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Sampling rate conversion using the filter banks is proposed in the field of digital signal processing. However, in sampling rate conversion with a rational factor, the computational complexity may become very large. Moreover whenever the sampling rate is changed, it is necessary to design the filter again. In this paper, hence, we present a kernel with block structure for sampling rate conversion. As the filter proposed has the impulse response approximated by polynomials, it is unnecessary to redesign whenever the sampling rate is changed. The kernel proposed has the block structure that the impulse response of the sampling section since the third is represented by the polynomial used for the second sampling section. Therefore, the filter has less memory. Moreover, the filter has the advantage that it is possible to correspond to arbitrary fractional sampling rate conversion.