This protected Filter subclass implements a
two-pole, two-zero digital filter. A method
is provided for creating a resonance in the
frequency response while maintaining a constant
filter gain.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a generic structure which
can be used to create a wide range of filters.
It can function independently or be subclassed
to provide more specific controls based on a
particular filter type.
In particular, this class implements the standard
difference equation:
a[0]*y[n] = b[0]*x[n] + ... + b[nb]*x[n-nb] -
a[1]*y[n-1] - ... - a[na]*y[n-na]
If a[0] is not equal to 1, the filter coeffcients
are normalized by a[0].
The \e gain parameter is applied at the filter
input and does not affect the coefficient values.
The default gain value is 1.0. This structure
results in one extra multiply per computed sample,
but allows easy control of the overall filter gain.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This protected Filter subclass implements
a one-pole digital filter. A method is
provided for setting the pole position along
the real axis of the z-plane while maintaining
a constant peak filter gain.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This protected Filter subclass implements
a two-pole digital filter. A method is
provided for creating a resonance in the
frequency response while maintaining a nearly
constant filter gain.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This protected Filter subclass implements
a one-zero digital filter. A method is
provided for setting the zero position
along the real axis of the z-plane while
maintaining a constant filter gain.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This protected Filter subclass implements
a two-zero digital filter. A method is
provided for creating a "notch" in the
frequency response while maintaining a
constant filter gain.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This protected Filter subclass implements
a one-pole, one-zero digital filter. A
method is provided for creating an allpass
filter with a given coefficient. Another
method is provided to create a DC blocking filter.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
Resonant low pass filter. 2nd order Butterworth.
(In the future, this class may be expanded so that order
and type of filter can be set).
Resonant high pass filter. 2nd order Butterworth.
(In the future, this class may be expanded so that order
and type of filter can be set).
Band pass filter. 2nd order Butterworth.
(In the future, this class may be expanded so that order
and type of filter can be set).
Band reject filter. 2nd order Butterworth.
(In the future, this class may be expanded so that order
and type of filter can be set).
Resonance filter. BiQuad with equal-gain zeros.
keeps gain under control independent of frequency.
Filter basic base class, with .freq, .Q, .set.
default limiter values:
slopeAbove = 0.1
slopeBelow = 1.0
thresh = 0.5
attackTime = 5 ms
releaseTime = 300 ms
externalSideInput = 0 (false)
default compressor values:
slopeAbove = 0.5
slopeBelow = 1.0
thresh = 0.5
attackTime = 5 ms
releaseTime = 300 ms
externalSideInput = 0 (false)
default expander values:
slopeAbove = 2.0
slopeBelow = 1.0
thresh = 0.5
attackTime = 20 ms
releaseTime = 400 ms
externalSideInput = 0 (false)
default noise gate values:
slopeAbove = 1.0
slopeBelow = 10000000
thresh = 0.1
attackTime = 11 ms
releaseTime = 100 ms
externalSideInput = 0 (false)
default ducker values:
slopeAbove = 0.5
slopeBelow = 1.0
thresh = 0.1
attackTime = 100 ms
releaseTime = 1000 ms
externalSideInput = 1 (true)
Note that the input to sideInput determines the level of
gain, not the direct signal input to Dyno.
Ported from rtcmix. See http://www.music.columbia.edu/cmix/makegens.html for more information on the GenX family of UGens. Currently coefficients past the 100th are ignored.
Lookup can either be done using the lookup() function, or by driving the table with an input UGen, typically a Phasor. For an input signal between [ -1, 1 ], using the absolute value for [ -1, 0 ), GenX will output the table value indexed by the current input.
Constructs a lookup table composed of sequential exponential curves. For a table with N curves, starting value of y', and value yn for lookup index xn, set the coefficients to [ y', y0, x0, ..., yN-1, xN-1 ]. Note that there must be an odd number of coefficients. If an even number of coefficients is specified, behavior is undefined. The sum of xn for 0 ≤ n < N must be 1. yn = 0 is approximated as 0.000001 to avoid strange results arising from the nature of exponential curves.
Constructs a lookup table composed of sequential line segments. For a table with N lines, starting value of y', and value yn for lookup index xn, set the coefficients to [ y', y0, x0, ..., yN-1, xN-1 ]. Note that there must be an odd number of coefficients. If an even number of coefficients is specified, behavior is undefined. The sum of xn for 0 ≤ n < N must be 1.
Constructs a lookup table of partials with specified amplitudes, phases, and harmonic ratios to the fundamental. Coefficients are specified in triplets of [ ratio, amplitude, phase ] arranged in a single linear array.
Constructs a lookup table of harmonic partials with specified amplitudes. The amplitude of partial n is specified by the nth element of the coefficients. For example, setting coefs to [ 1 ] will produce a sine wave.
Constructs a Chebyshev polynomial wavetable with harmonic partials of specified weights. The weight of partial n is specified by the nth element of the coefficients.
Primarily used for waveshaping, driven by a SinOsc instead of a Phasor. See http://crca.ucsd.edu/~msp/techniques/v0.08/book-html/node74.html and http://en.wikipedia.org/wiki/Distortion_synthesis for more information.
Constructs a wavetable composed of segments of variable times, values, and curvatures. Coefficients are specified as a single linear array of triplets of [ time, value, curvature ] followed by a final duple of [ time, value ] to specify the final value of the table. time values are expressed in unitless, ascending values. For curvature equal to 0, the segment is a line; for curvature less than 0, the segment is a convex curve; for curvature greater than 0, the segment is a concave curve.
LiSa provides basic live sampling functionality. An internal buffer stores samples chucked to LiSa's input. Segments of this buffer can be played back, with ramping and speed/direction control. Multiple voice facility is built in, allowing for a single LiSa object to serve as a source for sample layering and granular textures. LiSa is mono by default, but one can create multi-channel LiSas using LiSa10 (10-channel) and, as of 1.4.0.2, LiSa2 (stereo), LiSa4 (quad), LiSa6 (6-channel), LiSa8 (8-channel), and LiSa16 (16-channel). by Dan Trueman (2007) see LiSa Examples wiki for more, and also a slowly growing tutorial | mirror also see LiSa examples in the ChucK distribution
The following UGens subclass StkInstrument:
- BandedWG
- BlowBotl
- BlowHole
- Bowed
- Brass
- Clarinet
- Flute
- FM (and all its subclasses: BeeThree, FMVoices,
HevyMetl, PercFlut, Rhodey, TubeBell, Wurley)
- Mandolin
- ModalBar
- Moog
- Saxofony
- Shakers
- Sitar
- StifKarp
- VoicForm
This class uses banded waveguide techniques to
model a variety of sounds, including bowed
bars, glasses, and bowls. For more
information, see Essl, G. and Cook, P. "Banded
Waveguides: Towards Physical Modelling of Bar
Percussion Instruments", Proceedings of the
1999 International Computer Music Conference.
Control Change Numbers:
- Bow Pressure = 2
- Bow Motion = 4
- Strike Position = 8 (not implemented)
- Vibrato Frequency = 11
- Gain = 1
- Bow Velocity = 128
- Set Striking = 64
- Instrument Presets = 16
- Uniform Bar = 0
- Tuned Bar = 1
- Glass Harmonica = 2
- Tibetan Bowl = 3
by Georg Essl, 1999 - 2002.
Modified for Stk 4.0 by Gary Scavone.
This class implements a helmholtz resonator
(biquad filter) with a polynomial jet
excitation (a la Cook).
Control Change Numbers:
- Noise Gain = 4
- Vibrato Frequency = 11
- Vibrato Gain = 1
- Volume = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
register hole and one tonehole.
This class is based on the clarinet model,
with the addition of a two-port register hole
and a three-port dynamic tonehole
implementation, as discussed by Scavone and
Cook (1998).
In this implementation, the distances between
the reed/register hole and tonehole/bell are
fixed. As a result, both the tonehole and
register hole will have variable influence on
the playing frequency, which is dependent on
the length of the air column. In addition,
the highest playing freqeuency is limited by
these fixed lengths.
This is a digital waveguide model, making its
use possibly subject to patents held by Stanford
University, Yamaha, and others.
Control Change Numbers:
- Reed Stiffness = 2
- Noise Gain = 4
- Tonehole State = 11
- Register State = 1
- Breath Pressure = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a bowed string model, a
la Smith (1986), after McIntyre, Schumacher,
Woodhouse (1983).
This is a digital waveguide model, making its
use possibly subject to patents held by
Stanford University, Yamaha, and others.
Control Change Numbers:
- Bow Pressure = 2
- Bow Position = 4
- Vibrato Frequency = 11
- Vibrato Gain = 1
- Volume = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a simple brass instrument
waveguide model, a la Cook (TBone, HosePlayer).
This is a digital waveguide model, making its
use possibly subject to patents held by
Stanford University, Yamaha, and others.
Control Change Numbers:
- Lip Tension = 2
- Slide Length = 4
- Vibrato Frequency = 11
- Vibrato Gain = 1
- Volume = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a simple clarinet
physical model, as discussed by Smith (1986),
McIntyre, Schumacher, Woodhouse (1983), and
others.
This is a digital waveguide model, making its
use possibly subject to patents held by Stanford
University, Yamaha, and others.
Control Change Numbers:
- Reed Stiffness = 2
- Noise Gain = 4
- Vibrato Frequency = 11
- Vibrato Gain = 1
- Breath Pressure = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a simple flute
physical model, as discussed by Karjalainen,
Smith, Waryznyk, etc. The jet model uses
a polynomial, a la Cook.
This is a digital waveguide model, making its
use possibly subject to patents held by Stanford
University, Yamaha, and others.
Control Change Numbers:
- Jet Delay = 2
- Noise Gain = 4
- Vibrato Frequency = 11
- Vibrato Gain = 1
- Breath Pressure = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class inherits from PluckTwo and uses
"commuted synthesis" techniques to model a
mandolin instrument.
This is a digital waveguide model, making its
use possibly subject to patents held by
Stanford University, Yamaha, and others.
Commuted Synthesis, in particular, is covered
by patents, granted, pending, and/or
applied-for. All are assigned to the Board of
Trustees, Stanford University. For
information, contact the Office of Technology
Licensing, Stanford University.
Control Change Numbers:
- Body Size = 2
- Pluck Position = 4
- String Sustain = 11
- String Detuning = 1
- Microphone Position = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a number of different
struck bar instruments. It inherits from the
Modal class.
Control Change Numbers:
- Stick Hardness = 2
- Stick Position = 4
- Vibrato Gain = 11
- Vibrato Frequency = 7
- Direct Stick Mix = 1
- Volume = 128
- Modal Presets = 16
- Marimba = 0
- Vibraphone = 1
- Agogo = 2
- Wood1 = 3
- Reso = 4
- Wood2 = 5
- Beats = 6
- Two Fixed = 7
- Clump = 8
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This instrument uses one attack wave, one
looped wave, and an ADSR envelope (inherited
from the Sampler class) and adds two sweepable
formant (FormSwep) filters.
Control Change Numbers:
- Filter Q = 2
- Filter Sweep Rate = 4
- Vibrato Frequency = 11
- Vibrato Gain = 1
- Gain = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a "hybrid" digital
waveguide instrument that can generate a
variety of wind-like sounds. It has also been
referred to as the "blowed string" model. The
waveguide section is essentially that of a
string, with one rigid and one lossy
termination. The non-linear function is a
reed table. The string can be "blown" at any
point between the terminations, though just as
with strings, it is impossible to excite the
system at either end. If the excitation is
placed at the string mid-point, the sound is
that of a clarinet. At points closer to the
"bridge", the sound is closer to that of a
saxophone. See Scavone (2002) for more details.
This is a digital waveguide model, making its
use possibly subject to patents held by Stanford
University, Yamaha, and others.
Control Change Numbers:
- Reed Stiffness = 2
- Reed Aperture = 26
- Noise Gain = 4
- Blow Position = 11
- Vibrato Frequency = 29
- Vibrato Gain = 1
- Breath Pressure = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
PhISEM (Physically Informed Stochastic Event
Modeling) is an algorithmic approach for
simulating collisions of multiple independent
sound producing objects. This class is a
meta-model that can simulate a Maraca, Sekere,
Cabasa, Bamboo Wind Chimes, Water Drops,
Tambourine, Sleighbells, and a Guiro.
PhOLIES (Physically-Oriented Library of
Imitated Environmental Sounds) is a similar
approach for the synthesis of environmental
sounds. This class implements simulations of
breaking sticks, crunchy snow (or not), a
wrench, sandpaper, and more.
Control Change Numbers:
- Shake Energy = 2
- System Decay = 4
- Number Of Objects = 11
- Resonance Frequency = 1
- Shake Energy = 128
- Instrument Selection = 1071
- Maraca = 0
- Cabasa = 1
- Sekere = 2
- Guiro = 3
- Water Drops = 4
- Bamboo Chimes = 5
- Tambourine = 6
- Sleigh Bells = 7
- Sticks = 8
- Crunch = 9
- Wrench = 10
- Sand Paper = 11
- Coke Can = 12
- Next Mug = 13
- Penny + Mug = 14
- Nickle + Mug = 15
- Dime + Mug = 16
- Quarter + Mug = 17
- Franc + Mug = 18
- Peso + Mug = 19
- Big Rocks = 20
- Little Rocks = 21
- Tuned Bamboo Chimes = 22
by Perry R. Cook, 1996 - 1999.
This class implements a sitar plucked string
physical model based on the Karplus-Strong
algorithm.
This is a digital waveguide model, making its
use possibly subject to patents held by
Stanford University, Yamaha, and others.
There exist at least two patents, assigned to
Stanford, bearing the names of Karplus and/or
Strong.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a simple plucked string
algorithm (Karplus Strong) with enhancements
(Jaffe-Smith, Smith, and others), including
string stiffness and pluck position controls.
The stiffness is modeled with allpass filters.
This is a digital waveguide model, making its
use possibly subject to patents held by
Stanford University, Yamaha, and others.
Control Change Numbers:
- Pickup Position = 4
- String Sustain = 11
- String Stretch = 1
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This instrument contains an excitation singing
wavetable (looping wave with random and
periodic vibrato, smoothing on frequency,
etc.), excitation noise, and four sweepable
complex resonances.
Measured formant data is included, and enough
data is there to support either parallel or
cascade synthesis. In the floating point case
cascade synthesis is the most natural so
that's what you'll find here.
Control Change Numbers:
- Voiced/Unvoiced Mix = 2
- Vowel/Phoneme Selection = 4
- Vibrato Frequency = 11
- Vibrato Gain = 1
- Loudness (Spectral Tilt) = 128
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
Phoneme Names:
"eee" "ihh" "ehh" "aaa"
"ahh" "aww" "ohh" "uhh"
"uuu" "ooo" "rrr" "lll"
"mmm" "nnn" "nng" "ngg"
"fff" "sss" "thh" "shh"
"xxx" "hee" "hoo" "hah"
"bbb" "ddd" "jjj" "ggg"
"vvv" "zzz" "thz" "zhh"
This class controls an arbitrary number of
waves and envelopes, determined via a
constructor argument.
Control Change Numbers:
- Control One = 2
- Control Two = 4
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a simple 4 operator
topology, also referred to as algorithm 8 of
the TX81Z.
\code
Algorithm 8 is :
1 --.
2 -\|
+-> Out
3 -/|
4 --
\endcode
Control Change Numbers:
- Operator 4 (feedback) Gain = 2 (.controlOne)
- Operator 3 Gain = 4 (.controlTwo)
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements 3 carriers and a common
modulator, also referred to as algorithm 6 of
the TX81Z.
\code
Algorithm 6 is :
/->1 -\
4-|-->2 - +-> Out
\->3 -/
\endcode
Control Change Numbers:
- Vowel = 2 (.controlOne)
- Spectral Tilt = 4 (.controlTwo)
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements 3 cascade operators with
feedback modulation, also referred to as
algorithm 3 of the TX81Z.
Algorithm 3 is : 4--\
3-->2-- + -->1-->Out
Control Change Numbers:
- Total Modulator Index = 2 (.controlOne)
- Modulator Crossfade = 4 (.controlTwo)
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements algorithm 4 of the TX81Z.
\code
Algorithm 4 is : 4->3--\
2-- + -->1-->Out
\endcode
Control Change Numbers:
- Total Modulator Index = 2 (.controlOne)
- Modulator Crossfade = 4 (.controlTwo)
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
synthesis instrument.
This class implements two simple FM Pairs
summed together, also referred to as algorithm
5 of the TX81Z.
\code
Algorithm 5 is : 4->3--\
+ --> Out
2->1--/
\endcode
Control Change Numbers:
- Modulator Index One = 2 (.controlOne)
- Crossfade of Outputs = 4 (.controlTwo)
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
synthesis instrument.
This class implements two simple FM Pairs
summed together, also referred to as algorithm
5 of the TX81Z.
\code
Algorithm 5 is : 4->3--\
+ --> Out
2->1--/
\endcode
Control Change Numbers:
- Modulator Index One = 2 (.controlOne)
- Crossfade of Outputs = 4 (.controlTwo)
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
synthesis instrument.
This class implements two simple FM Pairs
summed together, also referred to as algorithm
5 of the TX81Z.
\code
Algorithm 5 is : 4->3--\
+ --> Out
2->1--/
\endcode
Control Change Numbers:
- Modulator Index One = 2 (.controlOne)
- Crossfade of Outputs = 4 (.controlTwo)
- LFO Speed = 11
- LFO Depth = 1
- ADSR 2 & 4 Target = 128
The basic Chowning/Stanford FM patent expired
in 1995, but there exist follow-on patents,
mostly assigned to Yamaha. If you are of the
type who should worry about this (making
money) worry away.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This protected Filter subclass implements
a non-interpolating digital delay-line.
A fixed maximum length of 4095 and a delay
of zero is set using the default constructor.
Alternatively, the delay and maximum length
can be set during instantiation with an
overloaded constructor.
A non-interpolating delay line is typically
used in fixed delay-length applications, such
as for reverberation.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This Delay subclass implements a fractional-
length digital delay-line using a first-order
allpass filter. A fixed maximum length
of 4095 and a delay of 0.5 is set using the
default constructor. Alternatively, the
delay and maximum length can be set during
instantiation with an overloaded constructor.
An allpass filter has unity magnitude gain but
variable phase delay properties, making it useful
in achieving fractional delays without affecting
a signal's frequency magnitude response. In
order to achieve a maximally flat phase delay
response, the minimum delay possible in this
implementation is limited to a value of 0.5.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This Delay subclass implements a fractional-
length digital delay-line using first-order
linear interpolation. A fixed maximum length
of 4095 and a delay of zero is set using the
default constructor. Alternatively, the
delay and maximum length can be set during
instantiation with an overloaded constructor.
Linear interpolation is an efficient technique
for achieving fractional delay lengths, though
it does introduce high-frequency signal
attenuation to varying degrees depending on the
fractional delay setting. The use of higher
order Lagrange interpolators can typically
improve (minimize) this attenuation characteristic.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a echo effect.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a simple envelope
generator which is capable of ramping to
a target value by a specified \e rate.
It also responds to simple \e keyOn and
\e keyOff messages, ramping to 1.0 on
keyOn and to 0.0 on keyOff.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This Envelope subclass implements a
traditional ADSR (Attack, Decay,
Sustain, Release) envelope. It
responds to simple keyOn and keyOff
messages, keeping track of its state.
The \e state = ADSR::DONE after the
envelope value reaches 0.0 in the
ADSR::RELEASE state.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class is derived from the CLM JCRev
function, which is based on the use of
networks of simple allpass and comb delay
filters. This class implements three series
allpass units, followed by four parallel comb
filters, and two decorrelation delay lines in
parallel at the output.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class is derived from the CLM NRev
function, which is based on the use of
networks of simple allpass and comb delay
filters. This particular arrangement consists
of 6 comb filters in parallel, followed by 3
allpass filters, a lowpass filter, and another
allpass in series, followed by two allpass
filters in parallel with corresponding right
and left outputs.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class is based on some of the famous
Stanford/CCRMA reverbs (NRev, KipRev), which
were based on the Chowning/Moorer/Schroeder
reverberators using networks of simple allpass
and comb delay filters. This class implements
two series allpass units and two parallel comb
filters.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a chorus effect.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class combines random and periodic
modulations to give a nice, natural human
modulation function.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class implements a simple pitch shifter
using delay lines.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
Generates a new random number every "rate" ticks
using the C rand() function. The quality of the
rand() function varies from one OS to another.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class generates a band-limited impulse train using a
closed-form algorithm reported by Stilson and Smith in
"Alias-Free Digital Synthesis of Classic Analog Waveforms",
1996. The user can specify both the fundamental frequency
of the impulse train and the number of harmonics contained
in the resulting signal.
The signal is normalized so that the peak value is +/-1.0.
If nHarmonics is 0, then the signal will contain all
harmonics up to half the sample rate. Note, however,
that this setting may produce aliasing in the signal
when the frequency is changing (no automatic modification
of the number of harmonics is performed by the
setFrequency() function).
Original code by Robin Davies, 2005.
Revisions by Gary Scavone for STK, 2005.
This class generates a band-limited sawtooth waveform
using a closed-form algorithm reported by Stilson and
Smith in "Alias-Free Digital Synthesis of Classic Analog
Waveforms", 1996. The user can specify both the
fundamental frequency of the sawtooth and the number
of harmonics contained in the resulting signal.
If nHarmonics is 0, then the signal will contain all
harmonics up to half the sample rate. Note, however,
that this setting may produce aliasing in the signal
when the frequency is changing (no automatic modification
of the number of harmonics is performed by the setFrequency()
function).
Based on initial code of Robin Davies, 2005.
Modified algorithm code by Gary Scavone, 2005.
This class generates a band-limited square wave signal.
It is derived in part from the approach reported by
Stilson and Smith in "Alias-Free Digital Synthesis of
Classic Analog Waveforms", 1996. The algorithm implemented
in this class uses a SincM function with an even M value to
achieve a bipolar bandlimited impulse train. This
signal is then integrated to achieve a square waveform.
The integration process has an associated DC offset but that
is subtracted off the output signal.
The user can specify both the fundamental frequency of the
waveform and the number of harmonics contained in the
resulting signal.
If nHarmonics is 0, then the signal will contain all
harmonics up to half the sample rate. Note, however, that
this setting may produce aliasing in the signal when the
frequency is changing (no automatic modification of the
number of harmonics is performed by the setFrequency() function).
Based on initial code of Robin Davies, 2005.
Modified algorithm code by Gary Scavone, 2005.
This class provides input support for various
audio file formats. It also serves as a base
class for "realtime" streaming subclasses.
WvIn loads the contents of an audio file for
subsequent output. Linear interpolation is
used for fractional "read rates".
WvIn supports multi-channel data in interleaved
format. It is important to distinguish the
tick() methods, which return samples produced
by averaging across sample frames, from the
tickFrame() methods, which return pointers to
multi-channel sample frames. For single-channel
data, these methods return equivalent values.
Small files are completely read into local memory
during instantiation. Large files are read
incrementally from disk. The file size threshold
and the increment size values are defined in
WvIn.h.
WvIn currently supports WAV, AIFF, SND (AU),
MAT-file (Matlab), and STK RAW file formats.
Signed integer (8-, 16-, and 32-bit) and floating-
point (32- and 64-bit) data types are supported.
Uncompressed data types are not supported. If
using MAT-files, data should be saved in an array
with each data channel filling a matrix row.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class inherits from WvIn and provides
audio file looping functionality.
WaveLoop supports multi-channel data in
interleaved format. It is important to
distinguish the tick() methods, which return
samples produced by averaging across sample
frames, from the tickFrame() methods, which
return pointers to multi-channel sample frames.
For single-channel data, these methods return
equivalent values.
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
This class provides output support for various
audio file formats. It also serves as a base
class for "realtime" streaming subclasses.
WvOut writes samples to an audio file. It
supports multi-channel data in interleaved
format. It is important to distinguish the
tick() methods, which output single samples
to all channels in a sample frame, from the
tickFrame() method, which takes a pointer
to multi-channel sample frame data.
WvOut currently supports WAV, AIFF, AIFC, SND
(AU), MAT-file (Matlab), and STK RAW file
formats. Signed integer (8-, 16-, and 32-bit)
and floating- point (32- and 64-bit) data types
are supported. STK RAW files use 16-bit
integers by definition. MAT-files will always
be written as 64-bit floats. If a data type
specification does not match the specified file
type, the data type will automatically be
modified. Uncompressed data types are not
supported.
Currently, WvOut is non-interpolating and the
output rate is always Stk::sampleRate().
by Perry R. Cook and Gary P. Scavone, 1995 - 2002.