AUDIO AND VIDEO SIGNAL PROCESSING

DESIGN OF DIGITAL FILTERS
 Relationship between difference equations, block diagrams and flow graphs. Structures for Finite Impulse Response (FIR) and Infinite Impulse Response (IIR) filters: direct forms, cascade and parallel forms; transposed forms; structures for linear phase FIR filters.
 Design methods for digital filters: specifications, choice of the response type, coefficients calculation and realization structure.
Synthesis of IIR filters from analog filters: review of Butterworth and Chebychev analog filters; pole-zero placing; the impulse invariance method; the matched Z transform; the bilinear transformation method.
 Synthesis of linear phase FIR filters: the windowing method; the Kaiser window method; the optimal (equiripple) method: the alternation theorem and the Parks-McClellan algorithm; the frequency sampling method.
 SPECTRAL ANALYSIS AND ESTIMATION
 Spectral estimation of stationary signals: the periodogram. Frequency resolution and Leakage. Spectral estimation of nonstationary signals: the time-varying DFt and the spectrogram. Applications to the analysis of speech signals.
THE DISCRETE COSINE TANSFORM (DCT)
Definitions and inverse Transforms: the DCT-1 and DCT-2. Energy compaction property. Applications to image compression.
SPEECH SIGNAL MODELS
Review of acoustic phonetics and the acoustic theory of speech production; digital models for speech signals. Analysis and synthesis of speech signals. Pitch freeuqncy and formant frequencies. The VOCODER model and its variations.
LINEAR PREDICTIVE CODING (LPC)
AR and MA filters; predictive filter with impulsive or noise input. Calculation of the Error signal power and its minimization. Yule-Walker equations. Interpretation in the frequency domain: predictive filter as a spectral estimator. Applications to speech signals.