void renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples)
{
if (angleDelta != 0.0)
{
if (tailOff > 0)
{
SynthesiserSound *sound = getCurrentlyPlayingSound();
while (--numSamples >= 0)
{
float currentSample;
// LFO used to create tremolo
const float lfoCurrentSample = (float) ((4.0 * sin (lfoCurrentAngle)) * level * tailOff) + 0.7;
currentSample = calculateSample(sound, currentSample, lfoCurrentSample, true);
for (int i = outputBuffer.getNumChannels(); --i >= 0;)
*outputBuffer.getSampleData (i, startSample) += (currentSample);
currentAngle += angleDelta;
lfoCurrentAngle += lfoAngleDelta;
++startSample;
tailOff *= 0.99;
if (tailOff <= 0.005)
{
clearCurrentNote();
angleDelta = 0.0;
lfoAngleDelta = 0.0;
break;
}
}
}
else
{
SynthesiserSound *sound = getCurrentlyPlayingSound();
while (--numSamples >= 0)
{
float currentSample;
// LFO used to create tremolo
const float lfoCurrentSample = (float) ((4.0 * sin (lfoCurrentAngle)) * level) + 0.7;
currentSample = calculateSample(sound, currentSample, lfoCurrentSample, false);
for (int i = outputBuffer.getNumChannels(); --i >= 0;)
*outputBuffer.getSampleData (i, startSample) += (currentSample);
currentAngle += angleDelta;
lfoCurrentAngle += lfoAngleDelta;
++startSample;
}
}
}
}
//==============================================================================
void SamplerVoice::renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples)
{
if (const SamplerSound* const playingSound = static_cast <SamplerSound*> (getCurrentlyPlayingSound().get()))
{
const float* const inL = playingSound->data->getSampleData (0, 0);
const float* const inR = playingSound->data->getNumChannels() > 1
? playingSound->data->getSampleData (1, 0) : nullptr;
float* outL = outputBuffer.getSampleData (0, startSample);
float* outR = outputBuffer.getNumChannels() > 1 ? outputBuffer.getSampleData (1, startSample) : nullptr;
while (--numSamples >= 0)
{
const int pos = (int) sourceSamplePosition;
const float alpha = (float) (sourceSamplePosition - pos);
const float invAlpha = 1.0f - alpha;
// just using a very simple linear interpolation here..
float l = (inL [pos] * invAlpha + inL [pos + 1] * alpha);
float r = (inR != nullptr) ? (inR [pos] * invAlpha + inR [pos + 1] * alpha)
: l;
l *= lgain;
r *= rgain;
if (isInAttack)
{
l *= attackReleaseLevel;
r *= attackReleaseLevel;
attackReleaseLevel += attackDelta;
if (attackReleaseLevel >= 1.0f)
{
attackReleaseLevel = 1.0f;
isInAttack = false;
}
}
else if (isInRelease)
{
l *= attackReleaseLevel;
r *= attackReleaseLevel;
attackReleaseLevel += releaseDelta;
if (attackReleaseLevel <= 0.0f)
{
stopNote (false);
break;
}
}
if (outR != nullptr)
{
*outL++ += l;
*outR++ += r;
}
else
{
*outL++ += (l + r) * 0.5f;
}
sourceSamplePosition += pitchRatio;
if (sourceSamplePosition > playingSound->length)
{
stopNote (false);
break;
}
}
}
}
