Download or read book A Study on Speech Enhancement Method Based on Two Microphone Adaptive Beamforming Technique written by Nur Diana Kamarudin and published by . This book was released on 2011 with total page 62 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Speech Enhancement written by Shoji Makino and published by Springer Science & Business Media. This book was released on 2005-03-17 with total page 432 pages. Available in PDF, EPUB and Kindle. Book excerpt: We live in a noisy world! In all applications (telecommunications, hands-free communications, recording, human-machine interfaces, etc) that require at least one microphone, the signal of interest is usually contaminated by noise and reverberation. As a result, the microphone signal has to be "cleaned" with digital signal processing tools before it is played out, transmitted, or stored. This book is about speech enhancement. Different well-known and state-of-the-art methods for noise reduction, with one or multiple microphones, are discussed. By speech enhancement, we mean not only noise reduction but also dereverberation and separation of independent signals. These topics are also covered in this book. However, the general emphasis is on noise reduction because of the large number of applications that can benefit from this technology. The goal of this book is to provide a strong reference for researchers, engineers, and graduate students who are interested in the problem of signal and speech enhancement. To do so, we invited well-known experts to contribute chapters covering the state of the art in this focused field.
Download or read book Speech Enhancement written by Jacob Benesty and published by Elsevier. This book was released on 2014-01-04 with total page 143 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech enhancement is a classical problem in signal processing, yet still largely unsolved. Two of the conventional approaches for solving this problem are linear filtering, like the classical Wiener filter, and subspace methods. These approaches have traditionally been treated as different classes of methods and have been introduced in somewhat different contexts. Linear filtering methods originate in stochastic processes, while subspace methods have largely been based on developments in numerical linear algebra and matrix approximation theory. This book bridges the gap between these two classes of methods by showing how the ideas behind subspace methods can be incorporated into traditional linear filtering. In the context of subspace methods, the enhancement problem can then be seen as a classical linear filter design problem. This means that various solutions can more easily be compared and their performance bounded and assessed in terms of noise reduction and speech distortion. The book shows how various filter designs can be obtained in this framework, including the maximum SNR, Wiener, LCMV, and MVDR filters, and how these can be applied in various contexts, like in single-channel and multichannel speech enhancement, and in both the time and frequency domains. - First short book treating subspace approaches in a unified way for time and frequency domains, single-channel, multichannel, as well as binaural, speech enhancement - Bridges the gap between optimal filtering methods and subspace approaches - Includes original presentation of subspace methods from different perspectives
Download or read book Speech Enhancement Using a Small Microphone Array written by Qingning Zeng and published by . This book was released on 2009 with total page 183 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book 20 Jahre Kunstpreis Junger Westen written by and published by . This book was released on 1968 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Array Processing written by Jacob Benesty and published by Springer. This book was released on 2019-02-28 with total page 194 pages. Available in PDF, EPUB and Kindle. Book excerpt: The focus of this book is on array processing and beamforming with Kronecker products. It considers a large family of sensor arrays that allow the steering vector to be decomposed as a Kronecker product of two steering vectors of smaller virtual arrays. Instead of directly designing a global beamformer for the original array, once the steering vector has been decomposed, smaller virtual beamformers are designed and separately optimized for each virtual array. This means the matrices that need to be inverted are smaller, which increases the robustness of the beamformers, and reduces the size of the observations. The book explains how to perform beamforming with Kronecker product filters using an unconventional approach. It shows how the Kronecker product formulation can be used to derive fixed, adaptive, and differential beamformers with remarkable flexibility. Furthermore, it demonstrates how fixed and adaptive beamformers can be intelligently combined, optimally exploiting the advantages of both. The problem of spatiotemporal signal enhancement is also addressed, and readers will learn how to perform Kronecker product filtering in this context.
Download or read book Microphone Arrays written by Michael Brandstein and published by Springer Science & Business Media. This book was released on 2013-04-17 with total page 401 pages. Available in PDF, EPUB and Kindle. Book excerpt: This is the first book to provide a single complete reference on microphone arrays. Top researchers in this field contributed articles documenting the current state of the art in microphone array research, development and technological application.
Download or read book Canonical Correlation Analysis in Speech Enhancement written by Jacob Benesty and published by Springer. This book was released on 2017-08-31 with total page 124 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book focuses on the application of canonical correlation analysis (CCA) to speech enhancement using the filtering approach. The authors explain how to derive different classes of time-domain and time-frequency-domain noise reduction filters, which are optimal from the CCA perspective for both single-channel and multichannel speech enhancement. Enhancement of noisy speech has been a challenging problem for many researchers over the past few decades and remains an active research area. Typically, speech enhancement algorithms operate in the short-time Fourier transform (STFT) domain, where the clean speech spectral coefficients are estimated using a multiplicative gain function. A filtering approach, which can be performed in the time domain or in the subband domain, obtains an estimate of the clean speech sample at every time instant or time-frequency bin by applying a filtering vector to the noisy speech vector. Compared to the multiplicative gain approach, the filtering approach more naturally takes into account the correlation of the speech signal in adjacent time frames. In this study, the authors pursue the filtering approach and show how to apply CCA to the speech enhancement problem. They also address the problem of adaptive beamforming from the CCA perspective, and show that the well-known Wiener and minimum variance distortionless response (MVDR) beamformers are particular cases of a general class of CCA-based adaptive beamformers.
Download or read book Speech Processing in Modern Communication written by Israel Cohen and published by Springer Science & Business Media. This book was released on 2009-12-18 with total page 342 pages. Available in PDF, EPUB and Kindle. Book excerpt: Modern communication devices, such as mobile phones, teleconferencing systems, VoIP, etc., are often used in noisy and reverberant environments. Therefore, signals picked up by the microphones from telecommunication devices contain not only the desired near-end speech signal, but also interferences such as the background noise, far-end echoes produced by the loudspeaker, and reverberations of the desired source. These interferences degrade the fidelity and intelligibility of the near-end speech in human-to-human telecommunications and decrease the performance of human-to-machine interfaces (i.e., automatic speech recognition systems). The proposed book deals with the fundamental challenges of speech processing in modern communication, including speech enhancement, interference suppression, acoustic echo cancellation, relative transfer function identification, source localization, dereverberation, and beamforming in reverberant environments. Enhancement of speech signals is necessary whenever the source signal is corrupted by noise. In highly non-stationary noise environments, noise transients, and interferences may be extremely annoying. Acoustic echo cancellation is used to eliminate the acoustic coupling between the loudspeaker and the microphone of a communication device. Identification of the relative transfer function between sensors in response to a desired speech signal enables to derive a reference noise signal for suppressing directional or coherent noise sources. Source localization, dereverberation, and beamforming in reverberant environments further enable to increase the intelligibility of the near-end speech signal.
Download or read book Speech and Computer written by Andrey Ronzhin and published by Springer. This book was released on 2016-08-15 with total page 747 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book constitutes the proceedings of the 18th International Conference on Speech and Computer, SPECOM 2016, held in Budapest, Hungary, in August 2016. The 85 papers presented in this volume were carefully reviewed and selected from 154 submissions.
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
Download or read book Proceedings of the International Conference on Advanced Intelligent Systems and Informatics 2020 written by Aboul Ella Hassanien and published by Springer Nature. This book was released on 2020-09-19 with total page 893 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book presents the proceedings of the 6th International Conference on Advanced Intelligent Systems and Informatics 2020 (AISI2020), which took place in Cairo, Egypt, from October 19 to 21, 2020. This international and interdisciplinary conference, which highlighted essential research and developments in the fields of informatics and intelligent systems, was organized by the Scientific Research Group in Egypt (SRGE). The book is divided into several sections, covering the following topics: Intelligent Systems, Deep Learning Technology, Document and Sentiment Analysis, Blockchain and Cyber Physical System, Health Informatics and AI against COVID-19, Data Mining, Power and Control Systems, Business Intelligence, Social Media and Digital Transformation, Robotic, Control Design, and Smart Systems.
Download or read book Array Signal Processing written by Don H. Johnson and published by Pearson. This book was released on 1993 with total page 558 pages. Available in PDF, EPUB and Kindle. Book excerpt: This is the first book on the market to bring together material on array signal processing in a coherent fashion, with uniform notation and convention of models. KEY TOPICS: Using extensive examples and problems, it presents not only the theories of propagating waves and conventional array processing algorithms, but also the underlying ideas of adaptive array processing and multi-array tracking algorithms. This manual will be valuable to engineers who wish to practice and advance their careers in the array signal processing field.
Download or read book Dual Microphone Speech Enhancement Algorithms on Android based Devices for Hearing Study written by Nikhil Shankar and published by . This book was released on 2018 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech Enhancement (SE) is a key module in the Hearing Aid (HA) signal processing pipeline and improves the listening comfort. Over the last few decades, researchers have developed many single and dual-microphone SE techniques. In this thesis, two novel dual-channel SE techniques have been proposed and are implemented on Android-based smartphones as an assistive device for HA. In the first algorithm, the coherence between speech and noise signals is used to obtain an SE gain function, in combination with a Super-Gaussian Joint Maximum a Posteriori (SGJMAP) single microphone SE gain function. The second technique uses the Minimum Variance Distortionless Response (MVDR) as a Signal to Noise Ratio (SNR) booster for the SE method. The considered SE gain is based on the Log Spectral Minimum Mean Square Error Amplitude Estimator (Log-MMSE) to improve the speech quality in the presence of different background noise. Objective evaluation and subjective results of the developed methods show significant improvements in speech quality and intelligibility in comparison with existing SE methods.
Download or read book Journal of Rehabilitation Research Development written by and published by . This book was released on 2001 with total page 954 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Speech Dereverberation written by Patrick A. Naylor and published by Springer Science & Business Media. This book was released on 2010-07-27 with total page 388 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech Dereverberation gathers together an overview, a mathematical formulation of the problem and the state-of-the-art solutions for dereverberation. Speech Dereverberation presents current approaches to the problem of reverberation. It provides a review of topics in room acoustics and also describes performance measures for dereverberation. The algorithms are then explained with mathematical analysis and examples that enable the reader to see the strengths and weaknesses of the various techniques, as well as giving an understanding of the questions still to be addressed. Techniques rooted in speech enhancement are included, in addition to a treatment of multichannel blind acoustic system identification and inversion. The TRINICON framework is shown in the context of dereverberation to be a generalization of the signal processing for a range of analysis and enhancement techniques. Speech Dereverberation is suitable for students at masters and doctoral level, as well as established researchers.
Download or read book Journal of Rehabilitation Research and Development written by and published by . This book was released on 1986 with total page 1136 pages. Available in PDF, EPUB and Kindle. Book excerpt:
