Index for this Document

3-dB bandwidth : 9.5 | 18.6

3dB-bandwidth : 18.7

abstraction (pd) : 24.5.1

additive synthesis : 8.6.5

affine function : 11.5

allpass condition : 16

allpass filter : 15.2

biquad case : 15.2.1

examples : 16.1

general case : 16

allpass filter design : 15.2.2

Amperes : 18.2 | 18.3

amplifier modeling : 19.7

amplitude : 14.1.1

amplitude envelope : 8.6.5

amplitude response : 2.2 | 2.3.1 | 8.2

analog filters : 18 | 22.2

allpass : 18.8

capacitor impedance : 18.2

damping ratio : 18.7.3

example : 18.1

example RC analysis : 18.4

example RLC analysis : 18.5

inductor impedance : 18.3

poles and zeros : 18.4.5

RL impulse response : 18.4.3

RLC impulse response : 18.5.4

second-order poles and zeros : 18.5.3

second-order transfer function : 18.5.2

transfer function : 18.4.2

analog prototype : 22.3.3

analytic continuation : 7.2 | 17.2 | 17.2 | 22.2.1

anticausal : 9.8

anticausal exponentials : 9.7

antiresonance frequency : 15.1.4 | 15.1.6

antiresonator : 15.1.4

antisymmetric impulse responses : 11.5

antisymmetric linear-phase filters : 11.5

banded Toeplitz filter matrix : 19.3

bandwidth : 18.7

bandwidth of a pole : 9.5

bandwidth of a pole : 18.6

bilateral z transform : 7.1

bilateral Laplace transform : 17

bilinear transformation : 22.3.1

frequency warping : 22.3.2

prototype analog filters : 22

binomial coefficient : 20.10.1

biquad filter section : 7.8.3 | 15.1.6

blocking capacitor : 15.3

boost : 15.5

Butterworth filters : 22.2

Butterworth lowpass example : 10.2.4

Butterworth lowpass filter design example : 22.2.2

canonical with respect to delay : 6.11.6

capacitor : 18.2

capacitor driving point impedance : 18.4.1

capacitors as springs : 18.2.1

carrier frequency : 8.6.3.1

carrier term : 14.3.3

carrier wave : 8.6.5

causal : 5.3 | 6.3 | 16

causal filters : 6.3

causal signal : 7.1 | 24.1

center frequency of a resonator : 15.1.3

cepstrum

complex : 9.8

minimum phase : 9.9

poles and zeros : 9.8

real : 9.8

characteristic polynomial : 20.6

circulant matrix : 19.4

clipping : 5.1

clipping dB magnitude : 23.10

coefficients, difference equation : 6.1

comb filter : 4

commutativity of series filters : 7.7.2.1

companding : 5.7

complete response : 6.12.5 | 20.3

complex amplitude : 2.4.2

complex analysis : 5.1

complex and trig identities : 14.2

complex cepstrum : 9.8

complex exponential : 2.4.1 | 7.8.4

complex filter : 5.3 | 6.1

complex numbers summary : 14.2

complex one-pole sections : 10.2.2.1

complex resonator : 15.1.5 | 18.6

complex signal : 5.1

complex sinusoid : 2.4.1

complex sinusoidal oscillator : 15.1.5

condition number : 20.10

conformal map : 22.3.2

constant peak-gain resonator : 15.6.4

constant resonance-gain resonator : 15.6.2

continuous-time complex one-pole resonator : 18.6

controllability and observability : 20.7.3

controllable modes : 20.7.1

controller canonical form : 19.6.1 | 20.7.1

convex optimization : 22.1

convolution : 7.8.10.2 | 20.1

convolution filter representation : 6.9 | 6.10

convolution is commutative : 7.7.2.1

convolution theorem for z transforms : 7.3

convolution theorem for z transforms : 7.3.2

convolution theorem for z transforms : 7.3.2

Coulombs : 18.2

cps : 14.1.1

critical damping : 18.7.2.1

critically damped : 18.7.2.1

current : 18.2

cut filter : 15.5

cut-off frequency : 2.2

cycles per second : 14.1.1

cyclic convolution : 19.4

damping constant : 18.7.2

damping factor : 18.7.2

damping ratio : 18.7.2 | 18.7.3

dB clipping : 23.10

dc blocker : 15.3

dc blocker frequency response : 15.3.1

dc blocking filter : 15.3

decay response : 6.12.4 | 6.12.4

decay time : 6.12.1

decay time-constant : 9.6

deconvolution : 7.8.10.3

degeneracy : 7.8.7

delay equalization : 8.6.4

delta function : 18.4.4

design of recursive digital filters : 22

determinant : 20.6

DFT matrix : 19.4

diagonalizing a state-space model : 20.9.1

difference equation : 6.1

differentiation theorem for Laplace transforms : 17.4.2

digital filter theory : 5

direct form filter implementation : 15.1.6

direct form filter implementations : 10.1

discrete Fourier transform (DFT) : 8.5.1

discrete time Fourier transform (DTFT) : 8.1

discrete-time sinusoid : 14.1.2

doublet : 17.5.1

driving point impedance

RLC network : 18.5.1

driving-point impedance : 18.2 | 18.3

DTFT : 8.1

Durbin recursion : 9.4.1 | 9.4.1

dynamic convolution : 5.9

dynamic range compression : 5.7

eigenvalues : 20.6 | 20.6

eigenvector : 20.9.1

electrical equivalent circuit : 17.5.1

equation error : 22.4.1

definition : 22.4.1

minimization : 22.4

equiripple : 8.6.4

error weighting function : 22.4.2

Euler's identity : 2.4

even impulse-response filter : 11.2

example elementary audio filters : 15

existence of the z transform : 7.2

explicit finite difference scheme : 6.1

exponential function summary : 14.2.1

exponential order : 17.1

exponentially swept sine analysis : 2.3.1

exponentially windowed : 17

externals (pd) : 24.5.1

factorial notation : 17.2

Farads : 18.2

Fast Fourier Transform (FFT) : 8.5.1

Faust programming language : 24

feedback coefficients : 6.1

feedback signal : 5.3

feedforward coefficients : 6.1

FFT convolution : 6.11.7 | 19.4

filter

allpass biquad : 15.2.1

allpass examples : 16.1

allpass sections : 15.2

amplitude response : 8.2

antiresonator : 15.1.4

antisymmetric impulse response : 11.5

biquad : 15.1.6

causal : 6.3

checking stability : 9.4.1

coefficients : 6.1

complete response : 6.12.5

complex : 6.1

complex one-pole resonator : 15.1.5

constant gain at resonance : 15.6.1

converting to minimum phase : 12.7

converting to parallel form : 7.8

dc blocker : 15.3

definition : 5.2 | 5.2

difference equation : 6.1

direct-form I : 6.2 | 6.5

direct-form II : 6.2 | 10.1.2

estimation from input/output data : 19.7

even impulse response : 11.2

examples : 5.3

feedback : 6.1

finite impulse response (FIR) : 6.11

first and second-order sections : 15.1

forming real second-order sections from two complex one-poles : 7.8.3

forward and backward : 11.6

frequency response : 8.1

frequency response in matlab : 8.5.1

general form of finite-order, causal, linear, time-invariant case : 6.4

graphical amplitude-response calculation from poles and zeros : 9.2

graphical phase response : 9.3

imaginary frequency response : 11.3

implementation structures : 6.2 | 10

complex resonators : 10.2.2.1

parallel second-order sections : 10.2.2

real second-order sections : 10.2.2.2

repeated pole : 10.2.2.3

second-order sections : 10.2

series second-order sections : 10.2.1

transposed direct-form II : 10.1.4

implementations

direct-form I : 10.1.1

direct-form II : 10.1.2

transposed direct forms : 10.1.3

internal overflow : 10.1.2

inverse : 19.5

linear : 5.4.2

linear phase : 11 | 11.4

linear time-varying : 21

linear, time invariant : 5

lossless : 16

LTI : 5.5

LTI matrix representation : 19.3

matrix representation : 19

minimum phase : 12

multi-input, multi-output (MIMO) allpass filters : 16.3

nonlinear example : 5.7

notch : 15.1.4

null : 15.1.4

odd impulse response : 11.3

one complex pole : 15.1.5

one pole : 15.1.2

one zero : 15.1.1

order : 6.4 | 6.4 | 9.1

paraconjugate : 16.2

parallel combination : 7.7

paraunitary, MIMO case : 16.3.1

paraunitary, SISO case : 16.2

partial fraction expansion : 7.8

peaking eq : 15.5

phase : 8.3

phase preserving : 11

phase response : 8.3

polar form of freq. response : 8.4

poles : 4.11

poles and zeros : 9

Q (quality factor) : 18.7

real : 6.1

real, digital : 5.2 | 5.2

recursive : 6.1

reflection coefficients : 9.4.1

resonance bandwidth of a pole : 18.6

resonator : 15.1.3

resonator bandwidth in terms of pole radius : 15.1.3.1

resonator center frequency : 15.1.3

series combination : 7.7

shelf : 15.4

shift-invariance : 5.5

signal flow graph (system diagram) : 6.2

simplest lowpass : 2

stability : 6.7 | 9.4

state space realization : 19.6

symmetric impulse response : 11.4

time-domain representations : 6

time-invariance : 5.5

transfer function : 7

transposition : 10.1.3

tunable resonator : 15.6.1

two pole : 15.1.3

two zero : 15.1.4

zero phase : 11.2

zeros : 4.11

filter design

analog prototype : 22.3.3

analog to digital conversion via bilinear transform : 22.3

Butterworth : 8.6.4 | 22.2

Chebyshev : 8.6.4

elliptic : 8.5.2 | 8.6.4

equation error method : 22.4

equation error minimization in the frequency domain : 22.4.4

frequency warping : 22.3.2

lowpass filter : 22.1

maximally flat amplitude response : 22.2

Padè-Prony method : 22.4.6

Prony's method : 22.4.5

Finite Impulse Response (FIR) digital filter : 6.11

finite support : 6.11.3

Finite-Impulse-Response (FIR) digital filter : 6.1

finite-order causal LTI digital filters : 6.4

FIR filter : 6.11 | 6.11

FIR filter design : 11.4.2

FIR part : 7.8.5

flip theorem for z transforms : 11.6

flow graph : 6.2

flow graph reversal : 10.1.3

folding a signal about index zero : 23.9

formant : 10.2.3

formant filtering : 10.2.3

forward-backward filtering : 11.6

frequencies : 14.1.1

frequency domain : 14.1.3

frequency response : 8.1

computation in matlab : 8.5.1

example in matlab : 8.5.2

imaginary : 11.3

frequency warping : 22.3.2 | 22.4.2

frequency-domain equation-error minimization : 22.4.4

frequency-response

measurement : 2.3

plotting in matlab : 23.4

gain at resonance : 15.6

generalized eigenvectors : 20.10

generalized function : 18.4.4

geometric sequence : 7.8.4

graphical computation of amplitude response from transfer-function poles and zeros : 9.2

graphical phase response calculation : 9.3

group delay : 8.6.3

computation : 8.6.6

example : 8.6.4

matlab function 1 : 23.8

matlab function 2 : 23.6

group delay equals modulation delay : 8.6.3.1

guard bits : 10.1.2.1

GUI generation : 24

Haar filter bank : 16.3.2

half-angle tangent identities : 14.2.3.1

half-open interval : 3.2

half-power bandwidth : 9.5 | 18.6

harmonic distortion : 5

Heaviside unit step function : 18.4.3

Henrys : 18.3

Hermitian : 11.2 | 11.3

Hertz (Hz) : 14.1.1

high shelf : 15.4

Hilbert transform relations : 9.10

Hooke's law for ideal springs : 18.2.1

Hurwitz polynomial : 18.8

impedance analysis : 18.4 | 18.5

implicit finite difference schemes : 6.1

impulse : 6.6

impulse invariant transformation : 18.6

impulse response : 6.6 | 6.11.1 | 12.2

example : 6.8

state-space model : 20.1

impulse signal : 4.6 | 6.6 | 6.11.1

impulse, continuous time : 18.4.4

inductor : 17.5.1 | 18.3

inductors as masses : 18.3.1

infinite-impulse-response (IIR) : 6.1

initial conditions : 20.2

initial state : 20.2

initial-condition response : 6.12.5

instantaneous frequency : 14.1.2

instantaneous phase : 14.1.2

intermodulation distortion : 5

interreciprocal : 10.1.3

inverse filter : 19.5

irreducible : 7.8.8

Jordan block : 20.10.1

Jordan canonical form : 20.10.1

Jordan form of a matrix : 20.10.1

ladder filter : 10.2.3

LADSPA plugins : 24.6

Laplace transform

analysis

linear systems : 17.5

mass-spring oscillator : 17.5.2

moving mass : 17.5.1

definition : 17

differentiation theorem : 17.4.2

existence : 17.1

linearity : 17.4.1

relation to z transform : 17.3

response to initial conditions : 17.5.1

theorems : 17.4

least-squares : 19.7

level-dependent gain : 5.7.1

Levinson recursion : 9.4.1

limiter : 5.7

linear algebra : 5.1

linear filter : 5.4.2

linear operator : 5.2

linear phase in audio applications : 12.6

linear prediction : 9.4.1

linear systems theory : 5

linear transformation : 5.2

linear, time-invariant filters : 5

linear-phase filter : 11 | 11.4

design : 11.4.2

examples : 11.4.1

linearity and time invariance : 5

log-swept sine-wave analysis : 2.3.1

logarithmic derivative : 8.6.6

long division : 7.8.5

lossless analog filters : 18.8.1

lossless filter : 15.2 | 16

examples : 16.1

lossless transfer function matrix : 16.3

losslessness implies allpass : 16

low shelf : 15.4

LTI filter matrix : 19.3

LTI filters : 5.5 | 5.10

LTI implications : 6.9

magnitude frequency response : 2.2 | 8.2

marginally stable : 9.4 | 9.4.2

Markov parameters : 20.1

Mason's gain formula : 10.1.3

Mason's gain theorem : 20.5

matched z transformation : 18.6

math summary : 14

matlab : 3

Matlab software : see softwaretextbf

matrices : 19.1 | 20

matrix : 5.2

matrix fraction descriptions : 5.2

matrix representations : 19

maximum-phase filters : 12.3

maximum-phase sequence : 12.3

median smoother : 5.3

memoryless nonlinearity : 5.3

message box (pd) : 24.5.1

MIMO digital filter : 5.2

minimum-delay sequence : 12.4

minimum-delay signals : 12.4

minimum-phase

filter : 12

polynomial : 12.2

sequence : 12.2

minimum-phase = fastest decay : 12.4

minimum-phase allpass decomposition : 12.5

minimum-phase computation from spectral magnitude data : 12.7

minimum-phase conversion of a spectrum : 23.11

minimum-phase filter design : 12.7

minimum-phase filters and signals : 12.7

minimum-phase sequence : 12.7

mixed-phase filter : 12.3

modal representation : 20.9 | 20.9 | 20.9.1

mode of vibration : 20.7.3

Moog VCF : 15.6.5

Moore-Penrose pseudoinverse : 19.7

moving average : 5.6

multi-input, multi-out (MIMO) digital filter : 5.2

multiplicity of a pole : 7.8.7.1

Muse Receptor : 24.7

negative-frequency component : 2.4.4

Newton's second law : 18.3.1

nonlinear distortion : 19.7

nonlinear filter : 5.7

analysis : 5.9

nonparametric signal processing : 12.7

nonrecursive digital filter : 6.1

normalized second-order resonator : 20.4.1

notch : 15.1.6

notch filter : 2.3.2

notch frequency : 15.1.4

null : 2.3.2

numerical issues : 10

observable modes : 20.7.3

observer canonical form : 20.7.1 | 20.7.2

Octave software : see softwaretextbf

odd impulse response : 11.3

one-off subpatch (pd) : 24.5.1

one-pole filter : 15.1.2

one-pole resonator, complex : 15.1.5

one-sided Laplace transform : 17

one-zero filter : 15.1.1

operator : 5.2

operator theory : 5.1

optimality in the Chebyshev sense : 8.6.4

order of a

filter : 6.4 | 9.1

pole : 17.2

polynomial : 9.1

rational function : 9.1

orthogonality principle : 19.7.1

output error minimization : 22.4.1

overdamped : 18.7.2.1

Padé-Prony method for filter design : 22.4.6

para-Hermitian conjugate : 8.6.6

paraconjugate transfer function : 16.2

parallel and series filter sections : 10.2.5

parallel combination : 7.7.2 | 18.5.1

parallel complex resonator : 10.2.2.1

parallel second-order filter sections : 4.12

parallel sos in matlab : 23.7

parametric equalizer : 15.5

paraunitary filter bank : 16.3.1.4

paraunitary MIMO filters : 16.3.1

partial fraction expansion : 4.12 | 7.8 | 10.2.2 | 15.1.5.1

alternate methods : 7.8.6

complex poles : 4.12.1

FIR part : 7.8.5

in matlab : 23.5

inversion : 7.8.4

repeated pole : 7.8.7 | 7.8.7.1

second order sections : 7.8.3

software : 7.8.10

summary : 7.8.9

passband : 2.2 | 8.5.2 | 11.2

pd

abstraction : 24.5.1

externals : 24.5.1

plugins : 24.5

subpatch : 24.5.1

peak filter : 15.5

peak gain : 15.6 | 15.6.3 | 18.7

peak gain versus resonance gain : 15.6.3

peaking eq filters : 15.5

perfect reconstruction filter bank : 16.3.1.4

periodic signal : 14.1.1

phase : 14.1.2

phase delay : 8.6.1

phase dispersion : 8.6.3 | 11.7

phase offset : 14.1.2

phase quadrature : 20.11.2

phase response : 2.3.1 | 8.3

phase unwrapping : 8.6.2

phasor : 2.4.3 | 14.3.3

phasor analysis : 2.4.3 | 14.3.2 | 14.3.3

phasor representation : 2.4.3

piecewise constant-phase filters : 11.2.2

plot

frequency data : 23.2

saving to disk : 23.3

plugin wrapper (pd) : 24.5.2

plugins : 24

LADSPA : 24.6

pd : 24.5

VST : 24.7

polar form of freq. response : 8.4

pole : 17.2

bandwidth : 9.5

frequency : 15.1.3

order : 17.2

time-constant : 9.6

pole-zero analysis : 4.11

poles : 4.11 | 7.6 | 7.8 | 9

poles and zeros : 9

poles of a state-space model : 20.6

poles outside unit circle : 9.7

polynomial

division in matlab : 7.8.10.3

long division : 7.8.10.3

multiplication : 7.8.10.2

multiplication in matlab : 7.8.10.2

order : 9.1

polynomial amplitude envelopes : 7.8.7.3

positive-frequency sinusoid : 2.4.4

predelay : 7.8.5

problems : see exercisestextbf

projection error : 19.7.1

Prony's method : 22.4.5

pseudoinverse : 19.7

Q (quality factor) : 18.7

relation to decay time : 18.7.4

Q of a complex resonator : 18.7.1

Q of a real resonator : 18.7.2

radians per second : 14.1.1

ratio test : 17.2

rational function : 9.1

RC time constant : 18.4.2

real filter : 2.3.2 | 5.2 | 6.1 | 7.8.3

real signal : 5.1

real, even-impulse-response filter : 11.2

real-frequency-response filter : 11.2

Receptor : 24.7

recursive filter : 5.3 | 6.1

reflecting zeros inside unit circle : 9.9

reflection coefficients : 9.4.1

region of convergence : 9.7

repeated pole : 20.10

impulse response : 7.8.7.3

residue : 7.8 | 7.8.7.1

resonance : 15.1.3

resonance frequency : 18.7 | 18.7.1

resonator : 15.1.3

resonator bandwidth : 18.6

response to initial conditions : 6.12.4

right-half plane : 17

ring time : 6.12.1

ripple : 8.5.2

rms level : 5.7.1

roll-off : 8.5.2

running weighted sum : 5.6

samples : 14.1.1

sampling interval : 14.1.1

scalars : 5.1

scaling property of linear systems : 5.4.2

Schur recursion : 9.4.1

Schur-Cohn stability test : 9.4.1 | 9.4.1

seconds : 14.1.1

series and parallel filter sections : 10.2.5

series and parallel transfer functions : 7.7

series connection : 7.7.1

series second-order sections : 10.2.1

set notation : 5.1

shelf filters : 15.4

shift operator : 5.5

shift theorem for z transforms : 7.3

shift theorem for z transforms : 7.3.1

shift-invariant filter : 5.5

sideband images : 5

sifting property : 18.4.4

signal

complex, discrete-time : 5.1 | 5.1

definition : 5.1

flow graph : 6.2

operator : 5.4.2

plotting in matlab : 23.1 | 23.13

real, discrete-time : 5.1 | 5.1

representation : 14.1

signal flow graph : 4.2

signal space : 5.1

similarity transformation : 20.8 | 20.9.1

simple lowpass filter

analysis in matlab : 3 | 3.2 | 3.3 | 3.4

matlab implementation : 3.1

simulation diagram : 2.2.1 | 6.2

sinc function : 11.2.2

sine-wave analysis : 2.3.1 | 2.3.2 | 2.4.5

single-input, single-output (SISO) digital filters : 5.2

singular matrix : 20.10

sinusoid : 14.1.2

SISO digital filter : 5.2

sliding linear combination : 5.6

software : 23

Faust programming : 24

Matlab

frequency-response plot : 23.4 | 23.13

signal plots : 23.1

Matlab or Octave

clipping dB magnitude : 23.10

folding a signal about index zero : 23.9

frequency plots : 23.2

frequency-response computation : 8.5.1

group delay computation : 23.8

minimum phase conversion : 23.11

parallel second-order sections : 23.7

partial fraction expansion : 23.5 | 23.6

saving plots : 23.3

Octave

signal plots : 23.1

spectrum : 14.1.3

speech modeling : 9.4.1

speech synthesis : 10.2.3

split-radix FFT : 3.4

spring

compliance : 18.2.1

constant : 18.2.1

stiffness : 18.2.1

stability of a digital filter : 6.7 | 7.8.8 | 9.4

state space filter : 20

analysis : 20

analysis example

the digital waveguide oscillator : 20.11

complete response : 20.3

computation : 20.7.6

diagonalization : 20.9.1

example : 19.6.1

from difference equations : 20.7

impulse response : 20.1

matlab : 20.7.8

modal representation : 20.9

poles : 20.6

realization : 19.6 | 20

response from initial conditions : 20.2

similarity transformation : 20.8

transfer function : 20.4

transfer function example : 20.4.1

transposition : 20.5

state space realization : 7.8.6

steady state

analysis : 18.2

response : 6.12

signal : 6.12.3

Steinberg Media Technologies : 24.7

step-down procedure : 9.4.1

stopband : 2.2 | 8.5.2

strict right-half plane : 17

strictly proper transfer function : 7.8.3

subpatch (pd) : 24.5.1

sum of sinusoids : 14.3

superposition property : 5.4.2

superposition property of linear systems : 5.4.2 | 6.10

swanalmainplot : 23.13

swanalplot : 23.12

symmetric impulse response : 11.1 | 11.1

symmetric linear-phase FIR filter : 11.4

synthesis filter bank : 16.3.1.4

system diagram : 4.2 | 6.2

system function : 7

system identification : 19.7 | 19.7

tapped delay line : 6.11

Taylor series expansion : 2.4

Tellegen's theorem : 10.1.3

time constant : 9.6

time constant of a pole : 9.6

time domain : 14.1.3

time reversal inverts the locations of all zeros : 12.3

time-delay spectrometry : 2.3.1

time-invariant filter : 5.5 | 5.5

time-varying

filter coefficients : 15.6

filter example : 5.8

two-pole digital filters : 15.6

Toeplitz linear operator : 19.3

Toeplitz matrix : 19.3

transfer characteristics : 7

transfer function : 6.11.5 | 7

factored : 7.6

matrix : 20.4

of a state space filter : 20.4

to second-order-section matlab function tf2sos : 10.2.1

transient : 6.12.3 | 6.12.3

transient response : 6.12

transition band : 8.5.2

transition frequency : 15.4

transpose of a filter : 10.1.3 | 20.5

transposed direct form I (TDF-I) : 10.1.3

transposed direct form II (TDF-II) : 10.1.3

transposing the signal flow graph : 20.5

transversal filter : 6.11

tremolo : 5.8

trig identities : 14.2

trigonometric identity summary : 14.2.2

tunable two-pole digital filters : 15.6

Tustin's Method : 22.3.1

two's complement wrap-around : 10.1.1.1

two-pole filter : 15.1.3

two-pole partial fraction expansion : 15.1.5.1

two-pole time-varying filter : 15.6

two-sided Laplace transform : 17

two-zero filter : 15.1.4

undamped : 18.7.2.1

underdamped : 18.7.2.1

unilateral z transform : 7.1

unilateral Laplace transform : 17

unit step function : 6.8 | 15.1.5

unstable poles : 9.7

unwrapping phase : 8.6.2

variable resonator : 15.6.1

variable two-pole digital filters : 15.6

VCF : 15.6

vector coordinate : 5.1

vector space : 5.1

vectorized algorithms : 3.1

virtual analog synthesis : 15.6.5

vocoder : 8.6.5

voltage divider rule : 18.4.2

voltage-controlled filters : 15.6

Volterra kernels : 5.9

vowel simulation : 10.2.3

VST plugins : 24.7

wave digital filters : 22.3.4

weighting function : 22.4.2

window method for FIR filter design : 22.1

z transform : 7.1

existence : 7.2

theorems

convolution : 7.3.2

shift : 7.3.1

zero at infinity : 18.4.5

zero initial state : 20.1

zero padding : 3.4 | 19.3 | 19.4

zero-input response : 19.6.1

zero-phase filter : 11.2

examples : 11.2.1 | 11.2.2

zero-state response : 6.12.5

zeros of a filter : 4.11