SoundTouch音频处理库源码分析及算法提取(9)

我们都知道，当播放速度改变的时候，会带来一个副作用，那就是音调也会随着播放的速度而改
变。播放速度加快，声音将升调，变得尖锐，播放速度减慢，声音将降调，变得低沉，当然这是题

外之话。改变音频播放速度的方法有很多，我只针对ST变速算法的核心，也就是线性插值算法的实

现。对于ST的插值算法前面已经提到。

k = (y - y0)/(y1-y0)

->y=(1-k) * y0+k * y1

k值也就是现在声音播放速度和原始声音播放速度的比。也就是加快/减慢的比率。从这个公式上看

，k为比率已知，y1为当前已知的一个采样值，y值和原来的位置y0有关，如何理解呢？假如你把一

连串的采样分成若干段，每次拿一段出来处理，在某段的结尾和紧接的一段之间做插值算法，就要

保存前一段结尾的采样点。这下子好理解了吧。因此针对16位双声道的音频输入，我们需要先定义

两个变量（左右声道）来保存前面的值。

short sPrevSampleL = 0,sPrevSampleR = 0;

然后算法实现如下：

第一个参数表示比率，可以理解为插值的间距，越小帧插得越多，播放速度也就越慢，越大帧插得

少或者直接跳过，播放速度也就越快。

uint InterpolationStereo(float fRate,short *dest,short *src,uint nSamples)

{

uint srcPos = 0, i = 0, used = 0;

float fSlopeCount = 0.0f;



if (nSamples == 0) return 0;

while (fSlopeCount <= 1.0f)

{

dest[2 * i] = (short)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src

[0]);

dest[2 * i + 1] = (short)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount *

src[1]);

i++;

fSlopeCount += fRate;

}

fSlopeCount -= 1.0f;

if (nSamples > 1)

{

while (1)

{

while (fSlopeCount > 1.0f)

{

fSlopeCount -= 1.0f;

used ++;

if (used >= nSamples - 1) goto end;

}

srcPos = 2 * used;

dest[2 * i] = (short)((1.0f - fSlopeCount) * src[srcPos]

+ fSlopeCount * src[srcPos + 2]);

dest[2 * i + 1] = (short)((1.0f - fSlopeCount) * src[srcPos + 1]

+ fSlopeCount * src[srcPos + 3]);

i++;

fSlopeCount += fRate;

}

}

end:

sPrevSampleL = src[2 * nSamples - 2];

sPrevSampleR = src[2 * nSamples - 1];

return i;

}

完整的例程：

//对16位双声道pcm编码的wav文件插值。

//作者：核桃

//修改main函数中float rate变量的值可以改变播放速度

//取值范围最好在0.5~2之间

#include <stdio.h>

#include <stdlib.h>

#define BUFF_SIZE 4096

#define SCALE 3

#ifndef uint

typedef unsigned int uint;

#endif

typedef struct

{

char riff_char[4];

int package_len;

char wave[4];

} WavRiff;

typedef struct

{

char fmt[4];

int format_len;

short fixed;

short channel_number;

int sample_rate;

int byte_rate;

short byte_per_sample;

short bits_per_sample;

} WavFormat;

typedef struct

{

char data_field[4];

uint data_len;

} WavData;

typedef struct

{

WavRiff riff;

WavFormat format;

WavData data;

} WavHeader;

short inbuffer[BUFF_SIZE];

short outbuffer[SCALE*BUFF_SIZE];

short sPrevSampleL = 0,sPrevSampleR = 0;

uint InterpolationStereo(float fRate,short *dest,short *src,uint nSamples);

uint InterpolationStereo(float fRate,short *dest,short *src,uint nSamples)

{

uint srcPos = 0, i = 0, used = 0;

float fSlopeCount = 0.0f;



if (nSamples == 0) return 0;

while (fSlopeCount <= 1.0f)

{

dest[2 * i] = (short)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src

[0]);

dest[2 * i + 1] = (short)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount *

src[1]);

i++;

fSlopeCount += fRate;

}

fSlopeCount -= 1.0f;

if (nSamples > 1)

{

while (1)

{

while (fSlopeCount > 1.0f)

{

fSlopeCount -= 1.0f;

used ++;

if (used >= nSamples - 1) goto end;

}

srcPos = 2 * used;

dest[2 * i] = (short)((1.0f - fSlopeCount) * src[srcPos]

+ fSlopeCount * src[srcPos + 2]);

dest[2 * i + 1] = (short)((1.0f - fSlopeCount) * src[srcPos + 1]

+ fSlopeCount * src[srcPos + 3]);

i++;

fSlopeCount += fRate;

}

}

end:

sPrevSampleL = src[2 * nSamples - 2];

sPrevSampleR = src[2 * nSamples - 1];

return i;

}

int main(int arg, char **argc)

{

FILE *pIn = NULL,*pOut = NULL;

WavHeader *wheader = NULL;

uint isamples = 0,numBytes = 0,totalBytes = 0;

float rate = 0.8f;

if (arg < 3)

{

printf("usage: %s input.wav output.wav/n",argc[0]);

return -1;

}

wheader = (WavHeader *)malloc(sizeof(WavHeader));

pIn = fopen(argc[1], "rb");

pOut = fopen(argc[2], "wb");

numBytes = fread(wheader,1,sizeof(WavHeader),pIn);

fwrite(wheader,1,sizeof(WavHeader),pOut);



while(!feof(pIn))

{

numBytes = fread(inbuffer,1,BUFF_SIZE,pIn);

numBytes = numBytes >> 2;

isamples = InterpolationStereo(rate,outbuffer,inbuffer,numBytes);

isamples = isamples << 2;

fwrite(outbuffer,1,isamples,pOut);

totalBytes += isamples;

}

fseek(pOut, 0, SEEK_SET);

wheader->data.data_len = totalBytes;

wheader->riff.package_len = totalBytes + sizeof(WavHeader)-4*sizeof(char)-



sizeof(int);

fwrite(wheader,1,sizeof(WavHeader),pOut);

free(wheader);

fclose(pIn);

fclose(pOut);

return 0;

}

PS:根据某某人的采样定理，采样频率一定要等于或者大于信号中最高频率的2倍。我们可以理解

为，最快播放速度不宜超过2倍，最慢播放速度不宜低于原来信号的1/2也就是0.5倍，如果不是声

音失真会非常严重。



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