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AES New York 2007
Paper Session P16

P16 - Signal Processing for Room Correction

Sunday, October 7, 1:00 pm — 4:00 pm
Chair: Rhonda Wilson, Meridian Audio - UK

P16-1 Sampling the Energy in a 3-D Sound FieldJan Abildgaard Pedersen, Lyngdorf Audio - Skive, Denmark
The energy in the 3-D sound field in a room holds crucial information needed when designing a room correction system. This paper shows how measured sound pressure in at least 4 randomly selected positions scattered across the entire listening room is a robust estimate of the energy in the 3-D sound field. The reproducibility was investigated for a different number of random positions, which lead to an assessment of the robustness of a room correction system based on different number of random microphone positions.
Convention Paper 7261 (Purchase now)

P16-2 Multi-Source Room Equalization: Reducing Room ResonancesJohn Vanderkooy, University of Waterloo - Waterloo, Ontario, Canada
Room equalization traditionally has been implemented as a single correction filter applied to all the channels in the audio system. Having more sources reproducing the same monophonic low-frequency signal in a room has the benefit of not exciting certain room modes, but it does not remove other strong room resonances. This paper explores the concept of using some of the loudspeakers as sources, while others are effectively sinks of acoustic energy, so that as acoustic signals cross the listening area, they flow preferentially from sources to sinks. This approach resists the buildup of room resonances, so that modal peaks and antimodal dips are reduced in level, leaving a more uniform low-frequency response. Impulse responses in several real rooms were measured with a number of loudspeaker positions and a small collection of observer positions. These were used to study the effect of source and sink assignment, and the derivation of an appropriate signal delay and response to optimize the room behavour. Particular studies are made of a common 5.0 loudspeaker setup, and some stereo configurations with two or more standard subwoofers. A measurable room parameter is defined that quantifies the deleterious effects of low-frequency room resonances, supported by a specific room equalization philosophy. Results are encouraging but not striking. Signal modification needs to be considered.
Convention Paper 7262 (Purchase now)

P16-3 A Low Complexity Perceptually Tuned Room Correction SystemJames Johnston, Serge Smirnov, Microsoft Corporation - Redmond, WA, USA
In many listening situations using loudspeakers, the actualities of room arrangements and the acoustics of the listening space combine to create a situation where the audio signal is unsatisfactorily rendered from the listener’s position. This is often true not only for computer-monitor situations, but also for home theater or surround-sound situations in which some loudspeakers may be too close to too far from the listener, in which some loudspeakers (center, surrounds) may be different than the main loudspeakers, or in which room peculiarities introduce problems in imaging or timbre coloration. In this paper we explain a room-correction algorithm that restores imaging characteristics, equalizes the first-attack frequency response of the loudspeakers, and substantially improves the listeners’ experience by using relatively simple render-side DSP in combination with a sophisticated room analysis engine that is expressly designed to capture room characteristics that are important for stereo imaging and timbre correction.
Convention Paper 7263 (Purchase now)

P16-4 Variable-Octave Complex SmoothingSunil Bharitkar, Audyssey Labs - Los Angeles, CA, USA, and University of Southern California, Los Angeles, CA, USA
In this paper we present a technique for processing room responses using a variable-octave complex-domain (viz., time-domain) smoother. Traditional techniques for room response processing, for equalization and other applications such as auralization, have focused on a constant-octave (e.g., 1/3 octave) and with magnitude domain smoothing of these room responses. However, recent research has shown that room responses need to be processed with a high resolution especially in the low-frequency region to characterize the discrete room modal structure as these are distinctly audible. Coupled this with the need for reducing the computational requirements associated with filters obtained from undesirable over-fitting the high-frequency part of the room response with such a high-Q complex-domain smoother, and knowledge of the fact that the auditory filters have wider bandwidth (viz., lower resolution) in the high-frequency part of the human hearing, the present paper proposes a variable-octave complex-domain smoothing. Thus this paper incorporates, simultaneously, the high low-frequency resolution requirement as well as the requirement of relatively lower-resolution fitting of the room response in the high-frequency part through a perceptually motivated approach.
Convention Paper 7264 (Purchase now)

P16-5 Multichannel Inverse Filtering With Minimal-Phase RegularizationScott Norcross, Communications Research Centre - Ottawa, Ontario, Canada; Martin Bouchard, University of Ottawa - Ottawa, Ontario, Canada
Inverse filtering methods are used in numerous audio applications such as loudspeaker and room correction. Regularization is commonly used to limit the amount of the original response that the inverse filter attempts to correct in an effort to reduce audible artifacts. It has been shown that the amount and type of regularization used in the inversion process must be carefully chosen so that it does not add additional artifacts that can degrade the audio signal. A method of designing a target function based on the regularization magnitude was introduced by the authors, where a minimal-phase target function could be used to reduce any pre-response caused by the regularization. In the current paper a multichannel inverse filtering scheme is introduced and explored where the phase of the regularization itself can be chosen to reduce the audibility of the added regularization. In the single-channel case, this approach is shown to be equivalent to the technique that was previously introduced by the authors.
Convention Paper 7265 (Purchase now)

P16-6 An In-flight Low-Latency Acoustic Feedback Cancellation AlgorithmNermin Osmanovic, Consultant - Seattle, WA, USA; Victor E. Clarke, Erich Velandia, Gables Engineering - Coral Gables, FL, USA
Acoustic feedback is a common problem in high gain systems; it is very unpredictable and unpleasant to the ear. Cockpit communication systems on aircraft may suffer from acoustic feedback between a pilot’s boomset microphone and high gain cockpit loudspeaker. The acoustic feedback tone can compromise flight safety by temporarily blocking communication between the pilot and ground control. This paper presents the design of an in-flight low latency (<6 ms) digital audio processing system that automatically detects and removes acoustic feedback tones from the microphone to loudspeaker audio path. We present information about the acoustic feedback cancellation algorithm including the calculation of feedback existence probability, as implemented in an aircraft cockpit communication system.
Convention Paper 7266 (Purchase now)

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