3:@?      !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~                                  ! " # $ % & ' ( ) * + , - . / 0 1 2 3 4 5 6 7 8 9 : ; < = > ? @ A B C D E F G H I J K L M N O P Q R S T U V WXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~            ! " # $ % & ' ( ) * + , - . / 01234567 89:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ [ \ ] ^ _ ` a b c d e f g h i j k l m n o p q r s t u v w x y z { | } ~           ! " # $ % & ' ( ) * + , - . / 0 1 2 3 4 5 6 7 8 9 : ; < = > ? @ A B C D E F G H I J K L M N O P Q R S T U V W X Y Z [ \ ] ^ _ ` a b c defghijklmnopqrstuvwxyz{|}~      !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~      !"#$%&'()*+,-./0123456789:;<=>#None234=K?Constructs a simple instrument that takes in a tuple of two arguments. They are amplitude and the frequency (in Hz or cycles per second).BmConstructs a drum-like instrument. Drum like instrument has a single argument that signifies an amplitude.*?@ABCD?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`ab?@ABCD&?@ABCD?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abNone234=KEFConverts a value to the midi-instrument. It's used with the functions  , !.@EFGcdefghijklmnopqrstuvwxyz{|}~EFG>EFGcdefghijklmnopqrstuvwxyz{|}~"None234=K HIJKHIJKHIJKNone?     9:;<=>?     >=<;:9None#None ?@ABCDEFGHIJKNone None None      V V     NoneLSets sample rate and block size setRates sampleRate blockSizeM#Sets hardware and software buffers. setBufs hardwareBuf ioBufWSets the output to nosound.XSets midi deviceYSets midi device to all.LMNOPQRSTUVWXY  !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~0123456LMNOPQRSTUVWXY6543210LMNOSRPQTUVWXY ( !"#$%&'A,-./0123456789:;<=>?@+)*IHGFEDCBSJKOPQRNMLXWVUT_^]\[ZYihgfedcba`qjklmnop~}|{zyxwvutsrLMNOPQRSTUVWXY NoneZ/Gets an init-rate value from the list by index.[8Gets an control/audio-rate value from the list by index.Z[H !"#$%&,-./0123456789:;<=>?@ABCDGHIJKLMNOPQRSTUZ[HHIPJKLMNTSRQU !"#$%DCBA@?>=<;:432198567,0/-.OG[&ZZ[ None3E#\0Represents a values with frequency of occurence.^UConstant event stream. It produces the same value (the first argument) all the time._ Behaves like $%, but returns an event stream.`Fires a single event right now. loadbang = pulseE 0a2Fires a single true value in the given time ahead.b-Fires a single event in the given time ahead.c*Makes an event stream from list of events.d Behaves like $&, but returns an event stream.e Behaves like $', but returns an event stream.f;the sync function but time is measured in beats per minute.g2Splits event stream on two streams with predicate.h9Splits a toggle event stream on on-events and off-events.iConstructs an event stream that contains an infinite repetition values from the given list. When an event happens this function takes the next value from the list, if there is no values left it starts from the beggining of the list.jbTurns an event of indices to the event of the values from the list. A value is taken with index. k 1range (xMin, xMax) === cycleE [xMin .. pred xMax]l>An event stream of the integers taken from the given diapason.m5An event stream of the random values in the interval (0, 1).n9An event stram of lists of random values in the interval (0, 1)5. The first argument is the length of the each list.oSkips elements at random.  randSkip probwhere prob@ is probability of includinng the element in the output stream. pSkips elements at random. randSkip probFunIt behaves just like randSkip', but probability depends on the value.qWhen something happens on the given event stream resulting event stream contains an application of some unary function to the given initial value. So the event stream contains the values: '[s0, f s0, f (f s0), f (f (f s0)), ...]rISubstitutes all values in the input stream with the given constant value.suAccumulates a values from the given event stream with binary function. It's a variant of the fold for event streams. appendE z f evt When value a happens with evt2, the resulting event stream contains a value (z f a) and in the next time z equals to this value.t"A special variant of the function s% for the monoids. Initial value is  and binary function is , which belong to the instance of the class .uiConstructs an event stream that contains values from the given list which are taken in the random order.vConstructs an event stream that contains values from the given list which are taken in the random order. In the list we specify not only values but the frequencies of occurrence. Sum of the frequencies should be equal to one.w%This function combines the functions  ( and v. We transform the values of the event stream with stateful function that produce not just values but the list of values with frequencies of occurrence. We apply this function to the current state and the value and then at random pick one of the values.xSpecialization of the function y.  every n [a, b, c, ..] evt#constructs a mask that skips first n elements and then produces an event and skips next (a - 1) events, then produces an event and skips next (b - 1) events and so on. It's useful for construction of the percussive beats. For example  every 0 [2] (metroE 2)ktriggers an event on the odd beats. With this function we can create a complex patterns of cyclic events.yFilters events with the mask. A mask is a list of ones and zeroes. n'th element from the given list should be included in the resulting stream if the n'th element from the list equals to one or skipped if the element equals to zero.z<Converts clicks to alternating 0 and 1 (toggle event stream){VConverts clicks to alternating 1 and 0 (toggle event stream with first value set to 1)|MTakes the ns events from the event stream and ignores the rest of the stream.}LDrops the ns events from the event stream and leaves the rest of the stream.~)Takes events while the predicate is true.)Drops events while the predicate is true.*\]^_`abcdefghijklmnopqrstuvwxyz{|}~6\]^_`abcdefghijklmnopqrstuvwxyz{|}~6]f_bde`a^ciqrstg|}~hz{\uvwmnlopkjxy*\]^_`abcdefghijklmnopqrstuvwxyz{|}~None234=K,Mixes the scores and plays them in the loop.0Mixes the procedures and plays them in the loop.kInvokes an instrument with first event stream and holds the note until the second event stream is active.VInvokes an instrument with toggle event stream (1 stands for on and 0 stands for off).kInvokes an instrument with first event stream and holds the note until the second event stream is active.FSets the same duration for all events. It's useful with the functions scheds, schedsBy, scheds_. FSets the same duration for all events. It's useful with the functions sched, schedBy, sched_. PTriggers an instrument with an event stream. The event stream contains triples: P(delay_after_event_is_fired, duration_of_the_event, argument_for_the_instrument)It's like the function trig, but delay is set to zero.2An instrument is triggered with event stream and delay time is set to zero (event fires immediately) and duration is set to infinite time. The note is held while the instrument is producing something. If the instrument is silent for some seconds (specified in the first argument) then it's turned off.kInvokes an instrument with first event stream and holds the note until the second event stream is active.)Triggers a procedure on the event stream.FTriggers a procedure on the event stream. A delay time is set to zero.kInvokes an instrument with first event stream and holds the note until the second event stream is active.IA closure to trigger an instrument inside the body of another instrument.IA closure to trigger an instrument inside the body of another instrument.IA closure to trigger an instrument inside the body of another instrument.>Executes some procedure for the whole lifespan of the program,6?@ABCDHIJK6HIJKBCD?@ANone 'Specifies the midi channel or programm.Produces midi amplitude and frequency as a signal. The signal fades out when nothing is pressed. It can be used in mono-synths. Arguments are portamento time and release time. A portamento time is time it takes for transition from one note to another. *monoMsg channel portamentoTime releaseTimeProduces midi amplitude and frequency as a signal and holds the last value till the next one is present. It can be used in mono-synths. Arguments are portamento time and release time. A portamento time is time it takes for transition from one note to another. holdMsg portamentoTime~Listens to midi on event on the given key as event stream. The event stream carries the level of volume (ranges from 0 to 1).;Listens to midi on event off the given key as event stream.,Initialization of the midi control-messages.BInitializes midi control and get the value in the specified range.BInitializes midi control and get the value in the range (-1) to 1.RUnipolar midiCtrl. Initializes midi control and get the value in the range 0 to 1.W8EFGW8EFGNone6A radio button. It takes a list of values with labels.A matrix of values.URadio button that returns functions. Useful for picking a waveform or type of filter.Matrix of functional values.Shortcut for press [ events.Shortcut for release [ events.GCreates an event in the output stream when one of the chars is pressed.ICreates an event in the output stream when one of the chars is depressed.eUnipolar linear slider. The value belongs to the interval [0, 1]. The argument is for initial value.cUnipolar linear knob. The value belongs to the interval [0, 1]. The argument is for initial value.DExponential slider (usefull for exploring frequencies or decibels).  xknob min max initValVThe value belongs to the interval [min, max]. The last argument is for initial value.BExponential knob (usefull for exploring frequencies or decibels).  xknob min max initValVThe value belongs to the interval [min, max]. The last argument is for initial value.Unit linear joystick.The sample and hold widget. You can pick a value from the list of doubles. The original value is a head of the list (the first element). The visual grouping is horizontal (notice the prefix h(). It's common to use it with function selector.The sample and hold widget. You can pick a value from the list of doubles. The original value is a head of the list (the first element). The visual grouping is vertical (notice the prefix v(). It's common to use it with function selector.The matrix of unipolar knobs. (knobPad columnNum rowNum names initVals It takes in the dimensions of matrix, the names (we can leave it empty if names are not important) and list of init values. It returns a function that takes in indices and produces the signal in the corresponding cell.The matrix of toggle buttons. *togglePad columnNum rowNum names initVals It takes in the dimensions of matrix, the names (we can leave it empty if names are not important) and list of init values (on/off booleans). It returns a function that takes in indices and produces the event stream in the corresponding cell.The matrix of buttons.  buttonPad columnNum rowNum namesIt takes in the dimensions of matrix, the names (we can leave it empty if names are not important). It returns a function that takes in indices and produces the event stream in the corresponding cell.A generic constructor for matrixes of sound source widgets. It takes the constructor of the widget, a default initial value, the dimensions of the matrix, the list of names and the list of initial values. It produces the function that maps indices to corresponding values.Horizontal radio group.Vertical radio group.Horizontal radio group.Vertical radio group.XYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~ZXY[\]^_`abcdefghijklmnopqrstuvwxyz{|}~None246Creates a window with the given name, size and content win name (width, height) guiThe shortcut for  mapSource.wCombines two sound sources. Visuals are aligned horizontally and the sound sources a grouped with the given function. uCombines two sound sources. Visuals are aligned vertically and the sound sources a grouped with the given function. It's just like the hlift2> but two more parameters change visual scaling of the widgets.It's just like the vlift2> but two more parameters change visual scaling of the widgets. The same as hlift2 but for three sound sources. The same as vlift2 but for three sound sources. The same as hlift2' but for three sound sources. The same as vlift2' but for three sound sources. The same as hlift2 but for four sound sources. The same as vlift2 but for four sound sources. The same as hlift2' but for four sound sources. The same as vlift2' but for four sound sources. The same as hlift2 but for five sound sources. The same as vlift2 but for five sound sources. The same as hlift2' but for five sound sources. The same as vlift2' but for five sound sources.XYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~     9:;<=>,None24Renders Csound file.0Render Csound file and save it to the give file.=Render Csound file with options and save it to the give file.9Render Csound file and save result sound to the wav-file.FRender Csound file with options and save result sound to the wav-file.qRenders Csound file, saves it to the given file, renders with csound command and plays it with the given program. playCsd program file csd Produces files file.csd (with )*) and file.wav (with csound) and then invokes: program "file.wav"Works just like )+# but you can supply csound options.Renders csound code to file tmp.csd with flags set to -odac and -Ma0 (sound output goes to soundcard in real time)., with options.)Output to dac with virtual midi keyboard.@Output to dac with virtual midi keyboard with specified options.Renders to file tmp.csd and invokes the csound on it.Renders to file tmp.csd and invokes the csound on it.6Renders to tmp.csd and tmp.wav and plays with mplayer.6Renders to tmp.csd and tmp.wav and plays with mplayer.;Renders to tmp.csd and tmp.wav and plays with totem player.;Renders to tmp.csd and tmp.wav and plays with totem player.54None2346=K ?A class for easy way to process the outputs of the instruments.?A class for easy way to process the outputs of the instruments.Scaling the sound. Crossfade. cfd coeff sig1 sig2[If coeff equals 0 then we get the first signal and if it equals 1 we get the second signal.Bilinear interpolation for four signals. The signals are placed in the corners of the unit square. The first two signals are the xy coordinates in the square.  cfd4 x y a b c d(0, 0) is for a(1, 0) is for b(1, 1) is for c(0, 1) is for d5Generic crossfade for n coefficients and n+1 signals. cfds coeffs sigsSpectral crossfade.!Spectral bilinear crossfade (see cfd4).Generic spectral crossfade. Weighted sum.V      !"#$%&'(R      !"#$%&'(NoneFilter without a resonance. f centerFreq q asigResonant filter. f centerFreq q asigLow-pass filter. lp cutoff resonance sigHigh-pass filter. hp cutoff resonance sigBand-pass filter. bp cutoff resonance sigBand-reject filter. br cutoff resonance sigAll-pass filter. alp cutoff resonance sigHigh-pass filter. bhp cutoff sigLow-pass filter. blp cutoff sigBand-pass filter. bbp cutoff bandwidth sigBand-regect filter. bbr cutoff bandwidth sigMoog's low-pass filter. %mlp centerFrequency qResonance signalMakes slides between values in the signals. The first value defines a duration in seconds for a transition from one value to another in piecewise constant signals.Produces smooth transitions between values in the signals. The first value defines a duration in seconds for a transition from one value to another in piecewise constant signals.>Applies a filter n-times. The n is given in the first rgument.WApplies a flat filter (without resonance) n-times. The n is given in the first rgument.7Low pass filter 18 dB with built in distortion module. )lp18 distortion centerFreq resonance asigdistortion's range is 0 to 1resonance's range is 0 to 1rAnother implementation of moog low pass filter (it's moogvcf in Csound). The arguments have are just like in the mlp filter.Mooglowpass filter with 18 dB.2First order low pass filter (tone in Csound, 6 dB) lp1 centerFreq asig4First order high pass filter (atone in Csound, 6 dB) hp1 centerFreq asigNResonance band pass filter (yet another implementation, it's reson in Csound) bp2 centerFreq q asigQResonance band reject filter (yet another implementation, it's areson in Csound) br2 centerFreq q asigFormant filter. $formant bandPassFilter formants asigIt expects a band pass filter, a list of formants and processed signal. The signal is processed with each filter the result is a sum of all proceessed signals. Formant filters are used to mimic the vocalization of the sound.Formant filter that sings an A. Formant filter that sings an O. Formant filter that sings an E. Formant filter that sings an U. Formant filter that sings an O."    )*+,-        "    )*+,-None Creates a midi instrument from sf2 sound font. Midi listens on all channels. It's useful to quickly test a sound font. The second argument is a sustain in seconds. How long it takes for the sound to decay.Creates a midi instrument from sf2 sound font file. The second argument is sustain in seconds. Reads samples with linear interpolation.Creates a midi instrument from sf2 sound font file. The second argument is sustain in seconds. Reads samples with cubic interpolation.Creates a midi instrument from sf2 sound font file. The second argument is sustain in seconds. Reads samples with linear interpolation. Produces mono output.Creates a midi instrument from sf2 sound font file. The second argument is sustain in seconds. Reads samples with cubic interpolation. Produces mono output.]Midi looper of the sf2 samples. The first arguments are: start, end, crossfade of the loop.Reads sf2 samples at given midi velocity and key (both are from 0 to 127). The second argument is sustain. Interpolation is linear.Reads sf2 samples at given midi velocity and key (both are from 0 to 127). The second argument is sustain. Interpolation is cubic.Reads sf2 samples at given midi velocity and key (both are from 0 to 127). The second argument is sustain. Interpolation is linear. The output is mono.Reads sf2 samples at given midi velocity and key (both are from 0 to 127). The second argument is sustain. Interpolation is cubic. The output is mono.XLooper of the sf2 samples. The first arguments are: start, end, crossfade of the loop.^Reads sf2 samples with amplitude in (0, 1) and frequency in Hz. The interpolation is linear.]Reads sf2 samples with amplitude in (0, 1) and frequency in Hz. The interpolation is cubic.sReads sf2 samples with amplitude in (0, 1) and frequency in Hz. The interpolation is linear. The output is mono.rReads sf2 samples with amplitude in (0, 1) and frequency in Hz. The interpolation is cubic. The output is mono.XLooper of the sf2 samples. The first arguments are: start, end, crossfade of the loop..frequency to midi/ 0123.  / 0123.-None     VWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~     89:;<=>?@ABCDEFGHIJK\]^_`abcdefghijklmnopqrstuvwxyz{|}~  None0+$The default table. It's rendered to (-1) in the Csound.,Creates a new table. The Tab could be used while the instrument is playing. When the instrument is retriggered the new tab is allocated.  newTab size-`Creates a new global table. It's generated only once. It's persisted between instrument calls. newGlobalTab identifier size.Calculates the number of samples needed to store the given amount of seconds. It multiplies the value by the current sample rate./;Calculates the closest power of two value for a given size.0QCalculates the closest power of two value in samples for a given size in seconds.1Loads wav or aiff file to table wavs fileName skipTime channel<skipTime specifies from what second it should read the file.?with channel argument we can read left, right or both channels.2Loads mp3 file to table: mp3s fileName skipTime format<skipTime specifies from what second it should read the file.}format is: 1 - for mono files, 2 - for stereo files, 3 - for left channel of stereo file, 4 for right channel of stereo file3Table contains all provided values (table is extended to contain all values and to be of the power of 2 or the power of two plus one). (by default it skips normalization).4#Segments of the exponential curves. exps [a, n1, b, n2, c, ...]where  a, b, c, ... are ordinate values n1, n2, ...U are lengths of the segments relative to the total number of the points in the table5.Equally spaced segments of exponential curves. eexps [a, b, c, ...] is the same as exps [a, 1, b, 1, c, ...]6Segments of cubic polynomials.  cubes [a, n1, b, n2, c, ...]where a, b, c .. - are ordinate values n1, n2, ...U are lengths of the segments relative to the total number of the points in the table7-Equally spaced segments of cubic polynomials. ecubes [a, b, c, ...] is the same as cubes [a, 1, b, 1, c, ...]8Segments of straight lines.  lins [a, n1, b, n2, c, ...]where a, b, c .. - are ordinate values n1, n2, ...U are lengths of the segments relative to the total number of the points in the table9*Equally spaced segments of straight lines. elins [a, b, c, ...] is the same as lins [a, 1, b, 1, c, ...]:Cubic spline curve. splines [a, n1, b, n2, c, ...]where a, b, c .. - are ordinate values n1, n2, ...U are lengths of the segments relative to the total number of the points in the table;Equally spaced spline curve. esplines [a, b, c, ...] is the same as splines [a, 1, b, 1, c, ...]<$Constant segments (sample and hold). consts [a, n1, b, n2, c, ...]where a, b, c .. - are ordinate values n1, n2, ...U are lengths of the segments relative to the total number of the points in the table=!Equally spaced constant segments. econsts [a, b, c, ...] is the same as consts [a, 1, b, 1, c, ...]>9Creates a table from a starting value to an ending value. GstartEnds [val1, dur1, type1, val2, dur2, type2, val3, ... typeX, valN],val1, val2 ... -- end points of the segments+dur1, dur2 ... -- durations of the segmentswtype1, type2 ... -- if 0, a straight line is produced. If non-zero, then it creates the following curve, for dur steps: >beg+(end-beg)*(1-exp(i*type))/(1-exp(type * dur))$beg, end - end points of the segmentdur - duration of the segment?.Equally spaced interpolation for the function  startEnds )estartEnds [val1, type1, val2, typ2, ...]is the same as 0estartEnds [val1, 1, type1, val2, 1, type2, ...]@Series of harmonic partials: sine = sines [1] #saw = sines $ fmap (1 / ) [1 .. 10] )square = sines $ fmap (1 / ) [1, 3 .. 11] Ntriangle = sines $ zipWith (\a b -> a / (b ** 2)) (cycle [1, -1]) [1, 3 .. 11]A Just like B$ but partial strength is set to one.B Just like C but phases are set to zero.C1Specifies series of possibly inharmonic partials.DESpecifies series of possibly inharmonic partials with direct current.ETable for pure sine wave.FTable for pure cosine wave.GTable for sigmoid wave.H'Generates values similar to the opcode $..  Cbuzzes numberOfHarmonics [lowestHarmonic, coefficientOfAttenuation]With buzzes n [l, r] you get n harmonics from l& that are attenuated by the factor of r on each step.ISModified Bessel function of the second kind, order 0 (for amplitude modulated FM).   bessels xint,the function is defined within the interval  [0, xint].J Polynomials. polys xl xr [c0, c1, c2, ..]whereNxl, xr - left and right values of the interval over wich polynomial is defined3[c0, c1, c2, ...] -- coefficients of the polynomial c0 + c1 * x + c2 * x * x + ...K(Chebyshev polynomials of the first kind. polys xl xr [h0, h1, h2, ..]whereNxl, xr - left and right values of the interval over wich polynomial is defined6[h0, h1, h2, ...] -- relative strength of the partialsL)Chebyshev polynomials of the second kind. polys xl xr [h0, h1, h2, ..]whereNxl, xr - left and right values of the interval over wich polynomial is defined6[h0, h1, h2, ...] -- relative strength of the partialsVDCreates a table of doubles (It's f-table in Csound). Arguments are: identificator of the GEN routineGEN routine arguments9All tables are created at 0 and memory is never released.W Adds guard point to the table size (details of the interpolation schemes: you do need guard point if your intention is to read the table once but you don't need the guard point if you read table in many cycles, the guard point is the the first point of your table). X Shortcut for W.YBSets an absolute size value. As you can do it in the Csound files.ZOSets the relative size value. You can set the base value in the options (see / at 0, with tabResolution you can easily change table sizes for all your tables). Here zero means the base value. 1 is the base value multiplied by 2, 2 is the base value multiplied by 4 and so on. Negative values mean division by the specified degree. [Sets degrees from -3 to 3.\Sets degrees from -3 to 3.]Sets degrees from -3 to 3.^Sets degrees from -3 to 3._Sets degrees from -3 to 3.`Sets degrees from -3 to 3.aSets degrees from -3 to 3.b&Writes tables in sequential locations.This opcode writes to a table in sequential locations to and from an a-rate variable. Some thought is required before using it. It has at least two major, and quite different, applications which are discussed below. kstart tablewa kfn, asig, koff csound docs: /http://www.csounds.com/manual/html/tablewa.htmlcVTransforms phasor that is defined in seconds to relative phasor that ranges in 0 to 1.\456789:;<= !"#$%&'()*+,-./0>1?2@ABCD3456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUEFVWXYZ[\]^_`abcGHf'()*+EFP !"#$%&'()*+,-./7 !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcfP++*()'7/.3,-./0'*)(!&%$#"12 @CBADHEFG<864:>=975;?JKLIMNOPQTURSVFEYZWX[\]^_`a-,+*)('&%$#" !bcK4 =<;:98765 !&%$#"'*)(+,-./0>1?2@ABCD3456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUEFVWXYZ[\]^_`abcGHNonedLow frequency oscillatoreA pure tone (sine wave).f*An oscillator with user provided waveform.gLTurns a bipolar sound (ranges from -1 to 1) to unipolar (ranges from 0 to 1)hMTurns an unipolar sound (ranges from 0 to 1) to bipolar (ranges from -1 to 1)iUnipolar pure tone.j Unipolar 12.kUnipolar sawtooth.lUnipolar integrated sawtooth.mUnipolar square wave.nUnipolar triangle wave.oUnipolar pulse.p!Unipolar band-limited oscillator.qFrequency modulation +fosc carrierFreq modulatorFreq modIndex cpsr.Pulse width modulation (width range is 0 to 1) pw dutyCycle cpss9Triangle wave with ramp factor (factor's range is 0 to 1) ramp factor cpst%Unipolar pulse width modulation wave.u'Unipolar triangle wave with ram factor.v1Rescaling of the bipolar signal (-1, 1) -> (a, b)  on a b biSigw1Rescaling of the unipolar signal (0, 1) -> (a, b)  on a b uniSigxGConstant random signal. It updates random numbers with given frequency. constRnd freq yELinear random signal. It updates random numbers with given frequency.  rndi freq z Unipolar rndh{ Unipolar rndi| White noise.} Pink noise.~Low frequency oscillator lfo shape depth ratedefghijklmnopqrstuvwxyz{|}~!defghijklmnopqrstuvwxyz{|}~!efrsghvwijklomtnupxzy{|}qd~defghijklmnopqrstuvwxyz{|}~None/+Linear adsr envelope generator with release  leg attack decay sustain release0Exponential adsr envelope generator with release  xeg attack decay sustain release>Makes time intervals relative to the note's duration. So that: onIdur [a, t1, b, t2, c] becomes: [a, t1 * idur, b, t2 * idur, c]>Makes time intervals relative to the note's duration. So that: onDur dt [a, t1, b, t2, c] becomes: , t1 * dt, b, t2 * dt, c] The opcode 34B with time intervals relative to the total duration of the note. The opcode 35B with time intervals relative to the total duration of the note. The opcode 34T with time intervals relative to the total duration of the note given by the user. The opcode 35T with time intervals relative to the total duration of the note given by the user. The opcode 36h with time intervals relative to the total duration of the note. Total time is set to the value of idur. linendur asig rise decay The opcode 36w with time intervals relative to the total duration of the note. Total time is set to the value of the first argument. linendurBy dt asig rise decay$Fades in with the given attack time.%Fades out with the given attack time.0Fades in by exponent with the given attack time.1Fades out by exponent with the given attack time.&A combination of fade in and fade out. "fades attackDuration decayDuration2A combination of exponential fade in and fade out. %expFades attackDuration decayDuration3Sample and hold cyclic signal. It takes the list of [a, dta, b, dtb, c, dtc, ...]4the a, b, c, ... are values of the constant segmentsAthe dta, dtb, dtc, are durations in seconds of constant segments.@The period of the repetition equals to the sum of all durations.It's just like linseg but it loops over the envelope.It's just like expseg but it loops over the envelope.NSample and hold sequence. It outputs the looping sequence of constan elements.&Step sequencer with unipolar triangle.$Step sequencer with unipolar square.&Step sequencer with unipolar sawtooth.,Step sequencer with unipolar inveted square..Step sequencer with unipolar inveted sawtooth.2Step sequencer with unipolar exponential sawtooth.;Step sequencer with unipolar inverted exponential sawtooth.2Step sequencer with unipolar exponential triangle.wA sequence of unipolar waves with pulse width moulation (see upw). The first argument is a duty cycle in range 0 to 1.A sequence of unipolar inverted waves with pulse width moulation (see upw). The first argument is a duty cycle in range 0 to 1.A sequence of unipolar triangle waves with ramp factor (see uramp). The first argument is a ramp factor cycle in range 0 to 1.A sequence of unipolar exponential triangle waves with ramp factor (see uramp). The first argument is a ramp factor cycle in range 0 to 1.A sequence of unipolar inverted triangle waves with ramp factor (see uramp). The first argument is a ramp factor cycle in range 0 to 1.A sequence of unipolar inverted exponential triangle waves with ramp factor (see uramp). The first argument is a ramp factor cycle in range 0 to 1.JLooping sample and hold envelope. The first argument is the list of pairs:  [a, durA, b, durB, c, durc, ...]It's a list of values and durations. The durations are relative to the period of repetition. The period is specified with the second argument. The second argument is the frequency of repetition measured in Hz. lpshold valDurs frequencyJLooping linear segments envelope. The first argument is the list of pairs:  [a, durA, b, durB, c, durc, ...]It's a list of values and durations. The durations are relative to the period of repetition. The period is specified with the second argument. The second argument is the frequency of repetition measured in Hz. loopseg valDurs frequencyOLooping exponential segments envelope. The first argument is the list of pairs:  [a, durA, b, durB, c, durc, ...]It's a list of values and durations. The durations are relative to the period of repetition. The period is specified with the second argument. The second argument is the frequency of repetition measured in Hz. loopxseg valDurs frequencyWIt's like lpshold but we can specify the phase of repetition (phase belongs to [0, 1]).WIt's like loopseg but we can specify the phase of repetition (phase belongs to [0, 1]).XIt's like loopxseg but we can specify the phase of repetition (phase belongs to [0, 1]).The looping ADSR envelope. 7xadsrSeq attack decay sustain release weights frequencyVThe sum of attack, decay, sustain and release time durations should be equal to one.The looping exponential ADSR envelope. there is a fifth segment at the end of the envelope during which the envelope equals to zero. 7xadsrSeq attack decay sustain release weights frequencyVThe sum of attack, decay, sustain and release time durations should be equal to one.3The looping ADSR envelope with the rest at the end. ;adsrSeq attack decay sustain release rest weights frequency\The sum of attack, decay, sustain, release and rest time durations should be equal to one.The looping exponential ADSR envelope. there is a fifth segment at the end of the envelope during which the envelope equals to zero. =xadsrSeq_ attack decay sustain release rest weights frequency\The sum of attack, decay, sustain, release and rest time durations should be equal to one.*The looping sequence of constant segments. DlinSeg [a, durA, b, durB, c, durC, ...] [scale1, scale2, scale3] cpsThe first argument is the list that specifies the shape of the looping wave. It's the alternating values and durations of transition from one value to another. The durations are relative to the period. So that lists *[0, 0.5, 1, 0.5, 0] and [0, 50, 1, 50, 0]produce the same results. The second list is the list of scales for subsequent periods. Every value in the period is scaled with values from the second list. The last argument is the rate of repetition (Hz).(The looping sequence of linear segments. DlinSeg [a, durA, b, durB, c, durC, ...] [scale1, scale2, scale3] cpsThe first argument is the list that specifies the shape of the looping wave. It's the alternating values and durations of transition from one value to another. The durations are relative to the period. So that lists *[0, 0.5, 1, 0.5, 0] and [0, 50, 1, 50, 0]produce the same results. The second list is the list of scales for subsequent periods. Every value in the period is scaled with values from the second list. The last argument is the rate of repetition (Hz).-The looping sequence of exponential segments. DexpSeg [a, durA, b, durB, c, durC, ...] [scale1, scale2, scale3] cpsThe first argument is the list that specifies the shape of the looping wave. It's the alternating values and durations of transition from one value to another. The durations are relative to the period. So that lists *[0, 0.5, 1, 0.5, 0] and [0, 50, 1, 50, 0]produce the same results. The second list is the list of scales for subsequent periods. Every value in the period is scaled with values from the second list. The last argument is the rate of repetition (Hz).?IJKLMNOPQRSTUVW00?IJKLMNOPQRSTUVWNone#Selects odd elements from the list.$Selects even elements from the list.)Reads table once during the note length. 0Reads table once during a given period of time. 3Reads table several times during the note length.  Mean value.UAdds vibrato to the sound unit. Sound units is a function that takes in a frequency. ^Adds a random vibrato to the sound unit. Sound units is a function that takes in a frequency. =Chorus takes a number of copies, chorus width and wave shape.Applies a resonator to the signals. A resonator is a list of band pass filters. A list contains the parameters for the filters: [(centerFrequency, bandWidth)]A resonator with user defined band pass filter. Warning: a filter takes in a center frequency, band width and the signal. The signal comes last (this order is not standard in the Csound but it's more convinient to use with Haskell).Mixes dry and wet signals.  dryWet ratio effect asigratio - of dry signal to weteffect - means to wet the signalasig -- processed signal%Chain of mass-spring-damping filters. modes params baseCps exciter params - a list of pairs '(resonantFrequencyRatio, filterQuality)baseCps" - base frequency of the resonator-exciter - an impulse that starts a resonator.1Doubles the mono signal to get the stereo signal.Random panningRandom panning'Random volume (with gauss distribution) gaussVol radiusOfDistribution Random volume gaussVol (minVolume, maxVolume)pHi-fi output for stereo signals. Saves the stereo signal to file. The length of the file is defined in seconds. "writeHifi fileLength fileName asig}It picks a signal from the list by integer index. The original value is taken from the head of the list (the first element).Creates running arpeggios.  +arpeggiBy ampWeights pitches instrument cpsIt plays an instrument with fast sequence of notes. We can specify the pitches and amplitude weights of the notes as well as frequency of repetition.Creates running arpeggios.  =arpeggiBy ampWave pitchwave ampWeights pitches instrument cpsIt plays an instrument with fast sequence of notes. We can specify amplitude envelope wave, pitch envelope wave, the pitches and amplitude weights of the notes as well as frequency of repetition.ZLow-pass filter pictured as joystick. Ox is for center frequency and Oy is for resonance.!Chains all functions in the list.HApplies all functions in the list to the given input and summs them up.XXNone 'Settings for cross filtering algorithm.#They are the defaults for opvodes: pvsifd, tradsyn, trcross and partials. ;Fft size degree -- it's the power of 2. The default is 12.8Hop size degree -- it's the power of 2. The default is 9.scale --amplitude scaling factor. default is 10pitch -- the pitch scaling factor. default is 1  maxTracksJ -- max number of tracks in resynthesis (tradsyn) and analysis (partials).winType$ -- O: Hamming, 1: Hanning (default)Search/ -- search interval length. The default is 1.05Depth -- depth of the effectThreshq -- analysis threshold. Tracks below ktresh*max_magnitude will be discarded (1 > ktresh >= 0).The default is 0.01 MinPointsY -- minimum number of time points for a detected peak to make a track (1 is the minimum).MaxGap -- maximum gap between time-points for track continuation (> 0). Tracks that have no continuation after kmaxgap will be discarded.Converts signal to spectrum.Converts spectrum to signal.7Applies a transformation to the spectrum of the signal.~Scales all frequencies. Usefull for transposition. For example, we can transpose a signal by the given amount of semitones: scaleSpec (semitone 1) asig+Adds given amount of Hz to all frequencies. addSpec hz asigScales frequency in semitones.LFilters the partials of the second signal with partials of the first signal.PSubstitutes the partials of the second signal with partials of the first signal.crossSpecFilter for mono signals.crossSpecVocoder for mono signals.YZ[ YZ[NoneD2Scaling factor for pitch. The 1 is inherent pitch.2Scaling factor for tempo. The 1 is inherent tempo.Fidelity corresponds to the size of the FFT-window that is used by functions of RAM-family. The function performs the FFT transform and it has to know the size. It's not the value for the size it's an integer value that proportional to the size. The higher the value the higher the size the lower the value the lower the size. The default value is 0. Zero is best for most of the cases. For drums we can lower it to (-2).PS: here is the formula for window size: 2 ** (fidelity + 11). So the fidelity is actually the degree for power of two. The FFT-algorithm requires the window size to be a power of two.AThe lower fidelity is the less power is consumed by the function.YLooping phasor. It creates a looping pointer to the file. It's used in the function ram.jTher arguments are: file name, start and end of the looping segment (in seconds), and the playback speed.The sample format.64-bit floats with a header24-bit integers with a header%8-bit unsigned integers with a header32-bit floats with a header32-bit integers with a header 16-bit integers with a headeru-law samples with a headerP16-bit integers with a header. The header type depends on the render (-o) format16-bit integers without header,32-bit floating point samples without headerKTakes only given amount (in seconds) from the signal (the rest is silence).0Delays signals by the given amount (in seconds).{Delays a signal by the first argument and takes only second argument amount of signal (everything is measured in seconds).)Repeats the signal with the given period.TPlays the first signal for some time (in seconds) and then switches to the next one. afterSnd dur sig1 sig2iCreates a sequence of signals. Each segment lasts for fixed amount of time given in the first argument.Creates a sequence of signals and loops over the sequence. Each segment lasts for fixed amount of time given in the first argument.1Converts stereosignal to mono with function mean.$Length in seconds of the sound file.;Produces repeating segments with the given time in seconds.=Reads stereo signal from the sound-file (wav or mp3 or aiff).nReads stereo signal from the sound-file (wav or mp3 or aiff) and loops it with the given period (in seconds).`Reads stereo signal from the sound-file (wav or mp3 or aiff) and loops it with the file length.Reads the wav file with the given speed (if speed is 1 it's a norma playback). We can use negative speed to read file in reverse.$Reads th wav file and loops over it.Reads a segment from wav file. Reads the wav file with the given speed (if speed is 1 it's a norma playback). We can use negative speed to read file in reverse. Scales the tempo with first argument.JReads th wav file and loops over it. Scales the tempo with first argument. !The mono variant of the function readSnd. !The mono variant of the function  loopSndBy. !The mono variant of the function loopSnd. !The mono variant of the function readWav. !The mono variant of the function loopWav.Reads a segment from wav file.Reads the mono wav file with the given speed (if speed is 1 it's a norma playback). We can use negative speed to read file in reverse. Scales the tempo with first argument.OReads th mono wav file and loops over it. Scales the tempo with first argument.Loads the sample in the table. The sample should be short. The size of the table is limited. It's up to 3 minutes for 44100 sample rate (sr), 2.9 minutes for 48000 sr, 1.4 minutes for 96000 sr.Loads the sample in the table. The sample should be short. The size of the table is limited. It's up to 6 minutes for 44100 sample rate (sr), 5.9 minutes for 48000 sr, 2.8 minutes for 96000 sr.rWrites a sound signal to the file with the given format. It supports only four formats: Wav, Aiff, Raw and Ircam.Writes wav files.Writes aiff files.!Writes mono signals to wav files."Writes mono signals to aiff files.)mincer  Phase-locked vocoder processing.mincer implements phase-locked vocoder processing using function tables containing sampled-sound sources, with GEN01, and mincer will accept deferred allocation tables.SThis opcode allows for time and frequency-independent scaling. Time is controlled by a time index (in seconds) to the function table position and can be moved forward and backward at any chosen speed, as well as stopped at a given position ("frozen"). The quality of the effect is generally improved with phase locking switched on. ?asig mincer atimpt, kamp, kpitch, ktab, klock[,ifftsize,idecim] csound doc: .http://www.csounds.com/manual/html/mincer.html=Mincer. We can playback a table and scale by tempo and pitch. $mincer fidelity table pointer pitch fidelity is the parameter that specifies the size of the window (for FFT transform). The size equals to formula (fidelity + 11) ^ 2. If you don't know what to choose choose 0 for pitched sounds and -2 for drums. The table contains the sample to playback. The pointer loops over the table. The pitch specifies a scaling factor for pitch. So we can raise tone an octave up by setting the pitch to 2.CCreates a pointer signal for reading audio from the table in loops. lphase length start end speedArguments are:length of the table in seconds,start and end points of the reading intervalplayback speed Forces phasor to play only once.Reads the file forth and back. Creates a phasor if segments are relative to the total length. It can be useful for drum loops. If we don't know the complete length but we know that loop contains four distinct parts.OCreates a phasor for reading the whole audio file in loops with given speed. Reads audio files in loops. The file is loaded in RAM. The size of the file is limited. It should be not more than 6 minutes for sample rate of 44100. 5.9 minutes for 48000.What makes this function so cool is that we can scale the sound by tempo without affecting pitch, and we can scale the sound by pitch without affecting the tempo. Let's study the arguments. ram fidelity phasor pitch fidelity corresponds to the size of the FFT-window. The function performs the FFT transform and it has to know the size. It's not the value for the size it's an integer value that proportional to the size. The higher the value the higher the size the lower the value the lower the size. The default value is 0. Zero is best for most of the cases. For drums we can lower it to (-2).#The phasor is a quadruple of values /(Phsr fileName startTime endTime playbackSpeed)we can read the file from startTime to endTime (in seconds) and we can set the speed for playback. If speed is negative file is played in reverse. The playback is looped. So to scale the tempo or play in reverse we can change the playbackSpeed.The last argument is pitch factor. We can rise by octave with factor 2. It's good place to use the function semitone. It produces factors for a number in semitones.Note that all parameters (except window size) are signals. It makes this function very flexible. We can change the speed of playback and start and end of the reading segment as we wish. !ram 0 (Phsr "file.wav" 0 1 1.2) 1=PS: here is the formula for window size: 2 ** (fidelity + 11)!1Reads file once and scales it by tempo and pitch."0Loop over file and scales it by tempo and pitch.#`Reads a segment from file once and scales it by tempo and pitch. Segment is defined in seconds.$^Loops over a segment of file and scales it by tempo and pitch. Segment is defined in seconds.%Reads a relative segment from file once and scales it by tempo and pitch. Segment is defined in seconds. The end ponits for the segment are relative to the total length of the file.&Loops over a relative segment of file and scales it by tempo and pitch. Segment is defined in seconds. The end ponits for the segment are relative to the total length of the file.'The mono version of readRam.(The mono version of loopRam.)The mono version of readSeg.*The mono version of loopSeg.+The mono version of readRel.,The mono version of loopRel.Q\      ]^!"#$%&'()*+,N      !"#$%&'()*+,N      !"#$%&'()*+,? \      ]^!"#$%&'()*+,None-A segment of the signal. The signal segment is a limited span of signal in time. The time can be measured in seconds or in events! The time span which is measured in events is the first occurence of the event in the event stream. (There are handy functions for scheduling the signal segments. we can delay the segment or loop over it or limit it with tme interval or play a sequence of segments. The main feature of the segments is the ability to schedule the signals with event streams (like button clicks or midi-events). .DConverts signals to segments. The segment is not limited in length./3Limits the length of the segment with event stream.0ALimits the length of the segment with constant length in seconds.11Plays the sequence of segments one ofter another.2jPlays a list of segments at the same time. the total length equals to the biggest length of all segments.3LLoops over a segment. The segment should be limited for loop to take effect.4BLimits a signal with an event stream and retriggers it after stop.5Converts segments to signals.6CA pause. Plays nothing until something happens on the event stream.7=Delays a segment until something happens on the event stream.8>A pause. Plays nothing for the given time interval in seconds.95Delays a segment by a given time interval in seconds._OTakes the first event from the event stream and ignores the rest of the stream.-`abcdef./012345ghij6789kl_mnop -./0123456789 -.5/123760984-edcba`f./012345ghij6789kl_mnopNone;RTriggers the signal with the first stream and turns it off with the second stream.q%Toggles the signal with event stream.<Consider note limiting? or performance degrades every note is held to infinity and it continues to produce zeroes. No it's not every sequence note triggers it but it's best to limit them anyway=Plays a list signals. It triggers the signal with event stream and silences all the rest in the list so that only one signal is playing. We can create simple costum monosynthes with this function. The last event stream stops all signals.>7Triggers one signal after another with an event stream.?Triggers a signal when one of the chars from the first string is pressed. Stos signal from playing when one of the chars from the second string is pressed.@&Plays a signal while a key is pressed.A'Toggles the signal when key is pressed.BConsider note limiting? or performance degrades every note is held to infinity and it continues to produce zeroes. No it's not every sequence note triggers it but it's best to limit them anywayCPlays a list of signals when corresponding key is pressed. Turns off all other signals in the group. The last string is for stopping the group from playing.DPlays signals one after another when key is pressed. Stops the group from playing when the char from the last argument is pressed.E%Scales the signal with the amplitude.FApplies a low pass filter to the signal. The first two arguments are the frequency range for center frequency of the filter and the second one is amount of resonance (ranges from 0 to 1).G6the midiLpInstr with audio range for center frequency.H=Ignores the amplitude and justplays back the original signal.IPlays a signal when the key is pressed. Retriggers the signal when the key is pressed again. The key is an integer midi code. The C1 is 60 and the A1 is 69.JPlays a signal when the key is pressed. Retriggers the signal when the key is pressed again. Turns off the signal after specified duration (n seconds). The key is an integer midi code. The C1 is 60 and the A1 is 69.KiPlyas a signal while the key is pressed. The key is an integer midi code. The C1 is 60 and the A1 is 69.LnPlays and stops a signal in the toggle mode. The key is an integer midi code. The C1 is 60 and the A1 is 69.MPlays a set of signals on the list of keys. When certain key is pressed the corresponding signal starts to play and all the rest are stopped. A- The key is an integer midi code. The C1 is 60 and the A1 is 69.NThe generic midiTrig. We can specify the midi function. The midi function takes in a signal and a volume of the pressed key (it ranges from 0 to 1). It produces some output. The default is scaling the signal with the amplitude.OThe generic midiTap. We can specify the midi function. The midi function takes in a signal and a volume of the pressed key (it ranges from 0 to 1). It produces some output. The default is scaling the signal with the amplitude.PThe generic midiPush. We can specify the midi function. The midi function takes in a signal and a volume of the pressed key (it ranges from 0 to 1). It produces some output. The default is scaling the signal with the amplitude.QThe generic midiToggle. We can specify the midi function. The midi function takes in a signal and a volume of the pressed key (it ranges from 0 to 1). It produces some output. The default is scaling the signal with the amplitude.RThe generic midiGroup. We can specify the midi function. The midi function takes in a signal and a volume of the pressed key (it ranges from 0 to 1). It produces some output. The default is scaling the signal with the amplitude.:;q<=>?@ABCDrEFGHIJKLMNOPQRs:;<=>?@ABCDEFGHIJKLMNOPQR;<=>?B@ACDIJKLMNOPQR:EFGH:;q<=>?@ABCDrEFGHIJKLMNOPQRsNone3WKDry/Wet mix value (ranges from 0 to 1). The 0 is all dry. The 1 is all wet.XFeedback for delayYThe delaya timeZThe maximum delay time.[7Mono version of the cool reverberation opcode reverbsc. 'reverbsc1 asig feedbackLevel cutOffFreq\Reverb with given time.]Mono reverb (based on reverbsc) rever1 feedback asig^Mono reverb (based on reverbsc) %rever2 feedback (asigLeft, asigRight)_Mono reverb for small room.`Mono reverb for small hall.aMono reverb for large hall.bThe magic cave reverb (mono).cStereo reverb for small room.dStereo reverb for small hall.eStereo reverb for large hall.fThe magic cave reverb (stereo).gNThe simplest delay with feedback. Arguments are: delay length and decay ratio. echo delayLength ratiohDelay with feedback. %fdelay delayLength decayRatio balanceiDelay with feedback. 2fdelay maxDelayLength delayLength feedback balancej?Multitap delay. Arguments are: max delay length, list of pairs (delayLength, decayRatio)1, balance of mixed signal with processed signal. *fdelay maxDelayLength delays balance asigk'Generic multitap delay. It's just like fvdelaysj but instead of constant feedbackLevel it expects a function for processing a delayed signal on the tap. *fdelay maxDelayLength delays balance asiglSDelay for functions that use some table (as a buffer). As granular synth or mincer. 8tabDelay fn maxDelayTime delayTime feedback balance asigm Distortion. distort distLevel asignChorus. chorus depth rate balance asigo$Flanger. Lfo depth ranges in 0 to 1.!flanger lfo feedback balance asigpFirst order phaser.qGSecond order phaser. Sweeping gaps in the timbre are placed harmonicalyr\Second order phaser. Sweeping gaps in the timbre are placed by powers of the base frequency.s Distortion  fxDistort level drive tone sigIntStereo distortion.uStereo chorus. $stChorus2 mix rate depth width sigInvPhaser +fxPhaser mix rate depth freq feedback sigInwStereo phaser.xFlanger -fxFlanger mix feedback rate depth delay sigInyStereo flangerz Analog delay. (analogDelay mix feedback time tone sigIn{Stereo analog delay.|@Filter effect (a pair of butterworth low and high pass filters). (fxFilter lowPassfFreq highPassFreq gain }GStereo filter effect (a pair of butterworth low and high pass filters).~ Equalizer (equalizer gainsAndFrequencies gain sigInStereo equalizer.@Equalizer with frequencies: 100, 200, 400, 800, 1600, 3200, 64000Equalizer with frequencies: 100, 400, 1600, 6400Gain fxGain gain sigIn'Adds filtered white noize to the signal fxWhite lfoFreq depth sigIn.Adds filtered white noize to the stereo signal&Adds filtered pink noize to the signal fxWhite lfoFreq depth sigIn-Adds filtered pink noize to the stereo signalSimplified delay ,fxEcho maxDelayLength delTime feedback sigInSimplified stereo delay.lInstrument plays an input signal in different modes. The segments of signal can be played back and forth.  &trackerSplice maxLength segLength mode maxLength9 -- the maximum length of the played segment (in seconds) segLength! -- the segment length in secondsmode! -- mode of the playing. If it's 1 - only a part of the sample is plyaed and it's played forward. The portion of the signal starts from the current playback point. It lasts for segLength. If it's 2 - the segment is played in reverse. Other values produce the normal input signal.Original author: Rory WalshExample: rmain = dac $ do let ev ch1 ch2 dt = fmap (\x -> (x, dt)) $ mconcat [ fmap (const 1.5) $ charOn ch1 , fmap (const 2.5) $ charOn ch2 , fmap (const 0) $ charOff ch1 <> charOff ch2] (k, dt) <- stepper (0, 0.1) $ ev 'q' 'w' 0.1 <> ev 'a' 's' 0.2 <> ev 'z' 'x' 0.4 mul 1.3 $ trackerSplice 0.8 dt (int' k) $ fst $ loopWav 1 "drumLoop.wav";STUVWXYZ[\]^_`abcdefghijklmnopqrtuvwstuvwxyz{|}~7STUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~7[]^\_`abcdefZYXWghijklmVUTSnopqrstuvwxyz{|}~;STUVWXYZ[\]^_`abcdefghijklmnopqrtuvwstuvwxyz{|}~None9 Defaults for fof2 opcode./Sndwarp secondary parameters. It's instance of Default, we can use the constant def to get the value.WinSizeL - window size in seconds (not in samples as in Csound!). The default is 0.1Randw - the bandwidth of a random number generator. The random numbers will be added to iwsize. It's measured in ratio to WinSize. So the 1 means the one WinSize length. The default is 0.3Overlapg - determines the density of overlapping windows. The default value is 50. It's in range (0 to 100)6Randomized parameters for arguments (in range 0 to 1).#Secondary parameters for syncgrain.Win8 -- grain window function (half-sine is used by default)OverlapC -- grain overlap (use values in range 0 to 100, the 25 is default)Secondary parameters for granule. We can use the def to get the defaults. Gap - gap between grains in sec.PVoice - number of voices (integer value in magnitude of 1 to 128, 64 is default)gRatio - ratio of the speed of the gskip pointer relative to output audio sample rate (the default is 1)Mode - playback mode (see  GranuleMode, play forward is the default)TSkip_os - gskip pointer random offset in sec, 0 will be no offset (0.5 is default). bGap_os - gap random offset in ratios (0 to 1) of the gap size, 0 gives no offset (0.5 is default).gSize_os -grain size random offset in ratios (0 to 1) of grain size, 0 gives no offset (0.5 is default).=Seed - seed for the random number generator (0.5 is default).XAtt - attack of the grain envelope in ratios (0 to 1) of grain size (0.3 is default).UDec - decay of the grain envelope in ratios (0 to 1) of grain size (0.3 is default).Granule playback mode.#Randomized parameters for function grainy. We can randomize pitch scaleing factor (0 to 1), read position (in ratio: 0 to 1), and duration of the grains (in seconds, in magnitude of 0.005 to 0.5). >Secondary parameters for the partikkel opcode. We can use the defN to get the defaults. See the official docs to know the complete description: csound doc: 1http://www.csounds.com/manual/html/partikkel.htmlGranular synthesizer with "per grain" control over many of its parameters. Has a sync input to sychronize its internal grain scheduler clock to an external clock source.@partikkel was conceived after reading Curtis Roads' book  Microsound , and the goal was to create an opcode that was capable of all time-domain varieties of granular synthesis described in this book. The idea being that most of the techniques only differ in parameter values, and by having a single opcode that can do all varieties of granular synthesis makes it possible to interpolate between techniques. Granular synthesis is sometimes dubbed particle synthesis, and it was thought apt to name the opcode partikkel to distinguish it from other granular opcodes. :partikkel spec grainrate grainsize kpitch ifiltabs apntersspec - secondary parameters grainrate - rate of the grain creation grainsize5 - grain size in sec (!!!not in ms as for Csound!!!).kpitch -- pitch scaling factor.apnters1 -- list of pointers (up to 4 values can be used)ifiltabs/ -- list of tables (up to 4 values can be used)?Simplified version of partikkel. The partikkel for mono sounds. 2grainy1 speed grainrate grainsize kfreqFactor filespeed - speed of the playback grainrate - rate of the grain creation grainsize - size of the grainsfile0 - filename of an audio file to read the grains.ASimplified version of partikkel. The partikkel for stereo sounds. 2grainy1 speed grainrate grainsize kfreqFactor filespeed - speed of the playback grainrate - rate of the grain creation grainsize - size of the grainsfile0 - filename of an audio file to read the grains.Randomized version of grainy1.Randomized version of grainy.`Simplified version of partikkel with pointer access to the table. The partikkel for mono sounds. 4ptrGrainy grainrate grainsize kfreqFactor tab apnterspeed - speed of the playback grainrate - rate of the grain creation grainsize - size of the grainstab - table with sampled sound.apnterD - pointer to the table. pointer is relative to total size (0 to 1).`Simplified version of partikkel with pointer access to the table. The partikkel for mono sounds. 4ptrGrainy grainrate grainsize kfreqFactor tab apnterspeed - speed of the playback grainrate - rate of the grain creation grainsize - size of the grainsfile - file with sampled sound.apnter" - pointer to the table in seconds`Simplified version of partikkel with pointer access to the table. The partikkel for mono sounds. 4ptrGrainy grainrate grainsize kfreqFactor tab apnterspeed - speed of the playback grainrate - rate of the grain creation grainsize - size of the grainsfile - file with sampled sound.apnter" - pointer to the table in secondsRandomized version of  ptrGrainy.4A more complex granular synthesis texture generator.granule is a Csound unit generator which employs a wavetable as input to produce granularly synthesized audio output. Wavetable data may be generated by any of the GEN subroutines such as GEN01 which reads an audio data file into a wavetable. This enable a sampled sound to be used as the source for the grains. Up to 128 voices are implemented internally. The maximum number of voices can be increased by redefining the variable MAXVOICE in the grain4.h file. granule has a build-in random number generator to handle all the random offset parameters. Thresholding is also implemented to scan the source function table at initialization stage. This facilitates features such as skipping silence passage between sentences. !granule spec chord grainSize ftabspec% -- secondary parameters. We can use def to get the defaults. chord :: [D] -- the list of pitch factors to scale the original sound. It can be up to 4 items long. This parameters allows us to create a chords out of grains. grainSize -- grain size in sec.ftab - table with sampled sound.granulei that is defined on stereo audio files. We provide the filename instead of table. The rest is the same. granulef that is defined on mono audio files. We provide the filename instead of table. The rest is the same.Synchronous granular synthesis.syncgrain implements synchronous granular synthesis. The source sound for the grains is obtained by reading a function table containing the samples of the source waveform. For sampled-sound sources, GEN01 is used. syncgrain will accept deferred allocation tables. 4syncgrain spec graidDuration timeScale PitchSig ftabspec - secondary params (use def to get the defaults) graidDuration! - duration of grains in seconds. timeScale - tempo scaling factor.PitchSig - pitch scaling factor.ftab - table with sampled sound.)The syncgrain with randomized parameters.qsyncgrain that is defined on stereo audio files. We provide the filename instead of table. The rest is the same.osyncgrain that is defined on mono audio files. We provide the filename instead of table. The rest is the same.trndSyncgrain that is defined on stereo audio files. We provide the filename instead of table. The rest is the same.rrndSyncgrain that is defined on mono audio files. We provide the filename instead of table. The rest is the same.:Simple sndwarp with scaling mode (corresponds to Csound's  initmode == 0). !sndwarp spec resample speed ftab spec - secondary params (use def to get the defaults)resample - the factor by which to change the pitch of the sound. For example, a value of 2 will produce a sound one octave higher than the original. The timing of the sound, however, will not be altered.speed: - the factor by which to change the tempo of the sound. ftab -- table with the samplesStereo version of the sndwarp.oSndwarp that is defined on stereo audio files. We provide the filename instead of table. The rest is the same.mSndwarp that is defined on mono audio files. We provide the filename instead of table. The rest is the same.(The simple sndwarp with pointer (Csound  initmode = 1).  sndwarp spec resample ftab ptrspec - secondary params (use def to get the defaults)resample - the factor by which to change the pitch of the sound. For example, a value of 2 will produce a sound one octave higher than the original. The timing of the sound, however, will not be altered.ftab -- table with the samplesptr, - pointer to read the table (in seconds). Stereo version of  ptrSndwarp.rptrSndwarp that is defined on stereo audio files. We provide the filename instead of table. The rest is the same.pptrSndwarp that is defined on mono audio files. We provide the filename instead of table. The rest is the same.8Reimplementation of fof2 opcode for stereo audio files.5Reimplementation of fof2 opcode for mono audio files. Reimplementation of fof2 opcode.GGranular delay effect for fof2. Good values for grain rate and size are grainRate = 25 grainSize = 2.5Granular delay effect for grainy.Granular delay effect for  rndGrainy.Granular delay effect for sndwarp.Granular delay effect for  syncgrain.Granular delay effect for  rndSyncgrain.Granular delay effect for  partikkel.Granular effect for grainy.Granular effect for  rndGrainy.Granular effect for sndwarp.Granular effect for  syncgrain.Granular effect for  rndSyncgrain.Granular effect for  partikkel.Granular effect for fof2.Granular synthesizer with "per grain" control over many of its parameters. Has a sync input to sychronize its internal grain scheduler clock to an external clock source.@partikkel was conceived after reading Curtis Roads' book  Microsound , and the goal was to create an opcode that was capable of all time-domain varieties of granular synthesis described in this book. The idea being that most of the techniques only differ in parameter values, and by having a single opcode that can do all varieties of granular synthesis makes it possible to interpolate between techniques. Granular synthesis is sometimes dubbed particle synthesis, and it was thought apt to name the opcode partikkel to distinguish it from other granular opcodes. a1 [, a2, a3, a4, a5, a6, a7, a8] partikkel agrainfreq, \ kdistribution, idisttab, async, kenv2amt, ienv2tab, ienv_attack, \ ienv_decay, ksustain_amount, ka_d_ratio, kduration, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, ktraincps, knumpartials, kchroma, \ ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, \ kwaveform4, iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains \ [, iopcode_id] csound doc: 1http://www.csounds.com/manual/html/partikkel.htmlSynchronous granular synthesis. syncgrain implements synchronous granular synthesis. The source sound for the grains is obtained by reading a function table containing the samples of the source waveform. For sampled-sound sources, GEN01 is used. syncgrain will accept deferred allocation tables. Wasig syncgrain kamp, kfreq, kpitch, kgrsize, kprate, ifun1, \ ifun2, iolaps csound doc: 1http://www.csounds.com/manual/html/syncgrain.html4A more complex granular synthesis texture generator.VThe granule unit generator is more complex than grain, but does add new possibilities. ares granule xamp, ivoice, iratio, imode, ithd, ifn, ipshift, igskip, \ igskip_os, ilength, kgap, igap_os, kgsize, igsize_os, iatt, idec \ [, iseed] [, ipitch1] [, ipitch2] [, ipitch3] [, ipitch4] [, ifnenv] csound doc: /http://www.csounds.com/manual/html/granule.html]Reads a mono sound sample from a table and applies time-stretching and/or pitch modification.sndwarp reads sound samples from a table and applies time-stretching and/or pitch modification. Time and frequency modification are independent from one another. For example, a sound can be stretched in time while raising the pitch! sares [, ac] sndwarp xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, \ irandw, ioverlap, ifn2, itimemode csound doc: /http://www.csounds.com/manual/html/sndwarp.html_Reads a stereo sound sample from a table and applies time-stretching and/or pitch modification.sndwarpst reads stereo sound samples from a table and applies time-stretching and/or pitch modification. Time and frequency modification are independent from one another. For example, a sound can be stretched in time while raising the pitch! ar1, ar2 [,ac1] [, ac2] sndwarpst xamp, xtimewarp, xresample, ifn1, \ ibeg, iwsize, irandw, ioverlap, ifn2, itimemode csound doc: 1http://www.csounds.com/manual/html/sndwarpst.htmlxZProduces sinusoid bursts including k-rate incremental indexing with each successive burst.EAudio output is a succession of sinusoid bursts initiated at frequency xfund with a spectral peak at xform. For xfund above 25 Hz these bursts produce a speech-like formant with spectral characteristics determined by the k-input parameters. For lower fundamentals this generator provides a special form of granular synthesis. ~ares fof2 xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur, kphs, kgliss [, iskip] csound doc: ,http://www.csounds.com/manual/html/fof2.htmlyz{|}~x}}Wy  z{|}~xNone24&The stereo signal processing function.Widget that represents a mixer.DWidget that represents a mixer with horizontal grouping of elements.ETransforms the mono signal to the stereo input for the mixer widget.Creates a widget that represents a stereo signal processing function. The parameters of the widget are updated with sliders. For example let's create a simple gain widget. It can be encoded like this: uiGain :: Bool -> Double -> Source FxFun uiGain isOn gain = fxBox "Gain" fx isOn [("gain", gain)] where fx :: Sig -> Sig2 -> Sig2 fx = mul+Let's look at the arguments of the function fxBox name fx isOn argsname -- is the name of the widgetfx1 -- is signal processing function (see the class FxUI). isOn& -- whether widget in the active stateargsD -- list of initial values for arguments and names of the arguments..It's cool to set the color of the widget with fxColorD function. we can make our widgets much more intersting to look at.Colors the source widgets.-Scales the gui for signal processing widgets.iGroups the signal processing widgets. The functions are composed the visuals are grouped horizontaly.gGroups the signal processing widgets. The functions are composed the visuals are grouped verticaly.3Applies a function to a signal processing function.(The distortion widget. The arguments are +uiDistort isOn levelOfDistortion drive tone $The chorus widget. The arguments are #uiChorus isOn mix rate depth width !%The flanger widget. The arguments are ,uiFlanger isOn mix feedback rate depth delay"$The phaser widget. The arguments are /uiPhaser isOn mix feedback rate depth frequency##The delay widget. The arguments are (uiDelay isOn mix feedback delayTime tone$.The simplified delay widget. The arguments are +uiEcho isOn maxDelayTime delayTime feedback%%The pair of low and high pass filters 5uiFilter isOn lowPassfrequency highPassFrequency gain&%The reverb widget. The arguments are: uiReverb mix depth'"The gain widget. The arguments are uiGain isOn amountOfGain(2The filtered white noize widget. The arguments are .uiWhite isOn centerFreqOfFilter amountOfNoize )1The filtered pink noize widget. The arguments are -uiPink isOn centerFreqOfFilter amountOfNoize *NThe constructor for signal processing functions with no arguments (controlls).+The reverb for room.,The reverb for hall.-The reverb for magic cave.,The widget for selecting a midi instrument. .0the widget for mixing in a signal to the signal./YA mixer widget represented as an effect. The effect sums the signals with given wieghts.2A widget with four standard waveforms: pure tone, triangle, square and sawtooth. The last parameter is a default waveform (it's set at init time).3Slider for master volume4Knob for master volume:      !"#$%&'()*+,-./01234.      !"#$%&'()*+,-./01234. !"#$%&'()*+,-./     01234/      !"#$%&'()*+,-./012341None    defghijklmnopqrstuvwxyz{|}~      !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~      !"#$%&'()*+,-./01234None 53D matrix (cube of vectors)62D matrix (grid of vecotrs)7 1D matrix8 Hvs vector9dAllows one-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables.ihvs1 allows one-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. R hvs1 kx, inumParms, inumPointsX, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] csound doc: ,http://www.csounds.com/manual/html/hvs1.html:dAllows two-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables.ihvs2 allows two-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. c hvs2 kx, ky, inumParms, inumPointsX, inumPointsY, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] csound doc: ,http://www.csounds.com/manual/html/hvs2.html;fAllows three-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables.khvs3 allows three-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. t hvs3 kx, ky, kz, inumParms, inumPointsX, inumPointsY, inumPointsZ, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] csound doc: ,http://www.csounds.com/manual/html/hvs3.html<POne dimensional Hyper vectorial synthesis. We can provide a list of vectors (of lists but the same length for all items is assumed) and a signal that ranges from 0 to 1. It interpolates between vectors in the list. As a result we get a n interpolated vector. It's a list but the actual length equals to the length of input vectors.bAn example. We can set the center frequency and resonance of the filter with the single parameter: let f = hvs1 [[100, 0.1], [300, 0.1], [600, 0.5], [800, 0.9]] dac $ lift1 (\x -> fmap (\[cps, q] -> mlp cps q (saw 110)) $ f x) (uknob 0.5)TNotice the exact pattern match with the list in the argument of the lambda function: '\[cps, q] -> mlp cps q (saw 110)) $ f x=It's determined by the length of the items in the input list.=Two dimensional Hyper vectorial synthesis. Now we provide a list of lists of vectors. The length of all vectors should be the same but there is no limit for the number! So that's how we can control a lot of parameters with pair of signals. The input 2D atrix is the grid of samples. It finds the closest four points in the grid and interpolates between them (it's a weighted sum). hvs2 matrix (x, y)(The usage is the same as in the case of hvs1 . An example: Vg = hvs2 [[[100, 0.1, 0.3], [800, 0.1, 0.5], [1400, 0.1, 0.8]], [[100, 0.5, 0.3], [800, 0.5, 0.5], [1400, 0.5, 0.8]], [[100, 0.8, 0.3], [800, 0.8, 0.5], [1400, 0.8, 0.8]]] main = dac $ do (g1, kx) <- uknob 0.5 (g2, ky) <- uknob 0.5 [cfq, q, w] <- g (kx, ky) panel $ hor [g1, g2] at (mlp cfq q) $ fmap (cfd w (saw 110)) (white)>The three dimensional 56789:;<=> 56789:;<=> 8765<=>9:; 56789:;<=>None p  !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~      !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~      !"#$%&'()*+,-./0123456789:;<=>?@ABCDEFGHIJKLMNOPQRSTUVWXYZ[\]^_`abcdefghijklmnopqrstuvwxyz{|}~                           ! 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" # $ % & ' ( ) * + , - . / 0 1 2 3 4 5 % 6 7 8 9 : ; & < = > ? @ A B C D E F G H I J K L M N O P Q R S T S U S V S W S X S Y S Z S [ S \ S ] S ^ S _ S ` a b a c a d a e a f a g a h i j i k i l i m n o n p n q n r n s n t n u n v n w n x n y n z n { n | n } n ~ n  n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n n                                          !  "  #  $  %  &  '  (  )  *  +  ,  -  .  /  0  1  2  3  4  5  6  7  8  9  :  ;  <  =  >  ?  @  A  B  C  D  E  F  G  H  I  J  K  L  M  N  O  P  Q  R  S  T  U  V  W  X  Y  Z  [  \  ]  ^  _  `  a  b  c  d  e  f  g  h  i  j  k  l  m  n  o  p  q  r  s t u t v t w t x t y t z t { t | t } t ~ t  t t t t t t t t t t t t t t t t t t          csound-expression-4.6.1Csound.Control.InstrCsound.OptionsCsound.Control.MidiCsound.Control.Gui.WidgetCsound.Control.GuiCsound.Air.WaveCsound.Control.SfCsound.Control.ChannelCsound.Control.Osc Csound.TypesCsound.Control.EvtCsound.Control.SE Csound.TabCsound.Control.Gui.PropsCsound.Control.Gui.Layout Csound.IOCsound.SigSpaceCsound.Air.FilterCsound.Air.EnvelopeCsound.Air.MiscCsound.Air.SpecCsound.Air.WavCsound.Air.SegCsound.Air.Sampler Csound.Air.FxCsound.Air.GranularCsound.Air.LiveCsound.Air.Hvs!Csound.Control.Overload.SpecInstr!Csound.Control.Overload.MidiInstr Csound.BasemidimidinCsound.Control.Overload.OutsCsound.Control.OverloadCsound.Opcode.BasicmetrochangedtriggeraccumECsound.Render.Mix renderCsdplayCsddacCsound.Controlbuzz tabResolution CsdOptions Csound.AiroscBy Csound.Opcodelinsegexpseglinen!csound-expression-dynamic-0.1.4.1Csound.Dynamic.Types.EventListCsdEventsingleCsdEventtoCsdEventListCsdScocsdEventListNotescsdEventListDur CsdEventListCsound.Dynamic.Types.Flags flagsVerbatimconfigdisplaysrtmidimidiRTmidiIO pulseAudiortaudioidTagsaudioFileOutputFlagsdithernopeaksnosoundinputoutput formatType formatSamplesAudioFileOutputNoHeader RewriteHeader FormatHeaderBit24AlawUcharSchar FloatSamplesUlawShortLong FormatSamples TriangularUniformDitherAiffAuAvrCafFlacHtkIrcamMat4Mat5NisPafPvfRawSd2SdsSvxVocW64WavWavexXi FormatTypeidTitle idSoftwareidDate idCopyright idCommentidArtistIdTags PortAudioAlsa jackOutport jackInport jackClientJackMme CoreAudio NoRtaudioRtaudiopaInputpaOutputpaServer PulseAudioterminateOnMidirawControllerMode muteTracks midiOutFilemidiFileMidiIO midiOutDevicemidiVelocityAmp midiVelocity midiKeyPch midiKeyOct midiKeyCpsmidiKey midiDeviceMidiRTPortMidiAlsaMidiCoreMidiMmeMidi WinmmeMidi VirtualMidiNoRtmidiRtmidi listOpcodesdisplayVerbosemsgColor mBenchmarksmColoursmDb mWarningsmRangemAmps messageLeveldisplayHeartbeat displayMode csdLineNumsDisplays NoDisplayPostScriptDisplay AsciiDisplay DisplayModesetTempo skipSecondsstrsetNschedNumsetSchedsmacroomacroscoreInnewSrnewKrioBufhwBufConfigcsound-expression-opcodes-0.0.2 Csound.Typed.Opcode.RealtimeMIDIcpsmidiampmidictrl7!csound-expression-typed-0.0.7.3.1Csound.Typed.Gui.WidgetkeyIn setToggle setToggleSigmeter setNumeric butBankSig1butBank1 butBankSigbutBank toggleSigtogglebuttonboxnumeric sliderBanksliderrollerknobjoycountcountSigsetTitledisplay sinkSourcesinksourcewidgetmscamvermhor mapGuiSource mapSource keyPanelBypanelBykeyPanelpanel keyPanelspanelsInputOutputInnerWidgetSinkSource SinkSourceDisplayCsound.Typed.ControlsetDurCsound.Typed.Control.Evtscheds_trigs_ schedHarpsBy schedHarpsschedsByschedsretrigstrigsBytrigsCsound.Typed.Control.Midipgmidi_midin_midi_pgmidiCsound.Typed.Control.Mixmix_mixBymixeffsco_scoMixCsound.Typed.Control.VcobloscsqrpulsetriisawsawCsound.Typed.Control.Sf2SfCsound.Typed.Control.Channel chnSetStr chnSetCtrl chnSetSigchnSetD chnGetStr chnGetSig chnGetCtrlchnGetDCsound.Typed.Control.OscsendOsc listenOscinitOscCsound.Typed.TypeswithSeedwithTabwithTabswithSigwithSigswithDwithDs withInitsCsound.Typed.Types.Evtsyncstepper evtToBoolsnapssnapshot filterAccumE filterAccumSEaccumSEfilterSEfilterEsigToEvt boolToEvtrunEvtEvtBamSnapCsound.Typed.Control.SERefglobalSensorsSEnewGlobalSERef sensorsSE modifySERefmixSERefnewSERef readSERef writeSERefSERefCsound.Typed.Types.TuplecaseArg guardedArgifArg caseTuple guardedTupleifTuplear8ar6ar4ar2ar1makeTupleMethods tupleMethodsTupleSigsSig2Sig3Sig4Sig5Sig6Sig8ArgCsound.Typed.Types.PrimftcpsftsrftchnlsftlennsampboolSigwhileDountilDowhenswhen1mod'div'rem'quot'round'int'frac'floor'ceil'sigirkrar getBlockSizegetControlRate getSampleRateidurtextintdouble forceNormskipNormunitSigDStrSpecWspecBoolSigBoolDUnitTabfromEtoGEfromGEValSigOrDCsound.Typed.GlobalState.SESECsound.Typed.GlobalState.GEMsgPressReleaseKeyEvtCharKeyF1F2F3F4F5F6F7F8F9F10F11F12ScrollCapsLook LeftShift RightShiftLeftCtrl RightCtrlEnterLeftAltRightAlt LeftWinKey RightWinKey BackspaceArrowUp ArrowLeft ArrowRight ArrowDownInsertHomePgUpDeleteEndPgDownNumLockNumDivNumMulNumSubNumHome NumArrowUpNumPgUp NumArrowLeftNumSpace NumArrowRightNumEnd NumArrowDown NumPgDownNumInsNumDelNumEnterNumPlusNum7Num8Num9Num4Num5Num6Num1Num2Num3Num0NumDotKeyCsound.Typed.Gui.Gui setKnobType setOrient setButtonType setTextType setSliderType setEmphasis setFontType setFontSize setTextColor setColors setColor2 setColor1 setBoxType setMaterialsetLabel setBorder 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pvstencilpvspitchpvsoutpvsoscpvsmorphpvsmoothpvsmixpvsmaskapvslockpvsinitpvsinfopvsinpvsifdpvshiftpvsgain pvsfwritepvsftwpvsftr pvsfreezepvsfread pvsfilterpvsdisp pvsdiskinpvsdemixpvscrosspvscentpvscale pvsbufread2 pvsbufread pvsbufferpvsblurpvsbinpvsbandrpvsbandppvsarppvsanalpvsadsynpartialsbinitspectrumspecsumspecscalspecptrkspechistspecfiltspecdispspecdiffspecaddmlpslotlpresonlpreadlpinterplpfresonvpvoc tablexsegtablesegpvreadpvocpvinterppvcross pvbufreadpvadd ktablesegCsound.Typed.Opcode.Strings strupperkstrupperstrtolkstrtolstrtodkstrtod strlowerkstrlowerstrcharkstrcharstrsubkstrsub strrindexk strrindexstrlenkstrlen strindexkstrindexstrcpykstrcpystrcmpkstrcmpstrcatkstrcatsprintfksprintfputsstrsetstrgetCsound.Typed.Opcode.Vectorialvcellacellvrandivrandhvportvecdelayvdelaykvwrapvmirrorvlimitvlinsegvexpsegvsubv_ivsubvvpowv_ivpowvvmultv_ivmultvvmapvexpv_ivexpvvdivv_ivdivvvcopy_ivcopyvaddv_ivaddvvpow_ivpowvmult_ivmultvexp_ivexpvadd_ivaddvtabwkvtabwivtabwavtablewkvtablewivtablewavtablekvtableivtableavtable1kvtabkvtabivtaba"Csound.Typed.Opcode.ZakPatchSystemzkwmzkwzkrzkmodzkclziwmziwzirzawmzawzargzarzamodzakinitzacl!Csound.Typed.Opcode.PluginHosting vstprogset vstparamget vstparamsetvstnote vstmidioutvstinitvstinfovstedit vstbankload vstaudiogvstaudiodssilistdssiinitdssictls dssiaudio dssiactivateCsound.Typed.Opcode.Networkstsend socksendssocksendstrecv sockrecvssockrecv remoteport!Csound.Typed.Opcode.RemoteOpcodesmidremot midglobalinsremot insglobal Csound.Typed.Opcode.MixerOpcodesmixerSetLevel_i mixerSetLevel mixerSend mixerReceive mixerGetLevel mixerClear*Csound.Typed.Opcode.ImageProcessingOpcodes imagesize imagesetpixel imagesave imageload imagegetpixel imagefree imagecreate!Csound.Typed.Opcode.Miscellaneous tableshufflei tableshufflesystemsystem_ipwd modmatrix fareylenifareylendata-default-class-0.0.1Data.Default.ClassDefaultdef