English

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The student will determine the spectral properties of 1D and 2D deterministic and stochastic signals using different base functions (Fourier analysis, wavelets) for multiple resolution. They will learn how to use multirate filter banks (for example, in methods of audio- and video-signal compression, ADSL transmission, etc.). They will also become familiar with linear prediction and cepstral analysis. They will program in the Matlab environment and use it for signal processing and system design.

The subject knowledge on the Bachelor´s degree level with emphasis on digital signal processing is required. Furthermore, the basic ability to program in the Matlab environment is necessary.

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Teaching methods depend on the type of course unit as specified in the article 7 of BUT Rules for Studies and Examinations.

Lab exercises are mandatory for successfully passing this course and students have to obtain the required credits. For computer lab tests they can get 30 out of 100 points. The remaining of 70 points can be obtained by successfully passing the final written examination.

1. Description of discrete signals and their division. Energy and power signals. Periodic discrete signals. Basic 1D and 2D signals (unit impulse, unit step, real and complex harmonic signals). The discrete Fourier series (DFS) and the discrete Fourier transform (DFT) spectra. The fast Fourier transform (FFT) algorithm. The Z transform.
2. External and internal (state-space) representations. Bounded Input Bounded Output (BIBO) stability, causality. Linear time-invariant 1D discrete system. Connection of partial sections. FIR and IIR systems. Frequency responses, fast convolution. Overlap-save and overlap-add methods. Linear shift-invariant 2D discrete system. The Fourier transform of 2D discrete signals, 2D frequency response.
3. Matrix representation of system equations and their solution. Semi-symbolic computer analysis. Signal flow graphs and Mason’s gain rule. Check of discrete system causality.
4. Definition of a periodic even sequence using an aperiodic sequence, definition of discrete cosine transform from DCT I to DCTIV. Relationship between DCT II and DFT. Definition of the discrete sine transform. Undersampling (decimation) and oversampling (interpolation) of discrete signal in an integer ratio. Description of the time and frequency domains. The transformation of sampling frequency in a rational number ratio. Optimization of the number of multiplier and memory registers of anti-aliasing low pass filter.
5. Zero-pole plot in the z domain, Minimal, maximal and mixed phases. All-pass filter, inverse discrete system. Sampling of bandpas signals. Real signal, analytical signal and complex lowpass signal. The Hilbert transform for continuous-time signals. Quadrature modulator and demodulator. The Hilbert transformer for discrete signals.
6. Analyzing part and synthesizing part of digital filter bank. Calculation of DFT spectrum of discrete signal using uniform-DFT filter banks. Sub-band coding. Quadrature mirror filters (QMF). Perfect signal reconstruction. Transmultiplexers.
7. The Gabor and the short-time Fourier transforms. Time-frequency resolution, The Heisenberg uncertainty principle. Orthogonal systems and their application to spectral analysis. Wavelets and their definition.
8. The continuous-time wavelet transform (CWT), the discrete wavelet transform (DWT). The discrete-time wavelet transform (DTWT). Relationship between DTWT and QMF digital filter banks.
9. Cumulative distribution function and probability density function, general and central moments. Stationary and ergodic continuous- and discrete-time stochastic processes. Estimates, consistent estimate. Random selection from probability distribution, statistics, statistical hypothesis testing, parametric and non-parametric tests, goodness of fit tests.
10. Forward and backward linear prediction. Calculation of linear prediction coefficients. Lattice structure of autoregressive (AR) and autoregressive moving average (ARMA) types and their application. Using linear predictive analysis for speech signal compression.
11. Definition of power spectral density and its properties. Definition of periodogram and its calculation. The Bartlett method of averaging periodograms. The Welch method of averaging modifmodified periodograms. The Blackman and Tukey method of smoothing the periodogram. Performance characteristics of nonparametric power spectral density estimators.
12. AR, MA or ARMA type stochastic processes. Model definition for power spectral density calculation. Relationship between autocorrelation coefficients and model parameters. The Yule-Walker and the Burg methods for AR type model. Selection of the order of type AR model.
13. Complex and real cepstra. Generalized superposition. Homomorphic filtering, definition and its application. Approximation of exponential function by continued fraction expansion.

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The aim of the course is to present modern methods of digital signal processing that are based primarily on analyses of 1D and 2D discrete-time and digital signals and systems. Furthermore, the students will learn about parametric and non-parametric spectral analysis of stochastic signals and about mathematical statistics. They will use multirate digital filter banks for signal processing in practise.

The content and forms of instruction in the evaluated course are specified by a regulation issued by the lecturer responsible for the course and updated for every academic year.

Not applicable.

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FLIEGE,N.J.: Multirate Digital Signal Processing. John Wiley, Chichester 1994. ISBN 0 471 93976 5
MADISETTI, V.K., WILLIAMS, D.B.: The Digital Signal Processing Handbook. CRC Press, 1998. ISBN 0-8493-8572-5
MITRA, S.K.: Digital Signal Processing. A Computer-Based Approach. The McGraw-Hill Companies, Inc. New York 1998. ISBN 0-07-042953-7
SHENOI, K.: Digital Signal Processing in Telecommunications. Prentice Hall, New Jersey 1995. ISBN 0-13-096751-3
VÍCH, R., SMÉKAL, Z.: Digital Filters (Číslicové filtry). Academia, Praha 2000. ISBN 80-200-0761-X (In Czech)

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