Noise Robust Speech Source Localization And Tracking Using Microphone Arrays For Smartphone Assisted Hearing Aid Devices PDF Download

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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.


Smartphone Based Multi-channel Dynamic-range Compression for Hearing Aid Research and Noise-robust Speech Source Localization Using Microphone Arrays

Smartphone Based Multi-channel Dynamic-range Compression for Hearing Aid Research and Noise-robust Speech Source Localization Using Microphone Arrays
Author: Yiya Hao
Publisher:
Total Pages:
Release: 2019
Genre: Compression (Audiology)
ISBN:

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Dynamic-range compression (DRC) is one technique of audio signal processing which maps the wide dynamic range of audio into the narrow range. In most of hearing aid devices (HADs), DRC plays as an important role to help hearing impaired people to listen to the speech with better quality and intelligibility. Compared to single-channel DRC, multi-channel DRC additionally improves the audio quality and intelligibility by setting appropriate adjustment parameters for every frequency band. Since most HADs are costly and unaffordable, fitting multi-channel DRC into smartphones becomes an outstanding solution since most people own smartphones. In this dissertation, three types of multi-channel DRCs has been proposed including their real-time implementations on smartphones. Speech Source Localization (SSL) (or Direction of Arrival (DOA) estimation) is another topic in this dissertation, which identifies the direction of the talker of interest in a noisy environment using multiple fixed microphones (known as a Microphone Array). This Proposal mainly covers only DRC, DOA and real-time implementation.


Microphone Arrays

Microphone Arrays
Author: Jacob Benesty
Publisher: Springer Nature
Total Pages: 232
Release: 2023-08-09
Genre: Technology & Engineering
ISBN: 3031369742

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This book explains the motivation for using microphone arrays as opposed to using a single sensor for sound acquisition. The book then goes on to summarize the most useful ideas, concepts, results, and new algorithms therein. The material presented in this work includes analysis of the advantages of using microphone arrays, including dimensionality reduction to remove the redundancy while preserving the variability of the array signals using the principal component analysis (PCA). The authors also discuss benefits such as beamforming with low-rank approximations, fixed, adaptive, and robust distortionless beamforming, differential beamforming, and a new form of binaural beamforming that takes advantage of both beamforming and human binaural hearing properties to improve speech intelligibility. The book makes the microphone array signal processing theory and applications available in a complete and self-contained text. The authors attempt to explain the main ideas in a clear and rigorous way so that the reader can easily capture the potentials, opportunities, challenges, and limitations of microphone array signal processing. This book is written for those who work on the topics of microphone arrays, noise reduction, speech enhancement, speech communication, and human-machine speech interfaces.


Audio Source Separation and Speech Enhancement

Audio Source Separation and Speech Enhancement
Author: Emmanuel Vincent
Publisher: John Wiley & Sons
Total Pages: 628
Release: 2018-07-24
Genre: Technology & Engineering
ISBN: 1119279917

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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.


Direction of Arrival Estimation and Localization of Multi-Speech Sources

Direction of Arrival Estimation and Localization of Multi-Speech Sources
Author: Nilanjan Dey
Publisher: Springer
Total Pages: 67
Release: 2017-12-23
Genre: Technology & Engineering
ISBN: 3319730592

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This book presents research and applications on arrival estimation and localization in speech processing to ensure that the broad vision of the direction of arrival estimation (DOAE) / localization of speech sources is well-established. The book first provides a brief overview of the most classical direction of arrival estimation and localization techniques. It then introduces the concept and model of acoustics sources and then highlights the most contemporary studies on this pervasive problem. In addition, the authors explore employing the optimization algorithms to improve the DOAE techniques. The book then highlights the concept and principles of the multi-DOAE approaches. Using a microphone array, the book introduces the localization and tracking problem of multiple speech/acoustic sources. It includes several applications and real-life speech sources localization based on the DOAE approaches. The book reports the challenges facing the DOAE techniques in speech-sources localization. The book pertains to researchers, designers, and engineers in speech processing fields.


Data-driven Multi-microphone Speaker Localization on Manifolds

Data-driven Multi-microphone Speaker Localization on Manifolds
Author: Bracha Laufter-Goldshtein
Publisher:
Total Pages: 161
Release: 2020
Genre: Acoustic localization
ISBN: 9781680837377

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Speech enhancement is a core problem in audio signal processing with commercial applications in devices as diverse as mobile phones, conference call systems, smart assistants, and hearing aids. An essential component in the design of speech enhancement algorithms is acoustic source localization. Speaker localization is also directly applicable to many other audio related tasks, e.g., automated camera steering, teleconferencing systems, and robot audition. From a signal processing perspective, speaker localization is the task of mapping multichannel speech signals to 3-D source coordinates. To obtain viable solutions for this mapping, an accurate description of the source wave propagation captured by the respective acoustic channel is required. In fact, the acoustic channels can be considered as the spatial fingerprints characterizing the positions of each of the sources in a reverberant enclosure. These fingerprints represent complex reflection patterns stemming from the surfaces and objects characterizing the enclosure. Hence, they are usually modelled by a very large number of coefficients, resulting in an intricate high-dimensional representation. We claim that in static acoustic environments, despite the high dimensional representation, the difference between acoustic channels can be attributed mainly to changes in the source position. Thus, the true intrinsic dimensionality of the variations of the acoustic channels are significantly smaller than the number of variables commonly used to represent them; that is, the acoustic channels pertain to a low-dimensional manifold that can be inferred from data using nonlinear dimensionality reduction techniques. A comprehensive experimental study carried out in a real-life acoustic environment demonstrates the validity of the proposed manifold-based paradigm. Motivated by this result, several high-performance localization and tracking methods were developed by harnessing novel mathematical tools for learning over manifolds, including diffusion maps, semi-supervised learning, optimization in reproducing kernel Hilbert spaces and Gaussian process inference. We present two localization algorithms that were designed for a single microphone array of two microphones. These algorithms were extended to several distributed arrays by merging the information of the different manifolds associated with each array. Tracking a moving source was also addressed by a data-driven propagation model relating movements on the abstract manifold to the actual source displacements. This data-driven propagation model was combined with a classical localization approach, in a hybrid algorithm that ties together the two worlds of classical and data-driven localization, while gaining the benefits of both. We show that the proposed algorithms outperform state-of-the-art localization methods, and obtain high accuracy in challenging noisy and reverberant environments.


Study and Design of Differential Microphone Arrays

Study and Design of Differential Microphone Arrays
Author: Jacob Benesty
Publisher: Springer
Total Pages: 0
Release: 2014-11-09
Genre: Technology & Engineering
ISBN: 9783642427565

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Microphone arrays have attracted a lot of interest over the last few decades since they have the potential to solve many important problems such as noise reduction/speech enhancement, source separation, dereverberation, spatial sound recording, and source localization/tracking, to name a few. However, the design and implementation of microphone arrays with beamforming algorithms is not a trivial task when it comes to processing broadband signals such as speech. Indeed, in most sensor arrangements, the beamformer output tends to have a frequency-dependent response. One exception, perhaps, is the family of differential microphone arrays (DMAs) who have the promise to form frequency-independent responses. Moreover, they have the potential to attain high directional gains with small and compact apertures. As a result, this type of microphone arrays has drawn much research and development attention recently. This book is intended to provide a systematic study of DMAs from a signal processing perspective. The primary objective is to develop a rigorous but yet simple theory for the design, implementation, and performance analysis of DMAs. The theory includes some signal processing techniques for the design of commonly used first-order, second-order, third-order, and also the general Nth-order DMAs. For each order, particular examples are given on how to form standard directional patterns such as the dipole, cardioid, supercardioid, hypercardioid, subcardioid, and quadrupole. The study demonstrates the performance of the different order DMAs in terms of beampattern, directivity factor, white noise gain, and gain for point sources. The inherent relationship between differential processing and adaptive beamforming is discussed, which provides a better understanding of DMAs and why they can achieve high directional gain. Finally, we show how to design DMAs that can be robust against white noise amplification.


Microphone Arrays

Microphone Arrays
Author: Michael Brandstein
Publisher: Springer Science & Business Media
Total Pages: 424
Release: 2001-05-02
Genre: Science
ISBN: 9783540419532

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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.


Microphone Array for Speech Processing and Recognition

Microphone Array for Speech Processing and Recognition
Author: Boopathy Prakasam
Publisher:
Total Pages: 164
Release: 2015
Genre:
ISBN:

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The performance of any sound signal processing system will be decreased when the distance between microphone and the sound source increases. While capturing the sound, various effects such as overlapping of speech from other speaker, background noise and room reverberation cause distortion in the interested sound signal quality. There are number of novel techniques developed in the area of sound signal processing to mitigate adverse effect of noise, room reverberation, and also possible to separate speech signal from the other overlapping sounds. One of the technique is, using "Microphone Array" instead of single microphone will provide a high-quality signal for the desired sound based on the sound source location. It has several advantages such as attenuating other interfering signals and ambient noises, localization and tracking of active talkers in the conference meetings, reduces the demand of hand-held or head-mounted microphone, capability to provide noise robustness, handsfree signal acquisition and the ability to capture distant speech signal which will improves the performance of Automatic Speech Recognition systems. Basically it is an array of multiple microphones placed at different spatial locations, used to capture sound signal by its propagation principle. Delay and Sum Beamforming algorithm is used here to increase the sensitivity of the required direction signal in the presence of corrupting noise sources.


Audio Signal Processing for Next-Generation Multimedia Communication Systems

Audio Signal Processing for Next-Generation Multimedia Communication Systems
Author: Yiteng (Arden) Huang
Publisher: Springer Science & Business Media
Total Pages: 375
Release: 2007-05-08
Genre: Technology & Engineering
ISBN: 1402077696

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Audio Signal Processing for Next-Generation Multimedia Communication Systems presents cutting-edge digital signal processing theory and implementation techniques for problems including speech acquisition and enhancement using microphone arrays, new adaptive filtering algorithms, multichannel acoustic echo cancellation, sound source tracking and separation, audio coding, and realistic sound stage reproduction. This book's focus is almost exclusively on the processing, transmission, and presentation of audio and acoustic signals in multimedia communications for telecollaboration where immersive acoustics will play a great role in the near future.