I have an app that selects a song from the iPod Library then copies that song into the app's directory as a '.caf' file. I now need to play and at the same time read that file into Apples FFT from the Accelerate framework so I can visualize the data like a spectrogram. Here is the code for the FFT:
void FFTAccelerate::doFFTReal(float samples[], float amp[], int numSamples)
{
int i;
vDSP_Length log2n = log2f(numSamples);
//Convert float array of reals samples to COMPLEX_SPLIT array A
vDSP_ctoz((COMPLEX*)samples,2,&A,1,numSamples/2);
//Perform FFT using fftSetup and A
//Results are returned in A
vDSP_fft_zrip(fftSetup, &A, 1, log2n, FFT_FORWARD);
//Convert COMPLEX_SPLIT A result to float array to be returned
amp[0] = A.realp[0]/(numSamples*2);
for(i=1;i<numSamples;i++)
amp[i]=sqrt(A.realp[i]*A.realp[i]+A.imagp[i]*A.imagp[i])/numSamples;
}
//Constructor
FFTAccelerate::FFTAccelerate (int numSamples)
{
vDSP_Length log2n = log2f(numSamples);
fftSetup = vDSP_create_fftsetup(log2n, FFT_RADIX2);
int nOver2 = numSamples/2;
A.realp = (float *) malloc(nOver2*sizeof(float));
A.imagp = (float *) malloc(nOver2*sizeof(float));
}
My question is how to I loop through the '.caf' audio file to feed the FFT while at the same time playing the song? I only need one channel. Im guessing I need to get 1024 samples of the song, process that in the FTT and then move further down the file and grab another 1024 samples. But I dont understand how to read an audio file to do this. The file has a sample rate of 44100.0 hz, is in linear PCM format, 16 Bit and I believe is also interleaved if that helps...