Source: Wikipedia. Pages: 205. Chapters: Fourier analysis, Analog-to-digital converter, Nyquist-Shannon sampling theorem, Discrete cosine transform, Codec, Discrete Fourier transform, Fast Fourier transform, Audio timescale-pitch modification, Dirac delta function, Sampling rate, Linear predictive coding, SIMD, Digital-to-analog converter, Bilinear transform, Low-pass filter, Whittaker-Shannon interpolation formula, Delta modulation, Digital filter, Source separation, Aliasing, Nyquist rate, System analysis, Digital signal processor, Adaptive predictive coding, Nyquist frequency, Adaptive filter, Filter design, Polyphase quadrature filter, Quadrature mirror filter, Multi-core processor, Delta-sigma modulation, LTI system theory, Dither, JESD204, Window function, Least-squares spectral analysis, Discrete-time Fourier transform, Parks-McClellan filter design algorithm, Almost periodic function, Minimum phase, Relations between Fourier transforms and Fourier series, Talk box, PLL multibit, Discrete wavelet transform, Voice activity detection, Least mean squares filter, Recursive least squares filter, Finite impulse response, Numerically-controlled oscillator, Sample rate conversion, SigSpec, Reconstruction filter, First-order hold, Bilinear time-frequency distribution, All-pass filter, Arkamys, Dbx Model 700 Digital Audio Processor, Time to digital converter, Goertzel algorithm, Polyphase matrix, DFT matrix, Super-resolution, Direct digital synthesizer, Anti-aliasing filter, Cascaded integrator-comb filter, Multiply-accumulate, A derivation of the discrete Fourier transform, BIBO stability, Pitch detection algorithm, Logarithmic number system, Infinite impulse response, Instantaneous phase, Successive approximation ADC, Noise shaping, Zero-order hold, Sinc filter, Impulse invariance, Pitch shift, Automatic control, Adaptive-additive algorithm, Bandlimiting, Oversampling, Shapiro polynomials, Pitch correction, Tricore, Nyquist ISI criterion, Ramer-Douglas-Peucker algorithm, Filter bank, Signal averaging, Sample and hold, Multidelay block frequency domain adaptive filter, Downsampling, IBM Mwave, Multiple signal classification, Cheung-Marks theorem, Similarities between Wiener and LMS, Multi-rate digital signal processing, ENOB, Causal system, Fast Fourier Transform Telescope, Upsampling, Spectrum continuation analysis, Discrete signal, Normalized frequency, Line spectral pairs, Media processor, Spectral slope, Advanced process control, Banded waveguide synthesis, Welch's method, Gerchberg-Saxton algorithm, Bin-centres, Digital signal controller, Beta encoder, Digital down converter, Frequency estimation, Spectral flatness, Matched Z-transform method, Full scale, Spectral centroid, DSSP, HADES, James A. Moorer, SoundDroid, Digital frequency, Digital delay line, Audio normalization, Discrete transform, FDOA, Pisarenko harmonic decomposition, Warped linear predictive coding, Resampling, Machine listening, Spurious-free dynamic range, Anticausal system, Fast Walsh-Hadamard transform, Geometric-Arithmetic Parallel Processor, Linear phase, Spectral flux, Computational auditory scene analysis, Adaptive equalizer, Differential nonlinearity, High Frequency Content measure, Bartlett's method, EXpressDSP, Decimation, Tristimulus, Discrete frequency domain, NTC Module, Audio Signal Processor, Integral nonlinearity, Ideal sampler, NeuroMatrix, Delay equalization, Adjoint filter, Waveform buffer, Super Bit Mapping, Native processing, Sogitec 4X, Unity am.