Acoustic Field Analysis In Small Microphone Arrays
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Acoustic Field Analysis In Small Microphone Arrays
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Author : Roman Scharrer
language : en
Publisher: Logos Verlag Berlin GmbH
Release Date : 2013
Acoustic Field Analysis In Small Microphone Arrays written by Roman Scharrer and has been published by Logos Verlag Berlin GmbH this book supported file pdf, txt, epub, kindle and other format this book has been release on 2013 with Science categories.
In this work, the possibilities of an acoustic field analysis in small microphone arrays are investigated. With the increased use of mobile communication devices, such as smartphones and hearing aids, and the increase in the number of microphones in such devices, multi-channel signal processing has gained popularity. Apart from the definite signal processing, this thesis evaluates what information on the acoustic sound field and environment can be gained from the signal of such small microphone arrays. For this purpose, an innovative sound field classification was developed that determines the energies of the single sound field components. The method is based on spatial coherences of two or more acoustical. The method was successfully verified with a set of simulated and measured input signals. An adaptive automatic sensor mismatch compensation was created, which proved able to fully compensate any slow sensor drift after an initial training. Further, a new method for the blind estimation of the reverberation time based on the dependency of the coherence estimate on the evaluation parameters was proposed. The method determines the reverberation time of a room from the spatial coherence between two or more acoustic sensors.
Theory And Applications Of Spherical Microphone Array Processing
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Author : Daniel P. Jarrett
language : en
Publisher: Springer
Release Date : 2016-08-26
Theory And Applications Of Spherical Microphone Array Processing written by Daniel P. Jarrett and has been published by Springer this book supported file pdf, txt, epub, kindle and other format this book has been release on 2016-08-26 with Technology & Engineering categories.
This book presents the signal processing algorithms that have been developed to process the signals acquired by a spherical microphone array. Spherical microphone arrays can be used to capture the sound field in three dimensions and have received significant interest from researchers and audio engineers. Algorithms for spherical array processing are different to corresponding algorithms already known in the literature of linear and planar arrays because the spherical geometry can be exploited to great beneficial effect. The authors aim to advance the field of spherical array processing by helping those new to the field to study it efficiently and from a single source, as well as by offering a way for more experienced researchers and engineers to consolidate their understanding, adding either or both of breadth and depth. The level of the presentation corresponds to graduate studies at MSc and PhD level. This book begins with a presentation of some of the essential mathematical and physical theory relevant to spherical microphone arrays, and of an acoustic impulse response simulation method, which can be used to comprehensively evaluate spherical array processing algorithms in reverberant environments. The chapter on acoustic parameter estimation describes the way in which useful descriptions of acoustic scenes can be parameterized, and the signal processing algorithms that can be used to estimate the parameter values using spherical microphone arrays. Subsequent chapters exploit these parameters including in particular measures of direction-of-arrival and of diffuseness of a sound field. The array processing algorithms are then classified into two main classes, each described in a separate chapter. These are signal-dependent and signal-independent beamforming algorithms. Although signal-dependent beamforming algorithms are in theory able to provide better performance compared to the signal-independent algorithms, they are currently rarely used in practice. The main reason for this is that the statistical information required by these algorithms is difficult to estimate. In a subsequent chapter it is shown how the estimated acoustic parameters can be used in the design of signal-dependent beamforming algorithms. This final step closes, at least in part, the gap between theory and practice.
Microphone Arrays
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Author : Michael Brandstein
language : en
Publisher: Springer Science & Business Media
Release Date : 2013-04-17
Microphone Arrays written by Michael Brandstein and has been published by Springer Science & Business Media this book supported file pdf, txt, epub, kindle and other format this book has been release on 2013-04-17 with Technology & Engineering categories.
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.
Acoustic Array Systems
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Author : Mingsian R. Bai
language : en
Publisher: John Wiley & Sons
Release Date : 2013-03-07
Acoustic Array Systems written by Mingsian R. Bai and has been published by John Wiley & Sons this book supported file pdf, txt, epub, kindle and other format this book has been release on 2013-03-07 with Science categories.
Presents a unified framework of far-field and near-field array techniques for noise source identification and sound field visualization, from theory to application. Acoustic Array Systems: Theory, Implementation, and Application provides an overview of microphone array technology with applications in noise source identification and sound field visualization. In the comprehensive treatment of microphone arrays, the topics covered include an introduction to the theory, far-field and near-field array signal processing algorithms, practical implementations, and common applications: vehicles, computing and communications equipment, compressors, fans, and household appliances, and hands-free speech. The author concludes with other emerging techniques and innovative algorithms. Encompasses theoretical background, implementation considerations and application know-how Shows how to tackle broader problems in signal processing, control, and transudcers Covers both farfield and nearfield techniques in a balanced way Introduces innovative algorithms including equivalent source imaging (NESI) and high-resolution nearfield arrays Selected code examples available for download for readers to practice on their own Presentation slides available for instructor use A valuable resource for Postgraduates and researchers in acoustics, noise control engineering, audio engineering, and signal processing.
Microphone Arrays
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Author : Jacob Benesty
language : en
Publisher: Springer Nature
Release Date : 2023-08-09
Microphone Arrays written by Jacob Benesty and has been published by Springer Nature this book supported file pdf, txt, epub, kindle and other format this book has been release on 2023-08-09 with Technology & Engineering categories.
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.
Microphone Array Signal Processing
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Author : Jacob Benesty
language : en
Publisher: Springer Science & Business Media
Release Date : 2008-03-11
Microphone Array Signal Processing written by Jacob Benesty and has been published by Springer Science & Business Media this book supported file pdf, txt, epub, kindle and other format this book has been release on 2008-03-11 with Technology & Engineering categories.
In the past few years we have written and edited several books in the area of acousticandspeechsignalprocessing. Thereasonbehindthisendeavoristhat there were almost no books available in the literature when we ?rst started while there was (and still is) a real need to publish manuscripts summarizing the most useful ideas, concepts, results, and state-of-the-art algorithms in this important area of research. According to all the feedback we have received so far, we can say that we were right in doing this. Recently, several other researchers have followed us in this journey and have published interesting books with their own visions and perspectives. The idea of writing a book on Microphone Array Signal Processing comes from discussions we have had with many colleagues and friends. As a c- sequence of these discussions, we came up with the conclusion that, again, there is an urgent need for a monograph that carefully explains the theory and implementation of microphone arrays. While there are many manuscripts on antenna arrays from a narrowband perspective (narrowband signals and narrowband processing), the literature is quite scarce when it comes to s- sor arrays explained from a truly broadband perspective. Many algorithms for speech applications were simply borrowed from narrowband antenna - rays. However, a direct application of narrowband ideas to broadband speech processing may not be necessarily appropriate and can lead to many m- understandings.
Study And Design Of Differential Microphone Arrays
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Author : Jacob Benesty
language : en
Publisher: Springer Science & Business Media
Release Date : 2012-10-23
Study And Design Of Differential Microphone Arrays written by Jacob Benesty and has been published by Springer Science & Business Media this book supported file pdf, txt, epub, kindle and other format this book has been release on 2012-10-23 with Technology & Engineering categories.
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.
Anthropometric Individualization Of Head Related Transfer Functions Analysis And Modeling
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Author : Ramona Bomhardt
language : en
Publisher: Logos Verlag Berlin GmbH
Release Date : 2017-09-04
Anthropometric Individualization Of Head Related Transfer Functions Analysis And Modeling written by Ramona Bomhardt and has been published by Logos Verlag Berlin GmbH this book supported file pdf, txt, epub, kindle and other format this book has been release on 2017-09-04 with Science categories.
Human sound localization helps to pay attention to spatially separated speakers using interaural level and time differences as well as angle-dependent monaural spectral cues. In a monophonic teleconference, for instance, it is much more difficult to distinguish between different speakers due to missing binaural cues. Spatial positioning of the speakers by means of binaural reproduction methods using head-related transfer functions (HRTFs) enhances speech comprehension. These HRTFs are influenced by the torso, head and ear geometry as they describe the propagation path of the sound from a source to the ear canal entrance. Through this geometry-dependency, the HRTF is directional and subject-dependent. To enable a sufficient reproduction, individual HRTFs should be used. However, it is tremendously difficult to measure these HRTFs. For this reason this thesis proposes approaches to adapt the HRTFs applying individual anthropometric dimensions of a user. Since localization at low frequencies is mainly influenced by the interaural time difference, two models to adapt this difference are developed and compared with existing models. Furthermore, two approaches to adapt the spectral cues at higher frequencies are studied, improved and compared. Although the localization performance with individualized HRTFs is slightly worse than with individual HRTFs, it is nevertheless still better than with non-individual HRTFs, taking into account the measurement effort.
Modeling The Radiation Of Modern Sound Reinforcement Systems In High Resolution
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Author : Stefan Feistel
language : en
Publisher: Logos Verlag Berlin GmbH
Release Date : 2014
Modeling The Radiation Of Modern Sound Reinforcement Systems In High Resolution written by Stefan Feistel and has been published by Logos Verlag Berlin GmbH this book supported file pdf, txt, epub, kindle and other format this book has been release on 2014 with Computers categories.
Starting from physical theory, this work develops a novel framework for the acoustic simulation of sound radiation by loudspeakers and sound reinforcement systems. First, a theoretical foundation is derived for the accurate description of simple and multi-way loudspeakers using an advanced point-source ''CDPS'' model that incorporates phase data. The model's practical implementation is presented including measurement requirements and the GLL loudspeaker data format specification. In the second part, larger systems are analyzed such as line arrays where the receiver may be located in the near field of the source. It is shown that any extended line source can be modeled accurately after decomposition into smaller CDPS elements. The influence of production variation among elements of an array is investigated and shown to be small. The last part of this work deals with the consequences of fluctuating environmental conditions such as wind and temperature on the coherence of sound signals from multiple sources. A new theoretical model is developed that allows predicting the smooth transition from amplitude to power summation as a function of the statistical properties of the environmental parameters. A part of this work was distinguished with the AES Publications Award 2010. Parts of the proposed data format have been incorporated into the international AES56 standard.
Structure Borne Sound Sources In Buildings
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Author : Matthias Lievens
language : en
Publisher: Logos Verlag Berlin GmbH
Release Date : 2013
Structure Borne Sound Sources In Buildings written by Matthias Lievens and has been published by Logos Verlag Berlin GmbH this book supported file pdf, txt, epub, kindle and other format this book has been release on 2013 with Architecture categories.
Structure-borne sound sources are vibrational sources connected in some way to the building structure. The mechanical excitation of the building structure leads to sound radiation. This is an important source of annoyance in modern light-weight buildings. The prediction of the sound pressure level from structure-borne sound sources is highly complicated because of the complexity involved in the coupling between source and receiver structure. The current standard on characterisation of service equipment in buildings EN 12354-5, can deal with sources on heavy structures (high-mobility source) but to date, there is no engineering method available for the case of coupling between source and receiver. A case study of a washing machine on a wooden joist floor is investigated in this thesis. In the first part, measurements in the coupled state are conducted. It is shown that the normal components are sufficient to predict the sound pressure level. However, this only applies to the coupled state. In the second part, a true prediction is calculated from independently measured source and receiver quantities. The difference between predicted and directly measured sound pressure level leads to considerable errors of up to 20 dB at low frequencies. This shows that the normal components are not sufficient to predict the coupling between a washing machine and a wooden floor.