C++ AudioBuffer::getSize方法代码示例

    void processAudio(AudioBuffer &buffer) {
        double rate = getSampleRate();

        float p1 = getRampedParameterValue(PARAMETER_A);
        float freq1 = p1*p1 * (MAX_FREQ-MIN_FREQ) + MIN_FREQ;
        double step1 = freq1 / rate;
        float amt1 = getRampedParameterValue(PARAMETER_B);

        float p2 = getRampedParameterValue(PARAMETER_C);
        float freq2 = p2*p2 * (MAX_FREQ-MIN_FREQ) + MIN_FREQ;
        float amt2 = getRampedParameterValue(PARAMETER_D);
        double step2 = freq2 / rate;

        int size = buffer.getSize();

        for(int ch = 0; ch<buffer.getChannels(); ++ch)
        {
            float* buf = buffer.getSamples(ch);

            for (int i=0; i<size; ++i)
            {
                float mod1 = sin(2 * M_PI * phase1) / 2 + .5; // 0..1
                float mod2 = sin(2 * M_PI * phase2) / 2 + .5; // 0..1
                float gain1 = (amt1 * mod1) + (1 - amt1);
                float gain2 = (amt2 * mod2) + (1 - amt2);
                buf[i] = (gain1 * gain2) * buf[i];
                phase1 += step1;
                phase2 += step2;
            }
        }

    }

    void processAudio(AudioBuffer &buffer){

    double rate = getSampleRate();
    

    unsigned int sampleDelay = getSampleDelay(getRampedParameterValue(PARAMETER_A), rate);
    sampleDelay = min(sampleDelay, bufferSize);
    float feedback = getRampedParameterValue(PARAMETER_B);
    float bias = getBiasExponent(1 - getRampedParameterValue(PARAMETER_C));
    float dryWetMix = getRampedParameterValue(PARAMETER_D);
    

    int size = buffer.getSize();

 	for(int ch = 0; ch<buffer.getChannels(); ++ch)
 	{
	    float* buf = buffer.getSamples(ch);

	    for (int i=0; i<size; ++i)
	    {
	      float delaySample = circularBuffer[writeIdx];
	      float v = buf[i] + circularBuffer[writeIdx] * feedback;
	      v = applyBias(v, bias);
	      circularBuffer[writeIdx] = min(1, max(-1, v)); // Guard: hard range limits.
	      buf[i] = linearBlend(buf[i], delaySample, dryWetMix);

	      writeIdx = (++writeIdx) % sampleDelay;
	    }
		
  	}
  }

  void processAudio(AudioBuffer& buf){
    float minf = getParameterValue(PARAMETER_A)*0.1 + 0.001;
    float maxf = min(0.4, minf + getParameterValue(PARAMETER_B)*0.2);
    // range should be exponentially related to minf
    //    int tones = getParameterValue(PARAMETER_C)*(TONES-1) + 1;
    int tones = 12;
    float spread = getParameterValue(PARAMETER_C) + 1.0;
    float rate = 1.0 + (getParameterValue(PARAMETER_D) - 0.5)*0.00002;
    int size = buf.getSize();
    FloatArray out = buf.getSamples(LEFT_CHANNEL);
    float amp;
    for(int t=1; t<tones; ++t)
      inc[t] = inc[t-1]*spread;
    for(int i=0; i<size; ++i){
      for(int t=0; t<tones; ++t){
	amp = getAmplitude((inc[t]-minf)/(maxf-minf));
	out[i] += amp * getWave(acc[t]);
        acc[t] += inc[t];
	if(acc[t] > 1.0)
	  acc[t] -= 1.0;
	else if(acc[t] < 0.0)
	  acc[t] += 1.0;
        inc[t] *= rate;
      }
    }
    if(inc[0] > maxf)
      inc[0] = minf;
      // while(inc[0] > minf)
      // 	inc[0] *= 0.5;
    else if(inc[0] < minf)
      inc[0] = maxf;
      // while(inc[0] < maxf)
      // 	inc[0] *= 2.0;
  }

 void processAudio(AudioBuffer &buffer){
   if(isButtonPressed(PUSHBUTTON))
     reset();
   dt = getParameterValue(PARAMETER_A)*getParameterValue(PARAMETER_A)*0.0250;
   float rotateX = getParameterValue(PARAMETER_B)*M_PI;
   float rotateY = getParameterValue(PARAMETER_C)*M_PI;
   float rotateZ = getParameterValue(PARAMETER_E)*M_PI;
   float gainL, gainR;
   gainL = gainR = getParameterValue(PARAMETER_D)*2/25.0;
   int size = buffer.getSize();
   float* left = buffer.getSamples(0);
   float* right = buffer.getSamples(1);
   float dx, dy, dz;
   updateMatrix(rotateX, rotateY, rotateZ);
   for(int i=0;i<size;i++){
     dx = a*(y - x);
     dy = (x * (c - z) - y);
     dz = (x*y - b * z);
     x += dx*dt;
     y += dy*dt;
     z += dz*dt;
     P[0] = x;
     P[1] = y;
     P[2] = z-25; // centre on z axis
     rotateP();
     left[i] = Pprime[0] * gainL;
     right[i] = Pprime[1] * gainR;
   }
   // debugMessage("x/y/z", x, y, z);
 }

  void processAudio(AudioBuffer &buffer) {
        
    int size  = buffer.getSize();
    float y;
        
    rate      = Rate(getParameterValue(PARAMETER_A));
    depth     = getParameterValue(PARAMETER_B);
    feedback  = getParameterValue(PARAMETER_C);
        
    //calculate and update phaser sweep lfo...
    float d  = _dmin + (_dmax-_dmin) * ((sin( _lfoPhase ) + 1.f)/2.f);
        
    _lfoPhase += rate;
    if( _lfoPhase >= M_PI * 2.f )
      _lfoPhase -= M_PI * 2.f;
        
    //update filter coeffs
    for( int i=0; i<6; i++ )
      _alps[i].Delay( d );
      
      
//       for (int ch = 0; ch<buffer.getChannels(); ++ch) {
          
            float* buf  = buffer.getSamples(0);
            for (int i = 0; i < size; i++) {
              //calculate output
              y = _alps[0].Update(_alps[1].Update(_alps[2].Update(_alps[3].Update(_alps[4].Update(
                                                      _alps[5].Update( buf[i] + _zm1 * feedback ))))));
              _zm1 = y;
                
              buf[i] = buf[i] + y * depth;
                
//             }
      }
  }

  void processAudio(AudioBuffer &buffer){
    float y[getBlockSize()];
    setCoeffs(getLpFreq(), 0.8f);
    float delayTime = getParameterValue(PARAMETER_A); // get delay time value    
    float feedback  = getParameterValue(PARAMETER_B); // get feedback value
    float wetDry    = getParameterValue(PARAMETER_D); // get gain value

    if(abs(time - delayTime) < 0.01)
      delayTime = time;
    else
      time = delayTime;
        
    float delaySamples = delayTime * (delayBuffer.getSize()-1);        
    int size = buffer.getSize();
    float* x = buffer.getSamples(0);
    process(size, x, y);     // low pass filter for delay buffer
    for(int n = 0; n < size; n++){
        
      //linear interpolation for delayBuffer index
      dSamples = olddelaySamples + (delaySamples - olddelaySamples) * n / size;
        
      y[n] = y[n] + feedback * delayBuffer.read(dSamples);
      x[n] = (1.f - wetDry) * x[n] + wetDry * y[n];  //crossfade for wet/dry balance
      delayBuffer.write(x[n]);
    }
    olddelaySamples = delaySamples;
  }

 void processAudio(AudioBuffer &buffer)
 {
     // Reasonably assume we will not have more than 32 channels
     float*  ins[32];
     float*  outs[32];
     int     n = buffer.getChannels();
     
     if ( (fDSP.getNumInputs() < 32) && (fDSP.getNumOutputs() < 32) ) {
         
         // create the table of input channels
         for(int ch=0; ch<fDSP.getNumInputs(); ++ch) {
             ins[ch] = buffer.getSamples(ch%n);
         }
         
         // create the table of output channels
         for(int ch=0; ch<fDSP.getNumOutputs(); ++ch) {
             outs[ch] = buffer.getSamples(ch%n);
         }
         
         // read OWL parameters and updates corresponding Faust Widgets zones
         fUI.update(); 
         
         // Process the audio samples
         fDSP.compute(buffer.getSize(), ins, outs);
     }
 }

  void processAudio(AudioBuffer &buffer) {
    float delayTime, feedback, wetDry;
    delayTime = getParameterValue(PARAMETER_A);
    feedback  = getParameterValue(PARAMETER_B);
    wetDry    = getParameterValue(PARAMETER_D);
    int size = buffer.getSize();
    int32_t newDelay;
    if(abs(time - delayTime) > 0.01){
      newDelay = delayTime * (delayBuffer.getSize()-1);
      time = delayTime;
    }else{
      newDelay = delay;
    }
    float* x = buffer.getSamples(0);
    float y;
    for (int n = 0; n < size; n++){
//       y = buf[i] + feedback * delayBuffer.read(delay);
//       buf[i] = wetDry * y + (1.f - wetDry) * buf[i];
//       delayBuffer.write(buf[i]);
      if(newDelay - delay > 4){
	y = getDelayAverage(delay-5, 5);
	delay -= 5;
      }else if(delay - newDelay > 4){
	y = getDelayAverage(delay+5, 5);
	delay += 5;
      }else{
	y = delayBuffer.read(delay);
      }
      x[n] = wetDry * y + (1.f - wetDry) * x[n];  // crossfade for wet/dry balance
      delayBuffer.write(feedback * x[n]);
    }
  }

  void processAudio(AudioBuffer &buffer){
    
    setCoeffs(getLpFreq(), 0.8f);
        
    float delayTime = getParameterValue(PARAMETER_A); // get delay time value    
    float feedback  = getParameterValue(PARAMETER_B); // get feedback value
    float wetDry    = getParameterValue(PARAMETER_D); // get gain value
        
    float delaySamples = delayTime * (DELAY_BUFFER_LENGTH-1);
        
    int size = buffer.getSize();
      
      for (int ch = 0; ch<buffer.getChannels(); ++ch) {
          
          float* buf = buffer.getSamples(ch);
          process(size, buf, outBuf);     // low pass filter for delay buffer
          
          for(int i = 0; i < size; i++){

              outBuf[i] = outBuf[i] + feedback * delayBuffer.read(delaySamples);
              buf[i] = (1.f - wetDry) * buf[i] + wetDry * outBuf[i];  //crossfade for wet/dry balance
              delayBuffer.write(buf[i]);
          }
      }
  }

void PatchController::process(AudioBuffer& buffer){
  if(activeSlot == GREEN && green.index != settings.patch_green){
    memset(buffer.getSamples(0), 0, buffer.getChannels()*buffer.getSize()*sizeof(float));
    // green must be active slot when patch constructor is called
    green.setPatch(settings.patch_green);
    codec.softMute(false);
    debugClear();
    return;
  }else if(activeSlot == RED && red.index != settings.patch_red){
    memset(buffer.getSamples(0), 0, buffer.getChannels()*buffer.getSize()*sizeof(float));
    // red must be active slot when constructor is called
    red.setPatch(settings.patch_red);
    codec.softMute(false);
    debugClear();
    return;
  }
  switch(mode){
  case SINGLE_MODE:
  case DUAL_GREEN_MODE:
    green.setParameterValues(getAnalogValues());
    green.patch->processAudio(buffer);
    break;
  case DUAL_RED_MODE:
    red.setParameterValues(getAnalogValues());
    red.patch->processAudio(buffer);
    break;
  case SERIES_GREEN_MODE:
    green.setParameterValues(getAnalogValues());
    green.patch->processAudio(buffer);
    red.patch->processAudio(buffer);
    break;
  case SERIES_RED_MODE:
    red.setParameterValues(getAnalogValues());
    green.patch->processAudio(buffer);
    red.patch->processAudio(buffer);
    break;
  case PARALLEL_GREEN_MODE:
    green.setParameterValues(getAnalogValues());
    processParallel(buffer);
    break;
  case PARALLEL_RED_MODE:
    red.setParameterValues(getAnalogValues());
    processParallel(buffer);
    break;
  }
}

 FixedDelayPatch() {
   AudioBuffer* buffer = createMemoryBuffer(1, REQUEST_BUFFER_SIZE);
   delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
   registerParameter(PARAMETER_A, "Feedback");
   registerParameter(PARAMETER_B, "Mix");
   registerParameter(PARAMETER_C, "");    
   registerParameter(PARAMETER_D, "");    
 }

 SimpleDriveDelayPatch() : delay(0)
   {
   registerParameter(PARAMETER_A, "Delay");
   registerParameter(PARAMETER_B, "Feedback");
   registerParameter(PARAMETER_C, "Drive");
   registerParameter(PARAMETER_D, "Wet/Dry ");
   AudioBuffer* buffer = createMemoryBuffer(1, REQUEST_BUFFER_SIZE);
   delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
     }

  void processAudio(AudioBuffer &buffer){
    float gain = getParameterValue(PARAMETER_A)*2;
    int size = buffer.getSize();
    for(int ch=0; ch<buffer.getChannels(); ++ch){
      float* buf = buffer.getSamples(ch);
      for(int i=0; i<size; ++i)
	buf[i] = gain*buf[i];
    }
  }

 LpfDelayPatch() : x1(0.0f), x2(0.0f), y1(0.0f), y2(0.0f), olddelaySamples(0.0f) {
   AudioBuffer* buffer = createMemoryBuffer(1, REQUEST_BUFFER_SIZE);
   delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
   registerParameter(PARAMETER_A, "Delay", "Delay time");
   registerParameter(PARAMETER_B, "Feedback", "Delay loop feedback");
   registerParameter(PARAMETER_C, "Fc", "Filter cutoff frequency");
   registerParameter(PARAMETER_D, "Dry/Wet", "Dry/wet mix");
   setCoeffs(getLpFreq()/getSampleRate(), 0.6f);
 }    

  void processAudio(AudioBuffer &buffer)
  {

    int size = buffer.getSize();
	
	float samp_float = getParameterValue(PARAMETER_A);
	int samp_freq = ceil(samp_float*63+0.1);
	
	float mayhem_rate = getParameterValue(PARAMETER_B);
	mayhem_rate *= 0.03;
	float mayhem = 1;
	
	if(abs(getParameterValue(PARAMETER_C)*2+1-prev_freq)>0.01)	//if the knob was turned
	{
		mayhem_freq = getParameterValue(PARAMETER_C);	//update center frequency					
		mayhem_freq *= 2;
		mayhem_freq += 1;			//mayhem_freq range = 1 to 3 --> 375 -- 1125 Hz
		prev_freq = mayhem_freq;	//store value to compare next time	
	}
	
	float mayhem_depth = getParameterValue(PARAMETER_D);
	mayhem_depth *= depth;
	
    //for(int ch=0; ch<buffer.getChannels(); ++ch){
      float* buf = buffer.getSamples(0);
      for(int i=0; i<size; ++i)
		{
			if(i%samp_freq==0)
			{	
				buf[i] = buf[i]*((1-mayhem)+mayhem*abs(cos(2*M_PI*mayhem_freq*(i+update_freq_cnt*size)/size)));
				samp = buf[i];	
			}
			else
				buf[i] = samp;
//				buf[i] = samp*(1-mayhem)+buf[i]*mayhem*abs(cos(2*M_PI*mayhem_freq*(i+update_freq_cnt*size)/size));
		}
//		update_freq_cnt++;
//		if(update_freq_cnt == 10)
		{
		update_freq_cnt = 0;
		if(mayhem_freq>=prev_freq+mayhem_depth || mayhem_freq>=3) inc_flag = 0;			//sets maximum freq 3*fs/size = 1125 Hz
		if(mayhem_freq<=prev_freq-mayhem_depth || mayhem_freq<=1)inc_flag = 1;		//minimum freq that can be achieved in 128 samples is 375 Hz
		if(inc_flag == 0) 
		{
			mayhem_freq /= 1+mayhem_rate*mayhem_depth/depth;
	//		freq = floor(fs/size*mayhem_freq);	//only integer frequencies
		}
		
		if(inc_flag == 1) 
		{
			mayhem_freq *= 1+mayhem_rate*mayhem_depth/depth;
	//		freq = ceil(fs/size*mayhem_freq);	//only integer frequencies
		}
		
	//	mayhem_freq = freq*size/fs;		//only integer frequencies
		}
	}