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Real-time Speech Processing Algorithms for Smartphone Based Hearing Aid Applications

Real-time Speech Processing Algorithms for Smartphone Based Hearing Aid Applications
Author: Gautam Shreedhar Bhat
Publisher:
Total Pages:
Release: 2021
Genre: Hearing aids
ISBN:

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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.


Smartphone-Based Real-Time Digital Signal Processing

Smartphone-Based Real-Time Digital Signal Processing
Author: Nasser Kehtarnavaz
Publisher: Morgan & Claypool Publishers
Total Pages: 169
Release: 2018-12-17
Genre: Technology & Engineering
ISBN: 1681734664

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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.


Smartphone-based Single and Dual Microphone Speech Enhancement Algorithms for Hearing Study

Smartphone-based Single and Dual Microphone Speech Enhancement Algorithms for Hearing Study
Author: Gautam Shreedhar Bhat
Publisher:
Total Pages:
Release: 2018
Genre: Assistive listening systems
ISBN:

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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.


Proceedings of the International Conference on Paradigms of Computing, Communication and Data Sciences

Proceedings of the International Conference on Paradigms of Computing, Communication and Data Sciences
Author: Rajendra Prasad Yadav
Publisher: Springer Nature
Total Pages: 765
Release: 2023-02-23
Genre: Technology & Engineering
ISBN: 9811987424

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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.


Real-time Smartphone Apps Integrating Signal Processing Modules of Hearing Aids

Real-time Smartphone Apps Integrating Signal Processing Modules of Hearing Aids
Author: Tahsin Ahmed Chowdhury
Publisher:
Total Pages:
Release: 2018
Genre: Hearing aids
ISBN:

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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.


Intelligent Speech Signal Processing

Intelligent Speech Signal Processing
Author: Nilanjan Dey
Publisher: Academic Press
Total Pages: 210
Release: 2019-06-15
Genre: Technology & Engineering
ISBN: 0128181303

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Intelligent Speech Signal Processing investigates the utilization of speech analytics across several systems and real-world activities, including sharing data analytics related information, creating collaboration networks between several participants, and implementing video-conferencing in different application areas. It provides a forum for readers to discover the characteristics of intelligent speech signal processing systems across different domains. Chapters focus on the latest applications of speech data analysis and management tools across different recording systems. The book emphasizes the multi-disciplinary nature of the field, presenting different applications and challenges with extensive studies on the design, implementation, development, and management of intelligent systems, neural networks, and related machine learning techniques for speech signal processing. Highlights different data analytics techniques in speech signal processing, including machine learning, and data mining Illustrates different applications and challenges across the design, implementation, and management of intelligent systems and neural networks techniques for speech signal processing Includes coverage of biomodal speech recognition, voice activity detection, spoken language and speech disorder identification, automatic speech to speech summarization, and convolutional neural networks


Noise-robust Speech Source Localization and Tracking Using Microphone Arrays for Smartphone-assisted Hearing Aid Devices

Noise-robust Speech Source Localization and Tracking Using Microphone Arrays for Smartphone-assisted Hearing Aid Devices
Author: Anshuman Ganguly
Publisher:
Total Pages:
Release: 2018
Genre: Hearing aids
ISBN:

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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.


User Customizable Real-time Single and Dual Microphone Speech Enhancement and Blind Speech Separation for Smartphone Hearing Aid Applications

User Customizable Real-time Single and Dual Microphone Speech Enhancement and Blind Speech Separation for Smartphone Hearing Aid Applications
Author: Chandan Karadagur Ananda Reddy
Publisher:
Total Pages:
Release: 2018
Genre: Blind source separation
ISBN:

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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.


Advancements in Hardware/firmware and Applications for CCi-MOBILE: a Cochlear Implant Research Platform

Advancements in Hardware/firmware and Applications for CCi-MOBILE: a Cochlear Implant Research Platform
Author: Ria Ghosh
Publisher:
Total Pages: 0
Release: 2021
Genre: Cochlear implants
ISBN:

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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.


Speech Enhancement, Modeling and Recognition- Algorithms and Applications

Speech Enhancement, Modeling and Recognition- Algorithms and Applications
Author: S. Ramakrishnan
Publisher: BoD – Books on Demand
Total Pages: 154
Release: 2012-03-14
Genre: Computers
ISBN: 9535102915

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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.