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Hybrid models for acoustic reverberation
Abstract
There is an increasing interest in creating interactive virtual worlds due to the wide variety of potential applications in entertainment and education. The 3D acoustic scene can be synthesized from two perspectives : the physical approaches and the perceptual approaches. Acoustic radiance transfer method is an efficient ray-based method to model the diffuse reflections and the late reverberation. An extension of the Radiance Transfer Method (RTM) is proposed in this thesis, which allows modeling the early part of specular reflections while keeping the advantage of the original model for the late reverberation simulation. Feedback delay networks are widely used structures to generate the late reverberation. A new method is presented in this thesis, which inherits the efficiency of the Feedback Delay Network (FDN) structure, but aims at linking the parameters of the FDN directly to the geometries of the modeled environment. The relation is achieved by assigning a physical meaning to each delay line and studying the sound energy exchange between them. Then the physical approachand the perceptual approach are combined. The simplified acoustic Radiance Transfer Method, with extension for both specular and diffuse reflections, is incorporated with the Feedback Delay Networks. The new reverberator, despite of modeling the diffuse and late reverberation, is also capable of simulating the early and specular reflections with accuracy.
Classical ultrasonic testing (UT) is a method used to evaluate the parameters of materials using mechanical wave. Most of these methods provide local information about the properties of the medium along the direct paths between emitter and receiver. Yet, some of these methods depend on the coda or late reflections, which provide global information about the medium. In this work, a method inspired by room acoustic techniques is proposed to determine medium properties. In fact, the reverberated signal is highly sensitive to mechanical parameters and boundary conditions. In acoustic room theory, which uses these reverberated signals, for determination of the reverberation time which is very useful to enhance the sound quality of rooms. However, this technique only uses one propagation mode, since the wave propagates through the air inside the acoustic room. Considering that a solid medium is analogous to the acoustic cavity, the walls are equivalent to boundary conditions. Firstly, a model is developed to estimate the acoustic intensity of each propagation mode taking into account the coupling due to mode conversion. It allows the reverberation time to be estimated in solid medium. Secondly, an experimental setup is proposed. A five piezoelectric (PZT) patches, randomly distributed on an aluminum block, one acting as an emitter and the others as receivers to perform experimental reverberation time measurements. Finally, a comparison between the simulated and experimental is done, and inverse problem solving is performed to retrieve the attenuations. A good agreement between the retrieved and literature attenuation values of both bulk waves has been observed in aluminum.
A computationally effcient digital filter structure for late reverberation modeling based on measured room acoustical data is presented. In our reverberator a dense response is obtained by inserting comb-allpass filters in comb filter loops which are connected in parallel. The sum of the comb filter outputs is fed back to their inputs. The advantage of the proposed structure is that a higher reflection density is obtained by a smaller computational burden than with former reverberators. To simulate the acoustics of an existing hall, the early reflections and the frequency dependent reverberation time are analyzed from measured room impulse responses. The data is used for deriving the parameters of the reverberator.
Recent research related to artificial reverberation is reviewed. The focus is on research published during the past few years after the writing of an overview article on the same topic by the same authors. Advances in delay networks, convolution-based techniques, physical room models, and virtual analog reverberation models are described. Many new developments are related to the feedback delay network, which is still the most popular parametric reverberation method. Additionally, three specific methods are discussed in detail: velvet-noise reverberation methods, scattering delay networks, and a modal architecture for artificial reverberation. It becomes evident that research on artificial reverberation and related topics continues to be as active as ever. The related conference paper is available from the AES E-Library at http://www.aes.org/e-lib/browse.cfm?elib=18061.
Computerized room acoustics
modeling has been practiced for almost 50 years up to date. These modeling techniques play an important role in room acoustic design nowadays, often including auralization, but can also help in the construction of virtual environments for such applications as computer games, cognitive research, and training. This overview describes the main principles, landmarks in the development, and state-of-the-art for techniques that are based on geometrical acoustics principles. A focus is given to their capabilities to model the different aspects of sound propagation: specular vs diffuse reflections, and diffraction.
Innovation in virtual reality and motion sensing devices is pushing the development of virtual communication platforms towards completely immersive scenarios, which require full user interaction and create complex sensory experiences. This evolution influences user experiences and creates new paradigms for interaction, leading to an increased importance of user evaluation and assessment on new systems interfaces and usability, to validate platform design and development from the users' point of view. The REVERIE research project aims to develop a virtual environment service for realistic inter-personal interaction. This paper describes the design challenges faced during the development process of user interfaces and the adopted methodological approach to user evaluation and assessment. Keywords: interactive virtual environments, interface design and assessment, sensors for entertainment, virtual characters.
In this work, a linear ray acoustic modelling theory is constructed. The theory forms a base for linear ray acoustic modelling methods. As such, the theory can be used to derive and analyse ray methods. Three existing ray modelling methods (the image source method, the radiosity method, and the ray tracing method) are shown to be derivable from the theory. It is also suggested that the theory can be used to derive acoustic characteristics estimators such as the average reverberation time of a room. To the author's knowledge, this is the first attempt to create a theory for acoustic ray modelling. The theory is divided into two parts: general and acoustic. The general theory consists of general definitions, time-dependent energy propagation equations, and detection equa-tions. The general part yields time-independent ray modelling theory by eliminating time dependency, thus linking the acoustic and the graphic ray modelling. The acoustic part specifies the general definitions as acoustic definitions. The theory lacks sub-surface scat-tering reflection and edge diffraction. A well-defined extension path for the inclusion is considered, however.
Reverberation is a well known effect that has
an important role in our listening experience. A great deal
of research has been devoted in the last decades aiming to
artificially reproduce the reverberation effect exploiting a hybrid
reverberation structure. In this context, several automatic
procedures have been presented in the literature in order to
derive the reverberator structure considering the mixing time
evaluation and the minimization functions definition for the late
reverberation device. Taking into consideration these aspects, a
deep analysis of hybrid digital reverberator audio quality is here
proposed, introducing a new parameter for the definition of the
mixing time and two new cost functions for the definition of the
late reverberation parameters. More in detail, starting from the
considerations derived from a previous accurate approach based
on the mel frequency cepstral coefficients, the new cost functions
are based on the evaluation of the perceptual linear predictive
and power normalized cepstral coefficients. Several results are
reported, in terms of objective measure, performance analysis
and subjective measures, taking into consideration different
real impulse responses and various input stimuli and making
a comparison with the state of the art. In particular, the
obtained results show that a good accuracy can be achieved also
considering a low number of coefficients, therefore improving the
computational performance.
In this paper we propose a methodology aimed at improving the resolution capabilities of plenacoustic imaging, which is based on deconvolution techniques mutuated from aerospace acoustic imaging. In order to reduce the computational burden, we also propose a modification of the minimization problem that exploits the highly structured information contained in the plenacoustic image. Experiments and simulations show the improvement of the accuracy gained by applying the deconvolution operator.
Modelling, simulation and auralisation of room acoustics plays an important role in computer games and virtual reality applications by increasing the level of realism. Accurate simulation of room acoustics is a computationally costly process which is often substituted with artificial reverberators that provide a computationally simpler alternative. However, such systems lack the accuracy and are not in general able to accurately simulate important aspects of room acoustics such as early reflections, source/microphone directivity, and frequency-dependent absorption. A new type of interactive and scalable room simulator named the scattering delay network (SDN) was recently proposed by the authors. A frequency-domain analysis and implementation of that simulator is presented in this paper. Numerical simulation examples which demonstrate the utility of the proposed system are provided.
This paper presents a method of determining a quantitative measure of the ray tracing technique error. This error results from replacing a wave structure of an acoustical field by a grain one. A standard deviation is used as a credibility parameter of the calculation result. Other kinds of credibility parameters are also considered.
An algorithm for the sound ray tracing in a three-dimensional space limited by either convex or concave solid of arbitrary shape is presented. The solid shape is approximated by an arbitrary number of quadrangular plane sectors. The successive parts of the algorithm are discussed from the viewpoint of economizing both calculation time and computer memory space. Some examples of the algorithm implementation in room acoustics are given.
We present an interactive sound propagation algorithm that can compute high orders of specular and diffuse reflections as well as edge diffractions in response to moving sound sources and a moving listener. Our formulation is based on a precomputed acoustic transfer operator, which we compactly represent using the Karhunen-Loeve transform. At runtime, we use a two-pass approach that combines acoustic radiance transfer with interactive ray tracing to compute early reflections as well as higher-order reflections and late reverberation. The overall approach allows accuracy to be traded off for improved performance at runtime, and has a low memory overhead. We demonstrate the performance of our algorithm on different scenarios, including an integration of our algorithm with Valve's Source game engine.
This paper reports the recent works and progress on a PC and C++ language-based virtual auditory environment (VAE) system platform. By tracing the temporary location and orientation of listener's head and dynamically simulating the acoustic propagation from sound source to two ears, the system is capable of recreating free-field virtual sources at various directions and distances as well as auditory perception in reflective environment via headphone presentation. Schemes for improving VAE performance, including PCA-based (principal components analysis) near-field virtual source synthesis, simulating six degrees of freedom of head movement, are proposed. Especially, the PCA-based scheme greatly reduces the computational cost of multiple virtual sources synthesis. Test demonstrates that the system exhibits improved performances as compared with some existing systems. It is able to simultaneously render up to 280 virtual sources using conventional scheme, and 4500 virtual sources using the PCA-based scheme. A set of psychoacoustic experiments also validate the performance of the system, and at the same time, provide some preliminary results on the research of binaural hearing. The functions of the VAE system is being extended and the system serves as a flexible and powerful platform for future binaural hearing researches and virtual reality applications.
Head-related transfer functions(HRTFs) are acoustical transfer functions from sound source to two ears and embody the main cues for sound source localization.They are therefore vital to the researches on binaural hearing.One of the important applications of HRTFs is virtual auditory display, in which input stimulus is filtered by a pair of HRTFs and then reproduced by headphone to create spatial auditory events.HRTFs and virtual auditory display are now the hot topics in the fields of acoustics, signal processing and hearing and have been widely used.The present article reviews the principle and advance in HRTFs and virtual auditory display, including some recent works in China, and then outlines the main applications of virtual auditory display.
A system has been constructed in which a digitally filtered combination of the two channels is fed to each loudspeaker in a traditional stereo setup. The crosstalk is effectively canceled while the good imaging properties of headphone reproduction are preserved. The effect is unexpectedly independent of head position as long as the distances to the loudspeakers are equal. The system is shown to work in an anechoic room, but it is not formally limited to this. For use in a normal living room, more computing power is needed.
We present a method for accelerating the computation of specular reflections in complex 3D enclosures, based on acoustic beam tracing. Our method constructs the beam tree on the fly through an iterative lookup process of a precomputed data structure that collects the information on the exact mutual visibility among all reflectors in the environment (region-to-region visibility). This information is encoded in the form of visibility regions that are conveniently represented in the space of acoustic rays using the Pl ¨ucker coordinates. During the beam tracing phase, the visibility of the environment from the source position (the beam tree) is evaluated by traversing the precomputed visibility data structure and testing the presence of beams inside the visibility regions. The Pl ¨ucker parameterization simplifies this procedure and reduces its computational burden, as it turns out to be an iterative intersection of linear subspaces. Similarly, during the path determination phase, acoustic paths are found by testing their presence within the nodes of the beam tree data structure. The simulations show that, with an average computation time per beam in the order of a dozen of microseconds, the proposed method can compute a large number of beams at rates suitable for interactive applications with moving sources and receivers.
The synthesis of an arbitrary enclosures room impulse response (RIR) may be performed using acoustic modelling. A number of acoustic modelling methods have been proposed previously each with their own advantages and limitations. This paper is concerned with mixing the RIRs from different modelling methods to synthesize a hybrid RIR. Low frequencies are modelled using the finite difference time domain method (FDTD), high frequencies are treated with geometric methods. A practical implementation for forming a hybrid RIR is discussed and further demonstrated in the context of a 2nd order B-Format spatial encode of the modelled sound field. The paper discusses the considerations and limitations of forming such hybrid RIRs using wave-based and geometric-based methods.
To enable efficient low latency convolution, a Fast Fourier Transform (FFT) is presented which balances processor and memory load across incoming blocks of input. The proposed FFT transforms a large block of input data in steps spread across the arrival of smaller blocks of input, and can be used to transform large partitions of an impulse response and input data for efficiency, while facilitating convolution at very low latency. Its primary advantage over a standard FFT, as used for a non-uniform partition convolution method, is that it can be performed in the same processing thread as the rest of the convolution, thereby avoiding problems associated with the combination of multithreading and near real-time calculations on general purpose computing architectures.
The basic requirements for an Auditory Virtual Environment (AVE) are presented and a system based on a physical approach (IKA-SIM), employing the mirror-image model to generate the early reflections, is described. The static and dynamic structure of the IKA-SIM software (written in C++) is shown in diagrams and the computational requirements for real-time performance are delineated. IKA-SIM is able to render rooms of arbitrary shape, to account for frequency dependent absorption factors, and to calculate high order reflections in real-time on a standard PC. The different interfaces for real-time interaction are presented. IKA-SIM supports headphone and loudspeaker reproduction. A new elevation panning algorithm for loudspeaker reproduction is introduced. Design aspects relevant to a real-time AVE system are presented.
In room acoustic modeling, Feedback Delay Networks (FDN) are known to efficiently model late reverberation due to their capacity to generate exponentially decaying dense impulses. However, this method relies on a careful tuning of the different synthesis parameters , either estimated from a pre-recorded impulse response from the real acoustic scene, or set manually from experience. In this paper we present a new method, which still inherits the efficiency of the FDN structure, but aims at linking the parameters of the FDN directly to the geometry setting. This relation is achieved by studying the sound energy exchange between each delay line using the acoustic Radiance Transfer Method (RTM). Experimental results show that the late reverberation modeled by this method is in good agreement with the virtual geometry setting.
Impulse response is an important quantity for room acoustics. To predict the impulse response, several types of calculation methods based on wave theory, such as the boundary element method
(BEM),
finite element method
(FEM) and finite difference method (FDM), are being investigated. Among them, the authors are exploring application of the finite difference time domain
(FDTD) method to calculate room sound fields. Using the method, by which the acoustical quantities at discrete grid points of a closed field are successively solved step‐by‐step according to vector operations, the impulse responses can be obtained directly in a time domain using little computer memory storage. Furthermore, changes of spatial distributions of room sound fields in time can be obtained easily by storing the sound pressure at grid points and displaying them by animation. In this presentation, several examples of the FDTD analyses of room impulse responses are reported and computer animation of their room sound fields is introduced. In addition, examination of the applicability of prediction of the impulse responses by comparison with in
situ measurement of a small hall is discussed.
Transaural stereo, generic for binaural stereo processed for cancellation of Loudspeaker-to-ear crosstalk, results from the use of minimum-phase filters in shuffler configuration. Simplifying the filters further at short wavelengths makes the listener position noncritical. Full spatial qualities appear in a conventional stereo playback that avoids early reflections. Inverse shufflers provide precise transaural pan functions for multitrack work.
Realistic 3D audio can greatly enhance the sense of presence in a virtual environment. We introduce a framework for capturing, transmitting and rendering of 3D audio in presence of other bandwidth savvy streams in a 3D Tele-immersion based virtual environment. This framework presents an efficient implementation for 3D Binaural Spatialization based on the positions of current objects in the scene, including animated avatars and on the fly reconstructed humans. We present a general overview of the framework, how audio is integrated in the system and how it can exploit the positions of the objects and room geometry to render realistic reverberations using head related transfer functions. The network streaming modules used to achieve lip-synchronization, high-quality audio frame reception, and accurate localization for binaural rendering are also presented. We highlight how large computational and networking challenges can be addressed efficiently. This represents a first step in adequate networking support for Binaural 3D Audio, useful for tele-presence. The subsystem is successfully integrated with a larger 3D immersive system, with state of art capturing and rendering modules for visual data.
In this paper we propose an extension for the Acoustic Radiance Transfer (ART) method for the modeling of room acoustics. The original ART method is very efficient for modeling diffuse reflections and the late reverberation but does not well represent the early echoes. We then propose, in this paper, an extension of the ART method which allows to model the early part while keeping the advantages of the original method for the late reverberation simulation. The experimental results confirm that the proposed method gives more accurate reconstruction of the early reflections than the traditional ART method in average and that comparable accuracy can be obtained at lower complexity and memory requirements than the traditional ART method.
Synthesizing the room impulse response (RIR) of an arbitrary enclosure may be performed using a number of alternative acoustic modeling methods, each with their own particular advantages and limitations. This article is concerned with obtaining a hybrid RIR derived from both wave and geometric-acoustics based methods, optimized for use across different regions of time or frequency. Consideration is given to how such RIRs can be matched across modeling domains in terms of both amplitude and boundary behavior and the approach is verified using a number of standardised case studies.
CORDIS-ANIMA is a digital, real-time object modeling and simulation system. The main purpose of the system is to model the instrumental world. This purpose is achieved by the computer simulation of music and by the animation of images. The synthesized music represents the real life instruments that produce sound vibrations when subjected to some action. Similarly, the animated images are taken from the real world as well. Total simulation effect (of the music and the images) is produced by optimizing the 'man-machine' interaction feature of this software.
In order to predict the aural impression of the listening conditions which is to be expected at the various seats of a room (e.g. a concert hall) in particular when this is still in the stage of design, binaural room impulse responses for given source and receiver positions must be determined. This paper describes an approach to add a diffuse background signal to the result of an image source method, in which only the specular parts of the reflections are considered. The missing diffusely scattered energies are handled by a method based on the result of a sound particle tracing procedure with a low time resolution of a few milliseconds. This combined method is used to predict the direct sound and the early reflections of the binaural room impulse response. The reverberant part and the diffuse background signal are determined with methods which require only little room-specific information and obtain the remaining data statistically. For the late part of the binaural response a simplified simulation method is presented, that is based on the frequency-dependent reverberation time. Furthermore, the paper discusses the connection of two parts of a room impulse response, which are calculated by means of different methods.
A geometrical cone-tracing method associated with the signal processing technique is used to calculate the binaural impulse response of a concert hall. Some inaccuracy and the computation time of the geometrical algorithm tend to limit the method for the high-reflection orders which are necessary to provide a good listening effect. In order to extend the response, a new approach is presented based on different statistical processes that depend on both the acoustical and geometrical characteristics of the hall. After a theoretical presentation (new statistical results are proposed to describe the sound field behavior in a concert hall), some simulations are given in order to illustrate the different statistical processes. This simulation technique seems to be a very convenient tool both for the design of a new concert hall and for the study of the important parameters in auditory spaciousness.
The assumption of diffuse reflection (Lambert's Law) leads to integral equations for the wall intensity in a reverberant sound field in the steady state and during decay. The latter equation, in the special case of a spherical enclosure with uniformly absorbent walls and uniform wall intensity, allows exponential decay with a decay time which agrees closely with the Norris--Eyring prediction. The sound-intensity and sound-energy density in the medium, during decay, are also calculated.
The impulse response of a linear system can be determined by exciting the system with white noise, and cross-correlating the input and output. As contrasted with the straightforward technique suing an impulsive excitation, this approach is capable of providing vastly superior dynamic range. In order to minimize the amount of computation required by the cross-correlation step, the system can be excited by a binary maximal-length sequence, and the cross correlation performed using the fast Hadamard transform. By this means, only additions are required, and the number of additions is approximately 2.5n log(2) n, where n is the length of the sequence.
Room acoustics has been digitally modeled with impulse responses calculated by ray-tracing orimage-source methods. Both of them are based on geometrical acoustics, which is well applicableonly to high frequencies, where diffraction can be neglected. Based on previous research onwaveguide meshes, we have implemented a finite difference scheme of arbitrary dimension andused it to model sound propagation in three-dimensional room spaces. The method has shownpromising results, particularly...
Imagine that you are blindfolded inside an unknown room. You snap your fingers and listen to the room's response. Can you hear the shape of the room? Some people can do it naturally, but can we design computer algorithms that hear rooms? We show how to compute the shape of a convex polyhedral room from its response to a known sound, recorded by a few microphones. Geometric relationships between the arrival times of echoes enable us to "blindfoldedly" estimate the room geometry. This is achieved by exploiting the properties of Euclidean distance matrices. Furthermore, we show that under mild conditions, first-order echoes provide a unique description of convex polyhedral rooms. Our algorithm starts from the recorded impulse responses and proceeds by learning the correct assignment of echoes to walls. In contrast to earlier methods, the proposed algorithm reconstructs the full 3D geometry of the room from a single sound emission, and with an arbitrary geometry of the microphone array. As long as the microphones can hear the echoes, we can position them as we want. Besides answering a basic question about the inverse problem of room acoustics, our results find applications in areas such as architectural acoustics, indoor localization, virtual reality, and audio forensics.
There are intractable problems with computer models of enclosed spaces using geometric acoustic assumptions. A new beam tracing method for finding image-receiver paths using triangular section beams is suggested where the source of these beams is omnidirectional and the beams do not overlap. A method of radiant exchange is proposed that produces a diffuse reverberant tail for an impulse response. There is no sharp division between the two models, rather the two overlap and the emphasis changes from the specular to the diffuse model.