Adaptive identification method and device, and adaptive echo...

Electrical audio signal processing systems and devices – Dereverberators

Reexamination Certificate

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C379S406010, C708S322000

Reexamination Certificate

active

06895094

ABSTRACT:
The device receives an input signal xtand an observation signal yt, one component of which is the response of a system to the input signal. It determines an error signal etat an instant t in accordance with the equation et=yt−XtTHt−1, where Ht−1is a vector made up of L coefficients of an identification filter having a finite impulse response representative of the impulse response of the system and XtT=(xt, xt−1, . . . , xt−L+1). A predictive analysis of the input signal is performed by a computation applied to frames of the input signal. The L coefficients of the identification filter are adapted by adding to the vector Ht−1a vector proportional toetXtT⁢Ut+λ⁢Ut,where Utis a vector made up of the L values of a prediction residue of the input signal at the instant t and at the preceding L−1 instants, and λ is a positive or zero coefficient.

REFERENCES:
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patent: 5428562 (1995-06-01), Gay
patent: 5570423 (1996-10-01), Walker et al.
patent: 5734715 (1998-03-01), Scalart et al.
patent: 6788785 (2004-09-01), Ding
P Scalart et al., [a system for speech enhancement in the context of hands-free radiotelephony with combined noise reduction and acoustic echo cancellation], speech communication, 1996, vol. 20 No 3, pp. 203-214.*
P Scalart et al., <<A System for Speech Enhancement in the Context of Hands-Free Radiotelephony with Combined Noise Reduction and Acoustic Echo Cancellation >>, Speech Communication, 1996, vol. 20 No. 3, pp. 203-214.

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