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P17 - Signal Processing - Part 2
Saturday, October 7, 2:30 pm — 6:00 pm
Chair: Jürgen Herre, Fraunhofer IIS - Erlangen, Germany
P17-1 Loudspeaker-Based 3-D Audio System Design Using the M-S Shuffler Matrix—Martin Walsh, Jean-Marc Jot, Creative Advanced Technology Center - Scotts Valley, CA, USA
Technology Center, Scotts Valley, CA, USA
This paper outlines a new design methodology that can help to achieve higher quality 3-D audio reproduction over loudspeakers for a variety of applications using only adapted M-S matrices. Several key M-S matrix-based topologies are summarized, and a new design methodology is presented that allows the design and efficient implementation of any new 3-D audio system using only M-S matrix-based topologies. A real-world design example is used to highlight how this new design methodology can not only help the 3-D audio system design process, but also improve the audio quality of the resulting reproduction.
Convention Paper 6949 (Purchase now)
P17-2 Binaural Simulation of Complex Acoustic Scenes for Interactive Audio—Jean-Marc Jot, Martin Walsh, Creative Advanced Technology Center - Scotts Valley, CA, USA; Adam Philp, Creative Labs – Sensaura - Egham, Surrey, UK
We describe a computationally efficient 3-D positional audio and spatial reverberation processing architecture for real-time virtual acoustics using headphones or loudspeakers. An advantageous method for binaural synthesis of massive numbers of sound sources is introduced. Extensions of the architecture are described for simulating nearfield emitters, modeling spatially extended sound events, rendering multiroom reverberation, and incorporating the perceptually salient features of early reflections and acoustic obstructions in the listener's immediate virtual environment. The proposed approach enables the implementation of scalable interactive 3-D audio rendering systems in personal computers, game consoles, set top boxes or mobile phones. The associated scene representation model is compatible with current interactive audio standards including OpenAL, MPEG-4, and JSR-234.
Convention Paper 6950 (Purchase now)
P17-3 A Technique for Nonlinear System Measurement—Jonathan Abel, David Berners, Universal Audio, Inc. - Santa Cruz, CA, USA, Stanford, University, Stanford, CA, USA
A method for measuring nonlinear systems having a certain type Volterra series is presented. The Volterra series studied is the parallel combination of elements having series input and output filters around a power-law distortion and may be used to represent a wide variety of systems combining filtering and memoryless distortion functions. The technique is to measure the system using a swept sinusoid at a variety of amplitudes and to use least squares to first separate the element responses and then identify the unknown input and output filters.
Convention Paper 6951 (Purchase now)
P17-4 Esophageal Voice Enhancement by Modeling Radiated Pulses in Frequency Domain—Alex Loscos, Jordi Bonada, Universitat Pompeu Fabra - Barcelona, Spain
Although esophageal speech has been demonstrated to be the most popular voice recovering method after laryngectomy surgery, it is difficult to master and shows a poor degree of intelligibility. This paper proposes a new method for esophageal voice enhancement using speech digital signal processing techniques based on modeling radiated voice pulses in the frequency domain. The analysis-transformation-synthesis technique creates a nonpathological spectrum for those utterances featured as voiced and filters those unvoiced. Healthy spectrum generation implies transforming the original timbre, modeling harmonic phase coupling from the spectral shape envelope, and deriving pitch from frame energy analysis. Resynthesized speech aims to improve intelligibility, minimize artificial artifacts, and acquire resemblance to patient’s presurgery original voice.
Convention Paper 6952 (Purchase now)
P17-5 A Novel IIR Equalizer for Nonminimum Phase Loudspeaker Systems—Avelino Marques,, Polytechnical Institute of Engineering of Porto - Porto, Portugal; Diamantino Freitas, University of Porto - Porto, Portugal
A novel approach for the equalization of nonminimum phase loudspeaker systems based on the design of an IIR inverse filter is presented. This IIR inverse filter is designed in time domain by minimization of the least squares error function that results from using the typical “Output Error” configuration in the inverse modeling of nonminimum phase systems, with an adjustable delay. Due to the nonlinear nature of the error function, iterative optimization methods for nonlinear least squares problems were applied, namely the Levenberg-Marquardt method. This approach allows the design of inverse filter-based equalization solutions with lower computational requirements, lower equalization error, and lower delay of the equalized loudspeaker system than the most used one, the FIR inverse filter. The advantages of this new approach are demonstrated with its application for the equalization of a two loudspeaker systems. The results of the objective evaluation of this application are outlined, presented, and discussed regarding time and frequency domain equalization errors and the delay of the equalized loudspeaker.
Convention Paper 6953 (Purchase now)
P17-6 Spring Reverb Emulation Using Dispersive Allpass Filters in a Waveguide Structure—Jonathan Abel, David Berners, Universal Audio, Inc. - Santa Cruz, CA, USA, Stanford University, Stanford, CA, USA; Sean Costello, Analog Devices - San Jose, CA, USA; Julius O. Smith III, Stanford University - Stanford, CA, USA
Wave propagation along springs in a spring reverberator is studied, and digital emulations of several popular spring reverberator models are presented. Measurements on a number of springs reveal several dispersive propagation modes and evidence of coupling among them. The torsional mode typically used by spring reverberators is seen to be highly dispersive, giving the spring its characteristic sound. Spring reverberators often have several springs operating in parallel, and the emulations presented here use a set of parallel waveguide structures, one for each spring element. The waveguides explicitly compute the left-going and right-going torsional waves, including dispersion, propagation, and reflection effects. Scattering from spring imperfections and from the rings coupling counter-wound springs are modeled via waveguide scattering junctions.
Convention Paper 6954 (Purchase now)
P17-7 Characteristics of Inharmonic Frequency Analysis of GHA and its Application to Audio Signal Processing—Teruo Muraoka, Tohru Fukube, The University of Tokyo - Tokyo, Japan
GHA is a frequency analysis originally proposed by N. Wiener in 1930. His aim was to analyze stochastic signals utilizing harmonic frequency analysis and clarified that any signal can be represented by almost periodic function whose frequency components are in inharmonic relationship. In 1993 Dr. Hirata proposed an inharmonic frequency analysis applicable to audio signal processing, and it became known as “GHA.” The authors have been engaged in its improvement and utilization and reported several applications. Among them, the authors have reported intensive noise reduction to damaged SP records at the last convention [AES 120th Convention, Paris, France, Convention Paper 6725]. In this paper the principle of GHA and its fundamental characteristics will be explained together with its application to noise reduction in comparison with conventional spectral subtraction method.
Convention Paper 6955 (Purchase now)