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Session O Monday, December 3 9:00 am-11:30 am
Chair: Louis D. Fielder, Dolby Laboratories, Inc., San Francisco, CA, USA

9:00 am

O-1 Low-Frequency Modal Equalization of Loudspeaker-Room Responses

Aki V. Mäkivirta, Genelec OY, Iisalmi, Finland
Poju Antsalo, Matti Karjalainen and Vesa Välimäki, Helsinki University of Technology, Espoo, Finland

In a room with strong low-frequency modes the control of excessively long decays is problematic or impossible with conventional passive means. In this paper we present a systematic methodology for active modal equalization able to correct the modal decay behavior of a loudspeaker-room system. Two methods of modal equalization are proposed. The first method modifies the primary sound such that modal decays are controlled. The second method uses separate primary and secondary radiators and controls modal decays with sound fed into the secondary radiator. Case studies of the first method of implementation are presented.

Convention Paper 5480


9:30 am

O-2 Practical Limits in Room Equalization

Louis D. Fielder, Dolby Laboratories, Inc., San Francisco, CA, USA

Traditionally, electronic equalization is used to improve the subjective quality of sound reproduction through the use of simple linear filters of low complexity. It will be shown that the properties of typical rooms combine with psychoacoustics to limit practical equalization to the use of minimum-phase filters of relatively low order despite the existence of new and powerful digital signal processing tools. The high Q and nonminimum-phase nature of the room-loudspeaker-listener transfer function due to wave interference effects creates severe problems for more complete equalization. Typical cinemas and a listening room will be used to investigate the difficulties of more powerful equalization approaches.

Convention Paper 5481


10:00 am

O-3 Pentium III-The DSP of Multimedia Audio?

Patrick McGrath, Audio Processing Technology, Belfast, Northern Ireland

This paper describes the evolution and application of personal computing power within the professional audio and broadcast arena. Utilizing the latest Pentium™ processors, the author will discuss how these may be used for digital signal processing within such areas as broadcast, editing, automation, and webcasting and streaming. With the full array of multimedia, including compression algorithms now available on a software platform, these can be utilized to the advantage of the broadcaster by harnessing the power available in your desktop machine. Current technology has relied on DSP devices on a hardware basis. This paper discusses how the internal PC architecture can take on a multitude of DSP functionality both right now and into the future.

No Convention Paper Printed


10:30 am

O-4 An Efficient Pipeline for Digital Signal Processing in Interactive Audio

Keith Weiner, DiamondWare, Ltd., Mesa, AZ, USA

This paper presents a general-purpose interactive audio pipeline. It supports a set of sound streams and a set of processing algorithms, such that each stream may traverse through any number of processing algorithms in any order, and that control parameters may be changed on the fly. Unique to this system is support for on the fly buffer length changes at any stage of the pipeline, and programmatic changes to control parameters.

Convention Paper 5483


11:00 am

O-5 An Intelligent System Approach to Sound Synthesis Parameter Optimisation

Brahim Hamadicharef and Emmanuel Ifeachor, University of Plymouth, Plymouth, Devon, UK

An intelligent audio system for sound design using artificial intelligence techniques is reported. The system is used to analyze acoustic recordings, extract salient sound features and to process them to generate parameters for sound synthesis, in a manner that mimics human audio experts. Preliminary tests show that the use of the system reduces design time and yet the quality of the resulting sound is considered high by audio experts.

Convention Paper 5484

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