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

International audienc

Creation and Perception of Sound Source width in Binaural Synthesis

東京藝術大学平成30年

Robust Personal Audio Geometry Optimization in the SVD-Based Modal Domain

© 2014 IEEE. Personal audio generates sound zones in a shared space to provide private and personalized listening experiences with minimized interference between consumers. Regularization has been commonly used to increase the robustness of such systems against potential perturbations in the sound reproduction. However, the performance is limited by the system geometry such as the number and location of the loudspeakers and controlled zones. This paper proposes a geometry optimization method to find the most geometrically robust approach for personal audio amongst all available candidate system placements. The proposed method aims to approach the most 'natural' sound reproduction so that the solo control of the listening zone coincidently accompanies the preferred quiet zone. Being formulated in the SVD-based modal domain, the method is demonstrated by applications in three typical personal audio optimizations, i.e., the acoustic contrast control, the pressure matching, and the planarity control. Simulation results show that the proposed method can obtain the system geometry with better avoidance of 'occlusion,' improved robustness to regularization, and improved broadband equalization

A system for room acoustic simulation for one's own voice

The real-time simulation of room acoustical environments for one’s own voice, using generic software, has been difficult until very recently due to the computational load involved: requiring real-time convolution of a person’s voice with a potentially large number of long room impulse responses. This thesis is presenting a room acoustical simulation system with a software-based solution to perform real-time convolutions with headtracking; to simulate the effect of room acoustical environments on the sound of one’s own voice, using binaural technology. In order to gather data to implement headtracking in the system, human head- movements are characterized while reading a text aloud. The rooms that are simulated with the system are actual rooms that are characterized by measuring the room impulse response from the mouth to ears of the same head (oral binaural room impulse response, OBRIR). By repeating this process at 2o increments in the yaw angle on the horizontal plane, the rooms are binaurally scanned around a given position to obtain a collection of OBRIRs, which is then used by the software-based convolution system. In the rooms that are simulated with the system, a person equipped with a near- mouth microphone and near-ear loudspeakers can speak or sing, and hear their voice as it would sound in the measured rooms, while physically being in an anechoic room. By continually updating the person’s head orientation using headtracking, the corresponding OBRIR is chosen for convolution with their voice. The system described in this thesis achieves the low latency that is required to simulate nearby reflections, and it can perform convolution with long room impulse responses. The perceptual validity of the system is studied with two experiments, involving human participants reading aloud a set-text. The system presented in this thesis can be used to design experiments that study the various aspects of the auditory perception of the sound of one’s own voice in room environments. The system can also be adapted to incorporate a module that enables listening to the sound of one’s own voice in commercial applications such as architectural acoustic room simulation software, teleconferencing systems, virtual reality and gaming applications, etc

A system for room acoustic simulation for one's own voice

The real-time simulation of room acoustical environments for one’s own voice, using generic software, has been difficult until very recently due to the computational load involved: requiring real-time convolution of a person’s voice with a potentially large number of long room impulse responses. This thesis is presenting a room acoustical simulation system with a software-based solution to perform real-time convolutions with headtracking; to simulate the effect of room acoustical environments on the sound of one’s own voice, using binaural technology. In order to gather data to implement headtracking in the system, human head- movements are characterized while reading a text aloud. The rooms that are simulated with the system are actual rooms that are characterized by measuring the room impulse response from the mouth to ears of the same head (oral binaural room impulse response, OBRIR). By repeating this process at 2o increments in the yaw angle on the horizontal plane, the rooms are binaurally scanned around a given position to obtain a collection of OBRIRs, which is then used by the software-based convolution system. In the rooms that are simulated with the system, a person equipped with a near- mouth microphone and near-ear loudspeakers can speak or sing, and hear their voice as it would sound in the measured rooms, while physically being in an anechoic room. By continually updating the person’s head orientation using headtracking, the corresponding OBRIR is chosen for convolution with their voice. The system described in this thesis achieves the low latency that is required to simulate nearby reflections, and it can perform convolution with long room impulse responses. The perceptual validity of the system is studied with two experiments, involving human participants reading aloud a set-text. The system presented in this thesis can be used to design experiments that study the various aspects of the auditory perception of the sound of one’s own voice in room environments. The system can also be adapted to incorporate a module that enables listening to the sound of one’s own voice in commercial applications such as architectural acoustic room simulation software, teleconferencing systems, virtual reality and gaming applications, etc

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

Analysis, modeling and wide-area spatiotemporal control of low-frequency sound reproduction

This research aims to develop a low-frequency response control methodology capable of delivering a consistent spectral and temporal response over a wide listening area. Low-frequency room acoustics are naturally plagued by room-modes, a result of standing waves at frequencies with wavelengths that are integer multiples of one or more room dimension. The standing wave pattern is different for each modal frequency, causing a complicated sound field exhibiting a highly position-dependent frequency response. Enhanced systems are investigated with multiple degrees of freedom (independently-controllable sound radiating sources) to provide adequate low-frequency response control. The proposed solution, termed a chameleon subwoofer array or CSA, adopts the most advantageous aspects of existing room-mode correction methodologies while emphasizing efficiency and practicality. Multiple degrees of freedom are ideally achieved by employing what is designated a hybrid subwoofer, which provides four orthogonal degrees of freedom configured within a modest-sized enclosure. The CSA software algorithm integrates both objective and subjective measures to address listener preferences including the possibility of individual real-time control. CSAs and existing techniques are evaluated within a novel acoustical modeling system (FDTD simulation toolbox) developed to meet the requirements of this research. Extensive virtual development of CSAs has led to experimentation using a prototype hybrid subwoofer. The resulting performance is in line with the simulations, whereby variance across a wide listening area is reduced by over 50% with only four degrees of freedom. A supplemental novel correction algorithm addresses correction issues at select narrow frequency bands. These frequencies are filtered from the signal and replaced using virtual bass to maintain all aural information, a psychoacoustical effect giving the impression of low-frequency. Virtual bass is synthesized using an original hybrid approach combining two mainstream synthesis procedures while suppressing each method‟s inherent weaknesses. This algorithm is demonstrated to improve CSA output efficiency while maintaining acceptable subjective performance