void ofxFFT::calc() {
//Generate a split complex vector from the real data
vDSP_ctoz((COMPLEX *)input, 2, &mDspSplitComplex, 1, mFFTLength);
//Take the fft and scale appropriately
vDSP_fft_zrip(mSpectrumAnalysis, &mDspSplitComplex, 1, log2n, FFT_FORWARD);
// vDSP_fft_zrip(mSpectrumAnalysis, &mDspSplitComplex, 1, log2n, FFT_INVERSE);
vDSP_vsmul(mDspSplitComplex.realp, 1, &mFFTNormFactor, mDspSplitComplex.realp, 1, mFFTLength);
vDSP_vsmul(mDspSplitComplex.imagp, 1, &mFFTNormFactor, mDspSplitComplex.imagp, 1, mFFTLength);
// /* The output signal is now in a split real form. Use the function
// * vDSP_ztoc to get a split real vector. */
// vDSP_ztoc(&mDspSplitComplex, 1, (COMPLEX *) output, 2, mFFTLength);
//
//Zero out the nyquist value
mDspSplitComplex.imagp[0] = 0.0;
//Convert the fft data to dB
vDSP_zvmags(&mDspSplitComplex, 1, amplitude, 1, mFFTLength);
//In order to avoid taking log10 of zero, an adjusting factor is added in to make the minimum value equal -128dB
float mAdjust0DB = ADJUST_0_DB;
vDSP_vsadd(amplitude, 1, &mAdjust0DB, power, 1, mFFTLength);
float one = 1;
vDSP_vdbcon(power, 1, &one, power, 1, mFFTLength, 0);
}
void FFTHelper::ComputeFFT(Float32* inAudioData, Float32* outFFTData)
{
if (inAudioData == NULL || outFFTData == NULL) return;
//Generate a split complex vector from the real data
vDSP_ctoz((COMPLEX *)inAudioData, 2, &mDspSplitComplex, 1, mFFTLength);
//Take the fft and scale appropriately
vDSP_fft_zrip(mSpectrumAnalysis, &mDspSplitComplex, 1, mLog2N, kFFTDirection_Forward);
vDSP_vsmul(mDspSplitComplex.realp, 1, &mFFTNormFactor, mDspSplitComplex.realp, 1, mFFTLength);
vDSP_vsmul(mDspSplitComplex.imagp, 1, &mFFTNormFactor, mDspSplitComplex.imagp, 1, mFFTLength);
//Zero out the nyquist value
mDspSplitComplex.imagp[0] = 0.0;
//Convert the fft data to dB
vDSP_zvmags(&mDspSplitComplex, 1, outFFTData, 1, mFFTLength);
//In order to avoid taking log10 of zero, an adjusting factor is added in to make the minimum value equal -128dB
vDSP_vsadd(outFFTData, 1, &kAdjust0DB, outFFTData, 1, mFFTLength);
Float32 one = 1;
vDSP_vdbcon(outFFTData, 1, &one, outFFTData, 1, mFFTLength, 0);
Float32 max = -100;
int index = -1;
for(unsigned long i = 0; i < mFFTLength; i++){
if(outFFTData[i] > max){
max = outFFTData[i];
index = i;
}
}
if(max > -40){ // Filter out anything else, as it is unlikely to be the microwave beep
recentMaxIndex = index;
//if(index == 181){ // We found the microwave beep
//printf("%d %f\n", index, max);
}else{
recentMaxIndex = 0;
}
}
void FFTHelper::ComputeFFT(Float32* inAudioData, Float32* outFFTData)
{
if (inAudioData == NULL || outFFTData == NULL) return;
//Generate a split complex vector from the real data
vDSP_ctoz((COMPLEX *)inAudioData, 2, &mDspSplitComplex, 1, mFFTLength);
//Take the fft and scale appropriately
vDSP_fft_zrip(mSpectrumAnalysis, &mDspSplitComplex, 1, mLog2N, kFFTDirection_Forward);
vDSP_vsmul(mDspSplitComplex.realp, 1, &mFFTNormFactor, mDspSplitComplex.realp, 1, mFFTLength);
vDSP_vsmul(mDspSplitComplex.imagp, 1, &mFFTNormFactor, mDspSplitComplex.imagp, 1, mFFTLength);
//Zero out the nyquist value
mDspSplitComplex.imagp[0] = 0.0;
//Convert the fft data to dB
vDSP_zvmags(&mDspSplitComplex, 1, outFFTData, 1, mFFTLength);
//In order to avoid taking log10 of zero, an adjusting factor is added in to make the minimum value equal -128dB
vDSP_vsadd(outFFTData, 1, &kAdjust0DB, outFFTData, 1, mFFTLength);
Float32 one = 1;
vDSP_vdbcon(outFFTData, 1, &one, outFFTData, 1, mFFTLength, 0);
}
Boolean FFTBufferManager::ComputeFFT(int32_t *outFFTData)
{
if (HasNewAudioData())
{
// for(int i=0;i<mFFTLength;i++)
// {
// if(mAudioBuffer[i]>0.15)
// printf("%f\n",mAudioBuffer[i]);
// }
//Generate a split complex vector from the real data
# define NOISE_FILTER 0.01;
for(int i=0;i<4096;i++)
{
//DENOISE
if(mAudioBuffer[i]>0)
{
mAudioBuffer[i] -= NOISE_FILTER;
if(mAudioBuffer[i] < 0)
{
mAudioBuffer[i] = 0;
}
}
else if(mAudioBuffer[i]<0)
{
mAudioBuffer[i] += NOISE_FILTER;
if(mAudioBuffer[i]>0)
{
mAudioBuffer[i] = 0;
}
}
else
{
mAudioBuffer[i] = 0;
}
}
vDSP_ctoz((COMPLEX *)mAudioBuffer, 2, &mDspSplitComplex, 1, mFFTLength);
//Take the fft and scale appropriately
vDSP_fft_zrip(mSpectrumAnalysis, &mDspSplitComplex, 1, mLog2N, kFFTDirection_Forward);
vDSP_vsmul(mDspSplitComplex.realp, 1, &mFFTNormFactor, mDspSplitComplex.realp, 1, mFFTLength);
vDSP_vsmul(mDspSplitComplex.imagp, 1, &mFFTNormFactor, mDspSplitComplex.imagp, 1, mFFTLength);
//Zero out the nyquist value
mDspSplitComplex.imagp[0] = 0.0;
//Convert the fft data to dB
Float32 tmpData[mFFTLength];
vDSP_zvmags(&mDspSplitComplex, 1, tmpData, 1, mFFTLength);
//In order to avoid taking log10 of zero, an adjusting factor is added in to make the minimum value equal -128dB
vDSP_vsadd(tmpData, 1, &mAdjust0DB, tmpData, 1, mFFTLength);
Float32 one = 1;
vDSP_vdbcon(tmpData, 1, &one, tmpData, 1, mFFTLength, 0);
//Convert floating point data to integer (Q7.24)
vDSP_vsmul(tmpData, 1, &m24BitFracScale, tmpData, 1, mFFTLength);
for(UInt32 i=0; i<mFFTLength; ++i)
outFFTData[i] = (SInt32) tmpData[i];
for(int i=0;i<mFFTLength/4;i++)
{
printf("%i:::%i\n",i,outFFTData[i]/16777216+120);
}
OSAtomicDecrement32Barrier(&mHasAudioData);
OSAtomicIncrement32Barrier(&mNeedsAudioData);
mAudioBufferCurrentIndex = 0;
return true;
}
else if (mNeedsAudioData == 0)
OSAtomicIncrement32Barrier(&mNeedsAudioData);
return false;
}
bool ofxAudioUnitFftNode::getAmplitude(std::vector<float> &outAmplitude)
{
getSamplesFromChannel(_sampleBuffer, 0);
// return empty if we don't have enough samples yet
if(_sampleBuffer.size() < _N) {
outAmplitude.clear();
return false;
}
// normalize input waveform
if(_outputSettings.normalizeInput) {
float timeDomainMax;
vDSP_maxv(&_sampleBuffer[0], 1, &timeDomainMax, _N);
vDSP_vsdiv(&_sampleBuffer[0], 1, &timeDomainMax, &_sampleBuffer[0], 1, _N);
}
PerformFFT(&_sampleBuffer[0], _window, _fftData, _fftSetup, _N);
// get amplitude
vDSP_zvmags(&_fftData, 1, _fftData.realp, 1, _N/2);
// normalize magnitudes
float two = 2.0;
vDSP_vsdiv(_fftData.realp, 1, &two, _fftData.realp, 1, _N/2);
// scale output according to requested settings
if(_outputSettings.scale == OFXAU_SCALE_LOG10) {
for(int i = 0; i < (_N / 2); i++) {
_fftData.realp[i] = log10f(_fftData.realp[i] + 1);
}
} else if(_outputSettings.scale == OFXAU_SCALE_DECIBEL) {
float ref = 1.0;
vDSP_vdbcon(_fftData.realp, 1, &ref, _fftData.realp, 1, _N / 2, 1);
float dbCorrectionFactor = 0;
switch (_outputSettings.window) {
case OFXAU_WINDOW_HAMMING:
dbCorrectionFactor = DB_CORRECTION_HAMMING;
break;
case OFXAU_WINDOW_HANNING:
dbCorrectionFactor = DB_CORRECTION_HAMMING;
break;
case OFXAU_WINDOW_BLACKMAN:
dbCorrectionFactor = DB_CORRECTION_HAMMING;
break;
}
vDSP_vsadd(_fftData.realp, 1, &dbCorrectionFactor, _fftData.realp, 1, _N / 2);
}
// restrict minimum to 0
if(_outputSettings.clampMinToZero) {
float min = 0.0;
float max = INFINITY;
vDSP_vclip(_fftData.realp, 1, &min, &max, _fftData.realp, 1, _N / 2);
}
// normalize output between 0 and 1
if(_outputSettings.normalizeOutput) {
float max;
vDSP_maxv(_fftData.realp, 1, &max, _N / 2);
if(max > 0) {
vDSP_vsdiv(_fftData.realp, 1, &max, _fftData.realp, 1, _N / 2);
}
}
outAmplitude.assign(_fftData.realp, _fftData.realp + _N/2);
return true;
}
