Download or read book Modal Array Signal Processing Principles and Applications of Acoustic Wavefield Decomposition written by Heinz Teutsch and published by Springer Science & Business Media. This book was released on 2007-01-10 with total page 267 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book deals with the problem of detecting and localizing multiple simultaneously active wideband acoustic sources by applying the notion of wavefield decomposition using circular and spherical microphone arrays. A rigorous derivation of modal array signal processing algorithms for unambiguous source detection and localization, as well as performance evaluations by means of measurements using an actual real-time capable implementation, are discussed.
Download or read book Acoustic Array Systems written by Mingsian R. Bai and published by John Wiley & Sons. This book was released on 2013-03-07 with total page 546 pages. Available in PDF, EPUB and Kindle. Book excerpt: 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.
Download or read book Proceedings of the EAA Joint Symposium on Auralization and Ambisonics 2014 written by Weinzierl, Stefan and published by Universitätsverlag der TU Berlin. This book was released on 2014 with total page 200 pages. Available in PDF, EPUB and Kindle. Book excerpt: In consideration of the remarkable intensity of research in the field of Virtual Acoustics, including different areas such as sound field analysis and synthesis, spatial audio technologies, and room acoustical modeling and auralization, it seemed about time to organize a second international symposium following the model of the first EAA Auralization Symposium initiated in 2009 by the acoustics group of the former Helsinki University of Technology (now Aalto University). Additionally, research communities which are focused on different approaches to sound field synthesis such as Ambisonics or Wave Field Synthesis have, in the meantime, moved closer together by using increasingly consistent theoretical frameworks. Finally, the quality of virtual acoustic environments is often considered as a result of all processing stages mentioned above, increasing the need for discussions on consistent strategies for evaluation. Thus, it seemed appropriate to integrate two of the most relevant communities, i.e. to combine the 2nd International Auralization Symposium with the 5th International Symposium on Ambisonics and Spherical Acoustics. The Symposia on Ambisonics, initiated in 2009 by the Institute of Electronic Music and Acoustics of the University of Music and Performing Arts in Graz, were traditionally dedicated to problems of spherical sound field analysis and re-synthesis, strategies for the exchange of ambisonics-encoded audio material, and – more than other conferences in this area – the artistic application of spatial audio systems. This publication contains the official conference proceedings. It includes 29 manuscripts which have passed a 3-stage peer-review with a board of about 70 international reviewers involved in the process. Each contribution has already been published individually with a unique DOI on the DepositOnce digital repository of TU Berlin. Some conference contributions have been recommended for resubmission to Acta Acustica united with Acustica, to possibly appear in a Special Issue on Virtual Acoustics in late 2014. These are not published in this collection.
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 Handbook of Signal Processing Systems written by Shuvra S. Bhattacharyya and published by Springer Science & Business Media. This book was released on 2010-09-10 with total page 1099 pages. Available in PDF, EPUB and Kindle. Book excerpt: It gives me immense pleasure to introduce this timely handbook to the research/- velopment communities in the ?eld of signal processing systems (SPS). This is the ?rst of its kind and represents state-of-the-arts coverage of research in this ?eld. The driving force behind information technologies (IT) hinges critically upon the major advances in both component integration and system integration. The major breakthrough for the former is undoubtedly the invention of IC in the 50’s by Jack S. Kilby, the Nobel Prize Laureate in Physics 2000. In an integrated circuit, all components were made of the same semiconductor material. Beginning with the pocket calculator in 1964, there have been many increasingly complex applications followed. In fact, processing gates and memory storage on a chip have since then grown at an exponential rate, following Moore’s Law. (Moore himself admitted that Moore’s Law had turned out to be more accurate, longer lasting and deeper in impact than he ever imagined. ) With greater device integration, various signal processing systems have been realized for many killer IT applications. Further breakthroughs in computer sciences and Internet technologies have also catalyzed large-scale system integration. All these have led to today’s IT revolution which has profound impacts on our lifestyle and overall prospect of humanity. (It is hard to imagine life today without mobiles or Internets!) The success of SPS requires a well-concerted integrated approach from mul- ple disciplines, such as device, design, and application.
Download or read book Techniques for Noise Robustness in Automatic Speech Recognition written by Tuomas Virtanen and published by John Wiley & Sons. This book was released on 2012-09-19 with total page 514 pages. Available in PDF, EPUB and Kindle. Book excerpt: Automatic speech recognition (ASR) systems are finding increasing use in everyday life. Many of the commonplace environments where the systems are used are noisy, for example users calling up a voice search system from a busy cafeteria or a street. This can result in degraded speech recordings and adversely affect the performance of speech recognition systems. As the use of ASR systems increases, knowledge of the state-of-the-art in techniques to deal with such problems becomes critical to system and application engineers and researchers who work with or on ASR technologies. This book presents a comprehensive survey of the state-of-the-art in techniques used to improve the robustness of speech recognition systems to these degrading external influences. Key features: Reviews all the main noise robust ASR approaches, including signal separation, voice activity detection, robust feature extraction, model compensation and adaptation, missing data techniques and recognition of reverberant speech. Acts as a timely exposition of the topic in light of more widespread use in the future of ASR technology in challenging environments. Addresses robustness issues and signal degradation which are both key requirements for practitioners of ASR. Includes contributions from top ASR researchers from leading research units in the field
Download or read book Parametric Time Frequency Domain Spatial Audio written by Ville Pulkki and published by John Wiley & Sons. This book was released on 2017-10-04 with total page 412 pages. Available in PDF, EPUB and Kindle. Book excerpt: A comprehensive guide that addresses the theory and practice of spatial audio This book provides readers with the principles and best practices in spatial audio signal processing. It describes how sound fields and their perceptual attributes are captured and analyzed within the time-frequency domain, how essential representation parameters are coded, and how such signals are efficiently reproduced for practical applications. The book is split into four parts starting with an overview of the fundamentals. It then goes on to explain the reproduction of spatial sound before offering an examination of signal-dependent spatial filtering. The book finishes with coverage of both current and future applications and the direction that spatial audio research is heading in. Parametric Time-frequency Domain Spatial Audio focuses on applications in entertainment audio, including music, home cinema, and gaming—covering the capturing and reproduction of spatial sound as well as its generation, transduction, representation, transmission, and perception. This book will teach readers the tools needed for such processing, and provides an overview to existing research. It also shows recent up-to-date projects and commercial applications built on top of the systems. Provides an in-depth presentation of the principles, past developments, state-of-the-art methods, and future research directions of spatial audio technologies Includes contributions from leading researchers in the field Offers MATLAB codes with selected chapters An advanced book aimed at readers who are capable of digesting mathematical expressions about digital signal processing and sound field analysis, Parametric Time-frequency Domain Spatial Audio is best suited for researchers in academia and in the audio industry.
Download or read book Ambisonics written by Franz Zotter and published by Springer. This book was released on 2019-04-30 with total page 210 pages. Available in PDF, EPUB and Kindle. Book excerpt: This open access book provides a concise explanation of the fundamentals and background of the surround sound recording and playback technology Ambisonics. It equips readers with the psychoacoustical, signal processing, acoustical, and mathematical knowledge needed to understand the inner workings of modern processing utilities, special equipment for recording, manipulation, and reproduction in the higher-order Ambisonic format. The book comes with various practical examples based on free software tools and open scientific data for reproducible research. The book’s introductory section offers a perspective on Ambisonics spanning from the origins of coincident recordings in the 1930s to the Ambisonic concepts of the 1970s, as well as classical ways of applying Ambisonics in first-order coincident sound scene recording and reproduction that have been practiced since the 1980s. As, from time to time, the underlying mathematics become quite involved, but should be comprehensive without sacrificing readability, the book includes an extensive mathematical appendix. The book offers readers a deeper understanding of Ambisonic technologies, and will especially benefit scientists, audio-system and audio-recording engineers. In the advanced sections of the book, fundamentals and modern techniques as higher-order Ambisonic decoding, 3D audio effects, and higher-order recording are explained. Those techniques are shown to be suitable to supply audience areas ranging from studio-sized to hundreds of listeners, or headphone-based playback, regardless whether it is live, interactive, or studio-produced 3D audio material.
Download or read book Media Production Delivery and Interaction for Platform Independent Systems written by Oliver Schreer and published by John Wiley & Sons. This book was released on 2013-12-06 with total page 369 pages. Available in PDF, EPUB and Kindle. Book excerpt: Presents current trends and potential future developments by leading researchers in immersive media production, delivery, rendering and interaction The underlying audio and video processing technology that is discussed in the book relates to areas such as 3D object extraction, audio event detection; 3D sound rendering and face detection, gesture analysis and tracking using video and depth information. The book will give an insight into current trends and developments of future media production, delivery and reproduction. Consideration of the complete production, processing and distribution chain will allow for a full picture to be presented to the reader. Production developments covered will include integrated workflows developed by researchers and industry practitioners as well as capture of ultra-high resolution panoramic video and 3D object based audio across a range of programme genres. Distribution developments will include script based format agnostic network delivery to a full range of devices from large scale public panoramic displays with wavefield synthesis and ambisonic audio reproduction to ’small screen’ mobile devices. Key developments at the consumer end of the chain apply to both passive and interactive viewing modes and will incorporate user interfaces such as gesture recognition and ‘second screen’ devices to allow manipulation of the audio visual content. Presents current trends and potential future developments by leading researchers in immersive media production, delivery, rendering and interaction. Considers the complete production, processing and distribution chain illustrating the dependencies and the relationship between different components. Proposes that a format-agnostic approach to the production and delivery of broadcast programmes will overcome the problems faced with the steadily growing number of production and delivery formats. Explains the fundamentals of media production in addition to the complete production chain, beyond current-state-of-the-art through to presenting novel approaches and technologies for future media production. Focuses on the technologies that will allow for the realization of an E2E media platform that supports flexible content representations and interactivity for users. An essential read for Researchers and developers of audio-visual technology in industry and academia, such as engineers in broadcast technology companies and students working toward a career in the rapidly changing area of broadcast both from a production and an engineering perspective.
Download or read book Multidimensional Signals and Systems written by Rudolf Rabenstein and published by Springer Nature. This book was released on 2023-07-24 with total page 514 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book covers the theory of multidimensional signals and systems and related practical aspects. It extends the properties and mathematical tools of one-dimensional signals and systems to multiple dimensions and covers relevant timeless topics including multidimensional transformations, multidimensional sampling as well as discrete multidimensional systems. A special emphasis is placed on physical systems described by partial differential equations, the construction of suitable integral transformations and the implementation of the corresponding discrete-time algorithms. To this end, signal spaces and functional transformations are introduced at a mathematical level provided by undergraduate programs in engineering and science. The presentation takes a comprehensive, illustrative and educational approach without reference to a particular application field. Instead, the book builds a solid theoretical concept of multidimensional signals and systems and shows the application to various problems relevant for practical scenarios.
Download or read book Psychoacoustic Music Sound Field Synthesis written by Tim Ziemer and published by Springer. This book was released on 2019-08-06 with total page 287 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book provides a broad overview of spaciousness in music theory, from mixing and performance practice, to room acoustics, psychoacoustics and audio engineering, and presents the derivation, implementation and experimental validation of a novel type of spatial audio system. Discussing the physics of musical instruments and the nature of auditory perception, the book enables readers to precisely localize synthesized musical instruments while experiencing their timbral variance and spatial breadth. Offering interdisciplinary insights for novice music enthusiasts and experts in the field of spatial audio, this book is suitable for anyone interested in the study of music and musicology and the application of spatial audio mixing, or those seeking an overview of the state of the art in applied psychoacoustics for spatial audio.
Download or read book The Journal of the Acoustical Society of America written by Acoustical Society of America and published by . This book was released on 2006 with total page 1114 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Wavefield Decomposition Using Microphone Arrays and Its Application to Acoustic Scene Analysis written by Heinz Teutsch and published by . This book was released on 2006 with total page 262 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Mathematical Reviews written by and published by . This book was released on 2007 with total page 984 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Download or read book Array Signal Processing Algorithms for Localization and Equalization in Complex Acoustic Channels written by Dumidu Talagala and published by . This book was released on 2013 with total page 310 pages. Available in PDF, EPUB and Kindle. Book excerpt: The reproduction of realistic soundscapes in consumer electronic applications has been a driving force behind the development of spatial audio signal processing techniques. In order to accurately reproduce or decompose a particular spatial sound field, being able to exploit or estimate the effects of the acoustic environment becomes essential. This requires both an understanding of the source of the complexity in the acoustic channel (the acoustic path between a source and a receiver) and the ability to characterize its spatial attributes. In this thesis, we explore how to exploit or overcome the effects of the acoustic channel for sound source localization and sound field reproduction. The behaviour of a typical acoustic channel can be visualized as a transformation of its free field behaviour, due to scattering and reflections off the measurement apparatus and the surfaces in a room. These spatial effects can be modelled using the solutions to the acoustic wave equation, yet the physical nature of these scatterers typically results in complex behaviour with frequency. The first half of this thesis explores how to exploit this diversity in the frequency-domain for sound source localization, a concept that has not been considered previously. We first extract down-converted subband signals from the broadband audio signal, and collate these signals, such that the spatial diversity is retained. A signal model is then developed to exploit the channel's spatial information using a signal subspace approach. We show that this concept can be applied to multi-sensor arrays on complex-shaped rigid bodies as well as the special case of binaural localization. In both cases, an improvement in the closely spaced source resolution is demonstrated over traditional techniques, through simulations and experiments using a KEMAR manikin. The binaural analysis further indicates that the human localization performance in certain spatial regions is limited by the lack of spatial diversity, as suggested in perceptual experiments in the literature. Finally, the possibility of exploiting known inter-subband correlated sources (e.g., speech) for localization in under-determined systems is demonstrated. The second half of this thesis considers reverberation control, where reverberation is modelled as a superposition of sound fields created by a number of spatially distributed sources. We consider the mode/wave-domain description of the sound field, and propose modelling the reverberant modes as linear transformations of the desired sound field modes. This is a novel concept, as we consider each mode transformation to be independent of other modes. This model is then extended to sound field control, and used to derive the compensation signals required at the loudspeakers to equalize the reverberation. We show that estimating the reverberant channel and controlling the sound field now becomes a single adaptive filtering problem in the mode-domain, where the modes can be adapted independently. The performance of the proposed method is compared with existing adaptive and non-adaptive sound field control techniques through simulations. Finally, it is shown that an order of magnitude reduction in the computational complexity can be achieved, while maintaining comparable performance to existing adaptive control techniques.
Download or read book Theory and Applications of Spherical Microphone Array Processing written by Daniel P. Jarrett and published by Springer. This book was released on 2016-09-02 with total page 187 pages. Available in PDF, EPUB and Kindle. Book excerpt: 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.
Download or read book Modal Based Solutions for the Acquisition and Rendering of Large Spatial Soundfields written by Prasanga Samarasinghe and published by . This book was released on 2015 with total page 0 pages. Available in PDF, EPUB and Kindle. Book excerpt: This thesis explores the design and analysis of acoustic signal processing algorithms for the acquisition and rendering of spatial soundfields over large regions. The principal function of spatial sound systems is to record and reproduce desired soundfields in real time. Although there exist numerous approaches to the above problem, the development of immersive audio over large spatial regions is impeded by several fundamental and technological limitations. Therefore, there is a growing interest in deriving new signal processing algorithms for the recreation of large acoustic events, along with a wide range of applications envisioned for the future. Our approach is to study the modal based solution to the classical wave equation, and exploit its intrinsic properties to develop new recording/reproduction techniques. Novel designs for microphone/loudspeaker arrays are derived where each unit records/reproduces lower order spatial sound samples of the larger soundfield of interest. These samples are collectively combined to generate the final outcome using a set of modal analysis tools referred to as the addition theorems. An efficient model of the underlying structure of reverberation is also incorporated to improve the fidelity of sound reproduction. Several outcomes resulting from this approach are: (i) derivation of a novel microphone array design for spatial sound recording with a substantial reduction in microphone numbers, (ii) derivation of a novel loudspeaker array design for spatial sound reproduction with a substantial reduction in loudspeaker numbers and, (iii) a new approach to acoustic equalization based upon an efficient parameterization of the room transfer function. Compared to the conventional approaches to sound recording, the proposed microphone design is shown to be more robust to measurement noise. Likewise, compared to the existing surround sound systems, the proposed loudspeaker design is shown to be highly advantageous with the added ability to cancel the outgoing soundfield. Application and performance of the above methods are illustrated using appropriate simulation examples.
