本文目錄一覽：
- 1、求：一個關於FIR帶通濾波器的C語言設計程序 代碼
- 2、一段matlab低通濾波器程序，求改編成C語言。
- 3、二階濾波器用C語言怎麼寫
- 4、c語言中值濾波問題？
- 5、如何用C語言實現低通濾波器
- 6、幫幫忙，能不能給我 基於C語言的FIR濾波器設計的程序代碼（包括CMD，C,ASM)，謝謝了 真的很急！！！
求：一個關於FIR帶通濾波器的C語言設計程序 代碼
short h[], short y[])
{
int i, j, sum; for (j = 0; j 100; j++) {
sum = 0;
for (i = 0; i 32; i++)
sum += x[i+j] * h[i];
y[j] = sum 15;
}
}
2
void fir(short x[], short h[], short y[])
{
int i, j, sum0, sum1;
short x0,x1,h0,h1; for (j = 0; j 100; j+=2) {
sum0 = 0;
sum1 = 0;
x0 = x[j];
for (i = 0; i 32; i+=2){
x1 = x[j+i+1];
h0 = h[i];
sum0 += x0 * h0;
sum1 += x1 * h0;
x0 = x[j+i+2];
h1 = h[i+1];
sum0 += x1 * h1;
sum1 += x0 * h1;
}
y[j] = sum0 15;
y[j+1] = sum1 15;
}
}
3
void fir(short x[], short h[], short y[])
{
int i, j, sum0, sum1;
short x0,x1,x2,x3,x4,x5,x6,x7,h0,h1,h2,h3,h4,h5,h6,h7; for (j = 0; j 100; j+=2) {
sum0 = 0;
sum1 = 0;
x0 = x[j];
for (i = 0; i 32; i+=8){
x1 = x[j+i+1];
h0 = h[i];
sum0 += x0 * h0;
sum1 += x1 * h0;
x2 = x[j+i+2];
h1 = h[i+1];
sum0 += x1 * h1;
sum1 += x2 * h1;
x3 = x[j+i+3];
h2 = h[i+2];
sum0 += x2 * h2;
sum1 += x3 * h2;
x4 = x[j+i+4];
h3 = h[i+3];
sum0 += x3 * h3;
sum1 += x4 * h3;
x5 = x[j+i+5];
h4 = h[i+4];
sum0 += x4 * h4;
sum1 += x5 * h4;
x6 = x[j+i+6];
h5 = h[i+5];
sum0 += x5 * h5;
sum1 += x6 * h5;
x7 = x[j+i+7];
h6 = h[i+6];
sum0 += x6 * h6;
sum1 += x7 * h6;
x0 = x[j+i+8];
h7 = h[i+7];
sum0 += x7 * h7;
sum1 += x0 * h7;
}
y[j] = sum0 15;
y[j+1] = sum1 15;
}
}
一段matlab低通濾波器程序，求改編成C語言。
這個我剛好做過一個濾波器，事實上對時域信號做FFT，截取一定點數再做逆FFT相當於理想濾波。設計濾波器代碼如下：
f1=100;f2=200;%待濾波正弦信號頻率
fs=2000;%採樣頻率
m=(0.3*f1)/(fs/2);%定義過度帶寬
M=round(8/m);%定義窗函數的長度
N=M-1;%定義濾波器的階數
b=fir1(N,f2/fs);%使用fir1函數設計濾波器
%輸入的參數分別是濾波器的階數和截止頻率
figure(1)
[h,f]=freqz(b,1,512);%濾波器的幅頻特性圖
%[H,W]=freqz(B,A,N)當N是一個整數時函數返回N點的頻率向量和幅頻響應向量
plot(f*fs/(2*pi),20*log10(abs(h)))%參數分別是頻率與幅值
xlabel(‘頻率/赫茲’);ylabel(‘增益/分貝’);title(‘濾波器的增益響應’);
figure(2)
subplot(211)
t=0:1/fs:0.5;%定義時間範圍和步長
s=sin(2*pi*f1*t)+sin(2*pi*f2*t);%濾波前信號
plot(t,s);%濾波前的信號圖像
xlabel(‘時間/秒’);ylabel(‘幅度’);title(‘信號濾波前時域圖’);
subplot(212)
Fs=fft(s,512);%將信號變換到頻域
AFs=abs(Fs);%信號頻域圖的幅值
f=(0:255)*fs/512;%頻率採樣
plot(f,AFs(1:256));%濾波前的信號頻域圖
xlabel(‘頻率/赫茲’);ylabel(‘幅度’);title(‘信號濾波前頻域圖’);
figure(3)
sf=filter(b,1,s);%使用filter函數對信號進行濾波
%參數分別為濾波器系統函數的分子和分母多項式係數向量和待濾波信號輸入
subplot(211)
plot(t,sf)%濾波後的信號圖像
xlabel(‘時間/秒’);ylabel(‘幅度’);title(‘信號濾波後時域圖’);
axis([0.2 0.5 -2 2]);%限定圖像坐標範圍
subplot(212)
Fsf=fft(sf,512);%濾波後的信號頻域圖
AFsf=abs(Fsf);%信號頻域圖的幅值
f=(0:255)*fs/512;%頻率採樣
plot(f,AFsf(1:256))%濾波後的信號頻域圖
xlabel(‘頻率/赫茲’);ylabel(‘幅度’);title(‘信號濾波後頻域圖’);
二階濾波器用C語言怎麼寫
這個可比你想像的複雜多了，s是個復變量，1/(s+1)極點在-1，要想用C語言寫，必須理解清楚下面幾個問題：
1、輸入必須是個有限序列，比如(x+yi)，x和y分別是兩個長度為N的數組
2、要過濾的頻率，必須是個整型值，或者是個整型區間
3、輸出結果同樣是兩個長度為N的數組(p+qi)
4、整個程序需要使用最基本的複數運算，這一點C語言本身不提供，必須手工寫複函數運算庫
5、實現的時候具體算法還需要編，這裡才是你問題的核心。
我可以送你一段FFT的程序，自己琢磨吧，和MATLAB的概念差別很大：
#include assert.h
#include math.h
#include stdio.h
#include stdlib.h
#include string.h
#include windows.h
#include “complex.h”
extern “C” {
// Discrete Fourier Transform (Basic Version, Without Any Enhancement)
// return – Without Special Meaning, constantly, zero
int DFT (long count, CComplex * input, CComplex * output)
{
assert(count);
assert(input);
assert(output);
CComplex F, X, T, W; int n, i;
long N = abs(count); long Inversing = count 0? 1: -1;
for(n = 0; n N ; n++){ // compute from line 0 to N-1
F = CComplex(0.0f, 0.0f); // clear a line
for(i = 0; i N; i++) {
T = input[i];
W = HarmonicPI2(Inversing * n * i, N);
X = T * W;
F += X; // fininshing a line
}//next i
// save data to outpus
memcpy(output + n, F, sizeof(F));
}//next n
return 0;
}//end DFT
int fft (long count, CComplex * input, CComplex * output)
{
assert(count);
assert(input);
assert(output);
int N = abs(count); long Inversing = count 0? -1: 1;
if (N % 2 || N 5) return DFT(count, input, output);
long N2 = N / 2;
CComplex * iEven = new CComplex[N2]; memset(iEven, 0, sizeof(CComplex) * N2);
CComplex * oEven = new CComplex[N2]; memset(oEven, 0, sizeof(CComplex) * N2);
CComplex * iOdd = new CComplex[N2]; memset(iOdd , 0, sizeof(CComplex) * N2);
CComplex * oOdd = new CComplex[N2]; memset(oOdd , 0, sizeof(CComplex) * N2);
int i = 0; CComplex W;
for(i = 0; i N2; i++) {
iEven[i] = input[i * 2];
iOdd [i] = input[i * 2 + 1];
}//next i
fft(N2 * Inversing, iEven, oEven);
fft(N2 * Inversing, iOdd, oOdd );
for(i = 0; i N2; i++) {
W = HarmonicPI2(Inversing * (- i), N);
output[i] = oEven[i] + W * oOdd[i];
output[i + N2] = oEven[i] – W * oOdd[i];
}//next i
return 0;
}//end FFT
void __stdcall FFT(
long N, // Serial Length, N 0 for DFT, N 0 for iDFT – inversed Discrete Fourier Transform
double * inputReal, double * inputImaginary, // inputs
double * AmplitudeFrequences, double * PhaseFrequences) // outputs
{
if (N == 0) return;
if (!inputReal !inputImaginary) return;
short n = abs(N);
CComplex * input = new CComplex[n]; memset(input, 0, sizeof(CComplex) * n);
CComplex * output= new CComplex[n]; memset(output,0, sizeof(CComplex) * n);
double rl = 0.0f, im = 0.0f; int i = 0;
for (i = 0; i n; i++) {
rl = 0.0f; im = 0.0f;
if (inputReal) rl = inputReal[i];
if (inputImaginary) im = inputImaginary[i];
input[i] = CComplex(rl, im);
}//next i
int f = fft(N, input, output);
double factor = n;
//factor = sqrt(factor);
if (N 0)
factor = 1.0f;
else
factor = 1.0f / factor;
//end if
for (i = 0; i n; i++) {
if (AmplitudeFrequences) AmplitudeFrequences[i] = output[i].getReal() * factor;
if (PhaseFrequences) PhaseFrequences[i] = output[i].getImaginary() * factor;
}//next i
delete [] output;
delete [] input;
return ;
}//end FFT
int __cdecl main(int argc, char * argv[])
{
fprintf(stderr, “%s usage:\n”, argv[0]);
fprintf(stderr, “Public Declare Sub FFT Lib \”wfft.exe\” \
(ByVal N As Long, ByRef inputReal As Double, ByRef inputImaginary As Double, \
ByRef freqAmplitude As Double, ByRef freqPhase As Double)”);
return 0;
}//end main
};//end extern “C”
c語言中值濾波問題？
1. 是規定做中值濾波的點不含邊緣的點(取決於中值濾波窗口大小)。 2,對圖像邊緣部分的信息進行鏡像處理。
如何用C語言實現低通濾波器
float middle_filter(float middle_value [] , intcount)
{
float sample_value, data;
int i, j;
for (i=1; i for(j=count-1; j=i,–j){
if(middle_value[j-1]=middle_value[j]{
data=middle_value[j-1];
middle_value[j-1]=middle_value[j]
middle_value[j]=data;
}
}
sample_value=middle_value(count-1)/2];
return(sample_value);
}
幫幫忙，能不能給我 基於C語言的FIR濾波器設計的程序代碼（包括CMD，C,ASM)，謝謝了 真的很急！！！
#include”math.h”
void firwin(n,band,fln,fhn,wn,h)
int n,band,wn;
double fln,fhn,h[];
{int i,n2,mid;
double s,pi,wc1,wc2,beta,delay;
double window();
beta=0.0;
if(wn==7)
{printf(“input beta parameter of Kaiser window(2beta10)\n”);
scanf(“%1f”,beta);
}
pi=4.0*atan(1.0);
if((n%2)==0)/*如果n是偶數*/
{n2=n/2+1;/*這行什麼意思*/
mid=1;
}
else
{n2=n/2;
mid=0;
}
delay=n/2.0;
wc1=2.0*pi*fln;
if(band=3) wc2=2.0*pi*fhn;/*先判斷用戶輸入的數據，如果band參數大於3*/
switch(band)
{case 1:
{for(i=0;i=n2;i++)
{s=i-delay;
h[i]=(sin(wc1*s)/(pi*s))*window(wn,n+1,i,beta);
h[n-i]=h[i];
}
if(mid==1) h[n/2]=wc1/pi;
break;
}
case 2:
{for(i=0;i=n2;i++)
{s=i-delay;
h[i]=(sin(pi*s)-sin(wc1*s))/(pi*s);
h[i]=h[i]*window(wn,n+1,i,beta);
h[n-i]=h[i];
}
if(mid==1) h[n/2]=1.0-wc1/pi;
break;
}
case 3:
{for(i=0;in2;i++)
{s=i-delay;
h[i]=(sin(wc2*s)-sin(wc1*s))/(pi*s);
h[i]=h[i]*window(wn,n+1,i,beta);
h[n-i]=h[i];
}
if(mid==1)h[n/2]=(wc2-wc1)/pi;
break;
}
case 4:
{for(i=0;i=n2;i++)
{s=i-delay;
h[i]=(sin(wc1*s)+sin(pi*s)-sin(wc2*s))/(pi*s);
h[i]=h[i]*window(wn,n+1,i,beta);
h[n-i]=h[i];
}
if(mid==1)h[n/2]=(wc1+pi-wc2)/pi;
break;
}
}
}
static double window(type,n,i,beta)
int i,n,type;
double beta;
{int k;
double pi,w;
double kaiser();
pi=4.0*atan(1.0);
w=1.0;
switch(type)
{case 1:
{w=1.0;
break;
}
case 2:
{k=(n-2)/10;
if(i=k)
w=0.5*(1.0-cos(i*pi/(k+1)));
break;
}
case 3:
{w=1.0-fabs(1.0-2*i/(n-1.0));
break;
}
case 4:
{w=0.5*(1.0-cos(2*i*pi/(n-1)));
break;
}
case 5:
{w=0.54-0.46*cos(2*i*pi/(n-1));
break;
}
case 6:
{w=0.42-0.5*cos(2*i*pi/(n-1))+0.08*cos(4*i*pi/(n-1));
break;
}
case 7:
{w=kaiser(i,n,beta);
break;
}
}
return(w);
}
static double kaiser(i,n,beta)
int i,n;
double beta;
{
double a,w,a2,b1,b2,beta1;
double bessel0();
b1=bessel0(beta);
a=2.0*i/(double)(n-1)-1.0;
a2=a*a;
beta1=beta*sqrt(1.0-a2);
b2=bessel0(beta1);
w=b2/b1;
return(w);
}
static double bessel0(x)
double x;
{int i;
double d,y,d2,sum;
y=x/2.0;
d=1.0;
sum=1.0;
for(i=1;i=25;i++)
{d=d*y/i;
d2=d*d;
sum=sum+d2;
if(d2sum*(1.0e-8)) break;
}
return(sum);
}
這是窗函數法的，當然還有其他的比如切比雪夫，零相位濾波什麼的，我也在研究，不是很懂哈
原創文章，作者：小藍，如若轉載，請註明出處：https://www.506064.com/zh-hk/n/159772.html
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