I’m writing a granular synthesizer. I originally partitioned multiple AudioBuffers, each of which corresponded to a grain. For performance reasons (I don’t want to have to re-partition every time grain size changes) and code clarity, I’d rather just store a start and end index in to one large AudioBuffer. Now, instead of storing a per-partitioned AudioBuffer of let’s say 10ms, my SynthesiserVoice subclass, sees one large buffer but I’m unable to restrict playback to a shorter range. What is unclear to me is why I can’t hardcode startSample and numSamples to be a 0 and 441 or something for let’s say a 10ms grain. Whenever I do this, it just plays the full buffer, so I think I’m missing something fundamental. This code is more or less copied from the SamplerVoice, code and is just called in standard fashion like:
void GranularAudioSource::getNextAudioBlock(const AudioSourceChannelInfo& bufferToFill)
{
...
synth.renderNextBlock(*bufferToFill.buffer, incomingMidi,
bufferToFill.startSample, bufferToFill.numSamples);
}
I think the change I need must be in the following code:
void GranularVoice::renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples)
{
if (auto* playingSound = static_cast<GranularSound*> (getCurrentlyPlayingSound().get()))
{
// grainData is an AudioBuffer<float>&
auto& data = playingSound->grainData;
const float* const inL = data.getReadPointer (0);
const float* const inR = data.getNumChannels() > 1 ? data.getReadPointer (1) : nullptr;
// Are we in stereo? If yes then set outR to the second write buffer pointer of data,
// else set outR to nullptr.
float* outL = outputBuffer.getWritePointer (0, startSample);
float* outR = outputBuffer.getNumChannels() > 1 ? outputBuffer.getWritePointer (1, startSample) : nullptr;
// We will fill the buffer with numSamples samples,
// but if the number of samples in a grain is less than numSamples,
// we'll need to loop through the grain some number of times to fill the buffer.
while (--numSamples >= 0)
{
auto pos = (int) sourceSamplePosition;
auto alpha = (float) (sourceSamplePosition - pos);
auto invAlpha = 1.0f - alpha;
// Using a very simple linear interpolation here.
float l = (inL[pos] * invAlpha + inL[pos + 1] * alpha);
// Again, only process right channel if we're in stereo.
float r = (inR != nullptr) ? (inR[pos] * invAlpha + inR[pos + 1] * alpha)
: l;
auto envelopeValue = adsr.getNextSample();
l *= lgain * envelopeValue;
r *= rgain * envelopeValue;
if (outR != nullptr)
{
*outL++ += l;
*outR++ += r;
}
else
{
*outL++ += (l + r) * 0.5f;
}
sourceSamplePosition += pitchRatio;
if (sourceSamplePosition > playingSound->length)
{
// stopNote (0.0f, true);
// break;
sourceSamplePosition = 0;
}
}
}
}