DocumentCode
316654
Title
Implementation of an adaptive noise canceller on the TMS320C31-50 for non-stationary environments
Author
Martínez, R. ; Álvarez, A. ; Nieto, V. ; Rodellar, V. ; Gómez, P.
Author_Institution
Dept. de Arquitectura y Tecnologia de Sistemas Inf., Univ. Politecnica de Madrid, Spain
Volume
1
fYear
1997
fDate
2-4 Jul 1997
Firstpage
49
Abstract
The results of the implementation of an adaptive noise cancelling algorithm for speech enhancement on a DSP platform are presented and discussed. Various algorithms where implemented and compared, but time-domain methods related with recursive least squares lattice-ladder filters with a posteriori updating showed the best cancelling behaviour, although at a higher computational cost, therefore, an optimised version of these filters was implemented for the DSP card. The influence of microphonic separation is also evaluated. The best results are obtained for cardioid microphones with a separation of around 20 cm, taking advantage of their angular properties to prevent speech crosstalk in the reference microphone, thus reducing the computational cost required to acceptable limits. This result is very important, since it ensures real time performance of the algorithm in a single-DSP card (farther distances involve more filter stages, which implies a higher computational cost)
Keywords
adaptive filters; adaptive signal processing; circuit optimisation; computational complexity; crosstalk; digital signal processing chips; interference suppression; lattice filters; least squares approximations; microphones; recursive filters; speech enhancement; time-domain synthesis; DSP platform; TMS320C31-50; a posteriori updating; adaptive noise canceller implementation; adaptive noise cancelling algorithm; angular properties; cardioid microphones; computational cost; filter stages; microphonic separation; nonstationary environments; optimised filters; real time performance; recursive least squares lattice-ladder filters; reference microphone; single-DSP card; speech crosstalk; speech enhancement; time-domain methods; Cardiology; Computational efficiency; Digital signal processing; Filters; Least squares methods; Microphones; Noise cancellation; Optimization methods; Speech enhancement; Time domain analysis;
fLanguage
English
Publisher
ieee
Conference_Titel
Digital Signal Processing Proceedings, 1997. DSP 97., 1997 13th International Conference on
Conference_Location
Santorini
Print_ISBN
0-7803-4137-6
Type
conf
DOI
10.1109/ICDSP.1997.627963
Filename
627963
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