在网上有很多关于混音的算法,但都是处理16bit的,而且是不带P的(左声道与右声道的数据是混在一起的),而我要做处理的是AAC格式混音,解码成PCM对应的是32bit,且左右声道是单独存储的,所以代签了一下他们的代码,整理如下:
#include
#include
#include
/************************************************
For n-bit sampling audio signal,If both A and B
are negative Y = A * B - (A * B / (-(pow(2, n-1) -1)))
else Y = A * B - (A * B / (POW(2,n-1)))
************************************************/
float Mix_001(const float *a, const float *b)
{
float c;
if(*a < 0 && *b < 0)
c = *a + *b - ((*a) * (*b) / -(pow(2,32-1) -1));
else
c = *a + *b - ((*a) * (*b) / (pow(2,32-1)));
c = c > 1 ? 1 : c;
c = c < -1 ? -1 : c;
return c;
}
/****************************
*线性叠加后求平均值
*优点:不会产生溢出,噪音较小
*缺点:衰减过大,影响通话质量
****************************/
float Mix_002(const float *a, const float *b)
{
return (*a + *b) / 2;
}
int main(int argc, char **argv)
{
/*仅支持音频数据为FLTP类型,即一个采样点占4个字节(32bits)*/
if(argc < 4)
{
fprintf(stderr,"Support fltp format,Use it like this:\n");
fprintf(stderr," %s in_1.pcm in_2.pcm out.pcm\n",argv[0]);
return 0;
}
FILE *in_0 = fopen(argv[1],"rb");
FILE *in_1 = fopen(argv[2],"rb");
FILE *out = fopen(argv[3],"wb");
char buf_0[4];
char buf_1[4];
while(1)
{
//读取一个声道的一个采样点
int ret_0 = fread(buf_0,4,1,in_0);
int ret_1 = fread(buf_1,4,1,in_1);
printf("ret_0=%d,ret_1=%d\n",ret_0,ret_1);
if(ret_0 == 0 || ret_1 == 0)
break;
//转换成float类型
float *a = (float*)buf_0;
float *b = (float*)buf_1;
float c = Mix_001(a,b);
//float c = Mix_002(a,b);
fwrite(&c,4,1,out);
}
}
16bit与32bit的算法是一致的,只不过获取音频的数据方法有点区别