MyApp(){
lpf.type(LOW_PASS); // Set filter to low-pass response
lpf.res(4); // Set resonance amount to emphasize filter
env.attack(0.01); // Set short (10 ms) attack
env.decay(0.4); // Set longer (400 ms) decay
tmr.freq(120./60.*4.); // Set timer frequency to 120 BPM
tmr.phaseMax(); // Ensures timer triggers on first sample
modCutoff.period(30); // Set period of cutoff modulation
modCutoff.phase(0.5); // Start half-way through cycle
freq.lag(0.1); // Lag time of portamento effect
step=0;
}
void Cascade::setLayout (const LayoutBase& proto)
{
const int numPoles = proto.getNumPoles();
m_numStages = (numPoles + 1)/ 2;
assert (m_numStages <= m_maxStages);
Biquad* stage = m_stageArray;
for (int i = 0; i < m_numStages; ++i, ++stage)
stage->setPoleZeroPair (proto[i]);
applyScale (proto.getNormalGain() /
std::abs (response (proto.getNormalW() / (2 * doublePi))));
}
void Cascade::setLayout (const LayoutBase& proto)
{
const int numPoles = proto.getNumPoles();
m_numStages = (numPoles + 1)/ 2;
if (m_numStages > m_maxStages)
throw std::invalid_argument("Number of stages is larger than the max stages.");
Biquad* stage = m_stageArray;
for (int i = 0; i < m_numStages; ++i, ++stage)
stage->setPoleZeroPair (proto[i]);
applyScale (proto.getNormalGain() /
std::abs (response (proto.getNormalW() / (2 * doublePi))));
}
void onAudio(AudioIOData& io){
while(io()){
if(tmr()){
// Our sequence of pitches
float pitches[] = {0,0,12,0,0,10,-5,0};
// Map pitch class to a frequency in Hz
float f = 55 * pow(2, pitches[step]/12.);
// Increment step counter
step = (step + 1) % 8;
// Set new target frequence of portamento
freq = f;
// Restart envelope using a soft reset (to avoid clicks)
env.resetSoft();
}
// Set saw frequency from portamento filter
saw.freq(freq());
// Get next envelope value
float e = env();
// Map envelope value to cutoff frequency
lpf.freq(e * (modCutoff.paraU()*6000 + 500) + 40);
// Generate next saw sample
float s = saw() * 0.3;
// Filter saw sample
s = lpf(s) * e;
// Send sample to DAC
io.out(0) = io.out(1) = s;
}
}
int main()
{
tmr.period(3); // Reset the envelope every 3 seconds
env.attack(0.01); // Attack time
env.decay(0.05); // Decay time
// Delays, lowpass, and allpass filters setup code goes here
//
lpf.type(LOW_PASS);
lpf.freq(2600);
lpf.res(sqrt(2.0)/2.0);
AudioIO audioIO(frameCount, samplingRate, audioCallBack, NULL, channelsOut, channelsIn);
Sync::master().spu(audioIO.framesPerSecond());
audioIO.start();
printf("Press 'enter' to quit...\n");
getchar();
return 0;
}
HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate, bool bassBoost)
: m_frameDelay(0)
, m_sampleRate(sampleRate)
{
ASSERT(channel);
// Determine the leading delay (average group delay) for the response.
m_frameDelay = extractAverageGroupDelay(channel, fftSize / 2);
float* impulseResponse = channel->data();
size_t responseLength = channel->length();
if (bassBoost) {
// Run through some post-processing to boost the bass a little -- the HRTF's seem to be a little bass-deficient.
// FIXME: this post-processing should have already been applied to the HRTF file resources. Once the files are put into this form,
// then this code path can be removed along with the bassBoost parameter.
Biquad filter;
filter.setLowShelfParams(700.0 / nyquist(), 6.0); // boost 6dB at 700Hz
filter.process(impulseResponse, impulseResponse, responseLength);
}
// We need to truncate to fit into 1/2 the FFT size (with zero padding) in order to do proper convolution.
size_t truncatedResponseLength = min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT
// Quick fade-out (apply window) at truncation point
unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate
ASSERT(numberOfFadeOutFrames < truncatedResponseLength);
if (numberOfFadeOutFrames < truncatedResponseLength) {
for (unsigned i = truncatedResponseLength - numberOfFadeOutFrames; i < truncatedResponseLength; ++i) {
float x = 1.0f - static_cast<float>(i - (truncatedResponseLength - numberOfFadeOutFrames)) / numberOfFadeOutFrames;
impulseResponse[i] *= x;
}
}
m_fftFrame = adoptPtr(new FFTFrame(fftSize));
m_fftFrame->doPaddedFFT(impulseResponse, truncatedResponseLength);
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
switch(cnt){
case 0: filt.type(Filter::LP); printf("Low-pass\n"); break;
case 1: filt.type(Filter::HP); printf("High-pass\n"); break;
case 2: filt.type(Filter::BP); printf("Band-pass\n"); break;
case 3: filt.type(Filter::BR); printf("Band-reject\n"); break;
}
++cnt %= 4;
}
float cutoff = scl::pow3(mod.triU()) * 10000;
filt.res(4);
filt.freq(cutoff);
float s = filt(src());
io.out(0) = io.out(1) = s * 0.1f;
}
}
