I am currently reading some code, so I can understand how this FFT output is actually been stored.
I have a function which computes FFT of audio file, and stores is in the format.
[real0, realN/2-1, real1, im1, real2, im2, ...]
which kind of confuses me, as I remember DFT/FFT converts from time domain to frequency domain, in which the real part is the amplitude, and the complex part is the phase, but aren't I missing some information regarding what frequency has what amplitude and what phase?
- Am I missing something?
- Or am I right when i say that I can't from these complex numbers extract the amplitude for a certain frequency?
I am currently using Kaldi framework, in which i am trying to compute the spectogram of an audio file...
The output file it has generated looks like this.
This output was generated from a 1 second audio file. sampled with 16 kHz, and frame length = 25 ms and hop_length = 10ms. The number of datapoint the function has generated is 25816. I am not sure how i should interpret it, i mean is each value supposed to be the power for a given frequency, and if so which frequency ?