AES Paris 2016
Paper Session P14
P14 - Audio Signal Processing: Part 3—Audio Applications
Monday, June 6, 08:45 — 12:15 (Room 352B)
Chair:
Iva Salom, Institute Mihajlo Pupin, University of Belgrade - Belgrade, Serbia
P14-1 Ensemble Effect Using Gaussian Matrices—Connor McCullough, Bose Corporation - Boston, MA, USA; University of Miami - Coral Gables, FL
The purpose of this paper is to propose an algorithm to serve as an alternative to the chorus effect, the current standard for simulating an ensemble from a single track. Due to the deterministic nature of chorus, specifically the use of an LFO to modulate the delay, chorus often has audible oscillation and does not truly model the behavior of musicians playing simultaneously. The proposed alternative is the implementation of a Gaussian-based algorithm that attempts to model the actual process of musicians playing together. This modeling will be achieved by generating a Gaussian matrix ([# of instruments] x [# of notes]), with each index containing a resampling factor that will temporally and tonally shift each note in a recording. While the Gaussian distribution will serve as the basis for the algorithm, additional constraints will be applied to the resampling factor in order to properly model ensemble behavior.
Convention Paper 9558 (Purchase now)
P14-2 A Loudness Function for Analog and Digital Sound Systems Based on Equal Loudness Level Contours—Sofus Birkedal Nielsen, Aalborg University - Aalborg, Denmark
A new and better loudness compensation has been designed based on the differences between the Equal Loudness Level Contours (ELLC) in ISO 226:2003. Sound productions are normally being mixed at a high Mixing Level (ML) in dB but often played at a lower listening level, which means that the perceived frequency balance will been changed both for LL lower or higher than ML. The differences in ELLC ask for a level based equalization using fractional-order filters. A designing technique for both analog and digital fractional-order filters has been developed. The analog solution is based on OPAMs and the digital solution is realized in a 16/32 bit fixed point DSP and could be implemented in any sound producing system.
Convention Paper 9559 (Purchase now)
P14-3 Spatial Multi-Zone Sound Field Reproduction Using Higher-Order Loudspeakers in Reverberant Rooms—Keigo Wakayama, NTT Service Evolution Laboratories - Kanagawa, Japan; Hideaki Takada, NTT Service Evolution Laboratories - Kanagawa, Japan
We propose a method for reproducing multi-zone sound fields in a reverberant room using an array of higher-order loudspeakers. This method enables sparse arrangement of loudspeakers and reproduction of independent sound fields for multiple listeners without the need for headphones. For multi-zone reproduction, global sound field coefficients are obtained using translation operator. By using the coefficient of the room transfer function measured or simulated with an extension of the image-source method, the loudspeakers’ coefficients are then calculated with the minimum norm method in the cylindrical harmonic domain. From experiments of two-zone and three-zone examples, we show that there was a 2N + 1-fold decrease in the number of Nth-order loudspeakers for accurate reproduction with the proposed method compared to conventional methods.
Convention Paper 9560 (Purchase now)
P14-4 WITHDRAWN—N/A
Convention Paper 9561 (Purchase now)
P14-5 Comparison of Simple Self-Oscillating PWM Modulators—Nicolai Dahl, Technical University of Denmark - Lyngby, Denmark; Niels Elkjær Iversen, Technical University of Denmark - Kogens Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
Switch-mode power amplifiers has become the conventional choice for audio applications due to their superior efficiency and excellent audio performance. These amplifiers rely on high frequency modulation of the audio input. Conventional modulators use a fixed high frequency for modulation. Self-oscillating modulators do not have a fixed modulation frequency and can provide good audio performance with very simple circuitry. This paper proposes a new type of self-oscillating modulator. The proposed modulator is compared to an already existing modulator of similar type and their performances are compared both theoretically and experimentally. The result shows that the proposed modulator provides a higher degree of linearity resulting in around 2% lower Total Harmonic Distortion (THD).
[Also a poster—see session P19-10]
Convention Paper 9562 (Purchase now)
P14-6 Low Energy Audio DSP Design: Going Beyond The Hardware Barrier—Jamie Angus, University of Salford - Salford, Greater Manchester, UK; JASA Consultancy - York, UK
Modern digital audio signal processors need to be energy efficient, both for mobile audio and environmental concerns. Improving technology has been reducing the power of these devices via better, smaller, transistors and reduced voltage swings between one and zero. However, there is a limit to how far this improvement can go. To further reduce processor energy consumption the number of transitions between one and zero must be reduced. This paper presents a method of doing this to, instructions, addresses, and data. By looking at the interaction between their usage statistics and their digital representation and modifying it to match the usage a reduction in energy consumption is achieved. The paper present both measured usage statistics, and bit allocation strategies to achieve this.
Convention Paper 9563 (Purchase now)
P14-7 WITHDRAWN—N/A
Convention Paper 9564 (Purchase now)
