Approximation of optimal step size control for acoustic echo cancellation

Antweiler, Christiane; Grunwald, Jan-Eric; Quack, H.

doi:10.1109/icassp.1997.599627

Cited by 14 publications

(9 citation statements)

References 2 publications

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“…That is, bounding the error signal can be interpreted as a reduction of the step-size parameter. What differentiates our approach is that while traditional variable step-size methods [6], [7], try to detect periods of double-talk and then take action, in our robust technique we use the signal before double-talk in order to be prepared for it. Due to this fact, the robust technique is faster and more efficient.…”

Section: Discussionmentioning

confidence: 99%

“…The robust algorithm developed here combines the PNLMS or PNLMS , [2], [3] algorithm with the appropriate nonlinearity. Another method to improve the overall performance is to allow variability of the global step-size parameter , [6], [7]. However, the algorithm by which the optimal step-size is found is fairly complex and it is difficult to adjust it fast enough when double-talk occurs.…”

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confidence: 99%

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Double-talk robust fast converging algorithms for network echo cancellation

Gänsler¹,

Gay²,

Sondhi³

et al. 2000

IEEE Trans. Speech Audio Process.

134

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Abstract-There is a need for echo cancelers for echo paths with long impulse responses ( 64 ms). This in turn creates a need for more rapidly converging algorithms in order to meet the specifications for network echo cancelers. Faster convergence, however, in general implies a higher sensitivity to near-end disturbances, especially "double-talk." Recently, a fast converging algorithm has been proposed called proportionate normalized least mean squares (PNLMS) algorithm. This algorithm exploits the sparseness of the echo path and has the advantage that no detection of active coefficients is needed. In this paper we propose a method for making the PNLMS algorithm more robust against double-talk. The slower divergence rate of these algorithms in combination with a standard Geigel double-talk detector improves the performance of a network echo canceler considerably during double-talk. The principle is based on a scaled nonlinearity which is applied to the residual error signal. This results in the robust PNLMS algorithm which diverges much slower than PNLMS and standard NLMS. Tradeoff between convergence and divergence rate is easily adjusted with one parameter and the added complexity is about seven instructions per sample which is less than 0.3% of the total load of a PNLMS algorithm with 512 filter coefficients. A generalization of the robust PNLMS algorithm to a robust proportionate affine projection algorithm (APA) is also presented. It converges very fast, and unlike PNLMS, is not as dependent on the assumption of a sparse echo path response. The complexity of the robust proportionate APA of order two is roughly the same as that of PNLMS.Index Terms-Adaptive filter, double-talk detection, echo cancellation, normalized least mean squares (NLMS), proportionate normalized least mean squares (PNLMS), robustness.

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Section: Discussionmentioning

confidence: 99%

mentioning

confidence: 99%

Double-talk robust fast converging algorithms for network echo cancellation

Gänsler¹,

Gay²,

Sondhi³

et al. 2000

IEEE Trans. Speech Audio Process.

134

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show abstract

“…In cases where the shift is more significant, it can freeze the system. This particular problem has been addressed by several authors earlier [16][17][18]. The problem of changes in the true echo paths causes the estimated echo path to be different from the current echo path.…”

Section: Accommodating the Altering Position Of Belt Microphonesmentioning

confidence: 99%

Signal processing techniques for seat belt microphone arrays

Rajan

Krini²,

Rodemer³

et al. 2016

EURASIP J. Adv. Signal Process.

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Microphones integrated on a seat belt are an interesting alternative to conventional sensor positions used for handsfree telephony or speech dialog systems in automobile environments. In the setup presented in this contribution, the seat belt consists of three microphones which usually lay around the shoulder and chest of a sitting passenger. The main benefit of belt microphones is the small distance from the talker's mouth to the sensor. As a consequence, an improved signal quality in terms of a better signal-to-noise ratio (SNR) compared to other sensor positions, e.g., at the rear view mirror, the steering wheel, or the center console, can be achieved. However, the belt microphone arrangement varies considerably due to movements of the passenger and depends on the size of the passenger. Furthermore, additional noise sources arise for seat belt microphones: they can easily be touched, e.g., by clothes, or might be in the path of an air-stream from the automotive ventilation system. This contribution presents several robust signal enhancement algorithms designed for belt microphones in multi-seat scenarios. The belt microphone with the highest SNR (usually closest to the speaker's mouth) is selected for speech signal enhancement. Further improvements can be achieved if all belt microphone signals are combined to a single output signal. The proposed signal enhancement system for belt microphones includes a robust echo cancelation scheme, three different microphone combining approaches, a sophisticated noise estimation scheme to track stationary as well as non-stationary noise, and a speech mixer to combine the signals from each seat belt to a single channel output in a multi-seat scenario.

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“…Andere Vorschläge gehen von der Tatsache aus, daß die Impulsantwort eines LRM-Systems exponentiell abklingt und bewerten die Verbesserung der einzelnen Koeffizienten entsprechend der Einhüllenden der Impulsantwort [17,18].…”

Section: Adaption Des Echokompensatorsunclassified

Freisprechen - Die Jagd nach akustischen Echos

Breining¹,

Hänsler²,

Schertler³

1997

Frequenz

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Übersicht:Befinden sich Lautsprecher und Mikrofon einer elektroakustischen Anlage im selben Raum, so nimmt das Mikrofon das Lautsprechersignal und seine Echos wieder auf. In rückgekoppelten Systemen kann dies zu Instabilitäten führen. Darüber hinaus hören Gesprächsteilnehmer ihr durch die Laufzeit des Systems verzögertes Echo. Ihre Unterhaltung kann dadurch wesentlich erschwert werden. Der vorliegende Beitrag beschäftigt sich mit dem Problem "Freisprechen" in Telefonsystemen. Er zeigt, welche besonderen Probleme bei dieser Anwendung auftreten und wieweit diese heute gelöst sind. Er konzentriert sich dann auf zwei Teilprobleme, deren Lösung wesentliche Verbesserungen erwarten lassen.Abstract: A microphone located in the same enclosure with the loudspeaker of an electro acoustic system picks up the loudspeaker output signal and its echoes. This may lead to instabilities. In addition, hearing delayed echoes of the own speech impedes a conversation. This paper deals with the hands-free telephone problem. The problem is explained and existing solutions are pointed out. Finally we focus onto two subproblems. Their solutions promise considerable improvements.Für die Dokumentation: Sprachverarbeitung / Echokompensation / adaptive Filter / digitale Signalverarbeitung / Fuzzy Control / Implementierung mit Festkomma-Arithmetik / Freisprechen

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Approximation of optimal step size control for acoustic echo cancellation

Cited by 14 publications

References 2 publications

Double-talk robust fast converging algorithms for network echo cancellation

Double-talk robust fast converging algorithms for network echo cancellation

Signal processing techniques for seat belt microphone arrays

Freisprechen - Die Jagd nach akustischen Echos

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