Download or read book A Fuzzy based Approach for Two microphone Speech Enhancement System written by Yanqin Dai and published by . This book was released on 2005 with total page 186 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Towards an Intelligent Fuzzy Based Multimodal Two Stage Speech Enhancement System written by Andrew Abel and published by . This book was released on 2013 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: This thesis presents a novel two stage multimodal speech enhancement system, making use of both visual and audio information to filter speech, and explores the extension of this system with the use of fuzzy logic to demonstrate proof of concept for an envisaged autonomous, adaptive, and context aware multimodal system. The design of the proposed cognitively inspired framework is scalable, meaning that it is possible for the techniques used in individual parts of the system to be upgraded and there is scope for the initial framework presented here to be expanded. In the proposed system, the concept of single modality two stage filtering is extended to include the visual modality. Noisy speech information received by a microphone array is first pre-processed by visually derived Wiener filtering employing the novel use of the Gaussian Mixture Regression (GMR) technique, making use of associated visual speech information, extracted using a state of the art Semi Adaptive Appearance Models (SAAM) based lip tracking approach. This pre-processed speech is then enhanced further by audio only beamforming using a state of the art Transfer Function Generalised Sidelobe Canceller (TFGSC) approach. This results in a system which is designed to function in challenging noisy speech environments (using speech sentences with different speakers from the GRID corpus and a range of noise recordings), and both objective and subjective test results (employing the widely used Perceptual Evaluation of Speech Quality (PESQ) measure, a composite objective measure, and subjective listening tests), showing that this initial system is capable of delivering very encouraging results with regard to filtering speech mixtures in difficult reverberant speech environments. Some limitations of this initial framework are identified, and the extension of this multimodal system is explored, with the development of a fuzzy logic based framework and a proof of concept demonstration implemented. Results show that this proposed autonomous,adaptive, and context aware multimodal framework is capable of delivering very positive results in difficult noisy speech environments, with cognitively inspired use of audio and visual information, depending on environmental conditions. Finally some concluding remarks are made along with proposals for future work.
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 Cognitively Inspired Audiovisual Speech Filtering written by Andrew Abel and published by Springer. This book was released on 2015-08-07 with total page 134 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book presents a summary of the cognitively inspired basis behind multimodal speech enhancement, covering the relationship between audio and visual modalities in speech, as well as recent research into audiovisual speech correlation. A number of audiovisual speech filtering approaches that make use of this relationship are also discussed. A novel multimodal speech enhancement system, making use of both visual and audio information to filter speech, is presented, and this book explores the extension of this system with the use of fuzzy logic to demonstrate an initial implementation of an autonomous, adaptive, and context aware multimodal system. This work also discusses the challenges presented with regard to testing such a system, the limitations with many current audiovisual speech corpora, and discusses a suitable approach towards development of a corpus designed to test this novel, cognitively inspired, speech filtering system.
Download or read book DFT Domain Based Single Microphone Noise Reduction for Speech Enhancement written by Richard C. Hendriks and published by Morgan & Claypool Publishers. This book was released on 2013-01-01 with total page 84 pages. Available in PDF, EPUB and Kindle. Book excerpt: As speech processing devices like mobile phones, voice controlled devices, and hearing aids have increased in popularity, people expect them to work anywhere and at any time without user intervention. However, the presence of acoustical disturbances limits the use of these applications, degrades their performance, or causes the user difficulties in understanding the conversation or appreciating the device. A common way to reduce the effects of such disturbances is through the use of single-microphone noise reduction algorithms for speech enhancement. The field of single-microphone noise reduction for speech enhancement comprises a history of more than 30 years of research. In this survey, we wish to demonstrate the significant advances that have been made during the last decade in the field of discrete Fourier transform domain-based single-channel noise reduction for speech enhancement.Furthermore, our goal is to provide a concise description of a state-of-the-art speech enhancement system, and demonstrate the relative importance of the various building blocks of such a system. This allows the non-expert DSP practitioner to judge the relevance of each building block and to implement a close-to-optimal enhancement system for the particular application at hand. Table of Contents: Introduction / Single Channel Speech Enhancement: General Principles / DFT-Based Speech Enhancement Methods: Signal Model and Notation / Speech DFT Estimators / Speech Presence Probability Estimation / Noise PSD Estimation / Speech PSD Estimation / Performance Evaluation Methods / Simulation Experiments with Single-Channel Enhancement Systems / Future Directions
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 Distant Speech Recognition written by Matthias Woelfel and published by John Wiley & Sons. This book was released on 2009-04-20 with total page 600 pages. Available in PDF, EPUB and Kindle. Book excerpt: A complete overview of distant automatic speech recognition The performance of conventional Automatic Speech Recognition (ASR) systems degrades dramatically as soon as the microphone is moved away from the mouth of the speaker. This is due to a broad variety of effects such as background noise, overlapping speech from other speakers, and reverberation. While traditional ASR systems underperform for speech captured with far-field sensors, there are a number of novel techniques within the recognition system as well as techniques developed in other areas of signal processing that can mitigate the deleterious effects of noise and reverberation, as well as separating speech from overlapping speakers. Distant Speech Recognitionpresents a contemporary and comprehensive description of both theoretic abstraction and practical issues inherent in the distant ASR problem. Key Features: Covers the entire topic of distant ASR and offers practical solutions to overcome the problems related to it Provides documentation and sample scripts to enable readers to construct state-of-the-art distant speech recognition systems Gives relevant background information in acoustics and filter techniques, Explains the extraction and enhancement of classification relevant speech features Describes maximum likelihood as well as discriminative parameter estimation, and maximum likelihood normalization techniques Discusses the use of multi-microphone configurations for speaker tracking and channel combination Presents several applications of the methods and technologies described in this book Accompanying website with open source software and tools to construct state-of-the-art distant speech recognition systems This reference will be an invaluable resource for researchers, developers, engineers and other professionals, as well as advanced students in speech technology, signal processing, acoustics, statistics and artificial intelligence fields.
Download or read book Speech Enhancement Exploiting the Source filter Model written by Samy Elshamy and published by . This book was released on 2020 with total page 0 pages. Available in PDF, EPUB and Kindle. Book excerpt: Imagining everyday life without mobile telephony is nowadays hardly possible. Calls are being made in every thinkable situation and environment. Hence, the microphone will not only pick up the user's speech but also sound from the surroundings which is likely to impede the understanding of the conversational partner. Modern speech enhancement systems are able to mitigate such effects and most users are not even aware of their existence. In this thesis the development of a modern single-channel speech enhancement approach is presented, which uses the divide and conquer principle to combat environmental noise in microphone signals. Though initially motivated by mobile telephony applications, this approach can be applied whenever speech is to be retrieved from a corrupted signal. The approach uses the so-called source-filter model to divide the problem into two subproblems which are then subsequently conquered by enhancing the source (the excitation signal) and the filter (the spectral envelope) separately. Both enhanced signals are then used to denoise the corrupted signal. The estimation of spectral envelopes has quite some history and some approaches already exist for speech enhancement. However, they typically neglect the excitation signal which leads to the inability of enhancing the fine structure properly. Both individual enhancement approaches exploit benefits of the cepstral domain which offers, e.g., advantageous mathematical properties and straightforward synthesis of excitation-like signals. We investigate traditional model-based schemes like Gaussian mixture models (GMMs), classical signal processing-based, as well as modern deep neural network (DNN)-based approaches in this thesis. The enhanced signals are not used directly to enhance the corrupted signal (e.g., to synthesize a clean speech signal) but as so-called a priori signal-to-noise ratio (SNR) estimate in a traditional statistical speech enhancement system. Such a traditional system consists of a noise power estimator, an a priori SNR estimator, and a spectral weighting rule that is usually driven by the results of the aforementioned estimators and subsequently employed to retrieve the clean speech estimate from the noisy observation. As a result the new approach obtains significantly higher noise attenuation compared to current state-of-the-art systems while maintaining a quite comparable speech component quality and speech intelligibility. In consequence, the overall quality of the enhanced speech signal turns out to be superior as compared to state-of-the-art speech ehnahcement approaches.
Download or read book 1999 IEEE International Conference on Acoustics Speech and Signal Processing written by IEEE Signal Processing Society and published by . This book was released on 1999 with total page 642 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book A Two microphone Speech Enhancement System for Monaural Listening written by Michael Vincent Carroll McConnell and published by . This book was released on 1985 with total page 206 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book DFT Domain Based Single Microphone Noise Reduction for Speech Enhancement written by Richard C. Hendriks and published by Springer. This book was released on 2013-02-11 with total page 70 pages. Available in PDF, EPUB and Kindle. Book excerpt: As speech processing devices like mobile phones, voice controlled devices, and hearing aids have increased in popularity, people expect them to work anywhere and at any time without user intervention. However, the presence of acoustical disturbances limits the use of these applications, degrades their performance, or causes the user difficulties in understanding the conversation or appreciating the device. A common way to reduce the effects of such disturbances is through the use of single-microphone noise reduction algorithms for speech enhancement. The field of single-microphone noise reduction for speech enhancement comprises a history of more than 30 years of research. In this survey, we wish to demonstrate the significant advances that have been made during the last decade in the field of discrete Fourier transform domain-based single-channel noise reduction for speech enhancement.Furthermore, our goal is to provide a concise description of a state-of-the-art speech enhancement system, and demonstrate the relative importance of the various building blocks of such a system. This allows the non-expert DSP practitioner to judge the relevance of each building block and to implement a close-to-optimal enhancement system for the particular application at hand. Table of Contents: Introduction / Single Channel Speech Enhancement: General Principles / DFT-Based Speech Enhancement Methods: Signal Model and Notation / Speech DFT Estimators / Speech Presence Probability Estimation / Noise PSD Estimation / Speech PSD Estimation / Performance Evaluation Methods / Simulation Experiments with Single-Channel Enhancement Systems / Future Directions
Download or read book Intelligent Systems written by Cornelius T. Leondes and published by CRC Press. This book was released on 2018-10-08 with total page 2400 pages. Available in PDF, EPUB and Kindle. Book excerpt: Intelligent systems, or artificial intelligence technologies, are playing an increasing role in areas ranging from medicine to the major manufacturing industries to financial markets. The consequences of flawed artificial intelligence systems are equally wide ranging and can be seen, for example, in the programmed trading-driven stock market crash of October 19, 1987. Intelligent Systems: Technology and Applications, Six Volume Set connects theory with proven practical applications to provide broad, multidisciplinary coverage in a single resource. In these volumes, international experts present case-study examples of successful practical techniques and solutions for diverse applications ranging from robotic systems to speech and signal processing, database management, and manufacturing.
Download or read book Neutrosophic approach for enhancing quality of signals written by Sudan Jha and published by Infinite Study. This book was released on with total page 32 pages. Available in PDF, EPUB and Kindle. Book excerpt: Information in a signal is often followed by undesirable disturbance which is termed as noise. Preventing noise in the signal leads to signal integrity, which also leads to better signal quality. The previous related works have the major issues while reducing noise in signals regarding assumptions, frequency and time domain, etc. This paper proposes a new Neutrosophic approach to reduce noises and errors in signal transmission. In the proposed method, confidence function is used as the truth membership function, which is associated with sampled time intervals.
Download or read book Multimedia Communications written by Francesco De Natale and published by Springer Science & Business Media. This book was released on 2012-12-06 with total page 586 pages. Available in PDF, EPUB and Kindle. Book excerpt: Multimedia Communications is at the core of the advanced interactive services that make up today's Information Society. Videoconferencing, teleworking, teleshopping and video-on-demand will benefit from developments in broadband and mobile telecommunication systems, intelligent multimedia terminals and digital signal processing. The latest research findings from these fields are presented here in the proceedings of the 10th Tyrrhenian Workshop on Digital Communications, held in Ischia, Italy, September 19 98. Focus is placed on the following four areas: Signal Processing for Multimedia Communications. Modeling, Analysis and Simulation of Multimedia Traffic Sources. Access Techniques. Multimode Multimedia Terminals. In particular, multimedia services and applications are presented. This comprehensive collection of papers will enable the reader to keep pace with the rapid changes that are taking place in this field. Experts have co-operated with top research centers worldwide, on an academic and industrial level, to make this an up-to-date reference volume for all those who are concerned with technological advances in Multimedia Distributed Systems.
Download or read book Dual microphone and Binaural Noise Reduction Techniques for Improved Speech Intelligibility by Hearing Aid Users written by Nima Yousefian Jazi and published by . This book was released on 2013 with total page 218 pages. Available in PDF, EPUB and Kindle. Book excerpt: Spatial filtering and directional discrimination has been shown to be an effective pre-processing approach for noise reduction in microphone array systems. In dual-microphone hearing aids, fixed and adaptive beamforming techniques are the most common solutions for enhancing the desired speech and rejecting unwanted signals captured by the microphones. In fact, beamformers are widely utilized in systems where spatial properties of target source (usually in front of the listener) is assumed to be known. In this dissertation, some dual-microphone coherence-based speech enhancement techniques applicable to hearing aids are proposed. All proposed algorithms operate in the frequency domain and (like traditional beamforming techniques) are purely based on the spatial properties of the desired speech source and does not require any knowledge of noise statistics for calculating the noise reduction filter. This benefit gives our algorithms the ability to address adverse noise conditions, such as situations where interfering talker(s) speaks simultaneously with the target speaker. In such cases, the (adaptive) beamformers lose their effectiveness in suppressing interference, since the noise channel (reference) cannot be built and updated accordingly. This difference is the main advantage of the proposed techniques in the dissertation over traditional adaptive beamformers. Furthermore, since the suggested algorithms are independent of noise estimation, they offer significant improvement in scenarios that the power level of interfering sources are much more than that of target speech. The dissertation also shows the premise behind the proposed algorithms can be extended and employed to binaural hearing aids. The main purpose of the investigated techniques is to enhance the intelligibility level of speech, measured through subjective listening tests with normal hearing and cochlear implant listeners. However, the improvement in quality of the output speech achieved by the algorithms are also presented to show that the proposed methods can be potential candidates for future use in commercial hearing aids and cochlear implant devices.
