Interpolation means increasing the sampling rate, hence it is called upsampling. Let the original sequence be $x(n)$, the sampling rate be $f_x$, and the interpolation factor be $L$. The interpolation process involves inserting $L-1$ zeros between every two adjacent samples of the original sequence to form a new sequence, mathematically expressed as
Let $f_y$ represent the sampling rate of $y(m)$, then the relationship between the sampling rates is
The illustration is as follows:
From the frequency domain perspective, the frequency spectrum of the original sequence is periodically extended with a period of $f_x$.
The new sequence after interpolation is periodically extended with the new sampling rate $f_y$.
It can be seen that the spectral components before and after interpolation remain unchanged, but the spectral components at integer multiples of $f_x$ are referred to as mirror components. Therefore, a low-pass filter must be added after interpolation to eliminate the mirror frequency. A typical interpolator is completed by a combination of an upsampler and an anti-mirror filter.
Since interpolation involves inserting zero values into the original sequence, meaning that the signal amplitude at certain sampling points becomes 0, it will alter the signal amplitude, thus causing a loss of signal amplitude. To ensure uniformity of signal amplitude before and after interpolation, a gain factor L can be set after the interpolation filter.