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Algorithm to mix sound

Asked 2008-12-17T21:08:19.533
71

I have two raw sound streams that I need to add together. For the purposes of this question, we can assume they are the same bitrate and bit depth (say 16 bit sample, 44.1khz sample rate).

Obviously if I just add them together I will overflow and underflow my 16 bit space. If I add them together and divide by two, then the volume of each is halved, which isn't correct sonically - if two people are speaking in a room, their voices don't become quieter by half, and a microphone can pick them both up without hitting the limiter.

  • So what's the correct method to add these sounds together in my software mixer?
  • Am I wrong and the correct method is to lower the volume of each by half?
  • Do I need to add a compressor/limiter or some other processing stage to get the volume and mixing effect I'm trying for?

-Adam

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4 Answers

21

Most audio mixing applications will do their mixing with floating point numbers (32 bit is plenty good enough for mixing a small number of streams). Translate the 16 bit samples into floating point numbers with the range -1.0 to 1.0 representing full scale in the 16 bit world. Then sum the samples together - you now have plenty of headroom. Finally, if you end up with any samples whose value goes over full scale, you can either attenuate the whole signal or use hard limiting (clipping values to 1.0).

This will give much better sounding results than adding 16 bit samples together and letting them overflow. Here's a very simple code example showing how you might sum two 16 bit samples together:

short sample1 = ...;
short sample2 = ...;
float samplef1 = sample1 / 32768.0f;
float samplef2 = sample2 / 32768.0f;
float mixed = samplef1 + sample2f;
// reduce the volume a bit:
mixed *= 0.8;
// hard clipping
if (mixed > 1.0f) mixed = 1.0f;
if (mixed < -1.0f) mixed = -1.0f;
short outputSample = (short)(mixed * 32768.0f)
answered 2008-12-18T10:12:22.320
8

I cannot believe that nobody knows the correct answer. Everyone is close enough but still, a pure philosophy. The nearest, i.e. the best was: (s1 + s2) -(s1 * s2). It's excelent approach, especially for MCUs.

So, the algorithm goes:

  1. Find out the volume in which you want the output sound to be. It can be the average or maxima of one of the signals.
    factor = average(s1) You assume that both signals are already OK, not overflowing the 32767.0
  2. Normalize both signals with this factor:
    s1 = (s1/max(s1))*factor
    s2 = (s2/max(s2))*factor
  3. Add them together and normalize the result with the same factor
    output = ((s1+s2)/max(s1+s2))*factor

Note that after the step 1. you don't really need to turn back to integers, you may work with floats in -1.0 to 1.0 interval and apply the return to integers at the end with the previously chosen power factor. I hope I didn't mistake now, cause I'm in a hurry.

answered 2012-12-14T15:43:11.497
2

I think that, so long as the streams are uncorrelated, you shouldn't have too much to worry about, you should be able to get by with clipping. If you're really concerned about distortion at the clip points, a soft limiter would probably work OK.

answered 2008-12-17T21:17:24.977
-1

I'd say just add them together. If you're overflowing your 16 bit PCM space, then the sounds you're using are already incredibly loud to begin with and you should attenuate them. If that would cause them to be too soft by themselves, look for another way of increasing the overall volume output, such as an OS setting or turning the knob on your speakers.

answered 2008-12-17T21:16:58.907

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