[linux-audio-dev] Audio sampling and fft accuracy.

Hi, If I have a sample 2kHz sine wave signal at 50% peak. I then add a white noise signal at 20%. I then do a FFT transform. If I design the FFT so that 2kHz is at the center of one of the FFT buckets, how close will the FFT figure be to the 50% sine wave input. Obviously, there will be a variation due to the amount of noise. So the output of the FFT buckets will be 50+-error. Will this variation as a result of noise decrease as I use more and more samples per fft block? I.e.Will the +-error value decrease? I would think it should decrease, as a result of the general averaging the fft routine will make to the noise signal. I would think that as the noise was random, it should average out to zero over time. I.e. 20% noise samples. Average each sample, the average should be zero as one reaches infinity number of samples. So, if my reasoning above is correct. I can draw the following conclusion: A 2kHz signal sampled at 8kHz and then FFTed, the 2kHz bucket will be less accurate than a 2kHz signal sampled at 16kHz and then FFTed. Now, I just need to work out how much more accurate it would be. James