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Last Updated: 20050913, wtm

P8 - Signal Processing for Audio -2

Saturday, October 8, 1:00 pm — 4:00 pm

Chair: Gilbert Soulodre, Communications Research Centre - Ottowa, Ontario, Canada

P8-1 VisualAudio—An Environment for Designing, Tuning, and Testing Embedded Audio ApplicationsDavid A. Jaffe, Paul Beckmann, Britton Peddie, Timothy Stilson, Scott Van Duyne, Analog Devices, Inc. - San Jose, CA, USA
Different hardware configurations and applications suggest different audio system design trade-offs. VisualAudio is focused on embedded processor applications and currently works with Analog Devices, Inc. SHARC and Blackfin processors. VisualAudio is appropriate for a wide range of applications, including general purpose audio, pro audio, music “stomp” boxes, consumer electronics (such as audio-visual receiver (AVR) systems), and automotive audio systems. This article describes the decisions that were made in the design of VisualAudio and how they are tailored to the embedded processing environment. It contrasts VisualAudio with previous systems created by the authors, particularly Staccato Systems’ “SynthCore,” currently known as Analog Devices’ “SoundMAX.”
Convention Paper 6560 (Purchase now)

P8-2 Analysis and Design Algorithm of Time Varying Reverberator for Low Memory ApplicationsTacksung Choi, Junho Lee, Young-Cheol Park, Dae Hee Youn, Yonsei University - Seoul, Korea
Development of an artificial reverberation algorithm with low memory requirements has been an issue of importance in applications such as mobile multimedia devices. One possible solution to this problem is to embed a time-varying all-pass filter to the feedback loop of the comb filter. In this paper theoretical and perceptual analyses of reverberators embedding time-varying all-pass filters in their feedback loops are presented. The analyses are to find a perceptually acceptable degree of phase variation by the all-pass filter. Based on the analyses, we propose a new methodology of designing reverberators embedding time-varying all-pass filters. Through the subjective tests, we showed that, even with smaller memory, the proposed method is capable of providing perceptually superior sound to the previous methods involving time-invariant parameters.
Convention Paper 6561 (Purchase now)

P8-3 A Comparison of the Performance of “Pruned Tree” versus “Stack” Algorithms for Look-Ahead Sigma Delta ModulatorsJames Angus, The University of Salford - Salford, Greater Manchester, UK
Look-ahead sigma-delta modulators look forward k samples before deciding to output a “one” or a “zero.” The Viterbi algorithm is then used to search the trellis of the exponential number of possibilities that such a procedure generates. This paper describes alternative tree-based algorithms. Tree-based algorithms are simpler to implement because they do not require backtracking to determine the correct output value. They can also be made more efficient using “Stack” algorithms. Both the tree algorithm and the more computationally efficient “Stack” algorithms are described. Implementations of both algorithms are described in some detail. In particular, the appropriate data structures for both the trial filters and score memories. Comparative results of their performance are also presented.
Convention Paper 6562 (Purchase now)

P8-4 Adaptive Strategies for Inverse FilteringScott Norcross, Communications Research Centre - Ottowa, Ontario, Canada; Martin Bouchard, University of Ottawa - Ottowa, Ontario, Canada; Gilbert Soulodre, Communications Research Centre - Ottowa, Ontario, Canada
Inverse filtering methods commonly use techniques such as regularization and/or smoothing to reduce artifacts created by the inverse filter. Previous studies have shown that these additional techniques can themselves introduce audible artifacts. Furthermore, the “optimal” amount of regularization or smoothing must be chosen by trial and error. This paper introduces some adaptive strategies based on analyzing the incoming audio to improve the subjective performance of various inverse filtering methods. The incoming audio signal is processed in blocks and the spectrum or masking curve can be calculated. One can then use the information from the audio signal to modify the inverse filter to help its performance. The characteristics of the incoming audio signal could also be used to determine if the application of an inverse filter is even necessary. In this paper two approaches are used to help define an inverse filter that is dependent on the incoming audio signal based on a frequency-domain fast-deconvolution method.
Convention Paper 6563 (Purchase now)

P8-5 New Understandings of the Use of Ferrites in the Prevention and Suppression of RF Interference to Audio SystemsJim Brown, Audio Systems Group, Inc. - Chicago, IL, USA
Building on the work of Muncy, the author has shown that radio-frequency current on cable shields is often coupled to audio systems by two mechanisms—“the pin 1 problem” and shield-current-induced noise (SCIN). An improved equivalent circuit for a ferrite choke is developed that addresses both dimensional resonance within ferrites and the self resonance of inductors formed using those materials, then compared with measured data. Field tests show that chokes formed by passing signal cables through ferrite cores can significantly reduce current-coupled interference over the range of 500 kHz to 1,000 MHz. Guidelines are presented for diagnosing the causes of EMI from sources as diverse as AM broadcast transmitters and cell phones. Solutions are presented both for use in new products and for RFI suppression in field installations.
Convention Paper 6564 (Purchase now)

P8-6 Parametric Control of Filter Slope versus Time Delay for Linear Phase CrossoversDavid McGrath, Justin Baird, Bruce Jackson, Lake Technology - Surry Hills, New South Wales, Australia
Linear phase crossover filters are a powerful tool for sound system designers. They deliver a near-ideal response with ruler-flat pass band, steep transition slopes, and adjustable stop-band rejection—all with zero phase shift. Transition slopes can be matched to a target response, for example 24 dB or 48 dB per octave, and can also be arbitrarily specified while still retaining a perfect-reconstruction characteristic. Practical application of linear phase crossovers requires manipulation of center frequency, transition slope, and stopband rejection. A graphical user interface is described that gives users new degrees of freedom in defining linear phase filter parameters. By setting bounds for parameters such as delay, a user can continuously vary other parameters while the graphical user interface optimizes the resulting filter. This paper presents new parameters for optimization of a target transition slope within a bounded delay parameter, providing fast and efficient user controls for working with and adjusting the crossover filters in real time.
Convention Paper 6565 (Purchase now)

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