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Author : Tim Black
Publisher :
ISBN 13 :
Total Pages : 128 pages
Book Rating : 4.:/5 (466 download)
Download or read book Pitch Determination of Speech Signals Using the Generalized Spectrum written by Tim Black and published by . This book was released on 2000 with total page 128 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author : W. Hess
Publisher : Springer Science & Business Media
ISBN 13 : 3642819265
Total Pages : 713 pages
Book Rating : 4.6/5 (428 download)
Download or read book Pitch Determination of Speech Signals written by W. Hess and published by Springer Science & Business Media. This book was released on 2012-12-06 with total page 713 pages. Available in PDF, EPUB and Kindle. Book excerpt: Pitch (i.e., fundamental frequency FO and fundamental period TO) occupies a key position in the acoustic speech signal. The prosodic information of an utterance is predominantly determined by this parameter. The ear is more sensitive to changes of fundamental frequency than to changes of other speech signal parameters by an order of magnitude. The quality of vocoded speech is essentially influenced by the quality and faultlessness of the pitch measure ment. Hence the importance of this parameter necessitates using good and reliable measurement methods. At first glance the task looks simple: one just has to detect the funda mental frequency or period of a quasi-periodic signal. For a number of reasons, however, the task of pitch determination has to be counted among the most difficult problems in speech analysis. 1) In principle, speech is a nonstationary process; the momentary position of the vocal tract may change abruptly at any time. This leads to drastic variations in the temporal structure of the signal, even between subsequent pitch periods, and assuming a quasi-periodic signal is often far from realistic. 2) Due to the flexibility of the human vocal tract and the wide variety of voices, there exist a multitude of possible temporal structures. Narrow-band formants at low harmonics (especially at the second or third harmonic) are an additional source of difficulty. 3) For an arbitrary speech signal uttered by an unknown speaker, the fundamental frequency can vary over a range of almost four octaves (50 to 800 Hz).
Author : Sean A. Fulop
Publisher : Springer Science & Business Media
ISBN 13 : 3642174787
Total Pages : 214 pages
Book Rating : 4.6/5 (421 download)
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.
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 : W. Bastiaan Kleijn
Publisher : Elsevier Science & Technology
ISBN 13 :
Total Pages : 784 pages
Book Rating : 4.3/5 (91 download)
Download or read book Speech Coding and Synthesis written by W. Bastiaan Kleijn and published by Elsevier Science & Technology. This book was released on 1995 with total page 784 pages. Available in PDF, EPUB and Kindle. Book excerpt: Hardbound. The fields of speech coding and synthesis have developed rapidly over the last decade. Text-to-text speech systems now produce reasonable quality speech, and currently available speech coders can transmit good quality speech at below 10kb/s. This, in combination with the ever-increasing speed of microprocessors and signal processing hardware, has resulted in a large number of practical applications. These applications in turn have stimulated research, and the number of papers published on speech coding and synthesis have proliferated rapidly. Reflecting periodically on such developments have inspired the publication of this book. Topics such as the effect of cross channel errors on coded speech and the determination of a proper pitch contour for synthesized speech are included.Both readers unfamiliar with the fields of speech coding and speech synthesis as well as those already working within the areas, will find the book of interest.
Author : Tianyu Tom Wang
Publisher :
ISBN 13 :
Total Pages : 179 pages
Book Rating : 4.:/5 (746 download)
Download or read book Toward an Interpretive Framework of Two-dimensional Speech-signal Processing written by Tianyu Tom Wang and published by . This book was released on 2011 with total page 179 pages. Available in PDF, EPUB and Kindle. Book excerpt: Traditional representations of speech are derived from short-time segments of the signal and result in time-frequency distributions of energy such as the short-time Fourier transform and spectrogram. Speech-signal models of such representations have had utility in a variety of applications such as speech analysis, recognition, and synthesis. Nonetheless, they do not capture spectral, temporal, and joint spectrotemporal energy fluctuations (or "modulations") present in local time-frequency regions of the time-frequency distribution. Inspired by principles from image processing and evidence from auditory neurophysiological models, a variety of twodimensional (2-D) processing techniques have been explored in the literature as alternative representations of speech; however, speech-based models are lacking in this framework. This thesis develops speech-signal models for a particular 2-D processing approach in which 2-D Fourier transforms are computed on local time-frequency regions of the canonical narrowband or wideband spectrogram; we refer to the resulting transformed space as the Grating Compression Transform (GCT). We argue for a 2-D sinusoidal-series amplitude modulation model of speech content in the spectrogram domain that relates to speech production characteristics such as pitch/noise of the source, pitch dynamics, formant structure and dynamics, and offset/onset content. Narrowband- and wideband-based models are shown to exhibit important distinctions in interpretation and oftentimes "dual" behavior. In the transformed GCT space, the modeling results in a novel taxonomy of signal behavior based on the distribution of formant and onset/offset content in the transformed space via source characteristics. Our formulation provides a speech-specific interpretation of the concept of "modulation" in 2-D processing in contrast to existing approaches that have done so either phenomenologically through qualitative analyses and/or implicitly through data-driven machine learning approaches. One implication of the proposed taxonomy is its potential for interpreting transformations of other time-frequency distributions such as the auditory spectrogram which is generally viewed as being "narrowband"/"wideband" in its low/high-frequency regions. The proposed signal model is evaluated in several ways. First, we perform analysis of synthetic speech signals to characterize its properties and limitations. Next, we develop an algorithm for analysis/synthesis of spectrograms using the model and demonstrate its ability to accurately represent real speech content. As an example application, we further apply the models in cochannel speaker separation, exploiting the GCT's ability to distribute speaker-specific content and often recover overlapping information through demodulation and interpolation in the 2-D GCT space. Specifically, in multi-pitch estimation, we demonstrate the GCT's ability to accurately estimate separate and crossing pitch tracks under certain conditions. Finally, we demonstrate the model's ability to separate mixtures of speech signals using both prior and estimated pitch information. Generalization to other speech-signal processing applications is proposed.
Author : Martin Cooke
Publisher :
ISBN 13 :
Total Pages : 406 pages
Book Rating : 4.3/5 (91 download)
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.
Author : Antonio Napolitano
Publisher : Academic Press
ISBN 13 : 0081027370
Total Pages : 626 pages
Book Rating : 4.0/5 (81 download)
Download or read book Cyclostationary Processes and Time Series written by Antonio Napolitano and published by Academic Press. This book was released on 2019-10-24 with total page 626 pages. Available in PDF, EPUB and Kindle. Book excerpt: Many processes in nature arise from the interaction of periodic phenomena with random phenomena. The results are processes that are not periodic, but whose statistical functions are periodic functions of time. These processes are called cyclostationary and are an appropriate mathematical model for signals encountered in many fields including communications, radar, sonar, telemetry, acoustics, mechanics, econometrics, astronomy, and biology. Cyclostationary Processes and Time Series: Theory, Applications, and Generalizations addresses these issues and includes the following key features. Presents the foundations and developments of the second- and higher-order theory of cyclostationary signals Performs signal analysis using both the classical stochastic process approach and the functional approach for time series Provides applications in signal detection and estimation, filtering, parameter estimation, source location, modulation format classification, and biological signal characterization Includes algorithms for cyclic spectral analysis along with Matlab/Octave code Provides generalizations of the classical cyclostationary model in order to account for relative motion between transmitter and receiver and describe irregular statistical cyclicity in the data
Author : Celia Shahnaz
Publisher :
ISBN 13 :
Total Pages : 0 pages
Book Rating : 4.:/5 (11 download)
Download or read book New Time-frequency Domain Pitch Estimation Methods for Speed Signals Under Low Levels of SNR written by Celia Shahnaz and published by . This book was released on 2009 with total page 0 pages. Available in PDF, EPUB and Kindle. Book excerpt: The major objective of this research is to develop novel pitch estimation methods capable of handling speech signals in practical situations where only noise-corrupted speech observations are available. With this objective in mind, the estimation task is carried out in two different approaches. In the first approach, the noisy speech observations are directly employed to develop two new time-frequency domain pitch estimation methods. These methods are based on extracting a pitch-harmonic and finding the corresponding harmonic number required for pitch estimation. Considering that voiced speech is the output of a vocal tract system driven by a sequence of pulses separated by the pitch period, in the second approach, instead of using the noisy speech directly for pitch estimation, an excitation-like signal (ELS) is first generated from the noisy speech or its noise- reduced version. In the first approach, at first, a harmonic cosine autocorrelation (HCAC) model of clean speech in terms of its pitch-harmonics is introduced. In order to extract a pitch-harmonic, we propose an optimization technique based on least-squares fitting of the autocorrelation function (ACF) of the noisy speech to the HCAC model. By exploiting the extracted pitch-harmonic along with the fast Fourier transform (FFT) based power spectrum of noisy speech, we then deduce a harmonic measure and a harmonic-to-noise-power ratio (HNPR) to determine the desired harmonic number of the extracted pitch-harmonic. In the proposed optimization, an initial estimate of the pitch-harmonic is obtained from the maximum peak of the smoothed FFT power spectrum. In addition to the HCAC model, where the cross-product terms of different harmonics are neglected, we derive a compact yet accurate harmonic sinusoidal autocorrelation (HSAC) model for clean speech signal. The new HSAC model is then used in the least-squares model-fitting optimization technique to extract a pitch-harmonic. In the second approach, first, we develop a pitch estimation method by using an excitation-like signal (ELS) generated from the noisy speech. To this end, a technique is based on the principle of homomorphic deconvolution is proposed for extracting the vocal-tract system (VTS) parameters from the noisy speech, which are utilized to perform an inverse-filtering of the noisy speech to produce a residual signal (RS). In order to reduce the effect of noise on the RS, a noise-compensation scheme is introduced in the autocorrelation domain. The noise-compensated ACF of the RS is then employed to generate a squared Hilbert envelope (SHE) as the ELS of the voiced speech. With a view to further overcome the adverse effect of noise on the ELS, a new symmetric normalized magnitude difference function of the ELS is proposed for eventual pitch estimation. Cepstrum has been widely used in speech signal processing but has limited capability of handling noise. One potential solution could be the introduction of a noise reduction block prior to pitch estimation based on the conventional cepstrum, a framework already available in many practical applications, such as mobile communication and hearing aids. Motivated by the advantages of the existing framework and considering the superiority of our ELS to the speech itself in providing clues for pitch information, we develop a cepstrum-based pitch estimation method by using the ELS obtained from the noise-reduced speech. For this purpose, we propose a noise subtraction scheme in frequency domain, which takes into account the possible cross-correlation between speech and noise and has advantages of noise being updated with time and adjusted at each frame. The enhanced speech thus obtained is utilized to extract the vocal-tract system (VTS) parameters via the homomorphic deconvolution technique. A residual signal (RS) is then produced by inverse-filtering the enhanced speech with the extracted VTS parameters. It is found that, unlike the previous ELS-based method, the squared Hilbert envelope (SHE) computed from the RS of the enhanced speech without noise compensation, is sufficient to represent an ELS. Finally, in order to tackle the undesirable effect of noise of the ELS at a very low SNR and overcome the limitation of the conventional cepstrum in handling different types of noises, a time-frequency domain pseudo cepstrum of the ELS of the enhanced speech, incorporating information of both magnitude and phase spectra of the ELS, is proposed for pitch estimation. (Abstract shortened by UMI.).
Author : Dumitru Baleanu
Publisher : BoD – Books on Demand
ISBN 13 : 9535102125
Total Pages : 314 pages
Book Rating : 4.5/5 (351 download)
Download or read book Wavelet Transforms and Their Recent Applications in Biology and Geoscience written by Dumitru Baleanu and published by BoD – Books on Demand. This book was released on 2012-03-02 with total page 314 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book reports on recent applications in biology and geoscience. Among them we mention the application of wavelet transforms in the treatment of EEG signals, the dimensionality reduction of the gait recognition framework, the biometric identification and verification. The book also contains applications of the wavelet transforms in the analysis of data collected from sport and breast cancer. The denoting procedure is analyzed within wavelet transform and applied on data coming from real world applications. The book ends with two important applications of the wavelet transforms in geoscience.
Author : Baris Bozkurt
Publisher : Presses univ. de Louvain
ISBN 13 : 2874630136
Total Pages : 125 pages
Book Rating : 4.8/5 (746 download)
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.
Author :
Publisher : Springer Science & Business Media
ISBN 13 : 0387776982
Total Pages : 1932 pages
Book Rating : 4.3/5 (877 download)
Download or read book Handbook of Signal Processing in Acoustics written by and published by Springer Science & Business Media. This book was released on 2008 with total page 1932 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author : David Havelock
Publisher : Springer Science & Business Media
ISBN 13 : 038730441X
Total Pages : 1932 pages
Book Rating : 4.3/5 (873 download)
Download or read book Handbook of Signal Processing in Acoustics written by David Havelock and published by Springer Science & Business Media. This book was released on 2008-10-26 with total page 1932 pages. Available in PDF, EPUB and Kindle. Book excerpt: The Handbook of Signal Processing in Acoustics brings together a wide range of perspectives from over 100 authors to reveal the interdisciplinary nature of the subject. It brings the key issues from both acoustics and signal processing into perspective and is a unique resource for experts and practitioners alike to find new ideas and techniques within the diversity of signal processing in acoustics.
Author : Alexey Karpov
Publisher : Springer
ISBN 13 : 3319664298
Total Pages : 845 pages
Book Rating : 4.3/5 (196 download)
Download or read book Speech and Computer written by Alexey Karpov and published by Springer. This book was released on 2017-09-01 with total page 845 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book constitutes the proceedings of the 19th International Conference on Speech and Computer, SPECOM 2017, held in Hatfield, UK, in September 2017. The 80 papers presented in this volume were carefully reviewed and selected from 150 submissions. The papers present current research in the area of computer speech processing (recognition, synthesis, understanding etc.) and related domains (including signal processing, language and text processing, computational paralinguistics, multi-modal speech processing, human-computer interaction).
Author : Curtis Roads
Publisher : Routledge
ISBN 13 : 1134379773
Total Pages : 501 pages
Book Rating : 4.1/5 (343 download)
Download or read book Musical Signal Processing written by Curtis Roads and published by Routledge. This book was released on 2013-12-19 with total page 501 pages. Available in PDF, EPUB and Kindle. Book excerpt: Compiled by an international array of musical and technical specialists, this book deals with some of the most important topics in modern musical signal processing. Beginning with basic concepts, and leading to advanced applications, it covers such essential areas as sound synthesis (including detailed studies of physical modelling and granular synthesis) ,control signal synthesis, sound transformation (including convolution), analysis/resynthesis (phase vocodor, wavelets, analysis by chaotic functions), object-oriented and artificial intelligence representations, musical interfaces and the integration of signal processing techniques in concert performance.
Author :
Publisher :
ISBN 13 :
Total Pages : 766 pages
Book Rating : 4.:/5 (318 download)
Download or read book ICASSP 82 written by and published by . This book was released on 1982 with total page 766 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author : Erik Larsen
Publisher : John Wiley & Sons
ISBN 13 : 0470858656
Total Pages : 312 pages
Book Rating : 4.4/5 (78 download)
Download or read book Audio Bandwidth Extension written by Erik Larsen and published by John Wiley & Sons. This book was released on 2005-04-08 with total page 312 pages. Available in PDF, EPUB and Kindle. Book excerpt: Bandwidth extension (BWE) refers to various methods that increase either the perceived or real frequency spectrum (bandwidth) of audio signals. Such frequency extension is desirable if at some point the frequency content of the audio signal has been reduced, as can happen for example during recording, transmission or reproduction. This volume, significant in dealing exclusively with BWE, discusses applications to music and speech and places particular emphasis on signal processing techniques. Presents an all-encompassing approach to BWE by covering theory, applications and algorithms Reviews important concepts in psychoacoustics, signal processing and loudspeaker theory Develops the theory and implementation of BWE applied to low-frequency sound reproduction, perceptually coded audio, speech and noise abatement Includes a BWE patent overview Audio Bandwidth Extension pulls together recent developments in to a single volume and presents a coherent framework to the reader. Such an approach will have instant appeal to engineers, specialists, researchers and postgraduate students in the fields of audio, signal processing and speech.
