Real-time stable adaptive filter based on DSP

By combining IIR filters and adaptive algorithms on the DSP platform, the filter parameters are adjusted in real time, and the problem of poor real-time stable filtering effect in the existing technology is solved, and the filtering effect of efficiently tracking the time-varying characteristics of the input signal in an unknown environment is achieved.

CN119814000APending Publication Date: 2025-04-11CHENGDU JINNUOXIN HIGH-TECH CO LTD
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Patent Information

Application Number
CN202411862345.8
Authority / Receiving Office
CN · China
Patent Type
Applications(China)
Current Assignee / Owner
Filing Date
2024-12-17
Publication Date
2025-04-11

AI Technical Summary

Technical Problem

In the prior art, the real-time stable filtering effect is poor, and it is difficult to work effectively in an unknown environment and track the time-varying characteristics of the input signal.

Method used

Using a DSP-based IIR filter and an adaptive algorithm, the filter parameters are adjusted in real time through the IIR digital filter and the adaptive algorithm module to minimize the mean square value of the error signal, thereby real-time and stable adaptive filtering.

Benefits of technology

It realizes filtering effect with high adaptability and stability in complex real-time environments. The circuit is simple, stable and reliable, and can effectively track the time-varying characteristics of the input signal.

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Abstract

The invention discloses a real-time stable adaptive filter based on a DSP, and relates to the technical field of time frequency. Comprising an analog signal input end, an analog-to-digital converter, a DSP-based IIR filtering control module, a digital-to-analog converter and an analog signal output end which are connected in sequence, the IIR filtering control module based on the DSP comprises a digital input end, an expected response input end, a digital output end, an IIR digital filter, a self-adaptive algorithm module and a subtracter; the IIR digital filter is used for generating response output y (n) according to the digital signal x (n); the subtracter is used for comparing the response output y (n) with the expected response d (n) to form an error signal e (n); and the adaptive algorithm module is used for adjusting parameters of the IIR digital filter according to the error signal e (n), so that the mean square value of the error signal e (n) is minimum. According to the method, the IIR filter and the DSP are combined to adapt to filtering characteristics required in different environments, the current output of the IIR filter depends on current and past input and past output, and higher real-time performance is achieved.
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Description

Technical Field

[0001] The invention relates to the field of time-frequency technology, and in particular to a real-time stable adaptive filter. Background Art

[0002] Adaptive filters are relative to fixed filters. When fixed design specifications are unknown, or the use of time-invariant filters cannot meet the design requirements, adaptive filters are needed.

[0003] The current adaptive filter is designed using FIR filters. According to changes in the test environment, an adaptive algorithm is used to change the filter parameters, such as Figure 1 As shown. The coefficients of the adaptive filter are time-varying coefficients updated by its own algorithm. That is, its coefficients automatically and continuously adapt to the given signal to obtain the desired output. The most important feature of the adaptive filter is that it can work effectively in an unknown environment and can track the time-varying characteristics of the input signal. Summary of the invention

[0004] The purpose of the present invention is to overcome the deficiencies of the prior art, provide a real-time stable adaptive filter based on DSP, and solve the problem of real-time stable filtering in the prior art.

[0005] The objective of the present invention is achieved through the following technical solutions: A real-time stable adaptive filter based on DSP, comprising an analog signal input terminal, an analog-to-digital converter, an IIR filter control module based on DSP, a digital-to-analog converter and an analog signal output terminal connected in sequence; The DSP-based IIR filter control module includes a digital input terminal, a desired response input terminal, a digital output terminal, an IIR digital filter, an adaptive algorithm module and a subtractor; The first input end of the IIR digital filter is connected to the digital input end, the second input end of the IIR digital filter is connected to the output end of the adaptive algorithm module, and the output end of the IIR digital filter is connected to the first input end and the digital output end of the subtractor, for x(n) Producing responsive output y(n) ; The second input terminal of the subtractor is connected to the expected response input terminal, and the output terminal of the subtractor is connected to the input terminal of the adaptive algorithm module, so as to output the response. y(n) Expected response d(n) Compare and form an error signal e (n) ; The adaptive algorithm module is used to e(n), Adjust the IIR digital filter parameters to make the error signal e(n) The mean square value of is the smallest.

[0006] Furthermore, an anti-aliasing filter is connected between the analog signal input terminal and the analog-to-digital converter to make the analog signal waveform clearer.

[0007] Furthermore, a smoothing filter for signal noise reduction is connected between the digital-to-analog converter and the analog signal output terminal.

[0008] Furthermore, the structure of the IIR digital filter adopts a parallel form.

[0009] Furthermore, the IIR digital filter is based on the digital signal x(n) Producing responsive output y(n) It can be expressed as a difference equation: in, x(n) is the current input, y(n) is the current output, b0,…,b k are the input coefficients, a0,…,a k is the output coefficient, M and N represent the input and output orders of the filter respectively.

[0010] The beneficial effects of the present invention are as follows: the present invention adopts a combination of IIR filter and DSP to adapt to the filtering characteristics required in different environments. The current output of the IIR filter depends on the current and past inputs, as well as the output at the past moment, and has stronger real-time performance, and has the following advantages: 1) The circuit is simple, stable and reliable, can be used in complex real-time environments, and has strong adaptability.

[0011] 2) The filtering effect is better by combining adaptive algorithm and feedback.

[0012] 3) Parallel filters are easy to adjust and have smaller errors. BRIEF DESCRIPTION OF THE DRAWINGS

[0013] Figure 1 It is the general structure diagram of adaptive filter; Figure 2 A structural diagram of a real-time stable adaptive filter based on DSP provided by the present invention; Figure 3 This is the internal structure diagram of the parallel form of the IIR digital filter; Figure 4 This is a comparison chart of filtering results. DETAILED DESCRIPTION

[0014] The technical solution of the present invention will be clearly and completely described below in conjunction with the embodiments. Obviously, the described embodiments are only part of the embodiments of the present invention, rather than all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative work are within the scope of protection of the present invention.

[0015] See also Figure 1-Figure 4 , the present invention provides a technical solution: A real-time stable adaptive filter based on DSP, such as Figure 2 As shown, it includes an analog signal input terminal, an analog-to-digital converter, an IIR filter control module based on DSP, a digital-to-analog converter and an analog signal output terminal connected in sequence; The DSP-based IIR filter control module includes a digital input terminal, a desired response input terminal, a digital output terminal, an IIR digital filter, an adaptive algorithm module and a subtractor; The first input end of the IIR digital filter is connected to the digital input end, the second input end of the IIR digital filter is connected to the output end of the adaptive algorithm module, and the output end of the IIR digital filter is connected to the first input end and the digital output end of the subtractor, for x(n) Producing responsive output y(n) ; The second input terminal of the subtractor is connected to the expected response input terminal, and the output terminal of the subtractor is connected to the input terminal of the adaptive algorithm module, so as to output the response. y(n) Expected response d(n) Compare and form an error signal e (n) ; The adaptive algorithm module is used to e(n), Adjust the IIR digital filter parameters to make the error signal e(n) The mean square value of is the smallest.

[0016] In this embodiment, an anti-aliasing filter is connected between the analog signal input terminal and the analog-to-digital converter to make the analog signal waveform clearer.

[0017] In this embodiment, a smoothing filter for signal noise reduction is further connected between the digital-to-analog converter and the analog signal output terminal.

[0018] like Figure 3 As shown, the structure of the IIR digital filter adopts a parallel form. Compared with the cascade form, it is easy to adjust, can minimize errors, and improve the stability of the entire system.

[0019] Furthermore, the IIR digital filter is based on the digital signal x(n) Producing responsive output y(n) It can be expressed as a difference equation: in, x(n) is the current input, y(n) is the current output, b0,…,b k are the input coefficients, a0,…,a k is the output coefficient, M and N represent the input and output orders of the filter respectively.

[0020] The filtering principle of the present invention is as follows: the analog input signal is converted into a digital signal through an anti-aliasing filter to make the signal waveform clearer. x(n) Generate response through adaptive filter y(n) , and compare it with the expected response d(n) Compare and form an error signal e(n) , and adjust the filter parameters through adaptive algorithm, and finally make e(n) The mean square value is the smallest, and the feedback circuit is combined to make it real-time. After obtaining the ideal signal, it is converted back to an analog signal, and then subjected to a noise reduction process by a smoothing filter to output the final signal.

[0021] In order to optimize the filter adaptability, the present invention omits the FIR filter and adopts the combination of IIR filter and DSP to adapt to the required filtering characteristics in different environments. The current output of the IIR filter depends on the current and past inputs, as well as the output at past moments, and has stronger real-time performance.

[0022] Figure 4 This is a comparison chart of the results of a general adaptive filter and the adaptive filter of the present invention. It can be seen that the output signal of the adaptive filter of the present invention is significantly better than that of the traditional adaptive filter, the effect is more stable, and it can be used in a real-time system.

[0023] The above is only a preferred embodiment of the present invention. It should be understood that the present invention is not limited to the form disclosed herein, and should not be regarded as excluding other embodiments, but can be used in various other combinations, modifications and environments, and can be modified within the scope of the concept described herein through the above teachings or the technology or knowledge of the relevant field. The changes and modifications made by those skilled in the art shall not deviate from the spirit and scope of the present invention, and shall be within the scope of protection of the claims attached to the present invention.

Claims

1. A real-time stable adaptive filter based on DSP, characterized in that: It includes an analog signal input terminal, an analog-to-digital converter, an IIR filter control module based on DSP, a digital-to-analog converter and an analog signal output terminal which are connected in sequence; The DSP-based IIR filter control module includes a digital input terminal, a desired response input terminal, a digital output terminal, an IIR digital filter, an adaptive algorithm module and a subtractor; The first input end of the IIR digital filter is connected to the digital input end, the second input end of the IIR digital filter is connected to the output end of the adaptive algorithm module, and the output end of the IIR digital filter is connected to the first input end and the digital output end of the subtractor, for x(n) Producing responsive output y(n) ; The second input terminal of the subtractor is connected to the expected response input terminal, and the output terminal of the subtractor is connected to the input terminal of the adaptive algorithm module, so as to output the response. y(n) Expected response d(n) Compare and form an error signal e(n) ; The adaptive algorithm module is used to e(n), Adjust the IIR digital filter parameters to make the error signal e(n) The mean square value of is the smallest.

2. A real-time stable adaptive filter based on DSP according to claim 1, characterized in that: An anti-aliasing filter is also connected between the analog signal input terminal and the analog-to-digital converter to make the analog signal waveform clearer.

3. A real-time stable adaptive filter based on DSP according to claim 1, characterized in that: A smoothing filter for signal noise reduction is also connected between the digital-to-analog converter and the analog signal output terminal.

4. A real-time stable adaptive filter based on DSP according to claim 1, characterized in that: The structure of the IIR digital filter adopts a parallel form.

5. A real-time stable adaptive filter based on DSP according to claim 1, characterized in that: The IIR digital filter is based on the digital signal x(n) Producing responsive output y(n) It can be expressed as a difference equation: in, x(n) is the current input, y(n) is the current output, b0,…,b k are the input coefficients, a0,…,a k is the output coefficient, M and N represent the input and output orders of the filter respectively.