2019-03-19 23:19:41 -03:00
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// See https://github.com/OneLoneCoder/synth
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#ifndef SYNTH_HPP
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#define SYNTH_HPP
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////////////////////////////////////////////////////////////
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// Headers
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////////////////////////////////////////////////////////////
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#include <SFML/Audio.hpp>
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#include <cmath>
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namespace Synth
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{
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enum eWaveType { OSC_SINE, OSC_SQUARE, OSC_TRIANGLE, OSC_SAW_ANA, OSC_SAW_DIG, OSC_NOISE };
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struct sEnvelope
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{
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double dAttackTime = 0.0;
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double dDecayTime = 0.0;
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double dSustainTime = 1.0;
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double dReleaseTime = 0.0;
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double dStartAmplitude = 1.0;
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double dSustainAmplitude = 1.0;
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};
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struct sTone
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{
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eWaveType waveType = OSC_SINE;
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double dStartFrequency = 440.0;
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double dEndFrequency = 440.0;
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double dAmplitude = 1.0;
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};
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2023-06-18 09:07:59 -03:00
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double w(const double dHertz);
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double osc(double dHertz, double dTime, eWaveType waveType);
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double amplitude(double dTime, sEnvelope env);
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bool generate(sf::SoundBuffer* buffer, sEnvelope env, std::vector<sTone> tones, unsigned uMasterVol, unsigned uSampleRate);
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bool generate(sf::SoundBuffer* buffer, sEnvelope env, sTone tone, unsigned uMasterVol, unsigned uSampleRate);
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2019-03-19 23:19:41 -03:00
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////////////////////////////////////////////////////////////
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// Converts frequency (Hz) to angular velocity
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////////////////////////////////////////////////////////////
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const double PI = 3.14159265359;
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double w(const double dHertz)
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{
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return dHertz * 2.0 * PI;
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}
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////////////////////////////////////////////////////////////
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// Multi-Function Oscillator
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//
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// This function was mostly "borrowed" from One Lone Coder blog at
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// wwww.onelonecoder.com
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//
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////////////////////////////////////////////////////////////
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double osc(double dHertz, double dTime, eWaveType waveType)
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{
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switch (waveType)
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{
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case OSC_SINE: // Sine wave bewteen -1 and +1
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return sin(w(dHertz) * dTime);
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case OSC_SQUARE: // Square wave between -1 and +1
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return sin(w(dHertz) * dTime) > 0 ? 1.0 : -1.0;
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case OSC_TRIANGLE: // Triangle wave between -1 and +1
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return asin(sin(w(dHertz) * dTime)) * (2.0 / PI);
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case OSC_SAW_ANA: // Saw wave (analogue / warm / slow)
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{
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double dOutput = 0.0;
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// this variable defines warmth, larger the number harsher and more similar do OSC_SAW_DIG sound becomes
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double dWarmth = 30.0;
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for (double n = 1.0; n < dWarmth; n++)
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dOutput += (sin(n * w(dHertz) * dTime)) / n;
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return dOutput * (2.0 / PI);
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}
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case OSC_SAW_DIG: // Saw Wave (optimised / harsh / fast)
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return (2.0 / PI) * (dHertz * PI * fmod(dTime, 1.0 / dHertz) - (PI / 2.0));
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case OSC_NOISE: // Pseudorandom noise
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return 2.0 * ((double)rand() / (double)RAND_MAX) - 1.0;
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default:
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return 0.0;
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}
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}
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////////////////////////////////////////////////////////////
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// Amplitude Modulator
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////////////////////////////////////////////////////////////
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double amplitude(double dTime, sEnvelope env)
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{
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// double dTimeOn = 0;
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double dTimeOff = env.dAttackTime + env.dDecayTime + env.dSustainTime;
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double dAmplitude = 0.0;
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if (dTime > dTimeOff) // Release phase
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dAmplitude = ((dTime - dTimeOff) / env.dReleaseTime) * (0.0 - env.dSustainAmplitude) + env.dSustainAmplitude;
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else if (dTime > (env.dAttackTime + env.dDecayTime)) // Sustain phase
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dAmplitude = env.dSustainAmplitude;
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else if (dTime > env.dAttackTime && dTime <= (env.dAttackTime + env.dDecayTime)) // Decay phase
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dAmplitude = ((dTime - env.dAttackTime) / env.dDecayTime) * (env.dSustainAmplitude - env.dStartAmplitude) + env.dStartAmplitude;
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else if (dTime <= env.dAttackTime) // Attack phase
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dAmplitude = (dTime / env.dAttackTime) * env.dStartAmplitude;
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// Amplitude should not be negative, check just in case
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if (dAmplitude <= 0.000)
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dAmplitude = 0.0;
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return dAmplitude;
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}
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////////////////////////////////////////////////////////////
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/// \brief Generate sound and store in SoundBuffer
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///
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/// This function uses case-in
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///
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/// \param buffer is address to SoundBuffer where the result will be stored
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/// \param envelope structure defining the ADSR Envelope
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/// \param tones vector of tone structures to be stacked
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/// \param master volume for the volume of sound
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/// \param sample rate to set quality of sound
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///
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/// \return True if the sound was generate, false if it failed
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///
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////////////////////////////////////////////////////////////
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bool generate(sf::SoundBuffer* buffer, sEnvelope env, std::vector<sTone> tones, unsigned uMasterVol, unsigned uSampleRate)
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{
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if (!buffer)
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return false;
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// Calculate and allocate buffer
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double dTotalDuration = env.dAttackTime + env.dDecayTime + env.dSustainTime + env.dReleaseTime;
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unsigned iBufferSize = unsigned(dTotalDuration * uSampleRate);
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sf::Int16 * iRaw;
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iRaw = new sf::Int16[iBufferSize];
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// Generate sound
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double dIncrement = 1.0 / double(uSampleRate);
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double dTime = 0.0;
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for (unsigned i = 0; i < iBufferSize; i++)
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{
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double signal = 0.0;
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// Generate multiple tones and stack them together
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for (sTone t : tones)
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{
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double dFrequency = t.dStartFrequency + ((t.dEndFrequency - t.dStartFrequency) * (double(i) / double(iBufferSize)));
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signal = signal + (osc(dFrequency, dTime, t.waveType) * t.dAmplitude);
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}
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// Calculate Amplitude based on ADSR envelope
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double dEnvelopeAmplitude = amplitude(dTime, env);
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// Apply master volume, envelope and store to buffer
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*(iRaw + i) = sf::Int16(uMasterVol * signal * dEnvelopeAmplitude);
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dTime += dIncrement;
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}
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// Load into SFML SoundBuffer
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bool bSuccess = buffer->loadFromSamples(iRaw, iBufferSize, 1, uSampleRate);
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delete[] iRaw;
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return bSuccess;
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}
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////////////////////////////////////////////////////////////
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/// \brief Generate sound and store in SoundBuffer
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///
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/// This function uses case-in
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///
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/// \param buffer is address to SoundBuffer where the result will be stored
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/// \param envelope structure defining the ADSR Envelope
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/// \param tone structure for simple tone definition
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/// \param master volume for the volume of sound
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/// \param sample rate to set quality of sound
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///
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/// \return True if the sound was generate, false if it failed
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///
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////////////////////////////////////////////////////////////
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bool generate(sf::SoundBuffer* buffer, sEnvelope env, sTone tone, unsigned uMasterVol, unsigned uSampleRate)
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{
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std::vector<sTone> tones;
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tones.push_back(tone);
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return generate(buffer, env, tones, uMasterVol, uSampleRate);
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}
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}
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#endif // SYNTH_HPP
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