Surround by Sound: A Review of Spatial Audio Recording and Reproduction

In this article, a systematic overview of various recording and reproduction techniques for spatial audio is presented. While binaural recording and rendering is designed to resemble the human two-ear auditory system and reproduce sounds specifically for a listener’s two ears, soundfield recording and reproduction using a large number of microphones and loudspeakers replicate an acoustic scene within a region. These two fundamentally different types of techniques are discussed in the paper. A recent popular area, multi-zone reproduction, is also briefly reviewed in the paper. The paper is concluded with a discussion of the current state of the field and open problemsThe authors acknowledge National Natural Science Foundation of China (NSFC) No. 61671380 and Australian Research Council Discovery Scheme DE 150100363

Microphone and Loudspeaker Array Signal Processing Steps towards a “Radiation Keyboard” for Authentic Samplers

To date electric pianos and samplers tend to concentrate on authenticity in terms of temporal and spectral aspects of sound. However, they barely recreate the original sound radiation characteristics, which contribute to the perception of width and depth, vividness and voice separation, especially for instrumentalists, who are located near the instrument. To achieve this, a number of sound field measurement and synthesis techniques need to be applied and adequately combined. In this paper we present the theoretic foundation to combine so far isolated and fragmented sound field analysis and synthesis methods to realize a radiation keyboard, an electric harpsichord that approximates the sound of a real harpsichord precisely in time, frequency, and space domain. Potential applications for such a radiation keyboard are conservation of historic musical instruments, music performance, and psychoacoustic measurements for instrument and synthesizer building and for studies of music perception, cognition, and embodiment

Äänikentän tila-analyysi parametrista tilaäänentoistoa varten käyttäen harvoja mikrofoniasetelmia

In spatial audio capturing the aim is to store information about the sound field so that the sound field can be reproduced without a perceptual difference to the original. The need for this is in applications like virtual reality and teleconferencing. Traditionally the sound field has been captured with a B-format microphone, but it is not always a feasible solution due to size and cost constraints. Alternatively, also arrays of omnidirectional microphones can be utilized and they are often used in devices like mobile phones. If the microphone array is sparse, i.e., the microphone spacings are relatively large, the analysis of the sound Direction of Arrival (DoA) becomes ambiguous in higher frequencies. This is due to spatial aliasing, which is a common problem in narrowband DoA estimation. In this thesis the spatial aliasing problem was examined and its effect on DoA estimation and spatial sound synthesis with Directional Audio Coding (DirAC) was studied. The aim was to find methods for unambiguous narrowband DoA estimation. The current State of the Art methods can remove aliased estimates but are not capable of estimating the DoA with the optimal Time-Frequency resolution. In this thesis similar results were obtained with parameter extrapolation when only a single broadband source exists. The main contribution of this thesis was the development of a correlation-based method. The developed method utilizes pre-known, array-specific information on aliasing in each DoA and frequency. The correlation-based method was tested and found to be the best option to overcome the problem of spatial aliasing. This method was able to resolve spatial aliasing even with multiple sources or when the source’s frequency content is completely above the spatial aliasing frequency. In a listening test it was found that the correlation-based method could provide a major improvement to the DirAC synthesized spatial image quality when compared to an aliased estimator.Tilaäänen tallentamisessa tavoitteena on tallentaa äänikentän ominaisuudet siten, että äänikenttä pystytään jälkikäteen syntetisoimaan ilman kuuloaistilla havaittavaa eroa alkuperäiseen. Tarve tälle löytyy erilaisista sovelluksista, kuten virtuaalitodellisuudesta ja telekonferensseista. Perinteisesti äänikentän ominaisuuksia on tallennettu B-formaatti mikrofonilla, jonka käyttö ei kuitenkaan aina ole koko- ja kustannussyistä mahdollista. Vaihtoehtoisesti voidaan käyttää myös pallokuvioisista mikrofoneista koostuvia mikrofoniasetelmia. Mikäli mikrofonien väliset etäisyydet ovat liian suuria, eli asetelma on harva, tulee äänen saapumissuunnan selvittämisestä epäselvää korkeammilla taajuuksilla. Tämä johtuu ilmiöstä nimeltä tilallinen laskostuminen. Tämän diplomityön tarkoituksena oli tutkia tilallisen laskostumisen ilmiötä, sen vaikutusta saapumissuunnan arviointiin sekä tilaäänisynteesiin Directional Audio Coding (DirAC) -menetelmällä. Lisäksi tutkittiin menetelmiä, joiden avulla äänen saapumissuunta voitaisiin selvittää oikein myös tilallisen laskostumisen läsnä ollessa. Työssä havaittiin, että nykyiset ratkaisut laskostumisongelmaan eivät kykene tuottamaan oikeita suunta-arvioita optimaalisella aikataajuusresoluutiolla. Tässä työssä samantapaisia tuloksia saatiin laajakaistaisen äänilähteen tapauksessa ekstrapoloimalla suunta-arvioita laskostumisen rajataajuuden alapuolelta. Työn pääosuus oli kehittää korrelaatioon perustuva saapumissuunnan arviointimenetelmä, joka kykenee tuottamaan luotettavia arvioita rajataajuuden yläpuolella ja useamman äänilähteen ympäristöissä. Kyseinen menetelmä hyödyntää mikrofoniasetelmalle ominaista, saapumissuunnasta ja taajuudesta riippuvaista laskostumiskuviota. Kuuntelukokeessa havaittiin, että korrelaatioon perustuva menetelmä voi tuoda huomattavan parannuksen syntetisoidun tilaäänikuvan laatuun verrattuna synteesiin laskostuneilla suunta-arvioilla

Measurement of head-related transfer functions : A review

A head-related transfer function (HRTF) describes an acoustic transfer function between a point sound source in the free-field and a defined position in the listener's ear canal, and plays an essential role in creating immersive virtual acoustic environments (VAEs) reproduced over headphones or loudspeakers. HRTFs are highly individual, and depend on directions and distances (near-field HRTFs). However, the measurement of high-density HRTF datasets is usually time-consuming, especially for human subjects. Over the years, various novel measurement setups and methods have been proposed for the fast acquisition of individual HRTFs while maintaining high measurement accuracy. This review paper provides an overview of various HRTF measurement systems and some insights into trends in individual HRTF measurements

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

International audienc

The plenacoustic function and its applications

This thesis is a study of the spatial evolution of the sound field. We first present an analysis of the sound field along different geometries. In the case of the sound field studied along a line in a room, we describe a two-dimensional function characterizing the sound field along space and time. Calculating the Fourier transform of this function leads to a spectrum having a butterfly shape. The spectrum is shown to be almost bandlimited along the spatial frequency dimension, which allows the interpolation of the sound field at any position along the line when a sufficient number of microphones is present. Using this Fourier representation of the sound field, we develop a spatial sampling theorem trading off quality of reconstruction with spatial sampling frequency. The study is generalized for planes of microphones and microphones located in three dimensions. The presented theory is compared to simulations and real measurements of room impulse responses. We describe a similar theory for circular arrays of microphones or loudspeakers. Application of this theory is presented for the study of the angular sampling of head-related transfer functions (HRTFs). As a result, we show that to reconstruct HRTFs at any possible angle in the horizontal plane, an angular spacing of 5 degrees is necessary for HRTFs sampled at 44.1 kHz. Because recording that many HRTFs is not easy, we develop interpolation techniques to achieve acceptable results for databases containing two or four times fewer HRTFs. The technique is based on the decomposition of the HRTFs in their carrier and complex envelopes. With the Fourier representation of the sound field, it is then shown how one can correctly obtain all room impulse responses measured along a trajectory when using a moving loudspeaker or microphone. The presented method permits the reconstruction of the room impulse responses at any position along the trajectory, provided that the speed satisfies a given relation. The maximal speed is shown to be dependent on the maximal frequency emitted and the radius of the circle. This method takes into account the Doppler effect present when one element is moving in the scenario. It is then shown that the measurement of HRTFs in the horizontal plane can be achieved in less than one second. In the last part, we model spatio-temporal channel impulse responses between a fixed source and a moving receiver. The trajectory followed by the moving element is modeled as a continuous autoregressive process. The presented model is simple and versatile. It allows the generation of random trajectories with a controlled smoothness. Application of this study can be found in the modeling of acoustic channels for acoustic echo cancellation or of time-varying multipath electromagnetic channels used in mobile wireless communications

Movements in Binaural Space: Issues in HRTF Interpolation and Reverberation, with applications to Computer Music

This thesis deals broadly with the topic of Binaural Audio. After reviewing the literature, a reappraisal of the minimum-phase plus linear delay model for HRTF representation and interpolation is offered. A rigorous analysis of threshold based phase unwrapping is also performed. The results and conclusions drawn from these analyses motivate the development of two novel methods for HRTF representation and interpolation. Empirical data is used directly in a Phase Truncation method. A Functional Model for phase is used in the second method based on the psychoacoustical nature of Interaural Time Differences. Both methods are validated; most significantly, both perform better than a minimum-phase method in subjective testing. The accurate, artefact-free dynamic source processing afforded by the above methods is harnessed in a binaural reverberation model, based on an early reflection image model and Feedback Delay Network diffuse field, with accurate interaural coherence. In turn, these flexible environmental processing algorithms are used in the development of a multi-channel binaural application, which allows the audition of multi-channel setups in headphones. Both source and listener are dynamic in this paradigm. A GUI is offered for intuitive use of the application. HRTF processing is thus re-evaluated and updated after a review of accepted practice. Novel solutions are presented and validated. Binaural reverberation is recognised as a crucial tool for convincing artificial spatialisation, and is developed on similar principles. Emphasis is placed on transparency of development practices, with the aim of wider dissemination and uptake of binaural technology

An Efficient Implementation of Parallel Parametric HRTF Models for Binaural Sound Synthesis in Mobile Multimedia

The extended use of mobile multimedia devices in applications like gaming, 3D video and audio reproduction, immersive teleconferencing, or virtual and augmented reality, is demanding efficient algorithms and methodologies. All these applications require real-time spatial audio engines with the capability of dealing with intensive signal processing operations while facing a number of constraints related to computational cost, latency and energy consumption. Most mobile multimedia devices include a Graphics Processing Unit (GPU) that is primarily used to accelerate video processing tasks, providing high computational capabilities due to its inherent parallel architecture. This paper describes a scalable parallel implementation of a real-time binaural audio engine for GPU-equipped mobile devices. The engine is based on a set of head-related transfer functions (HRTFs) modelled with a parametric parallel structure, allowing efficient synthesis and interpolation while reducing the size required for HRTF data storage. Several strategies to optimize the GPU implementation are evaluated over a well-known kind of processor present in a wide range of mobile devices. In this context, we analyze both the energy consumption and real-time capabilities of the system by exploring different GPU and CPU configuration alternatives. Moreover, the implementation has been conducted using the OpenCL framework, guarantying the portability of the code

Aerospace Medicine and Biology. A continuing bibliography with indexes

This bibliography lists 244 reports, articles, and other documents introduced into the NASA scientific and technical information system in February 1981. Aerospace medicine and aerobiology topics are included. Listings for physiological factors, astronaut performance, control theory, artificial intelligence, and cybernetics are included

Dynamic Measurement of Room Impulse Responses using a Moving Microphone

A novel technique for the recording of large sets of room impulse responses or head-related transfer functions is presented. The technique uses a microphone or a loudspeaker moving with constant speed. Given a setup (e.g. length of the room impulse response), a careful choice of the recording parameters (excitation signal, speed of movement) is shown to lead to the reconstruction of all impulse responses along the trajectory. In the case of moving element along a circle, the maximal angular speed is given in function of the length of the impulse response, its maximal temporal frequency, the speed of sound propagation and the radius of the circle. As result of this theory, it is shown that head-related transfer functions sampled at $44.1~$kHz can be measured at all angular positions along the horizontal plane in less than one second. The presented theory is compared with a real system implementation using a precision moving microphone holder. The practical setup is discussed together with its limitations