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Plenary Lecture

On Using Channel Impulse Response in Speech and Audio Processing - A Review

Associate Professor Septimiu Mischie
Faculty of Electronics and Telecommunications
Politehnica University of Timisoara
Romania
E-mail: septimiu.mischie@etc.upt.ro

Abstract: Channel Impulse Response is used to model the propagation of the sound from a source (a speaker) to a receiver (a microphone). Thus, the signal recorded by the microphone can be computed by the convolution between the original source and the channel impulse response. There are methods which measure the channel impulse response: sine sweep method, maximum length sequence method or image method. First two methods use a hardware setup and the last is a simulation method. Also some adaptive methods which use the original source and the recorded signal can be used. Solutions to implement all these methods are presented in this review.
A good knowledge of channel impulse response is very important in many speech or audio processing system. Three such systems will be presented in this review.
A speech separation system considers a number of speech sources (loudspeakers) and a number of microphones. These microphones record mixtures of the sources generated by the loudspeakers. The function of this system is to recover the original sources from the recorded mixtures.
A dereverberation system has the task to cancel a part of the reverberation components which are added to the original source. Such a signal is obtained when a microphone is used to record the signal in a room.
Estimation of the direction of arrival considers two microphones and an acoustic source and its task is to determine the angle which corresponds to the position of the source in comparison with the two microphones.
This review presents the most used methods to implement the three systems which were presented previously. These methods include experiments achieved using a Personal Computer together with the sound card equipped with microphones and loudspeakers and MATLAB environment but also dedicated systems with microcontrollers and digital system processors.

Brief Biography of the Speaker: Septimiu Mischie received the Bachelor Engineering and Ph.D. in electronics and telecommunications from “Politehnica” University of Timisoara in 1989 and 1998, respectively. He joined Faculty of Electronics and Telecommunications of “Politehnica” University of Timisoara in 1991 and now is an associate professor with the Department of Measurement and Optoelectronics. His research interests are in speech processing (vector quantization of Line Spectral Frequencies, speech separation), instrumentation and measurement, data acquisition and embedded systems. He is author of more than 40 research papers in Conference Proceedings and Journals (9 of these papers are indexed in ISI Web of Science). Also he is principal author for three books and coauthor for other three. Three of his papers presented at WSEAS Conferences have received the Best Paper Awards.

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