The course objective is to gain the knowledge of students about digital signal processing with the focus on audio signals.

A student who successfully completes this course will be able: - to analyze signal in time-frequency domain; - to do audio forensic work; - to understand, develop and implement algorithms for speech and music coding; - to understand, develop and implement algorithms for speech enhancement; - to understand, develop and implement algorithms for digital audio effects such as: equalization, chorus, delay, echo, flanging, reverberation, etc.; - to use specialized software tools (Praat, Matlab, Sox) for above-mentioned tasks.

- Audio signal analysis in time-frequency domain. - Sound perception. - Audio coding techniques based on perceptive model (MP3, AAC). - Speech production. - Speech coding techniques based on source model (LPC, CELP, GSM) - Speech enhancement algorithms. - Digital audio effects (equalization, chorus, delay, echo, flanging, reverberation). - Music information retrieval.

Lectures, practice, consultation. Lectures are dynamic and interactive. Presentations with specially created audio and video clips and animations demonstrate and illustrate key details in the lectures. In practice students implement parts of the algorithms that theoretical basis is presented in lectures. The students have a project assignment, whose defense is one of the exam prerequisites. Independent student work is supported through the web portal of the Chair of Telecommunications and Signal Processing.