C++ AudioBuffer::getReadPointer方法代码示例

void HRIR::update (double azimuth, double elevation)
{
    int nAzimuth = (int) azimuth;
    int nElevation = (int) elevation;
    int taps = mit_hrtf_availability (nAzimuth, nElevation, sampleRate, diffused);

    if (taps)
    {
        leftEar.setSize (1, taps);
        rightEar.setSize (1, taps);

        AudioBuffer<short> leftShort (1, taps);
        AudioBuffer<short> rightShort (1, taps);

        short* const writeLeft = leftShort.getWritePointer (0);
        short* const writeRight = rightShort.getWritePointer (0);

        taps = mit_hrtf_get (&nAzimuth, &nElevation, sampleRate, diffused, writeLeft, writeRight);

        float* const leftEarWrite = leftEar.getWritePointer (0);
        float* const rightEarWrite = rightEar.getWritePointer (0);

        const short* const readLeft = leftShort.getReadPointer (0);
        const short* const readRight = rightShort.getReadPointer (0);

        for (int n = 0; n < taps; ++n)
        {
            leftEarWrite[n] = readLeft[n] / 32768.0f;
            rightEarWrite[n] = readRight[n] / 32768.0f;
        }
    }
}

void VAOscillator::fillBufferSquarePulse(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer)
{
  if(isActive)
  {
    float* const data = buffer.getWritePointer(0);
    float const *phaseMod = phaseModBuffer.getReadPointer(0);
    float const *volMod = volumeModBuffer.getReadPointer(0);
    float const *pitchMod = pitchModBuffer.getReadPointer(0);
    
    //write momentary phase values into the buffer
    for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
    {
      double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex];
      
      while (phase < 0)
      {
        phase += 2 * double_Pi;
      }
      while (phase > 2 * double_Pi)
      {
        phase -= 2 * double_Pi;
      }
      
      currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
      
      if(currentPhase > 2 * double_Pi)
      {
        currentPhase -= 2 * double_Pi;
      }
      
      if(phase < double_Pi)
      {
        data[sampleIndex] = -1;
      }
      else
      {
        data[sampleIndex] = 1;
      }

      data[sampleIndex] += getBlep(phase, currentFrequency);
        
      if(phase < double_Pi)
      {
        data[sampleIndex] -= getBlep(phase + double_Pi, currentFrequency);
      }
      if(phase > double_Pi)
      {
        data[sampleIndex] -= getBlep(phase - double_Pi, currentFrequency);
      }
      
      data[sampleIndex] *= (float)std::abs(0.5 * (volMod[sampleIndex] + 1));
    }
  }
  else
  {
    buffer.clear();
  }
}// end square pulse

void VAOscillator::fillBufferTriangle(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer)
{
  if(isActive)
  {
    float* const data = buffer.getWritePointer(0);
    float const *phaseMod = phaseModBuffer.getReadPointer(0);
    float const *volMod = volumeModBuffer.getReadPointer(0);
    float const *pitchMod = pitchModBuffer.getReadPointer(0);
    
    for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
    {
      double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex];
      
      while (phase < 0)
      {
        phase += 2 * double_Pi;
      }
      while (phase > 2 * double_Pi)
      {
        phase -= 2 * double_Pi;
      }
          
      currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
      
      if(currentPhase > 2 * double_Pi)
      {
        currentPhase -= 2 * double_Pi;
      }
    
      if(phase < double_Pi)
      {
        data[sampleIndex] = (float)((2 * phase)/double_Pi - 1);
      }
      else
      {
        data[sampleIndex] = (float)((3 - 2 * phase/double_Pi));
      }
      
      //the 0.000026 is kind of a magic number i didnt calculate it just found it by trying out
      data[sampleIndex] += (float)(0.000026 * currentFrequency * getTriRes(phase, currentFrequency));
      if(phase < double_Pi)
      {
        data[sampleIndex] -= (float)(0.000026 * currentFrequency * getTriRes(phase + double_Pi, currentFrequency));
      }
      else if(phase >= double_Pi)
      {
        data[sampleIndex] -= (float)(0.000026 * currentFrequency * getTriRes(phase - double_Pi, currentFrequency));
      }
      
      data[sampleIndex] *= (float)std::abs(0.5 * (volMod[sampleIndex] + 1));
    }
  }
  else
  {
    buffer.clear();
  }
}// end triangle

void VAOscillator::fillBufferSine(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer, Array<int>& midiOns)// add midi buffer and know channel with control Voltage
{
  if(isActive)
  {
    float* const data = buffer.getWritePointer(0);
    float const *phaseMod = phaseModBuffer.getReadPointer(0);
    float const *volMod = volumeModBuffer.getReadPointer(0);
    float const *pitchMod = pitchModBuffer.getReadPointer(0);

    for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
    {
      //check if this is the best i can do
      if(midiOns.size() != 0)
      {
        if(sampleIndex == midiOns[0])
        {
          resetPhase();
          midiOns.remove(0);
        }
      }
      
      double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex] + currentPhaseOffset;
      
      while (phase < 0)
      {
        phase += 2 * double_Pi;
      }
      while (phase > 2 * double_Pi)
      {
        phase -= 2 * double_Pi;
      }
      
      data[sampleIndex] = static_cast<float>(sin(phase) * std::abs(0.5 * (volMod[sampleIndex] + 1)));
      
      currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
      
      if(currentPhase > 2 * double_Pi)
      {
        currentPhase -= 2 * double_Pi;
      }
    }
  }
  else
  {
    buffer.clear();
  }
}// end sine

void DaalDelAudioProcessor::processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiMessages)
{
    ScopedNoDenormals noDenormals;
    
    auto totalNumInputChannels  = getTotalNumInputChannels();
    auto 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.
    for (auto i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
        buffer.clear (i, 0, buffer.getNumSamples());
    
    // ====
    // Lengths for circular buffer
    const int bufferLength = buffer.getNumSamples();
    const int delayBufferLength = _delayBuffer.getNumSamples();
    
    
    // This is the place where you'd normally do the guts of your plugin's
    // audio processing...
    // Make sure to reset the state if your inner loop is processing
    // the samples and the outer loop is handling the channels.
    // Alternatively, you can process the samples with the channels
    // interleaved by keeping the same state.
    for (int channel = 0; channel < totalNumInputChannels; ++channel)
    {
        //auto* channelData = buffer.getWritePointer (channel);

        // ..do something to the data...
        
        // Set up circular buffer
        const float* bufferData = buffer.getReadPointer(channel);
        const float* delayBufferData = _delayBuffer.getReadPointer(channel);
        float* dryBuffer = buffer.getWritePointer(channel);
        
        // Apply gains (now do this before getting from delay)
        applyDryWetToBuffer(buffer, channel, bufferLength, dryBuffer);
        
        // Copy data from main to delay buffer
        fillDelayBuffer(channel, bufferLength, delayBufferLength, bufferData, delayBufferData);
        
        // Copy data from delay buffer to output buffer
        getFromDelayBuffer(buffer, channel, bufferLength, delayBufferLength, bufferData, delayBufferData);
        
        // Feedback
        feedbackDelay(channel, bufferLength, delayBufferLength, dryBuffer);
    }
    
    _writePosition += bufferLength; // Increment
    _writePosition %= delayBufferLength; // Wrap around position index
    
    // Update values from tree
    updateTreeParams();
}

void VAOscillator::fillBufferRisingSaw(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer)
{
  if(isActive)
  {
    float* const data = buffer.getWritePointer(0);
    float const *phaseMod = phaseModBuffer.getReadPointer(0);
    float const *volMod = volumeModBuffer.getReadPointer(0);
    float const *pitchMod = pitchModBuffer.getReadPointer(0);
    
    //write momentary phase values into the buffer
    for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
    {
      double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex];
      
      while (phase < 0)
      {
        phase += 2 * double_Pi;
      }
      while (phase > 2 * double_Pi)
      {
        phase -= 2 * double_Pi;
      }
      
      currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
      
      if(currentPhase > 2 * double_Pi)
      {
        currentPhase -= 2 * double_Pi;
      }
      
      data[sampleIndex] = (float)((2 * phase)/(2 * double_Pi) - 1);
      
      data[sampleIndex] += getBlep(phase , currentFrequency);// i have to watch out here because actually currentFrequency is not valid for this calculation
      
      data[sampleIndex] *= std::abs(0.5f * (volMod[sampleIndex] + 1));
    }
  }
  else
  {
    buffer.clear();
  }
}// end rising Saw

bool AudioFormatWriter::writeFromAudioSampleBuffer (const AudioBuffer<float>& source, int startSample, int numSamples)
{
    auto numSourceChannels = source.getNumChannels();
    jassert (startSample >= 0 && startSample + numSamples <= source.getNumSamples() && numSourceChannels > 0);

    if (startSample == 0)
        return writeFromFloatArrays (source.getArrayOfReadPointers(), numSourceChannels, numSamples);

    const float* chans[256];
    jassert ((int) numChannels < numElementsInArray (chans));

    for (int i = 0; i < numSourceChannels; ++i)
        chans[i] = source.getReadPointer (i, startSample);

    chans[numSourceChannels] = nullptr;

    return writeFromFloatArrays (chans, numSourceChannels, numSamples);
}

Result Upsampler::upsample(AudioBuffer<float> const &inputBuffer, int const inputChannel, AudioBuffer<float> &outputBuffer_, int const outputChannel, int &outputSampleCount_)
{
    int outputSamplesNeeded = roundDoubleToInt( (outputSampleRate * inputBuffer.getNumSamples())/ inputSampleRate);
    
    if (outputBuffer_.getNumSamples() < outputSamplesNeeded)
        return Result::fail("Output buffer too small");
    
    //DBG("upsample inputSampleCount:" << inputSampleCount);
    
    if (nullptr == resampler)
        return Result::fail("No resampler object");
    
    //
    // Clear the resample and pump some initial zeros into it
    //
    resampler->clear();
    
#if 0
    int preloadSamples = resampler->getInLenBeforeOutStart(MAX_RESAMPLER_INPUT_SAMPLES);
    inputBlockBuffer.clear(MAX_RESAMPLER_INPUT_SAMPLES);
    while (preloadSamples > 0)
    {
        int count = jmin((int)MAX_RESAMPLER_INPUT_SAMPLES, preloadSamples);
        double* outputBlock = nullptr;
        resampler->process(inputBlockBuffer, count, outputBlock);
        
        preloadSamples -= count;
    }
#endif
    
    //
    // Flush the output buffer
    //
    outputBuffer_.clear();
    
    //
    // Do the actual upsample
    //
    outputSampleCount_ = 0;
    
    int inputSampleCount = inputBuffer.getNumSamples();
    const float * source = inputBuffer.getReadPointer(inputChannel);
    while (inputSampleCount > 0)
    {
        //
        // Convert float to double
        //
        int inputConvertCount = jmin(inputSampleCount, (int)MAX_RESAMPLER_INPUT_SAMPLES);
        for (int i = 0; i < inputConvertCount; ++i)
        {
            inputBlockBuffer[i] = *source;
            source++;
        }
        inputSampleCount -= inputConvertCount;
        
        //
        // Run the SRC
        //
        double* outputBlock = nullptr;
        
        int outputBlockSampleCount = resampler->process(inputBlockBuffer, inputConvertCount, outputBlock);
        int outputSpaceRemaining = outputBuffer_.getNumSamples() - outputSampleCount_;
        int outputCopyCount = jmin( outputSpaceRemaining, outputBlockSampleCount);
        float *destination = outputBuffer_.getWritePointer(outputChannel, outputSampleCount_);
        
        for (int i = 0; i < outputCopyCount; ++i)
        {
            *destination = (float)outputBlock[i];
            destination++;
        }
        
        outputSampleCount_ += outputCopyCount;
    }
    
    //
    // Keep filling the output buffer
    //
    inputBlockBuffer.clear(MAX_RESAMPLER_INPUT_SAMPLES);
    
    while (outputSampleCount_ < outputBuffer_.getNumSamples())
    {
        //
        // Run the SRC
        //
        double* outputBlock = nullptr;
        
        int outputBlockSampleCount = resampler->process(inputBlockBuffer, MAX_RESAMPLER_INPUT_SAMPLES, outputBlock);
        int outputSpaceRemaining = outputBuffer_.getNumSamples() - outputSampleCount_;
        int outputCopyCount = jmin( outputSpaceRemaining, outputBlockSampleCount);
        float *destination = outputBuffer_.getWritePointer(outputChannel, outputSampleCount_);
        
        for (int i = 0; i < outputCopyCount; ++i)
        {
            *destination = (float)outputBlock[i];
            destination++;
        }
        
        outputSampleCount_ += outputCopyCount;
    }
    
//.........这里部分代码省略.........