Ultra Low Bit Rate Speech Coding

"Ultra Low Bit-Rate Speech Coding" focuses on the specialized topic of speech coding at very low bit-rates of 1 Kbits/sec and less, particularly at the lower ends of this range, down to 100 bps. The authors set forth the fundamental results and trends that form the basis for such ultra low bit-rates to be viable and provide a comprehensive overview of various techniques and systems in literature to date, with particular attention to their work in the paradigm of unit-selection based segment quantization. The book is for research students, academic faculty and researchers, and industry practitioners in the areas of speech processing and speech coding.

Digital Speech Coding for Low Bit Rate Communications Systems A. M. Kondoz University of Surrey UK Speech communication is still the dominant service in telecommunications networks. However, modern information requirements, such as multimedia, satellite and mobile communications, have created the need for a more robust and flexible alternative to the traditional analogue system. Digital speech transmission offers many advantages over its analogue counterpart, including consistent quality, versatility, security and successful integration into ISDN. This topical book provides an in-depth account of the latest developments in digital speech coders specifically designed for use in the evolving communications systems. Features include: * Fundamental material on sampling and quantisation. * Detailed coverage of speech signal analysis and modelling. * An examination of the key tools for speech coding. * Essential guidance on coding strategies and standards. * A review of a wide range of speech coding techniques, such as low delay speech coding and multi-band excitation speech coders. * Important information on error control in speech transmission. * A comprehensive assessment of system applications and real-time implementations, including unique coverage of the important code-excited linear prediction (CELP) method of speech coding. A complete examination of the theoretical design and practical applications of speech coders, this book will be invaluable for practising communications engineers, network designers and computer scientists. Advanced students in electrical, electronic and communications engineering, computer networking and digital signal processing will also find this text a helpful reference source.

Building on the success of the first edition Digital Speech offers extensive new, updated and revised material based upon the latest research. This Second Edition continues to provide the fundamental technical background required for low bit rate speech coding and the hottest developments in digital speech coding techniques that are applicable to evolving communication systems. Features new chapters on Pitch Estimation and Voice-Unvoiced Classification of Speech, Harmonic Speech Coding and Multimode Speech Coding Presents a comprehensively revised chapter entitled Analysis by Synthesis LPC Coding including specific examples of popular speech coders such as CELP (Code-Excited Linear Predictive) Coding Contains an updated chapter on Efficient LPC Quantization Methods including MSVQ and anti-aliasing filtering Discusses Voice Activity Detection (VAD) methods Offers expanded coverage of speech enhancement techniques such as echo cancellation and noise suppression Written by a well-known, highly respected academic, this authoritative volume will be invaluable to practising engineers, network designers, computer scientists and advanced students in communications, electrical and electronic engineering.

State of the art model based coders offer a perceptually acceptable reconstructed speech quality at bit-rates as low as 2000 bits per second. However, the performance of these coders rapidly deteriorates below this rate, primarily since very few bits are available to encode the model parameters with high fidelity. This thesis aims to meet the challenge of designing speech coders that operate at lower bit-rates while reconstructing the speech at the receiver at the same or even better quality than state of the art low bit-rate speech coders. In one of the contributions, we develop a plethora of techniques for efficient coding of the parameters obtained by the MELP algorithm, under the assumption that the classification of the frames of the MELP coder is available. Also, a simple and elegant procedure called dynamic codebook reordering is presented for use in the encoders and decoders of a vector quantization system that effectively exploits the correlation between vectors of parameters obtained from consecutiv speech frames without introducing any delay, distortion or suboptimality. The potential of this technique in significantly reducing the bit-rates of speech coders is illustrated. Additionally, the thesis also attempts to address the issues of designing such very low bit-rate speech coders so that they are robust to environmental noise. To impart robustness, a speech enhancement framework employing Kalman filters is presented. Kalman filters designed for speech enhancement in the presence of noise assume an autoregressive model for the speech signal. We improve the performance of Kalman filters in speech enhancement by constraining the parameters of the autoregressive model to belong to a codebook trained on clean speech. We then extend this formulation to the design of a novel framework, called the multiple input Kalman filter, that optimally combines the outputs from several speech enhancement systems. Since the low bit-rate speech coders compress the parameters significantly, it is very important to protect the transmitted information from errors in the communication channel. In this thesis, a novel channel-optimized multi-stage vector quantization codec is presented, in which the stage codebooks are jointly designed.

Speech coding has been an ongoing area of research for several decades, yet the level of activity and interest in this area has expanded dramatically in the last several years. Important advances in algorithmic techniques for speech coding have recently emerged and excellent progress has been achieved in producing high quality speech at bit rates as low as 4.8 kb/s. Although the complexity of the newer more sophisticated algorithms greatly exceeds that of older methods (such as ADPCM), today's powerful programmable signal processor chips allow rapid technology transfer from research to product development and permit many new cost-effective applications of speech coding. In particular, low bit rate voice technology is converging with the needs of the rapidly evolving digital telecom munication networks. The IEEE Workshop on Speech Coding for Telecommunications was held in Vancouver, British Columbia, Canada, from September 5 to 8, 1989. The objective of the workshop was to provide a forum for discussion of recent developments and future directions in speech coding. The workshop attracted over 130 researchers from several countries and its technical program included 51 papers.

Speech coding is a highly mature branch of signal processing deployed in products such as cellular phones, communication devices, and more recently, voice over internet protocol This book collects many of the techniques used in speech coding and presents them in an accessible fashion Emphasizes the foundation and evolution of standardized speech coders, covering standards from 1984 to the present The theory behind the applications is thoroughly analyzed and proved

Speech and Audio Coding for Wireless and Network Applications contains 34 chapters, loosely grouped into six topical areas. The chapters in this volume reflect the progress and present the state of the art in low-bit-rate speech coding, primarily at bit rates from 2.4 kbit/s to 16 kbit/s. Together they represent important contributions from leading researchers in the speech coding community. Speech and Audio Coding for Wireless and Network Applications contains contributions describing technologies that are under consideration as standards for such applications as digital cellular communications (the half-rate American and European coding standards). A brief Introduction is followed by a section dedicated to low-delay speech coding, a research direction which emerged as a result of the CCITT requirement for a universal low-delay 16 kbit/s speech coding technology and now continues with the objective of achieving toll quality with moderate delay at a rate of 8 kbit/s. A section on the important topic of speech quality evaluation is then presented. This is followed by a section on speech coding for wireless transmission, and a section on audio coding which covers not only 7 kHz bandwidth speech, but also wideband coding applicable to high fidelity music. The book concludes with a section on speech coding for noisy transmission channels, followed by a section addressing future research directions. Speech and Audio Coding for Wireless and Network Applications presents a cross-section of the key contributions in speech and audio coding which have emerged recently. For this reason, the book is a valuable reference for all researchers and graduate students in the speech coding community.

After alm ost three scores of years of basic and applied research, the field of speech processing is, at present, undergoing a rapid growth in terms of both performance and applications and this is fueHed by the advances being made in the areas of microelectronics, computation and algorithm design.Speech processing relates to three aspects of voice communications: -Speech Coding and transmission which is mainly concerned with man-to man voice communication. -Speech Synthesis which deals with machine-to-man communication. -Speech Recognition which is related to man-to-machine communication. Widespread application and use of low-bit rate voice codec.>, synthesizers and recognizers which are all speech processing products requires ideaHy internationally accepted quality assessment and evaluation methods as weH as speech processing standards so that they may be interconnected and used independently of their designers and manufacturers without costly interfaces. This book presents, in a tutorial manner, both fundamental and applied aspects of the above topics which have been prepared by weH-known specialists in their respective areas. The book is based on lectures which were sponsored by AGARD/NATO and delivered by the authors, in several NATO countries, to audiences consisting mainly of academic and industrial R&D engineers and physicists as weH as civil and military C3I systems planners and designers.