Hardware Optimized Sample Rate Conversion for Software Defined Radio

Frequenz ◽  
2010 ◽  
Vol 64 (11-12) ◽  
Author(s):  
Carina Schmidt-Knorreck ◽  
Raymond Knopp ◽  
Renaud Pacalet
Keyword(s):  
Software Defined Radio ◽  
Sample Rate ◽  
Rate Conversion

scholarly journals Modified CIC Filter for Rational Sample Rate Conversion

1970 ◽  
Vol 4 (1) ◽  
pp. 15-20 ◽  
Author(s):  
Gordana Jovanovic Dolecek
Keyword(s):  
Impulse Response ◽  
Software Defined Radio ◽  
Conversion Factor ◽  
Sampling Rate ◽  
Prime Numbers ◽  
Comb Filter ◽  
Triangular Form ◽  
Cic Filter ◽  
Sample Rate ◽  
Rate Conversion

The modification of the conventional CIC (cascadedintegrator-comb) filter for rational sample rate conversion (SRC) in software defined radio (SWR) systems is presented here. The conversion factor is a ratio of two mutually prime numbers, where the decimation factor M can be expressed as a product of two integers. The overall filter realization is based on a stepped triangular form of the CIC impulse response, the corresponding expanded cosine filter, and sine-based compensation filter. This filter performs sampling rate conversion efficiently by using only additions/subtractions making it attractive for software defined radio (SWR) applications.

Variable Ratio Sample Rate Conversion Based on Fractional Delay Filter

2015 ◽  
Vol 39 (2) ◽  
pp. 231-242 ◽  
Author(s):  
Marek Blok ◽  
Piotr Drózda
Keyword(s):  
Speech Signal Processing ◽  
Farrow Structure ◽  
Conversion Algorithm ◽  
Window Method ◽  
Voiced Speech ◽  
Fractional Delay ◽  
Sample Rate ◽  
Fractional Delay Filter ◽  
Speech Segments ◽  
Rate Conversion

Abstract In this paper a sample rate conversion algorithm which allows for continuously changing resampling ratio has been presented. The proposed implementation is based on a variable fractional delay filter which is implemented by means of a Farrow structure. Coefficients of this structure are computed on the basis of fractional delay filters which are designed using the offset window method. The proposed approach allows us to freely change the instantaneous resampling ratio during processing. Using such an algorithm we can simulate recording of audio on magnetic tape with nonuniform velocity as well as remove such distortions. We have demonstrated capabilities of the proposed approach based on the example of speech signal processing with a resampling ratio which was computed on the basis of estimated fundamental frequency of voiced speech segments.

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