void audioOutputCallbackFn(const AudioTimeStamp *inTimestamp) {
if (sys_userCallbackFn) {
(*sys_userCallbackFn)(inTimestamp);
}
sched_audio_callbackfn();
}
void ofxPd::renderAudio( float * input, float* output, int bufferSize, int nChannels ){
// adapted from AudioOutputController by bryan summerset
// sys_schedblocksize is How many frames we have per PD dsp_tick
// inNumberFrames is how many frames have been requested
int inNumberFrames = bufferSize;
// Make sure we're dealing with evenly divisible numbers between
// the number of frames requested vs the block size for a given PD dsp tick.
//Otherwise this looping scheme we're doing below doesn't make much sense.
assert(inNumberFrames % sys_schedblocksize == 0);
// How many times to generate a DSP event in PD
int times = inNumberFrames / sys_schedblocksize;
for(int i = 0; i < times; i++) {
for (int n = 0; n < sys_schedblocksize; n++) {
for(int ch = 0; ch < sys_ninchannels; ch++) {
sys_soundin[n+sys_schedblocksize*ch] = input[(n*sys_ninchannels+ch) + // offset in iteration
i*sys_schedblocksize*sys_ninchannels]; // iteration starting address
}
}
// do one Pd DSP block
sys_lock();
sched_audio_callbackfn();
sys_unlock();
// This should cover sys_noutchannels channels. Turns non-interleaved into
// interleaved audio.
for (int n = 0; n < sys_schedblocksize; n++) {
for(int ch = 0; ch < sys_noutchannels; ch++) {
t_sample fsample = CLAMP(sys_soundout[n+sys_schedblocksize*ch],-1,1);
output[(n*sys_noutchannels+ch) + // offset in iteration
i*sys_schedblocksize*sys_noutchannels] // iteration starting address
= fsample;
//sys_soundin[n+sys_schedblocksize*ch] = ofRandom(-0.5,0.5); // iteration starting address
}
}
// After loading the samples, we need to clear them for the next iteration
memset(sys_soundout, 0, sizeof(t_sample)*sys_noutchannels*sys_schedblocksize);
}
}
