// No audio is rendered in this method.
// This code is only here to get updates synchronized in audio frame time.
private void OnAudioFilterRead(float[] data, int channels)
{
bool playNote = (audioFrameCount % speed) == 0;
if (playNote)
{
int note = sequence[sequenceIdx++];
sequenceIdx %= sequence.Length;
if (note != -1)
{
note += pitch;
synth.HandleEventNow(new EventQueue.QueuedEvent(EventQueue.EventType.Note_off, 0, -1));
synth.HandleEventNow(new EventQueue.QueuedEvent(EventQueue.EventType.Note_on, note, -1));
}
}
if (lfo1enabled)
{
lfo1.set_freq(lfo1Freq * 1024);
lfo1.update();
synth.set_parameter(lfo1Param, lfo1BaseValue + lfo1.sin() * lfo1Strength);
}
if (lfo2enabled)
{
lfo2.set_freq(lfo2Freq * 1024);
lfo2.update();
synth.set_parameter(lfo2Param, lfo2BaseValue + lfo2.sin() * lfo2Strength);
}
audioFrameCount++;
}
private void render_float32_stereo_interleaved(float[] buffer, int sample_frames)
{
int smp = 0;
int buf_idx = 0;
//int time_smp = masterClock_smp;
update_params();
// Cache this for the entire buffer, we don't need to check for
// every sample if new events have been enqueued.
// This assumes that no other metdods call GetFrontAndDequeue.
int queueSize = queue.GetSize();
// Render loop
for (; smp < sample_frames; ++smp)
{
// Event handling
// This is sample accurate event handling.
// If it's too slow, we can decide to only handle 1 event per buffer and
// move this code outside the loop.
while (true)
{
if (eventIsWaiting == false && queueSize > 0)
{
//queueLock = true;
if (queue.GetFrontAndDequeue(ref nextEvent))
{
eventIsWaiting = true;
queueSize--;
}
//queueLock = false;
}
if (eventIsWaiting)
{
if (nextEvent.time_smp <= time_smp)
{
HandleEventNow(nextEvent);
eventIsWaiting = false;
}
else
{
// we assume that queued events are in order, so if it's not
// now, we stop getting events from the queue
break;
}
}
else
{
// no more events
break;
}
}
// Rendering
if (note_is_on)
{
// Render sample
float amp = aenv.process() * 0.5f;
float lfo_val = lfo.sin() * 0.48f * pwmStrength + 0.5f;
//float saw = osc1.saw() * sawAmp;
//float square = osc1.square(lfo_val) * squareAmp;
//float sawDPW = osc1.sawDPW() * sawDPWAmp;
float sine = osc2.sin() * subSine;
float sawPolyBLEP = osc1.sawPolyBLEP() * sawAmp;
float squarePolyBLEP = osc1.squarePolyBLEP(lfo_val) * squareAmp;
float sample = (sine + sawPolyBLEP + squarePolyBLEP)
* /*(current_velocity * 0.0079f) **/ amp;
buffer[buf_idx++] = sample;
buffer[buf_idx++] = sample;
// Update oscillators
osc1.update();
osc2.update();
lfo.update();
fenv.update_oneshot();
}
else
{
buffer[buf_idx++] = 0.0f;
buffer[buf_idx++] = 0.0f;
}
time_smp++;
}
// Filter entire buffer
if (filterEnabled >= 0.5f)
{
if (filterType == FilterType.Schmid)
{
filter1.process_mono_stride(buffer, sample_frames, 0, 2);
filter2.process_mono_stride(buffer, sample_frames, 1, 2);
}
#if LAZZARINI_FILTER
else if (filterType == FilterType.Lazzarini)
{
filter1Laz.process_mono_stride(buffer, sample_frames, 0, 2);
filter2Laz.process_mono_stride(buffer, sample_frames, 1, 2);
}
#endif
}
}