Proceedings of the EAA Spatial Audio Signal Processing symposium: SASP 2019

International audienc

Effizientes binaurales Rendering von virtuellen akustischen Realitäten : technische und wahrnehmungsbezogene Konzepte

Binaural rendering aims to immerse the listener in a virtual acoustic scene, making it an essential method for spatial audio reproduction in virtual or augmented reality (VR/AR) applications. The growing interest and research in VR/AR solutions yielded many different methods for the binaural rendering of virtual acoustic realities, yet all of them share the fundamental idea that the auditory experience of any sound field can be reproduced by reconstructing its sound pressure at the listener's eardrums. This thesis addresses various state-of-the-art methods for 3 or 6 degrees of freedom (DoF) binaural rendering, technical approaches applied in the context of headphone-based virtual acoustic realities, and recent technical and psychoacoustic research questions in the field of binaural technology. The publications collected in this dissertation focus on technical or perceptual concepts and methods for efficient binaural rendering, which has become increasingly important in research and development due to the rising popularity of mobile consumer VR/AR devices and applications. The thesis is organized into five research topics: Head-Related Transfer Function Processing and Interpolation, Parametric Spatial Audio, Auditory Distance Perception of Nearby Sound Sources, Binaural Rendering of Spherical Microphone Array Data, and Voice Directivity. The results of the studies included in this dissertation extend the current state of research in the respective research topic, answer specific psychoacoustic research questions and thereby yield a better understanding of basic spatial hearing processes, and provide concepts, methods, and design parameters for the future implementation of technically and perceptually efficient binaural rendering.Binaurales Rendering zielt darauf ab, dass der Hörer in eine virtuelle akustische Szene eintaucht, und ist somit eine wesentliche Methode für die räumliche Audiowiedergabe in Anwendungen der virtuellen Realität (VR) oder der erweiterten Realität (AR – aus dem Englischen Augmented Reality). Das wachsende Interesse und die zunehmende Forschung an VR/AR-Lösungen führte zu vielen verschiedenen Methoden für das binaurale Rendering virtueller akustischer Realitäten, die jedoch alle die grundlegende Idee teilen, dass das Hörerlebnis eines beliebigen Schallfeldes durch die Rekonstruktion seines Schalldrucks am Trommelfell des Hörers reproduziert werden kann. Diese Arbeit befasst sich mit verschiedenen modernsten Methoden zur binauralen Wiedergabe mit 3 oder 6 Freiheitsgraden (DoF – aus dem Englischen Degree of Freedom), mit technischen Ansätzen, die im Kontext kopfhörerbasierter virtueller akustischer Realitäten angewandt werden, und mit aktuellen technischen und psychoakustischen Forschungsfragen auf dem Gebiet der Binauraltechnik. Die in dieser Dissertation gesammelten Publikationen befassen sich mit technischen oder wahrnehmungsbezogenen Konzepten und Methoden für effizientes binaurales Rendering, was in der Forschung und Entwicklung aufgrund der zunehmenden Beliebtheit von mobilen Verbraucher-VR/AR-Geräten und -Anwendungen zunehmend an Relevanz gewonnen hat. Die Arbeit ist in fünf Forschungsthemen gegliedert: Verarbeitung und Interpolation von Außenohrübertragungsfunktionen, parametrisches räumliches Audio, auditive Entfernungswahrnehmung ohrnaher Schallquellen, binaurales Rendering von sphärischen Mikrofonarraydaten und Richtcharakteristik der Stimme. Die Ergebnisse der in dieser Dissertation enthaltenen Studien erweitern den aktuellen Forschungsstand im jeweiligen Forschungsfeld, beantworten spezifische psychoakustische Forschungsfragen und führen damit zu einem besseren Verständnis grundlegender räumlicher Hörprozesse, und liefern Konzepte, Methoden und Gestaltungsparameter für die zukünftige Umsetzung eines technisch und wahrnehmungsbezogen effizienten binauralen Renderings.BMBF, 03FH014IX5, Natürliche raumbezogene Darbietung selbsterzeugter Schallereignisse in virtuellen auditiven Umgebungen (NarDasS

Audio for Virtual, Augmented and Mixed Realities: Proceedings of ICSA 2019 ; 5th International Conference on Spatial Audio ; September 26th to 28th, 2019, Ilmenau, Germany

The ICSA 2019 focuses on a multidisciplinary bringing together of developers, scientists, users, and content creators of and for spatial audio systems and services. A special focus is on audio for so-called virtual, augmented, and mixed realities. The fields of ICSA 2019 are: - Development and scientific investigation of technical systems and services for spatial audio recording, processing and reproduction / - Creation of content for reproduction via spatial audio systems and services / - Use and application of spatial audio systems and content presentation services / - Media impact of content and spatial audio systems and services from the point of view of media science. The ICSA 2019 is organized by VDT and TU Ilmenau with support of Fraunhofer Institute for Digital Media Technology IDMT

Fast Numerical and Machine Learning Algorithms for Spatial Audio Reproduction

Audio reproduction technologies have underwent several revolutions from a purely mechanical, to electromagnetic, and into a digital process. These changes have resulted in steady improvements in the objective qualities of sound capture/playback on increasingly portable devices. However, most mobile playback devices remove important spatial-directional components of externalized sound which are natural to the subjective experience of human hearing. Fortunately, the missing spatial-directional parts can be integrated back into audio through a combination of computational methods and physical knowledge of how sound scatters off of the listener's anthropometry in the sound-field. The former employs signal processing techniques for rendering the sound-field. The latter employs approximations of the sound-field through the measurement of so-called Head-Related Impulse Responses/Transfer Functions (HRIRs/HRTFs). This dissertation develops several numerical and machine learning algorithms for accelerating and personalizing spatial audio reproduction in light of available mobile computing power. First, spatial audio synthesis between a sound-source and sound-field requires fast convolution algorithms between the audio-stream and the HRIRs. We introduce a novel sparse decomposition algorithm for HRIRs based on non-negative matrix factorization that allows for faster time-domain convolution than frequency-domain fast-Fourier-transform variants. Second, the full sound-field over the spherical coordinate domain must be efficiently approximated from a finite collection of HRTFs. We develop a joint spatial-frequency covariance model for Gaussian process regression (GPR) and sparse-GPR methods that supports the fast interpolation and data fusion of HRTFs across multiple data-sets. Third, the direct measurement of HRTFs requires specialized equipment that is unsuited for widespread acquisition. We ``bootstrap'' the human ability to localize sound in listening tests with Gaussian process active-learning techniques over graphical user interfaces that allows the listener to infer his/her own HRTFs. Experiments are conducted on publicly available HRTF datasets and human listeners

Proceedings of the EAA Joint Symposium on Auralization and Ambisonics 2014

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.European Acoustics Associatio

Head Related Transfer Function selection techniques applied to multimodal environments for spatial cognition

This thesis studies the pinna reflection model to extract the best possible HRTF for a user from a database. The performances of the selected HRTF are tested in a psychoacoustic experiment and compared with those recorded from KEMAR. This extraction increases the average performances in respect to the KEMAR. The selected HRTF is then integrated in a virtual multimodal environment together with the haptic device TAMO showing the relation and the benefit of audio in Orientation & Mobility task

PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS

Multichannel acoustic signal processing has undergone major development in recent years due to the increased complexity of current audio processing applications. People want to collaborate through communication with the feeling of being together and sharing the same environment, what is considered as Immersive Audio Schemes. In this phenomenon, several acoustic e ects are involved: 3D spatial sound, room compensation, crosstalk cancelation, sound source localization, among others. However, high computing capacity is required to achieve any of these e ects in a real large-scale system, what represents a considerable limitation for real-time applications. The increase of the computational capacity has been historically linked to the number of transistors in a chip. However, nowadays the improvements in the computational capacity are mainly given by increasing the number of processing units, i.e expanding parallelism in computing. This is the case of the Graphics Processing Units (GPUs), that own now thousands of computing cores. GPUs were traditionally related to graphic or image applications, but new releases in the GPU programming environments, CUDA or OpenCL, allowed that most applications were computationally accelerated in elds beyond graphics. This thesis aims to demonstrate that GPUs are totally valid tools to carry out audio applications that require high computational resources. To this end, di erent applications in the eld of audio processing are studied and performed using GPUs. This manuscript also analyzes and solves possible limitations in each GPU-based implementation both from the acoustic point of view as from the computational point of view. In this document, we have addressed the following problems: Most of audio applications are based on massive ltering. Thus, the rst implementation to undertake is a fundamental operation in the audio processing: the convolution. It has been rst developed as a computational kernel and afterwards used for an application that combines multiples convolutions concurrently: generalized crosstalk cancellation and equalization. The proposed implementation can successfully manage two di erent and common situations: size of bu ers that are much larger than the size of the lters and size of bu ers that are much smaller than the size of the lters. Two spatial audio applications that use the GPU as a co-processor have been developed from the massive multichannel ltering. First application deals with binaural audio. Its main feature is that this application is able to synthesize sound sources in spatial positions that are not included in the database of HRTF and to generate smoothly movements of sound sources. Both features were designed after di erent tests (objective and subjective). The performance regarding number of sound source that could be rendered in real time was assessed on GPUs with di erent GPU architectures. A similar performance is measured in a Wave Field Synthesis system (second spatial audio application) that is composed of 96 loudspeakers. The proposed GPU-based implementation is able to reduce the room e ects during the sound source rendering. A well-known approach for sound source localization in noisy and reverberant environments is also addressed on a multi-GPU system. This is the case of the Steered Response Power with Phase Transform (SRPPHAT) algorithm. Since localization accuracy can be improved by using high-resolution spatial grids and a high number of microphones, accurate acoustic localization systems require high computational power. The solutions implemented in this thesis are evaluated both from localization and from computational performance points of view, taking into account different acoustic environments, and always from a real-time implementation perspective. Finally, This manuscript addresses also massive multichannel ltering when the lters present an In nite Impulse Response (IIR). Two cases are analyzed in this manuscript: 1) IIR lters composed of multiple secondorder sections, and 2) IIR lters that presents an allpass response. Both cases are used to develop and accelerate two di erent applications: 1) to execute multiple Equalizations in a WFS system, and 2) to reduce the dynamic range in an audio signal.Belloch Rodríguez, JA. (2014). PERFORMANCE IMPROVEMENT OF MULTICHANNEL AUDIO BY GRAPHICS PROCESSING UNITS [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/40651TESISPremios Extraordinarios de tesis doctorale