CN104678170A - Power harmonic analysis method based on harmonic analyzer and harmonic analyzer - Google Patents

Abstract

The invention discloses a power harmonic analysis method based on a harmonic analyzer and a harmonic analyzer, wherein the method includes: determining a preset fundamental frequency, and designing a first FIR comb filter for an input signal according to the preset fundamental frequency filter; filter the input signal through the first FIR comb filter, and determine the frequency of the filtered signal; when the difference between the frequency of the filtered signal and the preset base frequency is less than the preset threshold, the filtered signal The frequency of the input signal is used as the fundamental frequency of the input signal; the frequency of the harmonic is determined according to the fundamental frequency, and a second FIR comb filter is designed for the harmonic; the input signal is filtered through the second FIR comb filter to determine the signal of the harmonic parameters, and send the signal parameters to the processing unit of the harmonic analyzer. The power harmonic analysis method is easy to implement, can conveniently calculate harmonic parameters, has high precision, and is not affected by frequency drift.

Description

A kind of electric harmonic analysis method and harmonic analyzer based on harmonic analyzer

technical field

The present invention relates to the technical field of signal processing, in particular to a harmonic analyzer-based power harmonic analysis method and a harmonic analyzer.

Background technique

With the rapid development of power electronic technology, the harmonic pollution of the power system is becoming more and more serious. The harmonic problems brought by power electronic devices pose a potential threat to the safety, stability and economic operation of the power system, and have a great impact on the surrounding electrical environment. , the problem of power quality has been highly valued. Harmonic analysis technology is widely used in many fields such as power quality monitoring, electronic product production inspection, and electrical equipment monitoring. It is an important technical means for power grid monitoring, quality inspection, and equipment monitoring. At present, the commonly used analysis methods mainly include discrete Fourier transform (DFT), fast Fourier transform (FFT), digital filtering algorithm based on FIR (Finite Impulse Response, finite impulse response), wavelet transform method and Prony algorithm, etc.

In engineering applications, harmonic analysis always performs sampling at limited points, and it is difficult to achieve strict synchronous sampling. In this way, when FFT and DFT are used for harmonic analysis, there will be long-range leakage due to the truncation effect and short-range leakage due to the fence effect, making the analysis results inaccurate or even unreliable.

The harmonic analysis of power system based on FIR digital filter is a simple and effective algorithm among these methods. FIR digital filters can filter out the desired frequency components from a signal, but this method has some side effects, it changes the amplitude and phase of the remaining frequency components in the signal.

Contents of the invention

The purpose of the present invention is to overcome the defect of fence effect in the harmonic analysis in the prior art, and according to one aspect of the present invention, a power harmonic analysis method based on a harmonic analyzer is proposed.

A method for analyzing power harmonics based on a harmonic analyzer according to an embodiment of the present invention, comprising:

Determine the preset base frequency, design the first FIR comb filter for the input signal according to the preset base frequency; filter the input signal through the first FIR comb filter, and determine the frequency of the filtered signal; when the filtered signal When the difference between the frequency and the preset fundamental frequency is less than the preset threshold, the frequency of the filtered signal is used as the fundamental frequency of the input signal; the frequency of the harmonic is determined according to the fundamental frequency, and a second FIR comb filter is designed for the harmonic The input signal is filtered by the second FIR comb filter, the signal parameters of the harmonics are determined, and the signal parameters are sent to the processing unit of the harmonic analyzer. The signal parameters include: signal amplitude and signal phase angle.

Preferably, the power harmonic analysis method based on the harmonic analyzer further includes: when the difference between the frequency of the filtered signal and the preset fundamental frequency is not less than the preset threshold, continue to pass through the first FIR comb filter pair The filtered signal is filtered until the difference between the frequency of the re-filtered signal and the preset fundamental frequency is less than a preset threshold, and the frequency of the re-filtered signal is used as the fundamental frequency of the input signal.

Preferably, determining the frequency of the filtered signal includes:

Obtain three consecutive sampling values x(k), x(k+1) and x(k+2) of the filtered signal;

Determine the frequency of the filtered signal as: in Δt is the sampling time interval.

Preferably, determining the signal parameters of the harmonics includes:

Obtain two consecutive sampling values x _i (k) and x _i (k+1) of the harmonic;

Determine the signal amplitude of the harmonic as: X _i =2|A _i |;

Determine the signal phase angle of the harmonic as: φ _i = angle(A _i );

in, f _i is the frequency of the harmonic, and Δt is the sampling time interval.

Preferably, the first FIR comb filter and the second FIR comb filter are FIR window filters;

For the sampling sequence x(k)=Xcos(2πfkΔt+φ), after filtering by FIR window filter:

$\tilde{x}\left(k\right)=\frac{x}{2}{e}^{\mathrm{j\φ}}{a}^k\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}+\frac{x}{2}{e}^{-\mathrm{j\φ}}{a}^{-k}\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right)a^{\mathrm{no}},$ where a=e ^j2πfΔt , $\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}$ is the window correction factor.

Preferably, when the second FIR comb filter is an FIR window filter, determining the signal parameters of the harmonics includes:

Get two consecutive samples of the harmonic and

Determine the signal amplitude of the harmonic as: X _i =2|A _i |;

Determine the signal phase angle of the harmonic as: φ _i = angle(A _i );

in, $A_i=\frac{{\tilde{x}}_i(k+1)a_i-{\tilde{x}}_i\left(k\right)}{{a_i}^k({a_i}^2-1)\×{\mathrm{WCF}}_i},$ $a_i=e^{j2\mathrm{\π}{f}_i\mathrm{\Δt}},$ ${\mathrm{WCF}}_i=\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right){a_i}^{\mathrm{no}},$ f _i is the frequency of the harmonic, and Δt is the sampling time interval.

A power harmonic analysis method based on a harmonic analyzer provided by an embodiment of the present invention is easy to implement and can conveniently calculate harmonic parameters with high precision, and the method is not affected by frequency drift, and the sampling frequency does not need to be It is 2n times of the fundamental frequency, which effectively overcomes the shortcomings of leakage, fence effect and aliasing effect that seriously affect the accuracy of harmonic analysis.

The purpose of the present invention is to overcome the defect of fence effect in the harmonic analysis in the prior art, and according to one aspect of the present invention, a harmonic analyzer is proposed.

A harmonic analyzer according to an embodiment of the present invention includes:

The first filtering design module is used to determine the preset fundamental frequency, and design a first FIR comb filter for the input signal according to the preset fundamental frequency;

The first filtering module is configured to filter the input signal through a first FIR comb filter, and determine the frequency of the filtered signal;

A base frequency determination module, configured to use the frequency of the filtered signal as the base frequency of the input signal when the difference between the frequency of the filtered signal and the preset base frequency is less than a preset threshold;

The second filter design module is used to determine the frequency of the harmonic according to the fundamental frequency, and to design a second FIR comb filter for the harmonic;

The second filter module is used to filter the input signal through the second FIR comb filter, determine the signal parameters of the harmonics, and send the signal parameters to the processing unit of the harmonic analyzer. The signal parameters include: signal amplitude and signal phase angle.

Preferably, the first filtering module is also used for: when the difference between the frequency of the filtered signal and the preset fundamental frequency is not less than the preset threshold, continue to filter the filtered signal through the first FIR comb filter, until the difference between the frequency of the filtered signal and the preset fundamental frequency is smaller than the preset threshold;

The base frequency determining module is also used for: taking the frequency of the filtered signal as the base frequency of the input signal.

Preferably, the first filter module includes:

The first acquisition unit is used to obtain three consecutive sampling values x(k), x(k+1) and x(k+2) of the filtered signal;

The first calculation unit is used to determine the frequency of the filtered signal as: in Δt is the sampling time interval.

Preferably, the second filter module includes:

The second acquisition unit is used to obtain two consecutive sampling values x _i (k) and x _i (k+1) of the harmonic;

The second calculation unit is used to determine the signal amplitude of the harmonic: X _i =2|A _i |; determine the phase angle of the signal of the harmonic: φ _i =angle(A _i );

in, f _i is the frequency of the harmonic, and Δt is the sampling time interval.

Preferably, the first FIR comb filter determined by the first filter design module and the second FIR comb filter determined by the second filter design module are FIR window filters;

For the sampling sequence x(k)=Xcos(2πfkΔt+φ), after filtering by FIR window filter:

$\tilde{x}\left(k\right)=\frac{x}{2}{e}^{\mathrm{j\φ}}{a}^k\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}+\frac{x}{2}{e}^{-\mathrm{j\φ}}{a}^{-k}\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right)a^{\mathrm{no}},$ where a=e ^j2πfΔt , $\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}$ is the window correction factor.

Preferably, when the second FIR comb filter is an FIR window filter, the second filter module includes:

The third acquisition unit is used to obtain two consecutive sampling values of harmonics and x

The third calculation unit is used to determine the signal amplitude of the harmonic: X _i =2|A _i |; determine the phase angle of the harmonic signal: φ _i =angle(A _i );

in, $A_i=\frac{{\tilde{x}}_i(k+1)a_i-{\tilde{x}}_i\left(k\right)}{{a_i}^k({a_i}^2-1)\×{\mathrm{WCF}}_i},$ $a_i=e^{j2\mathrm{\π}{f}_i\mathrm{\Δt}},$ ${\mathrm{WCF}}_i=\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\Σ}}}W\left(\mathrm{no}\right){a_i}^{\mathrm{no}},$ f _i is the frequency of the harmonic, and Δt is the sampling time interval.

Additional features and advantages of the invention will be set forth in the description which follows, and in part will be apparent from the description, or may be learned by practice of the invention. The objectives and other advantages of the invention may be realized and attained by the structure particularly pointed out in the written description and claims hereof as well as the appended drawings.

The technical solutions of the present invention will be described in further detail below with reference to the accompanying drawings and embodiments.

Description of drawings

The accompanying drawings are used to provide a further understanding of the present invention, and constitute a part of the description, and are used together with the embodiments of the present invention to explain the present invention, and do not constitute a limitation to the present invention. In the attached picture:

Fig. 1 is the flowchart of harmonic analysis method in the embodiment of the present invention;

Fig. 2 is the flowchart of harmonic analysis method in embodiment one;

Fig. 3 is a structural diagram of a harmonic analyzer in an embodiment of the present invention.

Detailed ways

The specific embodiments of the present invention will be described in detail below in conjunction with the accompanying drawings, but it should be understood that the protection scope of the present invention is not limited by the specific embodiments.

According to an embodiment of the present invention, a power harmonic analysis method based on a harmonic analyzer is provided, as shown in FIG. 1 , in the embodiment of the present invention, the flow of the harmonic analysis method is as follows:

Step 101: Determine a preset fundamental frequency, and design a first FIR comb filter for an input signal according to the preset fundamental frequency.

The preset fundamental frequency is the theoretical fundamental frequency of the input signal. For example, the theoretical fundamental frequency of the power signal is 50 Hz, and the preset fundamental frequency is set to 50 Hz. The preset fundamental frequency is mainly used for designing the first FIR comb filter and comparing with the frequency of the signal filtered by the first FIR comb filter.

Step 102: Filter the input signal through the first FIR comb filter, and determine the frequency of the filtered signal.

The first FIR comb filter can filter out all harmonics, leaving only the fundamental wave of the input signal, so the filtered signal is mainly composed of the fundamental wave.

Step 103: When the difference between the frequency of the filtered signal and the preset base frequency is smaller than a preset threshold, use the frequency of the filtered signal as the base frequency of the input signal.

Since there is an error between the fundamental frequency of the actual signal and the theoretical preset fundamental frequency, only when the difference between the frequency of the filtered signal and the preset fundamental frequency is smaller than the preset threshold, the frequency of the filtered signal is frequency as the fundamental frequency of the input signal. Wherein, the preset threshold is preset, and may be specifically 0.1 Hz or 0.01 Hz, etc., depending on actual conditions.

When the difference between the frequency of the filtered signal and the preset base frequency is not less than the preset threshold, continue to filter the filtered signal through the first FIR comb filter until the frequency of the filtered signal is the same as the preset base frequency. The difference between the frequencies is less than the preset threshold, and the frequency of the filtered signal is used as the fundamental frequency of the input signal. That is, when the difference between the frequency of the filtered signal and the preset base frequency is not less than the preset threshold, the filtered signal is filtered again, and then the frequency of the filtered signal and the preset base frequency are judged. Whether the difference between the two frequencies is less than the preset threshold, if the difference between the above two frequencies is still not less than the preset threshold, continue to filter the filtered signal until the difference between the two frequencies is less than the preset threshold threshold.

Step 104: Determine the frequency of the i-th harmonic according to the fundamental frequency, and design a second FIR comb filter for the i-th harmonic.

When the fundamental frequency of the input signal is determined, the frequency of the i-th harmonic in the input signal can be determined, which is i times the fundamental frequency, that is, the frequency of the i-th harmonic is f _i =if ₁ , where f ₁ is the base frequency. The second FIR comb filter is used to filter out other harmonics except the i-th harmonic in the input signal, that is, the second FIR comb filter outputs the i-th harmonic.

Step 105: filter the input signal through the second FIR comb filter, determine the signal parameters of the i-th harmonic, and send the signal parameters to the processing unit of the harmonic analyzer, the signal parameters include: signal amplitude and signal phase angle.

In step 104, the frequency of the i-th harmonic can be determined, and in step 105, the signal amplitude and signal phase angle of the i-th harmonic can be determined. The combination of the two steps can determine the frequency, amplitude and phase of the i-th harmonic , that is, all signal parameters of the i-th harmonic are determined. After the harmonic analyzer receives the signal parameters of the i-th harmonic, the harmonic analyzer can analyze and process the harmonic, and output it in the form of charts and curves for analysis and use by relevant technical personnel; harmonic analysis The instrument can also upload the harmonic signal parameters to the upper computer through the RS232, RS485 standard communication port or the network port (RJ45) communication interface. The upper computer is based on WINDOWS and other operating platforms, and can convert the collected harmonic signal parameters into various Reports, curves, bar graphs, at the same time, reports can be converted to WORD or EXCEL format as required.

In the embodiment of the present invention, the principle of the method for determining signal parameters in step 102 and step 105 is specifically as follows:

The ideal signal x(t) can be described as: x(t)=Xcos(2πft+φ); where X is the signal amplitude, f is the signal frequency, and φ is the signal phase angle. The continuous-time signal x(t) is discretely sampled at a fixed time interval Δt to generate a sampling sequence {x(k)}:

x(k)＝Xcos(2πfkΔt+φ) (2)

because Then formula (2) can be expressed as:

Let a=e ^j2πfΔt , z=Re(a), Among them, z is the real part of the complex number a, A ^* is the conjugate complex number of A, then the sampling sequence {x(k)} is:

x(k)＝Aa ^k +A ^* a ^-k (4)

According to Euler's formula e ^ix =cosx+isinx and formula (4), assuming m=2πfΔt, we can get:

x(k)+x(k+2)＝A(a ^k +a ^k+2 )+A ^* (a ^-k +a ^-k-2 )

＝A[cos(mk)+jsin(mk)+cos(mk+2m)+jsin(mk+2m)]+

A ^* [cos(mk)-jsin(mk)+cos(mk+2m)-jsin(mk+2m)]

＝A[2cos(mk+m)cosm+j2sin(mk+m)cosm]+

A ^* [2cos(mk+m)cosm-j2sin(mk+m)cosm]

=2[Ae ^j(mk+m) +A ^* e ^-j(mk+m) ]cosm=2x(k+1)cosm

At the same time, since z=Re(a)=Re(cos2πfΔt+jsin2πfΔt)=cos2πfΔt=cosm, combined with the above formula, we can get:

$\mathrm{<span\; class="notranslate">\; z</span>}<span\; class="notranslate">\; =</span>\frac{\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; 2</span>}<span\; class="notranslate">\; )</span>}{\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; )</span>}<span\; class="notranslate">\; -</span><span\; class="notranslate">\; -</span><span\; class="notranslate">\; -</span><span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; 5</span>}<span\; class="notranslate">\; )</span>$

And x(k+1)ax(k)=A[a ^k+1 aa ^k ]+A ^* (a ^-k-1 aa ^-k )=Aa ^k (a ² -1), therefore:

$\mathrm{<span\; class="notranslate">\; A</span>}<span\; class="notranslate">\; =</span>\frac{\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; )</span>\mathrm{<span\; class="notranslate">\; a</span>}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; )</span>}{{\mathrm{<span\; class="notranslate">\; a</span>}}^{\mathrm{<span\; class="notranslate">\; k</span>}}<span\; class="notranslate">\; (</span>{\mathrm{<span\; class="notranslate">\; a</span>}}^{\mathrm{<span\; class="notranslate">\; 2</span>}}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; )</span>}<span\; class="notranslate">\; -</span><span\; class="notranslate">\; -</span><span\; class="notranslate">\; -</span><span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; 6</span>}<span\; class="notranslate">\; )</span>$

Since z=cos2πfΔt, $A=\frac{x}{2}{e}^{\mathrm{j\&phi;}}=\frac{x}{2}(\cos \mathrm{\&phi;}+j\sin \mathrm{\&phi;}),$ Therefore, for a power signal x(t), the signal frequency Signal assignment X=2|A|, signal phase angle φ=angle(A).

In summary, for the signal x(t), z can be determined according to three discrete sampling values x(k), x(k+1), x(k+2), and then the signal frequency f of the signal x(t) can be determined ; When the signal frequency f and the sampling interval Δt are known, a and A can be determined, so that the signal amplitude X and the signal phase angle φ can be determined. The algorithm works for standard sine or cosine signals and therefore also for power harmonics. Using this algorithm to calculate the parameters of signal harmonics is easy to implement, has high precision, and is not affected by frequency drift. The sampling frequency does not have to be ²ⁿ times the fundamental frequency.

Applying the above algorithm to the embodiment of the present invention, in step 101, if the filtered signal is x(k), then determining the frequency of the filtered signal specifically includes:

Obtain three consecutive sampling values x(k), x(k+1) and x(k+2) of the filtered signal;

The frequency of the filtered signal is: in Δt is the sampling time interval.

In step 105, determining the signal parameters of the i-th harmonic specifically includes:

Obtain two consecutive sampling values x _i (k) and x _i (k+1) of the i-th harmonic;

The harmonic signal amplitude is: X _i =2|A _i |; the harmonic signal phase angle is: φ _i =angle(A _i );

in, $A_i=\frac{x_i(k+1)a_i-x_i\left(k\right)}{{a_i}^k({a_i}^2-1)},$ $a_i=e^{j2\mathrm{\&pi;}{f}_i\mathrm{\&Delta;t}},$ f _i is the frequency of the i-th harmonic, and Δt is the sampling time interval.

Preferably, both the first FIR comb filter and the second FIR comb filter are FIR window filters;

For the sampling sequence x(k)=Xcos(2πfkΔt+φ), after filtering by FIR window filter:

$\tilde{x}\left(k\right)=\frac{x}{2}{e}^{\mathrm{j\&phi;}}{a}^k\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}+\frac{x}{2}{e}^{-\mathrm{j\&phi;}}{a}^{-k}\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right)a^{\mathrm{no}},$ where a=e ^j2πfΔt , $\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}$ is the window correction factor.

According to formula (5) and formula (6), the signal of introducing window correction factor WCF The signal parameters of can be expressed as:

$\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; f</span>}}<span\; class="notranslate">\; =</span>\frac{{\mathrm{<span\; class="notranslate">\; cos</span>}}^{<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; (</span>\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; z</span>}}<span\; class="notranslate">\; )</span>}{\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; \&pi;\&Delta;t</span>}}<span\; class="notranslate">\; ,</span>\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; x</span>}}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 2</span>}<span\; class="notranslate">\; |</span>\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; A</span>}}<span\; class="notranslate">\; |</span><span\; class="notranslate">\; ,</span>\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; \&phi;</span>}}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; amgle</span>}<span\; class="notranslate">\; (</span>\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; A</span>}}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; ;</span>$

in: $\tilde{z}=\frac{\tilde{x}\left(k\right)+\tilde{x}(k+2)}{2\tilde{x}(k+1)},\tilde{A}=\frac{\tilde{x}(k+1)a-\tilde{x}\left(k\right)}{a^k(a^2-1)\&times;\mathrm{WCF}},\mathrm{WCF}=\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}.$

The ordinary FIR comb filter will change the amplitude and phase of the remaining frequency components in the signal while extracting the required frequency components, so the FIR window filter is used instead of the FIR comb filter. The window correction factor WCF method provided in the embodiment of the present invention is suitable for any finite sequence of digital filters, such as for Hamming, Hanning and Blackman window filters, and even DFT of finite complex sequences is also included.

A power harmonic analysis method based on a harmonic analyzer provided by the embodiment of the present invention is easy to implement and can conveniently calculate harmonic parameters with high precision, and the method is not affected by frequency drift, and the sampling frequency does not need to be It is 2n times of the fundamental frequency, which effectively overcomes the shortcomings of leakage, fence effect and aliasing effect that seriously affect the accuracy of harmonic analysis.

The flow of the power harmonic analysis method will be described in detail below through Embodiment 1.

Embodiment one

In the first embodiment, an FIR comb filter is used to filter the power signal. The FIR comb filter is processed by windowing, and its harmonic analysis process is shown in FIG. 2 . In Embodiment 1, the power signal waveform is described as: The signal x(t) is sampled at a fixed time interval Δt to generate a sampling sequence:

$\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; =</span>\underset{\mathrm{<span\; class="notranslate">\; i</span>}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 1</span>}}{\overset{\mathrm{<span\; class="notranslate">\; m</span>}}{\mathrm{<span\; class="notranslate">\; \&Sigma;</span>}}}{\mathrm{<span\; class="notranslate">\; x</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}\mathrm{<span\; class="notranslate">\; cos</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; \&pi;</span>}{\mathrm{<span\; class="notranslate">\; f</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}\mathrm{<span\; class="notranslate">\; k\&Delta;t</span>}<span\; class="notranslate">\; +</span>{\mathrm{<span\; class="notranslate">\; \&phi;</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; )</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 0,1,2,3</span>}<span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span>$

definition $a_i=e^{j2\mathrm{\&pi;}{f}_i\mathrm{\&Delta;t}},$ $A_i=\frac{x}{2}{e}^{j_i{\mathrm{\&phi;}}_i},$ z _i = Re(a _i ) = cos2πf _i Δt, ${A_i}^{*}=\frac{x}{2}{e}^{-j_i{\mathrm{\&phi;}}_i}.$

According to the unique properties of the FIR comb filter, the following conclusions can be drawn:

$\underset{\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 0</span>}}{\overset{\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; m</span>}}{\mathrm{<span\; class="notranslate">\; \&Sigma;</span>}}}\mathrm{<span\; class="notranslate">\; C</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span>\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 0</span>}<span\; class="notranslate">\; ;</span>$

C(n)={{1,-2z ₁ ,1}*{1,-2z ₂ ,1}*...*{1,-2z _m ,1}}, where z _i =cos2πf _i Δt, ^* is the convolution operator, and n is the number of FIR comb filter feedback loops. The above formula is a unique property of the FIR comb filter, which is a prior art, and will not be described in detail here.

When the FIR comb filter is used to filter out other harmonic signals except the i harmonic, the corresponding FIR comb filter expression is:

${\mathrm{<span\; class="notranslate">\; C</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; =</span>\frac{\mathrm{<span\; class="notranslate">\; C</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span>}{<span\; class="notranslate">\; \{</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; ,</span><span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}{\mathrm{<span\; class="notranslate">\; z</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; ,</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; \}</span>}<span\; class="notranslate">\; =</span><span\; class="notranslate">\; \{</span><span\; class="notranslate">\; \{</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; ,</span><span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}{\mathrm{<span\; class="notranslate">\; z</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; ,</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; \}</span><span\; class="notranslate">\; *</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; *</span><span\; class="notranslate">\; \{</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; ,</span><span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}{\mathrm{<span\; class="notranslate">\; z</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; ,</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; \}</span><span\; class="notranslate">\; *</span><span\; class="notranslate">\; \{</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; ,</span><span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}{\mathrm{<span\; class="notranslate">\; z</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; ,</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; \}</span><span\; class="notranslate">\; *</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; .</span><span\; class="notranslate">\; *</span><span\; class="notranslate">\; \{</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; ,</span><span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}{\mathrm{<span\; class="notranslate">\; z</span>}}_{\mathrm{<span\; class="notranslate">\; m</span>}}<span\; class="notranslate">\; ,</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; \}</span><span\; class="notranslate">\; \}</span><span\; class="notranslate">\; .</span>$

In Embodiment 1, the process of harmonic analysis is specifically as follows:

The preset fundamental frequency is f=60Hz, and the filter C ₁ (n) is designed.

Wherein, C ₁ (n)={{1,-2z ₂ ,1}*{1,-2z ₃ ,1}*...*{1,-2z _m ,1}}.

After the discrete power signal x(k) is filtered by the filter C1(n), the obtained signal waveform expression is:

${\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; x</span>}}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; =</span>\underset{\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 0</span>}}{\overset{\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; m</span>}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}}{\mathrm{<span\; class="notranslate">\; \&Sigma;</span>}}}{\mathrm{<span\; class="notranslate">\; C</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span>\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; .</span>$

Obtain Three consecutive samples of and According to formula (5), it can be determined that the frequency of the signal filtered by the filter C ₁ (n) is:

$f_{\mathrm{new}}=\frac{{\cos }^{-1}\left(z_1\right)}{2\mathrm{\&pi;\&Delta;t}},$ in, $z_1=\frac{{\tilde{x}}_1\left(k\right)+{\tilde{x}}_1(k+2)}{2{\tilde{x}}_1(k+1)}.$

Judging whether |f _new -f|<0.001Hz is true. Where 0.001Hz is the preset threshold, and |f _new -f|<0.001Hz is to judge whether the difference between the frequency f _new of the filtered signal and the preset base frequency 60Hz is less than the preset threshold 0.001Hz.

When |f _new -f|<0.001Hz is established, take f _new as the fundamental frequency f ₁ of the power signal x(k); when it is not established, continue to pass through the filter C ₁ (n) to Filtering is performed until the difference between the frequency of the filtered signal and the preset fundamental frequency of 60 Hz is smaller than the preset threshold.

When the fundamental frequency _f1 is determined, according to formula (6), the amplitude and phase angle of the fundamental wave can be determined:

X ₁ =2|A ₁ |, φ ₁ =angle(A ₁ ); where:

${\mathrm{<span\; class="notranslate">\; A</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; =</span>\frac{{\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; x</span>}}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; )</span>{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; -</span>{\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; x</span>}}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; )</span>}{{{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}}^{\mathrm{<span\; class="notranslate">\; k</span>}}<span\; class="notranslate">\; (</span>{{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}}^{\mathrm{<span\; class="notranslate">\; 2</span>}}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; \&times;</span>{\mathrm{<span\; class="notranslate">\; WCF</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}}<span\; class="notranslate">\; ,</span>{\mathrm{<span\; class="notranslate">\; WCF</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; =</span>\underset{\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 0</span>}}{\overset{\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; m</span>}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}}{\mathrm{<span\; class="notranslate">\; \&Sigma;</span>}}}{\mathrm{<span\; class="notranslate">\; C</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span>{{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}}^{\mathrm{<span\; class="notranslate">\; no</span>}}<span\; class="notranslate">\; .</span>$

According to the fundamental frequency f ₁ , the frequency of the i-th harmonic can be determined, and then the filter C _i (n) can be designed according to the required i-th harmonic, C _i (n)={{1,-2z ₁ ,1} *...*{1,-2z _i-1 ,1}*{1,-2z _i+1 ,1}*...*{1,-2z _m ,1}}.

After the discrete power signal x(k) is filtered by the filter C _i (n), the expression of the i-th harmonic obtained is:

${\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; x</span>}}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; =</span>\underset{\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 0</span>}}{\overset{\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; m</span>}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}}{\mathrm{<span\; class="notranslate">\; \&Sigma;</span>}}}{\mathrm{<span\; class="notranslate">\; C</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span>\mathrm{<span\; class="notranslate">\; x</span>}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span>$

The amplitude and phase angle of the i-th harmonic are:

X _i =2|A _i |, φ _i =angle(A _i ); where:

${\mathrm{<span\; class="notranslate">\; A</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; =</span>\frac{{\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; x</span>}}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; +</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; )</span>{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; -</span>{\stackrel{<span\; class="notranslate">\; ~</span>}{\mathrm{<span\; class="notranslate">\; x</span>}}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; k</span>}<span\; class="notranslate">\; )</span>}{{{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}}^{\mathrm{<span\; class="notranslate">\; k</span>}}<span\; class="notranslate">\; (</span>{{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}}^{\mathrm{<span\; class="notranslate">\; 2</span>}}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 1</span>}<span\; class="notranslate">\; )</span><span\; class="notranslate">\; \&times;</span>{\mathrm{<span\; class="notranslate">\; WCF</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}}<span\; class="notranslate">\; ,</span>{\mathrm{<span\; class="notranslate">\; WCF</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; =</span>\underset{\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; =</span>\mathrm{<span\; class="notranslate">\; 0</span>}}{\overset{\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; m</span>}<span\; class="notranslate">\; -</span>\mathrm{<span\; class="notranslate">\; 2</span>}}{\mathrm{<span\; class="notranslate">\; \&Sigma;</span>}}}{\mathrm{<span\; class="notranslate">\; C</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; (</span>\mathrm{<span\; class="notranslate">\; no</span>}<span\; class="notranslate">\; )</span>{{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}}^{\mathrm{<span\; class="notranslate">\; no</span>}}{<span\; class="notranslate">\; ,</span>\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; =</span>{\mathrm{<span\; class="notranslate">\; e</span>}}^{\mathrm{<span\; class="notranslate">\; j</span>}\mathrm{<span\; class="notranslate">\; 2</span>}\mathrm{<span\; class="notranslate">\; \&pi;</span>}{\mathrm{<span\; class="notranslate">\; f</span>}}_{\mathrm{<span\; class="notranslate">\; i</span>}}\mathrm{<span\; class="notranslate">\; \&Delta;t</span>}}<span\; class="notranslate">\; =</span>{{\mathrm{<span\; class="notranslate">\; a</span>}}_{\mathrm{<span\; class="notranslate">\; 1</span>}}}^{\mathrm{<span\; class="notranslate">\; i</span>}}<span\; class="notranslate">\; .</span>$

Therefore, to obtain the i-th harmonic Two consecutive sampling values of and The amplitude and phase angle of the i-th harmonic can be determined.

A power harmonic analysis method based on a harmonic analyzer provided by an embodiment of the present invention is easy to implement and can conveniently calculate harmonic parameters with high precision, and the method is not affected by frequency drift, and the sampling frequency does not need to be It is 2n times of the fundamental frequency, which effectively overcomes the shortcomings of leakage, fence effect and aliasing effect that seriously affect the accuracy of harmonic analysis. Using the window correction factor method, it can be applied to any finite sequence of digital filters, and different window algorithms and window sizes can be selected to obtain better performance. A power harmonic analysis method based on a harmonic analyzer provided by an embodiment of the present invention is suitable for off-line applications in need, and if the harmonic analysis method can be realized by parallel computing, it is also suitable for online applications.

The implementation process of a power harmonic analysis method based on a harmonic analyzer is introduced in detail above. Embodiments of the present invention also provide a harmonic analyzer. The structure of the harmonic analyzer is introduced below.

A harmonic analyzer provided by an embodiment of the present invention, as shown in FIG. 3 , includes: a first filter design module 301, a first filter module 302, a fundamental frequency determination module 303, a second filter design module 304 and a second filter Module 305.

The first filtering design module 301 is used to determine the preset fundamental frequency, and design a first FIR comb filter for the input signal according to the preset fundamental frequency;

The first filtering module 302 is configured to filter the input signal through a first FIR comb filter, and determine the frequency of the filtered signal;

The base frequency determination module 303 is used to use the frequency of the filtered signal as the base frequency of the input signal when the difference between the frequency of the filtered signal and the preset base frequency is less than a preset threshold;

The second filtering design module 304 is used to determine the frequency of the harmonic according to the fundamental frequency, and designs a second FIR comb filter for the harmonic;

The second filter module 305 is used to filter the input signal through the second FIR comb filter, determine the signal parameters of the harmonics, and send the signal parameters to the processing unit of the harmonic analyzer, the signal parameters include: signal amplitude and signal phase angle.

Preferably, the first filtering module 302 is also used for: when the difference between the frequency of the filtered signal and the preset fundamental frequency is not less than the preset threshold, continue to filter the filtered signal through the first FIR comb filter , until the difference between the frequency of the filtered signal and the preset fundamental frequency is smaller than the preset threshold;

The fundamental frequency determination module 303 is further configured to: use the frequency of the filtered signal again as the fundamental frequency of the input signal.

Preferably, the first filtering module 302 includes:

The first acquisition unit is used to obtain three consecutive sampling values x(k), x(k+1) and x(k+2) of the filtered signal;

The first calculation unit is used to determine the frequency of the filtered signal as: in Δt is the sampling time interval.

Preferably, the second filtering module 305 includes:

The second acquisition unit is used to obtain two consecutive sampling values x _i (k) and x _i (k+1) of the harmonic;

The second calculation unit is used to determine the signal amplitude of the harmonic: X _i =2|A _i |; determine the phase angle of the signal of the harmonic: φ _i =angle(A _i );

in, f _i is the frequency of the harmonic, and Δt is the sampling time interval.

Preferably, the first FIR comb filter determined by the first filter design module 301 and the second FIR comb filter determined by the second filter design module 304 are FIR window filters;

For the sampling sequence x(k)=Xcos(2πfkΔt+φ), after filtering by FIR window filter:

$\tilde{x}\left(k\right)=\frac{x}{2}{e}^{\mathrm{j\&phi;}}{a}^k\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}+\frac{x}{2}{e}^{-\mathrm{j\&phi;}}{a}^{-k}\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right)a^{\mathrm{no}},$ where a=e ^j2πfΔt , $\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right)a^{\mathrm{no}}$ is the window correction factor.

Preferably, when the second FIR comb filter is an FIR window filter, the second filtering module 305 includes:

The third acquisition unit is used to obtain two consecutive sampling values of harmonics and

The third calculation unit is used to determine the signal amplitude of the harmonic: X _i =2|A _i |; determine the phase angle of the harmonic signal: φ _i =angle(A _i );

in, $A_i=\frac{{\tilde{x}}_i(k+1)a_i-{\tilde{x}}_i\left(k\right)}{{a_i}^k({a_i}^2-1)\&times;{\mathrm{WCF}}_i},$ $a_i=e^{j2\mathrm{\&pi;}{f}_i\mathrm{\&Delta;t}},$ ${\mathrm{WCF}}_i=\underset{\mathrm{no}=0}{\overset{m-1}{\mathrm{\&Sigma;}}}W\left(\mathrm{no}\right){a_i}^{\mathrm{no}},$ f _i is the frequency of the harmonic, and Δt is the sampling time interval.

A harmonic analyzer provided by an embodiment of the present invention can quickly calculate harmonic parameters according to harmonic sampling values, has high precision, and is not affected by frequency drift. At the same time, the sampling frequency does not have to be 2 ⁿ times the fundamental frequency, effectively It overcomes the shortcomings that seriously affect the accuracy of harmonic analysis such as leakage, fence effect and aliasing effect.

The present invention can have multiple specific implementations in different forms. Above, the technical solutions of the present invention are illustrated by taking Fig. 1-Fig. 3 as an example in conjunction with the accompanying drawings. In the specific process or embodiment structure, those of ordinary skill in the art should understand that the specific implementations provided above are only some examples of various preferred usages, and any implementation that embodies the claims of the present invention shall be included in the present invention. Within the scope of protection required by the technical solution.

Finally, it should be noted that the above are only preferred embodiments of the present invention, and are not intended to limit the present invention. Although the present invention has been described in detail with reference to the foregoing embodiments, those skilled in the art can still understand The technical solutions recorded in the foregoing embodiments are modified, or some of the technical features are equivalently replaced. Any modifications, equivalent replacements, improvements, etc. made within the spirit and principles of the present invention shall be included within the protection scope of the present invention.

Claims (12)

1. A power harmonic analysis method based on a harmonic analyzer is characterized by comprising the following steps:

determining a preset fundamental frequency, and designing a first FIR comb filter for an input signal according to the preset fundamental frequency;

filtering the input signal through the first FIR comb filter and determining the frequency of the filtered signal;

when the difference value between the frequency of the filtered signal and the preset fundamental frequency is smaller than a preset threshold value, taking the frequency of the filtered signal as the fundamental frequency of the input signal;

determining the frequency of a harmonic according to the fundamental frequency, and designing a second FIR comb filter for the harmonic;

filtering the input signal through the second FIR comb filter, determining signal parameters of the harmonic waves, and sending the signal parameters to a processing unit of a harmonic wave analyzer, wherein the signal parameters include: signal amplitude and signal phase angle.

2. The method of claim 1, further comprising:

and when the difference value between the frequency of the filtered signal and the preset fundamental frequency is not less than a preset threshold value, continuing to filter the filtered signal through the first FIR comb filter until the difference value between the frequency of the re-filtered signal and the preset fundamental frequency is less than the preset threshold value, and taking the frequency of the re-filtered signal as the fundamental frequency of the input signal.

3. The method of claim 1 or 2, wherein determining the frequency of the filtered signal comprises:

acquiring three sampling values x (k), x (k +1) and x (k +2) of the filtered signal;

determining the frequency of the filtered signal as:whereinΔ t is the sampling time interval.

4. The method of claim 1 or 2, wherein said determining signal parameters of said harmonics comprises:

obtaining two consecutive sample values x of the harmonic_i(k) And x_i(k+1)；

Determining the signal amplitude of the harmonic as: x_i=2|A_i|；

Determining a signal phase angle of the harmonic as: phi is a_i=angle(A_i)；

Wherein, f_iat is the frequency of the harmonic, Δ t is the sampling time interval.

5. The method of claim 1 or 2, wherein the first FIR comb filter and the second FIR comb filter are FIR window filters;

for the sampling sequence x (k) = Xcos (2 pi fk Δ t + phi), after filtering processing is performed by an FIR window filter:

<math> <mrow> <mover> <mi>x</mi> <mo>~</mo> </mover> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mi>X</mi> <mn>2</mn> </mfrac> <msup> <mi>e</mi> <mi>j&phi;</mi> </msup> <msup> <mi>a</mi> <mi>k</mi> </msup> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>a</mi> <mi>n</mi> </msup> <mo>+</mo> <mfrac> <mi>X</mi> <mn>2</mn> </mfrac> <msup> <mi>e</mi> <mrow> <mo>-</mo> <mi>j&phi;</mi> </mrow> </msup> <msup> <mi>a</mi> <mrow> <mo>-</mo> <mi>k</mi> </mrow> </msup> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>a</mi> <mi>n</mi> </msup> <mo>,</mo> </mrow> </math> wherein a = e^j2πfΔt， <math> <mrow> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>a</mi> <mi>n</mi> </msup> </mrow> </math> Is a window correction factor.

6. The method of claim 5, wherein said determining signal parameters of said harmonics comprises:

obtaining two consecutive sample values of the harmonicAnd

determining the signal amplitude of the harmonic as: x_i=2|A_i|；

Determining a signal phase angle of the harmonic as: phi is a_i=angle(A_i)；

Wherein, <math> <mrow> <msub> <mi>A</mi> <mi>i</mi> </msub> <mo>=</mo> <mfrac> <mrow> <msub> <mover> <mi>x</mi> <mo>~</mo> </mover> <mi>i</mi> </msub> <mrow> <mo>(</mo> <mi>k</mi> <mo>+</mo> <mn>1</mn> <mo>)</mo> </mrow> <msub> <mi>a</mi> <mi>i</mi> </msub> <mo>-</mo> <msub> <mover> <mi>x</mi> <mo>~</mo> </mover> <mi>i</mi> </msub> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> </mrow> <mrow> <msup> <msub> <mi>a</mi> <mi>i</mi> </msub> <mi>k</mi> </msup> <mrow> <mo>(</mo> <msup> <msub> <mi>a</mi> <mi>i</mi> </msub> <mn>2</mn> </msup> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>&times;</mo> <msub> <mi>WCF</mi> <mi>i</mi> </msub> </mrow> </mfrac> <mo>,</mo> </mrow> </math> <math> <mrow> <msub> <mi>a</mi> <mi>i</mi> </msub> <mo>=</mo> <msup> <mi>e</mi> <mrow> <mi>j</mi> <mn>2</mn> <mi>&pi;</mi> <msub> <mi>f</mi> <mi>i</mi> </msub> <mi>&Delta;t</mi> </mrow> </msup> <mo>,</mo> </mrow> </math> <math> <mrow> <msub> <mi>WCF</mi> <mi>i</mi> </msub> <mo>=</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <msub> <mi>a</mi> <mi>i</mi> </msub> <mi>n</mi> </msup> <mo>,</mo> </mrow> </math> f_iat is the frequency of the harmonic, Δ t is the sampling time interval.

7. A harmonic analyzer, comprising:

the first filtering design module is used for determining a preset fundamental frequency and designing a first FIR comb filter for an input signal according to the preset fundamental frequency;

a first filtering module, configured to filter the input signal through the first FIR comb filter, and determine a frequency of the filtered signal;

a fundamental frequency determining module, configured to, when a difference between the frequency of the filtered signal and the preset fundamental frequency is smaller than a preset threshold, take the frequency of the filtered signal as the fundamental frequency of the input signal;

the second filtering design module is used for determining the frequency of the harmonic wave according to the fundamental frequency and designing a second FIR comb filter for the harmonic wave;

a second filtering module, configured to filter the input signal through the second FIR comb filter, determine a signal parameter of the harmonic, and send the signal parameter to a processing unit of a harmonic analyzer, where the signal parameter includes: signal amplitude and signal phase angle.

8. The harmonic analyzer of claim 7, wherein the first filtering module is further configured to: when the difference value between the frequency of the filtered signal and the preset fundamental frequency is not less than a preset threshold value, continuing to filter the filtered signal through the first FIR comb filter until the difference value between the frequency of the re-filtered signal and the preset fundamental frequency is less than the preset threshold value;

the fundamental frequency determination module is further configured to: and taking the frequency of the re-filtered signal as the fundamental frequency of the input signal.

9. The harmonic analyzer of claim 7 or 8, wherein the first filtering module comprises:

the first acquisition unit is used for acquiring sampling values x (k), x (k +1) and x (k +2) of the filtered signal in three consecutive times;

a first calculating unit, configured to determine that a frequency of the filtered signal is:whereinΔ t is the sampling time interval.

10. The harmonic analyzer of claim 7 or 8, wherein the second filtering module comprises:

a second acquisition unit for acquiring two consecutive sampling values x of the harmonic_i(k) And x_i(k+1)；

A second calculation unit for determining the signal amplitude of the harmonic as: x_i=2|A_iL, |; determining a signal phase angle of the harmonic as: phi is a_i=angle(A_i)；

Wherein, f_iat is the frequency of the harmonic, Δ t is the sampling time interval.

11. The harmonic analyzer of claim 7 or 8, wherein the first FIR comb filter determined by the first filter design module and the second FIR comb filter determined by the second filter design module are FIR window filters;

for the sampling sequence x (k) = Xcos (2 pi fk Δ t + phi), after filtering processing is performed by an FIR window filter:

<math> <mrow> <mover> <mi>x</mi> <mo>~</mo> </mover> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mi>X</mi> <mn>2</mn> </mfrac> <msup> <mi>e</mi> <mi>j&phi;</mi> </msup> <msup> <mi>a</mi> <mi>k</mi> </msup> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>a</mi> <mi>n</mi> </msup> <mo>+</mo> <mfrac> <mi>X</mi> <mn>2</mn> </mfrac> <msup> <mi>e</mi> <mrow> <mo>-</mo> <mi>j&phi;</mi> </mrow> </msup> <msup> <mi>a</mi> <mrow> <mo>-</mo> <mi>k</mi> </mrow> </msup> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>a</mi> <mi>n</mi> </msup> <mo>,</mo> </mrow> </math> wherein a = e^j2πfΔt， <math> <mrow> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>a</mi> <mi>n</mi> </msup> </mrow> </math> Is a window correction factor.

12. The harmonic analyzer of claim 11, wherein the second filtering module comprises:

a third acquisition unit for acquiring two consecutive sampling values of the harmonicAnd x

A third calculation unit for determining the signal amplitude of the harmonic as: x_i=2|A_iL, |; determining a signal phase angle of the harmonic as: phi is a_i=angle(A_i)；

Wherein, <math> <mrow> <msub> <mi>A</mi> <mi>i</mi> </msub> <mo>=</mo> <mfrac> <mrow> <msub> <mover> <mi>x</mi> <mo>~</mo> </mover> <mi>i</mi> </msub> <mrow> <mo>(</mo> <mi>k</mi> <mo>+</mo> <mn>1</mn> <mo>)</mo> </mrow> <msub> <mi>a</mi> <mi>i</mi> </msub> <mo>-</mo> <msub> <mover> <mi>x</mi> <mo>~</mo> </mover> <mi>i</mi> </msub> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> </mrow> <mrow> <msup> <msub> <mi>a</mi> <mi>i</mi> </msub> <mi>k</mi> </msup> <mrow> <mo>(</mo> <msup> <msub> <mi>a</mi> <mi>i</mi> </msub> <mn>2</mn> </msup> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>&times;</mo> <msub> <mi>WCF</mi> <mi>i</mi> </msub> </mrow> </mfrac> <mo>,</mo> </mrow> </math> <math> <mrow> <msub> <mi>a</mi> <mi>i</mi> </msub> <mo>=</mo> <msup> <mi>e</mi> <mrow> <mi>j</mi> <mn>2</mn> <mi>&pi;</mi> <msub> <mi>f</mi> <mi>i</mi> </msub> <mi>&Delta;t</mi> </mrow> </msup> <mo>,</mo> </mrow> </math> <math> <mrow> <msub> <mi>WCF</mi> <mi>i</mi> </msub> <mo>=</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>W</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <msub> <mi>a</mi> <mi>i</mi> </msub> <mi>n</mi> </msup> <mo>,</mo> </mrow> </math> f_iat is the frequency of the harmonic, Δ t is the sampling time interval.