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The first edition of this book has enjoyed a gratifying existence. 1s sued in 1965, it found its intended place as a research reference and as a graduate-Ievel text. Research laboratories and universities reported broad use. Published reviews-some twenty-five in number-were universally kind. Subsequently the book was translated and published in Russian (Svyaz; Moscow, 1968) and Spanish (Gredos, S.A.; Madrid, 1972). Copies of the first edition have been exhausted for several years, but demand for the material continues. At the behest of the publisher, and with the encouragement of numerous colleagues, a second edition was begun in 1970. The aim was to retain the original format, but to expand the content, especially in the areas of digital communications and com puter techniques for speech signal processing. As before, the intended audience is the graduate-Ievel engineer and physicist, but the psycho physicist, phonetician, speech scientist and linguist should find material of interest.
Human beings communicate expressively with each other in conversation: now in the computer age there is a perceived need for machines to communicate expressively with humans in dialogue. This title presents research examining expressive content in speech with a view to simulating expression in computer speech.
This book contains a complete and accurate mathematical treatment of the sounds of music with an emphasis on musical timbre. The book spans the range from tutorial introduction to advanced research and application to speculative assessment of its various techniques. All the contributors use a generalized additive sine wave model for describing musical timbre which gives a conceptual unity, but is of sufficient utility to be adapted to many different tasks.
This dissertation describes a speech system based on a combination of physiological and psychoacoustic results which has been developed. The system contains a nonuniform Filter/Detector bank. A new relationship between Filter/Detectors and the Short-time Fourier Transform magnitude is derived, and a generalized version of the Short-Time Fourier Transform magnitude is used to implement the anlaysis system. The new relationship is also applied to a discussion of channel vocoders, spectrograms, the sliding Discrete Fourier Transform, average power spectrum estimation, and nonuniform bandwidth analysis. Next, a new synthesis approach is used to reconstruct signals form the magnitude data produced by the nonuniform analysis. Apart form an overall sign factor, the analysis/synthesis system achieves exact reconstruction in the absence of data modification. The ability of the system to reconstruct signals from modified data is also demonstrated. Suggestions for further research, including data reduction and automatic speech recognition applications, are given. Keywords include: Auditory modeling, short-time fourier transform, magnitude-only reconstruction, Power spectrum estimation, Perception, Filter banks, Speech recognition, Spectrograms, and Vocoders.
With a growing need for understanding the process involved in producing and perceiving spoken language, this timely publication answers these questions in an accessible reference. Containing material resulting from many years’ teaching and research, Speech Synthesis provides a complete account of the theory of speech. By bringing together the common goals and methods of speech synthesis into a single resource, the book will lead the way towards a comprehensive view of the process involved in human speech. The book includes applications in speech technology and speech synthesis. It is ideal for intermediate students of linguistics and phonetics who wish to proceed further, as well as researchers and engineers in telecommunications working in speech technology and speech synthesis who need a comprehensive overview of the field and who wish to gain an understanding of the objectives and achievements of the study of speech production and perception.