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Index


FIR filter design
optimal least-squares impulse response : 15.2
absolutely integrable : 3.2.1
acyclic convolution : 3.3.5
acyclic FFT convolution : 9.1.2
additive synthesis : 2 | 8 | 8.1.4 | 11.4.6
alias component matrix : 12.3.8
aliased sinc function : 2.5
aliasing components : 3.3.12
aliasing theorem for the DTFT : 3.3.12
aliasing, time domain : 9.1.4.3
allpass filter : 12.5.2
amplitude envelope : 8
analysis modulation matrix : 12.3.8
analytic signal : 2.1 | 15.5
asinc function : 2.5
associate peaks : 8.3.2
audio spectrogram : 7.3
audio spectrogram hop size : 7.3.2
auditory filter : 7.3.3.3
auditory filter bank : 7.3.3.2
auditory filter banks : 7.3.1
autocorrelation : 3.3.7
autocorrelation computation : 6.9
autocorrelation function : 17.2.3
average : 17.1.8
bandlimited signals cannot be time limited : 3.4.16
bandpass filter : 15.5
Bark frequency scale : 18.5
Bark warping : 18.7
Bartlett window : 4.4
baseband signal : 10.1.2
bias : 5.7
bias of parabolic interpolation : 5.7
biased autocorrelation : 6
biased sample autocorrelation : 6.6
bibliography : 20
bilinear transform : 18.6
bilinear transform frequency warping : 18.2
bin number : 7.1.3
Blackman window : 4.3.3
Blackman window matlab example : 19.1.1
Blackman-Harris window : 4.3.6
Blackman-Harris window family : 4.3
Blackman-Harris window, frequency-domain implementation : 4.3.7
bounded variation : 3.5
brown noise : 6.14
cepstral smoothing : 11.2.1
cepstrum : 15.7
characteristic function : 16.13.4
Chebyshev polynomials : 4.9.4.1
chirp signal : 10.2.1
chirp, Gaussian-windowed : 16.5
chirplet : 16.5
chirplet modeling : 8.6.2
chirplets : 4.10 | 16
circular convolution : 9.1
coherent addition of signals : 6.15
COLA (constant overlap-add) : 7.1.1
COLA constraint : 9.2.1
COLA constraint, frequency domain : 9.3.2
COLA dual : 9.3
colored noise : 6.14
complex demodulation : 10.3.2
complex Gaussian integral : 16.3
compression : 12
Conjugate Quadrature Filters : 12.3.7
constant overlap-add (COLA) : 11.4.1
constant overlap-add (COLA) property : 7.1.1
constant overlap-add property : 9
constant-overlap-add : 9.2.1
continuous probability distribution : 17.1.3
convolution : 3.3.5 | 9.1
acyclic : 9.1.2
acyclic in matlab : 9.1.2.1
cyclic : 9.1.1
cyclic, or circular : 9.1
FFT overlap-add in matlab : 9.2.5
FFT, overlap-add : 9.2
fractional : 15.10.7
in Matlab or Octave : 9.1.3
short signals : 9.1
convolution theorem : 3.3.5 | 3.3.5 | 3.4.6
convolution, continuous time : 3.4.6
correlation : 3.3.6
correlation analysis : 17.2
correlation theorem : 3.3.6 | 3.3.7
covariance : 6.4
critical band of hearing : 7.3.2
cross synthesis : 11.1
cross-correlation : 17.2.1
cross-power spectral density : 17.2.2 | 17.2.2
cubic polynomial phase interpolation : 8.6.1
cut-off frequency : 15.1
cycles per second : 3.4.1
cyclic autocorrelation : 6.8
cyclic convolution : 9.1
cyclic FFT convolution : 9.1.1
dc sampling filter : 10.3.1
decimation operator : 12.1.2
deconvolution : 9.1.2
delta function : 3.4.9
demos : 11.7
denoising : 6.1.1
deterministic : 5.8.2
deterministic part : 8.4.1
detrend : 6.9
DFT filter bank : 10.3 | 10.3.4.2
differentiation theorem : 3.4.2 | 3.5
differentiation theorem dual, DTFT : 3.3.13
differentiation theorem dual, FT : 3.4.3
digital filter design, FIR : 15
digital prolate spheroidal sequence (DPSS) : 4.7
Dirichlet function : 2.5
discrete probability distribution : 16.10
Discrete Prolate Spheroidal Sequences (DPSS) : 4.7
discrete time Fourier transform (DTFT) : 3.1
Dolph window : 4.9
Dolph-Chebyshev and Hamming windows compared : 4.9.3
Dolph-Chebyshev window : 4.9 | 4.9
Dolph-Chebyshev window length computation : 4.9.4.4
Dolph-Chebyshev window, theory : 4.9.4
downsampling : 3.3.12
downsampling (decimation) operator : 12.1.2
DPSS window : 4.7
DTFT
aliasing theorem : 3.3.12
convolution theorem : 3.3.5
correlation theorem : 3.3.6
downsampling theorem : 3.3.12
energy theorem : 3.3.8
even symmetry : 3.3.3.1
linearity : 3.3.1
power theorem : 3.3.8
repeat operator : 3.3.10
repeat theorem : 3.3.11
scaling operator : 3.3.10
scaling theorem : 3.3.11
shift theorem : 3.3.4
stretch operator : 3.3.9
stretch theorem : 3.3.11
symmetry : 3.3.3
time reversal : 3.3.2
DTFT Fourier theorems : 3.3
effective length of a window : 2.7.1
energy theorem : 3.3.8
ensemble average : 17.1.6
entropy : 16.11.1 | 16.11.1
envelope break-points : 8.6.1
envelope follower : 7.3.3.6
equivalent rectangular bandwidth : 18.8
excitation pattern : 7.3.1 | 7.3.2 | 7.3.3.2
expected value : 17.1.6 | 17.1.6 | 17.3
exponential window : 4.5
extended lapped transforms : 12.7.2
FBS modifications : 10.8.2.1
FFT convolution speed : 9.1.4
FFT input buffer : 11.4.2
fftshift utility in matlab : 5.4.1
filter
overlap-add FFT convolution : 9.2
filter bank summation interpretation of the STFT : 10
filter bank, perfect reconstruction : 12.3
filter banks : 12
paraunitary : 12.5
filter design : 18
example of window method : 15.4.2
Hilbert transform filter : 15.5
least-squares, linear-phase FIR : 15.11.6
filter design, FIR
frequency-sampling method : 15.3
window method : 15.4
filter-bank interpretation of the STFT : 10.1.2
Filter-Bank Summation (FBS) : 10.3.4
filtered white noise : 6.14 | 6.14
filters
audio, FIR : 9.1.4.1
lossless : 12.5.2
lossless examples : 12.5.3
finite support : 6.6
finite-impulse-response : 15.4
FIR digital filter design
frequency-sampling method : 15.3
window method : 15.4
FIR filter design
by linear programming : 15.12
least-squares, linear phase : 15.11.6
optimal methods : 15.11
FIR fractional delay filter : 15.10
first-order moment : 16.13.1
flip operator : 3.4.7
formants : 7.2.1
Fourier dual : 5 | 10.5
Fourier theorems
continuous time : 3.4
discrete time : 3.3
DTFT
differentiation dual : 3.3.13
FT
differentiation dual : 3.4.3
Fourier theorems (continuous time)
convolution theorem : 3.4.6
differentiation : 3.4.2
flip theorem : 3.4.7
gaussian pulse : 3.4.10
impulse train : 3.4.13
power theorem : 3.4.8
rectangular pulse : 3.4.11
sampling theorem : 3.4.15
scaling or similarity : 3.4.4
shift theorem : 3.4.5
uncertainty principle : 3.4.16
Fourier transform : 3.2
Fourier transform existence : 3.2.1
Fourier transforms for continuous/discrete time/frequency : 3
fractional delay filter : 15.10
fractionally iterated convolution : 15.10 | 15.10.7
fractionally iterated multiplication : 15.10.1
frame : 7.1.3
frequency resolution : 2.5.2 | 2.7
frequency sampling for FIR filter design : 15.3
frequency shifting : 11.5
frequency trajectories : 8.3.2
frequency warping
allpass : 18
bilinear transform : 18.2
non-parametric : 19.3.5
frequency-shifting : 8.1.8
Gaussian chirp : 4.10 | 16
Gaussian distributed : 5.8.2
Gaussian distribution
maximum entropy property : 16.11
Gaussian function : 3.4.16.1 | 16
Gaussian integral : 16.2.1
gaussian pulse : 3.4.10
Gaussian random variable, closed under addition : 16.14
Gaussian window : 16.1
Gaussian window function : 4.10
Gaussian, Fourier transform of : 16.4
Gaussian-windowed chirp : 16.5
generalized function : 3.4.9
generalized Hamming window family : 4.2 | 4.2.6
Gibbs phenomenon : 2.5.1
glossary of notation : 14
graphic equalizer : 15.6
graphical convolution : 9.1
graphical equalizers : 9.3.3
Group-Additive Synthesis : 8.6.3.2
Haar filter bank : 12.3.3
Hamming and Dolph-Chebyshev windows compared : 4.9.3
Hamming window : 4.2.4
Hann window : 4.2.1 | 4.2.1
Hann-Poisson window : 4.6
hanning window : 4.2.1
harmonic : 2.7.1
Heisenberg uncertainty principle : 3.4.16.1
Hermitian : 3.3.3
Hermitian spectrum : 15.5
Hilbert transform : 15.9
Hilbert transform filter design : 15.5
hop size : 6.12 | 7.1.3 | 9.2.1
ideal lowpass filter : 15.4
impulse train : 3.4.13
impulse, continuous time : 3.4.9
impulse, sinc : 3.4.12
independent events : 17.1.2 | 17.3.1
independent random variables : 17.3.1
inner product : 3.3.8 | 3.4.8
instantaneous loudness : 7.3.2 | 7.3.3.6
interpolation kernel : 7.3.3.3
interpolation kernel, spectral, ideal : 19.3.5.1
interpolation of a DFT : 5.2
inverse FFT synthesis : 8.6.2
iterated convolution : 15.10.7
Kaiser window : 4.8
Kaiser window beta parameter : 4.8.3
Kaiser-Bessel window : 4.8
lagged product : 6.4
Laurent expansion : 15.8 | 15.8
least squares estimation : 5.8.1
least squares sinusoidal parameter estimation : 5.8.1
likelihood function : 5.8.3
linear least squares : 5.8.1.1
linear phase : 9.1.4.2
linear phase term : 3.3.4
linear prediction spectral envelope : 11.2.2
linearity of the DTFT : 3.3.1
long-term loudness : 7.3.3.6
lossless filter : 12.5.2
lossless filter examples : 12.5.3
lossless filters : 12.5.2
lossless transfer function matrix : 12.5.2
loudness : 7.3 | 7.3.1
loudness spectrogram : 7.3.2 | 7.3.2
loudness spectrogram, examples : 7.3.3
loudness versus time : 7.3.3.6
loudness versus time and frequency : 7.3.2
low-pass filtering by FFT : 9.1.4.2
lowpass filter, ideal : 15.1
magnitude-only analysis/synthesis : 11.4.7
main-lobe width : 2.6
masking : 11.6
matlab
discrete prolate spheroidal window : 19.1.2
DPSS window : 4.7.1
minimum zero-padding factor : 19.2.4
peak finder : 19.2
phase unwrapping : 19.3.4
spectrogram : 19.3
spectrum analysis windows : 19.1
window method for FIR filter design : 15.4.1
matlab examples : 19
matlab listing
dpssw : 19.1.2
findpeaks : 19.2.1
maxr : 19.2.2
npwarp : 19.3.5
oboeanal : 19.2.5
qint : 19.2.3
spectrogram : 19.3.1
testspectrogram : 19.3.2 | 19.3.3
unwrap : 19.3.4
zero-phase blackman : 19.1.1
zpfmin : 19.2.4
maximum likelihood estimator : 5.8.2
maximum likelihood sinusoidal parameter estimation : 5.8.2
mean of a distribution : 16.13.1
mean of a random process : 17.1.7
minimum phase : 15.7
minimum phase filters : 15.7
minimum phase means a causal cepstrum : 15.8
modulated lapped transform : 4.2.13
MPEG filter banks : 12.7
multi-resolution STFT : 7.3.2
multirate filter banks : 12
multirate noble identities : 12.2.5
multiresolution sinusoidal modeling : 8.1.7
multiresolution STFT : 7.3.3.1 | 7.3.3.1
munchkinization : 11.5
noble identities : 12.2.5
noise : 6.1.2
mean : 17.1.7
synthesis example : 6.14.2
white : 17.3
noise process : 17.1.4
noise spectral analysis
periodogram : 6.11
Welch's method : 6.12
noise spectrum analysis : 6
pink noise example : 6.14.3
noise, filtered : 6.14
non-coherent addition of signals : 6.15
non-parametric : 15.10
nonlinear spectral phase modification : 15.10
nonuniform resampling : 7.3.3.3
normal distribution : 5.8.2
normalized frequency : 3.1
normalized radian frequency : 2.2
notation glossary : 14
oboe spectrum analysis : 5.5
oddly-stacked Princen-Bradley filter bank : 12.7.2
OLA modifications : 10.8.2.1
optimized windows : 4.11
orthogonal two-channel filter banks : 12.3.8
orthogonality principle : 5.8.1.2
overlap-add convolution in matlab : 9.2.5
overlap-add decomposition : 9.2.1
overlap-add FFT convolution : 9.2
overlap-add FFT processor : 9
overlap-add interpretation of the STFT : 9 | 10.1.1
overlap-add method : 7.1.4
overlap-add, with modifications : 9.5
overtone : 8
panning : 6.16
paraconjugate : 12.3.8
paraconjugation : 12.5.1
parametric filter design : 15.10
paraunitary filter bank : 12.5.5
paraunitary filter banks : 12.5
Parseval's theorem : 3.3.8
PARSHL : 11.4
partial overtone : 8
partition of unity property : 9.2.1
PDF : 17.1.3
peak detection : 11.4.3
peak matching : 11.4.4
peak-finding : 5.8
perfect reconstruction : 10.1.3
perfect reconstruction filter bank, conditions for : 12.4.5
perfect reconstruction filter banks : 12.4
perfect reconstruction filter banks, critically sampled : 12.3
periodic sinc function : 2.5
periodogram : 6.11
periodogram method : 6.12 | 6.12
periodogram method for power spectrum estimation : 6.12
phase unwrapping : 15.10 | 19.3.4
phase vocoder : 8.1.3
phons : 7.3.3.6
pink noise : 6.14 | 6.14.2
pitch detection : 11.6
Poisson summation formula : 9.3.1
Poisson summation formula, continuous time : 3.4.14
Poisson window : 4.5
polyphase component filters : 12.2.1
polyphase components : 12.2
polyphase decomposition : 12.1.3 | 12.2.1 | 12.2.2
polyphase filter bank : 12.1.3
polyphase matrix : 12.4
polyphase signals : 12.1.3
Portnoff window : 10.7
power spectral density : 17.2.5
smoothed : 6.7
power spectrum : 17.2.5
power theorem : 3.3.8 | 3.4.8
pre-emphasis : 11.4.8
preemphasis : 11.6
preprocessing : 11.6
probability density function : 17.1.3
probability distribution : 16.10 | 17.1.1 | 17.1.1
processing gain : 6.15
prolate spheroidal wave function : 4.7
prolate spheroidal window : 4.7
Pseudo-QMF filter bank : 12.7.1
quadratic interpolation : 5.6
quadratically interpolated FFT (QIFFT) method : 5
quadrature mirror filterbanks (QMF) : 12.3.5
radians per second : 3.4.1
raised-cosine window : 4.2.1
random process : 17.1.4
random variable : 17.1.3
random variables : 17.1
Rayleigh's energy theorem : 3.3.8
rectangular pulse : 3.4.11
rectangular window : 2.3 | 2.5 | 4.1
rectangular window side-lobes : 2.5.1
references by topic : 20
Remez multiple exchange algorithm : 15.4.2.4
repeat operator : 3.3.10
repeat theorem : 3.3.11
residual signal : 8.4.1
resolution bandwidth : 2.6
resolution of frequencies : 2.7
resolution window length : 2.7
resolving sinusoids : 2.6
Riemann Lemma : 3.5
roll-off rate : 3.5
running-sum lowpass filter : 10.3.1
sample autocorrelation : 6 | 6.4
sample autocorrelation function : 6.9
sample mean : 17.1.8
sample mean of a random process : 17.1.8
sample power spectral density : 6.5
sample PSD : 6
sample variance : 6.4 | 17.1.10 | 17.1.10
sampled rectangular pulse : 3.4.13
sampling synthesis : 8.6.3.1
sampling theory : 3.4.15
scaling theorem : 3.4.4
second central moment : 16.13.2 | 17.1.9
second moments of a signal : 3.4.16.1
shah symbol : 3.4.13
shift operator : 3.3.4
shift theorem : 3.3.4 | 3.3.4 | 3.4.5
short time Fourier transform : 7
downsampling : 10.8
modifications : 10.9
short-term loudness : 7.3.3.6
short-time Fourier transform (STFT) : 7.1
side-lobe width : 2.6
sifting property : 2.1 | 3.4.9
signal model : 5.8.1
similarity theorem : 3.4.4
sinc function : 2.5 | 15.4
sinc function, aliased (periodic) : 2.5
sine window : 4.2.13 | 4.2.13
sines + noise + transients model : 11.4.13
sines + noise spectral modeling : 8.4
sines+noise+transients : 8
sinusoidal amplitude estimation : 5.8.1.1
sinusoidal modeling : 8 | 8
sinusoidal parameter estimation
general case : 5.8.1.3
known frequency : 5.8.1.2
known frequency and phase : 5.8.1.1
least squares : 5.8.1
sinusoidal spectrum analysis : 2
Slepian window : 4.7 | 4.7
sliding DFT : 10.3.4.2
sones : 7.3.3.6
specific loudness : 7.3.1 | 7.3.2 | 7.3.3.4
spectral display : 7.1
spectral envelope : 11.2
cepstral smoothing : 11.2.1
linear prediction : 11.2.2
spectral interpolation : 5
spectral interpolation, ideal : 5.1 | 19.3.5.1
spectral modeling : 8
history : 8.1
spectral modeling applications : 11.4.9
spectral modeling synthesis : 8
spectral modifications : 9 | 9.2
spectral transformations : 11.4.5
spectrogram : 7.2 | 19.3.1
spectrogram parameters : 7.2
spectrogram, for audio display : 7.3
spectrum : 2.1
spectrum analysis : 4
noise : 6
oboe data : 5.5
sinusoids or spectral peaks : 2
statistical formulation : 17
time varying : 7
speech spectrogram : 7.2.1
square integrable : 3.2.1
stationary : 6.1.1 | 17.1.6
stationary stochastic process : 17.1.5
statistical signal processing : 17
step size : 7.1.3
stereo panning : 6.16
STFT : see short-time Fourier transformtextbf
filter-bank interpretation : 10.1.2
overlap-add interpretation : 10.1.1
weighted overlap-add : 9.7
STFT filter bank, downsampled : 10.8.1
stochastic part : 8.4.1
stochastic process : 6 | 17.1.4
stop-band attenuation : 15.4.2.3
stretch operator : 3.3.9 | 3.3.9 | 12.1.1
stretch theorem : 3.3.11
strong COLA constraint : 9.3.2.1 | 9.3.2.1
subtractive synthesis : 8.4
symmetric Toeplitz operator : 4.7
symmetry of the DTFT for real signals : 3.3.3
third-octave filter bank : 7.3.1
time aliasing : 15.10.6
time compression/expansion : 11.5
time domain aliasing : 9.1.2.2
time limited : 15.4
time reversal and the DTFT : 3.3.2
time scale modification : 8.5.3 | 11.5
time-bandwidth product : 3.4.16.3
time-domain aliasing : 9.1.4.3
time-frequency displays : 7
time-frequency distributions : 7.1
time-frequency reassignment : 8.1.7.2
time-limited interpolation : 5.2
time-limited signals : 3.4.16.2
time-scale modification : 8.1.8
time-varying OLA modifications : 9.5
total variation : 3.5
transform coders : 7.1.4
transient detector : 8.5.2
transition band : 15.10.6
transpose, filter bank : 12.3.4 | 12.4.7
triangular window : 4.4
twiddle factor : 12.1.2
two-sided Taylor expansion : 15.8
type II polyphase decomposition : 12.2.3
unbiased estimator : 17.1.8 | 17.1.10
uncertainty principle : 3.4.16
unimodular polynomial matrix : 12.5.5
unwrapped phase : 15.10
unwrapping phase : 19.3.4
upsampling (stretch) operator : 12.1.1
variance : 17.1.9 | 17.1.9
variance of a distribution : 16.13.2
vocoder : 11.3
wah-wah pedal : 10.12
wavelet filter banks : 13
wavelets : 13.2
weak COLA constraint : 9.3.2
weighted overlap add : 9.7
weighted overlap-add : 9.7
Welch autocorrelation : 6.12.1 | 6.12.2
Welch's method for spectrum analysis : 6.12
Welch's method, windowed : 6.13
white noise : 6.1.1 | 6.1.2 | 6.3 | 6.3.1 | 6.4 | 6.4 | 6.5 | 6.5 | 6.7 | 6.10 | 6.11 | 6.11.1 | 6.14 | 6.14 | 6.14 | 6.14.2 | 17.3
window function : 4
window method, FIR filter design : 15.4 | 15.6
windowing effect : 2.4
windows
Bartlett : 4.4
Blackman : 4.3.3 | 19.1.1
Chebyshev : 4.9
Dolph-Chebyshev : 4.9
Dolph-Chebyshev theory : 4.9.4
DPSS : 4.7
exponential : 4.5
frequency resolution : 4.8.5
generalized Hamming : 4.2 | 4.2.6
Hann-Poisson : 4.6
Kaiser : 4.8
Kaiser-Bessel : 4.8
no side-lobes case : 4.6
optimized : 4.11
Poisson : 4.5
Prolate Spheroidal : 4.7
rectangular : 2.5 | 4.1
sine : 4.2.13
Slepian : 4.7
triangular : 4.4
windows for spectrum analysis : 4
zero padding : 5.3
zero padding, minimum : 19.2.4
zero padding, zero-phase form : 5.4
zero-centered : 2.3
zero-padding factor : 7.1.3
zero-phase windows : 2.5


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``Spectral Audio Signal Processing'', by Julius O. Smith III, (March 2007 Draft).
Copyright © 2008-05-20 by Julius O. Smith III
Center for Computer Research in Music and Acoustics (CCRMA),   Stanford University
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