Hello,
I've got some 4s recorded music samples (44100Hz, 16bits) for which I'd like to damage the frequency resolution. I've got some difficulties to achieve this task.
Actually, I first applied an FFT transform to get the spectrum (mod. and phase), then I damaged the frequency resolution (initially equal to 0.25Hz) by taking for example one point every 8 points from the first point (modified resolution equal to 2 Hz); in order to get back a modified signal sampled at 44100Hz; the damaged spectrum is linearly interpolated to get a value every 0.25 Hz; then several zero paddings are added to get oversampling before the complex iFFT is performed. After iFFT, I just keep the real part of the signal and undersample it to recover a sampling frequency equal to 44100Hz. But, when listening to the so transformed sound, it doesn't really sound like the original sound, there's a short impulse noise at the beginning of the sample and a short sequence of sound at the end which seems modulated (the sound is "accelerated"); Can anyone help me to detect where my reasonning is erroneous ?
Thanks in advance.
Skoobi.
