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A virtual environment aims at creating situations in which humans have perceptions that do not correspond to their physical environment but to a virtual one. The present chapter focuses on auditory virtual environments, i. e. on the auditory component of virtual environments as seen independently from the remaining modalities. Section 11.1 introduces and discusses concepts, such as auditory event and plausibility, required to the understanding of the remaining chapter. Section 11.2 presents the various components of a typical auditory virtual environment along with the techniques and models most often employed by each of them. Section 11.3 presents specific implementations of auditory virtual environments and addresses a selection of current research issues, namely, multi-modality, presence, quality and joint reality. Section 11.4 closes the chapter with a summary of the main conclusions.

Digital audio technology is allowing music and speech to be easily and readily accessible to most people, since these signals are treated as digital media and, hence, are significant components of the information-age revolution. From its commercial introduction via the audio Compact Disc, CD, approximately 20 years ago, this technology had a meteoric evolution which have seen the introduction of numerous methods, techniques, systems and formats and has allowed the users to benefit from reductions in the size of digital audio equipment and its cost. An overview of these developments is presented here with a critical assessment of their significance, along with a reference to many important publications and events. It is shown that this technology is mainly rooted on three constituent evolutionary components, namely, (a) digital electronics and computer technology, (b) DSP theory and techniques, (c) auditory modelling. Based on the analysis of these components, some conclusions are drawn, which allow the prediction of future trends concerning the evolution of this technology.

The mechanism of speech production is explained for the different speech sounds. The role of the most relevant articulators is discussed as well as, to a certain extent, sound excitation. The main cavities of the speech-production system are modelled by concatenation of short homogeneous acoustic tubes, interconnected by appropriate adaptors. Based on the concept of forward- and backward-travelling waves, signal-flow graphs are given, describing pressure, flow or power waves. Losses can be considered either by lumped impedances or distributed along the tube by properly designed filters. The estimation of a lossless un-branched tube system can easily be achieved by linear prediction from the speech signal. If losses are included and the tube system is branched, optimization algorithms can be applied for the parameter estimation. The termination at the glottis is assumed to be either fixed or time dependent according to the glottal opening. The mouth opening is described by a simple frequency-dependent termination. Applications in the fields of speech synthesis, source coding and recognition are briefly discussed.

The historic “coding gap” between high-rate coding of narrow- and wide-band speech on the one hand, and low-rate coding of narrow-band speech on the other hand, has been bridged more and more during the past 15 years. The GSM coder of 1990 was a very important milestone in this process, as it has prompted increasing research towards better quality and higher compression. These particular research efforts, together with other relevant activities worldwide, have helped closing the gap. In the following, the concepts behind this progress will be explained. A special focus will be put on the basis of this success, namely, on the fact that, finally, a break-through could be achieved for narrow-band speech, allowing for good quality at medium-to-low rates. For wide-band speech this holds true at medium rates. The same concepts as applied to speech coding were also applied to music, yet, ending up with some noticeable conceptual differences. While for speech time-domain approaches prevail, frequency-domain coding turned out to be more successful for audio. A characteristic, there, is extensive exploitation of psycho-acoustical phenomena.