Author: F. W. ZakiPublisher:
ISBN:
Category :
Languages : en
Pages :
Book Description
A Study of Adaptive Predictive Coding for Speech Signals
Author: F. W. ZakiSignal Prediction with Input Identification
Author: Jer-Nan JuangPublisher:
ISBN:
Category : Data compression (Computer science)
Languages : en
Pages : 36
Book Description
A novel coding technique is presented for signal prediction with applications including speech coding, system identification, and estimation of input excitation. The approach is based on the blind equalization method for speech signal processing in conjunction with the geometric subspace projection theory to formulate the basic prediction equation. The speech-coding problem is often divided into two parts, a linear prediction model and excitation input. The parameter coefficients of the linear predictor and the input excitation are solved simultaneously and recursively by a conventional recursive least-squares algorithm. The excitation input is computed by coding all possible outcomes into a binary notebook. The coefficients of the linear predictor and excitation, and the index of the codebook can then be used to represent the signal. In addition, a variable-frame concept is proposed to block the same excitation signal in sequence in order to reduce the storage size and increase the transmission rate. The results of this work can be easily extended to the problem of disturbance identification. The basic principles are outlined in this report and differences from other existing methods are discussed. Simulations are included to demonstrate the proposed method.
An Adaptive Approach to a 2.4 Kb/s LPC Speech Coding System
Author: R. YarlagaddaPublisher:
ISBN:
Category :
Languages : en
Pages : 255
Book Description
The goal of this research was to investigate (a) Adaptive estimation methods for noise suppression and performance enhancement of Narrowband Coding Systems for speech signals and (b) Autoregressive spectral estimation in noisy signals for speech analysis applications. Various prefiltering techniques for improving linear predictive coding systems were investigated. Filter coefficients were varied to optimize each filter technique to remove noise from the speech signal. A new prefilter consisting of an adaptive digital predictor (ADP) with pitch period delay was developed and evaluated. The one-dimension filter approach was expanded upon to use a two-dimensional approach to suppress noise in the short time fourier transform domain. The two dimensional approach was found to have significant potential. Fast algorithms for efficient solution of the linear estimation problem and a new recursive linear estimator suitable for rapid estimation of a signal in noise were developed. Keywords include: Speech analysis; Pitch estimation; Noise suppression; Signal estimation; Adaptive digital filter; and Speech enhancement.
adaptive predictive coding for speech signals
Author: kishkLinear Predictive Coding and the Internet Protocol
Author: Robert M. GrayPublisher: Now Publishers Inc
ISBN: 1601983484
Category : Computers
Languages : en
Pages : 181
Book Description
In December 1974 the first realtime conversation on the ARPAnet took place between Culler- Harrison Incorporated in Goleta, California, and MIT Lincoln Laboratory in Lexington, Massachusetts. This was the first successful application of realtime digital speech communication over a packet network and an early milestone in the explosion of realtime signal processing of speech, audio, images, and video that we all take for granted today. It could be considered as the first voice over Internet Protocol (VoIP), except that the Internet Protocol (IP) had not yet been established. In fact, the interest in realtime signal processing had an indirect, but major, impact on the development of IP. This is the story of the development of linear predictive coded (LPC) speech and how it came to be used in the first successful packet speech experiments. Several related stories are recounted as well. The history is preceded by a tutorial on linear prediction methods which incorporates a variety of views to provide context for the stories. This part is a technical survey of the fundamental ideas of linear prediction that are important for speech processing, but the development departs from traditional treatments and takes advantage of several shortcuts, simplifications, and unifications that come with years of hindsight. In particular, some of the key results are proved using short and simple techniques that are not as well known as they should be, and it also addresses some of the common assumptions made when modeling random signals. The reader interested only in the history and already familiar with or uninterested in the technical details of linear prediction and speech may skip Part I entirely.
A Study of Adaptive Algorithms for Modeling and Predictive Error Coding of Real Speech Signals
Author: Takeo KanaiPublisher:
ISBN:
Category : Speech processing systems
Languages : en
Pages : 334
Book Description
Algorithms and Software for Predictive and Perceptual Modeling of Speech
Author: Venkatraman AttiPublisher: Morgan & Claypool Publishers
ISBN: 160845388X
Category : Technology & Engineering
Languages : en
Pages : 124
Book Description
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
Practical Approaches to Speech Coding
Author: Panos E. PapamichalisPublisher: Prentice Hall
ISBN:
Category : Computers
Languages : en
Pages : 344
Book Description
Principles of Speech Coding
Author: Tokunbo OgunfunmiPublisher: CRC Press
ISBN: 1439882541
Category : Computers
Languages : en
Pages : 381
Book Description
It is becoming increasingly apparent that all forms of communication-including voice-will be transmitted through packet-switched networks based on the Internet Protocol (IP). Therefore, the design of modern devices that rely on speech interfaces, such as cell phones and PDAs, requires a complete and up-to-date understanding of the basics of speech
Digital Speech Processing
Author: A. Nejat IncePublisher: Springer Science & Business Media
ISBN: 147572148X
Category : Technology & Engineering
Languages : en
Pages : 254
Book Description
After alm ost three scores of years of basic and applied research, the field of speech processing is, at present, undergoing a rapid growth in terms of both performance and applications and this is fueHed by the advances being made in the areas of microelectronics, computation and algorithm design.Speech processing relates to three aspects of voice communications: -Speech Coding and transmission which is mainly concerned with man-to man voice communication. -Speech Synthesis which deals with machine-to-man communication. -Speech Recognition which is related to man-to-machine communication. Widespread application and use of low-bit rate voice codec.>, synthesizers and recognizers which are all speech processing products requires ideaHy internationally accepted quality assessment and evaluation methods as weH as speech processing standards so that they may be interconnected and used independently of their designers and manufacturers without costly interfaces. This book presents, in a tutorial manner, both fundamental and applied aspects of the above topics which have been prepared by weH-known specialists in their respective areas. The book is based on lectures which were sponsored by AGARD/NATO and delivered by the authors, in several NATO countries, to audiences consisting mainly of academic and industrial R&D engineers and physicists as weH as civil and military C3I systems planners and designers.