Method ProcessAudio

Source/Compressor.cpp:79–140

}

void Compressor::ProcessAudio(double time, ChannelBuffer* buffer)
{
   PROFILER(Compressor);

   if (!mEnabled)
      return;

   int bufferSize = buffer->BufferSize();
   mDelayBuffer.SetNumChannels(buffer->NumActiveChannels());

   for (int i = 0; i < bufferSize; ++i)
   {
      ComputeSliders(i);

      // create sidechain

      float input = 0;
      for (int ch = 0; ch < buffer->NumActiveChannels(); ++ch)
         input = MAX(input, fabsf(buffer->GetChannel(ch)[i]));
      input *= mDrive;

      /* if desired, one could use another EnvelopeDetector to smooth
       * the rectified signal.
       */

      // convert key to dB
      input += DC_OFFSET; // add DC offset to avoid log( 0 )
      mCurrentInputDb = lin2dB(input); // convert linear -> dB

      // threshold
      double overdB = mCurrentInputDb - mThreshold; // delta over threshold
      if (overdB < 0.0)
         overdB = 0.0;

      // attack/release

      overdB += DC_OFFSET; // add DC offset to avoid denormal
      mEnv.run(overdB, envdB_); // run attack/release envelope
      overdB = envdB_ - DC_OFFSET; // subtract DC offset

      /* REGARDING THE DC OFFSET: In this case, since the offset is added before
       * the attack/release processes, the envelope will never fall below the offset,
       * thereby avoiding denormals. However, to prevent the offset from causing
       * constant gain reduction, we must subtract it from the envelope, yielding
       * a minimum value of 0dB.
       */

      double invRatio = 1 / mRatio;

      // transfer function
      double reduction = overdB * (invRatio - 1.0); // gain reduction (dB)
      double makeup = (-mThreshold * .5) * (1.0 - invRatio);
      mOutputGain = ofLerp(1, dB2lin(reduction + makeup) * mDrive * mOutputAdjust, mMix);

      // output gain
      for (int ch = 0; ch < buffer->NumActiveChannels(); ++ch)
      {
         mDelayBuffer.Write(buffer->GetChannel(ch)[i], ch);