public float frequencyModulationRangeOut; //The Frequency Modulation Oscillator’s current value (range 0 to 1)
void Awake()
{
amplitudeModulationOscillator = new SineHP();
frequencyModulationOscillator = new SineHP();
// Grab the sample rate from the system
_FS = AudioSettings.outputSampleRate;
// Calculate how long each sample lasts
_sampleDuration = 1.0 / _FS;
}
void OnAudioFilterRead(float[] data, int channels)
{
/*
* This is "the current time of the audio system", as given
* by Unity. It is updated every time the OnAudioFilterRead() function
* is called. It's usually every 1024 samples.
*
* A note on the sample rate:
* We don't actually see real numbers for the sample rate, we instead
* read it from the system in the Start() function.
*/
//currentDspTime = AudioSettings.dspTime;
// goes through data chunk
for (int i = 0; i < data.Length; i += channels)
{
/*
* Sample duration is just 1/fs. Because dspTime isn't updated every
* sample, we "update" it ourselves so our envelope sounds smooth.
*/
// currentDspTime += _sampleDuration;
// envelope = ComputeAmplitude(currentDspTime) * volume;
// lets you modulate the frequency
double currentFreq = frequency;
// Applies Frequency Modulation
if (useFrequencyModulation)
{
phaseIncrementFM = (frequencyModulationOscillatorFrequency) * 2.0 * Mathf.PI / _FS;
phaseFM += phaseIncrementFM;
if (phaseFM > (Mathf.PI * 2))
{
phaseFM = phaseFM % (Mathf.PI * 2);
}
double freqOffset = (frequencyModulationOscillatorIntensity * frequency * 0.75) / 100.0;
currentFreq += mapValueD(SineHP.Sin((float)phaseFM), -1.0, 1.0, -freqOffset, freqOffset);
frequencyModulationRangeOut = (float)SineHP.Sin((float)phaseFM) * 0.5f + 0.5f;
}
else
{
frequencyModulationRangeOut = 0.0f;
}
/*
* The phase variable below increments by the calculated amount for
* every audio sample. We can then use this value to calculate what
* each waveform's value should be.
*
* 2pi * f
* phase = -------
* fs
*
* When phase is greater than 2pi, we just reset
* it so we don't have an ever-increasing variable that will cause
* an overflow error.
*/
phaseIncrement = (currentFreq * octave) * 2.0 * Mathf.PI / _FS;
phase += phaseIncrement;
if (phase > (Mathf.PI * 2))
{
phase = phase % (Mathf.PI * 2);
}
nextOutput = 0;
sinOutput = 0;
sawOutput = 0;
sqrOutput = 0;
noiseOutput = 0;
// Adds sinusoidal wave to the next output sample sinOutput = (float)(sinWeight * Mathf.Sin((float)phase));
sinOutput = (float)(sinWeight * SineHP.Sin((float)phase));
// nextOutput += (float)(sinWeight * Mathf.Sin((float)phase));
// Adds sawtooth wave to the next output sample
sawOutput = (float)sawWeight - (float)(sawWeight / Mathf.PI * phase);
// nextOutput += (float)(sawWeight * ((phase) / (2 * Mathf.PI)));
// Adds square wave to the next output sample
if (phase > Mathf.PI)
{
sqrOutput = (float)(sqrWeight);
//nextOutput += (float) (sqrWeight ) ;
}
else
{
sqrOutput = (-(float)(sqrWeight));
//nextOutput += (-(float) ( sqrWeight) ) ;
}
// Adds noise wave to the next output sample
noiseOutput = PinkNoise.Noise();
/* Mixa tutti gli output
* http://www.vttoth.com/CMS/index.php/technical-notes/68
* Let's say we have two signals, A and B. If A is quiet, we want to hear B on the output in unaltered form. If B
* is quiet, we want to hear A on the output (i.e., A and B are treated symmetrically.) If both A and B have a non-zero amplitude,
* the mixed signal must have an amplitude between the greater of A and B, and the maximum permissible amplitude.
* If we take A and B to have values between 0 and 1, there is actually a simple equation that satisfies all of the
* above conditions: Z= A + B − AB.
* Simple, isn't it! Moreover, it can be easily adapted for more than two signals.
* Consider what happens if we mix another signal, C, to Z: T= Z + C − Z C = A + B + C − AB − AC − BC + ABC.
*
*
*/
nextOutput = sinOutput + sawOutput + sqrOutput - (sinOutput * sawOutput) -
(sinOutput * sqrOutput) - (sawOutput * sqrOutput) + (sinOutput * sawOutput * sqrOutput);
nextOutput += noiseOutput * (float)noiseWeight; //da cambiare
/*
* Here we apply a single-pole low-pass filter. Even if the filter
* is completely open (lowPass = 1) we still compute this step so we
* don't have to have any conditional logic.
*/
nextOutput = (float)((nextOutput * lowPass) + (previousOutput * (1 - lowPass)));
nextOutput = nextOutput * 0.5f * volume;
// Applies Amplitude Modulation
if (useAmplitudeModulation)
{
phaseIncrementAM = (amplitudeModulationOscillatorFrequency) * 2.0 * Mathf.PI / _FS;
phaseAM += phaseIncrementAM;
if (phaseAM > (Mathf.PI * 2))
{
phaseAM = phaseAM % (Mathf.PI * 2);
}
nextOutput *= (float)mapValueD(SineHP.Sin((float)phaseAM), -1.0, 1.0, 0.0, 1.0);
amplitudeModulationRangeOut = (float)SineHP.Sin((float)phaseAM) * 0.5f + 0.5f;
}
else
{
amplitudeModulationRangeOut = 0.0f;
}
// Write the output to the audio filter
data[i] += nextOutput;
// This is for the next low-pass calculation
previousOutput = nextOutput;
// Copy the sound from one channel into the next channel for stereo
if (channels == 2)
{
data[i + 1] = data[i];
}
}
}
