FFT on the audio buffer in the AURenderCallback method:

FFT.m

static void performFFT(float* data, UInt32 numberOfFrames, SoundBufferPtr soundBuffer, UInt32 inBusNumber) {

    int bufferLog2 = round(log2(numberOfFrames));
    float fftNormFactor = 1.0/( 2 * numberOfFrames);
    
    FFTSetup fftSetup = vDSP_create_fftsetup(bufferLog2, kFFTRadix2);
    
    int numberOfFramesOver2 = numberOfFrames / 2;
    float outReal[numberOfFramesOver2];
    float outImaginary[numberOfFramesOver2];
    
    COMPLEX_SPLIT output = { .realp = outReal, .imagp = outImaginary };
    
    //Put all of the even numbered elements into outReal and odd numbered into outImaginary
    vDSP_ctoz((COMPLEX *)data, 2, &output, 1, numberOfFramesOver2);
    
    //Perform the FFT via Accelerate
    //Use FFT forward for standard PCM audio
    vDSP_fft_zrip(fftSetup, &output, 1, bufferLog2, FFT_FORWARD);
    
    //Scale the FFT data
    vDSP_vsmul(output.realp, 1, &fftNormFactor, output.realp, 1, numberOfFramesOver2);
    vDSP_vsmul(output.imagp, 1, &fftNormFactor, output.imagp, 1, numberOfFramesOver2);
    
    //Take the absolute value of the output to get in range of 0 to 1
    vDSP_zvabs(&output, 1, soundBuffer[inBusNumber].frequencyData, 1, numberOfFramesOver2);
    
    vDSP_destroy_fftsetup(fftSetup);
}