AE-1.1

Synthesized Stereo Combined with Acoustic Echo Cancellation for Desktop Conferencing
Jacob Benesty, Dennis R Morgan, Joseph L Hall, Man M Sondhi (Bell Labs)

One promising application in modern communications is desktop conferencing, which can involve several participants over a widely distributed area. Synthesized stereophonic sound will enable a listener to spatially separate one remote talker from another and thereby improve understanding. In such a scenario, we assume we are located in a hands-free environment where the composite acoustic signal is presented over loudspeakers, thus requiring acoustic echo cancellation. In this paper, we explain some of the methods that can be used to synthesize stereo sound and how such methods can be combined efficiently with stereo acoustic echo cancellation in the face of several difficult problems.

AE-1.2

A Stereo Echo Canceller Implemented Using a Stereo Shaker and a Duo-Filter Control System
Suehiro Shimauchi, Shoji Makino, Yoichi Haneda, Akira Nakagawa, Sumitaka Sakauchi (NTT Human Interface Laboratories)

Stereo echo cancellation has been achieved and used in daily teleconferencing. To overcome the non-uniqueness problem, a stereo shaker is introduced in eight frequency bands and adjusted so as to be inaudible and not affect stereo perception. A duo-filter control system including a continually running adaptive filter and a fixed filter is used for double-talk control. A second-order stereo projection algorithm is used in the adaptive filter. A stereo voice switch is also included. This stereo echo canceller was tested in two-way conversation in a conference room, and the strength of the stereo shaker was subjectively adjusted. A misalignment of 20 dB was obtained in the teleconferencing environment, and changing the talker's position in the transmission room did not affect the cancellation. This echo canceller is now used daily in a high-presence teleconferencing system and has been demonstrated to more than 300 attendees.

AE-1.3

Convergence Analysis of Stereophonic Echo Canceller with Pre-Processing - Relation between Pre-Processing and Convergence -
Akihiro Hirano, Kenji Nakayama, Kazunobu Watanabe (Dept. of Electrical and Computer Eng., Faculty of Eng., Kanazawa Univ.)

This paper presents convergence characteristics of stereophonic echo cancellers with pre-processing. The convergence analysis of the averaged tap-weights show that the convergence characteristics depends on the relation between the impulse response in the far-end room and the changes of the pre-processing filters. Examining the uniqueness of the solution in the frequency domain leads us to the same relation. Computer simulation results show the validity of these analyses.

AE-1.4

Implicit Decimation for FIR Systems and Its Application to Acoustic Echo Cancellation
Walter A Frank, Imre Varga (Siemens AG, Mobile Phones Dept.)

This paper presents a filter structure which performs implicit decimation of the impulse response. As a result, the number of required operations is reduced or, equivalently, the impulse response length of the filter can be increased. Analysis in the frequency domain shows that this implicit decimation can be applied to systems that exhibit low-pass characteristics or have a smooth transfer function at high frequencies. Such behaviour can be assumed for many technical systems. For the determination of the optimal coefficients many well known algorithms for FIR systems can be used after a slight modification of the signal vector. The performance of implicit decimation is demonstrated for acoustic echo cancellation. Comparison with different algorithms shows that implicit decimation outperforms conventional FIR filtering.

AE-1.5

A BLOCK LEAST SQUARES APPROACH TO ACOUSTIC ECHO CANCELLATION
Eric A Woudenberg (ATR Human Information Processing Laboratories, Kyoto, Japan), Frank K Soong, B.H. Juang (Bell Laboratories - Lucent Technologies, Murray Hill,New Jersey, USA)

We propose an efficient block least-squares (BLS) algorithm for acoustic echo cancellation. The high computation and memory requirements associated with a long room echo make the simple, gradient-based LMS filter a more acceptable commercial solution than a full-fledged LS canceler. However, the LMS echo canceler has slower convergence and worse steady-state performance than its LS counterpart. In the proposed BLS approach, the autocorrelation and cross-correlation of the source and echo, required in solving the LS normal equations, are performed once per block using FFT's. With appropriate data windowing the autocorrelation matrix is constrained to be Toeplitz, allowing the corresponding normal equations to be solved efficiently. The positive definiteness of the autocorrelation function eliminates the stability problems of other fast LS algorithms. BLS can reduce the echo residual to the level of background noise, allowing a residual power based, statistical near-end speech detector to be devised. Performance in real environments under various settings of filter length, SNR, near-end speech presence, etc., is investigated.

AE-1.6

A Postfilter for Echo and Noise Reduction Avoiding the Problem of Musical Tones
Stefan N.A. Gustafsson, Peter J. Jax, Axel Kamphausen, Peter Vary (Institute of Communication Systems and Data Processing, RWTH Aachen, Templergraben 55, D-52056 Aachen, Germany)

In this paper we address the problem of acoustic echo cancellation and noise reduction for narrow and wide band telephone applications. We combine a conventional echo canceller with a postfilter implemented in the frequency domain and derive an algorithm for the simultaneous attenuation of residual echo and noise. The main goals are a low level natural sounding background noise without artifacts such as musical tones, and an inaudible residual echo. This is achieved by considering the masking properties of the human auditory system. Simulation results verify that these goals are reached, while the distortion of the near end speech is comparable to conventional algorithms. Audio demonstrations are available via Internet from http://www.ind.rwth-aachen.de.

AE-1.7

Nonlinear Acoustic Echo Cancellation with 2nd Order Adaptive Volterra Filters
Alexander Stenger, Lutz Trautmann, Rudolf Rabenstein (University of Erlangen-Nuremberg, Telecommunications Laboratory, Erlangen, Germany)

Acoustic echo cancellers in today's speakerphones or video conferencing systems rely on the assumption of a linear echo path. Low-cost audio equipment or constraints of portable communication systems cause nonlinear distortions, which limit the echo return loss enhancement achievable by linear adaptation schemes. These distortions are a superposition of different effects, which can be modelled either as memoryless nonlinearities or as nonlinear systems with memory. Proper adaptation schemes for both cases of nonlinearities are discussed. An echo canceller for nonlinear systems with memory based on an adaptive second order Volterra filter is presented. Its performance is demonstrated by measurements with small loudspeakers. The results show an improvement in the echo return loss enhancement of 7 dB over a conventional linear adaptive filter. The additional computational requirement for the presented Volterra filter is comparable to that of existing acoustic echo cancellers.

AE-1.8

On the Poor Robustness of Sound Equalization in Reverberant Environments
Biljana D Radlovic, Robert C Williamson (Department of Engineering, Faculty of Engineering and Information Technology, Australian National University), Rodney A Kennedy (Telecommunications Engineering Group, Research School of Information Sciences and Engineering, The Institute of Advanced Studies, Australian National University)

This paper examines the sensitivity of sound equalization to source or microphone position changes in a reverberant room. It is demonstrate that even small displacements from the reference (equalization) point, of the order of a tenth of the acoustic wavelength, can cause large degradations in the equalized room response. The general theory developed in this paper, which implies that the sound equalization in practical environments may be an ill-posed problem, is verified by the simulation results averaged over different source and microphone positions.

AE-1.9

On the performance of a local active noise control system
Maria De-Diego, Alberto Gonzalez (Dept. of Comunicaciones, Universidad Politecnica de Valencia), Clemente Garcia

This paper presents a multichannel active system for the local control of sound around the headrest on the back of a seat placed inside an enclosure. The size of the zones of quiet produced makes the system practical only at relatively low frequencies. Finally, some results of cancellation for narrowband and broadband noise are presented. Two different system configurations algorithms have been tested on the adaptive controller. Both of them show similar results, but the new algorithm based on the minimization of the maximum error signal power, has shown computational saving and higher speed convergence than the multiple channel least squares algorithm.

AE-1.10

High Quality Signal Reception in the Presence of Stationary Interference-A Blind Signal Separation Approach
Wai Kuen Lai (Techical Services Division), Ting Wai Siu, Sze Fong Yau (Technical Services Division)

This paper considers the problem of interference rejection using sensor array with application to enhance signal reception. In contrast to conventional adaptive beamforming which requires knowledge of the array geometry and array response, we propose a two stage blind signal separation approach to achieve interference rejection. The proposed method is based on the practically viable assumptions that the desired signal is temporally non-stationary and the interference are temporally second-order stationary, and is applicable to arrays with uncalibrated response and geometry. Real-world experimental results are presented to demonstrate the performance of the proposed method.

AE-2 >

Last Update: February 4, 1999 Ingo Höntsch