Convolution algorithms on DSP processors

Author

Meyer, R. ; Reng, R. ; Schwarz, K.

Author_Institution

Lehrstuhl fuer Nachrichtentech., Erlangen-Nurnberg Univ., Germany

fYear

1991

fDate

14-17 Apr 1991

Firstpage

2193

Abstract

For state-of-the-art digital signal processors (DSPs) alternative ways to implement FIR (finite impulse response) filters are compared with respect to the necessary number of instruction cycles per tap. Also, the accuracy attainable both with fixed-point and floating-point arithmetic is discussed. Besides direct form filtering and fast convolution algorithms using FFTs (fast Fourier transforms), radix FIR filter structures are also considered. The latter are very interesting for most DSP architectures, offering speed improvements of up to 25% compared to the direct form

Keywords

computerised signal processing; digital arithmetic; digital filters; digital signal processing chips; fast Fourier transforms; DSP architectures; DSP processors; decimated subfilters; digital signal processors; direct form filtering; fast Fourier transforms; fast convolution algorithms; finite impulse response; fixed point arithmetic; floating-point arithmetic; instruction cycles per tap; radix FIR filter; Convolution; Digital filters; Digital signal processing; Digital signal processors; Fast Fourier transforms; Filtering algorithms; Finite impulse response filter; Flexible printed circuits; Floating-point arithmetic; Signal processing algorithms;

fLanguage

English

Publisher

ieee

Conference_Titel

Acoustics, Speech, and Signal Processing, 1991. ICASSP-91., 1991 International Conference on

Conference_Location

Toronto, Ont.

ISSN

1520-6149

Print_ISBN

0-7803-0003-3

Type

conf

DOI

10.1109/ICASSP.1991.150849

Filename

150849