I'm looking at the problem of integer upsampling (i.e. interpolation). This can be achieved by first increasing the sample rate by factor $L$ (i.e. inserting $L-1$ zeros between the original samples) and then low-pass filtering the signal (see e.g. Oppenheim/Schafer, "Discrete Time Signal Processing", Sect. 4.6.2).
When I compare the the samples of the original signal with the corresponding samples of the upsampled signal, I observe a slight difference in amplitude. Is there a way to avoid this and to leave the original samples unaltered? That's what I would expect of an interpolation method.
The filter I used is a symmetric FIR filter with an even number of coefficients and for comparison I compensate the delay introduced by that filter.
I know that there are other interpolation methods (linear, cubic, hermite, etc.), but I would like to use this particular method.
I hope I could make myself understood. Thanks in advance for your help!