Help with dynamic range compression function (audio)
- by MusiGenesis
I am writing a C# function for doing dynamic range compression (an audio effect that basically squashes transient peaks and amplifies everything else to produce an overall louder sound). I have written a function that does this (I think):
public static void Compress(ref short[] input, double thresholdDb, double ratio)
{
double maxDb = thresholdDb - (thresholdDb / ratio);
double maxGain = Math.Pow(10, -maxDb / 20.0);
for (int i = 0; i < input.Length; i += 2)
{
// convert sample values to ABS gain and store original signs
int signL = input[i] < 0 ? -1 : 1;
double valL = (double)input[i] / 32768.0;
if (valL < 0.0)
{
valL = -valL;
}
int signR = input[i + 1] < 0 ? -1 : 1;
double valR = (double)input[i + 1] / 32768.0;
if (valR < 0.0)
{
valR = -valR;
}
// calculate mono value and compress
double val = (valL + valR) * 0.5;
double posDb = -Math.Log10(val) * 20.0;
if (posDb < thresholdDb)
{
posDb = thresholdDb - ((thresholdDb - posDb) / ratio);
}
// measure L and R sample values relative to mono value
double multL = valL / val;
double multR = valR / val;
// convert compressed db value to gain and amplify
val = Math.Pow(10, -posDb / 20.0);
val = val / maxGain;
// re-calculate L and R gain values relative to compressed/amplified
// mono value
valL = val * multL;
valR = val * multR;
double lim = 1.5; // determined by experimentation, with the goal
// being that the lines below should never (or rarely) be hit
if (valL > lim)
{
valL = lim;
}
if (valR > lim)
{
valR = lim;
}
double maxval = 32000.0 / lim;
// convert gain values back to sample values
input[i] = (short)(valL * maxval);
input[i] *= (short)signL;
input[i + 1] = (short)(valR * maxval);
input[i + 1] *= (short)signR;
}
}
and I am calling it with threshold values between 10.0 db and 30.0 db and ratios between 1.5 and 4.0. This function definitely produces a louder overall sound, but with an unacceptable level of distortion, even at low threshold values and low ratios.
Can anybody see anything wrong with this function? Am I handling the stereo aspect correctly (the function assumes stereo input)? As I (dimly) understand things, I don't want to compress the two channels separately, so my code is attempting to compress a "virtual" mono sample value and then apply the same degree of compression to the L and R sample value separately. Not sure I'm doing it right, however.
I think part of the problem may the "hard knee" of my function, which kicks in the compression abruptly when the threshold is crossed. I think I may need to use a "soft knee" like this:
Can anybody suggest a modification to my function to produce the soft knee curve?
