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Single Channel Phase-Aware Signal Processing in Speech Communication

Single Channel Phase-Aware Signal Processing in Speech Communication
Author: Pejman Mowlaee
Publisher: John Wiley & Sons
Total Pages: 253
Release: 2016-12-27
Genre: Technology & Engineering
ISBN: 1119238811

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An overview on the challenging new topic of phase-aware signal processing Speech communication technology is a key factor in human-machine interaction, digital hearing aids, mobile telephony, and automatic speech/speaker recognition. With the proliferation of these applications, there is a growing requirement for advanced methodologies that can push the limits of the conventional solutions relying on processing the signal magnitude spectrum. Single-Channel Phase-Aware Signal Processing in Speech Communication provides a comprehensive guide to phase signal processing and reviews the history of phase importance in the literature, basic problems in phase processing, fundamentals of phase estimation together with several applications to demonstrate the usefulness of phase processing. Key features: Analysis of recent advances demonstrating the positive impact of phase-based processing in pushing the limits of conventional methods. Offers unique coverage of the historical context, fundamentals of phase processing and provides several examples in speech communication. Provides a detailed review of many references and discusses the existing signal processing techniques required to deal with phase information in different applications involved with speech. The book supplies various examples and MATLAB® implementations delivered within the PhaseLab toolbox. Single-Channel Phase-Aware Signal Processing in Speech Communication is a valuable single-source for students, non-expert DSP engineers, academics and graduate students.


A Perspective on Single-Channel Frequency-Domain Speech Enhancement

A Perspective on Single-Channel Frequency-Domain Speech Enhancement
Author: Jacob Benesty
Publisher: Morgan & Claypool Publishers
Total Pages: 111
Release: 2011-03-01
Genre: Technology & Engineering
ISBN: 1608456994

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This book focuses on a class of single-channel noise reduction methods that are performed in the frequency domain via the short-time Fourier transform (STFT). The simplicity and relative effectiveness of this class of approaches make them the dominant choice in practical systems. Even though many popular algorithms have been proposed through more than four decades of continuous research, there are a number of critical areas where our understanding and capabilities still remain quite rudimentary, especially with respect to the relationship between noise reduction and speech distortion. All existing frequency-domain algorithms, no matter how they are developed, have one feature in common: the solution is eventually expressed as a gain function applied to the STFT of the noisy signal only in the current frame. As a result, the narrowband signal-to-noise ratio (SNR) cannot be improved, and any gains achieved in noise reduction on the fullband basis come with a price to pay, which is speech distortion. In this book, we present a new perspective on the problem by exploiting the difference between speech and typical noise in circularity and interframe self-correlation, which were ignored in the past. By gathering the STFT of the microphone signal of the current frame, its complex conjugate, and the STFTs in the previous frames, we construct several new, multiple-observation signal models similar to a microphone array system: there are multiple noisy speech observations, and their speech components are correlated but not completely coherent while their noise components are presumably uncorrelated. Therefore, the multichannel Wiener filter and the minimum variance distortionless response (MVDR) filter that were usually associated with microphone arrays will be developed for single-channel noise reduction in this book. This might instigate a paradigm shift geared toward speech distortionless noise reduction techniques.


Latent Variable Analysis and Signal Separation

Latent Variable Analysis and Signal Separation
Author: Yannick Deville
Publisher: Springer
Total Pages: 583
Release: 2018-06-05
Genre: Computers
ISBN: 3319937642

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This book constitutes the proceedings of the 14th International Conference on Latent Variable Analysis and Signal Separation, LVA/ICA 2018, held in Guildford, UK, in July 2018.The 52 full papers were carefully reviewed and selected from 62 initial submissions. As research topics the papers encompass a wide range of general mixtures of latent variables models but also theories and tools drawn from a great variety of disciplines such as structured tensor decompositions and applications; matrix and tensor factorizations; ICA methods; nonlinear mixtures; audio data and methods; signal separation evaluation campaign; deep learning and data-driven methods; advances in phase retrieval and applications; sparsity-related methods; and biomedical data and methods.


Advances in Intelligent Computing and Communication

Advances in Intelligent Computing and Communication
Author: Swagatam Das
Publisher: Springer Nature
Total Pages: 713
Release: 2021-05-22
Genre: Technology & Engineering
ISBN: 9811606951

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This book presents high-quality research papers presented at the 3rd International Conference on Intelligent Computing and Advances in Communication (ICAC 2020) organized by Siksha ‘O’ Anusandhan Deemed to be University, Bhubaneswar, Odisha, India, in November 2020. This book brings out the new advances and research results in the fields of theoretical, experimental, and applied signal and image processing, soft computing, networking, and antenna research. Moreover, it provides a comprehensive and systematic reference on the range of alternative conversion processes and technologies.


Audio Source Separation and Speech Enhancement

Audio Source Separation and Speech Enhancement
Author: Emmanuel Vincent
Publisher: John Wiley & Sons
Total Pages: 506
Release: 2018-07-24
Genre: Technology & Engineering
ISBN: 1119279887

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Learn the technology behind hearing aids, Siri, and Echo Audio source separation and speech enhancement aim to extract one or more source signals of interest from an audio recording involving several sound sources. These technologies are among the most studied in audio signal processing today and bear a critical role in the success of hearing aids, hands-free phones, voice command and other noise-robust audio analysis systems, and music post-production software. Research on this topic has followed three convergent paths, starting with sensor array processing, computational auditory scene analysis, and machine learning based approaches such as independent component analysis, respectively. This book is the first one to provide a comprehensive overview by presenting the common foundations and the differences between these techniques in a unified setting. Key features: Consolidated perspective on audio source separation and speech enhancement. Both historical perspective and latest advances in the field, e.g. deep neural networks. Diverse disciplines: array processing, machine learning, and statistical signal processing. Covers the most important techniques for both single-channel and multichannel processing. This book provides both introductory and advanced material suitable for people with basic knowledge of signal processing and machine learning. Thanks to its comprehensiveness, it will help students select a promising research track, researchers leverage the acquired cross-domain knowledge to design improved techniques, and engineers and developers choose the right technology for their target application scenario. It will also be useful for practitioners from other fields (e.g., acoustics, multimedia, phonetics, and musicology) willing to exploit audio source separation or speech enhancement as pre-processing tools for their own needs.


Speech Enhancement

Speech Enhancement
Author: Philipos C. Loizou
Publisher: CRC Press
Total Pages: 715
Release: 2013-02-25
Genre: Technology & Engineering
ISBN: 1466599227

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With the proliferation of mobile devices and hearing devices, including hearing aids and cochlear implants, there is a growing and pressing need to design algorithms that can improve speech intelligibility without sacrificing quality. Responding to this need, Speech Enhancement: Theory and Practice, Second Edition introduces readers to the basic pr


Speech Enhancement

Speech Enhancement
Author: Shoji Makino
Publisher: Springer Science & Business Media
Total Pages: 432
Release: 2005-03-17
Genre: Computers
ISBN: 9783540240396

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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.


Phase-based Speech Processing

Phase-based Speech Processing
Author: Parham Aarabi
Publisher: World Scientific
Total Pages: 153
Release: 2006
Genre: Computers
ISBN: 9812566120

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This is the first book that takes a detailed look at the importance of phase in the design of speech processing systems. Phase, in comparison with amplitude, is often ignored for speech recognition applications. Thus, this book highlights some of the important ways in which the phase of speech signals can be utilized for sound localization, enhancement, and recognition.This book also discusses the state-of-the-art research in phase-based speech processing, starting from the basics of signal processing and recording, to single microphone speech recognition, the recognition of speech and the processing of speech by humans, as well as the importance of phase in human speech recognition and multi-microphone phase-based speech processing.


Single Channel Speech Enhancement in Severe Noise Conditions

Single Channel Speech Enhancement in Severe Noise Conditions
Author: Dariush David Farrokhi
Publisher:
Total Pages: 232
Release: 2011
Genre: Electronic circuits
ISBN:

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[Truncated abstract] SINGLE Channel Non Stationary Noise Speech Enhancement (SCNSNSE) algorithms can be used in many applications including enhancement of pre-recorded speech, hearing aids devices, speech recognition and telecommunication equipment. Many organizations such as medical, aviation and local or federal police are interested in having access to algorithms that can improve noisy speech signals. A combined set of existing and new algorithms were uniquely put together to produce a SCNSNSE system architecture. This novel SCNSNSE system architecture produces improved speech enhancement at low SNR (below 0 dB SNR). This SCNSNSE architecture contains novel algorithms at pre and post processing stages and enhances the speech signal which is contaminated with highly non-stationary noise. The SCNSNSE architecture consists of three major layers. Layer one consists of spectrum estimation (the preprocessing) of the SCNSNSE system architecture. At the spectrum estimation layer the concept of narrow band variation reduction is explained. The novel introduction of the Controlled Forward Moving Average (CFMA) algorithm greatly improves the reduction of narrow band variation. The CFMA is strategically placed in the SCNSNSE architecture to provide a better outcome. At the spectrum estimation layer a combination of existing algorithms cascaded with a new algorithm is applied as described below: 1. Discrete or Prolate Spheroidal Sequence (DPSS) multi-taper algorithm. 2. Controlled Forward Moving Average (CFMA) algorithm. 3. Stein s Unbiased Risk Estimator (SURE) wavelet thresholding. The second layer consists of the noise estimation (the post processing) algorithms. During the post-processing the concept of wide band variation estimation, Frequency Threshold Mapping (FTM) and Multi-Channel Threshold Mapping (MCTM) are introduced...