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Phase Spectrum Based Speech Processing And Spectral Energy Estimation For Robust Speech Recognition
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Book Synopsis Phase Spectrum Based Speech Processing and Spectral Energy Estimation for Robust Speech Recognition by : Anthony Stark
Download or read book Phase Spectrum Based Speech Processing and Spectral Energy Estimation for Robust Speech Recognition written by Anthony Stark and published by . This book was released on 2011 with total page 494 pages. Available in PDF, EPUB and Kindle. Book excerpt: Abstract : Speech is the dominant mode of communication between humans; simple to learn, easy to use and integral for modern life. Given the importance of speech, development of a human-machine speech interface has been greatly anticipated. This challenging task is encapsulated in the digital speech processing research field. In this dissertation, two specific areas of research are considered: 1) the use of short-time Fourier spectral phase in digital speech processing and 2) use of the minimum mean square error spectral energy estimator for environment-robust automatic speech recognition. In speech processing and modelling, the short-time Fourier spectral phase has been considered of minor importance. This is because classic psychoacoustic experiments have shown speech intelligibility to be closely related to short-time Fourier spectral magnitude. Given this result, it is unsurprising that the majority of speech processing literature has involved exploitation of the short-time magnitude spectrum. Despite this, recent studies have shown useful information can be extracted from the spectral phase of speech. As a result, it is now known that spectral phase possesses much of the same intelligibility information as spectral magnitude. It is this avenue of research that is explored in greater detail within this dissertation. In particular, we investigate two phase derived quantities {u2013} the short-time instantaneous frequency spectrum and the short-time group delay spectrum. The properties of both spectra are investigated mathematically and empirically, identifying the relationship between known speech features and the underlying phase spectrum. We continue the investigation by examining two related quantities {u2013} the instantaneous frequency deviation and the group delay deviation. As a result of this research, two novel phase-based spectral representations are proposed, both of which show a high degree information applicable to speech processing.
Book Synopsis Phase-based Speech Processing by : Parham Aarabi
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 Synopsis New Spectral Methods for Analysis of Source/filter Characteristics of Speech Signals by : Baris Bozkurt
Download or read book New Spectral Methods for Analysis of Source/filter Characteristics of Speech Signals written by Baris Bozkurt and published by Presses univ. de Louvain. This book was released on 2006 with total page 125 pages. Available in PDF, EPUB and Kindle. Book excerpt: This study proposes a new spectral representation called the Zeros of Z-Transform (ZZT), which is an all-zero representation of the z-transform of the signal. In addition, new chirp group delay processing techniques are developed for analysis of resonances of a signal. The combination of the ZZT representation with the chirp group delay processing algorithms provides a useful domain to study resonance characteristics of source and filter components of speech. Using the two representations, effective algorithms are developed for: source-tract decomposition of speech, glottal flow parameter estimation, formant tracking and feature extraction for speech recognition. The ZZT representation is mainly important for theoretical studies. Studying the ZZT of a signal is essential to be able to develop effective chirp group delay processing methods. Therefore, first the ZZT representation of the source-filter model of speech is studied for providing a theoretical background. We confirm through ZZT representation that anti-causality of the glottal flow signal introduces mixed-phase characteristics in speech signals. The ZZT of windowed speech signals is also studied since windowing cannot be avoided in practical signal processing algorithms and the effect of windowing on ZZT representation is drastic. We show that separate patterns exist in ZZT representations of windowed speech signals for the glottal flow and the vocal tract contributions. A decomposition method for source-tract separation is developed based on these patterns in ZZT. We define chirp group delay as group delay calculated on a circle other than the unit circle in z-plane. The need to compute group delay on a circle other than the unit circle comes from the fact that group delay spectra are often very noisy and cannot be easily processed for formant tracking purposes (the reasons are explained through ZZT representation). In this thesis, we propose methods to avoid such problems by modifying the ZZT of a signal and further computing the chirp group delay spectrum. New algorithms based on processing of the chirp group delay spectrum are developed for formant tracking and feature estimation for speech recognition. The proposed algorithms are compared to state-of-the-art techniques. Equivalent or higher efficiency is obtained for all proposed algorithms. The theoretical parts of the thesis further discuss a mixed-phase model for speech and phase processing problems in detail. Index Terms—spectral representation, source-filter separation, glottal flow estimation, formant tracking, zeros of z-transform, group delay processing, phase processing.
Book Synopsis Single Channel Phase-Aware Signal Processing in Speech Communication by : Pejman Mowlaee
Download or read book Single Channel Phase-Aware Signal Processing in Speech Communication written by Pejman Mowlaee and published by John Wiley & Sons. This book was released on 2016-12-27 with total page 253 pages. Available in PDF, EPUB and Kindle. Book excerpt: An overview on the challenging new topic of phase-aware signal processing Speech communication technology is a key factor in human-machine interaction, digital hearing aids, mobile telephony, and automatic speech/speaker recognition. With the proliferation of these applications, there is a growing requirement for advanced methodologies that can push the limits of the conventional solutions relying on processing the signal magnitude spectrum. Single-Channel Phase-Aware Signal Processing in Speech Communication provides a comprehensive guide to phase signal processing and reviews the history of phase importance in the literature, basic problems in phase processing, fundamentals of phase estimation together with several applications to demonstrate the usefulness of phase processing. Key features: Analysis of recent advances demonstrating the positive impact of phase-based processing in pushing the limits of conventional methods. Offers unique coverage of the historical context, fundamentals of phase processing and provides several examples in speech communication. Provides a detailed review of many references and discusses the existing signal processing techniques required to deal with phase information in different applications involved with speech. The book supplies various examples and MATLAB® implementations delivered within the PhaseLab toolbox. Single-Channel Phase-Aware Signal Processing in Speech Communication is a valuable single-source for students, non-expert DSP engineers, academics and graduate students.
Book Synopsis Speech Spectrum Analysis by : Sean A. Fulop
Download or read book Speech Spectrum Analysis written by Sean A. Fulop and published by Springer Science & Business Media. This book was released on 2011-05-26 with total page 214 pages. Available in PDF, EPUB and Kindle. Book excerpt: The accurate determination of the speech spectrum, particularly for short frames, is commonly pursued in diverse areas including speech processing, recognition, and acoustic phonetics. With this book the author makes the subject of spectrum analysis understandable to a wide audience, including those with a solid background in general signal processing and those without such background. In keeping with these goals, this is not a book that replaces or attempts to cover the material found in a general signal processing textbook. Some essential signal processing concepts are presented in the first chapter, but even there the concepts are presented in a generally understandable fashion as far as is possible. Throughout the book, the focus is on applications to speech analysis; mathematical theory is provided for completeness, but these developments are set off in boxes for the benefit of those readers with sufficient background. Other readers may proceed through the main text, where the key results and applications will be presented in general heuristic terms, and illustrated with software routines and practical "show-and-tell" discussions of the results. At some points, the book refers to and uses the implementations in the Praat speech analysis software package, which has the advantages that it is used by many scientists around the world, and it is free and open source software. At other points, special software routines have been developed and made available to complement the book, and these are provided in the Matlab programming language. If the reader has the basic Matlab package, he/she will be able to immediately implement the programs in that platform---no extra "toolboxes" are required.
Book Synopsis Speech Processing in Mobile Environments by : K. Sreenivasa Rao
Download or read book Speech Processing in Mobile Environments written by K. Sreenivasa Rao and published by Springer Science & Business Media. This book was released on 2014-01-28 with total page 129 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book focuses on speech processing in the presence of low-bit rate coding and varying background environments. The methods presented in the book exploit the speech events which are robust in noisy environments. Accurate estimation of these crucial events will be useful for carrying out various speech tasks such as speech recognition, speaker recognition and speech rate modification in mobile environments. The authors provide insights into designing and developing robust methods to process the speech in mobile environments. Covering temporal and spectral enhancement methods to minimize the effect of noise and examining methods and models on speech and speaker recognition applications in mobile environments.
Book Synopsis Perceptually Inspired Signal-processing Strategies for Robust Speech Recognition in Reverberant Environments by : Brian E.D. Kingsbury
Download or read book Perceptually Inspired Signal-processing Strategies for Robust Speech Recognition in Reverberant Environments written by Brian E.D. Kingsbury and published by . This book was released on 1998 with total page 438 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Book Synopsis Advanced Signal Processing and Digital Noise Reduction by : Saeed V. Vaseghi
Download or read book Advanced Signal Processing and Digital Noise Reduction written by Saeed V. Vaseghi and published by Vieweg+Teubner Verlag. This book was released on 1996-05 with total page 424 pages. Available in PDF, EPUB and Kindle. Book excerpt: Bayesian Estimation and classification. Hidden markov models. Wiener filters. Kalman and adaptive least squared error filters.
Book Synopsis Spectrum Estimation and System Identification by : S.Unnikrishna Pillai
Download or read book Spectrum Estimation and System Identification written by S.Unnikrishna Pillai and published by Springer Science & Business Media. This book was released on 2012-12-06 with total page 337 pages. Available in PDF, EPUB and Kindle. Book excerpt: Spectrum estimation refers to analyzing the distribution of power or en ergy with frequency of the given signal, and system identification refers to ways of characterizing the mechanism or system behind the observed sig nal/data. Such an identification allows one to predict the system outputs, and as a result this has considerable impact in several areas such as speech processing, pattern recognition, target identification, seismology, and signal processing. A new outlook to spectrum estimation and system identification is pre sented here by making use of the powerful concepts of positive functions and bounded functions. An indispensable tool in classical network analysis and synthesis problems, positive functions and bounded functions are well and their intimate one-to-one connection with power spectra understood, makes it possible to study many of the signal processing problems from a new viewpoint. Positive functions have been used to study interpolation problems in the past, and although the spectrum extension problem falls within this scope, surprisingly the system identification problem can also be analyzed in this context in an interesting manner. One useful result in this connection is regarding rational and stable approximation of nonrational transfer functions both in the single-channel case and the multichannel case. Such an approximation has important applications in distributed system theory, simulation of systems governed by partial differential equations, and analysis of differential equations with delays. This book is intended as an introductory graduate level textbook and as a reference book for engineers and researchers.
Book Synopsis Discrete-Time Speech Signal Processing by : Thomas F. Quatieri
Download or read book Discrete-Time Speech Signal Processing written by Thomas F. Quatieri and published by Pearson Education. This book was released on 2008-11-10 with total page 1226 pages. Available in PDF, EPUB and Kindle. Book excerpt: Essential principles, practical examples, current applications, and leading-edge research. In this book, Thomas F. Quatieri presents the field's most intensive, up-to-date tutorial and reference on discrete-time speech signal processing. Building on his MIT graduate course, he introduces key principles, essential applications, and state-of-the-art research, and he identifies limitations that point the way to new research opportunities. Quatieri provides an excellent balance of theory and application, beginning with a complete framework for understanding discrete-time speech signal processing. Along the way, he presents important advances never before covered in a speech signal processing text book, including sinusoidal speech processing, advanced time-frequency analysis, and nonlinear aeroacoustic speech production modeling. Coverage includes: Speech production and speech perception: a dual view Crucial distinctions between stochastic and deterministic problems Pole-zero speech models Homomorphic signal processing Short-time Fourier transform analysis/synthesis Filter-bank and wavelet analysis/synthesis Nonlinear measurement and modeling techniques The book's in-depth applications coverage includes speech coding, enhancement, and modification; speaker recognition; noise reduction; signal restoration; dynamic range compression, and more. Principles of Discrete-Time Speech Processing also contains an exceptionally complete series of examples and Matlab exercises, all carefully integrated into the book's coverage of theory and applications.
Book Synopsis Speech Signal Processing Based on Deep Learning in Complex Acoustic Environments by : Xiao-Lei Zhang
Download or read book Speech Signal Processing Based on Deep Learning in Complex Acoustic Environments written by Xiao-Lei Zhang and published by Elsevier. This book was released on 2024-09-04 with total page 282 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech Signal Processing Based on Deep Learning in Complex Acoustic Environments provides a detailed discussion of deep learning-based robust speech processing and its applications. The book begins by looking at the basics of deep learning and common deep network models, followed by front-end algorithms for deep learning-based speech denoising, speech detection, single-channel speech enhancement multi-channel speech enhancement, multi-speaker speech separation, and the applications of deep learning-based speech denoising in speaker verification and speech recognition. Provides a comprehensive introduction to the development of deep learning-based robust speech processing Covers speech detection, speech enhancement, dereverberation, multi-speaker speech separation, robust speaker verification, and robust speech recognition Focuses on a historical overview and then covers methods that demonstrate outstanding performance in practical applications
Book Synopsis Intelligent Speech Signal Processing by : Nilanjan Dey
Download or read book Intelligent Speech Signal Processing written by Nilanjan Dey and published by Academic Press. This book was released on 2019-06-15 with total page 210 pages. Available in PDF, EPUB and Kindle. Book excerpt: Intelligent Speech Signal Processing investigates the utilization of speech analytics across several systems and real-world activities, including sharing data analytics related information, creating collaboration networks between several participants, and implementing video-conferencing in different application areas. It provides a forum for readers to discover the characteristics of intelligent speech signal processing systems across different domains. Chapters focus on the latest applications of speech data analysis and management tools across different recording systems. The book emphasizes the multi-disciplinary nature of the field, presenting different applications and challenges with extensive studies on the design, implementation, development, and management of intelligent systems, neural networks, and related machine learning techniques for speech signal processing. Highlights different data analytics techniques in speech signal processing, including machine learning, and data mining Illustrates different applications and challenges across the design, implementation, and management of intelligent systems and neural networks techniques for speech signal processing Includes coverage of biomodal speech recognition, voice activity detection, spoken language and speech disorder identification, automatic speech to speech summarization, and convolutional neural networks
Book Synopsis Phase-based Speech Processing by : Guangji Shi
Download or read book Phase-based Speech Processing written by Guangji Shi and published by . This book was released on 2006 with total page 282 pages. Available in PDF, EPUB and Kindle. Book excerpt: The performance of automatic speech recognition (ASR) systems degrades significantly in adverse environments due to ambient noise and reverberation. This problem becomes even greater in hands-free speech applications, where the microphones can be placed far away from the speaker of interest. Environmental robustness has become a major barrier that prevents ASR from a wide range of applications such as voice recognition in a car and voice controlled hand-held devices. In this research, the importance of phase in robust speech recognition is explored. First, the effect of phase uncertainty on the recognition accuracy of human listeners is investigated. The goal is to get a quantitative measure on the importance of phase. The results show that the importance of phase varies with SNR (signal-to-noise ratio). At low SNR conditions, phase can have a significant impact on speech recognition accuracy. Next, motivated by the importance of phase in multi-microphone signal processing, a phase-based dual-microphone noise masking approach is proposed for speech enhancement. By utilizing the time delay of the speech source of interest to the two microphones and the actual phases of the signals recorded by both microphones, the algorithm filters the noise signal in the short-time Fourier transform domain. By doing so, the noise components are distorted beyond recognition and the speech recognition accuracy is improved. The effectiveness of this approach is demonstrated through performance comparison with alternative techniques. Lastly, an automatic parameter estimation technique is developed to further optimize its performance. The parameter of the phase-based dual-microphone filter is adjusted in run-time automatically by performing likelihood calculations of the enhanced speech features using a prior speech model. Speech recognition tests show that this adaptive approach not only achieves better recognition accuracy, but also improves the filter's robustness when time delay estimates are inaccurate.
Book Synopsis Robust Speech Recognition of Uncertain or Missing Data by : Dorothea Kolossa
Download or read book Robust Speech Recognition of Uncertain or Missing Data written by Dorothea Kolossa and published by Springer Science & Business Media. This book was released on 2011-07-14 with total page 387 pages. Available in PDF, EPUB and Kindle. Book excerpt: Automatic speech recognition suffers from a lack of robustness with respect to noise, reverberation and interfering speech. The growing field of speech recognition in the presence of missing or uncertain input data seeks to ameliorate those problems by using not only a preprocessed speech signal but also an estimate of its reliability to selectively focus on those segments and features that are most reliable for recognition. This book presents the state of the art in recognition in the presence of uncertainty, offering examples that utilize uncertainty information for noise robustness, reverberation robustness, simultaneous recognition of multiple speech signals, and audiovisual speech recognition. The book is appropriate for scientists and researchers in the field of speech recognition who will find an overview of the state of the art in robust speech recognition, professionals working in speech recognition who will find strategies for improving recognition results in various conditions of mismatch, and lecturers of advanced courses on speech processing or speech recognition who will find a reference and a comprehensive introduction to the field. The book assumes an understanding of the fundamentals of speech recognition using Hidden Markov Models.
Book Synopsis Recent Advances in Robust Speech Recognition Technology by : Javier Ramirez
Download or read book Recent Advances in Robust Speech Recognition Technology written by Javier Ramirez and published by Bentham Science. This book was released on 2011 with total page 223 pages. Available in PDF, EPUB and Kindle. Book excerpt: "This E-book is a collection of articles that describe advances in speech recognition technology. Robustness in speech recognition refers to the need to maintain high speech recognition accuracy even when the quality of the input speech is degraded, or whe"
Book Synopsis Springer Handbook of Speech Processing by : Jacob Benesty
Download or read book Springer Handbook of Speech Processing written by Jacob Benesty and published by Springer Science & Business Media. This book was released on 2007-11-28 with total page 1170 pages. Available in PDF, EPUB and Kindle. Book excerpt: This handbook plays a fundamental role in sustainable progress in speech research and development. With an accessible format and with accompanying DVD-Rom, it targets three categories of readers: graduate students, professors and active researchers in academia, and engineers in industry who need to understand or implement some specific algorithms for their speech-related products. It is a superb source of application-oriented, authoritative and comprehensive information about these technologies, this work combines the established knowledge derived from research in such fast evolving disciplines as Signal Processing and Communications, Acoustics, Computer Science and Linguistics.
Book Synopsis Visual Representations of Speech Signals by : Martin Cooke
Download or read book Visual Representations of Speech Signals written by Martin Cooke and published by . This book was released on 1993-04-14 with total page 406 pages. Available in PDF, EPUB and Kindle. Book excerpt: Presents a wide range of graphical representations of some speech signals and allows current speech analysis techniques to be assessed and directly compared. Describes time-frequency representations, auditory modeling, neural networks, pitch and multi-channel analysis. The study of over 40 different analyses of speech is represented in myriad images found throughout.
