ardupilot/libraries/AP_HAL_SITL/Synth.hpp

193 lines
6.4 KiB
C++
Raw Normal View History

// See https://github.com/OneLoneCoder/synth
#ifndef SYNTH_HPP
#define SYNTH_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio.hpp>
#include <cmath>
namespace Synth
{
enum eWaveType { OSC_SINE, OSC_SQUARE, OSC_TRIANGLE, OSC_SAW_ANA, OSC_SAW_DIG, OSC_NOISE };
struct sEnvelope
{
double dAttackTime = 0.0;
double dDecayTime = 0.0;
double dSustainTime = 1.0;
double dReleaseTime = 0.0;
double dStartAmplitude = 1.0;
double dSustainAmplitude = 1.0;
};
struct sTone
{
eWaveType waveType = OSC_SINE;
double dStartFrequency = 440.0;
double dEndFrequency = 440.0;
double dAmplitude = 1.0;
};
////////////////////////////////////////////////////////////
// Converts frequency (Hz) to angular velocity
////////////////////////////////////////////////////////////
const double PI = 3.14159265359;
double w(const double dHertz)
{
return dHertz * 2.0 * PI;
}
////////////////////////////////////////////////////////////
// Multi-Function Oscillator
//
// This function was mostly "borrowed" from One Lone Coder blog at
// wwww.onelonecoder.com
//
////////////////////////////////////////////////////////////
double osc(double dHertz, double dTime, eWaveType waveType)
{
switch (waveType)
{
case OSC_SINE: // Sine wave bewteen -1 and +1
return sin(w(dHertz) * dTime);
case OSC_SQUARE: // Square wave between -1 and +1
return sin(w(dHertz) * dTime) > 0 ? 1.0 : -1.0;
case OSC_TRIANGLE: // Triangle wave between -1 and +1
return asin(sin(w(dHertz) * dTime)) * (2.0 / PI);
case OSC_SAW_ANA: // Saw wave (analogue / warm / slow)
{
double dOutput = 0.0;
// this variable defines warmth, larger the number harsher and more similar do OSC_SAW_DIG sound becomes
double dWarmth = 30.0;
for (double n = 1.0; n < dWarmth; n++)
dOutput += (sin(n * w(dHertz) * dTime)) / n;
return dOutput * (2.0 / PI);
}
case OSC_SAW_DIG: // Saw Wave (optimised / harsh / fast)
return (2.0 / PI) * (dHertz * PI * fmod(dTime, 1.0 / dHertz) - (PI / 2.0));
case OSC_NOISE: // Pseudorandom noise
return 2.0 * ((double)rand() / (double)RAND_MAX) - 1.0;
default:
return 0.0;
}
}
////////////////////////////////////////////////////////////
// Amplitude Modulator
////////////////////////////////////////////////////////////
double amplitude(double dTime, sEnvelope env)
{
// double dTimeOn = 0;
double dTimeOff = env.dAttackTime + env.dDecayTime + env.dSustainTime;
double dAmplitude = 0.0;
if (dTime > dTimeOff) // Release phase
dAmplitude = ((dTime - dTimeOff) / env.dReleaseTime) * (0.0 - env.dSustainAmplitude) + env.dSustainAmplitude;
else if (dTime > (env.dAttackTime + env.dDecayTime)) // Sustain phase
dAmplitude = env.dSustainAmplitude;
else if (dTime > env.dAttackTime && dTime <= (env.dAttackTime + env.dDecayTime)) // Decay phase
dAmplitude = ((dTime - env.dAttackTime) / env.dDecayTime) * (env.dSustainAmplitude - env.dStartAmplitude) + env.dStartAmplitude;
else if (dTime <= env.dAttackTime) // Attack phase
dAmplitude = (dTime / env.dAttackTime) * env.dStartAmplitude;
// Amplitude should not be negative, check just in case
if (dAmplitude <= 0.000)
dAmplitude = 0.0;
return dAmplitude;
}
////////////////////////////////////////////////////////////
/// \brief Generate sound and store in SoundBuffer
///
/// This function uses case-in
///
/// \param buffer is address to SoundBuffer where the result will be stored
/// \param envelope structure defining the ADSR Envelope
/// \param tones vector of tone structures to be stacked
/// \param master volume for the volume of sound
/// \param sample rate to set quality of sound
///
/// \return True if the sound was generate, false if it failed
///
////////////////////////////////////////////////////////////
bool generate(sf::SoundBuffer* buffer, sEnvelope env, std::vector<sTone> tones, unsigned uMasterVol, unsigned uSampleRate)
{
if (!buffer)
return false;
// Calculate and allocate buffer
double dTotalDuration = env.dAttackTime + env.dDecayTime + env.dSustainTime + env.dReleaseTime;
unsigned iBufferSize = unsigned(dTotalDuration * uSampleRate);
sf::Int16 * iRaw;
iRaw = new sf::Int16[iBufferSize];
// Generate sound
double dIncrement = 1.0 / double(uSampleRate);
double dTime = 0.0;
for (unsigned i = 0; i < iBufferSize; i++)
{
double signal = 0.0;
// Generate multiple tones and stack them together
for (sTone t : tones)
{
double dFrequency = t.dStartFrequency + ((t.dEndFrequency - t.dStartFrequency) * (double(i) / double(iBufferSize)));
signal = signal + (osc(dFrequency, dTime, t.waveType) * t.dAmplitude);
}
// Calculate Amplitude based on ADSR envelope
double dEnvelopeAmplitude = amplitude(dTime, env);
// Apply master volume, envelope and store to buffer
*(iRaw + i) = sf::Int16(uMasterVol * signal * dEnvelopeAmplitude);
dTime += dIncrement;
}
// Load into SFML SoundBuffer
bool bSuccess = buffer->loadFromSamples(iRaw, iBufferSize, 1, uSampleRate);
delete[] iRaw;
return bSuccess;
}
////////////////////////////////////////////////////////////
/// \brief Generate sound and store in SoundBuffer
///
/// This function uses case-in
///
/// \param buffer is address to SoundBuffer where the result will be stored
/// \param envelope structure defining the ADSR Envelope
/// \param tone structure for simple tone definition
/// \param master volume for the volume of sound
/// \param sample rate to set quality of sound
///
/// \return True if the sound was generate, false if it failed
///
////////////////////////////////////////////////////////////
bool generate(sf::SoundBuffer* buffer, sEnvelope env, sTone tone, unsigned uMasterVol, unsigned uSampleRate)
{
std::vector<sTone> tones;
tones.push_back(tone);
return generate(buffer, env, tones, uMasterVol, uSampleRate);
}
}
#endif // SYNTH_HPP