Chris Kyriakakis

Alpha-stable modeling of noise and robust time-delay estimation in the presence of impulsive noise

A new representation of audio noise signals is proposed, based on symmetric /spl alpha/-stable (S/spl alpha/S) distributions in order to better model the outliers that exist in real signals. This representation addresses a shortcoming of the Gaussian model, namely, the fact that it is not well suited for describing signals with impulsive behavior. The /spl alpha/-stable and Gaussian methods are used to model measured noise signals. It is demonstrated that the /spl alpha/-stable distribution, which has heavier tails than the Gaussian distribution, gives a much better approximation to real-world audio signals. The significance of these results is shown by considering the time delay estimation (TDE) problem for source localization in teleimmersion applications. In order to achieve robust sound source localization, a novel time delay estimation approach is proposed. It is based on fractional lower order statistics (FLOS), which mitigate the effects of heavy-tailed noise. An improvement in TDE performance is demonstrated using FLOS that is up to a factor of four better than what can be achieved with second-order statistics.

Fundamental and technological limitations of immersive audio systems

Numerous applications are currently envisioned for immersive audio systems. The principal function of such systems is to synthesize, manipulate, and render sound fields in real time. We examine several fundamental and technological limitations that impede the development of seamless immersive audio systems. Such limitations stem from signal processing requirements, acoustical considerations, human listening characteristics, and listener movement. We present a brief historical overview to outline the development of immersive audio technologies and discuss the performance and future research directions of immersive audio systems with respect to such limits. Last, we present a novel desktop audio system with integrated listener-tracking capability that circumvents several of the technological limitations faced by todays digital audio workstations.

Surrounded by sound

The authors discuss immersive audio systems and the signal processing issues that pertain to the acquisition and subsequent rendering of 3D sound fields over loudspeakers. On the acquisition side, recent advances in statistical methods for achieving acoustical arrays in audio applications are reviewed. Classical array signal processing addresses two major aspects of spatial filtering, namely localization of a signal of interest, and adaptation of the spatial response of an array of sensors to achieve steering in a given direction. The achieved spatial focusing in the direction of interest makes array signal processing a necessary component in immersive sound acquisition systems. On the rendering side, 3D audio signal processing methods are described that allow rendering of virtual sources around the listener using only two loudspeakers. Finally, the authors discuss the commercial implications of audio DSP.

From remote media immersion to Distributed Immersive Performance

We present the architecture, technology and experimental applications of a real-time, multi-site, interactive and collaborative environment called Distributed Immersive Performance (DIP). The objective of DIP is to develop the technology for live, interactive musical performances in which the participants - subsets of musicians, the conductor and the audience - are in different physical locations and are interconnected by very high fidelity multichannel audio and video links. DIP is a specific realization of broader immersive technology - the creation of the complete aural and visual ambience that places a person or a group of people in a virtual space where they can experience events occurring at a remote site or communicate naturally regardless of their location. The DIP experimental system has interaction sites and servers in different locations on the USC campus and at several partners, including the New World Symphony of Miami Beach, FL. The sites have different types of equipment to test the effects of video and audio fidelity on the ease of use and functionality for different applications. Many sites have high-definition (HD) video or digital video (DV) quality images projected onto wide screen wall displays completely integrated with an immersive audio reproduction system for a seamless, fully three-dimensional aural environment with the correct spatial sound localization for participants. The system is capable of storage and playback of the many streams of synchronized audio and video data (immersidata), and utilizes novel protocols for the low-latency, seamless, synchronized real-time delivery of immersidata over local area networks and wide-area networks such as Internet2. We discuss several recent interactive experiments using the system and many technical challenges common to the DIP scenario and a broader range of applications. These challenges include: (1). low latency continuous media (CM) stream transmission, synchronization and data loss management; (2). low latency, real-time video and multichannel immersive audio acquisition and rendering; (3). real-time continuous media stream recording, storage, playback; (4). human factors studies: psychophysical, perceptual, artistic, performance evaluation; (5). robust integration of all these technical areas into a seamless presentation to the participants.

High-fidelity multichannel audio coding with Karhunen-Loeve transform

A new quality-scalable high-fidelity multichannel audio compression algorithm based on MPEG-2 advanced audio coding (AAC) is presented. The Karhunen-Loeve transform (KLT) is applied to multichannel audio signals in the preprocessing stage to remove interchannel redundancy. Then, signals in decorrelated channels are compressed by a modified AAC main profile encoder. Finally, a channel transmission control mechanism is used to re-organize the bitstream so that the multichannel audio bitstream has a quality scalable property when it is transmitted over a heterogeneous network. Experimental results show that, compared with AAC, the proposed algorithm achieves a better performance while maintaining a similar computational complexity at the regular bit rate of 64 kbit/sec/ch. When the bitstream is transmitted to narrowband end users at a lower bit rate, packets in some channels can be dropped, and slightly degraded, yet full-channel, audio can still be reconstructed in a reasonable fashion without any additional computational cost.

A cluster centroid method for room response equalization at multiple locations

We address the problem of simultaneous room response equalization for multiple listeners. Traditional approaches to this problem have used a single microphone at the listening position to measure impulse responses from a loudspeaker and then use an inverse filter to correct the frequency response. The problem with that approach is that it only works well for that one point and in most cases is not practical even for one listener with a typical ear spacing of 18 cm. It does not work at all for other listeners in the room, or if the listener changes positions even slightly. We propose a new approach that is based on the fuzzy c-means clustering technique. We use this method to design equalization filters and demonstrate that we can achieve better equalization performance for several locations in the room simultaneously as compared to single point or simple averaging methods.

Inverse filter design for immersive audio rendering over loudspeakers

Immersive audio systems can be used to render virtual sound sources in three-dimensional (3-D) space around a listener. This is achieved by simulating the head-related transfer function (HRTF) amplitude and phase characteristics using digital filters. In this paper, we examine certain key signal processing considerations in spatial sound rendering over headphones and loudspeakers. We address the problem of crosstalk inherent in loudspeaker rendering and examine two methods for implementing crosstalk cancellation and loudspeaker frequency response inversion in real time. We demonstrate that it is possible to achieve crosstalk cancellation of 30 dB using both methods, but one of the two (the Fast RLS Transversal Filter Method) offers a significant advantage in terms of computational efficiency. Our analysis is easily extendable to nonsymmetric listening positions and moving listeners.

Hybrid algorithm for robust, real-time source localization in reverberant environments

The location of an acoustical source can be found robustly using the steered response pattern-phase transform (SRP-PHAT) algorithm. However SRP-PHAT can be computationally expensive, requiring a search of a large number of candidate locations. The required spacing between these locations is dependent on sampling rate, microphone array geometry, and source location. In this work, a novel method is presented that calculates a smaller number of test points using an efficient closed-form localization algorithm. This method significantly reduces the number of calculations, while still remaining robust in acoustical environments.

Signal Processing, Acoustics, and Psychoacoustics for High Quality Desktop Audio

Integrated media workstations are increasingly being used for creating, editing, and monitoring sound that is associated with video or computer-generated images. While the requirements for high quality reproduction in large-scale systems are well understood, these have not yet been adequately translated to the workstation environment. In this paper we discuss several factors that pertain to high quality sound reproduction at the desktop, including acoustical and psychoacoustical considerations, signal processing requirements, and the importance of dynamically adapting the reproduced sound as the listeners head moves. We present a desktop audio system that incorporates several novel design requirements and integrates vision-based listener-tracking for accurate spatial sound reproduction. We conclude with a discussion of the role the pinnae play in immersive (3D) audio reproduction and present a method of pinna classification that allows users to select a set of parameters that closely match their individual listening characteristics.

Robust time delay estimation for sound source localization in noisy environments

This paper addresses the problem of robust localization of a sound source in a wide range of operating environments. We use fractional lower order statistics in the frequency domain of two-sensor measurements to accurately locate the source in impulsive noise. We demonstrate a significant improvement in detection via simulation experiments of a sound source in /spl alpha/-stable noise. Applications of this technique include the efficient steering of a microphone array in teleconference applications.