The Technology of Binaural Listening
This book reports on the application of advanced models of the human binaural hearing system in modern technology, among others, in the following areas: binaural analysis of aural scenes, binaural de-reverberation, binaural quality assessment of audio channels, loudspeakers and performance spaces, binaural perceptual coding, binaural processing in hearing aids and cochlea implants, binaural systems in robots, binaural/tactile human-machine interfaces, speech-intelligibility prediction in rooms and/or multi-speaker scenarios. An introduction to binaural modeling and an outlook to the future are provided. Further, the book features a MATLAB toolbox to enable readers to construct their own dedicated binaural models on demand.
Speech and Audio Processing in Adverse Environments
The book reflects the state of the art in important areas of speech and audio signal processing. It presents topics which are missed so far and most recent findings in the field. Leading international experts report on their field of work and their new results.Considerable amount of space is covered by multi-microphone systems, specific approaches for noise reduction, and evaluations of speech signals and speech processing systems.Multi-microphone systems include automatic calibration of microphones, localisation of sound sources, and source separation procedures. Also covered are recent approaches to the problem of adaptive echo and noise suppression. A novel solution allows the design of filter banks exhibiting bands spaced according to the Bark scale und especially short delay times. Furthermore, a method for engine noise reduction and proposals for improving the signal/noise ratio based on partial signal reconstruction or using a noise reference are reported. A number of contributions deal with speech quality. Besides basic considerations for quality evaluation specific methods for bandwidth extension of telephone speech are described. Procedures to reduce the reverberation of audio signals can help to increase speech intelligibility and speech recognition rates.In addition, solutions for specific applications in speech and audio signal processing are reported including, e.g., the enhancement of audio signal reproduction in automobiles and the automatic evaluation of hands-free systems and hearing aids.
10 Jahre wettbewerbsorientierte Regulierung von Netzindustrien in Deutschland
Vary, P. und Lüders, H.:
Mobiler Breitbandzugang in der Fläche durch gemeinsame Nutzung von Infrastruktur? – Technikaspekte und Regulierung –,
in: 10 Jahre wettbewerbsorientierte Regulierung von Netzindustrien in Deutschland, hrsg. von Picot, A., München: Verlag C. H. Beck, Feb. 2008, Kap. 9, ISBN: 978-3-40657-500-6.
Der Band erscheint aus Anlass der 10jährigen Gründung der Regulierungsbehörde für Telekommunikation und Post im Jahr 1998, die seit 2005 in Bundesnetzagentur umbenannt wurde. Die Beiträge in diesem Band machen deutlich, wie komplex und vielfältig die gelösten Aufgaben für die Regulierungsbehörde waren und wie viele offene Fragen die Gegenwart und Zukunft noch bereithalten.
Nachgegangen wird auch der zentrale Frage, wie viel bzw. wie wenig regulierende Eingriffe und Rahmenbedingungen benötigt werden, um Wettbewerb zu fördern und neue Dienste, Anbieter und Investitionen in den Netzindustrien auszulösen. Auch nach 10 Jahren sind die Meinungen hierzu nach wie vor kontrovers.
Mit Beiträgen von:Professor Dr. Dres. h.c. Arnold Picot, Universität München, Herrn Professor Dr. Juergen B. Donges, Universität zu Köln, Herrn Professor Dr. Charles B. Blankart, Humboldt-Universität zu Berlin, Herrn Professor Dr. Ludwig Gramlich, TU Chemnitz, Herrn Professor Dr. Dr. Franz Jürgen Säcker, Freie Universität Berlin, Herrn Professor Dr. Herbert Kubicek, Universität Bremen, Herrn Dr. Karl-Heinz Neumann, Wissenschaftliches Institut für Infrastruktur und Kommunikationsdienste GmbH, Herrn Professor Dr.-Ing. Peter Vary, Technische Hochschule Aachen, Herrn Professor Dr. Torsten J. Gerpott, Universität Duisburg, Herrn Professor Dr. Bernd Holznagel, LL.M., Universität Münster, Herrn Professor Dr.-Ing. Hans-Jürgen Haubrich, Technische Hochschule Aachen und Herrn Professor Dr. Dr. h.c. Wolfgang Ballwieser, Universität München.
Advances in Digital Speech Transmission
Adrat, M., Clevorn, T. und Schmalen, L.:
Iterative Source-Channel Decoding & Turbo DeCodulation,
in: Advances in Digital Speech Transmission, hrsg. von Martin, R., Heute, U. und Antweiler, C., Chichester, UK: John Wiley & Sons, Ltd., Jan. 2008, Kap. 13, S. 365–398, ISBN: 978-0-47051-739-0.
Multi-Channel System Identification with Perfect Sequences - Theory and Applications -,
in: Advances in Digital Speech Transmission, hrsg. von Martin, R., Heute, U. und Antweiler, C., Chichester, UK: John Wiley & Sons, Ltd., Jan. 2008, Kap. 7, S. 171–198, ISBN: 978-0-47051-739-0.
Geiser, B., Ragot, S. und Taddei, H.:
Embedded Speech Coding: From G.711 to G.729.1,
in: Advances in Digital Speech Transmission, hrsg. von Martin, R., Heute, U. und Antweiler, C., Chichester, UK: John Wiley & Sons, Ltd., Jan. 2008, Kap. 8, S. 201–247, ISBN: 978-0-47051-739-0.
Speech processing and speech transmission technology are expanding fields of active research. New challenges arise from the 'anywhere, anytime' paradigm of mobile communications, the ubiquitous use of voice communication systems in noisy environments and the convergence of communication networks toward Internet based transmission protocols, such as Voice over IP. As a consequence, new speech coding, new enhancement and error concealment, and new quality assessment methods are emerging.
Advances in Digital Speech Transmission provides an up-to-date overview of the field, including topics such as speech coding in heterogeneous communication networks, wideband coding, and the quality assessment of wideband speech.
- Provides an insight into the latest developments in speech processing and speech transmission, making it an essential reference to those working in these fields
- Offers a balanced overview of technology and applications
- Discusses topics such as speech coding in heterogeneous communications networks, wideband coding, and the quality assessment of the wideband speech
- Explains speech signal processing in hearing instruments and man-machine interfaces from applications point of view
- Covers speech coding for Voice over IP, blind source separation, digital hearing aids and speech processing for automatic speech recognition
Advances in Digital Speech Transmission serves as an essential link between the basics and the type of technology and applications (prospective) engineers work on in industry labs and academia. The book will also be of interest to advanced students, researchers, and other professionals who need to brush up their knowledge in this field.
Single and Multimicrophone Spectral Amplitude Estimation using a Super-Gaussian Speech Model,
in: Speech Enhancement, hrsg. von Benesty, J., Makino, S. und Chen, J., Springer Verlag, 2005.
We live in a noisy world! In all applications (telecommunications, hands-free communications, recording, human-machine interfaces, etc.) that require at least one microphone, the signal of interest is usually contaminated by noise and reverberation. As a result, the microphone signal has to be "cleaned" with digital signal processing tools before it is played out, transmitted, or stored.This book is about speech enhancement. Different well-known and state-of-the-art methods for noise reduction, with one or multiple microphones, are discussed. By speech enhancement, we mean not only noise reduction but also dereverberation and separation of independent signals. These topics are also covered in this book. However, the general emphasis is on noise reduction because of the large number of applications that can benefit from this technology.The goal of this book is to provide a strong reference for researchers, engineers, and graduate students who are interested in the problem of signal and speech enhancement. To do so, we invited well-known experts to contribute chapters covering the state of the art in this focused field.
Audio Bandwidth Extension
Bandwidth Extension for Speech,
in: Audio Bandwidth Extension, hrsg. von Larsen, E. und Aarts, R. M., New York, USA: John Wiley and Sons, Nov. 2004, Kap. 6, S. 171–236.
Bandwidth extension of signals may be defined as the deliberate process of expanding the frequency range (bandwidth) of a signal in which it contains an appreciable and useful content, and/or the frequency range in which its effects are such. Its significant advancement in recent years has led to the technology being adopted commercially in several areas including psychacoustic bass enhancement of small loudspeakers and the high frequency enhancement of perceptually coded audio. The increasing use of this technology in different areas creates a need for knowledge of the area. Written by an expert in the field, Signal Bandwidth Extension for Applications in Audio Signal Processing will consolidate the theories and applications surrounding signal bandwidth extension to help the reader develop a comprehensive knowledge of the area and its potential uses. An ideal reference for academics and professionals alike, it will be most suited to researchers and developers in audio engineering. However, it will also be suitable for those studying or working in areas such as signal processing, communications, audio and video engineering and coding theory.
Small Microphone Arrays with Postfilters for Noise and Acoustic Echo Reduction,
in: Microphone Arrays, hrsg. von Brandstein, M. und Ward, D., Springer, 2001, Kap. 12, S. 255–279.
This book provides, for the first time, a single complete reference on microphone arrays. Top researchers in this field contributed articles addressing their specific topics of study. The results cover the current state of the art in microphone array research, development, and technological application. Part I concerns the problem of enhancing the speech signal acquired by an array of microphones. Part II is devoted to the source localization problem. Part III details some specific applications of microphone array technology available today. Part IV presents expert summaries of current open problems in the field, as well as personal views of what the future of microphone array processing might hold. The individual chapters selected for the book were designed to be tutorial in nature with a specific emphasis on recent important results. They are of utility to a large audience, from the student or practising engineer just approaching the field to the experienced researcher.
Nato ASI Series F - Speech Recognition and Understanding: Recent Advances, Trends and Application
Robust Speech Recognition in Noisy and Reverberant Environment,
in: NATO ASI Series F - Speech Recognition and Understanding: Recent Advances, Trends and Application, hrsg. von Laface, P. und De Mori, R., Band 75, Springer-Verlag, 1992, S. 101–106.