ICASSP 2006 - May 15-19, 2006 - Toulouse, France

Technical Program

Paper Detail

Paper:AE-L2.2
Session:Loudspeaker and Microphone Array Processing
Time:Wednesday, May 17, 14:20 - 14:40
Presentation: Lecture
Topic: Audio and Electroacoustics: Loudspeaker and Microphone Array Signal Processing
Title: PROPORTIONATE FREQUENCY DOMAIN ADAPTIVE ALGORITHMS FOR BLIND CHANNEL IDENTIFICATION
Authors: Rehan Ahmad, Andy Khong, Patrick Naylor, Imperial College London, United Kingdom
Abstract: We present fast-converging adaptive blind channel identification algorithms for acoustic room impulse responses. These new algorithms exploit the fast-convergence of the improved proportionate normalized least-mean-square (IPNLMS) algorithm and address the problem of delay inherent in frequency domain algorithms by employing the multi-delay filter (MDF) structure. Simulation results for both speech and white Gaussian noise show that the proposed algorithms outperform current frequency domain blind channel estimation algorithms.



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