for (i=0;i<frame_size;i++)
for (i=0;i<frame_size*channels;i++)
int read_samples(FILE *fin,int frame_size, int bits, int channels, int lsb, float * input){      unsigned char in[MAX_FRAME_BYTES*2];   int i,d;   short *s;   /*Read input audio*/   fread(in,bits/8*channels, frame_size, fin);   if (feof(fin))      return 1;   s=(short*)in;   if(bits==8)   {      /* Convert 8->16 bits */      for(i=frame_size*channels-1;i>=0;i--)      {         s[i]=(in[i]<<8)^0x8000;      }   } else   {      /* convert to our endian format */      for(i=0;i<frame_size*channels;i++)      {         if(lsb)             s[i]=le_short(s[i]);          else            s[i]=be_short(s[i]);      }   }   if(channels==2)   {      /* downmix to mono */      for(i=0;i<frame_size;i++)      {         d=s[i*2]+s[i*2+1];         s[i]=d>>1;      }   }   /* copy to float input buffer */   for (i=0;i<frame_size;i++)   {      input[i]=(short)s[i];   }   return 0;}