PEVD-based Speech Enhancement
This website will demonstrate the PEVD-based Speech Enhancement algorithm developed by Vincent W. Neo, Christine Evers and Patrick A. Naylor. This is a joint work between the Speech and Audio Processing Lab at Imperial College London and the School of Electronics and Computer Science at University of Southampton.
The algorithm is demonstrated under 3 types of acoustic environments: noise-only (Noise Reduction), reverberation-only (Dereverberation), and noisy reverberant (Enhancement).
About This Work
Clean speech signals are taken from the TIMIT corpus. Room impulse response (RIR) and noise signals are taken from the 3-channel mobile phone recordings from the ACE corpus. Additional diffuse noise are generated using signals from NOISEX database and International Sound Effects Library. Noisy reverberant speech is generated by convolving the anechoic speech with the RIR before adding diffuse noise at the required SNR for each microphone.
The speech enhancement approaches used in the comparison include Log-MMSE, Single-channel subspace for coloured noise (COLSUB), Multi-channel subspace method (MCSUB), 2 versions of Multi-channel Wiener filter including a practical (MWF) and Oracle version (OMWF), Generalized Weighted Prediction Error (GWPE), Weighted Power minimisation Distortionless (WPD), the Integrated Sidelobe Canceller and Linear Prediction (ISCLP) Kalman filter, and the proposed (PEVD)-based algorithm [1].
[1] V. W. Neo, C. Evers, and P. A. Naylor, "Enhancement of noisy reverberant speech using polynomial matrix eigenvalue decomposition," IEEE/ACM Trans. Audio, Speech and Lang. Process., vol. 28, 2021. doi: 10.1109/TASLP.2021.3120630
Contact Us
Email: vincent.neo09@imperial.ac.uk
Visit Our Websites: SAP | Vincent W. Neo | Christine Evers | Patrick A. Naylor