Showing papers in "Journal of The Audio Engineering Society in 1993"

Auralization-An Overview

Abstract: Auralization is a term introduced to be used in analogy with visualization to describe rendering audible (imaginary) sound fields. Several modeling methods are available in architectural acoustics for this purpose. If auralization is done by computer modeling, it can be thought of as "true" acoustical computer-aided design. Together with new hardware implementations of signal processing routines, auralization forms the basis of a powerful new technology for room simulation and aural event generation. The history, trends, problems, and possibilities of auralization are described. The discussion primarily deals with auralization of auditorium acoustics andloudspeaker installations. The advantages and disadvantages of various approaches are discussed, as are possible testing and verification techniques. The possibility of using acoustic scale models for auralization is also discussed. \ Demonstrations of auralization have been made, but still the technology's ability to reproduce the subjective impression of the audible characteristics of a hall accurately remains to be verified. This limits the credibility of auralization as a design tool, and verification of auralization the foremost problem to be attacked at this time. The verification problem also applies to the basic room impulse response prediction programs. The combination of auralization with transaural reproduction, room equalization, and active noise control could make it possible to expand the applications of the technology beyond the laboratory and beyond simple headphone reproduction. A large number of interesting applications outside room and psychoacoustics research are conceivable, the most interesting of which are probably its use in information, education, and entertainment.

Auditory Distance Perception in Different Rooms

Abstract: A series of listening experiments was carried out to investigate how auditory distance is perceived in different rooms-an anechoic room, an IEC listening room, and a classroom. Loudspeakers playing back a voice signal were placed in the rooms

The AC-3 Multichannel Coder

Abstract: This paper describes the operation of the AC-3 multichannel adaptive transform coder. Coding a multiplicity of channels as a single entity is more efficient that coding individual channels, permitting a lower net data rate. The major processing blocks of the AC-3 coder are presented and discussed, along with the underlying psychoacoustic principles. Strengths and limitations of the coding techniques are cited.

Auralization of Impulse Responses Modeled on the Basis of Ray-Tracing Results

Abstract: It is shown how the information contained in an energy histogram obtained by ray tracing can be converted into an equivalent impulse response which, after binaural processing, is ready for auralization sufficient for the auralization process. Furthermore the steps are discussed which are needed to transform this information into a binaural impulse response ready for a convolution with music signals

Distortion Immunity of MLS-Derived Impulse Response Measurements

Abstract: Maximum-length-sequence (MLS) measurement of system impulse responses offers a potential enhancement in error immunity over periodic impulse testing, although care must be exercised in setting the MLS excitation amplitude in order to realize this potential. The effects of nonlinearity in MLS measurement are studied, in particular the way in which impulse response errors due to nonlinearity are distributed across the measurement period. The consequences of such errors in cumulative spectral dispIay plots are also investigated. Finally inverse-repeat sequences (IRS) are shown to have complete immunity to even-order nonlinearity while maintaining many of the advantages of MLS.

Methods for multiple wavetable synthesis of musical instrument tones

Abstract: We have explored two techniques for determining basis spectra and amplitude envelopes for resynthesizing tones via multiple fixed wavetable synthesis Breaking down the matching processes into efficient, robust subprocedures was central to the success of both the genetic algorithm and the principal component algorithm-based techniques

Horn Modeling with Conical and Cylindrical Transmission'"Line Elements*

Abstract: The applications to horns are described. Cylindrical and conical element models, both loss-free and dissipative, are presented

Low-Frequency Loudspeaker Models that Include Suspension Creep

Abstract: The traditional low frequency loudspeaker model is expanded to incorporate suspension creep (continued slow displacement under sustained force) by replacing the simple linear compliance with a dynamic transfer function. Three creep models are developed, partly from viscoelastic theory, partly from empirical considerations. Three different woofers are investigated.

Audibility of Changes in Geometric Shape, Source Directivity, and Absorptive Treatment-Experiments in Auralization

Abstract: Some demonstrations of realistic room acoustic situations are presented which use computer models based on actual acoustic spaces as well as on idealized rooms. The object has been to document the responses to the technique from acousticians as well as from those who would benefit most from auralization in the long run: musicians and concertgoers

EARS Auralization Software

Abstract: A new auralization software called EARS, for electronically auralized room simulation, is discussed in conjunction with the EASE (electro acoustic simulator for engineers) acoustic simulation software previously introduced by the authors. Starting from a room simulation in EASE 2.0, the resultant simulated monaural room impulse response, complete with directivity information (angles of incidence), and stored in EASE postprocessing files, is convolved with outer ear transfer functions to derive a binaural signal. This binaural signal is then convolved with dry music. The resultant binaural auralization is a function of the selected listening position in the room and the head orientation of the simulated listener

Axisymmetric Model of a Moving-Coil Loudspeaker

Abstract: The model integrates the traditional low frequency Thiele small parameters with a structure for modeling the high frequency acoustic acoustic response The method can be extended to model the enclosure volume and filling, to model driver nonlinearities, and to model first order diffraction effects from the outside edge of the enclosure

The boundary-element method and Horn design

Abstract: A description is given of both thefull boundary element and the simple source methods for predicting the acoustic field external to a vibrating body, together with an outline of the computational difficulties. Results from an axisymmetric code are compared with analystic and measured values in the context of horn design

Detection of Musical Pitch from Recorded Solo Performances

Abstract: Thispreprinthasbeen reproducedfrom the author'sadvance manuscript,withoutediting,correctionsor considerationby theReview Board. TheAES takesno responsibi/ityfor the contents. A// rightsreserved.Reproductionof thispreprint,or any portion thereof,isnot permitted withoutdirect permission fromthe Jouma/of theAudio EngineeringSociety. Our frequency-domain-based pitch detector graphs musical pitch as the log of detected fundamental frequency on a Unix computer. Recorded virtuosic solo performances are difficult to track when they contain frequency-obscuring reverberation, or when they are dry, since extraneous noises can then interfere. We tested our detector on dry and reverberated performance versions of two musical passages and found it to work quite well on dry versions, but performance was degraded on the reverberated versions. Also, we used the spectral and fundamental frequency data to resynthesize the recordings, with the advantage that most extraneous noises and reverberation were eliminated. The theory of the detector and possibilities for improving its performance are discussed. 1 System Overview Our objective is to develop a tool for accurately detecting and graphing the fundamental frequency or musical pitch as a function of time for solo performance. Although this area has had a long history of activity [1-4], no circuit or computer algorithm has been shown to work in a coinpletely reliable fashion, particularly for rapid passages recorded in either a studio or reverberant environment. Two applications are automatic music transcription and resynthesis with altered timbre, tempo, and pitch range. Also of interest are musicological applications where the detected output can be used to study timing and intonation of performances. Rapid musical passages present peculiar problems since notes are often very short and not necessarily stable. In other words, a large proportion of each note may consist of attack and decay transients, where the pitch is moving towards or away from a stable value. With studio recordings, noises caused by various mechanical devices (such as violin bows and clarinet keys) and the performer (e.g., breath noises) can cause problems, whereas in a hall the predominant factor seems to be the echo of previous notes occurring simultaneously with the current one, causing a chord effect. Our current solution to the problem is a program, implemented on a NeXT computer, which attempts to find the best match to a harmonic series within a time-variant spectrum at each time frame. The spectrum is found using an algorithm originally developed by McAulay and Quatieri [5] to determine spectral peaks which exceed a certain threshold. Then the fundamental frequency is estimated by a harmonic search and …

Auralization in Scale Models: Processing of Impulse Response

Abstract: Acoustical measurement using scale model offer a time- and cost-efficient method for determining the acoustical properties of enclosures. In order to listen into the models, measured impulse responses have to compensated for excessive air attenuation in the model, equalized for transducer response, and their dynamic range has to be extended. The corresponding processing, restricted to monaural signals, is described

An Engineering Study of the Four-Multiply Normalized Ladder Filter

Abstract: An engineering example of an orthogonal second order digital all pass filter implementation is presented in a fully parametric audio equalizer application. independent coefficients control frequency, Q, and boost/cut. The noise and scaling properties of the four multiply normalized ladder structure are analyzed and shown graphically