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Book Digital Processing of Speech Materials  A Critical Band Based Model of Speech Perception

Download or read book Digital Processing of Speech Materials A Critical Band Based Model of Speech Perception written by Robert D. Celmer and published by . This book was released on 1980 with total page 61 pages. Available in PDF, EPUB and Kindle. Book excerpt: Existing literature suggests that the hearing mechanism deals with incoming speech material by filtering the signals into a series of frequency bands. The width of these bands has been referred to as the critical band that is the perceptual frequency bandwidth observed in a variety of psychoacoustic contexts. Digital processing techniques have been developed for altering available recorded speech materials so that the frequency resolution available in the resultant stimuli may be controlled. Tapes have been produced wherein the frequency bandwidth resolution is limited to no better than one critical band and these tapes have been used in intelligibility testing. Some existing research indicates that the critical band is significantly widened in many individuals with sensorineural hearing loss of cochlear etiology. The digital processing routines described above were also used in developing tape recorded materials with bandwidth resolution limits considerably wider than the normal critical band. The bandwidths chosen for this stage of the digital processing were based on empirical observations of the critical band of sensorineural hearing impaired patients. These recordings were also used in intelligibility testing with normal listeners. Implications of these studies for the clinical measurement of speech intelligibility will be discussed. (Author).

Book Speech and Audio Signal Processing

Download or read book Speech and Audio Signal Processing written by Bernard Gold and published by . This book was released on 2000 with total page 562 pages. Available in PDF, EPUB and Kindle. Book excerpt: This text provides readers with a comprehensive coverage of speech and audio signal processing available. These topics include everything from the basic foundation material on digital signal processing, pattern recognition, acoustics, and hearing, to material of historical significance.

Book Speech and Audio Signal Processing

Download or read book Speech and Audio Signal Processing written by Ben Gold and published by John Wiley & Sons. This book was released on 2011-08-23 with total page 684 pages. Available in PDF, EPUB and Kindle. Book excerpt: When Speech and Audio Signal Processing published in 1999, it stood out from its competition in its breadth of coverage and its accessible, intutiont-based style. This book was aimed at individual students and engineers excited about the broad span of audio processing and curious to understand the available techniques. Since then, with the advent of the iPod in 2001, the field of digital audio and music has exploded, leading to a much greater interest in the technical aspects of audio processing. This Second Edition will update and revise the original book to augment it with new material describing both the enabling technologies of digital music distribution (most significantly the MP3) and a range of exciting new research areas in automatic music content processing (such as automatic transcription, music similarity, etc.) that have emerged in the past five years, driven by the digital music revolution. New chapter topics include: Psychoacoustic Audio Coding, describing MP3 and related audio coding schemes based on psychoacoustic masking of quantization noise Music Transcription, including automatically deriving notes, beats, and chords from music signals. Music Information Retrieval, primarily focusing on audio-based genre classification, artist/style identification, and similarity estimation. Audio Source Separation, including multi-microphone beamforming, blind source separation, and the perception-inspired techniques usually referred to as Computational Auditory Scene Analysis (CASA).

Book Robustness in Automatic Speech Recognition

Download or read book Robustness in Automatic Speech Recognition written by Jean-Claude Junqua and published by Springer. This book was released on 1996 with total page 480 pages. Available in PDF, EPUB and Kindle. Book excerpt: The domain of speech processing has come to the point where researchers and engineers are concerned with how speech technology can be applied to new products, and how this technology will transform our future. One important problem is to improve robustness of speech processing under adverse conditions, which is the subject of this book. Robust speech processing is a relatively new area which became a concern as technology started moving from laboratory to field applications. A method or an algorithm is robust if it can deal with a broad range of applications and adapt to unknown conditions. Robustness in Automatic Speech Recognition addresses all of the fundamental problems and issues in the area. The book is divided into three parts. The first provides the background necessary for understanding the rest of the material. It also emphasizes the problems of speech production and perception in noise along with popular techniques used in speech analysis and automatic speech recognition. Part Two discusses the problems relevant to robustness in automatic speech recognition and speech-based applications. It emphasizes intra- and inter-speaker variability as well as automatic speech recognition of Lombard, noisy and channel distorted speech. Finally, the third part covers recent advances in the field of robust automatic speech recognition. Audience: An invaluable reference. May be used as a text for advanced courses on the subject.

Book Speech and Audio Processing

Download or read book Speech and Audio Processing written by Ian McLoughlin and published by Cambridge University Press. This book was released on 2016-07-21 with total page 403 pages. Available in PDF, EPUB and Kindle. Book excerpt: An accessible introduction to speech and audio processing with numerous practical illustrations, exercises, and hands-on MATLAB® examples.

Book Algorithms and Software for Predictive and Perceptual Modeling of Speech

Download or read book Algorithms and Software for Predictive and Perceptual Modeling of Speech written by Venkatraman Atti and published by Springer Nature. This book was released on 2022-05-31 with total page 113 pages. Available in PDF, EPUB and Kindle. Book excerpt: From the early pulse code modulation-based coders to some of the recent multi-rate wideband speech coding standards, the area of speech coding made several significant strides with an objective to attain high quality of speech at the lowest possible bit rate. This book presents some of the recent advances in linear prediction (LP)-based speech analysis that employ perceptual models for narrow- and wide-band speech coding. The LP analysis-synthesis framework has been successful for speech coding because it fits well the source-system paradigm for speech synthesis. Limitations associated with the conventional LP have been studied extensively, and several extensions to LP-based analysis-synthesis have been proposed, e.g., the discrete all-pole modeling, the perceptual LP, the warped LP, the LP with modified filter structures, the IIR-based pure LP, all-pole modeling using the weighted-sum of LSP polynomials, the LP for low frequency emphasis, and the cascade-form LP. These extensions can be classified as algorithms that either attempt to improve the LP spectral envelope fitting performance or embed perceptual models in the LP. The first half of the book reviews some of the recent developments in predictive modeling of speech with the help of MatlabTM Simulation examples. Advantages of integrating perceptual models in low bit rate speech coding depend on the accuracy of these models to mimic the human performance and, more importantly, on the achievable "coding gains" and "computational overhead" associated with these physiological models. Methods that exploit the masking properties of the human ear in speech coding standards, even today, are largely based on concepts introduced by Schroeder and Atal in 1979. For example, a simple approach employed in speech coding standards is to use a perceptual weighting filter to shape the quantization noise according to the masking properties of the human ear. The second half of the book reviews some of the recent developments in perceptual modeling of speech (e.g., masking threshold, psychoacoustic models, auditory excitation pattern, and loudness) with the help of MatlabTM simulations. Supplementary material including MatlabTM programs and simulation examples presented in this book can also be accessed here. Table of Contents: Introduction / Predictive Modeling of Speech / Perceptual Modeling of Speech

Book Masters Theses in the Pure and Applied Sciences

Download or read book Masters Theses in the Pure and Applied Sciences written by W. H. Shafer and published by Springer Science & Business Media. This book was released on 2012-12-06 with total page 311 pages. Available in PDF, EPUB and Kindle. Book excerpt: Masters Theses in the Pure and Applied Sciences was first conceived, published, and dis seminated by the Center for Information and Numerical Data Analysis and Synthesis (CINDAS) * at Purdue University in 1957, starting its coverage of theses with the academic year 1955. Beginning with Volume 13, the printing and dissemination phases of the ac tivity were transferred to University Microfilms/Xerox of Ann Arbor, Michigan, with the thought that such an arrangement would be more beneficial to the academic and general scientific and technical community. After five years of this joint undertaking we had concluded that it was in the interest of all concerned if the printing and distribution of the volume were handled by an international publishing. house to assure improved service and broader dissemination. Hence, starting with Volume 18, Masters Theses in the Pure and Applied Sciences has been disseminated on a worldwide basis by Plenum Publishing Corporation of New York, and in the same year the coverage was broadened to include Canadian universities. All back issues can also be ordered from Plenum. We have reported in Volume 25 (thesis year 1980) a total of 10,308 theses titles from 27 Canadian and 214 United States universities. We are sure that this broader base for theses titles reported will greatly enhance the value of this important annual reference work. While Volume 25 reports theses submitted in 1980, on occasion, certain universities do report theses submitted in previous years but not reported at the time.

Book Dynamic Speech Models

Download or read book Dynamic Speech Models written by Li Deng and published by Springer Nature. This book was released on 2022-05-31 with total page 105 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech dynamics refer to the temporal characteristics in all stages of the human speech communication process. This speech “chain” starts with the formation of a linguistic message in a speaker's brain and ends with the arrival of the message in a listener's brain. Given the intricacy of the dynamic speech process and its fundamental importance in human communication, this monograph is intended to provide a comprehensive material on mathematical models of speech dynamics and to address the following issues: How do we make sense of the complex speech process in terms of its functional role of speech communication? How do we quantify the special role of speech timing? How do the dynamics relate to the variability of speech that has often been said to seriously hamper automatic speech recognition? How do we put the dynamic process of speech into a quantitative form to enable detailed analyses? And finally, how can we incorporate the knowledge of speech dynamics into computerized speech analysis and recognition algorithms? The answers to all these questions require building and applying computational models for the dynamic speech process. What are the compelling reasons for carrying out dynamic speech modeling? We provide the answer in two related aspects. First, scientific inquiry into the human speech code has been relentlessly pursued for several decades. As an essential carrier of human intelligence and knowledge, speech is the most natural form of human communication. Embedded in the speech code are linguistic (as well as para-linguistic) messages, which are conveyed through four levels of the speech chain. Underlying the robust encoding and transmission of the linguistic messages are the speech dynamics at all the four levels. Mathematical modeling of speech dynamics provides an effective tool in the scientific methods of studying the speech chain. Such scientific studies help understand why humans speak as they do and how humans exploit redundancy and variability by way of multitiered dynamic processes to enhance the efficiency and effectiveness of human speech communication. Second, advancement of human language technology, especially that in automatic recognition of natural-style human speech is also expected to benefit from comprehensive computational modeling of speech dynamics. The limitations of current speech recognition technology are serious and are well known. A commonly acknowledged and frequently discussed weakness of the statistical model underlying current speech recognition technology is the lack of adequate dynamic modeling schemes to provide correlation structure across the temporal speech observation sequence. Unfortunately, due to a variety of reasons, the majority of current research activities in this area favor only incremental modifications and improvements to the existing HMM-based state-of-the-art. For example, while the dynamic and correlation modeling is known to be an important topic, most of the systems nevertheless employ only an ultra-weak form of speech dynamics; e.g., differential or delta parameters. Strong-form dynamic speech modeling, which is the focus of this monograph, may serve as an ultimate solution to this problem. After the introduction chapter, the main body of this monograph consists of four chapters. They cover various aspects of theory, algorithms, and applications of dynamic speech models, and provide a comprehensive survey of the research work in this area spanning over past 20~years. This monograph is intended as advanced materials of speech and signal processing for graudate-level teaching, for professionals and engineering practioners, as well as for seasoned researchers and engineers specialized in speech processing

Book Introduction to Digital Speech Processing

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.

Book Neural Modeling of Speech Processing and Speech Learning

Download or read book Neural Modeling of Speech Processing and Speech Learning written by Bernd J. Kröger and published by Springer. This book was released on 2019-07-11 with total page 280 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book explores the processes of spoken language production and perception from a neurobiological perspective. After presenting the basics of speech processing and speech acquisition, a neurobiologically-inspired and computer-implemented neural model is described, which simulates the neural processes of speech processing and speech acquisition. This book is an introduction to the field and aimed at students and scientists in neuroscience, computer science, medicine, psychology and linguistics.

Book Speechreading by Humans and Machines

Download or read book Speechreading by Humans and Machines written by David G. Stork and published by Springer Science & Business Media. This book was released on 2013-11-11 with total page 681 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book is one outcome of the NATO Advanced Studies Institute (ASI) Workshop, "Speechreading by Man and Machine," held at the Chateau de Bonas, Castera-Verduzan (near Auch, France) from August 28 to Septem ber 8, 1995 - the first interdisciplinary meeting devoted the subject of speechreading ("lipreading"). The forty-five attendees from twelve countries covered the gamut of speechreading research, from brain scans of humans processing bi-modal stimuli, to psychophysical experiments and illusions, to statistics of comprehension by the normal and deaf communities, to models of human perception, to computer vision and learning algorithms and hardware for automated speechreading machines. The first week focussed on speechreading by humans, the second week by machines, a general organization that is preserved in this volume. After the in evitable difficulties in clarifying language and terminology across disciplines as diverse as human neurophysiology, audiology, psychology, electrical en gineering, mathematics, and computer science, the participants engaged in lively discussion and debate. We think it is fair to say that there was an atmosphere of excitement and optimism for a field that is both fascinating and potentially lucrative. Of the many general results that can be taken from the workshop, two of the key ones are these: • The ways in which humans employ visual image for speech recogni tion are manifold and complex, and depend upon the talker-perceiver pair, severity and age of onset of any hearing loss, whether the topic of conversation is known or unknown, the level of noise, and so forth.

Book Digital Speech Processing

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.

Book Phase based Speech Processing

Download or read book Phase based Speech Processing written by Parham Aarabi and published by World Scientific. This book was released on 2006 with total page 153 pages. Available in PDF, EPUB and Kindle. Book excerpt: This is the first book that takes a detailed look at the importance of phase in the design of speech processing systems. Phase, in comparison with amplitude, is often ignored for speech recognition applications. Thus, this book highlights some of the important ways in which the phase of speech signals can be utilized for sound localization, enhancement, and recognition.This book also discusses the state-of-the-art research in phase-based speech processing, starting from the basics of signal processing and recording, to single microphone speech recognition, the recognition of speech and the processing of speech by humans, as well as the importance of phase in human speech recognition and multi-microphone phase-based speech processing.

Book Speech Analysis Synthesis and Perception

Download or read book Speech Analysis Synthesis and Perception written by James L. Flanagan and published by Springer. This book was released on 1972 with total page 468 pages. Available in PDF, EPUB and Kindle. Book excerpt: 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.

Book Speaker Perception and Recognition  An Integrative Framework for Computational Speech Processing

Download or read book Speaker Perception and Recognition An Integrative Framework for Computational Speech Processing written by Oxana Lapteva and published by kassel university press GmbH. This book was released on 2011 with total page 192 pages. Available in PDF, EPUB and Kindle. Book excerpt:

Book Advances in Non Linear Modeling for Speech Processing

Download or read book Advances in Non Linear Modeling for Speech Processing written by Raghunath S. Holambe and published by Springer Science & Business Media. This book was released on 2012-02-21 with total page 109 pages. Available in PDF, EPUB and Kindle. Book excerpt: Advances in Non-Linear Modeling for Speech Processing includes advanced topics in non-linear estimation and modeling techniques along with their applications to speaker recognition. Non-linear aeroacoustic modeling approach is used to estimate the important fine-structure speech events, which are not revealed by the short time Fourier transform (STFT). This aeroacostic modeling approach provides the impetus for the high resolution Teager energy operator (TEO). This operator is characterized by a time resolution that can track rapid signal energy changes within a glottal cycle. The cepstral features like linear prediction cepstral coefficients (LPCC) and mel frequency cepstral coefficients (MFCC) are computed from the magnitude spectrum of the speech frame and the phase spectra is neglected. To overcome the problem of neglecting the phase spectra, the speech production system can be represented as an amplitude modulation-frequency modulation (AM-FM) model. To demodulate the speech signal, to estimation the amplitude envelope and instantaneous frequency components, the energy separation algorithm (ESA) and the Hilbert transform demodulation (HTD) algorithm are discussed. Different features derived using above non-linear modeling techniques are used to develop a speaker identification system. Finally, it is shown that, the fusion of speech production and speech perception mechanisms can lead to a robust feature set.