void OnAudioFilterRead(float[] data, int channels)
{
currentDspTime = AudioSettings.dspTime;
dataLen = data.Length / channels;
chunkTime = dataLen / sampleRate;
dspTimeStep = chunkTime / dataLen;
double preciseDspTime;
for (int i = 0; i < dataLen; i++)
{
preciseDspTime = currentDspTime + i * dspTimeStep;
double signalValue = 0.0;
double currentFreq = mainFrequency;
if (useFrequencyModulation)
{
double freqOffset = (frequencyModulationOscillatorIntensity * mainFrequency * 0.75) / 100.0;
currentFreq += mapValueD(frequencyModulationOscillator.calculateSignalValue(preciseDspTime, frequencyModulationOscillatorFrequency), -1.0, 1.0, -freqOffset, freqOffset);
frequencyModulationRangeOut = (float)frequencyModulationOscillator.calculateSignalValue(preciseDspTime, frequencyModulationOscillatorFrequency) * 0.5f + 0.5f;
}
else
{
frequencyModulationRangeOut = 0.0f;
}
if (useSinusAudioWave)
{
signalValue += 1 * sinusAudioWave.calculateSignalValue(preciseDspTime, currentFreq);
}
if (useAmplitudeModulation)
{
signalValue *= mapValueD(amplitudeModulationOscillator.calculateSignalValue(preciseDspTime, amplitudeModulationOscillatorFrequency), -1.0, 1.0, 0.0, 1.0);
}
else
{
}
float x = masterVolume * 0.5f * (float)signalValue;
for (int j = 0; j < channels; j++)
{
data[i * channels + j] = x;
}
}
}
/*private void LogFrequency ()
* {
* Debug.Log("zeroL: " + zerosL.Count + "\t zeroR: " + zerosR.Count + "\t aligned:" + zerosAlign);
* ZeroCrossing l = zerosL.Last<ZeroCrossing>();
* ZeroCrossing r = zerosR.Last<ZeroCrossing>();
* zerosL.Clear();
* zerosR.Clear();
* zerosL.Add(l);
* zerosR.Add(r);
* zerosAlign = 0;
* }
*/
// Update is called once per frame
//void Update()
//{
// // Debug.Log("avg : " + audioRange.Average() + " | max: " + audioRange.Max() + " | min: " + audioRange.Min());
//}
private void OnAudioFilterRead(float[] data, int channels)
{
currentDspTime = AudioSettings.dspTime;
dataLen = data.Length / channels;
chunkTime = dataLen / sampleRate;
dspTimeStep = chunkTime / dataLen;
double preciseDspTime;
double f = (double)frequency;
double bf = f + (double)brainwave;
for (int i = 0; i < dataLen; i++)
{
preciseDspTime = currentDspTime + i * dspTimeStep;
double signalValueA = 0.0;
double signalValueB = 0.0;
double stepPercentage = (double)i / dataLen;
signalValueA = sinAAudio.calculateSignalValue(preciseDspTime, f, (double)amplitude, stepPercentage);
signalValueB = (double)amplitude * sinBAudio.calculateSignalValue(preciseDspTime, bf, (double)amplitude, stepPercentage);
data[i * channels] = (float)signalValueA;
data[i * channels + 1] = (float)signalValueB;
}
}
void OnAudioFilterRead(float[] data, int channels)
{
/* This is called by the system
* suppose: sampleRate = 48000
* suppose: data.Length = 2048
* suppose: channels = 2
* then:
* dataLen = 2048/2 = 1024
* chunkTime = 1024 / 48000 = 0.0213333... so the chunk time is around 21.3 milliseconds.
* dspTimeStep = 0.0213333 / 1024 = 2.083333.. * 10^(-5) = 0.00002083333..sec = 0.02083 milliseconds
* keep note that 1 / dspTimeStep = 48000 ok!
*/
currentDspTime = AudioSettings.dspTime;
dataLen = data.Length / channels; // the actual data length for each channel
chunkTime = dataLen / sampleRate; // the time that each chunk of data lasts
dspTimeStep = chunkTime / dataLen; // the time of each dsp step. (the time that each individual audio sample (actually a float value) lasts)
double preciseDspTime;
for (int i = 0; i < dataLen; i++) // go through data chunk
{
preciseDspTime = currentDspTime + i * dspTimeStep;
double signalValue = 0.0;
double currentFreq = mainFrequency;
if (useFrequencyModulation)
{
double freqOffset = (frequencyModulationOscillatorIntensity * mainFrequency * 0.75) / 100.0;
currentFreq += mapValueD(frequencyModulationOscillator.calculateSignalValue(preciseDspTime, frequencyModulationOscillatorFrequency), -1.0, 1.0, -freqOffset, freqOffset);
frequencyModulationRangeOut = (float)frequencyModulationOscillator.calculateSignalValue(preciseDspTime, frequencyModulationOscillatorFrequency) * 0.5f + 0.5f;
}
else
{
frequencyModulationRangeOut = 0.0f;
}
if (useSinusAudioWave)
{
signalValue += sinusAudioWaveIntensity * sinusAudioWave.calculateSignalValue(preciseDspTime, currentFreq);
}
if (useSawAudioWave)
{
signalValue += sawAudioWaveIntensity * sawAudioWave.calculateSignalValue(preciseDspTime, currentFreq);
}
if (useSquareAudioWave)
{
signalValue += squareAudioWaveIntensity * squareAudioWave.calculateSignalValue(preciseDspTime, currentFreq);
}
if (useAmplitudeModulation)
{
signalValue *= mapValueD(amplitudeModulationOscillator.calculateSignalValue(preciseDspTime, amplitudeModulationOscillatorFrequency), -1.0, 1.0, 0.0, 1.0);
amplitudeModulationRangeOut = (float)amplitudeModulationOscillator.calculateSignalValue(preciseDspTime, amplitudeModulationOscillatorFrequency) * 0.5f + 0.5f;
}
else
{
amplitudeModulationRangeOut = 0.0f;
}
float x = masterVolume * 0.5f * (float)signalValue;
for (int j = 0; j < channels; j++)
{
data[i * channels + j] = x;
}
}
}
/// <summary>
/// OnAudioFilterRead consente di intercettare ciò che l' audio source collegato a questo oggetto sta riproducendo. In questo modo possiamo modificare
/// ciò che l'audio source sta riproducendo, sia scrivendo nuovi campionamenti all'interno, che modificando l'audio clip che l'audio source sta riproducendo.
/// </summary>
/// <param name="data">Buffer contenente i campionamenti dell'aduio source.. possiamo scrivere qui per far rispodurre suoni all'oggetto audiosource</param>
/// <param name="channels"> Numero di canali disponibili dal motore audio unity. Per audio stereo, channels = 2.</param>
void OnAudioFilterRead(float[] data, int channels)
{
if (_GM.isActive)
{
/*
* 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.
*
*/
currentDspTime = AudioSettings.dspTime; // the current time of the audio system
dataLen = data.Length / channels; // the actual data length for each channel
chunkTime = dataLen / sampleRate; // the time that each chunk of data lasts
dspTimeStep = chunkTime / dataLen; // the time of each dsp step. (the time that each individual audio sample (actually a float value) lasts)
double preciseDspTime; // used to get a precise approximation of the time
nextOutput = 0;
sinOutput = 0;
sawOutput = 0;
sqrOutput = 0;
for (int i = 0; i < dataLen; i++) // go through data chunk
{
preciseDspTime = currentDspTime + i * dspTimeStep; // we calculate the current dsp time adding the time of every step
double currentFreq = frequency; // this lets us modulate the frequency
// Applies Frequency Modulation
if (useFrequencyModulation)
{
double freqOffset = (frequencyModulationOscillatorIntensity * frequency * 0.75) / 100.0;
currentFreq += mapValueD(frequencyModulationOscillator.calculateSignalValue(preciseDspTime, frequencyModulationOscillatorFrequency), -1.0, 1.0, -freqOffset, freqOffset);
frequencyModulationRangeOut = (float)frequencyModulationOscillator.calculateSignalValue(preciseDspTime, frequencyModulationOscillatorFrequency) * 0.5f + 0.5f;
}
else
{
frequencyModulationRangeOut = 0.0f;
}
// the samples calculated for the sine wave
sinOutput = (float)(sinWeight * sinusAudioWave.calculateSignalValue(preciseDspTime, currentFreq));
// the samples calculated for the saw wave
sawOutput = (float)(sawWeight * sawAudioWave.calculateSignalValue(preciseDspTime, currentFreq));
// the samples calculated for the square wave
sqrOutput = (float)(sqrWeight * squareAudioWave.calculateSignalValue(preciseDspTime, currentFreq));
/* Mixa assieme 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);
// Applies Amplitude Modulation
if (useAmplitudeModulation)
{
nextOutput *= (float)mapValueD(amplitudeModulationOscillator.calculateSignalValue(preciseDspTime, amplitudeModulationOscillatorFrequency), -1.0, 1.0, 0.0, 1.0);
amplitudeModulationRangeOut = (float)amplitudeModulationOscillator.calculateSignalValue(preciseDspTime, amplitudeModulationOscillatorFrequency) * 0.5f + 0.5f;
}
else
{
amplitudeModulationRangeOut = 0.0f;
}
// regulates the output based on the current volume of the synth
float x = volume * (float)nextOutput;
// Copies the samples on every available channels of the sound system
for (int j = 0; j < channels; j++)
{
data[i * channels + j] = x;
}
}
}
}