Getting the amplitude of two frequencies over time - ios

Question

I am using this GitHub project called "The Amazing Audio Engine", to capture audio from the microphone. So I am using this method:
id<AEAudioReceiver> receiver = [AEBlockAudioReceiver audioReceiverWithBlock: ^(void *source, const AudioTimeStamp *time, UInt32 frames, AudioBufferList *audio) {
// Do something with 'audio'
}];
This method fires every 23 ms delivering an audio array containing all amplitudes of the sound wave over that 23 ms interval.
This is the catch. This audio sound I am dealing with is a FM signal, composed of two frequencies, one at 1000 Hz and one at twice the frequency that represents zeros and ones of a digital stream.
This is my problem. At that point I have an array of audio amplitudes over 0.23 ms.
So I thought I could do a FFT to convert the signal into frequency levels. I used this code:
// Setup the length
vDSP_Length log2n = log2f(numFrames);
// Calculate the weights array. This is a one-off operation.
FFTSetup fftSetup = vDSP_create_fftsetup(log2n, FFT_RADIX2);
// For an FFT, numSamples must be a power of 2, i.e. is always even
int nOver2 = numFrames/2;
// Populate *window with the values for a hamming window function
float *window = (float *)malloc(sizeof(float) * numFrames);
vDSP_hamm_window(window, numFrames, 0);
// Window the samples
vDSP_vmul(data, 1, window, 1, data, 1, numFrames);
// Define complex buffer
COMPLEX_SPLIT A;
A.realp = (float *) malloc(nOver2*sizeof(float));
A.imagp = (float *) malloc(nOver2*sizeof(float));
// Pack samples:
// C(re) -> A[n], C(im) -> A[n+1]
vDSP_ctoz((COMPLEX*)data, 2, &A, 1, numFrames/2);
// RUN THE FFT
//Perform a forward FFT using fftSetup and A
//Results are returned in A
vDSP_fft_zrip(fftSetup, &A, 1, log2n, FFT_FORWARD);
Because each interval is 172 Hz and I want to isolate 1000Hz, I think the 6th "bucket" of the FFT result would be the one, so I have this code:
//Convert COMPLEX_SPLIT A result to magnitudes
float amp[numFrames];
amp[0] = A.realp[0]/(numFrames*2);
for(int i=1; i<numFrames; i++) {
amp[i]=A.realp[i]*A.realp[i]+A.imagp[i]*A.imagp[i];
}
// I need the 6th and the 12th bucket, so I need a[5] and a[11]
but then I am starting to think that the FFT is not what I want because a[5] and a[11] will give me the amplitudes of ~1000Hz and ~2000Hz over 0.23 ms but in fact what I need are all the variations of the 1000 Hz and 2000 Hz sounds had over the 0.23ms time. In fact I need to obtain arrays, not single values.
In broad lines what should I do to obtain the amplitudes over time of the two frequencies, 1000 and 2000 Hz?

Answer 1

If you know what time resolution you want, two Goertzel filters slid by that length would allow you to measure the amplitudes of your two frequencies with much less overhead than using FFTs. The length of the filter or FFT need not (usually should not) be the same length as the number of frames from each audio callback. You can use a circular buffer or fifo to decouple the lengths. (In iOS, the numFrames can be different on different device models, and may suddenly change depending on other factors outside the apps control).