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Signals, Spectra and Signal Processing

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Signals, Spectra and Signal Processing

ANALOG-TO-DIGITAL AND DIGITAL-TO ANALOG CONVERSION


Activity No. 4

I. INTENDED LEARNING OUTCOMES

1. Understand the process of analog-to-digital and digital-to-analog conversion and


the concepts, principles and theories behind.
2. Create a VI that will demonstrate the sampling process, the effect of aliasing, the
quantization and coding processes, quantization error and signal reconstruction.
3. Apply the concepts learned in practical signals.

II. BACKGROUND INFORMATION

The process of analog-to-digital signal conversion consists of converting a continuous-


time and amplitude signal into discrete-time and amplitude values. Sampling, quantization
and coding constitute the steps needed to achieve analog-to-digital conversion.

Block diagram of analog-to-digital converter

Sampling is the process of generating discrete-time samples from an analog signal. The analog and
digital frequencies are related in the following manner. Consider an analog sinusoidal
signal

. Sampling this signal at (note that sampling is time-
scaling of analog signals), with the sampling time interval of , generates the discrete-time
signal

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ᄉᄃ
where with , the sampling time interval, is the reciprocal of ᄂ, the sampling frequency. Note

that the discrete-time angular frequency


, where is the discrete-time frequency. Therefore,

or that the discrete-time frequency is related to the continuous-time frequency by


dividing it with the sampling frequency. With this, we deduce the following properties of
the discrete-time frequency in relation to the analog frequency:
• The same discrete-time signal is obtained from different continuous-time signals
if the product remains the same, as illustrated in the figure below:

Sampling of two different analog signals leading to the same discrete-time signals.

• Different discrete-time signals are obtained from the same continuous-time signals
when the sampling frequency is changed, as illustrated in the figure below:

Sampling of the same analog signals at different sampling frequencies.

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Recall in the previous lesson that the range of frequencies allowable for continuous-time sinusoids is

−∞ < ᄂ <

−∞ < <

and that of the discrete-time sinusoids is

− 1

1
< <


< <
When a continuous-time sinusoid is sampled at a rate of ᄂ - , its frequency ᄂ must fall between


1 ᄂ


≤ ᄂ≤

1

ᄂ ≤ ᄂ≤

Since sampling a continuous-time signal involves mapping a frequency with infinite range into a
frequency with a finite range, sampling introduces ambiguity. This can be illustrated in the two properties
presented before, that is two different continuous-time signals can lead to the same discrete-time signals
when sampled and a continuous-time signal can have different discrete-time signals when sampled.
Since the maximum frequency of a discrete-time signal is
or ᄉ, it follows that, for a sampling frequency ᄂ, the continuous-time signal that can be sampled has
a maximum frequency of

!"

!"

Beyond ᄂ !" ( !" ), continuous-time signals that will be sampled will have the same set of
discrete-time signals as a continuous-time signal with a frequency less than ᄂ !" . This
phenomenon is called aliasing. This will be illustrated in one of the activities. It would be
interesting to note that the effect of sampling can also be analyzed in frequency domain, but
the tools required for such analysis are not yet discussed. Thus, the discussion of the effect
of sampling on the frequency domain of signals will be left out for now.

A digital signal (note that a discrete-time signal is NOT a digital signal) is a sequence of numbers (samples)
in which each number is represented by a finite number of digits (finite precision). Thus, a sampled continuous-
time signal (a discrete-time signal) must have its values discretized to make it a digital signal. The process of
converting a discrete-time continuous-value signal into a digital signal by expressing each sample value as a
finite (instead of an infinite) number of digits is called quantization. Since each

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sample is being converted into one with discrete values during the process of quantization,
an error, called the quantization error or quantization noise is introduced. The quantizing
operation is denoted as Q, thus the quantized signal of the sampled signal is
$ Q% &

The quantization noise is a sequence e $ defined as the difference between the


quantized value and the actual sample value, or
e$ $ −

There are two methods by which quantization are done: truncation and rounding. Truncation discards
excess decimal digits that are in excess of the discrete-values, while rounding off involves rounding the
values to the nearest discrete-value. In truncation, the maximum quantization error per sample depends
on the step size, or the difference between the quantized values, while for rounding, it is fixed at 0.5 per
sample with the quantization error evenly distributed over the range -0.5 to 0.5.

The quantization process, as well as the analysis of quantization error is illustrated in


one of the activities.

The quantized signal, since this is already discrete in time and in value, is a digital signal. To complete
the process, the quantized signal is coded, that is bit streams of zeros and ones are assigned to each
discrete-value. This process is called coding, and is also illustrated in one of the activities.

According to Nyquist theorem, an analog signal can be recovered from it sample


values using the interpolation function
g i
ᄂ !"
ᄂ !"

if and only if it satisfies the Nyquist criterion (the Nyquist criterion will be deduced during
the performance of activity). Thus, can be expressed as
-
* + ,g+− ,
ᄂ ᄂ
./0-

where / ᄂ or the samples of the analog signal . The interpolation function g is called the i
function and the operation used to recover from its samples is called the convolution of the samples
and the i
function. The reconstruction process will be illustrated in one of the activities.

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III. LEARNING ACTIVITIES

Activity 4.1 – Sampling of Continuous-time Signals and Aliasing


1. Create the VI act04-01.vi with the following FP and BD.

Block diagram for True Case

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Block diagram for False Case.

• Place a Pointer Slide control on the FP. Name this control as Analog Frequency F
(Hz). Adjust the size of this control and the maximum value of the pointer to 1000.
• Place two Numeric Indicators on the FP, naming them as Normalized Frequency
(Fs = 1 kHz) and Aliased Frequency (Hz). Resize these indicators accordingly.
• Divide the output of the Analog Frequency F (Hz) node by 1000 using the Divide
node. Wire the quotient to the input of the Normalized Frequency (Fs = 1 kHz)
node. Call this quotient as the normalized frequency.
• On the BD, place three (3) Sine Wave.vi nodes. On the first Sine Wave.vi node, create constants
10 to the samples input, and 90 to the phase input. Wire the frequency input to the output of the
Divide node. Insert a Build Waveform node, and expand it downwards, in which an attributes
terminal will appear. Right-click on the Build Waveform node, point to Select Item and check the
dt item. The attributes terminal will be changed to dt terminal. Wire the sine wave output of the
Sine Wave.vi to the Y terminal of the Build Waveform node. Place a Reciprocal node, create a
constant 1000 at its input and wire the output to the dt terminal of the Build Waveform node.
• On the second Sine Wave.vi node, create constants 1000 and 90 on the samples and phase in
terminals of this node respectively. Insert a Divide node, this time dividing the normalized
frequency with 100. The output of this Divide node is wired to the frequency terminal of the
second Sine Wave.vi node. Insert another Build Waveform node with which one of the terminals
must be dt. Wire the sine wave output of the Sine Wave.vi node to the Y input terminal of the
second Build Waveform node inserted. Insert a Reciprocal node, create, 100000 constant at its
input terminal and wire the output to the dt terminal of the second Build Waveform node.

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• Insert a Case Structure. Then insert a Greater or Equal? node. Wire the first input of
this node to the normalized frequency and on he second input, create a constant 0.5.
Wire the output of this node to the selector terminal of the Case Structure.
• In the True case of the Case Structure, create an Expression Node. Inside the Expression Node,
type the formula −
2 34 − 567, then wire its input to the normalized frequency output while its output must be divided
by 100 and the quotient wired as an input to the frequency terminal of the third Sine Wave.vi. On
the samples and phase in of the third Sine Wave.vi node, create the constants 1000 and 90
respectively. Insert another Build Waveform node, with Y and dt as the input terminals. Wire the
sine wave output of the Sine Wave.vi node to the Y input of the Build Waveform node, and on the
dt terminal, the reciprocal of 100000. Insert a Multiply node outside the Case Structure, and wire
the output of the Expression Node to one of the inputs of the Multiply node. Create a constant
1000 the other input terminal. Wire the output of this Multiply node to the input of the Aliased
Frequency (Hz) numeric indicator. Wire the outputs of the second and third Build Waveform
nodes to the edge of the Case Structure, creating tunnels. Note: The tunnel for the third Build
Waveform, as well as the one going to the Multiply node is not yet solid at this point. This is
because the False case is not yet wired.
• Switch to False case. Connect the tunnel created from the second Build Waveform node
to the one created from the third Build Waveform node. Then on the tunnel connecting to
the Multiply node, create a constant 0. Note: The tunnels are now solid.
• Insert Build Array node, expanding it to have three input terminals. Wire each of the
Build Waveform node to the input of the Build Array node (take note of the order!). On
the FP, create a Waveform Graph. Switch to BD. The node for the Waveform Graph
appears in the BD. Wire the input of this to the output of the Build Array node. Switch to
FP. Expand the Legend (the one containing Plot 1) to include three plots. Rename each
as Sampled Signal, Original Signal and Aliased Signal, respectively. Right-click on the
Waveform Graph, and choose Properties. The Graph Properties window appears. On
the Plots tab, select Sampled Signal as Plot, and configure it as follows:

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• Enclose everything inside a While Loop structure. Create a Stop control at the
Loop Condition terminal of this structure.

2. Save and run the VI.

Question: Discuss the functionality of the VI.


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Question: Explain the choice of constants for the number of samples and the dt of the Sine Wave.vi
and Build Waveform nodes respectively.
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Question: Adjusting the values of the Analog Frequency F (Hz), complete the table below.

Input Analog Frequency F (Hz) Normalized Frequency (f) Alias Frequency (Fa)
0
100
300
500
700
800
900
1000

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Question: Based on the table above, what analog frequencies did have an alias
frequency? In general, what range of frequencies of signals will have aliases based on
the observations that can be drawn from the table above? What is the relationship of
frequencies of signals which did have aliases to the sampling frequency?
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Question: As stated in the Background Information part of this activity, a signal can be
reconstructed from its samples if the sampling process satisfies the Nyquist criterion.
The Nyquist criterion requires that the samples of the signal do not belong to the alias of
the original signal. What then is this Nyquist criterion?
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4.2 – Quantization and Quantization Error


1. Create the act04-02.vi. This VI will quantize the signal

76
e p−15 i
5
67

whose plot is shown below. Note that this signal has a value between 5 and 7. The
values of this signal will be discretized to integer values of the signal. The quantization
error e $ when this signal is quantized will be analyzed using a histogram.

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0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0. 45 0. 5

Plot for x(t)=5.2 exp (-10t) sin (20πt) + 2.5

2. Build the BD and FP as shown.

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• Place a Formula Waveform.vi. Then create the constants 10 for the frequency terminal,
5.2 for the amplitude terminal, the string constant exp(-10*t)*a*sin(w*t)+2.5 for the formula
terminal, and 1000 (for Fs) and 300 (for #s) for the sampling info terminal.
Note: The values at the frequency and amplitude terminals of the Formula
Waveform.vi node are automatically assigned in the variables w and a respectively.
• Insert a Get Waveform Components node. Wire the signal out terminal of the Formula
Waveform.vi node to the waveform input terminal of the Get Waveform Components node.
• Insert a To Unsigned Byte Integer node. Wire the Y output terminal of the Get Waveform
Components node to the number input of the To Unsigned Byte Integer node.
• Insert a Build Waveform node. Wire the waveform input terminal of this node to
the signal out terminal of the Formula Waveform.vi and the unsigned 8bit integer
to the Y input terminal of this node.
• Insert a Build Array node. Expand the node so that two input terminals appear. The
first input terminal is to be wired from the signal out of the Formula Waveform.vi, while
the second one is from the output waveform terminal of the Build Waveform node.
• Insert a Subtract node. Wire the output waveform terminal of the Build Waveform
node on the first input terminal, and the signal out of the Formula Waveform.vi
node on the second input terminal of the Subtract node.
• Insert a Create Histogram Express VI. The Configure Create Histogram configuration window
appears. Set the Number of bins to 10, the Maximum value to 0.5, the Minimum value to -0.5,

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and choose Sample count for the Amplitude Representation. Click OK. Right-click
on the Histogram output terminal of the Express VI, then choose Create > Graph
Indicator. Adjust the appearance of this indicator as it appears in the FP.
• Insert two other Waveform Graphs in the FP. Wire the first one at the output of the
Build Array node, and name this as Input/Output Signal. Wire the second one at
the output of the Subtract node, and name this as Quantization Error.
• In the FP, configure the Input/Output Signal waveform graph. Expand the plot legend to display
Plot 0 and Plot 1, renaming this as Input Signal and Output Signal respectively. Right-click on this
waveform graph, choose Properties and on the Plots tab, choose Output Signal from the Plots
drop-down menu, then choose the stepwise horizontal, indicated by icon as
the Plot Interpolation method. Click OK.

3. Save and run the VI.

Question: Discuss the functionality of the VI. How did the VI quantize the input signal?

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Question: Discuss how the plot of the Quantization Error waveform graph was generated.
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Question: What is a histogram? Interpret the histogram of the quantization error as


displayed in the waveform graph.
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4. Modify the BD as shown below. This will write the output signal and the quantization
error (the output of the Subtract node) vectors onto a spreadsheet.

• Place two Get Waveform Components node. Wire the first one to the output signal terminal of
the Formula Waveform.vi node and the second one to the output of the Subtract node.
• Place a Build Array node. Expand this node so that two input terminals appear. Wire the Y outputs
of the Get Waveform Components to the input terminals of the Build Array node as shown.
• Place a Write To Spreadsheet File.vi node. Wire the 2D input terminal of this node to
the output terminal of the Build Array node. Then create a constant on the transpose
terminal of the Write To Spreadsheet File.vi, clicking it to change it to True (T) state.

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5. Save and run the VI. When prompted, name the file as SQNR.xls. Open the
spreadsheet file using MS Excel or OpenOffice Calc. Click Yes when asked if the file is
to be opened even though the format is different than specified by the extension.

Question: What is the data that is contained in the first and second columns of the
spreadsheet file?
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6. Create a formula in the spreadsheet that will compute for the energy of the signal in the first and

second columns. Determine the power of the signals for a certain number of samples by dividing the
energy by the number of samples of the signal. Call the power of the signal in the first column as P
and on the second column, P;. The ratio P/P; gives the signal-to-quantization noise ratio (SQNR).

Question: Complete the table below.

Signal Power <= for Quantization Noise SQNR SQNR


Power <A for
> ≤ ? ≤ @>> (absolute) (decibel)
> ≤ ? ≤ @>>

Question: What does the SQNR parameter indicate with regards to the performance of
an analog-to-digital converter?
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Activity 4.3 – Coding
1. In this part of the activity, the signal from Activity 4.2 will be coded to a bitstream of
3 bits, 4B , 4 ᄉ and 4. Create the VI act04-03.vi. Build the FP and BD as shown.

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• Place a Formula Waveform.vi. Then create the constants 10 for the frequency terminal,
5.2 for the amplitude terminal, the string constant exp(-10*t)*a*sin(w*t)+2.5 for the formula
terminal, and 1000 (for Fs) and 300 (for #s) for the sampling info terminal.
Note: The values at the frequency and amplitude terminals of the Formula
Waveform.vi node are automatically assigned in the variables w and a respectively.
• Insert a Get Waveform Components node. Wire the signal out terminal of the Formula
Waveform.vi node to the waveform input terminal of the Get Waveform Components node.
• Insert a To Unsigned Byte Integer node. Wire the Y output terminal of the Get Waveform
Components node to the number input of the To Unsigned Byte Integer node.
• Insert an Array Size node. Wire the input of this node to the output of the To
Unsigned Byte Integer node.
• Place a For Loop structure. In the count (labeled by N) terminal of the For Loop, wire the
output of the Array Size node. Then create a tunnel at the edge of the For Loop structure
(which is, by default automatically indexed). Inside the For Loop structure, place a
Number to Boolean Array node. The input to this node is the tunnel from the Array Size
node. Place an Index Array node inside the structure, wiring its array input from the
Number to Boolean Array node output terminal. Expand the Index Array node so that
three input terminals appear. Create constants 2, 1 and 0 in each of the input terminals as
shown. Each of the outputs of the Index Array node is wired to a Boolean To (0, 1) node.
• Insert four waveform graphs, naming each as b3, b2, b1 and Quantized Signal. Wire
each of the Boolean To (0, 1) to b3, b2, and b1 as shown, while the Quantized Signal
is to be connected to the To Unsigned Byte Integer output terminal.
• Switch to FP. Right-click each of the graphs, then choose Visible Items and uncheck Plot
Legend to hide the plot legend at the top of the waveform graphs. Right-click each of the
graphs again, then choose Properties. On the Plots tab, choose the stepwise horizontal
as the Plot Interpolation method (see step 1 of the Activity 4.1). Click OK.

2. Save and run the VI.

Question: Interpret each of the waveform graphs.


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Question: Discuss the process of coding the quantized signal as observed in the activity.
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Activity 4.4 – Signal Reconstruction


1. In this part of the activity, a unity amplitude sinusoid whose frequency is 10 Hz is sampled at 80
Hz. It will be reconstructed using the i
function. The main VI consists of two subVIs. Build the first subVI act04-04_AddZeros.vi as shown.
The input terminals to this subVI are the waveform and the Oversample rate. The output terminals
are Zero-Inserted Samples and Number of Samples.

2. Build the second subVI as act04-04Sinc.vi as shown. The inputs to this VI are # of period of sinc
function and Oversampling rate. The outputs are # of samples sinc pattern and Sinc Pattern.

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3. Build the main VI act04-04.vi as shown.

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• The Add Zeros and Sinc Function are subVIs created previously.
• Configure the Analog Signal waveform graph as follows: Expand the Plot Legend. Rename
the signals as Analog Signal for the first plot and Samples for the second plot. Right-click the
waveform graph, choose Properties. On the Plots tab, choose the Samples plot, then choose the

Plot Interpolation Method as None (indicated by the icon), then choose zero
under the Fill to drop-down menu. Click OK.

4. Save and run the VI. Adjust the Oversampling Rate and # of Period of Sinc
Function then observe the waveform graphs.

Question: Discuss the functionality of the Add Zeros and Sinc Function subVIs.
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Question: Discuss the functionality of the main VI. Interpret the output of the waveform graphs.
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Question: Adjusting the Oversampling Rate value while keeping the # of Period of Sinc
Function constant, what can be said about the reconstructed signal in comparison with
the original Analog signal?
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Question: Adjusting the # of Period of Sinc Function value while keeping the
Oversampling Rate constant, what can be said about the reconstructed signal in
comparison with the original Analog signal?
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IV. CONCLUSIONS
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V. MACHINE PROBLEM

Create a VI that will illustrate the analog-to-digital and digital-to-analog conversion of an audio
signal. The VI will read a recorded audio, convert it into digital form and back to the analog form. The
VI must illustrate the ADC process (sampling, quantizing, coding) as well as the signal reconstruction
using the i
function and write the reconstructed signal back to WAV file. It must also demonstrate what will
happen if aliasing occurs, and how to prevent it from occurring. Call the main VI of this machine
problem as macpro04.vi. The subVIs (if any) that will be used in the activity must be named
macpro04_<function>.vi. The <function> name must suggest the functionality of the subVI (e.g. the
subVI macmpro04_sampler.vi is a subVI that will perform the sampling of the audio).

VI. REFERENCES
Kehtarnavaz, N. & Kim, N. (2005). Digital Signal Processing System-Level Design
Using LabVIEW. Oxford, United Kingdom: Elsevier, Inc.

Proakis, J. & Manolakis D. (2006). Digital Signal Processing: Principles, Algorithms and Applications 4th
Edition. New Jersey, USA: Prentice-Hall.

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