Let us now look at increasing the sample rate. Since it is less obvious how
to achieve this, let us first consult theory.
Given are the samples
of a band-limited
signal
taken at frequency
is above the
Nyquist rate
. We want to compute the samples
of
at a higher sampling rate
. This means that the new sampling step
should be
or,
.
Since the original sampling rate
is above Nyquist, we can in theory
reconstruct the entire signal
using the reconstruction formula
[link] using
. Once the continuous-time (finite energy) signal
is obtained, we only need to sample it at
. This reads as follows
This formula allows indeed to compute
from
, at least in principle.
However, a closer look at theory is required to understand the effect when usingonly finite many samples.
The reconstruction formula
[link] is best understood in the frequency domain: it
amounts to removing the spectral copies of
via filtering
with cut-off frequency
. To make this filtering step visible we need to write
[link] in form of a convolution. To this end, we write
where the new sequence
is obtained by “upsampling” and is given as:
The convolution
[link] allows for more
convenient data processing via
digital filtering and for a
simple spectral interpretation.
Interpolation by
or resampling
at
times larger rate consists of the following steps:
Up-sampling (in French: “interpolation”)
Multiplication by
and
Low-pass filtering at cutoff frequency
using the ideal filter
:
Spectral picture of Interpolation by
In analogy to the decimation we set
. Then, the samples of
taken at the same rate
constitute the samples of
taken at rate
.
Analogously to the decimation we find quickly
which indicates how the spectrum at rate
is obtained: the spectrum
at rate
gets contracted in the frequency axis by
and expanded in amplitude
by
.
Note that the spectral copies of
are at distance
just like those
of
.
We may break the procedure down into the individual steps:
(i) Upsampling (introducing the zero-samples) leaves the Fourier transform, and thus the spectrum
almost intact, leading only to a rescaling of the frequencies(contraction of
):
Indeed, the Fourier transform
of the samples
becomes
The corresponding signal
(with samples
at sampling rate
) is not of interest. If you want to know about it any way,
see
Comment 7 . Note, however, that the fundamental period of
amounts to
and contains
copies of
(see
[link] ).
Comment 7 For clarity: the Fourier transform of
is found by
removing the spectral copies of
outside
.
These copies are caused by sampling. The Fourier transform of
consists of
contracted copies
of
at distance
of each other which are caused by upsampling. Using the reconstruction
formula
[link] with the samples
, sample rate
and
correct pass-band
yields the signal
. Using that
while
for all integer
we find quickly that the samples of
are indeed
, i.e.,
.
(ii) Multiplication with
restores the average value of the samples.
The Fourier transform is now
which consists of copies of
, or
,
at distance
(as for
, there are
copies in one period).
(iii) The digital low-pass filtering of
at cut-off
frequency
removes all of the copies of
except the ones centered at 0,
,
etc. and leaves only one copy per period,
in other words, only copies at distance
.
What remains is exactly
as we have pointed out earlier.
(see
[link] ).
Power and Interpolation
Similarly as with the decimation
the power of a periodic signal does not change under interpolation.In fact, we only need to use
and replace
by
in the computation
done with the decimation. We conclude that
the power of discrete samples does not change under interpolation provided that
, at least
approximatively.
To verify this, let us move through the 3 steps above. Step (i), upsampling, reduces power by a factor
since the sum of squares of the samples is the same (the zeros
added don't contribute), but there are now
times more samples. Power is an average.
Step (ii) obviously multiplies power with
. Step (iii), the low-pass filtering,
removes
spectral copies and leaves only 1, thus divides power by
. All steps
together leave the power as it is.