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 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 Proceedings of the International Conference on Paradigms of Computing Communication and Data Sciences written by Rajendra Prasad Yadav and published by Springer Nature. This book was released on 2023-02-23 with total page 765 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book gathers selected high-quality research papers presented at International Conference on Paradigms of Communication, Computing and Data Sciences (PCCDS 2022), held at Malaviya National Institute of Technology Jaipur, India, during 05 – 07 July 2022. It discusses high-quality and cutting-edge research in the areas of advanced computing, communications and data science techniques. The book is a collection of latest research articles in computation algorithm, communication and data sciences, intertwined with each other for efficiency.

Download or read book Real time Smartphone Apps Integrating Signal Processing Modules of Hearing Aids written by Tahsin Ahmed Chowdhury and published by . This book was released on 2018 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: This thesis covers the integration of several signal processing modules that appear in digital hearing aids as real-time smartphone apps. The modules that are considered for integration in the developed apps in this thesis are voice activity detector, supervised noise classifier, noise reduction, and compression. The objective of this work has been to lay the foundation for realtime implementation of integrating various signal processing pipelines that are used in digital hearing aids on the smartphone platform as an open source platform which is widely used and carried by people. More specifically, two integrations are covered in this thesis. The first integration involves combining a previously developed voice activity detector and a previously developed supervised noise classifier into a real-time smartphone app. The second integration involves combining another previously developed voice activity detector, a previously developed noise reduction, and the compression module in the MATLAB Audio System Toolbox into a real-time smartphone app. The results obtained indicate that the developed integrations run in real-time on both iOS and Android smartphones.

Download or read book Intelligent Speech Signal Processing written by Nilanjan Dey and published by Academic Press. This book was released on 2019-04-02 with total page 210 pages. Available in PDF, EPUB and Kindle. Book excerpt: Intelligent Speech Signal Processing investigates the utilization of speech analytics across several systems and real-world activities, including sharing data analytics, creating collaboration networks between several participants, and implementing video-conferencing in different application areas. Chapters focus on the latest applications of speech data analysis and management tools across different recording systems. The book emphasizes the multidisciplinary nature of the field, presenting different applications and challenges with extensive studies on the design, development and management of intelligent systems, neural networks and related machine learning techniques for speech signal processing.

Download or read book Applied Speech Processing written by Nilanjan Dey and published by Academic Press. This book was released on 2021-01-19 with total page 208 pages. Available in PDF, EPUB and Kindle. Book excerpt: Applied Speech Processing: Algorithms and Case Studies is concerned with supporting and enhancing the utilization of speech analytics in several systems and real-world activities, including sharing data analytics related information, creating collaboration networks between several participants, and the use of video-conferencing in different application areas. The book provides a well-standing forum to discuss the characteristics of the intelligent speech signal processing systems in different domains. The book is proposed for professionals, scientists, and engineers who are involved in new techniques of intelligent speech signal processing methods and systems. It provides an outstanding foundation for undergraduate and post-graduate students as well. - Includes basics of speech data analysis and management tools with several applications, highlighting recording systems - Covers different techniques of big data and Internet-of-Things in speech signal processing, including machine learning and data mining - Offers a multidisciplinary view of current and future challenges in this field, with extensive case studies on the design, implementation, development and management of intelligent systems, neural networks, and related machine learning techniques for speech signal processing

Download or read book Noise robust Speech Source Localization and Tracking Using Microphone Arrays for Smartphone assisted Hearing Aid Devices written by Anshuman Ganguly and published by . This book was released on 2018 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech Source Localization (SSL) (or Direction of Arrival estimation) is a powerful pre-processing tool that helps identify the direction of the talker of interest in a noisy environment using multiple fixed microphones (known as a Microphone Array). This information is very helpful to the speech-processing pipeline and can be utilized to improve the performance of the overall system. With recent advancements, smartphones now possess the requisite hardware and computational power to perform real-time SSL for different applications. In this work, we propose application-specific SSL algorithms for three types of microphone arrays and show their effectiveness for smartphone implementation under realistic background noise conditions. We evaluate our proposed approaches in several realistic noisy conditions and present object evaluations to demonstrate the effectiveness of the proposed methods. We also propose the real-time implementation of some of our methods on the latest smartphones and smartphone-assisted devices.

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 Advancements in Hardware firmware and Applications for CCi MOBILE a Cochlear Implant Research Platform written by Ria Ghosh and published by . This book was released on 2021 with total page 0 pages. Available in PDF, EPUB and Kindle. Book excerpt: Hearing impairment is a pervasive problem which occurs due to the detrimental damage caused to the inner ear. Assistive Hearing Devices such as Cochlear Implants (CIs) and Hearing Aids (HAs) are designed to restore hearing, personalize rehabilitation, and enrich the listening experience. Although signal processing and machine learning research has greatly improved audio processing, the rigid design requirements of commercial CI sound processors make it difficult to explore novel algorithms for research investigations and conduct longitudinal studies. This thesis presents the design, development, clinical evaluation, and applications of CCi-MOBILE, a computationally powerful signal processing testing platform built specifically for researchers in the CI/HA field along with implementing multiple additional features to the platform. This custom-made, portable research platform allows researchers to design and perform complex speech processing algorithm assessment offline and in real-time through user-friendly, software-mediated open-source tools with implants manufactured by Cochlear Corporation. The design includes a lightweight custom circuit board comprising of an on-board FPGA to be used in conjunction with a computing platform such as a PC/tablet/laptop/smartphone based on the requirement of CI/HA signal processing algorithms. The processing pipeline for CI and HA stimulation is discussed followed by results from an acute study with implant users’ speech intelligibility in quiet and noisy conditions. The platform supports testing of algorithms for unilateral, bilateral, and bimodal hearing impairment. A major obstruction to accurate source localization for bimodal and bilateral CI users is the distortion of interaural time and level difference cues (ITD and ILD), and limited ITD sensitivity. Various CI research interfaces developed by either academic or industry sponsored research teams support proposed signal processing and psychoacoustic investigations but have limited ability to efficiently validate bimodal and/or bilateral algorithms. To overcome such challenges; verification, and validation of the synchronized bilateral (electric-electric) and bimodal (electric-acoustic) outputs is performed, in an authenticated and efficient way, to support localization algorithmic and experimental investigations. It has been hypothesized that variable stimulation rate for exciting the electrode array can aid for better speech perception and increased spectral information. Hence, a new multi-rate implant strategy including time-varying stimulation rates has been proposed in this work. Lastly, expanding the capabilities of the platform to ensure long-term sustainability, a real-time data streaming link between the platform and a cloud-based data repository is established to enable remote-test facilities along with an algorithm implementation and testing in naturalistic environments. We discuss implementation feasibility, and hypothesized performance of these approaches individually, and collectively, on the perceptual benefit for researchers working towards the welfare of the hearing-impaired community.

Download or read book Speech Enhancement Modeling and Recognition Algorithms and Applications written by S. Ramakrishnan and published by BoD – Books on Demand. This book was released on 2012-03-14 with total page 154 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book on Speech Processing consists of seven chapters written by eminent researchers from Italy, Canada, India, Tunisia, Finland and The Netherlands. The chapters covers important fields in speech processing such as speech enhancement, noise cancellation, multi resolution spectral analysis, voice conversion, speech recognition and emotion recognition from speech. The chapters contain both survey and original research materials in addition to applications. This book will be useful to graduate students, researchers and practicing engineers working in speech processing.

Download or read book Smartphone Based Real Time Digital Signal Processing written by Nasser Kehtarnavaz and published by Morgan & Claypool Publishers. This book was released on 2018-12-17 with total page 169 pages. Available in PDF, EPUB and Kindle. Book excerpt: Real-time or applied digital signal processing courses are offered as follow-ups to conventional or theory-oriented digital signal processing courses in many engineering programs for the purpose of teaching students the technical know-how for putting signal processing algorithms or theory into practical use. These courses normally involve access to a teaching laboratory that is equipped with hardware boards, in particular DSP boards, together with their supporting software. A number of textbooks have been written discussing how to achieve real-time implementation on these hardware boards. This book discusses how to use smartphones as hardware boards for real-time implementation of signal processing algorithms as an alternative to the hardware boards that are used in signal processing laboratory courses. The fact that mobile devices, in particular smartphones, have become powerful processing platforms led to the development of this book enabling students to use their own smartphones to run signal processing algorithms in real-time considering that these days nearly all students possess smartphones. Changing the hardware platforms that are currently used in applied or real-time signal processing courses to smartphones creates a truly mobile laboratory experience or environment for students. In addition, it relieves the cost burden associated with using dedicated signal processing boards noting that the software development tools for smartphones are free of charge and are well-maintained by smartphone manufacturers. This book is written in such a way that it can be used as a textbook for real-time or applied digital signal processing courses offered at many universities. Ten lab experiments that are commonly encountered in such courses are covered in the book. This book is written primarily for those who are already familiar with signal processing concepts and are interested in their real-time and practical aspects. Similar to existing real-time courses, knowledge of C programming is assumed. This book can also be used as a self-study guide for those who wish to become familiar with signal processing app development on either Android or iPhone smartphones.

Download or read book Metaheuristic Applications to Speech Enhancement written by Prajna Kunche and published by Springer. This book was released on 2016-04-12 with total page 126 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book serves as a basic reference for those interested in the application of metaheuristics to speech enhancement. The major goal of the book is to explain the basic concepts of optimization methods and their use in heuristic optimization in speech enhancement to scientists, practicing engineers, and academic researchers in speech processing. The authors discuss why it has been a challenging problem for researchers to develop new enhancement algorithms that aid in the quality and intelligibility of degraded speech. They present powerful optimization methods to speech enhancement that can help to solve the noise reduction problems. Readers will be able to understand the fundamentals of speech processing as well as the optimization techniques, how the speech enhancement algorithms are implemented by utilizing optimization methods, and will be given the tools to develop new algorithms. The authors also provide a comprehensive literature survey regarding the topic.

Download or read book Hearing Health Care for Adults written by National Academies of Sciences, Engineering, and Medicine and published by National Academies Press. This book was released on 2016-10-06 with total page 325 pages. Available in PDF, EPUB and Kindle. Book excerpt: The loss of hearing - be it gradual or acute, mild or severe, present since birth or acquired in older age - can have significant effects on one's communication abilities, quality of life, social participation, and health. Despite this, many people with hearing loss do not seek or receive hearing health care. The reasons are numerous, complex, and often interconnected. For some, hearing health care is not affordable. For others, the appropriate services are difficult to access, or individuals do not know how or where to access them. Others may not want to deal with the stigma that they and society may associate with needing hearing health care and obtaining that care. Still others do not recognize they need hearing health care, as hearing loss is an invisible health condition that often worsens gradually over time. In the United States, an estimated 30 million individuals (12.7 percent of Americans ages 12 years or older) have hearing loss. Globally, hearing loss has been identified as the fifth leading cause of years lived with disability. Successful hearing health care enables individuals with hearing loss to have the freedom to communicate in their environments in ways that are culturally appropriate and that preserve their dignity and function. Hearing Health Care for Adults focuses on improving the accessibility and affordability of hearing health care for adults of all ages. This study examines the hearing health care system, with a focus on non-surgical technologies and services, and offers recommendations for improving access to, the affordability of, and the quality of hearing health care for adults of all ages.

Download or read book Advances in Speech Recognition written by Amy Neustein and published by Springer Science & Business Media. This book was released on 2010-09-21 with total page 383 pages. Available in PDF, EPUB and Kindle. Book excerpt: Two Top Industry Leaders Speak Out Judith Markowitz When Amy asked me to co-author the foreword to her new book on advances in speech recognition, I was honored. Amy’s work has always been infused with c- ative intensity, so I knew the book would be as interesting for established speech professionals as for readers new to the speech-processing industry. The fact that I would be writing the foreward with Bill Scholz made the job even more enjoyable. Bill and I have known each other since he was at UNISYS directing projects that had a profound impact on speech-recognition tools and applications. Bill Scholz The opportunity to prepare this foreword with Judith provides me with a rare oppor- nity to collaborate with a seasoned speech professional to identify numerous signi- cant contributions to the field offered by the contributors whom Amy has recruited. Judith and I have had our eyes opened by the ideas and analyses offered by this collection of authors. Speech recognition no longer needs be relegated to the ca- gory of an experimental future technology; it is here today with sufficient capability to address the most challenging of tasks. And the point-click-type approach to GUI control is no longer sufficient, especially in the context of limitations of mode- day hand held devices. Instead, VUI and GUI are being integrated into unified multimodal solutions that are maturing into the fundamental paradigm for comput- human interaction in the future.

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 Advances in Digital Speech Transmission written by Prof Rainer Martin and published by John Wiley & Sons. This book was released on 2008-02-28 with total page 572 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech processing and speech transmission technology are expanding fields of active research. New challenges arise from the 'anywhere, anytime' paradigm of mobile communications, the ubiquitous use of voice communication systems in noisy environments and the convergence of communication networks toward Internet based transmission protocols, such as Voice over IP. As a consequence, new speech coding, new enhancement and error concealment, and new quality assessment methods are emerging. Advances in Digital Speech Transmission provides an up-to-date overview of the field, including topics such as speech coding in heterogeneous communication networks, wideband coding, and the quality assessment of wideband speech. Provides an insight into the latest developments in speech processing and speech transmission, making it an essential reference to those working in these fields Offers a balanced overview of technology and applications Discusses topics such as speech coding in heterogeneous communications networks, wideband coding, and the quality assessment of the wideband speech Explains speech signal processing in hearing instruments and man-machine interfaces from applications point of view Covers speech coding for Voice over IP, blind source separation, digital hearing aids and speech processing for automatic speech recognition Advances in Digital Speech Transmission serves as an essential link between the basics and the type of technology and applications (prospective) engineers work on in industry labs and academia. The book will also be of interest to advanced students, researchers, and other professionals who need to brush up their knowledge in this field.

Download or read book Recent Advances in Robust Speech Recognition Technology written by Javier Ramirez and published by Bentham Science. This book was released on 2011 with total page 223 pages. Available in PDF, EPUB and Kindle. Book excerpt: "This E-book is a collection of articles that describe advances in speech recognition technology. Robustness in speech recognition refers to the need to maintain high speech recognition accuracy even when the quality of the input speech is degraded, or whe"