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 Smartphone based Single and Dual Microphone Speech Enhancement Algorithms for Hearing Study written by Gautam Shreedhar Bhat 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 elemental in many real world applications. In the last two decades, extensive studies have been carried out on single and multi-channel SE techniques. In this thesis, three novel SE algorithms have been proposed that can be used for Hearing Aid Devices using a smartphone as their assistive device. The first SE method exploits the information of formant locations to improve the speech quality and intelligibility of the Super-Gaussian Joint Maximum aposterori (SGJMAP) SE method. The second method is the extension of this work on the Log Spectral Minimum Mean Square Error Amplitude Estimator (Log-MMSE) which is a well-known SE algorithm. The third method is a real time Blind Source Separation (BSS) method based on Independent Vector Analysis (IVA) for convolutive mixtures. Objective and subjective evaluation of the developed techniques show substantial improvements in speech quality and intelligibility.
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 Audio Source Separation and Speech Enhancement written by Emmanuel Vincent and published by John Wiley & Sons. This book was released on 2018-07-24 with total page 506 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 Real time Single and Dual channel Speech Enhancement on Edge Devices for Hearing Applications written by Nikhil Shankar and published by . This book was released on 2021 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech Enhancement (SE) is an important module in the signal processing pipeline for hearing applications and it helps enhance the comfort of listening. Many single and dualmicrophone SE techniques have been developed by researchers over the last few decades. In this thesis, novel single and dual-channel SE techniques have been proposed and are implemented on edge devices as an assistive tool for hearing applications. The smartphone is considered as the processing platform for real-time implementation and testing. In this work, both statistical signal processing and deep learning algorithms are proposed for SE. Firstly, we compare different two-channel beamformers for SE. Later, the Minimum Variance Distortionless Response (MVDR) beamformer assisted by a voice activity detector (VAD) is used as a Signal to Noise Ratio (SNR) booster for the SE method. Deep neural network architectures comprising of convolutional neural network (CNN) and recurrent neural network (RNN) layers are proposed in this thesis for real-time SE. Finally to filter out background noise, the SE gain estimation for noisy speech mixture is smoothed along the frequency axis by a Mel filter-bank, resulting in a Mel-warped frequency-domain gain estimation. In comparison with existing SE methods, objective assessment and subjective results of the developed methods indicate substantial improvements in speech quality and intelligibility.
Download or read book Real time Speech Processing Algorithms for Smartphone Based Hearing Aid Applications written by Gautam Shreedhar Bhat and published by . This book was released on 2021 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: Signal processing algorithms are extensively used in hearing aid applications to improve the quality and intelligibility of speech. The hearing aid device (HAD) signal processing pipeline consists of several key modules that help to improve the perception for hearing-impaired listeners. In this dissertation, novel speech processing algorithms have been proposed that can be used in smartphone-based hearing aid (HA) setup. Every chapter of this dissertation concentrates on the individual modules of the signal processing pipeline in HADs. The first algorithm is developed for speech enhancement (SE) to suppress the background noise. A voice activity detector (VAD) to classify the incoming signal into speech or noise is developed. Signal enhancement techniques like blind source separation and dereverberation are developed. The algorithms are developed using conventional and supervised learning techniques. Objective and subjective evaluations are conducted for each of the proposed techniques to show substantial improvements in speech quality and intelligibility.
Download or read book Digital Speech Transmission and Enhancement written by Peter Vary and published by John Wiley & Sons. This book was released on 2023-11-23 with total page 596 pages. Available in PDF, EPUB and Kindle. Book excerpt: DIGITAL SPEECH TRANSMISSION AND ENHANCEMENT Enables readers to understand the latest developments in speech enhancement/transmission due to advances in computational power and device miniaturization The Second Edition of Digital Speech Transmission and Enhancement has been updated throughout to provide all the necessary details on the latest advances in the theory and practice in speech signal processing and its applications, including many new research results, standards, algorithms, and developments which have recently appeared and are on their way into state-of-the-art applications. Besides mobile communications, which constituted the main application domain of the first edition, speech enhancement for hearing instruments and man-machine interfaces has gained significantly more prominence in the past decade, and as such receives greater focus in this updated and expanded second edition. Readers can expect to find information and novel methods on: Low-latency spectral analysis-synthesis, single-channel and dual-channel algorithms for noise reduction and dereverberation Multi-microphone processing methods, which are now widely used in applications such as mobile phones, hearing aids, and man-computer interfaces Algorithms for near-end listening enhancement, which provide a significantly increased speech intelligibility for users at the noisy receiving side of their mobile phone Fundamentals of speech signal processing, estimation and machine learning, speech coding, error concealment by soft decoding, and artificial bandwidth extension of speech signals Digital Speech Transmission and Enhancement is a single-source, comprehensive guide to the fundamental issues, algorithms, standards, and trends in speech signal processing and speech communication technology, and as such is an invaluable resource for engineers, researchers, academics, and graduate students in the areas of communications, electrical engineering, and information technology.
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 Speech Enhancement Techniques for Digital Hearing Aids written by Komal R. Borisagar and published by Springer. This book was released on 2018-11-15 with total page 162 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book provides various speech enhancement algorithms for digital hearing aids. It covers information on noise signals extracted from silences of speech signal. The description of the algorithm used for this purpose is also provided. Different types of adaptive filters such as Least Mean Squares (LMS), Normalized LMS (NLMS) and Recursive Lease Squares (RLS) are described for noise reduction in the speech signals. Different types of noises are taken to generate noisy speech signals, and therefore information on various noises signals is provided. The comparative performance of various adaptive filters for noise reduction in speech signals is also described. In addition, the book provides a speech enhancement technique using adaptive filtering and necessary frequency strength enhancement using wavelet transform as per the requirement of audiogram for digital hearing aids. Presents speech enhancement techniques for improving performance of digital hearing aids; Covers various types of adaptive filters and their advantages and limitations; Provides a hybrid speech enhancement technique using wavelet transform and adaptive filters.
Download or read book Robust Single Channel Speech Enhancement and Speaker Localization in Adverse Environments written by Saeed Mosayyebpour and published by . This book was released on 2014 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: In speech communication systems such as voice-controlled systems, hands-free mobile telephones and hearing aids, the received signals are degraded by room reverberation and background noise. This degradation can reduce the perceived quality and intelligibility of the speech, and decrease the performance of speech enhancement and source localization. These problems are difficult to solve due to the colored and nonstationary nature of the speech signals, and features of the Room Impulse Response (RIR) such as its long duration and non-minimum phase. In this dissertation, we focus on two topics of speech enhancement and speaker localization in noisy reverberant environments. A two-stage speech enhancement method is presentedto suppress both early and late reverberation in noisy speech using only one microphone. It is shown that this method works well even in highly reverberant rooms.Experiments under different acoustic conditions confirm that the proposed blind method is superior in terms of reducing early and late reverberation effects and noise compared to other well known single-microphone techniques in the literature.Time Difference Of Arrival (TDOA)-based methods usually provide the most accurate source localization in adverse conditions. The key issue for these methods is to accurately estimate the TDOA using the smallest number of microphones.Two robust Time Delay Estimation (TDE) methods are proposed which use the information from only two microphones. One method is based on adaptive inverse filtering which provides superior performance even in highly reverberant and moderately noisy conditions. It also has negligible failure estimation which makes it a reliable method in realistic environments. This method has high computational complexity due to the estimation in the first stage for the first microphone. As a result, it can not be applied in time-varying environments and real-time applications. Our second method improves this problem by introducing two effective preprocessing stages for the conventional Cross Correlation (CC)-based methods. The results obtained in different noisy reverberant conditions including a real and time-varying environment demonstrate that the proposed methods are superior compared to the conventional TDE methods.
Download or read book Blind Speech Separation written by Shoji Makino and published by Springer. This book was released on 2007-09-20 with total page 432 pages. Available in PDF, EPUB and Kindle. Book excerpt: This is the world’s first edited book on independent component analysis (ICA)-based blind source separation (BSS) of convolutive mixtures of speech. This book brings together a small number of leading researchers to provide tutorial-like and in-depth treatment on major ICA-based BSS topics, with the objective of becoming the definitive source for current, comprehensive, authoritative, and yet accessible treatment.
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 Robust Microphone Array Processing for Speech Enhancement in Hearing Aids written by Michael W. Hoffman and published by . This book was released on 1992 with total page 424 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Personalization of Noise Reduction and Compression for Hearing Enhancement written by Nasim Taghizadeh Alamdari and published by . This book was released on 2021 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: Noise reduction and dynamic range compression constitute two main signal processing modules for hearing enhancement in modern digital hearing aid devices. This dissertation covers personalization solutions for both of these two modules. First, in contrast to the great majority of previous works that are designed based on pre-collected datasets, in this dissertation, a personalized noise reduction solution is developed for field deployment which is capable of dealing with unseen noisy audio environments. More specifically, a deep learning-based approach is devised to improve speech denoising in real-world audio environments by not requiring the availability of clean speech signals as reference. A fully convolutional neural network is designed based on two noisy realizations of the same speech signal, one used as the input and the other as the target of the network. The results of extensive experimentations are reported to show the superiority of the developed personalized deep learning-based speech denoising approach over existing approaches. In support of personalized noise reduction, a personalized noise classification is also developed in this dissertation by performing the noise classification in an unsupervised manner. Second, in contrast to the existing prescriptive compression strategies used in hearing aids which are devised based on gain averages from a group of users, a personalized compression solution is developed in this dissertation via a human-in-the-loop deep reinforcement learning approach. The developed approach is designed to learn a specific user's hearing preferences in order to set compression ratios based on the user's preference feedbacks. Both simulation and subject testing results are reported to show the superiority of the developed personalized deep learning-based compression over conventional prescriptive compression. In addition, four smartphone apps in support of the above two modules are developed in this dissertation which include the real-time implementation of noise classification, noise reduction, and dynamic range compression.
Download or read book Microphone Array Algorithms and Architectures for Hearing Aid and Speech Enhancement Applications written by Arthur Wei-Tsu Wang and published by . This book was released on 1997 with total page 338 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Transform Domain Model based Wideband Speech Enhancement with Hearing Aid Applications written by Brady Nicholas Mahoney Laska and published by . This book was released on 2010 with total page 390 pages. Available in PDF, EPUB and Kindle. Book excerpt:
