Fractional sampling rate conversion method for complex coefficient Hilbert band-pass filter

Abstract

The invention provides a fractional sampling rate conversion method for a complex coefficient Hilbert band-pass filter. A real coefficient band-pass filter with the sampling rate of L*fsi is designed by engineering simulation design software Matrix Laboratory (MATLAB), then an impulse response sequence of real numbers hr(n) is converted into an impulse response sequence of complex numbers hc(n) by a Hilbert transformation function provided by the MATLAB, and the hc(n) is taken as the complex coefficient of the Hilbert band-pass filter; an L phase subfilter complex coefficient is acquired, and the L phase subfilter is realized through an implementation structure of a common finite impulse response (FIR) filter in a field programmable gate array (FPGA), so that an L times of parallel interpolation filter structure with increased sampling rate is formed; M times of extraction rate conversion is realized through write operation of a first in first out (FIFO) data buffer, and through independent write operation on the FIFO in a clock domain with sampling rate of fsi and independent read operation on the FIFO in the clock domain with the output sampling rate of fso=(L / M)*fsi, direct data transmission between the clock domain with the sampling rate of fsi and the clock domain with the sampling rate of fso is isolated, so that fractional rate L / M is changed.

Description

technical field [0001] The invention relates to a complex coefficient Hilbert bandpass filter for broadband complex signal processing in a digital signal processing system, in particular to a method for converting a fractional sampling rate by using the complex coefficient Hilbert bandpass filter. Background technique [0002] In the prior art, the sampling rate conversion technology is also called the resampling technology, which is a key technology in software radio digital signal processing and is widely used. The sampling rate is the sampling frequency of the analog signal waveform per unit time (for example, per second). The higher the sampling rate, the smaller the distortion of the original analog signal represented by the digital signal obtained by sampling. Using a high sampling rate means that the distortion of the analog signal is small, but the amount of data per unit time is large, which means that the required storage space is large, and it puts forward high ha...

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Problems solved by technology

[0009] The object of the present invention is to provide a kind of wideband complex signal from input sampling rate f to the problem existing in the above-mentioned prior art signal processing si to the output sampling rate f so =(L / M)*f si The implementation method of L / M times interpolation sampling rate conversion is used to solve the problem that the general real coefficient band-pass filter cannot completely suppress the image spectrum in the process of band-pass complex signal interpolation

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