Friday, October 10 10:00 am – 11:30 am

Session Z1 Posters: Acoustics and Sound Reproduction

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****Z1-1** **Wavelet-Based Multiple Point Equalization of Room Transfer Function**—*Jae-Jin Jeon, Lae-Hoon Kim, Koeng-Mo Sung*, Seoul National University, Seoul, Korea

A Multiple Point Equalization scheme based on the least square (LS) method for a wavelet-filtered signal is proposed. As the variations of the room transfer function (RTF) are different at different frequency bins depending on wavelength of frequencies, equalization with different frequency resolution is desirable. Using a decomposing received signal with discrete wavelet transform, we can assign different kinds of filters to each bandpass signal. Moreover, RTF measurements at various receiver positions should be utilized to make the inverse filter insensitive to source/receiver position changes. These two methods are well combined to guarantee a wider sweet region in a listening room. Real measurement data are used to construct an inverse filter.

**Z1-2** **The Time When the Reverberation Tail in a Binaural Room Impulse Response Begins**—*Kittiphong Meesawat, Dorte Hammershøi*, Aalborg University, Aalborg, Denmark

This paper aims to determine the time in a binaural room impulse response (BRIR) where the transition from early reflections to the reverberation tail occurs. The reverberation tail is defined as the part of the BRIR that is perceptually independent of directions and locations of measurements in a room. This implies that there are no perceptual differences among the reverberation tails. The audible differences between two binaural signals created from an original BRIR and those created from a modified BRIR were tested. The modified BRIRs are identical to the original BRIRs up to a chosen time and concatenated with the later part taken from another BRIR. Signal processing for BRIR concatenation, the experimental methods and results are presented.

**Z1-3** **Hybrid M Sequences for Room Impulse Response Estimation**—*Joel Preto Paulo, Carlos Rodrigues Martins,* Escola Náutica Infante D. Henrique, Paço D’Arcos, Oeiras, Portugal;* José L. Bento Coelho*, Instituto Superior Técnico, Lisbon, Portugal

The measurement of the room impulse response is often evaluated in the presence of nonstationary noise showing an rms value and a power spectral density that significantly varies with time. Under these conditions, the mean square value, MS, of the sequence must be minimized to improve the overall SNR. This suggests that the analysis should be performed by considering the energy of the noise in the time domain and in the frequency domain.

A modified MLS measurement method working in the time and in the frequency domain for applications in the room acoustics field is presented. Experimental results obtained in real conditions are described and shown in the paper.

The new approach, the hybrid sequences technique proved to lead to a significant increase of the SNR, when compared with the classical MLS technique.

**Z1-4** **Active Field Control (AFC)—Reverberation Enhancement System Using Acoustical Feedback Control**—*Hideo Miyazaki, Takayuki Watanabe, Shinji Kishinaga, Fukushi Kawakami*, Yamaha Corporation, Hamamatsu, Shizuoka, Japan

Technology for controlling sound field by electro-acoustic means is often called Active Field Control (AFC), which is used to improve auditory impressions such as liveness, loudness, and spaciousness in auditoria. The AFC system, which has been developed at Yamaha, utilizes feedback control techniques to recreate natural reverberation based on the existing acoustics of the room. Time varying control, including EMR (Electric Microphone Rotator) and fluc-FIR (fluctuating FIR), is implemented in the AFC system to improve stability, preventing the coloration caused by a feedback loop in the system. In this paper these technologies are summarized, together with an introduction to the recent representative venues using AFC. A system plan using core devices named AFC1, which has been developed at Yamaha and released recently in the U.S., is also presented.

**Z1-5** **Designing a Spherical Microphone Array for the Directional Analysis of Reflections and Reverberation**—*Bradford N. Gover*, National Research Council, Ottawa, Ontario, Canada; *James G. Ryan*, Gennum Corporation, Kanata, Ontario, Canada; *Michael R. Stinson*, National Research Council, Ottawa, Ontario, Canada

Spherical microphone array designs were investigated from the point of view of suitability for directional analysis of reverberant sound fields. Four array geometries (tetrahedron, cube, dodecahedron, geodesic sphere) were considered. Beamforming filters were designed using a constrained gain maximization process. The theoretical performance of each array was then predicted. A room acoustic simulator was used to help assess sufficient directionality and evaluate the suitability of each design. A 32-element geodesic sphere array was constructed and used to make directional measurements in real sound fields.

**Z1-6** **Practical Implementation of Constant Beamwidth Transducer (CBT) Loudspeaker Circular-Arc Line Arrays**—*D. B. (Don) Keele, Jr.*, Harman/Becker Automotive Systems, Martinsville, IN, USA

To maintain constant beamwidth behavior, CBT circular-arc loudspeaker line arrays require that the individual transducer drive levels be set according to a continuous Legendre shading function. This shading gradually tapers the drive levels from maximum at the center of the array to zero at the outside edges of the array. This paper considers approximations to the Legendre shading that both discretize the levels and truncate the extent of the shading so that practical CBT arrays can be implemented. It was determined by simulation that a 3 dB stepped approximation to the shading maintained out to –12 dB did not significantly alter the excellent vertical pattern control of the CBT line array. Very encouraging experimental measurements were exhibited by a pair of passively-shaded prototype CBT arrays using miniature wide-band transducers.

**Z1-7** **Acoustical Evaluation of Virtual Rooms by Means of Binaural Activity Patterns**—*Wolfgang Hess*, Universitaet Bochum, Bochum, Germany; Harman/Becker Automotive Systems, Ittersbach, Germany; *Jonas Braasch, Jens Blauert*, Universitaet Bochum, Bochum, Germany

The output of computational auditory localization models (e.g., [Lindemann, 1986], [Gaik, 1993]) is often given in the form of a time-intensity-lateralization correlogram, the so-called binaural activity pattern. These patterns can be used for detection, identification, and separation of incoherent sound sources, for determination of the azimuth, detection of echoes, and the estimation of the amount of auditory spaciousness. It is investigated how the binaural activity patterns of head-related room impulse responses can be used for the perceived quality judgment of different virtual rooms with the aim to present the binaural activity patterns in the optimal form to visualize a general overview of the perceived room-acoustics and to characterize perceived lateral reflections, reverberation, and energy distribution over time and lateralization.