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科技英语论文
科技英语论文

信号与系统课程论文(报告、案例分析)

院系物理与电子工程学院

专业电子信息工程

班级 1211电工

学生姓名刘慧

学号 2012128040

任课教师罗倩倩

2014年 12 月 01 日

Liu Hui

Physics and Electronics Engineering College of Hubei University of

Arts and Science

1138804992@https://www.wendangku.net/doc/6f12003167.html,

Abstract:The characteristics of Butterworth filter is the pass band frequency response curves of the

maximally flat, no ups and downs, and in the resistance band is gradually decreased to zero. First order Butterworth filter attenuation rate of 6 dB per octave doubling every ten to 20 dB. The two order Butterworth filter attenuation rate of 12 dB per octave, and so on. The amplitude frequency monotone Butterworth filter down the diagonal, and whether the order number is unique, the amplitude frequency curves of the diagonal filter to keep the shape. But the higher the order of the filter, in the stop band

attenuation faster.

Key words:Butterworth passband frequency response sampling stopband

1 Requirements

using the impulse response method to design the Butterworth digital low pass filter, passband cutoff frequency 100HZ, the sampling frequency of 1000HZ, the maximum passband attenuation is 0.5HZ, minimum stopband attenuation i s 10HZ, the amplitude frequency and phase frequency, draw the corresponding curve. And the assumption of a signal x (T) =sin (2*pi*f1*t) +sin (2*pi*f2*t), wherein f1=50HZ, f2=200HZ. Use this signal to verify the correctness of filter design.

2 design principle

2.1 The basic concept of the digital filter

The so-called digital filter, refers to the input, output the digital signal, through numerical calculation processing relative proportion change of the input signal frequency components contained in, or filter in addition to certain frequency components of digital device or program, therefore, the concept of digital filter and analog filter is same, just form and realize the different filtering methods. Because of the digital filter to achieve filtering through numerical calculation, so the digital filter processing of high accuracy, stability, s mall volume and light weight, flexible, does not exist the problem of impedance matching, can special filtering function experiment simulation filter cannot achieve. If you want to deal with is the analog signal, through A\DC and D\AC conversion, matching in signal form, can also be filtered analog signal digital fi lter.

2.2 The working principle of the digital filter

Digital filter is a discrete time system, input x (n) is a time series, the output of Y (n) is a time series. Such as the system function of digital filter for H (Z), the pulse response of H (n), there is the following relationship in time domain;Y (n) =x (n)* H (n)

In the Z domain, the relationship between the input and output:Y (Z) =H (Z) *X (Z).In the formula, X (Z), Y (Z) x (n) respectively for the input and output y (n) Z transform.

Also in the frequency domain, the relationship between the input and output:Y (JW) =X (JW) *H (JW).In the formula, H (JW) for frequency response of digital filter, X (JW) and Y (JW) were x (n) and Y (n) spectrum. W is a digital angular frequency, unit rad. Usually the design of H (JW) in response to certain frequency value is 1, in response to certain frequency bands for 0.X (JW) and H (JW) of the product in the frequency response for 1 of those frequency value is X (JW), which can filter the unhindered by the amplitude of these bands, these bands for the pass band. X (JW) and H (JW) product response to those of 0 frequency value regardless of the X in the frequency band (JW) size is zero, which is not in the amplitude in these bands through the filter, the band called the stopband.

A suitable digital filter system function H (Z) may according to need to input x (n) frequency characteristic, the Y processed signal digital filter after the (n) retention signal x (n) useful frequency component in the removal of unwanted frequency components.

2.3 Butterworth filter design principle

2.3.1 the basic properties

Butterworth filter with the Butterworth function to approximate the system function filter. Butterworth filter is a filter according to the amplitude frequency characteristics has the most flat characteristics defined in the pass band..

The following summarizes the main characteristics of Butterworth filter

A .on all N,.()

1a j H 20=Ω=Ω B. on all N,()

707.0a j 2c =ΩΩH =Ω ()dB 3a lg 20j H c

=Ω=ΩΩ C .

()Ωj H a 2is monotone decreasing function. D. ()Ωj H a 2

with the increase of order N and more close to the ideal lowpass filter. As shown in Figure 2, we can see that the amplitude frequency characteristic of filter with order N filter increase and become better and better in the cut-off frequency Ωa function at C value is always 1/2, pass band more band area is close to 1; within the stopband more rapidly approaching zero.

Figure 2 Butterworth low pass filter amplitude frequency characteristics of the square

2.3.2 the system function

The system function design for Ha Butterworth (s), then:

2.3.3 the design process

Butterworth low pass type relation index filtering technology

Ap>-20log|Ha (J) |,

Ωa < ΩaP As<-20log|Ha (J) |, Ωa > Ωas Of which: ΩP

passband

boundary frequency, Ωa s stopband edge frequency. The type 1.4.1 can be:

After simplification finishing available:

For the smallest integer N type as the order of the filter. Then N into available:

Or

The look-up table to obtain normalized transfer function H (s), make s/ Ωa C instead of normalized prototype filter system function s, to obtain the actual filter transfer function.

2.4 he impulse response method

The so-called impulse response method is the digital filter impulse response sequence H (n) in response to the HA is equal to the analog filter (T) value of the sample, i.e.

H (n) =ha (T) |t=nT=ha (nT)

In the formula, T is the sampling period.

Therefore, digital filter system function H (Z) can be calculated by the formula

H (z) =Z[h (n)]=Z[ha (nT)]

Z[-] said the content of transform transform [-], please refer to the contents of the digital signal processing of the corresponding materials.

If you have already acquired the meet the transfer function of the Ha analog filter performance index (s), the H digital filter transfer function for the corresponding (z) method is the:

(1), for analog filter unit impulse response ha (t).

In the formula, L[Ha (s)] Ha (s) expressed the Laplace. inversion. Laplace transform, please refer to the contents of the integral transform or signal processing teaching of higher mathematics.

(2), for analog filter unit impulse response ha (T) sampling values, namely digital filter impulse response sequence H (n).

(3), rushed to the digital filter (n) response of the Z transform, transfer function H (Z).

By the above method to infer more directly by analog filter system function Ha (s) to calculate the digital filter system function H (z) steps:

(1) the use of the partial fraction expansion of the transfer function of the H analog filter (z) is expanded into

Ha (s) = Rk\ (S-Pk)

In MA TLAB this step can be implemented by residue function

If you call the residue function is of the form [b (R, a]=residue, P, K) form.

If [R, P, K]=residue (a, b) is the reverse process of the above call form.

(2) the Pk transform for the digital simulation of pole pole e^pkT to obtain the transfer function of the digital system

H (z) = Rk\ (1-e^pkT*z* (-1))

Type T is the sampling interval.

(3) the type conversion to transfer function form, can use [R, P, K]=residue (B, a).

For the above steps, has provided the impulse response function method to design digital filter, call format

[bz, az]=impinvar (B, a[, Fs], Fp)

In the formula, B, a analog filter the numerator and denominator polynomial coefficient vector; Fs is the sampling frequency (the filter data), unit Hz, the default for 1Hz, pre distortion frequency (prewarped frequency), is a "match" in frequency, the frequency, frequency response and the change before and after the simulation of frequency in the the variable can be matched precisely. Do not consider the general design. BZ, AZ respectively for the digital filter the numerator and denominator polynomial coefficient vector.

2.5 the MATLAB function that number.

(1) [N, wc]=buttord (WP, WS, RP, As, 's')

The format used in the order of N and 3dB to calculate Butterworth analog filter cutoff frequency wc. Wp, WS and WC is the actual simulation of the angular frequency (rad\s). Rp and As for the maximum passband attenuation and minimum attenuation.

(2) [Z, P, k]=buttap (N)

The format used in the calculation of N order Butterworth normalized analog low pass prototype filter system function zeros and poles and the gain factor returns the length of the N column vectors Z and P, were given N a zero and pole position, said K filter gain.

(3) Y=filter (B, a, x)

Type B said in the coefficient matrix of the system transfer function of the numerator polynomial coefficient matrix; a represents the denominator polynomial of the system transfer function; X represents the input sequence; filter represents the output sequence. A direct form of IIR functions.

(4) [b, a]=butter (N,Ωc, 'ftype')

The coefficient vector calculation of N order Butterworth digital filter system function the numerator and denominator polynomials B, a.

Note: the order number and the call to 3dB parameters N and WC are respectively the Butterworth digital filter cutoff frequency normalized value is generally called buttord format (1) calculation of N and wc. Coefficient b, a is in accordance with the Z-1 power arrangement.

(5) [B, A]=butter (N, Ωc 'ftype', 's')

Calculation of molecular, Butterworth analog filter system function of the denominator polynomial coefficient vector.

Description: C respectively as the Butterworth simulation filter call parameters N and Ωorder and 3dB cutoff frequency (the actual angular frequency), call the buttord (2) N and Ωc format calculation. Coefficients B, A are arranged according to descending s.

Tfype is a type of filter:

Ftype=high, high pass;Ωc only 1 values.

Ftype=stop, band stop; Ωc=[Cl, Omega cu], respectively, band stop filter passband 3dB cutoff frequency and the upper cutoff frequency.

Ftype by default: if Ωc only 1 values, the default is low pass; if Ωc2 values, the default is bandpass; the pass band frequency interval Ωc1< Ωa < Ωcu.

(6) [H, w]=freqz (B, a, N)

The coefficient vector B and a respectively for the discrete system system function the numerator and denominator polynomial, back H consists of discrete frequency response in the range of 0 ~ PI N frequency points values (where N is a positive integer), the w contains a frequency range of N points. Call the default N, 512 of its value. Can call freqz () calculation of frequency response function of the system, and then use the ABS () and angle () function and plot () function, draw out the frequency response curve of system.

(7) the lp2lp function [bt, at]=lp2lp (B, a, W0)

The function used to achieve the frequency transform by the low pass analog low-pass filter prototype filter, can be used to transfer function and the state space conversion, but no matter what kind of form, the input must be the analog filter prototype.

(8) [bz, az]=impinvar (B, a, FS)

With [b, the a] simulation of the filter transfer function model is converted into the sampling frequency

3 design ideas

Set signal

↓Analog low-pass filter prototype→frequency conversion→ analog discrete→IIR digital filter→the output signal

4 the design content

4.1MATLAB program design

Wp=2*pi*100; Ws=2*pi*150; % filter cutoff frequency

Rp=0.5; Rs=10; % maximum passband attenuation and the minimum stopband attenuation

Fs=1000 % of the sampling frequency;

Nn=128; % frequency points used by calling freqz

[N, Wc]=buttord (Wp, Ws, Rp, Rs,'s'); % minimum order analog filter number

[z, P, k]=buttap (N); % design analog lowpass Butterworth filter

[Bap, Aap]=zp2tf (Z, P, K); % converts the zero pole gain form as the form of transfer function [b, a]=lp2lp (Bap, Aap, Wc) % for frequency conversion

[bz, az]=impinvar (B, a, Fs); % by using the impulse response method to obtain the transfer function of the digital filter

Figure (1)

[H, W]=freqz (BZ, AZ, Nn, Fs); % amplitude characteristic drawing of digital filter and the phase frequency characteristic

Subplot (2,1,1)

Plot (W, 20*log10 (ABS (H)));

Xlabel ('frequency');

Ylabel ('range');

Grid on;

Subplot (2,1,2);

Plot (W, 180\pi*unwrap (angle (H)));

Xlabel ('frequency');

Ylabel ('range');

Grid on;

Figure (2)

F1=50; f2=200; % of the input signal frequency

N=100; % data length

Dt=1\Fs; n=0:N-1; t=n*dt; % sampling interval and time series

X=sin (2*pi*f1*t) +sin (2*pi*f2*t) % filter input signal

Subplot (2,1,1); % to draw the input signal

Plot (T, X);

Title ('input signals'); % on the input signal filtering using filter function y1=filter (BZ, AZ, X); Subplot (2,1,2); % plotted output waveform

Xlabel ('time')

Title ('output signal');

4.2 Experiment results and analysis

The two picture as obtained from the experiment shown in the first picture, less than 100 attenuation is less than 3Hz, and in the greater than 120Hz attenuation greater than 15dB, meet the design index filter. By the second pictures to see signal on 50Hz and 200Hz frequency components of the filter, filter the 200dB signal, achieve the effect of filtering.

5 The experience of experiment

Program the simulation process, I found the function and based on the low-pass filter to the design I. According to the design of the filter of the WP, WS, Rp, Rs four parameters to determine the frequency of the input signal in order to achieve the purpose of filtering. I also meet a lot of problems in the course of the experiment, but through the search data have been resolved. Through the curriculum design of learning, we have the simulation and design of MA TLAB, enhance mutual understanding and technical indexes of the low pass filter design principle of setting. The curriculum design not only let me on the digital signal processing theory of knowledge of the discipline to deepen understanding, but also let me use for many of the functions in the MA TLAB is more flexible, but also enhances the ability to solve the problem .

6 Ref.

[1] digital signal processing "(Third Edition) Gao Xiquan Ding Meiyu, Xi'an Electronic and Science University press

[2] "MA TLAB toolbox application" Su Jinming Zhang Lianhua Paul Liu, publishing house of electronics industry

[3] "digital signal processing MA TLAB implementation" edited by Wan Y ongji Science Press

科技英语论文

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