LSP-based Acoustic Echo Cancellation and Noise Reduction PDF Download
Are you looking for read ebook online? Search for your book and save it on your Kindle device, PC, phones or tablets. Download LSP-based Acoustic Echo Cancellation and Noise Reduction PDF full book. Access full book title LSP-based Acoustic Echo Cancellation and Noise Reduction by Zhongwei Zhuang. Download full books in PDF and EPUB format.
Author: Zhongwei Zhuang
Publisher:
ISBN:
Category : Noise control
Languages : en
Pages : 198
Get Book Here
Book Description
Author: Zhongwei Zhuang
Publisher:
ISBN:
Category : Noise control
Languages : en
Pages : 198
Get Book Here
Book Description
Author:
Publisher:
ISBN:
Category : Acoustical engineering
Languages : en
Pages : 380
Get Book Here
Book Description
Author: Jacob Benesty
Publisher: Springer Science & Business Media
ISBN: 3642225748
Category : Technology & Engineering
Languages : en
Pages : 141
Get Book Here
Book Description
Single-channel hands-free teleconferencing systems are becoming popular. In order to enhance the communication quality of these systems, more and more stereophonic sound devices with two loudspeakers and two microphones are deployed. Because of the coupling between loudspeakers and microphones, there may be strong echoes, which make real-time communication very difficult. The best way we know to cancel these echoes is via a stereo acoustic echo canceller (SAEC), which can be modelled as a two-input/two-output system with real random variables. In this work, the authors recast this problem into a single-input/single-output system with complex random variables thanks to the widely linear model. From this new convenient formulation, they re-derive the most important aspects of a SAEC, including identification of the echo paths with adaptive filters, double-talk detection, and suppression.
Author: Jang Chyuan Jenq
Publisher:
ISBN: 9783330822221
Category :
Languages : en
Pages :
Get Book Here
Book Description
Author: Nils Anders Stefan Gustafsson
Publisher:
ISBN: 9783860738306
Category :
Languages : en
Pages : 142
Get Book Here
Book Description
Author:
Publisher:
ISBN:
Category :
Languages : en
Pages :
Get Book Here
Book Description
Author: Venkatraman Atti
Publisher: Springer Nature
ISBN: 3031015169
Category : Technology & Engineering
Languages : en
Pages : 113
Get Book Here
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
Author: J. Benesty
Publisher: Springer
ISBN: 9783642075070
Category : Technology & Engineering
Languages : en
Pages : 222
Get Book Here
Book Description
This book brings together many advanced topics in network and acoustic echo cancellation aimed towards enhancing the echo cancellation performance of next-generation telecommunication systems. The resulting compendium provides a coherent treatment of such topics not found otherwise in journals or other books.
Author: Robert W. Stewart (Electrical engineer)
Publisher:
ISBN:
Category :
Languages : en
Pages : 118
Get Book Here
Book Description
Author: Michael Oliver Vukadinovic
Publisher:
ISBN:
Category :
Languages : en
Pages : 0
Get Book Here
Book Description
Hands-free communication techniques are becoming increasingly important in the world of modern telephony. The convenience of hands-free links is invaluable in teleconferencing situations, automobiles and even in personal communications. The loud-speaker and microphone on a hands-free set are inherently coupled acoustically, producing an echo which degrades the quality of conversation over such a link. The conventional acoustic echo cancellation method is gain control which provides a half-duplex communication link. Research into adaptive filters which can provide a full-duplex communication link is intense but transparent transmission of speech over a hands-free link remains an elusive goal. Conventional acoustic echo cancellers are concerned with minimizing an objective measure of the level of echo. This thesis presents a study of two adaptive filtering structures which are based on the goal of providing the best subjective echo cancellation possible. The adaptive filtering algorithms attempt to give the echo the same spectral shape as the absolute hearing threshold curve, in order to minimize the echo's audibility. The filtered-E LMS algorithm, considered first, was found not to perform the kind of frequency weighting desired. A proposed subband adaptive filter with perceptually-based adaptive tap assignment, when trained with natural speech, is capable of providing a noticeable improvement in audible echo cancellation; however, when trained with bandlimited speech, the algorithm's performance was found to be only slightly better or comparable to the fixed tap assignment algorithm in terms of audible echo cancellation. The subband adaptive filtering algorithm is evaluated by means of simple listening tests performed by the author.
