I’m currently working on the iPhone with Audio Units and I’m playing four tracks simultaneously. To improve the performance of my setup, I thought it would be a good idea to minimize the number of Audio Units / threads, by mixing down the four tracks into one.
With the following code I’m processing the next buffer by adding up the samples of the four tracks, keep them in the SInt16 range and add them to a temporary buffer, which will later on be copied into the ioData.mBuffers of the Audio Unit.
Although it works, I don’t have the impression that this is the most efficient way to do this.
SInt16* buffer = bufferToWriteTo;
int reads = bufferSize/sizeof(SInt16);
SInt16** files = circularBuffer->files;
float tempValue;
SInt16 values[reads];
int k,j;
int numFiles=4;
for (k=0; k<reads; k++)
{
tempValue=0.f;
for (j=0; j<numFiles; j++)
{
tempValue += files[j][packetNumber];
}
if (tempValue > 32767.f) tempValue = 32767.f;
else if (tempValue < -32768.f) tempValue =- 32768.f;
values[k] = (SInt16) tempValue;
values[k] += values[k] << 16;
packetNumber++;
if (packetNumber >= totalPackets) packetNumber=0;
}
memcpy(buffer,values,bufferSize);
Any ideas or pointers to speed this up? Am I right?
The biggest improvement you can get from this code would be by not using floating point arithmetic. While the arithmetic by itself is fast, the conversions which happen in the nested loops, take a long time, especially on the ARM processor in the iPhone. You can achieve exactly the same results by using ‘SInt32’ instead of ‘float’ for the ‘tempValue’ variable.
Also, see if you can get rid of the memcpy() in the last string: perhaps you can construct the ‘buffer’ directly, without using a temporary buffer called ‘values’. That saves one copy, which would be significant improvement for such a function.
Other notes: the last two lines of the loop probably belong outside of the loop and the body of the nested loop should use ‘k’ as a second index, instead of ‘packetNumber’, but I’m not sure about this logic.
And the last note: you’re squashing the peaks of your resulting sound. While this seems like a good idea, it will sound pretty rough. You probably want to scale the result down instead of cropping it. Like that: instead of this code
you probably want something like this:
Edit: and please do not forget to measure the performance before and after, to see which one of the improvements gave the biggest impact. This is the best way to learn performance: trial and measurement