Wav文件直接反映了一个声音在每个时刻的大小值，比如说以下一段波形：   
我们按每人0.1秒取一点，得到的wav文件数值就是0,1,1,-1,0,1。因此，假如我们能把许多Wav文件的数据直接相加，你听到的就是所有的声音，这就是混音器的原理。   Step 1, Get the Raw data of the two files, (Example, of the sample 8bit and 8Kh, means one sample is of 8bit)
Step 2 Let the two audio signal be A and B respectively, the range is between 0 and 255. Where A and B are the Sample Values (Each raw data) And store the resultant into the Y
If Both the samples Values are possitv  Y = A + B - A * B / 255 
Where Y is the resultant signal which contains both signal A and B, merging two audio streams into single 
stream by this method solves the problem of overflow and information loss to an extent. 
If the range of 8-bit sampling is between -127 to 128 
If both A and B are negative Y = A +B - (A * B / (-127)) 
Else Y = A + B - A * B / 128 
Similarly for the nbit (ex 16bit data)
For n-bit sampling audio signal 
If both A and B are negative Y = A + B - (A * B / (-(2 pow(n-1) -1))) 
Else Y = A + B - (A * B / (2 pow(n-1)) 
Step 3.
Add the Header to the Resultant (mixed) data and play back.
If some thing is unclear and ambigious let me know.
Regards
Ranjeet Gupta.
还有简单C程序示意代码，但是其中包含了核心算法： #include 
#include 
#include 
#include 

int main(int argc,char *argv[]) {
char mixname[255];
FILE *pcm1, *pcm2, *mix;
char sample1, sample2;
int value;

pcm1 = fopen(argv[1],"r");
pcm2 = fopen(argv[2],"r");

strcpy (mixname, argv[1]);
strcat (mixname, "_temp.wav");
mix = fopen(mixname, "w");

while(!feof(pcm1)) {

sample1 = fgetc(pcm1);
sample2 = fgetc(pcm2);

if ((sample1 < 0) && (sample2 < 0)) {
value = sample1 + sample2 - (sample1 * sample2 / -(pow(2,16-1)-1));
}else{
value = sample1 + sample2 - (sample1 * sample2 / (pow(2,16-1)-1));
}

fputc(value, mix);
}


fclose(pcm1);
fclose(pcm2);
fclose(mix);

return 0;
}
自己的混音(混音麦克风和扬声器)：16位的数据，双声道              //将PCM叠加
for (int i = 0; i < oAcc->frame_size*2; i=i+2)
{
uint8_t* pMicOut = frame_audioMicOut->extended_data[0] + i;
uint8_t* pMicIn  = frame_audioMicIn->extended_data[0] + i;

short tempMicOut = *(short*)pMicOut;
short tempMicIn  = *(short*)pMicIn;


int tempOut = 0;
if (tempMicOut < 0 && tempMicIn < 0)
tempOut = tempMicOut + tempMicIn - tempMicOut*tempMicIn / (-(pow(2, 15) - 1));
else if (tempMicOut > 0 && tempMicIn > 0)
tempOut = tempMicOut + tempMicIn - tempMicOut*tempMicIn / (pow(2, 15));
pMicIn = (uint8_t*)tempOut;
}

线性叠加后求平均

优点：不会产生溢出，噪音较小； 
缺点：衰减过大，影响通话质量； short remix(short buffer1,short buffer2) { int value = buffer1 + buffer2; return (short)(value/2); }

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归一化混音(自适应加权混音算法)

思路：

使用更多的位数(32 bit)来表示音频数据的一个样本，混完音后在想办法降低其振幅，使其仍旧分布在16 bit所能表示的范围之内，这种方法叫做归一法.

方法：

为避免发生溢出，使用一个可变的衰减因子对语音进行衰减。这个衰减因子也就代表语音的权重，衰减因子随着音频数据的变化而变化，所以称为自适应加权混音。当溢出时，衰减因子较小，使得溢出的数据在衰减后能够处于临界值以内，而在没有溢出时，又让衰减因子慢慢增大，使数据较为平缓的变化. 
代码: void Mix(char sourseFile[10][SIZE_AUDIO_FRAME],int number,char *objectFile) { //归一化混音 int const MAX=32767; int const MIN=-32768; double f=1; int output; int i = 0,j = 0; for (i=0;i2;i++) { int temp=0; for (j=0;jshort*)(sourseFile[j]+i*2); } output=(int)(temp*f); if (output>MAX) { f=(double)MAX/(double)(output); output=MAX; } if (outputdouble)MIN/(double)(output); output=MIN; } if (f<1) { f+=((double)1-f)/(double)32; } *(short*)(objectFile+i*2)=(short)output; } }

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下面是我从newlc上找到的一个关于PCM脉冲编码的音频信号的混音实现，其中包含了一个关键的混音算法！

if( data1 < 0 && data2 < 0) date_mix = data1+data2 - (data1 * data2 / -(pow(2,16-1)-1)); else date_mix = data1+data2 - (data1 * data2 / (pow(2,16-1)-1));

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切割时间片，重采样算法

可以把各个通道的声音叠到一起，让声音的采样率按倍增加，如果提高声音的播放频率，声音可以正常的播放，声音实现了叠加；如果不想修改声音的播放输出频率，可以通过声音的重采样后输出自己想要的输出频率；

下面是上面的混音的测试代码：

#include #include #include #define IN_FILE1 "1.wav" #define IN_FILE2 "2.wav" #define OUT_FILE "remix.pcm" #define SIZE_AUDIO_FRAME (2) void Mix(char sourseFile[10][SIZE_AUDIO_FRAME],int number,char *objectFile) { //归一化混音 int const MAX=32767; int const MIN=-32768; double f=1; int output; int i = 0,j = 0; for (i=0;i2;i++) { int temp=0; for (j=0;jshort*)(sourseFile[j]+i*2); } output=(int)(temp*f); if (output>MAX) { f=(double)MAX/(double)(output); output=MAX; } if (outputdouble)MIN/(double)(output); output=MIN; } if (f<1) { f+=((double)1-f)/(double)32; } *(short*)(objectFile+i*2)=(short)output; } } int main() { FILE * fp1,*fp2,*fpm; fp1 = fopen(IN_FILE1,"rb"); fp2 = fopen(IN_FILE2,"rb"); fpm = fopen(OUT_FILE,"wb"); short data1,data2,date_mix; int ret1,ret2; char sourseFile[10][2]; while(1) { ret1 = fread(&data1,2,1,fp1); ret2 = fread(&data2,2,1,fp2); *(short*) sourseFile[0] = data1; *(short*) sourseFile[1] = data2; if(ret1>0 && ret2>0) { Mix(sourseFile,2,(char *)&date_mix); /* if( data1 < 0 && data2 < 0) date_mix = data1+data2 - (data1 * data2 / -(pow(2,16-1)-1)); else date_mix = data1+data2 - (data1 * data2 / (pow(2,16-1)-1));*/ if(date_mix > pow(2,16-1) || date_mix < -pow(2,16-1)) printf("mix error "); } else if( (ret1 > 0) && (ret2==0)) { date_mix = data1; } else if( (ret2 > 0) && (ret1==0)) { date_mix = data2; } else if( (ret1 == 0) && (ret2 == 0)) { break; } fwrite(&date_mix,2,1,fpm); } fclose(fp1); fclose(fp2); fclose(fpm); printf("Done! "); }