私が読んだものに基づいて、FMサウンド合成のアルゴリズムを作成しました。正しくやったかどうかわかりません。ソフトウェアシンセサイザーを作成する場合、機能を使用してオシレーターを生成し、モジュレーターを使用してこのオシレーターの周波数をモジュール化できます。FMシンセシスが正弦波の変調にのみ機能するのかどうかわかりませんか？
このアルゴリズムは、周波数変調器の機器の波動関数と変調器のインデックスと比率を取得します。ノートごとに、周波数を取得し、キャリアオシレーターとモジュレーターオシレーターの位相値を保存します。変調器は常に正弦波を使用します。
これは、擬似コードのアルゴリズムです。
function ProduceSample(instrument, notes_playing)
for each note in notes_playing
if note.isPlaying()
# Calculate signal
if instrument.FMIndex != 0 # Apply FM
FMFrequency = note.frequency*instrument.FMRatio; # FM frequency is factor of note frequency.
note.FMPhase = note.FMPhase + FMFrequency / kGraphSampleRate # Phase of modulator.
frequencyDeviation = sin(note.FMPhase * PI)*instrument.FMIndex*FMFrequency # Frequency deviation. Max deviation is a factor of the FM frequency. Modulation is done by a sine wave.
note.phase = note.phase + (note.frequency + frequencyDeviation) / kGraphSampleRate # Adjust phase with deviation
# Reset the phase value to prevent the float from overflowing
if note.FMPhase >= 1
note.FMPhase = note.FMPhase - 1
end if
else # No FM applied
note.phase = note.phase + note.frequency / kGraphSampleRate # Adjust phase without deviation
end if
# Calculate the next sample
signal = signal + instrument.waveFunction(note.phase,instrument.waveParameter)*note.amplitude
# Reset the phase value to prevent the float from overflowing
if note.phase >= 1
note.phase = note.phase - 1
end if
end if
end loop
return signal
end function
したがって、ノートの周波数が100Hzの場合、FMRatioは0.5に設定され、FMIndexは0.1であり、50Hzサイクルで95Hzから105Hzの間の周波数を生成するはずです。これは正しい方法ですか。私のテストでは、特にのこぎり波や方形波を変調する場合は、必ずしも正しく聞こえるとは限りません。このように鋸波と方形波を変調しても大丈夫ですか、それとも正弦波のみですか？
これは、CおよびCoreAudioでの実装です。
static OSStatus renderInput(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData){
AudioSynthesiser * audioController = (AudioSynthesiser *)inRefCon;
// Get a pointer to the dataBuffer of the AudioBufferList
AudioSampleType * outA = (AudioSampleType *) ioData->mBuffers[0].mData;
if(!audioController->playing){
for (UInt32 i = 0; i < inNumberFrames; ++i){
outA[i] = (SInt16)0;
}
return noErr;
}
Track * track = &audioController->tracks[inBusNumber];
SynthInstrument * instrument = (SynthInstrument *)track;
float frequency_deviation;
float FMFrequency;
// Loop through the callback buffer, generating samples
for (UInt32 i = 0; i < inNumberFrames; ++i){
float signal = 0;
for (int x = 0; x < 10; x++) {
Note * note = track->notes_playing[x];
if(note){
//Envelope code removed
//Calculate signal
if (instrument->FMIndex) { //Apply FM
FMFrequency = note->frequency*instrument->FMRatio; //FM frequency is factor of note frequency.
note->FMPhase += FMFrequency / kGraphSampleRate; //Phase of modulator.
frequency_deviation = sinf(note->FMPhase * M_PI)*instrument->FMIndex*FMFrequency; //Frequency deviation. Max deviation is a factor of the FM frequency. Modulation is done by a sine wave.
note->phase += (note->frequency + frequency_deviation) / kGraphSampleRate; //Adjust phase with deviation
// Reset the phase value to prevent the float from overflowing
if (note->FMPhase >= 1){
note->FMPhase--;
}
}else{
note->phase += note->frequency/ kGraphSampleRate; //Adjust phase without deviation
}
// Calculate the next sample
signal += instrument->wave_function(note->phase,instrument->wave_parameter)*track->note_amplitude[x];
// Reset the phase value to prevent the float from overflowing
if (note->phase >= 1){
note->phase--;
}
} //Else nothing added
}
if(signal > 1.0){
signal = 1;
}else if(signal < -1.0){
signal = -1.0;
}
audioController->wave[audioController->wave_last] = signal;
if (audioController->wave_last == 499) {
audioController->wave_last = 0;
}else{
audioController->wave_last++;
}
outA[i] = (SInt16)(signal * 32767.0f);
}
return noErr;
}
回答は大歓迎です。