1:00, AUDIO-L2.1
AN EFFICIENT MULTICHANNEL LINE ECHO CANCELER ALGORITHM FOR PSTN AND VOIP/VODSL APPLICATIONS
A. SUGIYAMA, T. YAMAJI
This paper proposes an efficient multichannel line echo canceler algorithm with interchannel distribution of computations. A limited number of coefficient adaptations are distributed among channels based on the degree of convergence and the input signal power to achieve up to 50\% reduction of total computations. Adaptations are more frequently performed in the channels with sufficient input power, where convergence is behind others. As the index to the degree of convergence, a gradient of squared coefficient sum is used. This index weighted with the input signal power is evaluated for interchannel distribution of coefficient adaptations. This weighting improves the coefficient adaptation efficiency by 30%. Simulation results with white Gaussian signals and speech signals demonstrate good convergence.The computational savings can be used to accommodate more channels on the same chip, or to cover a longer echo-path by codecs and/or ATM cell assembly/disassembly. It is also promising to analog interface in VoIP/VoDSL applications.

1:20, AUDIO-L2.2
DYNAMIC RESOURCE ALLOCATION FOR NETWORK ECHO CANCELLATION
T. GANSLER, J. BENESTY, M. SONDHI, S. GAY
Network echo canceler chips are designed to handle several channels simultaneously. With the processing speeds now available, a single chip might handle several hundred channels. In current implementations, however, the adaptation algorithm is designed for a single channel, and the computations are replicated Nc times, where Nc is the number of channels. With such an implementation, the computational requirement is Nc times the peak load for a single channel. The number of computations required in each channel, however, varies widely over time. Therefore, a considerable reduction in computational load can be achieved by designing the system for the average load plus a margin to account for load variations. The reduction in complexity is achieved by exploiting three features: (a) the inherent pauses in conversations,(b) the sparseness of network echo paths, and (c) the fact that an adaptive filter does not need to be updated when the error signal is small. In this paper it is shown that, in principle, such a design can reduce the computational load by a very large factor -- perhaps as large as thirty. It remains to be seen whether a customized hardware architecture can be implemented to fully take advantage of the proposed algorithm.

1:40, AUDIO-L2.3
AN ACOUSTIC ECHO CANCELLATION STRUCTURE FOR SYNTHETIC SURROUND SOUND
T. YENSEN, R. GOUBRAN
This paper proposes an acoustic echo cancellation structure for hands-free synthetic surround sound applications, such as multiple participant conferencing, and virtual reality applications. Voice over Internet protocol (VoIP) and other virtual reality applications can benefit from the addition of 3D spatial audio generated by more than two loudspeakers. When full-duplex audio is present in a system, however, acoustic echo cancellation is required to eliminate the feedback echo path. The acoustic echo cancellation structure proposed by this paper is based on the acoustic echo canceller per spatial region allocation scheme previously introduced by the authors for two channel synthetic stereo. This paper will show that the spatial region allocation scheme is extensible to any number of channels which makes it extremely versatile and flexible, especially for surround sound audio. Microsoft Direct X 7, a commonly used application programmer interface (API), was used in our simulations to generate the 3D spatial audio on a PC.

2:00, AUDIO-L2.4
LIMITS FOR GENERALIZED SIDELOBE CANCELLERS WITH EMBEDDED ACOUSTIC ECHO CANCELLERS
W. HERBORDT, W. KELLERMANN
This paper analyzes positive synergies and theoretical limits of the combination of acoustic echo cancellation (AEC) and Generalized Sidelobe Cancellers (GSC) for the removal of echoes of loudspeaker signals and local interferers. While the proposed system only requires one AEC for an arbitrary number of microphones, the array gain is limited by the number of sensor channels when all interferers arrive from different directions-of-arrival (DOAs). The paper also shows that the degrees of freedom of the adaptive sidelobe cancelling path are not sufficient when local interferers and acoustic echoes have common DOAs.

2:20, AUDIO-L2.5
EXACT PERFORMANCES ANALYSIS OF A SELECTIVE COEFFICIENT ADAPTIVE ALGORITHM IN ACOUSTIC ECHO CANCELLATION
I. KAMMOUN, M. JAIDANE
In the hands-free communications, the identification of long impulse response in acoustic echo cancellation requires very important load calculation. One way to reduce the complexity of the Normalized Least Mean Square (NLMS) adaptive algorithm, is to use the Mmax NLMS algorithm. It is shown that this algorithm is a very promising one, that maintains a closest performance to the full update NLMS filter in spite of the updating of a small number of coefficients. However, due to its complexity, the mean square analysis uses unrealistic hypothesis. It was then not possible to consider practical context such as high input correlation or high step size. In this paper, we present an exact performances analysis, when the input signal remains in a finite alphabet set. With this realistic hypothesis, dedicated to the digital context, we can describe accurately the Mmax NLMS'behavior without any unrealistic assumption. In particular, we evaluate the exact value of critical and optimal step size and we provide the exact Mean Square Error (MSE) for all step size and input correlation. The influence of high order statistics can be enhanced.

2:40, AUDIO-L2.6
PERFORMANCE IMPROVEMENT OF DOUBLE-TALK DETECTION ALGORITHM IN THE ACOUSTIC ECHO CANCELLER
K. KIM, S. KIM, H. KWON, K. BAE, K. BYUN
This paper deals with the delay problem in the endpoint detection of double-talk detection algorithm in the acoustic echo canceller. In case that past power is much bigger than current power like at the end of double-talking period, the power estimated using forgetting factor decreases slowly to cause the delay problem in the endpoint detection. In this paper two methods are proposed to solve this problem. One is that the current power is periodically replaced by a new average power and the other is that the past power in recursive equation is periodically removed or replaced by other values. The simulation results show that proposed methods outperform conventional method in the endpoint detection of double-talking periods without increasing the computational burden much more.