按照一般的做法，实现不同频域的音量调节，是使用快速傅氏变换FFT，将PCM时序信号转换成频域值处理，然后再将处理过的数据再做一次ifft转换为时序信号。
一直在考虑有没有更高效的办法来实现这一功能，后来看一下ffdshow在这部分内容的实现，感觉非常神奇，完全没用fft做变换，而是使用bp2函数构造出一个常量表再对信号进行处理，因为程序没有什么注释，函数命名也很简单，没看出什么门道。

大牛们给俺指指门道


/*
* Copyright (c) 2003-2006 Milan Cutka
* implementation (c) 2001 Anders Johansson ajh@atri.curtin.edu.au
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/

#include "stdafx.h"
#include "TaudioFilterEQ.h"

const double TaudioFilterEQ::Q=1.2247449; // Q value for band-pass filters 1.2247=(3/2)^(1/2) gives 4dB suppression @ Fc*2 and Fc/2

TaudioFilterEQ::TaudioFilterEQ(IffdshowBase *Ideci,Tfilters *Iparent):TaudioFilter(Ideci,Iparent)
{
old.eq0=INT_MAX;
oldsf=0;
}

void TaudioFilterEQ::Tequalizer::bp2(float* a, float* b, double fc, double q)
{
double th=2*M_PI*fc;
double C=(1-tan(th*q/2))/(1+tan(th*q/2));
static const double bp2mult=1;

a[0]=float(bp2mult*(1+C)*cos(th));
a[1]=float(bp2mult*-1*C);

b[0]=float(bp2mult*(1.0-C)/2.0);
b[1]=float(bp2mult*-1.0050);
}
void TaudioFilterEQ::Tequalizer::init(const TeqSettings *cfg,double freq)
{
reset();
static const double qmult=1;
for(int i=0;i<AF_NCH;i++)
for(int j=0;j<KM;j++)
{
float db=(cfg->highdb-cfg->lowdb)*(&cfg->eq0)[j]/200.0f+cfg->lowdb;
g[i][j]=(float)(qmult*( db2value(db,100)/* pow(10.0,db/(100*20.0))*/ -1.0));
}
K=KM;
while (((&cfg->f0)[K-1]/100.0)>freq/2.3)
K--;
for (int k=0;k<K;k++)
bp2(a[k],b[k],((&cfg->f0)[k]/100.0)/freq,Q);
}
template<class sample_t> float TaudioFilterEQ::Tequalizer::equalize(const float *g,sample_t in,unsigned int ci)
{
float yt=(float)in;
for (int k=0;k<K;k++)
{
float *wq = this->wq[ci][k];
float w=yt*b[k][0] + wq[0]*a[k][0] + wq[1]*a[k][1];
yt+=(w + wq[1]*b[k][1])*g[k];
wq[1] = wq[0];
wq[0] = w;
}
return yt;
}
template<class sample_t> void TaudioFilterEQ::Tequalizer::process(sample_t *samples,size_t numsamples,unsigned int nchannels)
{
unsigned int ci=nchannels;
while (ci--)
{
const float *g=this->g[ci];
sample_t *inout=samples+ci;
sample_t *end=inout+nchannels*numsamples;
while (inout<end)
{
float yt=equalize(g,*inout,ci);
*inout=TsampleFormatInfo<sample_t>::limit(yt);
inout+=nchannels;
}
}
}

HRESULT TaudioFilterEQ::process(TfilterQueue::iterator it,TsampleFormat &fmt,void *samples,size_t numsamples,const TfilterSettingsAudio *cfg0)
{
const TeqSettings *cfg=(const TeqSettings*)cfg0;

samples=init(cfg,fmt,samples,numsamples);

if (!cfg->equal(old))
{
old=*cfg;
eq.init(cfg,fmt.freq);
}

if (oldsf!=fmt.sf)
{
oldsf=fmt.sf;
eq.reset();
}

switch (fmt.sf)
{
case TsampleFormat::SF_PCM16:eq.process((int16_t*)samples,numsamples,fmt.nchannels);break;
case TsampleFormat::SF_PCM32:eq.process((int32_t*)samples,numsamples,fmt.nchannels);break;
case TsampleFormat::SF_FLOAT32:eq.process((float*)samples,numsamples,fmt.nchannels);break;
}
return parent->deliverSamples(++it,fmt,samples,numsamples);
}

void TaudioFilterEQ::onSeek(void)
{
eq.reset();
}