Signal Processing Part 2

Paper Sessions

Chair:
Rémi Audfray, Magic Leap, Inc. - San Francisco, CA, USA

P05-1 A Pseudoinverse Technique for the Pressure-Matching Beamforming Method—Miller Puckette, University of California San Diego - San Diego, CA, USA; Tahereh Afghah, University of California San Diego - San Diego, CA, USA; Elliot Patros, University of California San Diego - San Diego, CA, USA
In this work an extension to the pressure-matching beamforming method (PMM) that’s well-suited for transaural sound field control is presented. The method aims to improve performance at dark points, locations relative to the array where sound pressure is minimized; without producing noticeable artifacts at bright points, locations where acoustic interference is minimized. The method’s new performance priorities result from replacing Tikhonov regularization, which is conventionally used in PMM, with a purpose-built regularization strategy for solving the pseudoinverse of ill-conditioned matrices. Discussions of how this method’s formulation affects the filter design process and of performance comparisons between this and conventional PMM filters are included. 

P05-2 Analog Circuits and Port-Hamiltonian Realizability Issues: A Resolution Method for Simulations via Equivalent Components—Judy Najnudel, IRCAM - Paris, France; Thomas Hélie, IRCAM - CNRS-Sorbonne Université - Paris, France; Henri Boutin, IRCAM - Paris, France; David Roze, CNRS - Paris, France; STMS lab (UMR 9912, CNRS - Ircam - Sorbonne Université) - Paris, France; Thierry Maniguet, CNRS-Musé de la Musique - Paris, France; Stéphane Vaiedelich, CNRS-Musé de la Musique - Paris, France
In order to simulate the Ondes Martenot, a classic electronic musical instrument, we aim to model its circuit using Port-Hamiltonian Systems (PHS). PHS have proven to be a powerful formalism to provide models of analog electronic circuits for audio applications, as they guarantee the stability of simulations, even in the case of non-linear systems. However, some systems cannot be converted directly into PHS because their architecture cause what are called realizability conflicts. The Ondes Martenot circuit is one of those systems. In this paper a method is introduced to resolve such conflicts automatically: problematic components are replaced by equivalent components without altering the overall structure nor the content of the modeled physical system. 

P05-3 Practical Realization of Dual-Shelving Filter Using Proportional Parametric Equalizers—Rémi Audfray, Magic Leap, Inc. - San Francisco, CA, USA; Jean-Marc Jot, Magic Leap - San Francisco, CA, USA; Sam Dicker, Magic Leap, Inc. - San Francisco, CA, USA
Proportional Parametric Equalizers have been proposed as an efficient tool for accurate magnitude response control within defined constraints. In particular, a combination of shelving filters can be used to create a 3-band parametric equalizer or tone control with minimal processing overhead. This paper picks up on this concept, fully develops the filter control equations, and proposes a look up table based implementation of the dual-shelving filter design. 

P05-4 Measuring Audio when Clocks Differ—Mark Martin, Audio Precision - Beaverton, OR, USA; Jayant Datta, Audio Precision - Beaverton, OR, USA; Xinhui Zhou, Audio Precision - Beaverton, OR, USA
This paper examines what happens when a digital audio signal is measured with respect to a digital reference signal that was created based on a different clock. The resultant change in frequency and time depends on the degree of mismatch between clocks and may introduce a significant amount of distortion into the measurements. But the distortion is different from the types normally considered important in audio systems and may obscure other types of distortion, e.g., harmonic, IMD, and noise, that are more important to accurately assess. A difference in clocking essentially creates a difference in sample rates. Therefore, sample rate conversion methods can be used to mitigate the discrepancy. Although this approach is effective, it cannot be used when the sample rate difference is too small, it can be computationally intensive, and it cannot entirely eliminate the effects of a clocking difference. This paper describes a much simpler and more effective technique that requires minimal computation. 

P05-5 Reducing Musical Noise in Transform Based Audio Codecs—Elias Nemer, XPERI/DTS - Calabasas, CA, USA; Jeff Thompson, XPERI/DTS - Calabasas, CA, USA; Ton Kalker, XPERI/DTS - Calabasas, CA, USA
This paper addresses the problem of musical noise in transform-based audio codecs. This artifact occurs when encoding audio segments with a noise-like spectrum—at low bit rates where signal quantization results in significant zero-valued coefficients. Due to the quantization commonly used, bands containing several non-zero coefficients are quantized to only one or two, giving rise to a musical artifact. This has been identified in other codecs, such as in CELT/OPUS , where a special transform is used prior to quantization to remedy the problem. In this paper we provide a modified approach consisting of a Hadamard transform combined with an interleaving scheme. Simulation shows the proposed method has a lower complexity and yields improved perceptual scores as measured by PEAQ. 

P05-6 Statistical and Analytical Approach to System Alignment—Juan Sierra, Stanford University - Stanford, CA, USA; Meyer Sound Laboratories - Berkeley, CA, USA; Jonathan Kamrava, Meyer Sound Laboratories - Berkeley, CA, USA; Pablo Espinosa, Meyer Sound Laboratories - Berkeley, CA, USA; Jon Arneson, Meyer Sound Laboratories - Berkeley, CA, USA; Paul Kohut, Meyer Sound Laboratories - Berkeley, CA, USA
The current project describes the design of a tuning or alignment processor for a system based on multiple satellite speakers and a single subwoofer. It explains the methodology used to solve this problem and the procedure to arrive to a viable solution. On one side, the procedure was based on filter design techniques to optimize phase relationships; however, these phase relations are often unknown due to the possibility of changing the relative positions of the speakers. Accordingly, a statistical analysis was used to determine the most stable set of parameters across different speaker locations and acoustical environments, implying that the same alignment parameters can be implemented in multiple circumstances without significant performance degradation.