Dereverberation of speech signals in a hands-free scenario by inverse filtering has been a research topic for several years now. However, it is still a challenging problem because of the nature of common room impulse responses (RIRs), which are time-variant mixed phase systems having a large number of zeros close to, on, and even outside the unit circle in the z-domain. In this contribution an adaptive multi-channel equalization algorithm based on a decoupled version of the modified filtered-X LMS (mFxLMS) will be derived in the partitioned frequency domain. This new algorithm allows for fast convergence, computationally efficient implementation, and a low system delay under realistic conditions such as ambient noise and imperfect RIR estimates.
|Title of host publication
|2008 42nd Asilomar Conference on Signals, Systems and Computers
|Number of pages
|Published - 01.10.2008
|2008 42nd Asilomar Conference on Signals, Systems and Computers - Pacific Grove, United States
Duration: 26.10.2008 → 29.10.2008
Conference number: 77519