C++ MidiMessage::getMidiNoteInHertz方法代码示例

void LyrebirdAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
    
    // Do Midi things
    buffer.clear();
    
    int time;
    MidiMessage m;
    
    for (MidiBuffer::Iterator i (midiMessages); i.getNextEvent (m, time);)
    {
        sawGenerator->setWavelength(currentSampleRate, m.getMidiNoteInHertz(m.getNoteNumber()));
        if (m.isNoteOff())
        {
            sawGenerator->setWavelength(currentSampleRate, 0);
        }
    }
    
    
    // In case we have more outputs than inputs, this code clears any output
    // channels that didn't contain input data, (because these aren't
    // guaranteed to be empty - they may contain garbage).
    // I've added this to avoid people getting screaming feedback
    // when they first compile the plugin, but obviously you don't need to
    // this code if your algorithm already fills all the output channels.
    for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i)
        buffer.clear (i, 0, buffer.getNumSamples());


    float* leftData = buffer.getWritePointer (0);
    float* rightData = buffer.getWritePointer(1);
    for (int sample = 0; sample < buffer.getNumSamples(); sample++)
    {
        leftData[sample] = sawGenerator->getCurrentAmplitude();
        rightData[sample] = sawGenerator->getCurrentAmplitude();
        sawGenerator->incrementSaw();
    }
}

void SynthAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
    const int totalNumInputChannels  = getTotalNumInputChannels();
    const int totalNumOutputChannels = getTotalNumOutputChannels();

    // In case we have more outputs than inputs, this code clears any output
    // channels that didn't contain input data, (because these aren't
    // guaranteed to be empty - they may contain garbage).
    // This is here to avoid people getting screaming feedback
    // when they first compile a plugin, but obviously you don't need to keep
    // this code if your algorithm always overwrites all the output channels.

    MidiBuffer Midi;
    int time;
    MidiMessage m;
    
    for(MidiBuffer::Iterator i(midiMessages); i.getNextEvent(m, time);){
        
        //handle monophonic on/off of notes
        if(m.isNoteOn()){
            noteOn++;
        }
        if(m.isNoteOff()){
            noteOn--;
        }
        if(noteOn > 0){
            monoNoteOn = 1.0f;
            env.reset();
            //handle the pitch of the note
            noteVal = m.getNoteNumber();
            osc.setF(m.getMidiNoteInHertz(noteVal));
        }else{
            monoNoteOn = 0.0f;
        }
    }
    
    for (int i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
        buffer.clear (i, 0, buffer.getNumSamples());

    for (int channel = 0; channel < totalNumOutputChannels; ++channel){
        
        //just do the synth stuff on one channel.
        if(channel == 0){


            for(int sample = 0; sample < buffer.getNumSamples(); ++sample){
  
				//do this stuff here.  it's terribly inefficient..
				freqValScaled = 20000.0f * pow(freqP->get(), 3.0f);
				envValScaled = 10000.0f *  pow(envP->get(), 3.0f);
				speedValScaled = pow((1.0f - speedP->get()), 2.0f);
				oscValScaled = (oscP->get() - 0.5f) * 70.0f;
				detValScaled = (detP->get() - 0.5f) * 24.0f;

                filter.setFc(freqSmoothing.process(freqValScaled + (envValScaled * pow(env.process(),3.0f))) / UPSAMPLING);
                env.setSpeed(speedValScaled);
                filter.setQ(qP->get());
				float frequency = noteVal + 24.0f + oscValScaled + modOsc.process(0) + (driftSmoothing.process(random.nextFloat() - 0.5f) * 20.0f);
                float frequency2 = exp((frequency + detValScaled + (driftSmoothing2.process(random.nextFloat() - 0.5f) * 10.0f)) / 17.31f) / UPSAMPLING;
				frequency = exp(frequency / 17.31f) / UPSAMPLING;
				osc.setF(frequency);
                osc2.setF(frequency2);
                float monoNoteOn2 = ampSmoothing.process(monoNoteOn);
                
                float data;
                
                for(int i = 0; i < UPSAMPLING; i++){
                    data = 20.0f * filter.process(0.1f * osc.process() + ampP->get() * 0.1f * osc2.process());
                }
                
                data *= monoNoteOn2;
                
                buffer.setSample(0, sample, data);
                buffer.setSample(1, sample, data);
            }
        }
    }
}