Journal of the Audio Engineering Society

2004 May - Volume 52 Number 5


A set of mathematical expressions to analyze the performance of loudspeaker line arrays is provided. At first a set of expressions for straight-line arrays is developed, including the directivity function, polar response, quarter-power angle, on-axis and off-axis pressure responses, and two-dimensional pressure field. Since in practice many loudspeaker line arrays are not actually straight, expressions are provided for curved (arc), J, and progressive line arrays. In addition, since loudspeaker systems are often not perfect sources, the effects of spherical radiating sources and gaps between sources are analyzed. Several examples of how to apply the models are given, and modeled performance is compared to measured polar data.

Convolution is one of the key operations in signal processing and control. Despite its importance, convolution is also known to be a computationally expensive process such that direct implementation in the time domain becomes prohibitive for various real-time applications. To overcome the difficulties encountered in convolution using long filters, an efficient method based on a nonuniform sampling scheme is proposed. This method exploits the proportional-band property of human hearing and seeks an optimal design of an exponentially sampled finite impulse response (FIR) filter, in accord with a proportional-band frequency domain template. To justify the proposed technique, experiments were carried out for two audio signal processing applications: head-related transfer functions (HRTFs) and reverberators. Using the proposed method, the memory requirement as well as the computation loading can be reduced drastically without any significant degradation in performance. Although the implementations using the proposed fast convolution technique are distinguishable from direct convolution, the listening tests reveal satisfactory and pleasant impressions.

Measurement of the reverberation time of a passenger car compartment is important because it is a major parameter in determining the absorption rates of the absorptive materials used inside a car. However, it has been difficult to measure this reverberation time using a traditional method because the reverberation time is very short. A short reverberation time makes the value of the BT product of filter bandwidth B and reverberation time T small. In order to measure the reverberation time with accuracy using a traditional method, the value of this BT product has to be at least 16. To overcome this problem, the wavelet transform- based method using a wavelet filter bank, has been used for the measurement of very short reverberation times of a passenger car compartment, and the results are compared with the traditional method, which uses an infinite impulse response filter.

Wavetable Matching of Pitched Inharmonic Instrument Tones

Authors: So, Clifford; Horner, Andrew B.

Wavetable matching is the process of finding the parameters needed to resynthesize a musical instrument tone using wavetable synthesis. The most important parameters are the wavetable basis spectra. Previous wavetable matching has often assumed the original tone was harmonic or nearly harmonic. This assumption does not hold for instruments such as the plucked strings. A semiautomated process has recently been proposed to group the partials based on their normalized frequency deviations, and then match each group. However, the user must adjust the group size manually to find the best match. The method is also inefficient at matching harmonic instruments. A new method for hierarchically grouping partials with similar normalized frequency deviations is introduced. Ordinary wavetable matching then finds the basis spectra for individual groups. The new method is fully automatic and requires no prior knowledge about the inharmonicity of the tone. Results for 18 instrument tones show that the new method slightly improves the perceived match on harmonic tones and greatly improves the perceived match on pitched inharmonic tones.

[feature] Wavefield synthesis (WFS) is essentially a means by which a soundfield can be reconstructed within a listening area using an array of loudspeakers, enabling faithful spatial reproduction. It represents an important but potentially complex advance over conventional loudspeaker stereophony, the topic having been presented and discussed at AES events for many years now. During this time the primary development team, based at the Technical University of Delft in the Netherlands, was working on the theoretical principles and proving the concept of WFS. In recent years the technology has experienced wider interest, and WFS is a key element of a recently completed European research project entitled CARROUSO. In this article we review concepts from a selection of recent AES papers arising from this work, showing how WFS ideas might be integrated into parts of an audio production chain, incorporating elements of object representation, conventional stereophony, and room compensation.

Engineering reports

Measurement of Reverberation Times Using a Wavelet Filter Bank and Application to a Passenger Car

Wavetable Matching of Pitched Inharmonic Instrument Tones

Standards and Information Documents

AES Standards Committee News


Wavefield Synthesis: Evolution from Stereophony and Some Practical Challenges

117th Convention, San Francisco, Call for Workshop Participants


Review of Acoustical Patents

News of the Sections

New Products and Developments

Upcoming Meetings

Available Literature

Membership Information

Advertiser Internet Directory

Sections Contacts Directory

AES Conventions and Conferences


Cover & Sustaining Members List

VIP List & Editorial Staff

Institutional Subscribers: If you would like to log into the E-Library using your institutional log in information, please click HERE.

Choose your country of residence from this list:

Skip to content