v7.0, 20040922, me
Thursday, October 28, 1:00 pm 5:00 pm
Session C LOUDSPEAKERS, Part 2
Chair: Marshall Buck, Psychotechnology, Inc., Los Angeles, CA, USA
C-1 Do Higher Order Modes at the Horn Driver's Mouth Contribute to the Sound Field of a Horn Loudspeaker?Michael Makarski, Aachen University, Aachen, Germany
The Boundary Element Method (BEM) is a well-known tool in acoustics for the calculation of radiation from vibrating surfaces. When using BEM for the calculation of horn loudspeakers, the horn surface is described by its surface admittance; the connected driver is modeled by the velocity distribution at the common junction of driver and horn. Measurements of the velocity distribution have shown that higher order modes within the horn throat can be excited by the horn driver (presented at the 116th AES Convention). On the other hand, a two-port description of the driver together with a plane-wave velocity distribution for the BEM calculation leads to good results. It is investigated to what extend higher order modes at the drivers mouth contribute to the sound radiation.
Convention Paper 6188
C-2 Investigating the Potential Benefits to Both the Objective and Subjective Performance of a Two-Way Loudspeaker Obtained by Using a Wide-Band Tweeter to Place the Cross-Over at a Lower than Usual FrequencyNeil Harris, Alan Hildyard, Valerie Taylor, New Transducers Ltd. (NXT), Huntingdon, UK
The tweeter in a two-way loudspeaker was replaced by a unit having a natural bandwidth of 300 Hz to 20 kHz. This gave a much greater degree of freedom to the choice of cross-over frequency than would normally be possible. The first part of this paper looks at the potential benefits such freedom could bring to the acoustical performance of the loudspeaker. The second part reports results of early listening tests, which were conducted to discover the most preferred cross-over frequency in the range 700 Hz to 3 kHz.
Convention Paper 6189
C-3 A Multiple Regression Model for Predicting Loudspeaker Preference Using Objective Measurements: Part IIDevelopment and Verification of ModelSean Olive, Harman International Industries, Inc., Northridge, CA, USA
A new model is presented that accurately predicts listener preference ratings of loudspeakers based on anechoic measurements. The model was tested using 70 different loudspeakers evaluated in 19 different listening tests. Its performance was compared to two models based on in-room measurements with 1/3-octave and 1/20-octave resolution, and two models based on sound power measurements, including the Consumers Union (CU) model, tested in Part One. The correlations between predicted and measured preference ratings were: 1.0 (our model), 0.91 (inroom, 1/20th-octave), 0.87 (sound power model), 0.75 (in-room, 1/3-octave), and -0.22 (CU model). Models based on sound power are less accurate because they ignore the qualities of the perceptually important direct and early reflected sounds. The premise of the CU model is that the sound power response of the loudspeaker should be flat, which we show is negatively correlated with preference rating. It is also based on 1/3-octave measurements that are shown to produce less accurate predictions of sound quality.
Convention Paper 6190
C-4 Compact Magnetic Suspension TransducerKenneth Kantor, Ioannis Kanellakopoulos, Ali Jabbari, Tymphany Corporation, Cupertino, CA, USA
The role of compliant parts in the operation of loudspeaker drivers is discussed, and a new method of construction employing a magnetic suspension system is presented. Audio transducers require a complex interaction between moving and non-moving structures, placing conflicting demands on the compliant parts typically employed to interface between them. The limitations of current materials and of manufacturing technology, suggest that replacing flexible and compliant mechanical parts with a system based on magnetic forces might yield several benefits. Such a system, which utilizes a moving magnet balanced between static repulsive forces, is discussed conceptually, analytically and experimentally. Proposed advantages include increased linear excursion, convenient form-factor, reduced wear and fatigue, and the simplification of certain production processes.
Convention Paper 6191
C-5 Comparative Analysis of Nonlinear Distortion in Compression Drivers and HornsAlexander Voishvillo, JBL Professional, Northridge, CA, USA
Nonlinear effects in horn drivers are the inseparable part of the principle of their operation. In addition to the distortion caused by electrodynamic and mechanical effects, the distortion is generated in the compression chamber by the nonlinear adiabatic compression, modulation of the airs mechanical stiffness, mass, and viscous losses, and by the nonlinear relationship between the particle velocity and the sound pressure. A new more accurate nonlinear model of compression chamber has been developed. A significant part of distortion is generated in the phasing plug and the horn due to the nonlinear propagation of the high pressure sound waves. Quantitative comparison of the nonlinear effects in a compression chamber and horn is carried out. The comparison is performed by using such criteria as harmonic distortion and two-tone intermodulation distortion.
Convention Paper 6192
C-6 Maximum Efficiency of Compression DriversD. B. (Don) Keele, Jr., Harman International, Northridge, CA, USA
Small-signal calculations show that the maximum nominal efficiency of a horn loudspeaker compression driver is 50 percent and the maximum true efficiency is 100 percent. Maximum efficiency occurs at the drivers resonance frequency. In the absence of driver mechanical losses, the maximum nominal efficiency occurs when the reflected acoustic load resistance equals the drivers voice-coil resistance. The maximum true efficiency occurs when the reflected acoustic load resistance is much higher than the drivers voice-coil resistance. To maximize the drivers broad-band true efficiency, the Bl force factor must be increased as much as possible, while jointly reducing moving mass, voice-coil inductance, mechanical losses, and front air chamber volume. Higher compression ratios will raise high-frequency efficiency but may decrease mid-band efficiency. This paper will explore in detail the efficiency and design implications of both the nominal and true efficiency relationships including gain-bandwidth tradeoffs.
Convention Paper 6193
C-7 Analysis and Modeling of the Bi-Directional Fluid Flow in Loudspeaker PortsZachary Rapoport, Allan Devantier, Harman International, Northridge, CA, USA
Bass reflex ports are used in loudspeakers to enhance low frequency performance. At low sound levels the port extends the low frequency response by supplying one of the components of a Helmholtz resonator. At higher sound levels the turbulent intensity in the port increases disrupting the Helmholtz resonance causing distortion, noise, and compression. Although there has been significant work done to reduce these negative effects, no optimal solution has been found. To better understand the flow phenomena within the port, Computational Fluid Dynamics (CFD) was used to model the flow. The flow was simulated for six port profiles over a wide range of sound levels. In order to correlate the results of the CFD work to the real world, the same six ports were prototyped and subjected to several objective and subjective tests.
Convention Paper 6194
C-8 Comparison of Inverse Filter Real-Time Equalization Methods for Nonminimum Phase Loudspeaker SystemsAvelino Marques, Polytechnic Institute of Engineering of Porto (ISEP), Porto, Portugal; Diamantino Freitas, Faculty of Engineering of Porto (FEUP), Porto, Portugal
Three time domain digital inverse filter design techniques are considered for non-minimum phase loudspeaker systems equalization, namely: FIR filter obtained with adjustable modeling delay, IIR filter followed by excess-phase compensation, and warped filter also followed by excess-phase compensation. Off-line inverse filtering results using real measured impulse responses of loudspeaker systems are compared and discussed for each design technique on the basis of the time equalization error, similar responses magnitude flatness, phase linearity and filter order. Real-time inverse filter implementations requirements on a real set-up, using a digital signal processor of the Texas Instruments TMS320 family are also compared based on computational load and memory needs. Results show that loudspeaker equalization with an inverse IIR filter followed by excess-phase compensation appears as a good compromise solution
Convention Paper 6195