v7.0, 20040922, me

Saturday, October 30, 9:30 am – 11:30 am
Session Z5 Posters: MULTICHANNEL SOUND
NOTE: During the first 10 minutes of the session all authors will present a brief outline of their presentation.

9:30 am
Z5-1 An Ear Training System for Identifying Parameters of Artificial Reverberation in Multichannel Audio—Jason Corey, University of Michigan, Ann Arbor, MI, USA
Artificial reverberation is a signal processing tool used by recording engineers to help create a sense of spaciousness, depth, and envelopment in a sound recording. A system is proposed for training listeners to detect and identify various aspects of artificial reverberation reproduced in multichannel audio. The training helps increase listeners’ auditory sensitivity to parameters of artificial reverberation in sound scene comparisons. The exercises progress from simple matching to identification of the more subtle aspects of artificial reverberation.
Convention Paper 6262

9:30 am
Z5-2 A Novel Multichannel Panning Method for Standard and Arbitrary Loudspeaker Configurations—Ramy Sadek, Chris Kyriakakis, University of Southern California, Marina Del Rey, CA, USA
This paper presents a novel panning algorithm called Speaker-Placement Correction Amplitude Panning (SPCAP), which guarantees conservation of loudspeaker power output. The method is appropriate for any speaker arrangement (e.g., ITU 5.1, 10.2, etc.) and scales with the number of speakers. SPCAP works by correcting initial pan values based on speaker placement to achieve constant power output. Because panning occurs over an arbitrary number of speakers (i.e., is not pair-wise), SPCAP provides two significant advantages over discrete panning schemes. First, pan values for current and future surround-sound formats (e.g., 5.1 and 10.2) are guaranteed to conserve power under any lower-resolution setup, making dynamic up/down mixing in nonstandard setups feasible. Second, SPCAP provides a framework for producing wide (non point-source) sounds.
Convention Paper 6263

9:30 am
Z5-3 Multi-Source Low Frequency Room Simulation Using Finite Difference Time Domain Approximations—Adrian Celestinos, Sofus Birkedal Nielsen, Aalborg University, Aalborg, Denmark
Sound level distribution generated by loudspeakers placed in a room can be simulated using numerical methods. The purpose of this paper is to present an application based on finite-difference time-domain approximations (FDTD) for the study of low frequencies in audio reproduction such as ordinary stereo to multichannel surround setups. A rectangular room is simulated by using a discrete model in time and space. This technique has been used extensively and gives good performance at low frequencies. The impulse response can be obtained in addition to the sound level distribution. Simulation of multiple loudspeakers in a room can be achieved to evaluate and visualize their coupling with the room. A high frequency resolution can be obtained for auralization purposes.
Convention Paper 6264

9:30 am
Z5-4 A Host-Based Real-Time Multichannel Immersive Sound Playback and Processing System—Ramy Sadek, University of Southern California, Marina Del Rey, CA, USA
This paper presents ARIA (Application Rendering Immersive Audio). This system provides a means for the research community to easily test and integrate algorithms into a multichannel playback/recording system. ARIA uses a host-based architecture, meaning that programs can be developed and debugged in standard C++ without the need for expensive, specialized DSP programming and testing tools. ARIA allows developers to exploit the speed and low cost of modern CPUs, provides cross-platform portability, and simplifies the modification and sharing of codes. This system is designed for real-time playback and processing, thus closing the gap between research testbed and delivery systems.
Convention Paper 6265

9:30 am
Z5-5 Optimum Loudspeaker System with Subwoofer and Digital Equalization—Tsakiris Vassilis, Crystal Audiovideo Ltd., London, UK; Orinos Chris, Crystal Audio SA, Vrilissia, Greece
In this paper we investigate the subwoofer concept in relation to the various benefits of digital equalization and the way it can be used together with today’s small sized multichannel loudspeaker systems. We try to systematize a somewhat objective method of comparing between different subwoofer positions and crossover frequencies regarding their optimum response in a listening area. All these show that we can raise the subwoofer frequency at 120 Hz and thus relieve the main loudspeakers from the task of reproducing frequencies down to 80 Hz. Thus it is possible to create a high-end system using slim, line array, main loudspeakers, with all their known advantages, which can be correctly integrated, both aesthetically and acoustically, in any listening room.
Convention Paper 6266