This question is more or less the following of a previous question.
The aim is to equalize some sounds signals. By equalizing, I mean increasing or decreasing some frequency ranges, a bit like an analogic equalizer. The process I am applying now is :
1/ getting the temporal sound signal
2/ getting the "spectral correction" coefficients for each frequency sample
3/ converting the spectral correction to a "temporal correction" by using a FFT
4/ convoluting the temporal sound signal with the "temporal correction".
I have many questions with that process.
The first one is : what about the phasis ? The temporal sound frames are integers given by by wav file. So, if I write them with a complex number, the real part will be the value of the sample and the imaginary part will be 0.
But when i convert the spectral correction to a "temporal correction", some imaginary values may appears. And, after my convolution, I'll get a temporal signal with some imaginary parts wich will be different from 0. So, how, coming back to a temporal signal with a phasis = 0 ? Shall I use the modulus of the complex get by my convoluted signal ?
I also have a question about the "temporal correction". For testing, I'm using a "blank correction" : all the spectral coefficients = 1 so, my signal convoluted should be equal to the original signal. When I calculate the FFT of the spectral correction, I get a tab with those values {1, 0, 0, 0 (...)}.
So, my second question is : Does this result is logical or shall I get {1, 0, 0 (...), 1} because of the hermitian symmetry ?
And my third (and last) question is : What about the "padding" ? I'm following this website to compute the convolution but, I don't understand how many "0" I do have to add before or after the input signal. Let's say I have a 512 frames sound signal to convolute with a 511 frames "correction signal". Shall I add 511 "zero" before the signal ? or 255 before and 255 after ?
Thank you for all your answers.
Dr_Click