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 Masters Theses in the Pure and Applied Sciences written by Wade H. Shafer and published by Springer Science & Business Media. This book was released on 2012-12-06 with total page 414 pages. Available in PDF, EPUB and Kindle. Book excerpt: Masters Theses in the Pure and Applied Sciences was first conceived, published, SIld disseminated by the Center for Information and Numerical Data Analysis and Synthesis (CINDAS) * at Purdue University in 1957, starting its coverage of theses with the academic year 1955. Beginning with Volume 13, the printing and dissemination phases of the activity were transferred to University Microfilms/Xerox of Ann Arbor, Michigan, with the thought that such an arrangement would be more beneficial to the academic and general scientific and technical community. After five years of this joint undertaking we had concluded that it was in the interest of all con cerned if the printing and distribution of the volumes were handled by an interna and broader dissemination. tional publishing house to assure improved service Hence, starting with Volume 18, Masters Theses in the Pure and Applied Sciences has been disseminated on a worldwide basis by Plenum Publishing Cor poration of New York, and in the same year the coverage was broadened to include Canadian universities. All back issues can also be ordered from Plenum. We have reported in Volume 30 (thesis year 1985) a total of 12,400 theses titles from 26 Canadian and 186 United States universities. We are sure that this broader base for these titles reported will greatly enhance the value of this important annual reference work.
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 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 RLE Progress Report written by Massachusetts Institute of Technology. Research Laboratory of Electronics and published by . This book was released on 1986 with total page 300 pages. Available in PDF, EPUB and Kindle. Book excerpt:
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 Audio Source Separation and Speech Enhancement written by Emmanuel Vincent and published by John Wiley & Sons. This book was released on 2018-10-22 with total page 517 pages. Available in PDF, EPUB and Kindle. Book excerpt: 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.
Download or read book DFT Domain Based Single Microphone Noise Reduction for Speech Enhancement written by Richard C. Hendriks and published by Springer Nature. This book was released on 2022-05-31 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 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.
Download or read book The Technology of Binaural Listening written by Jens Blauert and published by Springer Science & Business Media. This book was released on 2013-06-07 with total page 516 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book reports on the application of advanced models of the human binaural hearing system in modern technology, among others, in the following areas: binaural analysis of aural scenes, binaural de-reverberation, binaural quality assessment of audio channels, loudspeakers and performance spaces, binaural perceptual coding, binaural processing in hearing aids and cochlea implants, binaural systems in robots, binaural/tactile human-machine interfaces, speech-intelligibility prediction in rooms and/or multi-speaker scenarios. An introduction to binaural modeling and an outlook to the future are provided. Further, the book features a MATLAB toolbox to enable readers to construct their own dedicated binaural models on demand.
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 Convolutional and Recurrent Neural Networks for Real time Speech Separation in the Complex Domain written by Ke Tan and published by . This book was released on 2021 with total page 181 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech signals are usually distorted by acoustic interference in daily listening environments. Such distortions severely degrade speech intelligibility and quality for human listeners, and make many speech-related tasks, such as automatic speech recognition and speaker identification, very difficult. The use of deep learning has led to tremendous advances in speech enhancement over the last decade. It has been increasingly important to develop deep learning based real-time speech enhancement systems due to the prevalence of many modern smart devices that require real-time processing. The objective of this dissertation is to develop real-time speech enhancement algorithms to improve intelligibility and quality of noisy speech. Our study starts by developing a strong convolutional neural network (CNN) for monaural speech enhancement. The key idea is to systematically aggregate temporal contexts through dilated convolutions, which significantly expand receptive fields. Our experimental results suggest that the proposed model consistently outperforms a feedforward deep neural network (DNN), a unidirectional long short-term memory (LSTM) model and a bidirectional LSTM model in terms of objective speech intelligibility and quality metrics. Although significant progress has been made on deep learning based speech enhancement, most existing studies only exploit magnitude-domain information and enhance the magnitude spectra. We propose to perform complex spectral mapping with a gated convolutional recurrent network (GCRN). Such an approach simultaneously enhances magnitude and phase of speech. Evaluation results show that the proposed GCRN substantially outperforms an existing CNN for complex spectral mapping. Moreover, the proposed approach yields significantly better results than magnitude spectral mapping and complex ratio masking. To achieve strong enhancement performance typically requires a large DNN, making it difficult to deploy such speech enhancement systems on devices with limited hardware resources or in applications with strict latency requirements. We propose two compression pipelines to reduce the model size for DNN-based speech enhancement. We systematically investigate these techniques and evaluate the proposed compression pipelines. Experimental results demonstrate that our approach reduces the sizes of four different models by large margins without significantly sacrificing their enhancement performance. An important application of real-time speech enhancement lies in mobile speech communication. We propose a deep learning based real-time enhancement algorithm for dual-microphone mobile phones. The proposed algorithm employs a new densely-connected convolutional recurrent network to perform dual-channel complex spectral mapping. By compressing the model with a structured pruning technique, we derive an efficient system amenable to real-time processing. Experimental results suggest that the proposed algorithm consistently outperforms an earlier algorithm to dual-channel speech enhancement for mobile phone communication, as well as a deep learning based beamformer. Multi-channel complex spectral mapping (CSM) has proven to be effective in speech separation, assuming a fixed geometry of the microphone array. We comprehensively investigate this approach, and find that multi-channel CSM achieves separation performance better than or comparable to conventional and masking-based beamforming for different array geometries and speech separation tasks. Our investigation demonstrates that this all-neural approach is a general and effective spatial filter for multi-channel speech separation.
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 User Customizable Real time Single and Dual Microphone Speech Enhancement and Blind Speech Separation for Smartphone Hearing Aid Applications written by Chandan Karadagur Ananda Reddy 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 vital algorithmic component in the Hearing Aid pipeline. Over the years, several algorithms have been developed to work in real-time and to improve the quality and intelligibility of speech. However, noise suppression with minimal distortion to speech is still a prime challenge that needs to be addressed. In this work, a new single microphone SE is introduced that is implemented on a smartphone to work as an assistive device to Hearing Aids via wireless connectivity. The uniqueness of the developed method is in the introduction of varying parameters that allow the smartphone user to control the amount of noise suppression and speech distortion in real-time, which allows the user to customize the perceptual audio to their preference. A super-Gaussian extension of this approach is explored and analyzed. With the recent accessibility of the two microphones on the smartphones, doors were opened to use beamformer as a pre-filtering stage to the proposed single microphone SE. Real-time blind speech separation technique is also proposed to yield superior quality for speech. Objective and subjective results show that the developed methods outperform traditional SE techniques.
Download or read book The Journal of the Acoustical Society of America written by Acoustical Society of America and published by . This book was released on 2006 with total page 1222 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Nonlinear Analyses and Algorithms for Speech Processing written by Marcos Faundez-Zanuy and published by Springer. This book was released on 2006-02-08 with total page 393 pages. Available in PDF, EPUB and Kindle. Book excerpt: Refereed postproceedings of the International Conference on Non-Linear Speech Processing, NOLISP 2005. The 30 revised full papers presented together with one keynote speech and 2 invited talks were carefully reviewed and selected from numerous submissions for inclusion in the book. The papers are organized in topical sections on speaker recognition, speech analysis, voice pathologies, speech recognition, speech enhancement, and applications.
