Acoustic Echo Cancellation Algorithms Research Articles

Alternating fixed-point algorithm for stereophonic acoustic echo cancellation

A novel algorithm for use in stereophonic acoustic echo cancellation is introduced. The alternating fixed-point structure of the algorithm descends each of the channels alternatively and avoids nonuniqueness in the solutions by employing inter- and intra-channel orthogonal projections. For a noisy environment, the algorithm is shown to provide improved misalignment performance and echo cancellation over leading competitors at a low computational cost. The inherent noise robustness is uncovered using H∞ estimation techniques and noise-error bounds are derived.

State detection for hands-free telephone sets by means of fuzzy LVQ and SOM

The step gain of adaptive algorithms for acoustic echo cancellation should vary with time in dependence of the far-end signal, the local signal, and the adaptation quality. It can be described as state-dependent, which is also the case for the control parameters of other algorithms in a hands-free telephone set, such as adaptive loss control or noise reduction. Most of the known state detection methods can only distinguish between subsets of the states, and are unreliable for certain state transitions, so that they should be combined for improved reliability of the state detection. In this contribution, state detection is performed by a fuzzy learning vector quantization/self-organizing map approach to combine several state detectors. The step gain is inferred from state information using fuzzy logic concepts. Finally, adaptation results are compared to the best conventional step-gain control method used in the complete structure.

Cascaded power symmetric IIR filter banks and continuity constrained adaptive algorithms for acoustic echo cancellation in subbands

Teleconferencing systems and hands-free mobile terminals use acoustic echo cancellation (AEC) for high-quality full-duplex speech communication. The problem of aliasing in subband AEC is addressed. Filter banks with implicit notch filtering are derived from cascaded power symmetric-infinite impulse response (CFS-IIR) filters. It is shown that adaptive filters used with these filter banks must be coupled via continuity constraints to reduce the aliasing in the residual echo. A continuity constrained NLMS algorithm is therefore proposed and evaluated.

An adaptation control for acoustic echo cancellers

Acoustic echo cancellation algorithms need a reliable adaptation control to cope with changing systems and nonstationary signals. A variable stepsize is proposed that is based on the detection of speakers' activities, and involves estimates of system parameters and background noise. Real-time experiments demonstrate the workability in practice.

Combined acoustic echo cancellation, dereverberation and noise reduction: a two microphone approach

This paper presents new algorithms for acoustic echo cancellation and noise reduction which use two (or possibly more) microphone signals. In contrast to the single microphone method the multimicrophone approach can exploit the spatial coherence properties of sound fields which arise from noise and reverberated speech. Besides the standardfir echo canceller the proposed algorithms comprise an adaptive filter to eliminate non coherent signal components. The combined system achieves better Erle than thefir echo canceller alone, attenuates ambient noise, dereverberates near end speech, and possibly leads to implementations with reduced complexity. The paper analyzes the acoustical properties of typical environments, presents the algorithms and experimental results.

Acoustic echo cancellation with a transform domain adaptive filter

The adaptive filtering technique is one of the major approaches to control the acoustic echoes in teleconferencing systems or hands‐free telephony systems. A brief overview of the performance of different adaptive algorithms for acoustic echo cancellation [cf, S. M. Kuo and H. Zhao, J. Acoust. Soc. Am. Suppl. 1 86, S56 (1989)] will be presented. A transform domain adaptive filter (TDAF) algorithm is proposed to construct an acoustic echo canceler that has a convergence rate faster than that of the conventional LMS adaptive transversal filter with colored signals. The proposed approach does not introduce extra delay during the process so that it can be used in the application where delay is not allowed. Different orthogonal transforms such as the discrete Fourier transform (DFT), discrete cosine transform (DCT), and discrete Hartley transform (DHT) are used for signals with different eigenvalue spread. Real‐time implementations of this class of canceler with digital signal processors (DSP) will also be discussed.

Adaptive acoustic echo cancellation algorithms in teleconferencing systems

In teleconferencing systems, acoustic echoes consist of acoustic coupling between louder speakers and microphones and the room reverberations. The echoes could be very annoying and even cause howling if the echo level is high enough. Among many ways of canceling acoustic echoes in teleconferencing system, the adaptive digital signal processing approach is very promising since adaptive filtering has the capability to trace the change of the echo path and cancel echoes actively. Adaptive algorithms to cancel acoustic echoes in both the time and frequency domains are presented, and the computer simulation results are compared in a proposed paper. The algorithms included are LMS, NLMS, LLMS, SDLMS, SELMS, SSLMS for FIR adaptive filtering and RLMS, SHARF, and Pole-Zero for IIR adaptive filtering. Performance of these adaptation algorithms will be discussed. Subband adaptive filtering algorithm, which has a faster convergence rate, will also be presented.