Sampling Rate Converison by a Fractional Factor

Author(s):  
Ljiljana Milic

We have discussed so far the decimation and interpolation where the sampling rate conversion factor is an integer. However, the need for a non-integer sampling rate conversion appears when the two systems operating at different sampling rates have to be connected, or when there is a need to convert the sampling rate of the recorded data into another sampling rate for further processing or reproduction. Such applications are very common in telecommunications, digital audio, multimedia and others. In this chapter, we consider the sampling rate conversion by a rational factor, called sometimes a fractional sampling rate conversion. We use MATLAB functions from the Signal Processing and Filter Design Toolbox to demonstrate the fractional sampling rate conversion. We present the technique for constructing efficient fractional sampling rate converters based on FIR filters and the polyphase decomposition. In the sequel, we consider the sampling rate alteration with an arbitrary conversion factor. We present the polynomial-based approximation of the impulse response of a hybrid analog/digital model, and the implementation based on the Farrow structure. We also consider the fractional-delay filter problem. This chapter concludes with MATLAB exercises for individual study.

2014 ◽  
Vol 687-691 ◽  
pp. 4093-4096
Author(s):  
Yan Xue ◽  
Fei Yang

At present, in the digital audio processing sampling rate is respectively 32 kHz, 44.1 kHz, 48 kHz [1]. Because of the different criteria, there is much inconvenience in the process of research. Therefore, the sampling rate converter is a must, between any two kinds of sampling rate. In synchronous sampling rate conversion, you can use decimation and interpolation for sampling rate conversion, but in the asynchronous sampling rate system, due to the different input clock pulse with the output clock pulse, the above method cannot achieve. Therefore we introduce the fractional delay filter sampling rate conversion. This article introduces the principle of the sampling rate conversion and the fractional delay filter based on Farrow structure. At last, we simulate asynchronous sampling rate conversion of audio signal through the MATLAB.


2015 ◽  
Vol 39 (2) ◽  
pp. 231-242 ◽  
Author(s):  
Marek Blok ◽  
Piotr Drózda

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.


2002 ◽  
pp. 105-142 ◽  
Author(s):  
Ljiljana D. Milic ◽  
Miroslav D. Lutovac

Application of multirate techniques to improve digital filter design and implementation are considered in this chapter. FIR and IIR filter design and implementation for sampling rate conversion by integer and rational factors are presented. Sharp narrow-band and wide-band multirate design techniques are discussed. Accurate designs of FIR and IIR half-band filters are described in detail. Several examples are provided to illustrate the multirate approach to filter design.


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