Author: Dan Jurafsky
Publisher: Pearson Education India
ISBN: 9788131716724
Category :
Languages : en
Pages : 912
Book Description
Speech & Language Processing
Author: Dan Jurafsky
Publisher: Pearson Education India
ISBN: 9788131716724
Category :
Languages : en
Pages : 912
Book Description
Publisher: Pearson Education India
ISBN: 9788131716724
Category :
Languages : en
Pages : 912
Book Description
Digital Processing of Speech Signals
Author: Lawrence R. Rabiner
Publisher: Pearson Education India
ISBN: 9788131705131
Category : Digital electronics
Languages : en
Pages : 532
Book Description
Publisher: Pearson Education India
ISBN: 9788131705131
Category : Digital electronics
Languages : en
Pages : 532
Book Description
Introduction to Digital Speech Processing
Author: Lawrence R. Rabiner
Publisher: Now Publishers Inc
ISBN: 1601980701
Category : Computers
Languages : en
Pages : 212
Book Description
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.
Publisher: Now Publishers Inc
ISBN: 1601980701
Category : Computers
Languages : en
Pages : 212
Book Description
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.
Noise Reduction in Speech Processing
Author: Jacob Benesty
Publisher: Springer Science & Business Media
ISBN: 364200296X
Category : Technology & Engineering
Languages : en
Pages : 236
Book Description
Noise is everywhere and in most applications that are related to audio and speech, such as human-machine interfaces, hands-free communications, voice over IP (VoIP), hearing aids, teleconferencing/telepresence/telecollaboration systems, and so many others, the signal of interest (usually speech) that is picked up by a microphone is generally contaminated by noise. As a result, the microphone signal has to be cleaned up with digital signal processing tools before it is stored, analyzed, transmitted, or played out. This cleaning process is often called noise reduction and this topic has attracted a considerable amount of research and engineering attention for several decades. One of the objectives of this book is to present in a common framework an overview of the state of the art of noise reduction algorithms in the single-channel (one microphone) case. The focus is on the most useful approaches, i.e., filtering techniques (in different domains) and spectral enhancement methods. The other objective of Noise Reduction in Speech Processing is to derive all these well-known techniques in a rigorous way and prove many fundamental and intuitive results often taken for granted. This book is especially written for graduate students and research engineers who work on noise reduction for speech and audio applications and want to understand the subtle mechanisms behind each approach. Many new and interesting concepts are presented in this text that we hope the readers will find useful and inspiring.
Publisher: Springer Science & Business Media
ISBN: 364200296X
Category : Technology & Engineering
Languages : en
Pages : 236
Book Description
Noise is everywhere and in most applications that are related to audio and speech, such as human-machine interfaces, hands-free communications, voice over IP (VoIP), hearing aids, teleconferencing/telepresence/telecollaboration systems, and so many others, the signal of interest (usually speech) that is picked up by a microphone is generally contaminated by noise. As a result, the microphone signal has to be cleaned up with digital signal processing tools before it is stored, analyzed, transmitted, or played out. This cleaning process is often called noise reduction and this topic has attracted a considerable amount of research and engineering attention for several decades. One of the objectives of this book is to present in a common framework an overview of the state of the art of noise reduction algorithms in the single-channel (one microphone) case. The focus is on the most useful approaches, i.e., filtering techniques (in different domains) and spectral enhancement methods. The other objective of Noise Reduction in Speech Processing is to derive all these well-known techniques in a rigorous way and prove many fundamental and intuitive results often taken for granted. This book is especially written for graduate students and research engineers who work on noise reduction for speech and audio applications and want to understand the subtle mechanisms behind each approach. Many new and interesting concepts are presented in this text that we hope the readers will find useful and inspiring.
Intelligent Speech Signal Processing
Author: Nilanjan Dey
Publisher: Academic Press
ISBN: 0128181303
Category : Technology & Engineering
Languages : en
Pages : 210
Book Description
Intelligent Speech Signal Processing investigates the utilization of speech analytics across several systems and real-world activities, including sharing data analytics, creating collaboration networks between several participants, and implementing video-conferencing in different application areas. Chapters focus on the latest applications of speech data analysis and management tools across different recording systems. The book emphasizes the multidisciplinary nature of the field, presenting different applications and challenges with extensive studies on the design, development and management of intelligent systems, neural networks and related machine learning techniques for speech signal processing.
Publisher: Academic Press
ISBN: 0128181303
Category : Technology & Engineering
Languages : en
Pages : 210
Book Description
Intelligent Speech Signal Processing investigates the utilization of speech analytics across several systems and real-world activities, including sharing data analytics, creating collaboration networks between several participants, and implementing video-conferencing in different application areas. Chapters focus on the latest applications of speech data analysis and management tools across different recording systems. The book emphasizes the multidisciplinary nature of the field, presenting different applications and challenges with extensive studies on the design, development and management of intelligent systems, neural networks and related machine learning techniques for speech signal processing.
Multilingual Speech Processing
Author: Tanja Schultz
Publisher: Elsevier
ISBN: 0080457622
Category : Computers
Languages : en
Pages : 540
Book Description
Tanja Schultz and Katrin Kirchhoff have compiled a comprehensive overview of speech processing from a multilingual perspective. By taking this all-inclusive approach to speech processing, the editors have included theories, algorithms, and techniques that are required to support spoken input and output in a large variety of languages. Multilingual Speech Processing presents a comprehensive introduction to research problems and solutions, both from a theoretical as well as a practical perspective, and highlights technology that incorporates the increasing necessity for multilingual applications in our global community. Current challenges of speech processing and the feasibility of sharing data and system components across different languages guide contributors in their discussions of trends, prognoses and open research issues. This includes automatic speech recognition and speech synthesis, but also speech-to-speech translation, dialog systems, automatic language identification, and handling non-native speech. The book is complemented by an overview of multilingual resources, important research trends, and actual speech processing systems that are being deployed in multilingual human-human and human-machine interfaces. Researchers and developers in industry and academia with different backgrounds but a common interest in multilingual speech processing will find an excellent overview of research problems and solutions detailed from theoretical and practical perspectives. - State-of-the-art research with a global perspective by authors from the USA, Asia, Europe, and South Africa - The only comprehensive introduction to multilingual speech processing currently available - Detailed presentation of technological advances integral to security, financial, cellular and commercial applications
Publisher: Elsevier
ISBN: 0080457622
Category : Computers
Languages : en
Pages : 540
Book Description
Tanja Schultz and Katrin Kirchhoff have compiled a comprehensive overview of speech processing from a multilingual perspective. By taking this all-inclusive approach to speech processing, the editors have included theories, algorithms, and techniques that are required to support spoken input and output in a large variety of languages. Multilingual Speech Processing presents a comprehensive introduction to research problems and solutions, both from a theoretical as well as a practical perspective, and highlights technology that incorporates the increasing necessity for multilingual applications in our global community. Current challenges of speech processing and the feasibility of sharing data and system components across different languages guide contributors in their discussions of trends, prognoses and open research issues. This includes automatic speech recognition and speech synthesis, but also speech-to-speech translation, dialog systems, automatic language identification, and handling non-native speech. The book is complemented by an overview of multilingual resources, important research trends, and actual speech processing systems that are being deployed in multilingual human-human and human-machine interfaces. Researchers and developers in industry and academia with different backgrounds but a common interest in multilingual speech processing will find an excellent overview of research problems and solutions detailed from theoretical and practical perspectives. - State-of-the-art research with a global perspective by authors from the USA, Asia, Europe, and South Africa - The only comprehensive introduction to multilingual speech processing currently available - Detailed presentation of technological advances integral to security, financial, cellular and commercial applications
Speech and Audio Signal Processing
Author: Ben Gold
Publisher: John Wiley & Sons
ISBN: 0470195363
Category : Technology & Engineering
Languages : en
Pages : 684
Book Description
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).
Publisher: John Wiley & Sons
ISBN: 0470195363
Category : Technology & Engineering
Languages : en
Pages : 684
Book Description
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).
Discrete-Time Speech Signal Processing
Author: Thomas F. Quatieri
Publisher: Pearson Education
ISBN: 0132441233
Category : Technology & Engineering
Languages : en
Pages : 1226
Book Description
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.
Publisher: Pearson Education
ISBN: 0132441233
Category : Technology & Engineering
Languages : en
Pages : 1226
Book Description
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.
Springer Handbook of Speech Processing
Author: Jacob Benesty
Publisher: Springer Science & Business Media
ISBN: 3540491252
Category : Technology & Engineering
Languages : en
Pages : 1170
Book Description
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.
Publisher: Springer Science & Business Media
ISBN: 3540491252
Category : Technology & Engineering
Languages : en
Pages : 1170
Book Description
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.
Speech Processing and Synthesis Toolboxes
Author: D. G. Childers
Publisher: John Wiley & Sons
ISBN:
Category : Computers
Languages : en
Pages : 504
Book Description
Strike a balance between theory and practice! With this text, you'll, find a balance between theory and practice that allows you to build your understanding of the basic concepts, assumptions, and limitations of the theory of speech analysis and synthesis. The methods for data analysis as well as the theoretical background are provided to help you comprehend the analysis results. And you'll be able to study the features and properties of speech as a signal without having to record data and write software to analyze the data. The text includes two CDs that contain stand-alone and MATLAB software and speech and electroglottographic data. The CDs illustrate the effects that speech models and speech analysis procedures have on the quality of synthesized speech. An extensive speech database provides numerous speech files and other data. Examples included in each chapter demonstrate how to use the software. The CDs allow you to: * Calculate the parameters of linear prediction speech models. * Examine procedures for converting the speech of one speaker to sound like that of another speaker (i.e., voice conversion). * Analyze and alter the temporal structure of the speech signal. This allows you to automatically parse speech into various features, such as voiced segments, unvoiced segments, nasal and non-nasal segments, fricatives, stops, and more. * Create speech with a "high speaking rate" or generate speech with a "slow speaking rate." * Adjust the parameters of the vocal fold model to change the vocal fold tension, length, thickness, mass, etc., in order to observe the effects of these parameters on the vibratory motion of the vocal folds.
Publisher: John Wiley & Sons
ISBN:
Category : Computers
Languages : en
Pages : 504
Book Description
Strike a balance between theory and practice! With this text, you'll, find a balance between theory and practice that allows you to build your understanding of the basic concepts, assumptions, and limitations of the theory of speech analysis and synthesis. The methods for data analysis as well as the theoretical background are provided to help you comprehend the analysis results. And you'll be able to study the features and properties of speech as a signal without having to record data and write software to analyze the data. The text includes two CDs that contain stand-alone and MATLAB software and speech and electroglottographic data. The CDs illustrate the effects that speech models and speech analysis procedures have on the quality of synthesized speech. An extensive speech database provides numerous speech files and other data. Examples included in each chapter demonstrate how to use the software. The CDs allow you to: * Calculate the parameters of linear prediction speech models. * Examine procedures for converting the speech of one speaker to sound like that of another speaker (i.e., voice conversion). * Analyze and alter the temporal structure of the speech signal. This allows you to automatically parse speech into various features, such as voiced segments, unvoiced segments, nasal and non-nasal segments, fricatives, stops, and more. * Create speech with a "high speaking rate" or generate speech with a "slow speaking rate." * Adjust the parameters of the vocal fold model to change the vocal fold tension, length, thickness, mass, etc., in order to observe the effects of these parameters on the vibratory motion of the vocal folds.