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Author : Steven T. Stoiber
Publisher :
ISBN 13 :
Total Pages : 39 pages
Book Rating : 4.:/5 (33 download)
Download or read book Low Power Digital Signal Processing for Speech Recognition written by Steven T. Stoiber and published by . This book was released on 1994 with total page 39 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author : Keshab K. Parhi
Publisher : John Wiley & Sons
ISBN 13 : 9788126510986
Total Pages : 960 pages
Book Rating : 4.5/5 (19 download)
Download or read book VLSI DIGITAL SIGNAL PROCESSING SYSTEMS: DESIGN AND IMPLEMENTATION written by Keshab K. Parhi and published by John Wiley & Sons. This book was released on 2007 with total page 960 pages. Available in PDF, EPUB and Kindle. Book excerpt: Market_Desc: · Students in graduate level courses· Electrical Engineers· Computer Scientists· Computer Architecture Designers· Circuit Designers· Algorithm Designers· System Designers· Computer Programmers in the Multimedia and Wireless Communications Industries· VLSI System Designers Special Features: This example-packed resource provides invaluable professional training for a rapidly-expanding industry. · Presents a variety of approaches to analysis, estimation, and reduction of power consumption in order to help designers extend battery life.· Includes application-driven problems at the end of each chapter· Features six appendices covering shortest path algorithms used in retiming, scheduling, and allocation techniques, as well as determining the iteration bound· The Author is a recognized expert in the field, having written several books, taught several graduate-level classes, and served on several IEEE boards About The Book: This book complements the other Digital Signaling Processing books in our list, which include an introductory treatment (Marven), a comprehensive handbook (Mitra), a professional reference (Kaloupsidis), and others which pertain to a specific topic such as noise control. This graduate level textbook will fill an important niche in a rapidly expanding market.
Author : Geoff Bristow
Publisher :
ISBN 13 :
Total Pages : 422 pages
Book Rating : 4.F/5 ( download)
Download or read book Electronic Speech Recognition written by Geoff Bristow and published by . This book was released on 1986 with total page 422 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author : Lawrence R. Rabiner
Publisher : Now Publishers Inc
ISBN 13 : 1601980701
Total Pages : 212 pages
Book Rating : 4.6/5 (19 download)
Download or read book Introduction to Digital Speech Processing written by Lawrence R. Rabiner and published by Now Publishers Inc. This book was released on 2007 with total page 212 pages. Available in PDF, EPUB and Kindle. Book excerpt: Provides the reader with a practical introduction to the wide range of important concepts that comprise the field of digital speech processing. Students of speech research and researchers working in the field can use this as a reference guide.
Author : A. Nejat Ince
Publisher : Springer Science & Business Media
ISBN 13 : 147572148X
Total Pages : 254 pages
Book Rating : 4.4/5 (757 download)
Download or read book Digital Speech Processing written by A. Nejat Ince and published by Springer Science & Business Media. This book was released on 2013-03-09 with total page 254 pages. Available in PDF, EPUB and Kindle. Book excerpt: 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.
Author : Keshab K. Parhi
Publisher : Wiley-Interscience
ISBN 13 :
Total Pages : 824 pages
Book Rating : 4.3/5 (91 download)
Download or read book VLSI Digital Signal Processing Systems written by Keshab K. Parhi and published by Wiley-Interscience. This book was released on 1999 with total page 824 pages. Available in PDF, EPUB and Kindle. Book excerpt: Digital audio, speech recognition, cable modems, radar, high-definition television-these are but a few of the modern computer and communications applications relying on digital signal processing (DSP) and the attendant application-specific integrated circuits (ASICs). As information-age industries constantly reinvent ASIC chips for lower power consumption and higher efficiency, there is a growing need for designers who are current and fluent in VLSI design methodologies for DSP. Enter VLSI Digital Signal Processing Systems-a unique, comprehensive guide to performance optimization techniques in VLSI signal processing. Based on Keshab Parhi's highly respected and popular graduate-level courses, this volume is destined to become the standard text and reference in the field. This text integrates VLSI architecture theory and algorithms, addresses various architectures at the implementation level, and presents several approaches to analysis, estimation, and reduction of power consumption. Throughout this book, Dr. Parhi explains how to design high-speed, low-area, and low-power VLSI systems for a broad range of DSP applications. He covers pipelining extensively as well as numerous other techniques, from parallel processing to scaling and roundoff noise computation. Readers are shown how to apply all techniques to improve implementations of several DSP algorithms, using both ASICs and off-the-shelf programmable digital signal processors. The book features hundreds of graphs illustrating the various DSP algorithms, examples based on digital filters and transforms clarifying key concepts, and interesting end-of-chapter exercises that help match techniques with applications. In addition, the abundance of readily available techniques makes this an extremely useful resource for designers of DSP systems in wired, wireless, or multimedia communications. The material can be easily adopted in new courses on either VLSI digital signal processing architectures or high-performance VLSI system design. An invaluable reference and practical guide to VLSI digital signal processing. A tremendous source of optimization techniques indispensable in modern VLSI signal processing, VLSI Digital Signal Processing Systems promises to become the standard in the field. It offers a rich training ground for students of VLSI design for digital signal processing and provides immediate access to state-of-the-art, proven techniques for designers of DSP applications-in wired, wireless, or multimedia communications. Topics include: * Transformations for high speed using pipelining, retiming, and parallel processing techniques * Power reduction transformations for supply voltage reduction as well as for strength or capacitance reduction * Area reduction using folding techniques * Strategies for arithmetic implementation * Synchronous, wave, and asynchronous pipelining * Design of programmable DSPs. An Instructor's Manual presenting detailed solutions to all the problems in the book is available from the Wiley editorial department.
Author : Perez-Meana, Hector
Publisher : IGI Global
ISBN 13 : 1599041340
Total Pages : 462 pages
Book Rating : 4.5/5 (99 download)
Download or read book Advances in Audio and Speech Signal Processing: Technologies and Applications written by Perez-Meana, Hector and published by IGI Global. This book was released on 2007-02-28 with total page 462 pages. Available in PDF, EPUB and Kindle. Book excerpt: "This book provides a comprehensive approach of signal processing tools regarding the enhancement, recognition, and protection of speech and audio signals. It offers researchers and practitioners the information they need to develop and implement efficient signal processing algorithms in the enhancement field"--Provided by publisher.
Author : Sadaoki Furui
Publisher : CRC Press
ISBN 13 : 1351990926
Total Pages : 338 pages
Book Rating : 4.3/5 (519 download)
Download or read book Digital Speech Processing written by Sadaoki Furui and published by CRC Press. This book was released on 2018-05-04 with total page 338 pages. Available in PDF, EPUB and Kindle. Book excerpt: A study of digital speech processing, synthesis and recognition. This second edition contains new sections on the international standardization of robust and flexible speech coding techniques, waveform unit concatenation-based speech synthesis, large vocabulary continuous-speech recognition based on statistical pattern recognition, and more.
Author : Muhammad S. Elrabaa
Publisher : Springer Science & Business Media
ISBN 13 : 1441985468
Total Pages : 210 pages
Book Rating : 4.4/5 (419 download)
Download or read book Advanced Low-Power Digital Circuit Techniques written by Muhammad S. Elrabaa and published by Springer Science & Business Media. This book was released on 2012-10-28 with total page 210 pages. Available in PDF, EPUB and Kindle. Book excerpt: Advanced Low-Power Digital Circuit Techniques presents several novel high performance digital circuit designs that emphasize low-power and low-voltage operation. These circuits represent a wide range of circuits that are used in state-of-the-art VLSI systems and hence serve as good examples for low-power design. Each chapter contains a brief introduction that serves as a quick background and gives the motivation behind the design. Each chapter also ends with a summary that briefly explains the contributions contained therein. This makes the book very readable. The reader can skim through the chapters very quickly to get a feel for the design problems presented in the book and the solutions proposed by the authors. Examples of circuits used in systems where low-power is important from reliability and portability points of view (such as general-purpose and DSP processors) are presented in Chapters 2, 3 and 4. Chapters 5 and 7 give examples of circuits used in systems where reliability and more system integration are the main driving forces behind lowering the power consumption. Chapter 6 gives an example of a general purpose high-performance low-power circuit design. Advanced Low-Power Digital Circuit Techniques is a real designer's book. It investigates alternative circuit styles, as well as architectural alternatives, and gives quantitative results for comparison in realistic technologies. Several of the circuits presented have been fabricated so that simulations can be checked. The circuits covered are the most important building blocks for many designs, so the text will be of direct use to designers. MOS designs are covered, as well as BiCMOS, and there are several novel circuits.
Author : Lawrence R. Rabiner
Publisher : Prentice Hall
ISBN 13 :
Total Pages : 536 pages
Book Rating : 4.3/5 (91 download)
Download or read book Digital Processing of Speech Signals written by Lawrence R. Rabiner and published by Prentice Hall. This book was released on 1978 with total page 536 pages. Available in PDF, EPUB and Kindle. Book excerpt: The material in this book is intended as a one-semester course in speech processing. The purpose of this text is to show how digital signal processing techniques can be applied to problems related to speech communication. The book gives an extensive description of the physical basis for speech coding including fourier analysis, digital representation and digital and time domain models of the wave form. It goes on to discuss homomorphic speech processing, linear predictive coding and digital processing for machine communication by voice.
Author : Vijay Madisetti
Publisher : CRC Press
ISBN 13 : 1420046098
Total Pages : 616 pages
Book Rating : 4.4/5 (2 download)
Download or read book Video, Speech, and Audio Signal Processing and Associated Standards written by Vijay Madisetti and published by CRC Press. This book was released on 2018-09-03 with total page 616 pages. Available in PDF, EPUB and Kindle. Book excerpt: Now available in a three-volume set, this updated and expanded edition of the bestselling The Digital Signal Processing Handbook continues to provide the engineering community with authoritative coverage of the fundamental and specialized aspects of information-bearing signals in digital form. Encompassing essential background material, technical details, standards, and software, the second edition reflects cutting-edge information on signal processing algorithms and protocols related to speech, audio, multimedia, and video processing technology associated with standards ranging from WiMax to MP3 audio, low-power/high-performance DSPs, color image processing, and chips on video. Drawing on the experience of leading engineers, researchers, and scholars, the three-volume set contains 29 new chapters that address multimedia and Internet technologies, tomography, radar systems, architecture, standards, and future applications in speech, acoustics, video, radar, and telecommunications. This volume, Video, Speech, and Audio Signal Processing and Associated Standards, provides thorough coverage of the basic foundations of speech, audio, image, and video processing and associated applications to broadcast, storage, search and retrieval, and communications.
Author : Jacob Benesty
Publisher : Elsevier
ISBN 13 : 0128002530
Total Pages : 143 pages
Book Rating : 4.1/5 (28 download)
Download or read book Speech Enhancement written by Jacob Benesty and published by Elsevier. This book was released on 2014-01-04 with total page 143 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech enhancement is a classical problem in signal processing, yet still largely unsolved. Two of the conventional approaches for solving this problem are linear filtering, like the classical Wiener filter, and subspace methods. These approaches have traditionally been treated as different classes of methods and have been introduced in somewhat different contexts. Linear filtering methods originate in stochastic processes, while subspace methods have largely been based on developments in numerical linear algebra and matrix approximation theory. This book bridges the gap between these two classes of methods by showing how the ideas behind subspace methods can be incorporated into traditional linear filtering. In the context of subspace methods, the enhancement problem can then be seen as a classical linear filter design problem. This means that various solutions can more easily be compared and their performance bounded and assessed in terms of noise reduction and speech distortion. The book shows how various filter designs can be obtained in this framework, including the maximum SNR, Wiener, LCMV, and MVDR filters, and how these can be applied in various contexts, like in single-channel and multichannel speech enhancement, and in both the time and frequency domains. First short book treating subspace approaches in a unified way for time and frequency domains, single-channel, multichannel, as well as binaural, speech enhancement Bridges the gap between optimal filtering methods and subspace approaches Includes original presentation of subspace methods from different perspectives
Author : Kai-Fu Lee
Publisher : Springer Science & Business Media
ISBN 13 : 9780898382969
Total Pages : 232 pages
Book Rating : 4.3/5 (829 download)
Download or read book Automatic Speech Recognition written by Kai-Fu Lee and published by Springer Science & Business Media. This book was released on 1988-10-31 with total page 232 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech Recognition has a long history of being one of the difficult problems in Artificial Intelligence and Computer Science. As one goes from problem solving tasks such as puzzles and chess to perceptual tasks such as speech and vision, the problem characteristics change dramatically: knowledge poor to knowledge rich; low data rates to high data rates; slow response time (minutes to hours) to instantaneous response time. These characteristics taken together increase the computational complexity of the problem by several orders of magnitude. Further, speech provides a challenging task domain which embodies many of the requirements of intelligent behavior: operate in real time; exploit vast amounts of knowledge, tolerate errorful, unexpected unknown input; use symbols and abstractions; communicate in natural language and learn from the environment. Voice input to computers offers a number of advantages. It provides a natural, fast, hands free, eyes free, location free input medium. However, there are many as yet unsolved problems that prevent routine use of speech as an input device by non-experts. These include cost, real time response, speaker independence, robustness to variations such as noise, microphone, speech rate and loudness, and the ability to handle non-grammatical speech. Satisfactory solutions to each of these problems can be expected within the next decade. Recognition of unrestricted spontaneous continuous speech appears unsolvable at present. However, by the addition of simple constraints, such as clarification dialog to resolve ambiguity, we believe it will be possible to develop systems capable of accepting very large vocabulary continuous speechdictation.
Author : Michael D. Riley
Publisher : Springer Science & Business Media
ISBN 13 : 1461310792
Total Pages : 169 pages
Book Rating : 4.4/5 (613 download)
Download or read book Speech Time-Frequency Representations written by Michael D. Riley and published by Springer Science & Business Media. This book was released on 2012-12-06 with total page 169 pages. Available in PDF, EPUB and Kindle. Book excerpt: 1.1. Steps in the initial auditory processing. 4 2 THE TIME-FREQUENCY ENERGY REPRESENTATION 2.1. Short-time spectrum of a steady-state Iii. 9 2.2. Smoothed short-time spectra. 9 2.3. Short-time spectra of linear chirps. 13 2.4. Short-time spectra of /w /'s. 15 2.5. Wide band spectrograms of /w /'s. 16 Spectrograms of rapid formant motion. 2.6. 17 2.7. Wigner distribution and spectrogram. 21 2.8. Wigner distribution and spectrogram of cos wot. 23 2.9. Concentration ellipses for transform kernels. 28 2.10. Concentration ellipses for complementary kernels. 42 42 2.11. Directional transforms for a linear chirp. 47 2.12. Spectrograms of /wioi/ with different window sizes. 2.13. Wigner distribution of /wioi/. 49 2.14. Time-frequency autocorrelation function of /wioi/. 49 2.15. Gaussian transform of Iwioi/. 50 2.16. Directional transforms of lwioi/. 52 3 TIME-FREQUENCY FILTERING 3.1. Recovering the transfer function by filtering. 57 3.2. Estimating 'aliased' transfer function. 61 3.3. T-F autocorrelation function of an impulse train. 70 3.4. T-F autocorrelation function of LTI filter output. 70 Windowing recovers transfer function. 3.5. 72 3.6. Shearing the time-frequency autocorrelation function. 75 3.7. T-F autocorrelation function for FM filter. 76 3.8. T-F autocorrelation function of FM filter output. 77 3.9. Windowing recovers transfer function. 79 4 THE SCHEMATIC SPECTROGRAM Problems with pole-fitting approach.
Author : Shaila Dinkar Apte
Publisher : CRC Press
ISBN 13 : 1498782019
Total Pages : 463 pages
Book Rating : 4.4/5 (987 download)
Download or read book Random Signal Processing written by Shaila Dinkar Apte and published by CRC Press. This book was released on 2017-08-15 with total page 463 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book covers random signals and random processes along with estimation of probability density function, estimation of energy spectral density and power spectral density. The properties of random processes and signal modelling are discussed with basic communication theory estimation and detection. MATLAB simulations are included for each concept with output of the program with case studies and project ideas. The chapters progressively introduce and explain the concepts of random signals and cover multiple applications for signal processing. The book is designed to cater to a wide audience starting from the undergraduates (electronics, electrical, instrumentation, computer, and telecommunication engineering) to the researchers working in the pertinent fields. Key Features: • Aimed at random signal processing with parametric signal processing-using appropriate segment size. • Covers speech, image, medical images, EEG and ECG signal processing. • Reviews optimal detection and estimation. • Discusses parametric modeling and signal processing in transform domain. • Includes MATLAB codes and relevant exercises, case studies and solved examples including multiple choice questions
Author : Heejong Yoo
Publisher :
ISBN 13 :
Total Pages : pages
Book Rating : 4.:/5 (67 download)
Download or read book Low-power Audio Input Enhancement for Portable Devices written by Heejong Yoo and published by . This book was released on 2005 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: With the development of VLSI and wireless communication technology, portable devices such as personal digital assistants (PDAs), pocket PCs, and mobile phones have gained a lot of popularity. Many such devices incorporate a speech recognition engine, enabling users to interact with the devices using voice-driven commands and text-to-speech synthesis. The power consumption of DSP microprocessors has been consistently decreasing by half about every 18 months, following Gene's law. The capacity of signal processing, however, is still significantly constrained by the limited power budget of these portable devices. In addition, analog-to-digital (A/D) converters can also limit the signal processing of portable devices. Many systems require very high-resolution and high-performance A/D converters, which often consume a large fraction of the limited power budget of portable devices. The proposed research develops a low-power audio signal enhancement system that combines programmable analog signal processing and traditional digital signal processing. By utilizing analog signal processing based on floating-gate transistor technology, the power consumption of the overall system as well as the complexity of the A/D converters can be reduced significantly. The system can be used as a front end of portable devices in which enhancement of audio signal quality plays a critical role in automatic speech recognition systems on portable devices. The proposed system performs background audio noise suppression in a continuous-time domain using analog computing elements and acoustic echo cancellation in a discrete-time domain using an FPGA.
Author : Sen M. Kuo
Publisher : John Wiley & Sons
ISBN 13 : 1118706684
Total Pages : 532 pages
Book Rating : 4.1/5 (187 download)
Download or read book Real-Time Digital Signal Processing written by Sen M. Kuo and published by John Wiley & Sons. This book was released on 2013-08-05 with total page 532 pages. Available in PDF, EPUB and Kindle. Book excerpt: Combines both the DSP principles and real-time implementations and applications, and now updated with the new eZdsp USB Stick, which is very low cost, portable and widely employed at many DSP labs. Real-Time Digital Signal Processing introduces fundamental digital signal processing (DSP) principles and will be updated to include the latest DSP applications, introduce new software development tools and adjust the software design process to reflect the latest advances in the field. In the 3rd edition of the book, the key aspect of hands-on experiments will be enhanced to make the DSP principles more interesting and directly interact with the real-world applications. All of the programs will be carefully updated using the most recent version of software development tools and the new TMS320VC5505 eZdsp USB Stick for real-time experiments. Due to its lower cost and portability, the new software and hardware tools are now widely used in university labs and in commercial industrial companies to replace the older and more expensive generation. The new edition will have a renewed focus on real-time applications and will offer step-by-step hands-on experiments for a complete design cycle starting from floating-point C language program to fixed-point C implementation, code optimization using INTRINSICS, and mixed C-and-assembly programming on fixed-point DSP processors. This new methodology enables readers to concentrate on learning DSP fundamentals and innovative applications by relaxing the intensive programming efforts, namely, the traditional DSP assembly coding efforts. The book is organized into two parts; Part One introduces the digital signal processing principles and theories, and Part Two focuses on practical applications. The topics for the applications are the extensions of the theories in Part One with an emphasis placed on the hands-on experiments, systematic design and implementation approaches. The applications provided in the book are carefully chosen to reflect current advances of DSP that are of most relevance for the intended readership. Combines both the DSP principles and real-time implementations and applications using the new eZdsp USB Stick, which is very low cost, portable and widely employed at many DSP labs is now used in the new edition Places renewed emphasis on C-code experiments and reduces the exercises using assembly coding; effective use of C programming, fixed-point C code and INTRINSICS will become the main focus of the new edition. Updates to application areas to reflect latest advances such as speech coding techniques used for next generation networks (NGN), audio coding with surrounding sound, wideband speech codec (ITU G.722.2 Standard), fingerprint for image processing, and biomedical signal processing examples. Contains new addition of several projects that can be used as semester projects; as well as new many new real-time experiments using TI’s binary libraries – the experiments are prepared with flexible interface and modular for readers to adapt and modify to create other useful applications from the provided basic programs. Consists of more MATLAB experiments, such as filter design, algorithm evaluation, proto-typing for C-code architecture, and simulations to aid readers to learn DSP fundamentals. Includes supplementary material of program and data files for examples, applications, and experiments hosted on a companion website. A valuable resource for Postgraduate students enrolled on DSP courses focused on DSP implementation & applications as well as Senior undergraduates studying DSP; engineers and programmers who need to learn and use DSP principles and development tools for their projects.
