FFT order on different buffer sizes

Audio Plugins
1

Goodmorning there,
i am trying to use dsp::FFT
I have managed to apply forward and inverse with perfect results in 44100 sampleRate and 448 samples/buffer using the following code.
I choose 7th order FFT so that FFTSize is an integer product of buffer’s samples. If i dont do that, the last window will remain incomplete.

How do we cope with different sample rates and different buffer sizes for this to work?

processor.h:

fftOrder = 7,           
fftSize = 1 << fftOrder //  FFT size = 64 = buffer samples/7
std::unique_ptr<dsp::FFT> myFFT; 
float fifo[fftSize];           
float fftData[2 * fftSize];
int fifoIndex = 0;

processor.cpp/constructor:

//initialise FFT obgject
myFFT.reset(new dsp::FFT(fftOrder));

processor.cpp/processBlock:

 for (int channel = 0; channel < totalNumInputChannels; ++channel){
    auto* channelData = buffer.getWritePointer (channel);
    for (int i = 0; i < buffer.getNumSamples(); i++) {
		//fill fft array
		fifo[fifoIndex] = channelData[i];
		fifoIndex++;

		//when fft window is full
		if (fifoIndex == fftSize)
		{
			//reset window index
			fifoIndex = 0;

			//create FFT input data in appropriate size
			zeromem(fftData, sizeof(fftData));
			memcpy(fftData, fifo, sizeof(fifo));

			//perform forward FFT on fftData
            myFFT->performRealOnlyForwardTransform(fftData, true);


			//fftData now contains frequency domain information
			
            
			//perform inverse FFT on frequency domain information fftData
            myFFT->performRealOnlyInverseTransform(fftData);


			//write fftData back to audio buffer
            for (int j = 0; j < (fftSize); j++){
                channelData[(i - fftSize + 1) + j] =fftData[j];
            }
		}