inline float Process(float input)
{
float g = 1.0f;
float s = FastClip(input * input, 1.0e-11f, 100.0f);
float timeConst = (s > env) ? atk : rel;
env += (s - env) * timeConst + 1.0e-16f; // add small constant to always positive number to avoid denormal numbers
float sideChainLevel = 10.0f * log10f(env); // multiply by 10 (not 20) because duckEnvelope is RMS
float t = sideChainLevel - thr;
if (fabsf(t) < knee)
{
t += knee;
g = powf(exp1, t * t);
}
else if (t > 0.0f)
g = powf(exp2, t);
reduction = g;
return input * g;
}
UNITY_AUDIODSP_RESULT UNITY_AUDIODSP_CALLBACK ProcessCallback(UnityAudioEffectState* state, float* inbuffer, float* outbuffer, unsigned int length, int inchannels, int outchannels)
{
EffectData::Data* data = &state->GetEffectData<EffectData>()->data;
const float atksamples = data->p[P_ATK] * state->samplerate;
const float relsamples = data->p[P_REL] * state->samplerate;
const float atkconst = (atksamples <= 1.0f) ? 1.0f : (1.0f - powf(0.01f, 1.0f / atksamples));
const float relconst = (relsamples <= 1.0f) ? 1.0f : (1.0f - powf(0.01f, 1.0f / relsamples));
const float bw = powf(1.0f - 0.999f * data->p[P_RESO], 3.0f);
for (int i = 0; i < inchannels; i++)
{
EffectData::Data::Channel& ch = data->channels[i];
ch.filter1.bandwidth = bw;
ch.filter2.bandwidth = bw;
float* src = inbuffer + i;
float* dst = outbuffer + i;
float* sc = state->sidechainbuffer + i;
for (unsigned int n = 0; n < length; n++)
{
float s = *src;
float a = fabsf(s + (*sc - s) * data->p[P_SIDECHAIN]);
ch.env += (a - ch.env) * ((a > ch.env) ? atkconst : relconst);
ch.filter1.cutoff = FastClip(data->p[P_BASE] + ch.env * data->p[P_SENS], 0.0f, 1.4f);
ch.filter2.cutoff = ch.filter1.cutoff;
ch.filter2.ProcessLPF(ch.filter1.ProcessLPF(*src));
float lpf = ch.filter1.lpf + (ch.filter2.lpf - ch.filter1.lpf) * data->p[P_DEPTH];
float bpf = ch.filter1.bpf + (ch.filter2.bpf - ch.filter1.bpf) * data->p[P_DEPTH];
*dst = lpf + (bpf - lpf) * data->p[P_TYPE];
src += inchannels;
dst += outchannels;
sc += inchannels;
}
}
return UNITY_AUDIODSP_OK;
}
