MFC中 中对一组数据进行FFT转换，要得到频率和幅值

去VIRTINS TECHNOLOGY的网站免费下载multi-instrument后，在其\DAQDAOAPIs目录下有个TestDAQVC例程，其中包含一个专门用于信号处理与分析的vtSPA.dll软件包枣世侍，其中包凳吵返盯括一个求频率和幅值的API，SPA_PeakFrequencyDetection( double * xr, double * xi, double * xp, long FFTSize, double SamplingFrequency, double * PeakFrequency, double *PeakFrequencyRMS, double * PeakFrequencyPhase )可供调用。 具体说明参考：www.virtins.com/Signal-Processing-and-Analysis-APIs.pdf。TestDAQVC例程中有调用例子

我做过，但播放模块使用的是Direct Sound。

频谱缺启分析需要用FFT变换，给你提供一下我的代码（c++）

/*==============================================================

快速傅立叶变换 模块

================================================================*/

inline unsigned int NumberOfBitsNeeded( unsigned int p_nSamples )

{

int i

if( p_nSamples <2 )

{

return 0

}

for ( i=0i++ )

{

if( p_nSamples &(1 <<i) ) return i

}

}

inline unsigned int ReverseBits(unsigned int p_nIndex, unsigned int p_nBits)

{

unsigned int i, rev

for(i=rev=0i <p_nBitsi++)

{

rev = (rev <<1) | (p_nIndex &1)

p_nIndex >>= 1

}

return rev

}

inline double Index_to_frequency(unsigned int p_nBaseFreq, unsigned int p_nSamples, unsigned int p_nIndex)

{

if(p_nIndex >= p_nSamples)

{

return 0.0

}

else if(p_nIndex <= p_nSamples/2)

{

return ( (double)p_nIndex / (double)p_nSamples * p_nBaseFreq )

}

else

{

return ( -(double)(p_nSamples-p_nIndex) /乎扮茄岁察 (double)p_nSamples * p_nBaseFreq )

}

}

void FFT (unsigned int p_nSamples, bool p_bInverseTransform, double *p_lpRealIn, double *p_lpImagIn, double *p_lpRealOut, double *p_lpImagOut)

{

if(!p_lpRealIn || !p_lpRealOut || !p_lpImagOut) return

unsigned int NumBits

unsigned int i, j, k, n

unsigned int BlockSize, BlockEnd

double angle_numerator = 2.0 * PI

double tr, ti

if( p_bInverseTransform ) angle_numerator = -angle_numerator

NumBits = NumberOfBitsNeeded ( p_nSamples )

for( i=0i <p_nSamplesi++ )

{

j = ReverseBits ( i, NumBits )

p_lpRealOut[j] = p_lpRealIn[i]

p_lpImagOut[j] = (p_lpImagIn == NULL) ? 0.0 : p_lpImagIn[i]

}

BlockEnd = 1

for( BlockSize = 2BlockSize <= p_nSamplesBlockSize <<= 1 )

{

double delta_angle = angle_numerator / (double)BlockSize

double sm2 = sin ( -2 * delta_angle )

double sm1 = sin ( -delta_angle )

double cm2 = cos ( -2 * delta_angle )

double cm1 = cos ( -delta_angle )

double w = 2 * cm1

double ar[3], ai[3]

for( i=0i <p_nSamplesi += BlockSize )

{

ar[2] = cm2

ar[1] = cm1

ai[2] = sm2

ai[1] = sm1

for ( j=i, n=0n <BlockEndj++, n++ )

{

ar[0] = w*ar[1] - ar[2]

ar[2] = ar[1]

ar[1] = ar[0]

ai[0] = w*ai[1] - ai[2]

ai[2] = ai[1]

ai[1] = ai[0]

k = j + BlockEnd

tr = ar[0]*p_lpRealOut[k] - ai[0]*p_lpImagOut[k]

ti = ar[0]*p_lpImagOut[k] + ai[0]*p_lpRealOut[k]

p_lpRealOut[k] = p_lpRealOut[j] - tr

p_lpImagOut[k] = p_lpImagOut[j] - ti

p_lpRealOut[j] += tr

p_lpImagOut[j] += ti

}

}

BlockEnd = BlockSize

}

if( p_bInverseTransform )

{

double denom = (double)p_nSamples

for ( i=0i <p_nSamplesi++ )

{

p_lpRealOut[i] /= denom

p_lpImagOut[i] /= denom

}

}

}

/*===================================================================

=====================================================================*/

如果你是做软件的，这些应该看得懂，

还有个使用的实例，我做的声音变调函数：

//*************************音调变换*****************************************

void __stdcall Sd72_Pitch(char *in_fn,//输入文件名

char *in_outfn,//输出文件名

long in_s,//开始位置（单位：采样）

long in_l,//处理长度（单位：采样）

double in_v)//变调幅度（(1,1.8]升调，[0.6,1)降调）

{

jFile fp,dp

fp=in_fn

dp=in_outfn

dp.toend()

unsigned long filelength=dp.len()+in_l*2

long l,i,j=0,k,m=2

double a=in_v,ca=(abs(a-1)>0.3)?0.85:0.75,w,winpower=(a>1)?1:2,at=(a>1)?0.5:2

l=pow(2,13)

long cl,el,jl=1

cl=(long)(ca*l)

el=l-cl

short *e=(short*)malloc(sizeof(short)*l*m)

short *be=(short*)malloc(sizeof(short)*l*m)

short *f=(short*)malloc(sizeof(short)*l*m)

short *bf=(short*)malloc(sizeof(short)*l*m)

double *rin=(double*)malloc(sizeof(double)*l*m)

double *iin=(double*)malloc(sizeof(double)*l*m)

double *rout=(double*)malloc(sizeof(double)*l*m)

double *iout=(double*)malloc(sizeof(double)*l*m)

fp.pointto(in_s*2-1)

fp.in(0,e,sizeof(short)*l)

memcpy(be,e,sizeof(short)*l)

////

ZeroMemory(rin,sizeof(double)*l*m)

ZeroMemory(iin,sizeof(double)*l*m)

ZeroMemory(rout,sizeof(double)*l*m)

ZeroMemory(iout,sizeof(double)*l*m)

w=2*PI/l

for(i=0i<li++)

rin[i]=(double)e[i]*pow(0.5*(cos(w*(i-l*0.5+0.5))+1),winpower)//余弦窗

C_DoEvents()

FFT(l,false,rin,iin,rout,iout)

C_DoEvents()

ZeroMemory(&rout[(long)(l*0.5)],sizeof(double)*l*0.5)

ZeroMemory(&iout[(long)(l*0.5)],sizeof(double)*l*0.5)

ZeroMemory(rin,sizeof(double)*l*m)

ZeroMemory(iin,sizeof(double)*l*m)

if(a>1 &&a<=1.8)

{

FcZero(rout,l*0.5,rin,(l*0.5)*a)

FcZero(iout,l*0.5,iin,(l*0.5)*a)

}

if(a>=0.6 &&a<1)

{

FcTo(rout,l*0.5,rin,(l*0.5)*a)

FcTo(iout,l*0.5,iin,(l*0.5)*a)

}

ZeroMemory(rout,sizeof(double)*l*m)

ZeroMemory(iout,sizeof(double)*l*m)

C_DoEvents()

FFT(l*m,true,rin,iin,rout,iout)

C_DoEvents()

FcTo(rout,l*m,iout,l)//iout出

for(i=0i<li++)

f[i]=(short)(iout[i]*at)

////

memcpy(bf,f,sizeof(short)*l)

while(jl==1)

{

fp.in(0,&e[cl],sizeof(short)*el)

memcpy(e,&be[el],sizeof(short)*cl)

memcpy(be,e,sizeof(short)*l)

////

ZeroMemory(rin,sizeof(double)*l*m)

ZeroMemory(iin,sizeof(double)*l*m)

ZeroMemory(rout,sizeof(double)*l*m)

ZeroMemory(iout,sizeof(double)*l*m)

w=2*PI/l

for(i=0i<li++)

rin[i]=(double)e[i]*pow(0.5*(cos(w*(i-l*0.5+0.5))+1),winpower)//余弦窗

C_DoEvents()

FFT(l,false,rin,iin,rout,iout)

C_DoEvents()

ZeroMemory(&rout[(long)(l*0.5)],sizeof(double)*l*0.5)

ZeroMemory(&iout[(long)(l*0.5)],sizeof(double)*l*0.5)

ZeroMemory(rin,sizeof(double)*l*m)

ZeroMemory(iin,sizeof(double)*l*m)

if(a>1 &&a<=1.8)

{

FcZero(rout,l*0.5,rin,(l*0.5)*a)

FcZero(iout,l*0.5,iin,(l*0.5)*a)

}

if(a>=0.6 &&a<1)

{

FcTo(rout,l*0.5,rin,(l*0.5)*a)

FcTo(iout,l*0.5,iin,(l*0.5)*a)

}

ZeroMemory(rout,sizeof(double)*l*m)

ZeroMemory(iout,sizeof(double)*l*m)

C_DoEvents()

FFT(l,true,rin,iin,rout,iout)

C_DoEvents()

for(i=0i<li++)

f[i]=(short)(rout[i]*at)

////

for(k=0k<clk++)

{

f[k]=f[k]+bf[k+el]// ^_^ 重叠叠加，使声音变平滑

}

dp.out(0,bf,sizeof(short)*el)

in_l-=el

if(in_l<=0)

jl=0

memcpy(bf,f,sizeof(short)*l)

}

fp.leave(NULL,JFILE_NORMAL)

dp.leave(NULL,filelength)

}

//*********************************************************************************

如果不懂，等我高考完了可以和我探讨探讨，QQ：994373259

%mfcc

function mfc=mfcc(x)

%%%%%%%%%%%%%%%%%%%%%%%%%

%对输入的语音序列x进行mfcc参数提取，返回mfcc参数和一阶差分mfcc参数，mel滤波器的阶数为24

%fft变换长度为256，采样频率为8000HZ,对x 256点分为一帧

%%%%%%%%%%%%%%%%%%%%%%%%%%%%

bank=melbankm(24,256,8000,0,0.5,'m')

%归一化mel滤波器组参数

bank=full(bank)

bank=bank/颤滑max(bank((:))

%DCT系数，12*24

for k=1:12

n=0:23

dctcoef(:,k)=cos((2*n+1)*k*pi/(2*24))

end

%归穗银一化倒谱提升窗口

w=1+6*sin(pi*[1:12]./12)

w=w/max(w)

%预加重滤波器

xx=double(x)

xx=filter([1 -0.9375],1,xx)

%语音信号分茄族腊帧

xx=enframe(xx,256,80)

%计算每帧的mfcc参数

for i=1:size(xx,1) %

y=xx(i,:)

s=y'.*hamming(256)

t=abs(fft(s))

t=t.^2%计算能量

c1=dctcoef*log(bank*t(1:129))%dctcoef为dct系数，bank归一化mel滤波器组系数

c2=c1.*w'%w为归一化倒谱提升窗口

m(i,:)=c2'

end

%差分系数

dtm=zeros(size(m))

for i=3:size(m,1)-2

dtm(i,：)=-2*(i-2,:)-m(i-1,1)+2*m(i+2,:)

end

dtm=dtm/3

%合并mfcc参数和一阶差分mfcc参数

mfc=[m dtm]

%去除首尾两帧，因为这两帧的一阶差分参数为0

mfc=mfc(3:size(m,1)-2,:)

---