加速フレームワークでケプストラム分析のピーク値を見つけようとしています。フレームの最後または最初に常にピーク値を取得します。マイクから音声をリアルタイムで取得して分析しています。この私のコードの何が問題になっていますか? 私のコードは以下です：
OSStatus microphoneInputCallback (void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData){
// get reference of test app we need for test app attributes
TestApp *this = (TestApp *)inRefCon;
COMPLEX_SPLIT complexArray = this->fftA;
void *dataBuffer = this->dataBuffer;
float *outputBuffer = this->outputBuffer;
FFTSetup fftSetup = this->fftSetup;
uint32_t log2n = this->fftLog2n;
uint32_t n = this->fftN; // 4096
uint32_t nOver2 = this->fftNOver2;
uint32_t stride = 1;
int bufferCapacity = this->fftBufferCapacity; // 4096
SInt16 index = this->fftIndex;
OSStatus renderErr;
// observation objects
float *observerBufferRef = this->observerBuffer;
int observationCountRef = this->observationCount;
renderErr = AudioUnitRender(rioUnit, ioActionFlags,
inTimeStamp, bus1, inNumberFrames, this->bufferList);
if (renderErr < 0) {
return renderErr;
}
// Fill the buffer with our sampled data. If we fill our buffer, run the
// fft.
int read = bufferCapacity - index;
if (read > inNumberFrames) {
memcpy((SInt16 *)dataBuffer + index, this->bufferList->mBuffers[0].mData, inNumberFrames*sizeof(SInt16));
this->fftIndex += inNumberFrames;
} else {
// If we enter this conditional, our buffer will be filled and we should PERFORM FFT.
memcpy((SInt16 *)dataBuffer + index, this->bufferList->mBuffers[0].mData, read*sizeof(SInt16));
// Reset the index.
this->fftIndex = 0;
/*************** FFT ***************/
//multiply by window
vDSP_vmul((SInt16 *)dataBuffer, 1, this->window, 1, this->outputBuffer, 1, n);
// We want to deal with only floating point values here.
vDSP_vflt16((SInt16 *) dataBuffer, stride, (float *) outputBuffer, stride, bufferCapacity );
/**
Look at the real signal as an interleaved complex vector by casting it.
Then call the transformation function vDSP_ctoz to get a split complex
vector, which for a real signal, divides into an even-odd configuration.
*/
vDSP_ctoz((COMPLEX*)outputBuffer, 2, &complexArray, 1, nOver2);
// Carry out a Forward FFT transform.
vDSP_fft_zrip(fftSetup, &complexArray, stride, log2n, FFT_FORWARD);
vDSP_ztoc(&complexArray, 1, (COMPLEX *)outputBuffer, 2, nOver2);
complexArray.imagp[0] = 0.0f;
vDSP_zvmags(&complexArray, 1, complexArray.realp, 1, nOver2);
bzero(complexArray.imagp, (nOver2) * sizeof(float));
// scale
float scale = 1.0f / (2.0f*(float)n);
vDSP_vsmul(complexArray.realp, 1, &scale, complexArray.realp, 1, nOver2);
// step 2 get log for cepstrum
float *logmag = malloc(sizeof(float)*nOver2);
for (int i=0; i < nOver2; i++)
logmag[i] = logf(sqrtf(complexArray.realp[i]));
// configure float array into acceptable input array format (interleaved)
vDSP_ctoz((COMPLEX*)logmag, 2, &complexArray, 1, nOver2);
// create cepstrum
vDSP_fft_zrip(fftSetup, &complexArray, stride, log2n-1, FFT_INVERSE);
//convert interleaved to real
float *displayData = malloc(sizeof(float)*n);
vDSP_ztoc(&complexArray, 1, (COMPLEX*)displayData, 2, nOver2);
float dominantFrequency = 0;
int currentBin = 0;
float dominantFrequencyAmp = 0;
// find peak of cepstrum
for (int i=0; i < nOver2; i++){
//get current frequency magnitude
if (displayData[i] > dominantFrequencyAmp) {
// DLog("Bufferer filled %f", displayData[i]);
dominantFrequencyAmp = displayData[i];
currentBin = i;
}
}
DLog("currentBin : %i amplitude: %f", currentBin, dominantFrequencyAmp);
}
return noErr;
}
