp, kpitch, ktab, [kdetect, kwrap, ioffset,ifftsize, ihop, idbthresh] -- -- doc: <http://www.csounds.com/manual/html/pvstanal.html> pvstanal :: Sig -> Sig -> Sig -> Sig -> Spec pvstanal = opc4 "pvstanal" [(f, ks 6 ++ is 4)] -- | Resynthesise phase vocoder data (f-signal) using a FFT overlap-add. -- -- > ares pvsynth fsrc, [iinit] -- -- doc: <http://www.csounds.com/manual/html/pvsynth.html> pvsynth :: Spec -> Sig pvsynth = opc1 "pvsynth" [(a, [f,i])] -- | Resynthesize using a fast oscillator-bank. -- -- > ares pvsadsyn fsrc, inoscs, kfmod [, ibinoffset] [, ibinincr] [, iinit] -- -- doc: <http://www.csounds.com/manual/html/pvsadsyn.html> pvsadsyn :: Spec -> I -> Sig -> Sig pvsadsyn = opc3 "pvsadsyn" [(a, [f,i,k,i,i,i])] ----------------------------------------------------- -- ** Writing FFT Data To A File And Reading From It -- | This opcode writes a fsig to a PVOCEX file (which in turn can be read by pvsfread or other programs that support PVOCEX file input). -- -- > pvsfwrite fsig, ifile -- -- doc: <http://www.csounds.com/manual/html/pvsfwrite.html> pvswrite :: Spec -> Str -> SE () pvswrite a1 a2 = se_ $ opc2 "pvswrite" [(x, [f,s])] a1 a2 -- | Create an fsig stream by reading a selected channel from a PVOC-EX analysis file loaded into memory, with frame -- interpolation. Only format 0 files (amplitude+frequency) are currently supported. The operation of this opcode -- mirrors that of pvoc, but outputs an fsig instead of a resynthesized signal. -- -- > fsig pvsfread ktimpt, ifn [, ichan] -- -- doc: <http://www.csounds.com/manual/html/pvsfread.html> pvsfread :: Sig -> Str -> Spec pvsfread = opc2 "pvsfread" [(f, [k,s,i])] -- | Create an fsig stream by reading a selected channel from a PVOC-EX analysis file, with frame interpolation. -- -- > fsig pvsdiskin SFname,ktscal,kgain[,ioffset, ichan] -- -- doc: <http://www.csounds.com/manual/html/pvsdiskin.html> pvsdiskin :: Str -> Sig -> Sig -> Spec pvsdiskin = opc3 "pvsdiskin" [(f, [s,k,k,i,i])] ----------------------------------------------------- -- ** FFT Info -- | Get format information about fsrc, whether created by an opcode such as pvsanal, or obtained from -- a PVOCEX file by pvsfread. This information is available at init time, and can be used to set parameters -- for other pvs opcodes, and in particular for creating function tables (e.g. for pvsftw), or setting -- the number of oscillators for pvsadsyn. -- -- > ioverlap, inumbins, iwinsize, iformat pvsinfo fsrc -- -- doc: <http://www.csounds.com/manual/html/pvsinfo.html> pvsinfo :: Spec -> (I, I, I, I) pvsinfo = mopc1 "pvsinfo" ([i,i,i,i], [f]) -- | Obtain the amp and freq values off a PVS signal bin as k-rate variables. -- -- > kamp, kfr pvsbin fsig, kbin -- -- doc: <http://www.csounds.com/manual/html/pvsbin.html> pvsbin :: Spec -> Sig -> (Sig, Sig) pvsbin = mopc2 "pvsbin" ([k,k], [f,k]) -- | Calculate the spectral centroid of a signal from its discrete Fourier transform. -- -- > kcent pvscent fsig -- -- doc: <http://www.csounds.com/manual/html/pvscent.html> pvscent :: Spec -> Sig pvscent = opc1 "pvscent" [(k, [f])] ----------------------------------------------------- -- ** Manipulating FFT Signals -- | Scale the frequency components of a pv stream, resulting in pitch shift. Output amplitudes can be optionally modified in order to attempt formant preservation. -- -- > fsig pvscale fsigin, kscal[, kkeepform, kgain, kcoefs] -- -- doc: <http://www.csounds.com/manual/html/pvscale.html> pvscale :: Spec -> Sig -> Spec pvscale = opc2 "pvscale" [(f, [f,k,k,k,k])] -- | Shift the frequency components of a pv stream, stretching/compressing its spectrum. -- -- > fsig pvshift fsigin, kshift, klowest[, kkeepform, igain, kcoefs] -- -- doc: <http://www.csounds.com/manual/html/pvshift.html> pvshift :: Spec -> Sig -> Sig -> Spec pvshift = opc3 "pvshift" [(f, [f,k,k,k,i,k])] -- | Filter the pvoc frames, passing bins whose frequency is within a band, and with linear interpolation for transitional bands. -- -- > fsig pvsbandp fsigin, xlowcut, xlowfull, \ -- > xhighfull, xhighcut[, ktype] -- -- doc: <http://www.csounds.com/manual/html/pvsbandp.html> pvsbandp :: Spec -> Sig -> Sig -> Sig -> Sig -> Spec pvsbandp = opc5 "pvsbandp" [(f, [f,x,x,x,x,k])] -- | Filter the pvoc frames, rejecting bins whose frequency is within a band, and with linear interpolation for transitional bands. -- -- > fsig pvsbandr fsigin, xlowcut, xlowfull, \ -- > xhighfull, xhighcut[, ktype] -- -- doc: <http://www.csounds.com/manual/html/pvsbandr.html> pvsbandr :: Spec -> Sig -> Sig -> Sig -> Sig -> Spec pvsbandr = opc5 "pvsbandr" [(f, [f,x,x,x,x,k])] -- | Mix 'seamlessly' two pv signals. This opcode combines the most prominent components of two pvoc streams into a single mixed stream. -- -- > fsig pvsmix fsigin1, fsigin2 -- -- doc: <http://www.csounds.com/manual/html/pvsmix.html> pvsmix :: Spec -> Spec -> Spec pvsmix = opc2 "pvsmix" [(f, [f,f])] -- | Performs cross-synthesis between two source fsigs. -- -- > fsig pvscross fsrc, fdest, kamp1, kamp2 -- -- doc: <http://www.csounds.com/manual/html/pvscross.html> pvscross :: Spec -> Spec -> Sig -> Sig -> Spec pvscross = opc4 "pvscross" [(f, [f,f,k,k])] -- | Multiply amplitudes of a pvoc stream by those of a second pvoc stream, with dynamic scaling. -- -- > fsig pvsfilter fsigin, fsigfil, kdepth[, igain] -- -- doc: <http://www.csounds.com/manual/html/pvsfilter.html> pvsfilter :: Spec -> Spec -> Sig -> Spec pvsfilter = opc3 "pvsfilter" [(f, [f,f,k,i])] -- | This opcode provides support for cross-synthesis of amplitudes and frequencies. It takes the amplitudes -- of one input fsig and combines with frequencies from another. It is a spectral version of the well-known channel vocoder. -- -- > fsig pvsvoc famp, fexc, kdepth, kgain [,kcoefs] -- -- doc: <http://www.csounds.com/manual/html/pvsvoc.html> pvsvoc :: Spec -> Spec -> Sig -> Sig -> Spec pvsvoc = opc4 "pvsvoc" [(f, [f,f,k,k,k])] -- | Performs morphing (or interpolation) between two source fsigs. -- -- > fsig pvsmorph fsrc, fdest, kamp1, kamp2 -- -- doc: <http://www.csounds.com/manual/html/pvsmotph.html> pvsmorph :: Spec -> Spec -> Sig -> Sig -> Spec pvsmorph = opc4 "pvsmorph" [(f, [f,f,k,k])] -- | This opcodes 'freezes' the evolution of pvs stream by locking into steady amplitude and/or frequency -- values for each bin. The freezing is controlled, independently for amplitudes and frequencies, by a -- control-rate trigger, which switches the freezing 'on' if equal to or above 1 and 'off' if below 1. -- -- > fsig pvsfreeze fsigin, kfreeza, kfreezf -- -- doc: <http://www.csounds.com/manual/html/pvsfreeze.html> pvsfreeze :: Spec -> Sig -> Sig -> Sig pvsfreeze = opc3 "pvsfreeze" [(f, [f,k,k])] -- | Modify amplitudes of fsrc using function table, with dynamic scaling. -- -- > fsig pvsmaska fsrc, ifn, kdepth -- -- doc: <http://www.csounds.com/manual/html/pvsmaska.html> pvsmaska :: Spec -> Tab -> Sig -> Spec pvsmaska = opc3 "pvsmaska" [(f, [f,i,k])] -- | Average the amp/freq time functions of each analysis channel for a specified time (truncated to number of frames). As a side-effect the input pvoc stream will be delayed by that amount. -- -- > fsig pvsblur fsigin, kblurtime, imaxdel -- -- doc: <http://www.csounds.com/manual/html/pvsblur.html> pvsblur :: Spec -> Sig -> D -> Spec pvsblur = opc3 "pvsblur" [(f, [f,k,i])] -- | Transforms a pvoc stream according to a masking function table; if the pvoc stream amplitude -- falls below the value of the function for a specific pvoc channel, it applies a gain to that channel. -- -- The pvoc stream amplitudes are compared to a masking table, if the fall below the table values, they -- are scaled by kgain. Prior to the operation, table values are scaled by klevel, which can be used as masking depth control. -- -- Tables have to be at least fftsize/2 in size; for most GENS it is important to use an extended-guard -- point (size power-of-two plus one), however this is not necessary with GEN43. -- -- One of the typical uses of pvstencil would be in noise reduction. A noise print can be analysed with -- pvanal into a PVOC-EX file and loaded in a table with GEN43. This then can be used as the masking table -- for pvstencil and the amount of reduction would be controlled by kgain. Skipping post-normalisation will -- keep the original noise print average amplitudes. This would provide a good starting point for a successful -- noise reduction (so that klevel can be generally set to close to 1). -- -- Other possible transformation effects are possible, such as filtering and `inverse-masking'. -- -- > fsig pvstencil fsigin, kgain, klevel, iftable -- -- doc: <http://www.csounds.com/manual/html/pvstencil.html> pvstencil :: Spec -> Sig -> Sig -> Tab -> Sig pvstencil = opc4 "pvstencil" [(f, [f,k,k,i])] -- | This opcode arpeggiates spectral components, by amplifying one bin and attenuating all the others around -- it. Used with an LFO it will provide a spectral arpeggiator similar to Trevor Wishart's CDP program specarp. -- -- > fsig pvsarp fsigin, kbin, kdepth, kgain -- -- doc: <http://www.csounds.com/manual/html/pvsarp.html> pvsarp :: Spec -> Sig -> Sig -> Sig -> Spec pvsarp = opc4 "pvsarp" [(f, [f,k,k,k])] -- | Smooth the amplitude and frequency time functions of a pv stream using a 1st order lowpass IIR with time-varying -- cutoff frequency. This opcode uses the same filter as the tone opcode, but this time acting separately on the amplitude -- and frequency time functions that make up a pv stream. The cutoff frequency parameter runs at the control-rate, but unlike -- tone and tonek, it is not specified in Hz, but as fractions of 1/2 frame-rate (actually the pv stream sampling rate), -- which is easier to understand. This means that the highest cutoff frequency is 1 and the lowest 0; the lower the -- frequency the smoother the functions and more pronounced the effect will be. -- -- These are filters applied to control signals so the effect is basically blurring the spectral evolution. The effects -- produced are more or less similar to pvsblur, but with two important differences: 1.smoothing of amplitudes and -- frequencies use separate sets of filters; and 2. there is no increase in computational cost when higher amounts -- of "blurring" (smoothing) are desired. -- -- > fsig pvsmooth fsigin, kacf, kfcf -- -- doc: <http://www.csounds.com/manual/html/pvsmooth.html> pvsmooth :: Spec -> Sig -> Sig -> Spec pvsmooth = opc3 "pvsmooth" [(f, [f,k,k])] ----------------------------------------------------- -- * Physical Models and FM Instruments ----------------------------------------------------- -- ** Waveguide Physical Modelling -- | An audio signal is modified by a string resonator with variable fundamental frequency. -- -- > ares streson asig, kfr, ifdbgain -- -- doc: <http://www.csounds.com/manual/html/streson.html> streson :: Sig -> Sig -> D -> Sig streson = opc3 "streson" [(a, [a,k,i])] -- | Audio output is a naturally decaying plucked string or drum sound based on the Karplus-Strong algorithms. -- -- > ares pluck kamp, kcps, icps, ifn, imeth [, iparm1] [, iparm2] -- -- doc: <http://www.csounds.com/manual/html/pluck.html> pluck :: Sig -> Sig -> D -> Tab -> I -> Sig pluck = opc5 "pluck" [(a, [k,k,i,i,i,i,i])] -- | repluck is an implementation of the physical model of the plucked string. A user can control the pluck point, -- the pickup point, the filter, and an additional audio signal, axcite. axcite is used to excite the 'string'. Based on the Karplus-Strong algorithm. -- -- > ares repluck iplk, kamp, icps, kpick, krefl, axcite -- -- doc: <http://www.csounds.com/manual/html/repluck.html> repluck :: D -> Sig -> D -> Sig -> Sig -> Sig -> Sig repluck = opc6 "repluck" [(a, [i,k,i,k,k,a])] -- | Audio output is a tone similar to a bowed string, using a physical model developed from Perry Cook, but re-coded for Csound. -- -- > ares wgbow kamp, kfreq, kpres, krat, kvibf, kvamp, ifn [, iminfreq] -- -- doc: <http://www.csounds.com/manual/html/wgbow.html> wgbow :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Sig wgbow = opc7 "wgbow" [(a, ks 6 ++ is 2)] -- | A physical model of a bowed bar, belonging to the Perry Cook family of waveguide instruments. -- -- > ares wgbowedbar kamp, kfreq, kpos, kbowpres, kgain [, iconst] [, itvel] \ -- > [, ibowpos] [, ilow] -- -- doc: <http://www.csounds.com/manual/html/wgbowedbar.html> wgbowedbar :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig wgbowedbar = opc5 "wgbowedbar" [(a, ks 5 ++ is 2)] -- | Audio output is a tone related to a brass instrument, using a physical model developed from Perry Cook, but re-coded for Csound. -- -- > ares wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn [, iminfreq] -- -- doc: <http://www.csounds.com/manual/html/wgbrass.html> wgbrass :: Sig -> Sig -> Sig -> D -> Sig -> Sig -> Tab -> Sig wgbrass = opc7 "wgbrass" [(a, [k,k,k,i,k,k,i,i])] -- | Audio output is a tone similar to a clarinet, using a physical model developed from Perry Cook, but re-coded for Csound. -- -- > ares wgclar kamp, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, ifn \ -- > [, iminfreq] -- -- doc: <http://www.csounds.com/manual/html/wgclar.html> wgclar :: Sig -> Sig -> Sig -> D -> D -> Sig -> Sig -> Sig -> Tab -> Sig wgclar = opc9 "wgclar" [(a, [k,k,k,i,i,k,k,k,i,i])] -- | Audio output is a tone similar to a flute, using a physical model developed from Perry Cook, but re-coded for Csound. -- -- > ares wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, ifn \ -- > [, iminfreq] [, ijetrf] [, iendrf] -- -- doc: <http://www.csounds.com/manual/html/wgflute.html> wgflute :: Sig -> Sig -> Sig -> D -> D -> Sig -> Sig -> Sig -> Tab -> Sig wgflute = opc9 "wgflute" [(a, [k,k,k,i,i,k,k,k,i,i,i,i])] -- | A high fidelity simulation of a plucked string, using interpolating delay-lines. -- -- > ares wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite -- -- doc: <http://www.csounds.com/manual/html/wgpluck.html> wgpluck :: D -> D -> Sig -> D -> D -> D -> Sig -> Sig wgpluck = opc7 "wgplusk" [(a, [i,i,k,i,i,i,a])] -- | wgpluck2 is an implementation of the physical model of the plucked string, with control over the pluck point, the pickup point and the filter. Based on the Karplus-Strong algorithm. -- -- > ares wgpluck2 iplk, kamp, icps, kpick, krefl -- -- doc: <http://www.csounds.com/manual/html/wgpluck2.html> wgpluck2 :: D -> Sig -> D -> Sig -> Sig -> Sig wgpluck2 = opc5 "wgpluck2" [(a, [i,k,i,k,k])] -- | A simple waveguide model consisting of one delay-line and one first-order lowpass filter. -- -- > ares wguide1 asig, xfreq, kcutoff, kfeedback -- -- doc: <http://www.csounds.com/manual/html/wguide1.html> wguide1 :: Sig -> Sig -> Sig -> Sig -> Sig wguide1 = opc4 "wguide1" [(a, [a,x,k,k])] -- | A model of beaten plate consisting of two parallel delay-lines and two first-order lowpass filters. -- -- > ares wguide2 asig, xfreq1, xfreq2, kcutoff1, kcutoff2, \ -- > kfeedback1, kfeedback2 -- -- doc: <http://www.csounds.com/manual/html/wguide2.html> wguide2 :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Sig wguide2 = opc7 "wguide2" [(a, [a,x,x,k,k,k,k])] ---------------------------------------------------- -- ** FM Instrument Models fmGen :: Name -> Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig fmGen name = opc11 name [(a, ks 6 ++ is 6)] -- | Uses FM synthesis to create a Hammond B3 organ sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. -- -- > ares fmb3 kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ -- > ifn4, ivfn -- -- doc: <http://www.csounds.com/manual/html/fmb3.html> fmb3 :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig -- | Uses FM synthesis to create a tublar bell sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. -- -- > ares fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ -- > ifn4, ivfn[, isus] -- -- doc: <http://www.csounds.com/manual/html/fmbell.html> fmbell :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig -- | Uses FM synthesis to create a “Heavy Metal” sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. -- -- > ares fmmetal kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ -- > ifn4, ivfn -- -- doc: <http://www.csounds.com/manual/html/fmmetal.html> fmmetal :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig -- | Uses FM synthesis to create a percussive flute sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. -- -- > ares fmpercfl kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, \ -- > ifn3, ifn4, ivfn -- -- doc: <http://www.csounds.com/manual/html/fmpercfl.html> fmpercfl :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig -- | Uses FM synthesis to create a Fender Rhodes electric piano sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. -- -- > ares fmrhode kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, \ -- > ifn3, ifn4, ivfn -- -- doc: <http://www.csounds.com/manual/html/fmrhode.html> fmrhode :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig -- | FM Singing Voice Synthesis -- -- > ares fmvoice kamp, kfreq, kvowel, ktilt, kvibamt, kvibrate, ifn1, \ -- > ifn2, ifn3, ifn4, ivibfn -- -- doc: <http://www.csounds.com/manual/html/fmvoice.html> fmvoice :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig fmwurlie :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Tab -> Tab -> Sig fmb3 = fmGen "fmb3" fmbell = fmGen "fmbell" fmmetal = fmGen "fmmetal" fmpercfl = fmGen "fmpercfl" fmrhode = fmGen "fmrhode" fmvoice = fmGen "fmvoice" fmwurlie = fmGen "fmwurlie" -------------------------------------------------------------------- -- PhISEM percussion opcodes. -- | bamboo is a semi-physical model of a bamboo sound. -- -- > ares bamboo kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ -- > [, ifreq1] [, ifreq2] -- -- doc: <http://www.csounds.com/manual/html/bamboo.html> bamboo :: Sig -> D -> Sig bamboo = opc2 "bamboo" [(a, k:is 7)] -- | cabasa is a semi-physical model of a cabasa sound. -- -- > ares cabasa iamp, idettack [, inum] [, idamp] [, imaxshake] -- -- doc: <http://www.csounds.com/manual/html/cabasa.html> cabasa :: D -> D -> Sig cabasa = opc2 "cabasa" [(a, is 5)] -- | crunch is a semi-physical model of a crunch sound. -- -- > ares crunch iamp, idettack [, inum] [, idamp] [, imaxshake] -- -- doc: <http://www.csounds.com/manual/html/crunch.html> crunch :: D -> D -> Sig crunch = opc2 "crunch" [(a, is 5)] -- | dripwater is a semi-physical model of a water drop. -- -- > ares dripwater kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ -- > [, ifreq1] [, ifreq2] -- -- doc: <http://www.csounds.com/manual/html/dripwater.html> dripwater :: Sig -> D -> Sig dripwater = opc2 "dripwater" [(a, k:is 7)] -- | guiro is a semi-physical model of a guiro sound. -- -- > ares guiro kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] -- -- doc: <http://www.csounds.com/manual/html/guiro.html> guiro :: Sig -> D -> Sig guiro = opc2 "guiro" [(a, k:is 7)] -- | sandpaper is a semi-physical model of a sandpaper sound. -- -- > ares sandpaper iamp, idettack [, inum] [, idamp] [, imaxshake] -- -- doc: <http://www.csounds.com/manual/html/sandpaper.html> sandpaper :: D -> D -> Sig sandpaper = opc2 "sandpaper" [(a, is 5)] -- | sekere is a semi-physical model of a sekere sound. -- -- > ares sekere iamp, idettack [, inum] [, idamp] [, imaxshake] -- -- doc: <http://www.csounds.com/manual/html/sekere.html> sekere :: D -> D -> Sig sekere = opc2 "sekere" [(a, is 5)] -- | sleighbells is a semi-physical model of a sleighbell sound. -- -- > ares sleighbells kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ -- > [, ifreq1] [, ifreq2] -- -- doc: <http://www.csounds.com/manual/html/sleighbells.html> sleighbells :: Sig -> D -> Sig sleighbells = opc2 "sleighbells" [(a, k:is 7)] -- | stix is a semi-physical model of a stick sound. -- -- > ares stix iamp, idettack [, inum] [, idamp] [, imaxshake] -- -- doc: <http://www.csounds.com/manual/html/stix.html> stix :: D -> D -> Sig stix = opc2 "stix" [(a, is 5)] -- | tambourine is a semi-physical model of a tambourine sound. -- -- > ares tambourine kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ -- > [, ifreq1] [, ifreq2] -- -- doc: <http://www.csounds.com/manual/html/tambourine.html> tambourine :: Sig -> D -> Sig tambourine = opc2 "tambourine" [(a, k:is 7)] ------------------------------------------------------------- -- by Perry Cook -- | Audio output is a tone related to the striking of a cow bell or similar. -- -- > ares gogobel kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivfn -- -- doc: <http://www.csounds.com/manual/html/gogobel.html> gogobel :: Sig -> Sig -> D -> D -> D -> Sig -> Sig -> Tab -> Sig gogobel = opc8 "gogobel" [(a, [k,k,i,i,i,k,k,i])] -- | Audio output is a tone related to the striking of a wooden block as found in a marimba. -- -- > ares marimba kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec \ -- > [, idoubles] [, itriples] -- -- doc: <http://www.csounds.com/manual/html/marimba.html> marimba :: Sig -> Sig -> D -> D -> Tab -> Sig -> Sig -> Tab -> D -> Sig marimba = opc9 "marimba" [(a, [k,k,i,i,i,k,k,i,i,i,i])] -- | Audio output is a tone related to the shaking of a maraca or similar gourd instrument. -- -- > ares shaker kamp, kfreq, kbeans, kdamp, ktimes [, idecay] -- -- doc: <http://www.csounds.com/manual/html/shaker.html> shaker :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig shaker = opc5 "shaker" [(a, ks 5 ++ [i])] -- | Audio output is a tone related to the striking of a metal block as found in a vibraphone. -- -- > ares vibes kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec -- -- doc: <http://www.csounds.com/manual/html/vibes.html> vibes :: Sig -> Sig -> D -> D -> Tab -> Sig -> Sig -> Tab -> D -> Sig vibes = opc9 "vibes" [(a, [k,k,i,i,i,k,k,i,i])] --------------------------------------------------------------- -- other models -- | Audio output is a tone similar to a struck metal bar, using a physical model developed from solving the partial -- differential equation. There are controls over the boundary conditions as well as the bar characteristics. -- -- > ares barmodel kbcL, kbcR, iK, ib, kscan, iT30, ipos, ivel, iwid -- -- doc: <http://www.csounds.com/manual/html/barmodel.html> barmodel :: Sig -> Sig -> D -> D -> Sig -> D -> D -> D -> D -> Sig barmodel = opc9 "barmodel" [(a, [k,k,i,i,k,i,i,i,i])] -- | An emulation of a mandolin. -- -- > ares mandol kamp, kfreq, kpluck, kdetune, kgain, ksize, ifn [, iminfreq] -- -- doc: <http://www.csounds.com/manual/html/mandol.html> mandol :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Sig mandol = opc7 "mandol" [(a, ks 6 ++ is 2)] -- | An emulation of a mini-Moog synthesizer. -- -- > ares moog kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn, iwfn, ivfn -- -- doc: <http://www.csounds.com/manual/html/moog.html> moog :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Tab -> Sig moog = opc9 "moog" [(a, ks 6 ++ is 3)] -- | An emulation of a human voice. -- -- > ares voice kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn -- -- doc: <http://www.csounds.com/manual/html/voice.html> voice :: Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Sig -> Tab -> Tab -> Sig voice = opc9 "voice" [(a, [k,k,k,k,k,k,k,i,i])]