Reconstruction of sinusoidal signals

In this section we will describe the reconstruction process of the D-to-C convertor which converts the sequence of discrete-time samples x[n] into the continuous-time signal x(t).

Screencast video [⯈]



Ideal D-to-C convertor

Because of the uniqueness issue in the discrete-time domain, the ideal D-to-C convertor makes a choice which frequencies are used for the conversion to the continuous-time domain. For this reason the ideal D-to-C convertor selects only frequencies which are inside the FI.

Ideal D-to-C converter.
Ideal D-to-C converter.

Let's start the description of the reconstruction from discrete-time to continuous-time by using again a simple sinusoidal signal from which the sequence of samples is described by x[n]=Acos(θn+ϕ). Because of the uniqueness issue we assume that the relative frequency θ is within the Fundamental Interval (FI), thus θ<π. In case this is not true, we first map the relative frequency into the FI. Furthermore we use the relation between relative and absolute frequency given by θ=2πffs, from which we assume as before that the ratio ffs is a rational number. Finally, since we have for this simple example the mathematical description of the signal samples as a sinusoidal function, we can obtain the mathematical description of the continuous-time signal x(t) by replacing the integer index n by the continuous-time variable tfs, leading to x(t)=x[n]|n=tfs=Acos(2πffsfst+ϕ)=Acos(2πft+ϕ) Fig. 1 shows an example where we converted the sequence of samples x[n]=cos(0.4πn)=,1,0.3,0.8,0.8,0.3,1, to the continuous-time signal x(t), with an ideal D-to-C convertor, which runs at a sample rate of fs=1000 [samples/sec]. In Fig. 2 the sequence of input signal samples x[n] is represented in a plot.

The reconstruction process where the input $x[n]$ is represented as a series of numbers.
The reconstruction process where the input x[n] is represented as a series of numbers.
The reconstruction process where the input $x[n]$ is represented as a plot.
The reconstruction process where the input x[n] is represented as a plot.
An ideal D-to-C convertor chooses the relative frequencies within the FI: π<θπ. In case a C-to-D convertor produces a frequency outside the FI, such a frequency will always first be mapped inside the FI and the D-to-C convertor will use this frequency for the conversion of the discrete-time signal into a continuous-time signal. This mapping back into the FI is called Aliasing.

Example


A continuous-time input signal x(t) is converted to the discrete-time domain samples x[n] with an ideal C-to-D convertor which runs at a sample rate of fs,in [samples/sec]. After that the samples x[n] are converted back to the continuous-time signal y(t) with an ideal D-to-C convertor which runs at a sample rate of fs,out [samples/sec]. The input signal x(t) and the sample rate fs,in are x(t)=x1(t)+x2(t), with x1(t)=4cos(400πt+π4), x2(t)=2sin(1000πtπ3) and fs,in=600 [samples/sec]. In first instance we choose the sample rate of the D-to-C convertor equal to the sample rate of the C-to-D convertor, thus fs,out=fs,in=600 [samples/sec]. After that we change the sample rate of the D-to-C convertor to fs,out=900 [samples/sec].
Calculate an expression for the output signal y(t) in both cases.
The reconstruction chain represented by ideal converters.
The reconstruction chain represented by ideal converters.

The first steps are the same of the last example of the previous subsection. The largest absolute frequency of the FI after the C-to-D convertor equals fs,in/2=300 [Hz]. Thus the frequency of x1(t) (200 [Hz]) is converted inside the FI, at relative frequency 2π3, while the frequency of x2(t) (500 [Hz]) is converted outside the FI, at relative frequency 5π3. After mapping back this frequency inside the FI, into relative frequency π3, we obtain the following mathematical description of the sequence of discrete-time samples x[n]: x[n]=4cos(2π3n+π4)2sin(π3n+π3) When converting the relative frequencies of this discrete-time signal into absolute frequencies of a continuous-time signal, we can make use of θ=2πffs. In case the sample rate of the D-to-C convertor fs,out=fs,in=600 [samples/sec] this leads to the following continuous-time output signal: y(t)=4cos(400πt+π4)2sin(200πt+π3). Thus the frequency of the first 200 [Hz] component did not change while the frequency of the second 500 [Hz] component has changed due to aliasing. However if we change the sample rate of the D-to-C convertor into fs,out=900 [samples/sec] the resulting continuous-time signal becomes: y(t)=4cos(600πt+π4)2sin(300πt+π3). Thus both frequencies are changed. The frequency of x2(t) has changed because of aliasing while above that both frequencies of x1(t) and x2(t) have changed because of the fact that the sample rates of C-to-D and D-to-C are different.



Reconstruction D-to-C convertor based on samples x[n]

The D-to-C convertor constructs a continuous-time signal x(t), based on the discrete-time samples x[n]. In the previous subsection we have seen that for sampled sinusoidal signals the D-to-C convertor in effect replaces the discrete-time integer variable n by the continuous-time variable tfs. As a result we can construct, for such a case, the continuous-time signal x(t) based on the mathematical description of the sinusoidal function. This procedure is only valid and possible in case of sampled sinusoidal signals and an example is shown at the left-hand side plot of Fig. 3. The continuous-time sinusoidal signal x(t) (in blue) is constructed by replacing n into tfs in the sinusoidal description of the samples x[n].

The reconstruction process with different types of reconstruction waveforms.
The reconstruction process with different types of reconstruction waveforms.
Only in case of sampled sinusoidal signals the ideal D-to-C convertor in effect replaces n by tfs.

In the more general case the continuous-time output signal y(t) is constructed by placing a continuous-time function p(t) at each new position of a sample x[n] and adding together all the resulting continuous-time signals. This reconstruction is performed via the so called Interpolation equationy(t)=n=x[n]p(tnTs) in which the continuous-time function p(t) represents the characteristic of the D-to-C convertor. In practice there are many different characteristic forms p(t), e.g. a square pulse: p(t)={1,12Ts<t12Ts0,otherwise

which leads to the so called ‘zero order hold’ procedure, which is depicted in the middle-part of Fig. 3. In this procedure the electronics of the D-to-C convertor are build in such a way that around each new discrete-time sample a continuous-time signal is constructed which has the same value as the discrete-time sample, while this value is kept constant during one sampling period of Ts [sec]. Thus the reconstructed continuous-time signal looks like a ‘stair-case’ approximation of the sinusoidal signal as depicted (in red) in the left-hand side figure. The electronics needed for this conversion procedure are very simple and cheap. For this reason the ‘zero order hold’ procedure is applied very often in practice.

Another characteristic form of p(t) is a triangular pulse: p(t)={1|t|Ts,|t|Ts0,otherwise

which leads to the conversion as shown in the right-hand side of Fig. 3. The signal samples are interpolated in a linear way. The reconstructed continuous-time signal looks like a ‘triangular-like’ approximation of the sinusoidal signal as depicted (in red) in the left-hand side figure.



Sampling theorem

From the previous section it follows that for a system as depicted in Fig. 1, with fs,in=fs,out=fs, the output y(t) of the D-to-C convertor is equal to input x(t) of the ideal C-to-D convertor if the continuous-time signal x(t) contains no frequencies higher than fmax and the sampling frequency fs is larger than 2fmax. This results into the following important sampling theorem, which is usually assigned to Shannon, who introduced this theorem around 1940. Almost at the same time Kotelnikov did the same. However some years before the theoretical basis was created by E.T. and J.A. Whittaker. This theorem states:

A continuous-time signal x(t) with frequencies no higher than fmax can be reconstructed exactly from its samples x[n]=x(t)t=nTs, if samples are taken at a rate fs=1/Ts, that is greater than 2fmax.