Labview

Labview Temperature Measurement and First-Order Dynamic Response

2010-09-25T08:01:00.001-07:00

Summary: In this lab exercise, you will determine how various temperature measurement systems respond to different inputs. This analysis will include a thermometer, a thermocouple, and a thermistor.

Introduction:

In this lab exercise, you will determine how various temperature measurement systems respond to different inputs. This analysis will include a thermometer, a thermocouple, and a thermistor.

Teaching Objectives:

The purpose of this laboratory is for each student to understand and do the following:

1. Instrument calibration.

2. Measurements with thermocouples and thermistors.

3. Determination of time constants and comparisons of the dynamic responses of three measuring systems.

4. Application of first-order dynamic behavior to dynamic temperature measurement problems.

Preparatory Reading:

Figliola and Beasley

Measurement System Behavior, First Order Systems: pp. 72–87

Thermistors: p. 295

Thermoelectric Temperature Measurement: pp. 300-313

Overview:

In this lab, we will use first-order models to approximate the response of a thermometer, a thermocouple, and a thermistor to temperature inputs. Each of these temperature sensors measure temperature in a different way.

A thermometer senses a change in temperature as a change in the density of a fluid.
A thermocouple consists of two wires of different metals joined at one end (the junction). When a voltage is applied across the free ends of the two wires, the differing properties of the wires create an induced voltage that it proportional to the temperature change at the junction.
A thermistor is a thermally sensitive resistor. When the temperature of the thermistor changes, a proportional change in resistance occurs.

Procedure:

Part 1: Modify a VI for temperature measurements

In this lab, we will build on the skills that were developed in Introduction to Benchtop Equipment and Data Acquisition. It is assumed that the student understands the concepts of a Front Panel, a Block Diagram, the Control Palette, and the Functions Palette. It is also assumed that the student can search the control and functions palette for necessary features.

1. Connect a thermocouple to channel 0 of the SCXI-1112 module.

2. Open LabVIEW 8.0.

3. Open the VI that you developed for Introduction to Benchtop Equipment and Data Acquisition.

4. Click File>>Save As…>>Substitute Copy for Original to save your VI with a new name.

5. On the Block Diagram, double click the DAQ Assistant to open the configuration dialogue box.

**Figure 1:** DAQ Assistant Configuration Box

6. Configure the DAQ Assistant as shown above.

1. Click Voltage and press Delete to delete the channel you used in the first lab.
2. A dialog box will ask if you want to delete the selected channel. Click Yes.
3. Add a new measurement channel.
  1. Click Add Channels>>Voltage.
  2. Select the channel that the thermocouple is connected to (SC1Mod2>>ai0).
  3. Click OK.
4. The input range should be 80mV max to -80mV min. (Thermocouples have a low voltage output.)
5. The Terminal Configuration should be Differential. (The ground terminal is not used.)
6. Change Acquisition Mode to Continuous.
7. Click OK.
8. A dialog box will open asking if you would like a While Loop to be made around the DAQ Assistant. Click No. (You will place a While Loop in the next step that includes the subVIs that are already on the Block Diagram.

7. Place a While Loop around the entire Block Diagram as shown below. LabVIEW While Loops are similar to While Loops in any text-based programming language. The routine inside the loop will run repeatedly until a true value is passed to the loop condition, represented by a red button at the bottom right corner of the loop.

**Figure 2:** Block diagram with while loop

8. A toggle button can be used to end the While Loop routine. Right click on the loop condition icon and click Create>>Control.

9. Double click the Write to Measurement File ExpressVI to open the configuration box. Under If a file already exists, select Append to File. Click OK. (Because the Write to File function now occurs in a loop, the data must be appended to the end of the file, rather than overwriting the data from the previous loop.)

10. On the front panel, place a Thermometer.

11. Right Click the Thermometer. Click Visible Items>>Digital Display.

12. On the Block Diagram, inside the While Loop, place a Time Delay Express VI. When the Configuration Box opens, change the Time Delay (seconds) to 0.001. (Without a delay in the While Loop, the program will repeat as quickly as it can and consume processor power.)

13. Your Block Diagram should resemble Figure 3.

14. Click File>>Save As…>>Copy>>Create Unopened Disk Copy. Save the VI with “Lab3” in the file name. (You will use the VI as it is now as a starting point in Operational Amplifiers and Signal Conditioning.)

15. Save a second copy with “Lab4” in the file name. (You will also use the VI as it is now as a starting point in Strain Gages and Force Measurement.)

**Figure 3:** Completed block diagram

Part 2: Calibration Curve

You will create a static calibration curve to correlate voltage measurements with temperature readings. We will use the thermometer as the standard. (Note: We use the thermometer as a standard for convenience. Do not assume that an alcohol-in-glass thermometer is a temperature measurement device with superior accuracy.)

2.1 Calibrate the thermocouple

1. Take temperature readings with the thermometer in ice water and boiling water. Be careful not to allow the devices to come into contact with the hot plate or the bottom of the beaker.

2. Record the temperatures measured.

3. Use the thermocouple to measure both temperatures. Again, be careful not to allow the devices to come in contact with the hot plate or the bottom of the beaker.

4. In Excel:

1. Plot the temperatures measured with the known standard values on the y-axis and the ADC output (voltage) resulting from the thermocouple input on the x-axis.
2. Create a linear curve fit using the two data points.

The linear fit results in an equation that relates the ADC output obtained from the thermocouple to the known temperature from the thermometer:

T=k(Vadc) + b

where “T” is the temperature, “k” is the gain (°C/V), “Vadc” is the voltage read in to the computer, and “b” is the zero-voltage offset. The gain “k” is the inverse of the static sensitivity, which has units of V/°C.

5. Repeat the calibration procedure for the thermistor.

2.2 Modify your VI for calibration

**Figure 4:** Modified block diagram

In this section, you will multiply the data by the conversion scalar “k” that you found experimentally. You will also add an offset “b”.

1. Delete the Tone Measurement ExpressVI. Delete the Frequency and Amplitude indicators.

2. Delete the wires that connect the DAQ Assistant to the thermometer, the Waveform Graph, and the Write to File ExpressVI. Press Ctrl+b to remove any broken wire remnants.

3. Place a Multiply and an Add on the block diagram.

4. Create a control and connect it to the y terminal of the Multiply icon. Rename the control “k”.

5. Create a control and connect it to the y terminal of the Add icon. Rename the control “b”.

6. Wire your block diagram as shown in Figure 4.

7. On the front panel, enter the values you found experimentally for the “k” and “b” values.

8. When you run the VI, your measurements should now be displayed as temperatures.

Part 3: Time Constants

The calibration curves you found represent static calibrations. In the following section, you will consider the dynamic behavior of the sensors. The dynamics of all the temperature sensors used in this lab can be modeled accurately as first-order systems. The natural response of a first-order system can be characterized by a single parameter — the time constant.

Time constants can be found by observing the response of the instrument to a step input in temperature. Note that the second medium governs the physics of the transient response. For each of the sensors, you will determine the time constant for transfers from ambient air to ice water and from hot air to ambient air.

An ice water bath will be provided for your tests. A heat gun will be used to generate hot air. Caution: Keep all temperature sensors (particularly the thermometer and the thermistor) at least 9 inches away from the heat gun. Holding them closer to the heat gun will result in temperatures outside the operating range of the sensors.

Four methods will be used to calculate various time constants. Table 1 summarizes the tests to be performed. The testing methods are described below.

Table 1. Time constant calculation methods.

TABLE 1
	thermometer	thermocouple	thermistor
ambient air to ice water	1, 2	3, 4	3, 5
hot air to ambient air	1	3	3
ice water to ambient air	-	3	-

Method 1- Time Elapsed at 63.2% Sensor Accuracy

Thermometer: Perform this test for ambient air to ice water. Repeat for hot air to ambient air.

1. Record the temperature of the initial medium, T0, and the temperature of the final medium, T_inf.

2. Calculate the intermediate reference temperature, Tr that is 63.2 percent of the way from T0 to T_inf.

3. With the instrument at steady state (reading T0) in the initial medium, move it quickly to the final medium.

4. The time from when the step in temperature began to when the instrument reads Tr is the time constant, . This time can be measured using a stopwatch.

Method 2 - Mathematic Definition of Tau

Thermometer: Perform this test moving from ambient air to ice water.

1. Record the temperature of the initial medium, T0, and the temperature of the final medium, T_inf.

2. Choose an arbitrary reference temperature, Tr that is between 60 and 80 percent of the way from T0 to T_inf.

3. With the instrument at steady state (reading T0) in the initial medium, move it quickly to the final medium.

4. Record the time trthat it takes to reach the reference temperature Tr.

5. The following expression can be used to calculate the time constant tau:

**Figure 5:** equation (1)

Method 3 – Analyze Acquired Data

Thermocouple: Perform this test for ambient air-to-ice water, hot air-to-ambient air, and ice water-to-ambient air.

Thermistor: Perform this test for ambient air-to-ice water, and hot air- to-ambient air.

1. Run the VI.

2. Insert the thermocouple into the initial medium.

3. Click the Enable toggle control as you transfer the thermocouple to the final medium.

4. When the sensor comes to equilibrium, stop the VI.

5. Open your measurement file in Excel.

6. Determine the time required for the sensor to complete 63.2 percent of the transient from the initial temperature to the final temperature.

7. Change the file path to prevent overwriting data. (You can either change the name of the file after it is written, or you can change the name of the path that the next file will be written to.) You will use the saved data for subsequent calculations.

Method 4- Error Fraction

Thermocouple: Complete this step using the data from the ambient air-to-ice water transfer.

Thermistor: Repeat this step using the data from the ambient air-to-ice water transfer.

Equation 2 below gives an expression for the error fraction.

**Figure 6:** equation (2)

1. Calculate the error fraction at each instant of time for the data you have gathered.

2. Take the natural log of the error fraction data.

3. Plot the natural log of the error fraction versus time.

4. Find the negative reciprocal of the slope of this curve. This is the time constant.

The error fraction versus time curve provides an easy way to judge the accuracy of a first-order model for the data gathered. If the error fraction plot is linear, then the first-order model is a good approximation of the dynamic behavior of the system. (See pages 78–79 of Figliola and Beasley for a more detailed discussion of the error fraction and its use to calculate the time constant.) Are the thermocouple and thermistor in water adequately modeled by the first-order system approximation?

Method 5- Least Squared Error

Thermistor: Complete this step using the ambient air-to-ice water data.

Equation (3) below gives the theoretical expression for the temperature, T(t), as a function of time when the temperature of the environment is suddenly changed from T0 to T∞. The thermocouple is assumed to be at temperature T0 when the change occurs at t = 0.

**Figure 7:** equation (3)

1. Using Excel, compute a value for tau so that the square of the error between the theory and your experimental result is minimized. This is called a least square error approach and is very common in engineering.

2. Plot both the test data and the results from Equation (3) on the same axes to compare how well this model fits the data.

Part 4: Additional Tests

Once you have performed all of the tests outlined above, consider the following tests:

•Transfer the thermocouple from ice water to room temperature air.

How does the time constant compare with the one obtained from the air-to-ice-water transition? Is it different? Why or why not?

•When calculating the time constant for the thermocouple in transition from air to ice water, you measured the time to go 63.2 percent of the way from the initial temperature (around 21 C) to the final temperature (around 0 C). Using the same graph, recalculate the time constant using14 C as the starting temperature (i.e., what is the time required to go 63.2 percent of the way from 14 C to 0 C?). Is this time constant value significantly different from the initial value you computed? Why or why not?

Part 5: Thought Questions

Document your response to these questions in your lab book.

1. How do the time constants of the thermometer, thermocouple, and thermistor compare? Why is the thermometer “slow” compared to the other devices?

2. How do the time constants obtained for air and water compare? Are they significantly different? If so, explain why.

3. Does the instrument need to be at steady state before transferring it to the final medium?

4. How accurate (qualitatively) is the first-order model approximation for the different sensors and media?

Part 6: Frequency Response

From the measured time responses of the thermometer, the thermocouple, and the thermistor, it can be seen that the response time of these sensors is not instantaneous. From this we can infer that these measurement systems may have some difficulty following inputs that change quickly. Frequency response is a means of characterizing the dynamic response capabilities of a system.

The frequency response of a system can be determined by calculating the response of a system to sinusoidal inputs at different frequencies. Two important characteristics are:

the magnitude ratio between the output and input waveforms
the phase difference between the output and input waveforms

For a first-order system, we can create the frequency response plot using only our knowledge of the system time constant. The magnitude ratio can be calculated from the following expression.

**Figure 8:** equation (4)

The phase difference is determined from expression 5.

**Figure 9:** equation (5)

In Excel, create a frequency response plot for the thermometer, thermocouple, and thermistor using the time constant from the ice water tests. Plot the magnitude ratio on a log-log scale and the phase difference on a semilog scale (log scale of frequency axis). Consider frequencies from 0.1 rad/s to 1000 rad/s. Your plots will look nice if you use ten points per decade of frequency (i.e., 0.1, 0.2, 0.3, … ,1, 2, 3, … , 10, 20, 30, …). Overlay the three magnitude responses on one plot and the three phase responses on another. How do the frequency response characteristics of the thermometer, thermocouple, and thermistor compare?

Part 7: Dynamic Temperature Measurement Problem

Include the analysis for this problem in your lab book.

Assume that the thermocouple you used is placed in a water stream at a point where the temperature varies sinusoidally between 60 C and 80 C at a frequency of 40 rad/s. Assuming that the thermocouple time constant is the same as the value you determined in the ice water case, what will be the range of sinusoidal variation in the temperature readings? Repeat the calculations for a temperature frequency of 2 rad/s. Now assume the thermocouple is in air instead of water. Determine the range of the variation using input frequencies of 10 rad/s and 1 rad/s.

Clean Up:

Please straighten up your lab work area. Turn off the data acquisition hardware and leave your work area clean. Your help in keeping the lab neat is appreciated.

Report:

For this lab, you are to write up the Method section of a full formal report. Guidelines are given in the example report in the Undergraduate Guide. Avoid presenting results in the Methods section. Your writing assignment is relatively short. Take the time to do a good job. Since you will not be submitting Results or Discussion of Results in the report, you will need to document your results in your lab book. Also, be sure to answer the questions from the lab handout in your lab book.

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labview Strain Gages and Force Measurement

2010-09-25T07:59:00.001-07:00

Summary: In this lab, you will learn about strain gages and the Wheatstone bridge circuit. You will see how they can be used for strain and force measurement. You will modify an existing program to measure the dynamic characteristics of a second-order system.

Introduction

In this lab, you will learn about strain gages and the Wheatstone bridge circuit. You will see how they can be used for strain and force measurement. You will modify an existing program to measure the dynamic characteristics of a second-order system.

Teaching Objectives

Gain practical experience with resistance strain-measurement techniques.
Learn about the Wheatstone bridge and how it is used in strain measurement.
Use a beam instrumented with strain gages as a force measurement device.
Use strain gages to measure the natural frequency and damping in a beam.
Design a force transducer for measuring thrust from a model rocket motor.

Preparatory Reading:

Figliola and Beasley

Strain Measurement: pp. 425–446

Procedure

Part 1: Strain Gages and the Wheatstone Bridge

The metal foil strain gages used in this lab are resistors with a nominal (unstrained) resistance of 120 ohms. As they are put in tension, their resistance increases; as they are compressed, their resistance decreases. The Wheatstone bridge provides a way to convert these changes in resistance to changes in voltage, which are easy to work with. These voltages can be conditioned, transmitted, or stored digitally.

**Figure 1:** Wheatstone Bridge Circuit

Figure 1 shows a Wheatstone bridge configuration.

Four resistors are connected in an end-to-end fashion.
The input or excitation voltage is connected to the bridge between top and bottom nodes of the circuit.
The output is the difference between the voltage at the left node and the voltage at the right node.
An excitation voltage is required to convert the change in resistance (in the legs of the bridge) to a change in voltage at the output of the bridge.

For the bridge shown, the output voltage is expressed as

**Figure 2:** equation (1)

When building a Wheatstone bridge with strain gages, all four resistors have the same nominal value. Bridges can be built in the following configurations:

Quarter Bridge-One strain gage and three fixed resistors
Half Bridge- Two strain gages and two fixed resistors
Full Bridge- Four strain gages

**Figure 3:** Quarter Bridge Configuration

Quarter Bridges

Figure 3 illustrates a quarter bridge configuration. The quarter bridge has one active leg, i.e., one leg with a changing resistance. From equation (1) above we can derive an expression for the output voltage as a function of the resistance change ∆R:

**Figure 4:** equation (2)

Half and Full Bridges

Figure 5 and Figure 6 show half-bridge and full-bridge configurations respectively.

Half bridge: two active legs, one in tension and one in compression. These legs are adjacent legs in the bridge.
Full bridge: four active legs, two in tension and two in compression. The gages in tension are on opposite legs of the bridge.

Using equation 1 and Figure 1 as a guide, derive expressions for the output voltage of the half-bridge and full-bridge circuits.

Thought Question

A half bridge could be made with two gages in tension on opposite legs. When would this be useful? What would be the main problem with doing this?

**Figure 5:** Half Bridge Configuration

**Figure 6:** Full Bridge Configuration

Part 2: Calibration of the Strain-gaged Cantilever Beam

Your TA will provide an aluminum beam instrumented with strain gages.

1. Clamp the beam securely to the edge of the lab table with the gaged portion of the beam cantilevered beyond the edge of the bench.

2. Using the junction box provided, connect a single strain gage in a quarter bridge configuration. (Refer to Figure 3 as a guide.)

3. Connect the junction box output cable to Channel 2 of the SCXI-1520 strain gage module. The SCXI-1520 provides an excitation voltage to the bridge and amplifies the output voltage from the bridge to ranges that are easily observable or acceptable for data acquisition. The excitation voltage Vi is 5 V and the gain is 500. When connected to the signal conditioning board, the amplified bridge output can be read by the data acquisition software.

4. Open the “Lab4” VI that you created in Temperature Measurement and First-Order Dynamic Response.

5. Hang weights from the end of the beam.

6. Record the voltage measured with LabVIEW in your lab book.

7. Repeat steps 5 and 6 for several different weights.

8. In Excel, plot the voltage (input) versus weight (output).

9. Find the best fit linear relationship for the data. The resulting equation can be used to calibrate the voltage output of the strain gages.

10. Apply several loads not used for calibration to test the validity of the linear curve fit.

Part 3: Comparison of Theoretical Strain to Measured Strain

In this step, we will compare a theoretically based estimate of strain for a given load to that which was measured earlier. First we will determine the strain corresponding to the voltage measurements of step 2.

1. Measure the excitation voltage for bridge circuit, Vi.

The circuit analyses from step one showed that the bridge output Vo is related to the excitation voltage by the following relationship

**Figure 7:** equation (2)

where K equals 1/4 for a quarter bridge, 1/2 for a half bridge, and 1 for a full bridge. The voltage measured in LabVIEW is related to the bridge output by

**Figure 8:** equation (3)

where Kamp is the gain of the signal conditioning board. Because the bridge resistances are not balanced exactly and the weight of the beam itself produces some strain, you will observe a nonzero output voltage when there is no load applied.

2. Measure the output voltage with no load. Call this voltage Voffset.

3. To determine the strain induced by the applied loads, measure the changes in the display voltages relative to this offset voltage (Vdisplay–Voffset).

For a strain gage, the gage factor is defined as

**Figure 9:** equation (4)

where epsilon is the strain experienced by the gage. The gages used in this lab have a gage factor of 2.12±0.8%.

4. Derive an expression for the strain in the beam using equations (2) through (4). Your empirical strain estimate should be in the range of 0 to 2000 microstrains.

A theoretical estimate of the strain can be obtained by drawing on topics from CEEn 203. The stress on the surface of a beam in bending is given by

**Figure 10:** equation (5)

where M is the applied moment at the location of interest, y is the distance from the neutral axis (in this case, the half height of the beam cross section), I is the area moment of inertia of the cross section with respect to the neutral axis. Recall that for a rectangular cross section,

**Figure 11:** equation (6)

Recall also that stress and strain are related by Young’s modulus:

**Figure 12:** equation (7)

For aluminum, E = 10.4x10^6 psi.

5. Using equations (5) through (7), estimate the strain where the gages are bonded to the beam. How does the theoretically obtained strain compare to the value determined from measurements? If they are different, what are some possible reasons?

Part 4: Experiments with Half and Full Bridges

Repeat parts 2 and 3 for the half bridge and the full bridge configurations.

Part 5: Measurement of an Unknown Load

Based upon the calibration determined in step 2, use your beam to determine the weight of an arbitrary object. You may use the quarter bridge, the half bridge, or full bridge configuration for this test. Measure the actual weight using a precision scale. How does the weight determined with your beam compare to the object's true weight? How certain is your measurement? What are some possible sources of uncertainty?

Part 6: Sensitivity to Extraneous Loads and Temperature Variations

Strain gages are often used as transducers to measure force in a structure or mechanical device. A perfect sensor is sensitive only to changes in the variable of interest and is completely insensitive to changes in extraneous parameters. (Such a sensor is impossible to create.) In practice, every effort should be made to minimize sensitivity to extraneous variables. In the case of the cantilever-beam scale, the objective is to measure the weight of objects or alternatively, to measure forces in the vertical plane. As such, a good design will be much less sensitive to loads in other directions (axial, lateral, torsional, etc.). Also a good design will not be sensitive to strains induced by other means, such as thermal expansion.

Your TA will supervise in the following two demonstrations.

6.1 Lateral Sensitivity

Use the instrumented beam of square cross section provided by your TA.

1. Clamp the beam in a cantilever fashion so that it will measure vertically applied loads.

2. Using your hand, gently apply lateral and axial loads to the beam. Do your best to not apply vertical bending loads to the beam.

3. Compare the sensitivity of the quarter bridge to that of either the half bridge or full bridge to these extraneous loads. Which configuration seems to work the best overall? Explain why this is so. Which extraneous loads are "rejected" by which configurations? Explain why. (Note: The steps of this paragraph are more easily said than done. The point is that the quarter bridge is more sensitive to off-axis loads than either the half bridge or full bridge. If you have difficulty demonstrating this, don't stress out.)

6.2 Temperature Sensitivity

Using a quarter bridge configuration, apply a load to the beam and measure the voltage output.

Apply heat uniformly to the beam (140 F max) near the gages and observe any changes in the voltage output. Do not apply heat to the gages directly! Why does the voltage (which corresponds to the measured strain) change? Repeat the procedure using a full-bridge configuration. Compare the sensitivity of the quarter bridge and full bridge configurations to temperature variations. How would you expect the half bridge to behave?

Part 7: Dynamic Characteristics

Up to this part of the lab, you have examined the static characteristics of the strain-gage bridge. Now you will modify the VI that you developed for Temperature Measurement and First-Order Dynamic Response to measure the dynamic characteristics of a signal from the strain-gage bridge. You will excite the dynamics of the cantilever beam by plucking it. You will use your VI to display the response of the vibrating beam.

7.1 Differential Model

The first mode of vibration of the cantilever beam can be modeled using a simple mass-spring-damper model. This model results in a second-order differential equation that describes the dynamics of the system. You will observe that the response of the strain –gages to an initial deflection is a damped sinusoid. This is the expected response for a second-order system. Using acquired data; you will compute and display values for the damping ratio and the natural frequency of the first vibrational mode of the beam.

7.2 Damping Ratio and Damped Natural Frequency

Figure 5 shows the response of a second-order system to an initial condition. This response plot will be used to define the damping ratio and the natural frequency for this system. From the amplitudes x1and xn, the damping ratio can be calculated using the following expression:

**Figure 13:** equation (8)

The damped natural frequency can be determined by measuring the period of the damped oscillations. An accurate measurement of the period T is obtained by considering several periods:

**Figure 14:** equation (9)

**Figure 15:** equation (10)

Once and are known, the natural frequency of oscillation can be calculated by

**Figure 16:** equation (11)

**Figure 17:** Second-order system Response

7.3 Programming Exercise

You will modify your VI to automate the calculation of and for the cantilever beam.

1. Your TA will direct you to a subVI named “Damping Ratio” that is has been started for you.

2. To place the subVI on your block diagram, open the functions palette and select Select a VI…

3. Connect the Data wire from the DAQ Assistant to the Measured Data input of the Damping Ratio subVI.

4. Open the block diagram by double-clicking the subVI icon. (The block diagram is shown in Figure 18.)

5. You will need to modify the block diagram for the VI to work. Refer to equations 8-11 as you design a block diagram that will calculate the Damping Ratio of your signal.

**Figure 18:** Damping Ratio SubVI Block Diagram

6. You will notice that the necessary inputs for the subVI are placed along the left of the block diagram. The outputs are placed along the right of the block diagram. The Peak Detector VI is used to locate the peaks (or valleys) of the signal. The locations of the peaks are given as indexed data points. For example, if the first peak occurs in between the sixth and seventh data points, the location will be 6.5. The time at which this peak occurs can be calculated using the sample rate.

7. For more information about the Peak Detector or other VIs, press Ctrl+H to activate the context help window.

(Hint: you may want to use the index array function to pull values such as x1, xn, t1 and tn out of the amplitude and location arrays.)

This exercise is meant to give you a feel for the possibilities for data acquisition, processing, and display using computer software tools. Be sure to save a copy of your data for plotting. You will need it for your writing assignment.

Part 8: Design of a Force Transducer

For your course project, you will design a force transducer for measuring the thrust of a model rocket motor. This exercise is intended to help you initiate the design process. You may assume that the maximum thrust that will be sensed by your transducer is 5 lb.

Cantilever-Beam Force Transducer

Assume that you will use a cantilever-beam configuration similar to what was used in this lab. A full strain-gage bridge will be used. The width of two gages side-by-side is 1/2 inch. A good design will result in 2000 microstrains under the maximum load and will have a natural frequency above 100 Hz. Meeting both of these objectives might not be possible with a cantilever-beam design. You are to determine the thickness of the cross section where the gages are to be mounted and the distance from where the motor is mounted to the center of the gages (i.e., the length of the cantilever).

Spreadsheet Analysis

Develop a spreadsheet to perform your calculations. A spreadsheet will simplify the evaluation of different designs. This analysis will be a good starting point for the analysis of your project. Note: Your project will involve different loads and different dimensions than the transducer used in this lab. The calculations, however, are the same.

Lab Report

For this lab you will write the Results and Discussion of Results sections of a full report. As you perform the lab, think about what data should be saved or recorded for presentation and why these data are important. For some of your data, tabulation is sufficient (e.g., calculated strain vs. measured strain). Other data should be recorded using LabVIEW (e.g., dynamic response of strain when the beam is plucked). Make sure that you include your thoughts about the results you obtained and why they are important. Discuss their agreement with your expectations.

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labview Algorithms for ECG Signal Analysis

2010-09-25T07:58:00.001-07:00

Summary: In order to accurately detect the heart rate in the ECG signal, filter banks analysis in Matlab is used on the filtered ECG signal. This algorithm can also be used to detect abnormalities in the ECG signal. LabVIEW is then used to notify the user of the results. This includes displaying the heart rate and alerting the user if an abnormality is detected.

ECG Signal Analysis: Abnormality Detection

The flowchart below breaks down the tasks needed to accomplish signal analysis in greater detail. Both LabVIEW and Matlab were used to accomplish these tasks.

**Figure 1**
Signal Analysis Flow Chart

Matlab Implementation of Filter Banks Analysis

Many scenarios deal with signals which contain specific energy distributions in the frequency domain. For example, with regard to the ECG, a significant proportion of the energy from the QRS complex extends to a frequency of 40 Hz, and even more if the Q, R, and S waves have very sharp morphologies. The P and T waves, in general, have a significant proportion of their energy only up to 10 Hz. Thus, a strategic approach to detecting heartbeats is to analyze different sub-bands of the ECG, rather than just the output of one filter which maximizes SNR of the QRS.

In this filter bank analysis technique we use 5 sub-bands, each one has bandwidth 5Hz. The ECG signal is processed by those 5 sub-band filters and downsampled. The downsampled signal is

**Figure 2**

A variety of features which are indicative of the QRS complex can be designed by combining sub-bands of interest from l = 0, 1, … , M-1. For example a sum-of-absolute values feature P1 can be computed using sub-bands 1, 2, and 3.

**Figure 3**

P1 has a value which corresponds to the energy in the frequency band [5.6, 22.5] Hz. Similarly, P2 and P3 can be computed using sub-bands {1, 2, 3, 4}, and {2, 3, 4}, respectively, and these values are proportional to the energy in their respective sub-bands. Heuristic beat detection logic can be used to incorporate some of the above features which are indicative of the QRS complex.

Figure 4 gives an overview of the sequential levels in the beat detection algorithm. The goal of the detection algorithm is to maximize the number of true positives (TP’s), while keeping the number of false negatives (FN’s) and false positives (FP’s) to a minimum. Since it is not possible to arrive at this goal using one simple detector, multiple detectors with complementary FN’s and FP’s performances are simultaneously operated and the results of each fused together to arrive at an overall decision. The advantage of this strategy is that multiple features which are indicative of the QRS complex can be used to detect beats.

**Figure 4**
Beat Detection Algorithm Schematic

A one-channel beat detection block is described in Figure 5.

**Figure 5**
Schematic of One-Channel Beat Detection Block

The detection strength Ds of an incoming feature (e.g., P1, P2, P3) is determined by comparing with the signal and noise levels (SL and NL, respectively):

**Figure 6**

If a feature’s value is less than NL then Ds is limited at 0, and if it is above SL then DS is limited to one. When a feature has a greater than a specified threshold (preset between zero and one) it is classified as a signal peak and the signal history is updated with the feature’s value. If the feature has a smaller than the threshold it is classified as a noise peak and the noise history is updated with the feature’s value.

The detail operations of each level can be found in the paper by V. Afonso et al (ECG beat detection using filter banks, IEEE Trans. Biomed. Eng. 46: 1999). Also, download and runECGmain.m to test out filter banks. Supplementary mfiles: nqrsdetect.m, ECGSigProc.m, t.mat, x.mat.

LabVIEW Programming for Signal Processing and Interfacing

LabVIEW is used to not only process and acquire the signal, but to also develop an user friendly interface that automatically alerts the user to any abnormalities or arrhythmias detected. This particular ECG is developed to detect left ventricular hypertrophy and old myocardial infarction. It will also automatically display the heart rate and whether or not the patient’s heart rate is within a healthy range.

**Figure 7:** This view of the the GUI shows a normal, healthy ECG signal. The indicator lights (6) will change color depending on the diagnosis. Additionally, the green trace turns red if any abnormality or arrhythmia is detected.
The Graphical User Interface

The user interface consists of several useful features. First, there is a text input for the patient’s name to avoid confusion in the clinic. Second, the ECG chart is automatically updated with the name and live signal data. Third, the user can control the data acquisition periods. The heart rate and any abnormalities are displayed in the patient report. The warning lights to the left of the patient report correspond to the current condition of the patient’s ECG. If everything is normal, the green light is on and the signal is green. If an arrhythmia is detected, the yellow light turns on and a warning message will appear in the patient report that indicates the type. Similarly, the orange light turns on if an abnormality is detected and a warning message appears in the report. For either of these, the ECG trace turns red in the chart window. A red light indicates that no recognizable signal is being detected. This can mean that the patient has flat-lined, that the circuit board is not powered or that the leads are not connected properly. As a result, the warning message will ask the user to check the lead and power connections. Finally, the form can be reset between patients. You can download our VI here to test it out.

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labview Multicore Programming

2010-09-25T07:56:00.001-07:00

Summary: The creation of this content was supported in some part by NSF grant 0538934.

If you have written parallel programs in G and have a multicore computer, CONGRATULATIONS!!! You have been successfully developing interactive parallel programs that execute in multicore PC processors.

**Figure 1:** Interactive Multicore G Program

The following sections discuss some multicore programming techniques to improve the performance of G programs.

Data Parallelism

Matrix multiplication is a compute intensive operation that can leverage data parallelism. Figure 2 shows a G program with 8 sequential frames to demonstrate the performance improvement via data parallelism.

**Figure 2:** Data Parallelism

The Create Matrix function generates a square matrix based of size indicated by Size containing random numbers between 0 and 1. The Create Matrix function is shown in Figure 3.

**Figure 3:** Creating a Square Matrix

The Split Matrix function determines the number of rows in the matrix and shifts right the resulting number of rows by one (integer divide by 2). This value is used to split the input matrix into the top half and bottom half matrices. The Split Matrix function is shown in Figure 4.

**Figure 4:** Split Matrix into Top & Bottom

TABLE 1
Sequence Frame	Operation Description
First Frame	Generates two square matrices initialized with random numbers
Second Frame	Records start time for single core matrix multiply
Third Frame	Performs single core matrix multiply
Fourth Frame	Records stop time of single core matrix multiply
Fifth Frame	Splits the matrix into top and bottom matrices
Sixth Frame	Records start time for multicore matrix multiply
Seventh Frame	Performs multicore matrix multiply
Eighth Frame	Records stop time of multicore matrix multiply

The rest of the calculations determine the execution time in milliseconds of the single core and multicore matrix multiply operations and the performance improvement of using data parallelism in a multicore computer.

The program was executed in a dual core 1.83 GHz laptop. The results are shown in Figure 5. By leveraging data parallelism, the same operation has nearly a 2x performance improvement. Similar performance benefits can be obtained with higher multicore processors

**Figure 5:** Data Parallelism Performance Improvement

Task Pipelining

A variety of applications require tasks to be programmed sequentially and continually iterate on these tasks. Most notably are telecommunications applications require simultaneous transmit and receive. In the following example, a simple telecommunications example illustrates how these sequential tasks can be pipelined to leverage multicore environments.

Consider the following simple modulation - demodulation example where a noisy signal is modulated transmitted and demodulated. A typical diagram is shown in Figure 6.

**Figure 6:** Sequential Tasks

Adding a shift register to the loop allows tasks to be pipelined and be executed in parallel in separate cores should they be available. Task pipelining is shown in Figure 7.

**Figure 7:** Pipelined Tasks

The program below times the sequential task and the pipelined tasks to establish its performance improvement when executed in multicore computers.

**Figure 8:** Task Pipelining Program Example

Figure 9 shows the results of running the above G program in a dual core 1.8 GHz laptop. Pipelining shows nearly 2x performance improvement.

**Figure 9:** Pipelining Performance Improvement

Pipelining Using Feedback Nodes

Feedback Nodes provide a storage mechanism between loop iterations. They are programmatically identical to the Shift Registers. Feedback Nodes consist of an Initializer Terminal and the Feedback Node itself (see Figure 10).

**Figure 10:** Feedback Node

To add a Feedback Node, right click on the Block Diagram window and select Feedback Node from the Functions >> Programming >> Structures pop-up menu. The direction of theFeedback Node can be changed by right clicking on the node and selecting Change Direction.

**Figure 11:** Feedback Node Direction

The diagram shown in Figure 12 is programmatically identical to the diagram in Figure 7.

**Figure 12:** Pipelining with Feedback Node

Similarly, the diagram in Figure 13 is programmatically identical to that in Figure 8.

**Figure 13:** Pipelining Tasks with Feedback Nodes

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labview Sound Recorder

2010-09-25T07:55:00.001-07:00

Summary: Various types of filters can be used to improve the quality of an audio signal. In the Sound Recorder Lab, experimenters use a Low Cost National Instruments USB Data Acquisition device to acquire sound data from a microphone element. LabVIEW software is then used to implement a band-pass filter that can act to clarify the audio signal. The Sound Recorder Lab is designed as a labratory or at-home experiment.

Sound Recorder Using National Instruments Low Cost Data Acquisition

Introduction

Any signal, including a sound wave, can be thought of as the sum of different frequency sine waves (where each frequency wave has a specific amplitude and phase angle). In addition to the frequencies we expect to see in a signal, some undesirable frequencies may also be present.

Filters allow us to select which frequencies we care about, and discard certain frequencies that are undesirable (such as noise). Various types of filters exist including low-pass, high-pass, band-pass, etc. There are also many different ways of constructing each filter type with each filter implementation having its own specific characteristics. More information on filters will be provided in the theory section.

In this exercise, the experimenters will use a microphone element to convert a sound wave into an electrical signal. This signal will then be digitized using a Low Cost USB DAQ device. Finally, a LabVIEW application will be constructed to filter undesired frequencies from the signal and play the clarified signal back.

Pre-Lab Assignment

1) In your own words, describe what an ideal band-pass filter transfer function would look like. You may research this by searching for “ideal band pass filter” in google, or using a textbook.

2) When filtering an audio signal, what frequencies must be preserved? Do some research to determine the frequency range produced by the human voice, and the audible range of the human ear.

3) Elaborate on some possible sources of undesirable frequencies in a signal. Where does signal noise come from?

4) Become familiar with the National Instruments USB 6008 and 6009 data acquisition devices. These datasheets are available at www.ni.com.

Theory

Electret Microphone Elements

One way to convert sound pressure waves into an electrical signal is using an electret microphone element. A picture of such an element is shown below:

**Figure 1:** Electret Microphone Element

Inside the electret microphone element, a dielectric material is made to hold a permanent charge. When the element vibrates, the internal capacitance changes and an electrical signal is produced. A variety of additional components complete the microphone element circuitry by adding a small amplifier to the output.

Band Pass Filters

The Fourier Transform tells us that it is possible to think of any signal as being composed of various frequency sine waves (with each frequency having an associated amplitude and phase angle)

Imagine that you have just used a microphone to convert a sound wave into an electrical signal. If the sound wave consisted only of a human voice, then only the frequencies that human vocal chords can produce should be present. Therefore, the overall signal should roughly be composed of frequencies between 80 Hz and 1.2 kHz.

Unfortunately, when playing back your audio signal, you may find that it does not sound very good! Perhaps the lights in your room added some 60 Hz electrical noise to the signal that shouldn’t have been there. Maybe the wind was blowing on your microphone, causing the signal acquired to be fuzzy-sounding. There are an enormous number of factors that could affect your sound signal.

Using a filter can help clarify the signal so that it sounds clear once again. Since you know that any frequencies outside of the 80 Hz – 1.2 kHz range are obviously noise, you can attempt to attenuate these frequencies as much as possible. Specifically, a band-pass filter can be used to accomplish this objective.

An ideal band-pass filter will completely attenuate any signals outside of a desired range (known as the passband). In the real world, it is impossible to construct an ideal filter, but with a large enough circuit or complex digital filtering it is possible to obtain a fairly sharp cutoff.

Remember, all filtering is essentially “frequency selection”. By filtering a signal, we are attempting to “choose” which frequency components can pass through and which we want to discard.

Hardware and Software Required

1. 10 Ohm resistor

2. 4.7 uF capacitor

3. Electret microphone element

4. National Instruments Low Cost USB DAQ

5. LabVIEW 8.2 software (LabVIEW 8.0 will work as well)

Laboratory Exercise

During this exercise, the experimenter will acquire a sound signal from an electret microphone element. This sound signal will then be run through an optional band-pass filter and played back using speakers.

When completed, the completed sound recorder front panel will resemble the following:

**Figure 2:** Completed Sound Recorder Front Panel in LabVIEW

1) Connect the following circuit to the Low Cost USB DAQ as shown. The microphone element can be purchased cheaply at Radio Shack, etc. Note that the +5V power supply can be obtained directly from the National Instruments USB 6008 or 6009 devices.

**Figure 3:** Electret Microphone Circuit

2) Using an event structure in LabVIEW, replicate the following block diagram for the “playback” event. Note that the “Bandpass Filter” Boolean control allows the user to play back the filtered or original signal.

**Figure 4:** Sound Recorder Block Diagram Showing Playback Event

3) Using the DAQ Assistant Express VI, complete the block diagram for the “record” event as indicated below:

**Figure 5:** Sound Recorder Block Diagram Showing Record Event

4) Experiment with the sound recorder VI by recording a simple voice message. Attempt to play back both the original and filtered signals. Modify the filter cut-off frequencies and see how narrow you can make the passband before the played back signal is difficult to decipher.

Post-Lab Questions

1) What sample rate did you use when recording your sound signal? Explain why you chose this rate and what issues could occur with too low or high of a sample rate.

2) Could you have used any other filter types to clarify the sound signal? Would a low-pass, high-pass, or other filter have accomplished the same objective?

3) How can you tell if high frequency noise is present in your sound signal without playing it back? Before filtering the signal, how could you have determined what frequencies the signal contained?

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labview simulated microphone array

2010-09-25T07:53:00.001-07:00

Labview VIs Used in Simulation

Top Level Diagram and Overall Organization

The figure below is a top level diagram of a six microphone array simulation VI. This VI is named "Multiple Frequency Simulation" Starting at the left with the signal generation, the code flows to the right. This top VI is made out of smaller VIs that each perform a specific function. Detailed descriptions of each of these can be found in the following paragraphs.

**Figure 1:** Top Level Diagram
Top Level Diagram (Multiple Frequency Simulation)

Simulated Signal Generation VI

**Figure 2:** Simulate Signal Icon
Simulate Signal Icon

Signal Generation VI

The above icon corresponds to the Signal Generation VI. At the far left are four signal generation VIs. Each of these VIs produces six different signals that simulate the delayed signals that each of the microphones would hear in real life. These six signals are then bunched together in a cluster (pink wire) to keep the amount of wires down. The three inputs to this VI are the frequency of the sinusoid desired, the direction of the signal, and the distnace between microphones (9.9 cm in this example). The user sets these paramaters on the front panel and is free to change them at any time.

Our VI uses the formula discussed in previous modules that relates the time delay of signals to the distance between microphones. Using the desired angle that the signal is coming from, the speed of sound (C), the distance between microphones, and a conversion between degrees and radians, the VI first computes the time delay between microphones. Becuase we are interested in the delay between the first microphone and all others, the amount of delay is multiplied by zero for the first mic, one for the second mic, two for the third, and so on. These delays are then used as phase inputs to the six identical Simulate Signal VIs. Adding phase to a periodic signal like the sinusoids being used here has the same effect as delaying the signal in time. Finally the six signals are merged together into a cluster so that they will be easier to deal with as a group in other VIs.

**Figure 3:** Simulate Signal Block Diagram
Simulate Signal Block Diagram

In this simulation we simulate four different frequencies and directions. Once we have the data of these four separate signals, we sum the signals on each of the channels to get the final output on each of the microphones. Doing so leaves us solely with what we would hear on the six microphones if this were set up in real life. If more signals are required for future testing or other applications, onc can simply copy and paste the signal generation VI and then sum it with the other signals.

Once this signal is comple, we move to the right in our VI. From here we branch into two separate areas. Following the pink line up takes us to where we calculate the angle that a given frequency is coming from,and to the right is where we perform the calculations to listen to signals from a given direction.

Frequency and Spatial FFT Computation VIs

**Figure 4:** 1rst FFT Icon
1rst FFT Icon (Time - Frequency)

1rst FFT VI

In order to listen to a certain direction we first need to transform our signals from the microphone in to the frequency domain. To do this we made the "1rst FFT" Vi (see icon above). This sub-VI is fairly simple so there is no need to show the block diagram. It takes the FFT of each of the six channels channels, transfroming the data from the time domain to the freqeuncy domain For simulation, we looked at 1 second samples at 4000hz. This VI then takes the FFTs of the six 4000 element long signals from the simulated microphones.

**Figure 5:** Six Point FFT Icon(Frequency - Spatial)
Six Point FFT Icon (Frequency - Spatial)

Six Point FFT VI

Moving to the right after the first FFT, we find the "6 pt FFT" VI. This VI is used to transform our frequency domain data into the spatial domain. This VI needs to run for every frequency of interest, so in the top level diagram this VI is found inside of a for loop. The user can control what range of frequencies are of interest on the front panel of the main VI, and the loop will run for those frequencies.

Inside this VI, the complex frequency coefficient for the given frequency is extracted from the array for each microphone channel. These six values are then concantonated into a lengh six array. Next, the array is zeropadded to a user specified length (we used 256 in our simulation) so more resolution can be observed in the resulting spatial transform. Finally, the FFT is performed on these values transforming them into the spatial domain. With the data in the spatial domain we are easily able to figure out the magnitude of any frequency from any direction in our 180 degrees of interest. How to do this can be found in the magnitude VI.

**Figure 6:** Six Point FFT block Diagram (Frequency - Spatial)
Six Point FFT block Diagram (Frequency - Spatial)

Coefficient, Angle, and Magnitude Calculation VIs

**Figure 7:** Coefficient Calculator
Coefficient Calculator

Coefficient Calculator

Now that the signals have been transformed to their spatial representations, the correct coefficients to construct the signal of interest must be computed. To do this we created the Coefficient Calculator VI.. This VI takes in the angle of interest, a frequency, the number of microphones, a constant factor (determined by the setup of the microphones and the amount of zero padding...in this example it is -93), and our original sampling frequency to compute the correct coefficient to look at for the given angle. Once this coefficient is found, we extract the value at that coefficient and append it to our ouptut array as the value at that given frequency. Below is the block diagram for this VI. It consists of a basic formula node and some logic on the front and back to make sure that the calculations are correct when working with angles from negative directions.

Because this formula is dependent on the frequnecy of interest, we are required to run this VI for every frequency we are intereseted in. In order to do this, we put this VI inside a for loop that is controlled by our frequency range. Any coefficient for frequencies outside of this range are simply given a value of zero. The array modules ouside of this for loop are used to do just that. They append arrays with value zero on the front and back of the output of the for loop to return our vector to its original size. From here, we run this vector through a few multiplies to amplify the differnce between the coefficients with high and low magnitudes, and finally we inverse FFT it to get our output array. This array represents a signal in the time domain and is graphed on the front panel along with a graph of its frequency components. We also included a small VI that will play the output waveform on computer speakers. This VI uses a matlab script and requires the user to have matlab 6.5 or earlier.

**Figure 8:** Magnitude Graph of Coefficients After Spatial FFT
Magnitude Graph of Coefficients After Spatial FFT

**Figure 9:** Magnitude Calculation VI
Magniutde Calculation VI

Magnitude Calculation

If we go back to the branch in the pink wire immediatly after the signal generation VIs and move upwards instead of to the right, we come across the code that is used to calculate the angle at which the largest magnitude signal of a given frequency is approaching the array. Another "6 pt fft" VI is used, but this one is slightly modified. It also includes the initial FFTs of all 6 channels. We grouped these two VIs together because ony one spacial FFT is being computed (that at the frequeny of interest).

The resulting vector of the previous six point FFT is immediately used as the input to the Magnitude Calculation VI. The vecor of spatial coefficients from the "six pt FFT" vis are complex, so this VI is used to calculate the magnitudes of the coefficients so the maximum coefficient can be found. The output of this VI is also used to create a visual representation of what direction the specified frequency is coming from. Below is a graph of the magnitude of the coefficients of the spatial FFT. As discussed before, we see the peak correspoinding to the incoming direction and the smaller ripples to each side.

**Figure 10:** Magniutde of Coefficients after Spatial FFT
Magnitude Graph of Coefficients After Spatial FFT

EXAMPLE 1: Magnitude Graph Example

As we can see in the previous figure, the magnitude of the spatial FFT is greatest around coefficient 82. There are also smaller ripples that die off around each end. This graph tells us that the the direction that the given frequency is coming from corresponds to the angle represneted by coefficient 82. To figure out what angle this was, we would use our Coefficient to Angle VI.

Calculation of Angle

Finally, we isolate the index of the maximum angle and use it to compute the angle of incidence. Using the same formula used in the Coefficient Angle Calculator, the Coefficient to Angle VI deduces the angle of incidence. This VI uses the same formula found in the Coefficient Angle Calculator, but is arranged differently so that we can find the angle based on the coefficient instead of the coefficient based on the angle. Once the VI computes this value, the result is output on the front panel.

Links to Labview Code

Code Remarks

Overall, the code involved in this method of array signal processing can be broken up into smaller parts that are easy to understand. By combining these smaller parts we are able to create an upper level VI that performs a complicated task that would be difficult to get working using other methods. The major problem with this VI is that it requires a large number of calculations. To increase performance (without upgrading computers) one could decrease the frequency range of interest, or they could lower the amount of zeropadding. They could aslo look at a smaller time period.

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labview Speaker-Air-Microphone (SAM) Channel Characterization

2010-09-25T07:50:00.001-07:00

Summary: A speaker and microphone connected to a computer serve as an excellent communications channel because the transmitted information is audible. Listening to the channel while making parameter adjustments and viewing plots builds additional insight into the specific modulation scheme under study. This project describes how to characterize a "speaker-air-microphone" (SAM) channel for use in subsequent projects that use bandpass modulation techniques.

TABLE 1
	This module refers to LabVIEW, a software development environment that features a graphical programming language. Please see the LabVIEW QuickStart Guide module for tutorials and documentation that will help you:
	• Apply LabVIEW to Communications / Signal Processing
	• Get started with LabVIEW
	• Obtain a fully-functional evaluation edition of LabVIEW

NOTE:

Visit LabVIEW Setup to learn how to adjust your own LabVIEW environment to match the settings used by the LabVIEW screencast video(s) in this module. Click the "Fullscreen" button at the lower right corner of the video player if the video does not fit properly within your browser window.

Summary

A speaker and microphone connected to a computer serve as an excellent communications channel because the transmitted information is audible. Listening to the channel while making parameter adjustments and viewing plots builds additional insight into the specific modulation scheme under study.

The passband limits and bandwidth of the SAM channel (Speaker-to-Air-to-Microphone) must be characterized to effectively choose modulation parameters such as carrier frequency, bit rate, and signal pulse shape. This project describes how to characterize a SAM channel for use in subsequent projects that use bandpass modulation techniques.

Objectives

Measure the passband limits and bandwidth of a speaker-air-microphone (SAM) channel
Learn how to use LabVIEW Express subVIs for audio and power spectrum measurement

Deliverables

Summary write-up of your results
Hardcopy of all LabVIEW code that you develop (block diagrams and front panels)
Any plots or diagrams requested

NOTE:

You can easily export LabVIEW front-panel waveform plots directly to your report. Right-click on the waveform indicator and choose "Export Simplified Image."

Setup

LabVIEW 8.5 or later version
Computer soundcard
Speaker
Microphone

Textbook Linkages

Refer to the following textbooks for additional background on the project activities of this module; see the "References" section below for publication details:

Carlson, Crilly, and Rutledge -- Ch 9
Couch -- Ch 6
Haykin -- Ch 1
Haykin and Moher -- Ch 8
Lathi -- Ch 3
Proakis and Salehi (FCS) -- Ch 5
Proakis and Salehi (CSE) -- Ch 4
Stern and Mahmoud -- Ch 2

Prerequisite Modules

If you are relatively new to LabVIEW, consider taking the course LabVIEW Techniques for Audio Signal Processing which provides the foundation you need to complete this project activity, including: block diagram editing techniques, essential programming structures, subVIs, arrays, and audio.

Introduction

Figure 1 illustrates the speaker-to-air-to-microphone (SAM) channel.

**Figure 1:** Speaker-air-microphone channel

The magnitude frequency response of the SAM channel can be easily measured by applying white noise to the speaker, recording the result, and calculating the power spectrum of the of the recorded signal. White noise contains equal contribution from all frequencies; therefore, any deviation from a flat power spectrum at the channel output must be the response of the channel.

The computer sound card forms part of the channel, as well. Sound card microphone inputs are often AC-coupled and do not pass DC. Microphone inputs are sometimes intentionally lowpass filtered to reduce high-frequency hiss, and all soundcard inputs include lowpass filters to serve as anti-aliasing filters.

Most modern sound cards are full duplex, meaning that they can simultaneously generate and record sound -- a necessary feature for this project. Older half duplex sound cards can still be used but require two computers, one to transmit the sound and another to receive the sound.

Procedure

White noise source

Create a white noise sound source to serve as the SAM channel excitation:

Use the "Express | Output | Play Waveform" Express VI to generate audio for the speaker.
Use the "Express | Signal Analysis | Simulate Signal" Express VI to generate uniform white noise in the range -1 to +1.
Set the Express VI parameters to generate a signal with 44.1 kHz sampling frequency and duration of 10 seconds.

Refer to the video of Figure 2 for LabVIEW coding techniques for the white noise source.

**Figure 2:** [video] LabVIEW coding techniques for white noise source

Recording device

Create a device to record the microphone signal:

Use the "Express | Input | Acquire Sound" Express VI configured for the same sampling frequency as the white noise source (44.1 kHz) and a 0.1 second duration.
Monitor the recorded signal as a time-domain plot.
Use the "Express | Signal Manipulation | Extract Portion of Signal" Express VI to discard a user-specified number of samples at the beginning of the waveform to account for transient startup.

Refer to the video of Figure 3 for LabVIEW coding techniques for the recording device.

**Figure 3:** [video] LabVIEW coding techniques for recording device

Power spectrum display

Create a graphical indicator to display the power spectrum of the recorded microphone signal:

Use the "Express | Signal Analysis | Spectral Measurements" Express VI configured for power spectrum measurement.
Create a "Waveform Graph" indicator to display the power spectrum. Calibrate the Y-axis in decibels and the X-axis in hertz. Use logarithmic mapping for the frequency axis.
Enclose the power spectrum measurement and recording-related subVIs in a for-loop structure (N=80). This arrangement takes 80 short recordings, calculates the power spectrum for each, and averages them together to improve the estimated frequency response of the channel. The value of N is chosen to allow the recording/measurement process to complete before the white noise source stops.

NOTE:

The white noise source and recording/measurement sections of the overall VI must be kept separate from each other to allow them to operate in parallel.

Refer to the video of Figure 4 for LabVIEW coding techniques for the power spectrum display.

**Figure 4:** [video] LabVIEW coding techniques for power spectrum display

SAM channel physical setup

Set up the hardware for the SAM channel. Place the microphone within a few inches of the speaker. Adjust the soundcard setup and volume controls to generate and record sound to satisfy two goals:

Signal applied to sound card lies within the range ±1 with no clipping (saturation)
Signal recorded from sound card fills as much as possible the range ±1 without clipping

The video of Figure 5 for LabVIEW coding techniques for the power spectrum display.

**Figure 5:** [video] Adjust the audio settings in Windows XP

SAM channel measurements

Apply the white noise source to the SAM channel and record its response for 5 to 10 seconds. Also record the noise floor of the channel; use the identical speaker/microphone arrangement but do not generate any sound (set the "Noise Amplitude" parameter of the noise source to zero).

Subtract the noise floor measurement from the white noise measurement. The result should be approximately zero dB outside the passband region of the SAM channel.

Make hardcopy of your two measurements on the SAM channel as well as the measurement with the noise floor subtracted (right-click on the "Waveform Graph" indicator and choose "Export Simplified Image"). Annotate the plots with hand-written labels to identify the lower passband limit, the upper passband limit, and the overall bandwidth of the channel.

State your numerical criterion for choosing the passband limits. Half-power frequency, the frequency at which the response drops by 3 dB, is a standard method.

Optional: To Go Further

Your instructor may ask you to complete one or more of the following activities:

If supported by your sound card, change the soundcard recording device to "Stereo Mix" and repeat the two measurements above. This soundcard loopback test helps you distinguish which parts of the frequency response to attribute to the soundcard itself as opposed to the speaker/microphone combination.
Repeat the SAM channel measurement for other combinations of speakers and microphones
Enhance the overall VI to record the white noise source on a first run of the VI and the noise floor on the next run of the VI. Subtract the noise floor from the white noise recording and display the result.

The video of Figure 6 shows how to retain a measurement from one run of the VI to the next.

**Figure 6:** [video] Retain a measurement from one run of a VI to the next

References

Carlson, A. Bruce, Paul B. Crilly, and Janet C. Rutledge, "Communication Systems," 4th ed., McGraw-Hill, 2002. ISBN-13: 978-0-07-011127-1
Couch, Leon W. II, "Digital and Analog Communication Systems," 7th ed., Pearson Prentice Hall, 2007. ISBN-10: 0-13-142492-0
Haykin, Simon. "Communication Systems," 4th ed., Wiley, 2001. ISBN-10: 0-471-17869-1
Haykin, Simon, and Michael Moher, "Introduction to Analog and Digital Communication Systems," 2nd ed., Wiley, 2007. ISBN-13: 978-0-471-43222-7
Lathi, Bhagwandas P., "Modern Digital and Analog Communication Systems," 3rd ed., Oxford University Press, 1998. ISBN-10: 0-19-511009-9
Proakis, John G., and Masoud Salehi, "Fundamentals of Communication Systems," Pearson Prentice Hall, 2005. ISBN-10: 0-13-147135-X
Proakis, John G., and Masoud Salehi, "Communication Systems Engineering," 2nd ed., Pearson Prentice Hall, 2002. ISBN-10: 0-13-061793-8
Stern, Harold P.E., and Samy A. Mahmoud, "Communication Systems," Pearson Prentice Hall, 2004. ISBN-10: 0-13-040268-0

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labview Caller ID Decoder

2010-09-25T07:49:00.001-07:00

Summary: The telephone company's "Caller ID" service provides the calling party's directory information as well as the time and date of the call as an FSK (frequency shift keying) signal between the first and second rings of a telephone call. In this project, develop a complete Caller ID decoder that analyzes an audio recording of the FSK signal to extract the directory and date information for display.

TABLE 1
	This module refers to LabVIEW, a software development environment that features a graphical programming language. Please see the LabVIEW QuickStart Guide module for tutorials and documentation that will help you:
	• Apply LabVIEW to Audio Signal Processing
	• Get started with LabVIEW
	• Obtain a fully-functional evaluation edition of LabVIEW

NOTE:

Summary

The telephone company's "Caller ID" service provides the calling party's directory information as well as the time and date of the call as an FSK (frequency shift keying) signal between the first and second rings of a telephone call. In this project, develop a complete Caller ID decoder that analyzes an audio recording of the FSK signal to extract the directory and date information for display.

Objectives

Describe the Caller ID standard at the signal level
Express a set of Caller ID information as a series of message bytes
Decode by hand a Caller ID data block bitstream
Implement a complete Caller ID decoder application in LabVIEW

Deliverables

Summary write-up of your results
Hardcopy of all LabVIEW code that you develop (block diagrams and front panels)
Any plots or diagrams requested

NOTE:

You can easily export LabVIEW front-panel waveform plots directly to your report. Right-click on the waveform indicator and choose "Export Simplified Image."

Setup

LabVIEW 8.5 or later version
Modulation Toolkit 4.0 or later version
Computer soundcard
Speaker

Textbook Linkages

Refer to the following textbooks for additional background on the project activities of this module; see the "References" section below for publication details:

Carlson, Crilly, and Rutledge -- Ch 14
Couch -- Ch 5
Haykin and Moher -- Ch 7
Lathi -- Ch 13
Proakis and Salehi (FCS) -- Ch 10
Stern and Mahmoud -- Ch 5
Wheeler -- Ch 14 (an excellent reference, provides much detail about Caller ID)

Prerequisite Modules

Introduction

You are no doubt familiar with Caller ID, the telephone company service that provides the name and phone number of your incoming caller. The Caller ID service transmits the calling party'sdirectory information (name and telephone number) as well as the date and time of the call between the first and second ring as a binary FSK (frequency shift keying) signal. ClickCallerID_audio_example.mp3 to listen to a typical Caller ID FSK signal embedded between the first and second ringer pulses.

After successfully completing this project, your LabVIEW application will be able to read audio recordings such as CallerID_audio_example.mp3 and then extract the Caller ID message for display.

Figure 1 illustrates the Caller ID decoder system to be constructed in this project.

**Figure 1:** Caller ID decoder system to be constructed in this project

The process begins with a call placed by the calling party. The telephone company's subscriber line interface card (SLIC) in the telephone company central office (CO) signals thecustomer premises equipment (CPE) -- telephone, modem, CallerID unit, etc. -- with a ringing pulse (90 VRMS, 20 Hz, 2 seconds on, 4 seconds off). The CO repeats the ringing pulse as long as the CPE is on hook, that is, the phone has not been answered. Answering the phone call places the CPE in the off hook state, and the CO connects the calling party to the CPE. The terms "on hook" and "off hook" refer to the position of the ear piece or handset in early telephone equipment. The SLIC detects the CPE hook state by the amount of DC current in the customer loop: zero current means on-hook, and non-zero current (about 10 to 20mA) indicates off-hook. The Caller ID FSK signal is transmitted between the first and second ringing pulses provided the CPE is on-hook. For this reason, the interface circuit indicated in Figure 1 must be AC-coupled to the SLIC to prevent drawing DC current.

FSK Signal

Caller ID uses the Bell 202 modem standard:

Binary FSK (two-level frequency shift keying)
Symbol rate: 1,200 symbols/second
Bit rate: 1 bit per symbol
Logic 0 ("space"): 2,200 Hz
Logic 1 ("mark"): 1,300 Hz

Caller ID Message Format

The complete Caller ID message is less than eight tenths of a second in duration. Listen to reduced_tempo_FSK.wav, a reduced tempo version of just the FSK signal; the signal is stretched out in time by a factor of 4, but the original frequencies are preserved.

Hopefully you could discern three distinct regions of the signal:

Alternating 1's and 0's for 250 ms -- this channel seizure region "wakes up" the demodulator and gives the symbol synchronization subsystem enough time to generate pulses synchronized to the FSK signal
Constant 1's for 150 ms -- this steady mark region separates the channel seizure region from the datablock; the relatively long interval of constant "mark" level ensures that the first "space" symbol of the data block can be easily detected
Message bits for approximately 350 ms (the total time varies depending on the caller information) -- this data block region contains the Caller ID information

The message consists of a sequence of 10-bit frames. A start bit of value 0 begins the frame, 8 bits of information follow, and the frame concludes with a stop bit of value 1. The 8 information bits begin with the LSB (least significant bit) and end with the MSB (most significant bit). The information bits form one byte.

The data block message bytes are organized as follows:

Message type, 1 byte -- 0x80 (i.e., hexadecimal 80) for Multiple Data Message Format (MDMF), or 0x04 for Single Data Message Format (SDMF). Number-only Caller ID service uses SDMF, and number-plus-name service uses MDMF. Number-plus-name service is much more common today, and is used in this project.
Length of complete message, 1 byte -- This length value excludes the single-byte checksum at the end of the data block
Data type, 1 byte -- 0x01 = date and time, 0x02 = phone number, 0x04 = number not present, 0x07 = name, and 0x08 = name not present
Length of data, 1 byte
Data bytes, variable number according to length
Repeat Items 3, 4, and 5 as needed to complete the Caller ID message
Checksum, 1 byte -- Add this to the modulo-256 sum of all the previous bytes in the message, including the message type and message length; a zero result indicates no errors detected

FSK Demodulator

The phase-lock loop (PLL) can easily discern the change in frequencies of an FSK signal. The LabVIEW Modulation Toolkit provides a PLL component that serves as an FSK demodulator for this project. Refer to the theory-of-operation screencast video in cid_Demodulator.vi learn how to use this PLL.

Timing Recovery

The Caller ID message symbol rate is 1,200 symbols per second. The baseband output of the FSK demodulator must be compared to a threshold and sampled at the symbol rate to recover the serial bit stream. The timing recovery system ensures that the thresholded demodulator output is sampled near the midpoint of the symbol. In this project a "local oscillator" produces a squarewave at a nominal frequency of 1,200 Hz. The local oscillator phase is synchronized to the thresholded FSK demodulator output. That is, the local oscillator phase is reset each time a transition is detected on the FSK demodulator output.

Procedure

Download required project files

Download the required project files contained in CID_Decoder_Project_Files.zip; unpack the files to the same folder in which you plan to build your LabVIEW subVIs and top-level application VIs.

The .zip archive contains the following files:

cid_ParseMessage.vi -- Accepts a text string containing the Caller ID data block bytes, and parses the string to extract the Caller ID data fields, i.e., date, time, number, and name; also returns information about the data block itself, namely, message type (SDMF or MDMF), message length, checksum value, and result of checksum calculation. All of the values are returned in a single cluster.
cid_Display.ctl -- Custom front-panel control to display Caller ID data block information contained in the cluster generated by cid_ParseMessage.vi. Follow these steps to place the control on the front panel: Display the front panel, right-click and choose "Select a Control...", and then choose cid_Display.ctl.
CallerID-N.wav -- Two audio recordings of Caller ID signal embedded between the first and second ringer pulses. The recordings include three ringer pulses, are approximately 17 seconds in duration, and are sampled at 44.1 kHz.
CallerID-N_19.2kHz.wav -- The same audio recordings downsampled to 19.2 kHz to produce 16 samples per symbol, the default value for many of the LabVIEW Modulation Toolkit subVIs. Either audio file can be used for this project, although the downsampled versions shorten run times of the Caller ID decoder application.
cid_Recorder.vi -- LabVIEW VI to monitor the soundcard input for ringer activity; when detected, record for a fixed time interval and save to a .wav file. Useful to collect several Caller ID signals automatically. Requires a suitable interface circuit between the telephone wall jack and the computer sound card.

Study the Caller ID signal audio recordings

Open the CallerID-1.wav audio recording in Audacity or an equivalent sound editor to view and listen to the signal. Use the zoom features to study the fine detail of the FSK signal, especially at the transitions between frequencies.

Repeat for the other .wav audio files included in the download distribution.

Are you able to discern any obvious differences among the various audio recordings?

SubVI construction

Possible approaches to analyze the signal generated by the telephone central office include: (1) process each sample as it arrives and generate the decoded message "on the fly," or (2) record the entire signal and then make repeated passes over the recording as needed to extract the message. Real-time implementation requires the former approach, while the latter "off-line" approach is easier to implement as a sequence of subVI calls, and is therefore the method of choice for this project.

Build the subVIs listed below. You may already have some of these available from previous projects.

Demonstrate that each of these subVIs works properly before continuing to the next part. The order in which you build the subVIs does not matter, however, the order presented roughly corresponds to the general processing flow that begins with the audio recording and ends with a collection of bytes.

Isolate and demodulate the FSK signal

Build a top-level VI that isolates and demodulates the FSK portion of the complete Caller ID audio signal. util_GetAudio to load the .wav audio file, and then pass this signal tocid_Demodulator.vi. Connect front-panel controls to the four parameter inputs of cid_Demodulator.vi. Create a mixed-signal waveform chart to display the four signals associated with cid_Demodulator.vi, namely: FSK signal (the original audio signal applied to the demodulator input), baseband signal (the demodulated output), phase error magnitude, and PLL locked.

The Figure 2 screencast video describes how to set up a LabVIEW "Mixed Signal Graph" to plot the waveform data type and Boolean 1-D array on a common timescale, much like an oscilloscope display with multiple analog and digital signals.

**Figure 2:** [video] Set up a "Mixed Signal Graph" to display both "analog" and "digital" signals on a common timescale

Set the demodulator VCO carrier frequency to the average value of the mark and space frequencies of the FSK signal.

Experiment with the remaining demodulator parameters VCO gain (start with values in the range 0.05 to 0.20), phase error LPF cutoff frequency, and comparator threshold for PLL lock to satisfy the following goals:

Baseband signal in FSK region "looks good", i.e., reasonably quick rise time without high-frequency ringing or other non-ideal artifacts
PLL locked is active (T) only during the FSK signal and is inactive during silence and ringer pulses.

Report your four demodulator parameter values and plot the mixed signal graph that demonstrates the ability of your system to identify the time region over which the FSK signal is active.

Sample the baseband signal

Use regen_Sampler.vi to extract the FSK signal region from the baseband signal produced by cid_Demodulator.vi. Note that regen_Sampler.vi is used like a "gating" circuit here: the PLL locked signal is active over the entire time that the FSK signal is detected, therefore regen_Sampler.vi picks out every value from the original audio recording over whichPLL locked is active.

Add the bit sync system to your VI using a zero-crossing comparator, util_EdgeDetector.vi (two instances), regen_BitClock.vi set to 1,200 Hz, and another instance ofregen_Sampler.vi. Use the zero-crossing comparator to convert the FSK signal to a Boolean 1-D array. This signal is not a bitstream yet, but rather serves to identify the beginning of bit intervals. Use one edge detector to produce indicator pulses for each zero crossing of the baseband signal (both rising edges and falling edges). These indicator pulses serve as the synchronization for the bit clock oscillator, which produces a square wave synchronized to the baseband signal. The square wave transitions low-to-high at the beginning of the symbol interval and transitions high-to-low at the midpoint of the signal interval. Use an edge detector to produce indicator pulses at the midpoint of the symbol interval which control when the sampler should take samples from the isolated FSK signal.

Create another mixed signal waveform graph to display the isolated FSK signal, the thresholded version of this signal, the indicator pulses that synchronize the bit clock, the bit clock output, and "actual sampling instants" produced by the sampler.

Study your results to ensure that the demodulated baseband signal is sampled properly.

The Figure 3 screencast video shows how you can conserve front-panel real estate by placing two waveform graphs inside a tabbed control.

**Figure 3:** [video] Conserve front-panel real estate with a tabbed control

Decode the message bitstream

Use a zero-crossing comparator (specifically of the "less than zero" type) to convert the sampled baseband signal to a bitstream. Process this bitstream with cid_DetectStartBit.vi to extract only the data block portion of the bitstream.

Use util_BitstreamToText.vi to convert the bitstream to a sequence of 8-bit values ("string" data type), and then use cid_ParseMessage to convert the text string into an information cluster to be displayed with the custom front-panel control CID Display.

Include front panel indicators for the framing error and text out outputs from util_BitstreamToText.vi. The string indicator can be easily switched from ASCII display to hexadecimal display, as needed: right-click on the front panel indicator and choose either "Normal Display" or "Hex Display."

The framing error indicator should be dark for the entire data block region; some "left over" bits are likely after the data block ends due to the delay until the FSK demodulator's PLL locked output returns to F, and the resulting framing errors may be safely ignored.

Evaluate the completed system

Choose one of the Caller ID signal recordings, and run your VI to decode the FSK signal. When all goes well, the CID Display indicator will show "OK" for both the parsing process and the checksum calculation.

Confirm that the data block values are correctly extracted by manually decoding the hex values in the text string front panel indicator. More specifically, copy the hex values to a piece of paper, and work through the interpretation of each byte. For example, the first byte of the data block is 0x80, which indicates the Caller ID message is of the MDMF (Multiple Data Message Format) type. The next byte is an 8-bit unsigned integer that indicates message length; convert the hexadecimal value to decimal and report this value. Continue in this way to demonstrate that you understand the significance of each byte in the data block.

Try your Caller ID decoder on other signal recordings. Report the CID Display indicator values for each audio signal recording.

References

Carlson, A. Bruce, Paul B. Crilly, and Janet C. Rutledge, "Communication Systems," 4th ed., McGraw-Hill, 2002. ISBN-13: 978-0-07-011127-1
Couch, Leon W. II, "Digital and Analog Communication Systems," 7th ed., Pearson Prentice Hall, 2007. ISBN-10: 0-13-142492-0
Haykin, Simon, and Michael Moher, "Introduction to Analog and Digital Communication Systems," 2nd ed., Wiley, 2007. ISBN-13: 978-0-471-43222-7
Lathi, Bhagwandas P., "Modern Digital and Analog Communication Systems," 3rd ed., Oxford University Press, 1998. ISBN-10: 0-19-511009-9
Proakis, John G., and Masoud Salehi, "Fundamentals of Communication Systems," Pearson Prentice Hall, 2005. ISBN-10: 0-13-147135-X
Stern, Harold P.E., and Samy A. Mahmoud, "Communication Systems," Pearson Prentice Hall, 2004. ISBN-10: 0-13-040268-0
Wheeler, Tom, "Electronic Communications for Technicians," 2nd ed., Pearson Prentice Hall, 2006. ISBN-10: 0-13-113049-8

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LabVIEW 3D Control Simulation using SolidWorks 3D Models

2010-09-25T07:48:00.001-07:00

Summary: Importing a SolidWorks VRML into LabVIEW gives engineers and students the ability to integrate the power of SolidWorks 3D computer aided design with LabVIEW simulation. It is extremely useful for test and simulation applications that require complex visualization of a 3D model. To combine 3D visualization with simulation, you can use a combination of LabVIEW, the Simulation Module and the 3D Picture Control Toolkit.

3D Control Simulation in LabVIEW

The LabVIEW development environment is an ideal platform for prototyping, designing, and deploying control alogithms. As part of the prototyping step, 3D simulation can be used to assist in visualizing both the behavior of the plant and the effects of the control algorithm on the control plant. In this demonstration, the LabVIEW Simulation Module and LabVIEW 3D Picture Control Toolkit are used to simulate the behavior and control of a Quanser Linear Inverted Pendulum.

Download the source code for the Linear Inverted Pendulum 3D Simulation here. This VI was created in LabVIEW 8.2 using the 3D Picture Control Toolit add-on which is not a supported feature.

NEW: An updated version of this 3D demo, using the new supported 3D Picture Control, now ships with the Control Design and Simulation Module and can be found using the Example Finder within LabVIEW.

**Figure 1:** Matrix A,B,C, and D define the controller for the simulation. The Acceration of Gravity and Pendulum Length in centimeters can be cutomized by editing the plant model. The Array of inital states allow the angle, angular velocity, and linear velocity variables to be defined before the simulation is executed. The gains K1, K2, and K3 respresent the gain for the controller.
LabVIEW LIP Simulation Front Panel

**Figure 2:** This simulation utilizes the LabVIEW Simulation Module with the RK1 continuous time solver. The controller is containined inside the 'Insert Control Code Here' box allowing students to design their own unique controller while the plant model and 3D simulaiton model are preserved.
LabVIEW LIP Simulation Block Diagram

When downloading this VI, you must have a minimum of LabVIEW 8.0.1 with the LabVIEW Simulation Module installed, as well as the free 3D Picture Control Toolkit. This 3D simualtion is one of many developed for educational purposes. Other demonstrations include the 3D helicopter, auto suspension, and 3D rotational inverted pendulum.

Creating Your Own 3D Simulations

To create LabVIEW simulations using SolidWorks VRML97 export file format, you will need to create a program that contains three VI’s. The top level VI contains the simulation loop and the state-space matrices information; the first subVI draws and updates the picture in accordance with the simulation parameters; the third and lowest level VI imports and parses the VRML97 file to create objects names and parent/child relationships for each object in your system.

For the lowest level VI, you can load WRL files into LabVIEW using the 3D picture control toolkit VI called Read WRL file.vi, which reads a WRL file and creates the data necessary to render the scene into a picture control. It is located in the Objects Functions Palette. You can add objects to the system using the Insert Into Array.vi. The first step needed in the VRML parser subVI is an indexed for loop with an unbundle.vi followed directly by a bundle.vi and you need to manually name each component of the system for reference. After this is completed, use the Find Object.vi to create object refnums for each object and connect those to a case structure with an identical unbundled/bundle feature to assign parent/child relationships (this allows for the pendulum to move when the base moves and not the other way around).

After you have successfully parsed the VRML97 file, the drawObject subVI can take your object array. Once again use the Find Object.vi to create references for each object you would like move with the movement VI’s such as Translate Object To.vi and Rotate Object To.vi. The drawObject.vi creates the camera and its controller as well as the light source. It also performs the scene rendering and scene drawing.

The top level VI is responsible for the simulation parameters and equations that describe the system. The simulation dynamics can be altered on the fly using the CD Create State-Space Model.vi located in the Control Design Toolkit. This VI allows the user to enter variable formulae into the state-space matrices and then alter the system very quickly by changing the variable values. Combined with the 3D picture control toolkit, you can witness the visual effects of the variable values on the system immediately.

This roughly outlined method describes the basic functions needed to perform a LabVIEW simulation with 3D visualization provided by SolidWorks VRML97 export. The VRML parsing is only necessary because at the time of this writing, SolidWorks software does not provide object names and parent/child relationships with VRML97 export.

Temperature Running Average VI

2010-09-25T07:47:00.001-07:00

Summary: Use shift registers to perform a running average.

EXERCISE 1

Complete the following steps to modify the Temperature Monitor VI to average the last three temperature measurements and display the average on a waveform chart.

1.A) Front Panel

Open the Temperature Monitor VI.
Select File>>Save As and rename the VI Temperature Running Average.vi in the C:\Exercises\LabVIEW Basics I directory.

1.B) Block Diagram

Display the block diagram.
Right-click the right or left border of the While Loop and select Add Shift Register from the shortcut menu to create a shift register.
Right-click the left terminal of the shift register and select Add Element from the shortcut menu to add an element to the shift register.
Modify the block diagram as in Figure 1.

Figure 1
1. Press the <Ctrl> key while you click the Thermometer VI and drag it outside the While Loop to create a copy of the subVI. The Thermometer VI returns one temperature measurement from the temperature sensor and initializes the left shift registers before the loop starts.
2. Place the Compound Arithmetic function, located on the Functions>>Arithmetic & Comparison>>Express Numeric palette, on the block diagram. This function returns the sum of the current temperature and the two previous temperature readings. Use the Positioning tool to resize the function to have three left terminals.
3. Place the Divide function, located on the Functions>>Arithmetic & Comparison>>Express Numeric palette, on the block diagram. This function returns the average of the last three temperature readings.
4. Right-click the y terminal of the Divide function, select Create>>Constant, type 3, and press the <Enter> key.

1.C) Run the VI

Run the VI. During each iteration of the While Loop, the Thermometer VI takes one temperature measurement. The VI adds this value to the last two measurements stored in the left terminals of the shift register. The VI divides the result by three to find the average of the three measurements, the current measurement plus the previous two. The VI displays the average on the waveform chart. Notice that the VI initializes the shift register with a temperature measurement.

1.D) Block Diagram

Modify the block diagram as shown in Figure 2.

Figure 2
1. Place the Bundle function, located on the Functions>>All Functions>>Cluster palette, on the block diagram. This function bundles the average and current temperature for plotting on the waveform chart.
Save the VI. You will use this VI later in the course.

1.E) Run the VI

Run the VI. The VI displays two plots on the waveform chart. The plots are overlaid. That is, they share the same vertical scale.
If time permits, complete the optional steps. Otherwise, close the VI.

1.F) Optional

Customize the waveform chart as shown in Figure 3. You can display a plot legend, a scale legend, a graph palette, a digital display, and a scrollbar. By default, a waveform chartdisplays the plot legend.

**Figure 3**

Customize the y-axis.
1. Use the Labeling tool to double-click 70.0 in the y-axis, type 75.0, and press the <Enter> key.
2. b: Use the Labeling tool to double-click the second number from the bottom on the y-axis, type 80.0, and press the <Enter> key. This number determines the numerical spacing of the y-axis divisions. For example, if the number above 75.0 is 77.5, it indicates a y-axis division of 2.5, changing the 77.5 to 80.0 reformats the y-axis to multiples of 5.0 (75.0, 80.0, 85.0, and so on).
NOTE:
The waveform chart size has a direct effect on the display of axis scales. Increase the waveform chart size if you encounter problems while customizing the axis.
Right-click the waveform chart and select Visible Items>>Scale Legend from the shortcut menu to display the scale legend, as shown in Figure 4. You can place the scale legend anywhere on the front panel.

Figure 4: 1. X-axis, 2. Y-axis, 3. Scale Labels, 4. Scale Lock Button, 5. Autoscale Button, 6. Scale Format Button
Use the scale legend to customize each axis.
1. Make sure the Lock Autoscale button appears locked and the Autoscale LED is green so the y-axis adjusts the minimum and maximum values to fit the data in the chart.
2. Click the Scale Format button to change the format, precision, mapping mode, scale visibility, and grid options for each axis.
Use the plot legend to customize the plots.
1. Use the Positioning tool to resize the plot legend to include two plots.
2. Use the Labeling tool to change Temp to Running Avg and to change Plot 1 to Current Temp. If the text does not fit, use the Positioning tool to resize the plot legend.
3. Right-click the plot in the plot legend to set the line and point styles and the color of the plot background or traces.
Right-click the waveform chart and select Visible Items>>Graph Palette from the shortcut menu to display the graph palette, as shown in Figure 5. You can place the graph palette anywhere on the front panel.

Figure 5: 1. Cursor Movement Tool, 2. Zoom Button, 3. Panning Tool, 4. Zoom Pull-down Menu

Use the Zoom button on the graph palette to zoom in or out of sections of the chart or the whole chart. Use the Panning tool to pick up the plot and move it around on the display. Use the Cursor Movement tool to move the cursor on the graph.
Run the VI. While the VI runs, use the buttons in the scale legend and graph palette to modify the waveform chart.

NOTE:
If you modify the axis labels, the display might become larger than the maximum size that the VI can correctly present.
Use the Operating tool to click the Power switch and stop the VI.
Save and close the VI.

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Measurement Studio Components

2010-09-25T07:45:00.001-07:00

Summary: This section introduces the three main components of Measurement Studio—ActiveX controls, C++ class libraries, and .NET class libraries.

Links

[hidelinks]

Supplemental links

Measurement Studio consists of three main types of components—ActiveX controls, C++ class libraries, and .NET class libraries.

The ActiveX controls provide language-native APIs for Visual Basic 6 as well as provide UI components for Measurement Studio C++ applications. Additionally, APIs exist for NI driver software, such as Traditional DAQ, NI-VISA, GPIB, IMAQ, IVI class drivers, and Motion.

The C++ class libraries provide language-native APIs for Visual C++ MFC applications. The C++ class libraries provide an object-oriented C++ interface and a consistent set of data types, data type transformations, and integration into the MFC framework for driver software including NI-DAQmx, NI-VISA, GPIB, IVI class drivers, IVI specific drivers, NI modular instrument drivers, such as NI-Scope, NI-DMM, and FieldPoint. The C++ class libraries provide the same sort of interface to the Measurement Studio ActiveX controls, automatically converting VARIANT and SAFEARRAY types to C++ MFC types and automatically handling ActiveX threading issues.

The .NET class libraries provide language-native APIs for .NET languages. Measurement Studio specifically supports C# and Visual Basic .NET through code generation, example programs, and customer support, but the .NET class libraries are built to be Common Language Specification (CLS)-compliant, so they should work with any CLS-compliant .NET language.

The DAQ ActiveX control is complimentary and is installed with the Traditional DAQ driver. All the other ActiveX controls require Measurement Studio.

Measurement Studio provides the following ActiveX controls:

User Interface controls, such as CWGraph, CWButton,CWKnob
NI VISA
NI 488.2 (CWInstr)
Analysis (CWDSP, CWArray, CWStat, CWMatrix, CWComplex)
DataSocket
IVI
Traditional DAQ (Complimentary with Traditional DAQ Driver)
3D Graph

Measurement Studio C++ Class Libraries

Measurement Studio provides the following C++MFC libraries:

3D Graph
Analysis
DataSocket
NI-DAQmx
NI-488.2
NI-VISA
IVI
LabVIEW RT
Microsoft Excel Interface
Microsoft Word Interface
NI-Reports
User Interface (Graph, Button, Knob)
Utility (File, Sound, System, Timer)

Measurement Studio .NET Libraries

The Measurement Studio class libraries are designed specifically for engineers and scientists who build virtual instrumentation systems with Visual Basic .NET or Visual C# in Visual Studio. With integrated libraries for acquiring, analyzing, and displaying data, Measurement Studio includes everything you need for building advanced measurement and automation applications.

Measurement Studio provides the following .NET libraries:

Analysis .NET class library
Common .NET class library
DataSocket .NET class library
Network Variable .NET class library
User Interface .NET class library
Windows Forms .NET class library
ASP.NET Web Forms class library

User Interface Controls – Windows Forms

Visual Studio provides a wide variety of general-purpose controls, including calendar controls, text boxes, buttons, and data grids. With Measurement Studio, you can use the standard Visual Studio controls, in addition to test and control specific controls such as, graphs, gauges, and tanks to create complete applications running on your Windows desktop.

Out of the box, Measurement Studio contains managed .NET controls, such as graphs, dials, gauges, LEDs, sliders, tanks, numeric edits (double data type), and knobs. All of these controls can be configured to create robust and professional user interfaces that integrate seamlessly with your .NET application. Attributes, such as color, font size, and graph type can be instantly reconfigured, and you can even rotate, zoom, and pan your graphs at runtime.

All Measurement Studio controls are designed to be extensible, thus you can use them as a base class and apply your own customization to create entirely new controls.

**Figure 1**

User Interface Controls—Web Forms

In addition to providing Windows Forms user interface controls, Measurement Studio also provides user interface controls for web forms. These controls include graphs, gauges, knobs, switches, tanks, slides, and so on.

Drag and drop the appropriate controls onto the Visual Studio designer. Run your application, it automatically launches a Web browser and displays your controls. You can even interact with these Web controls.

These controls are based upon Microsoft ASP.NET, providing you a powerful platform to create Web pages. Web clients do not require any runtime engine to display and interact with ASP.NET web pages. As a result, any browser, Operating system, or Web device can view your Measurement Studio ASP.NET.

**Figure 2**

Additional User Interface Controls

Some of the newest Measurement Studio controls include the Digital Waveform Graph, Complex Graph, Property Editor, Instrument Control Strip Control, and several array user interface controls.

Digital Waveform Graph—With the digital waveform graph, you can plot digital waveform data.

**Figure 3**

Complex Graph—With the complex graph, you can plot and chart complex double data consisting of both real and imaginary components.

**Figure 4**

Instrument Control Strip Control—This control is a container for ToolStripPropertyEditor and other items derived from ToolStripItem. The control displays a set of Property Editor controls.

**Figure 5**

Property Editor—With the property editor control, you create applications that enable your end users to configure properties for Windows Forms controls at run time. You can edit properties of any .NET type at run time, including collections, and you can edit expandable properties that represent nested properties of another object, such as major divisions of an axis.

**Figure 6**

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Interactive Ball and Beam Experiment

2010-09-25T07:44:00.001-07:00

Summary: The ball and beam experiment represents a simple second order system. This system is easily simulated using four states, but is more difficult to implement in real-life because not all necessary variables are easily observed. This simulation assumes all four states are easily observed and the system is linear throught the operating region. The plant model and the controller of the system are simulated in LabVIEW using the LabVIEW Simulation Module.

Content actionsThe ball and beam experiment is built with two moving parts, the ball and a beam, typically having a track so to limit the balls movement to one dimension along the beam. A motor is used to change the angle of the beam from the center. Both the angel of the beam and the position of the ball must be measured. The state space model utilizes all four states for control of the model. Those include the x position of the ball, the velocity of the ball, the angle of the bar and the angular velocity of the bar. The controller makes use of these values to derive an error signal used for feedback.

Download the source code for the Ball and Beam 3D Simulation here. This VI was created in LabVIEW 8.2 using the 3D Picture Control Toolit add-on which is not a supported feature.

NEW: If you find value in this 3D demo, please take some time to build on my work by updating it to support the new supported 3D Picture Control tool now shipping within LabVIEW. Then post it to Connexions!

**Figure 1:** LabVIEW Interactive Ball and Beam Experiment
Interactive Ball and Beam Experiment

The Ball and Beam source code can be downloaded here: Ball and Beam Simulation VI

**Figure 2:** This LabVIEW VI utilizes the LabVIEW Simulation module to simulate the controller effects on the Ball and Beam plant model.
Ball and Beam VI Block Diagram

This VI was created with LabVIEW 8.20 and the LabVIEW Simulation Module 8.20.

Reference: Solution to the Ball and Beam Problem Using the State-space Design Method by William C. Messner, Carnegie Mellon University and Dawn Tilbury, University of Michigan

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Serial Port Communication

2010-09-25T07:43:00.001-07:00

Summary: In this lesson, you will learn about serial communication.

Serial communication is a popular means of transmitting data between a computer and a peripheral device such as a programmable instrument or even another computer. Serial communication uses a transmitter to send data, one bit at a time, over a single communication line to a receiver. You can use this method when data transfer rates are low or you must transfer data over long distances. Serial communication is popular because most computers have one or more serial ports, so no extra hardware is needed other than a cable to connect the instrument to the computer or two computers together.

**Figure 1:** 1: RS-232 Instrument,2: RS-232 Cable, 3: Serial Port

Serial communication requires that you specify the following four parameters:

The baud rate of the transmission
The number of data bits encoding a character
The sense of the optional parity bit
The number of stop bits

Each transmitted character is packaged in a character frame that consists of a single start bit followed by the data bits, the optional parity bit, and the stop bit or bits. Figure 2 shows a typical character frame encoding the letter m.

**Figure 2**

Baud rate is a measure of how fast data are moving between instruments that use serial communication. RS-232 uses only two voltage states, called MARK and SPACE. In such a two-state coding scheme, the baud rate is identical to the maximum number of bits of information, including control bits, that are transmitted per second.

MARK is a negative voltage, and SPACE is positive. Figure 2 shows how the idealized signal looks on an oscilloscope. The following is the truth table for RS-232: Signal>3V=0 Signal>-3V=1

The output signal level usually swings between +12 V and -12 V. The dead area between +3 V and -3 V is designed to absorb line noise.

A start bit signals the beginning of each character frame. It is a transition from negative (MARK) to positive (SPACE) voltage. Its duration in seconds is the reciprocal of the baud rate. If the instrument is transmitting at 9,600 baud, the duration of the start bit and each subsequent bit is about 0.104 ms. The entire character frame of eleven bits would be transmitted in about 1.146 ms.

Data bits are transmitted upside down and backwards. That is, inverted logic is used, and the order of transmission is from least significant bit (LSB) to most significant bit (MSB). To interpret the data bits in a character frame, you must read from right to left and read 1 for negative voltage and 0 for positive voltage. This yields 1101101 (binary) or 6D (hex). An ASCII conversion table shows that this is the letter m.

An optional parity bit follows the data bits in the character frame. The parity bit, if present, also follows inverted logic, 1 for negative voltage and 0 for positive voltage. This bit is included as a simple means of error handling. You specify ahead of time whether the parity of the transmission is to be even or odd. If the parity is chosen to be odd, the transmitter then sets the parity bit in such a way as to make an odd number of ones among the data bits and the parity bit. This transmission uses odd parity. There are five ones among the data bits, already an odd number, so the parity bit is set to 0.

The last part of a character frame consists of 1, 1.5, or 2 stop bits. These bits are always represented by a negative voltage. If no further characters are transmitted, the line stays in the negative (MARK) condition. The transmission of the next character frame, if any, is heralded by a start bit of positive (SPACE) voltage.

How Fast Can I Transmit?

Knowing the structure of a character frame and the meaning of baud rate as it applies to serial communication, you can calculate the maximum transmission rate, in characters per second, for a given communication setting. This rate is just the baud rate divided by the bits per frame. In the previous example, there are a total of eleven bits per character frame. If the transmission rate is set at 9,600 baud, you get

9,600

=872 characters per second. Notice that this is the maximum character transmission rate. The hardware on one end or the other of the serial link might not be able to reach these rates, for various reasons.

Hardware Overview

There are many different recommended standards of serial port communication, including the following most common types.

RS-232

The RS-232 is a standard developed by the Electronic Industries Association (EIA) and other interested parties, specifying the serial interface between Data Terminal Equipment(DTE) and Data Communications Equipment (DCE). The RS-232 standard includes electrical signal characteristics (voltage levels), interface mechanical characteristics (connectors), functional description of interchange circuits (the function of each electrical signal), and some recipes for common kinds of terminal-to-modem connections. The most frequently encountered revision of this standard is called RS-232C. Parts of this standard have been adopted (with various degrees of fidelity) for use in serial communications between computers and printers, modems, and other equipment. The serial ports on standard IBM-compatible personal computers follow RS-232.

RS-449, RS-422, RS-423

The RS-449, RS-422, and RS-423 are additional EIA serial communication standards related to RS-232. RS-449 was issued in 1975 and was supposed to supersede RS-232, but few manufacturers have embraced the newer standard. RS-449 contains two subspecifications called RS-422 and RS-423. While RS-232 modulates a signal with respect to a common ground, or single-ended transmission, RS-422 modulates two signals against each other, or differential transmission. The RS-232C receiver senses whether the received signal is sufficiently negative with respect to ground to be a logical 1, whereas the RS-422 receiver senses which line is more negative than the other. This makes RS-422 more immune to noise and interference and more versatile over longer distances. The Macintosh serial ports follow RS-422, which can be converted to RS-423 by proper wiring of an external cable. RS-423 can then communicate with most RS-232 devices over distances of 15 m or so.

RS-232 Cabling

Devices that use serial cables for their communication are split into two categories. These are DCE and DTE. DCE are devices such as a modem, TA adapter, plotter, and so on, while DTE is a computer or terminal. RS-232 serial ports come in two sizes, the D-Type 25-pin connector and the D-Type 9-pin connector. Both of these connectors are male on the back of the PC. Thus, you require a female connector on the device. Table 1 shows the pin connections for the 9-pin and 25-pin D-Type connectors.

**Figure 3**

TABLE 1
Function	Signal	PIN	DTE	DCE
Data	TxD	3	Output	Input
Data	RxD	2	Input	Output
Handshake	RTS	7	Output	Input
	CTS	8	Input	Output
	DSR	6	Input	Output
	DCD	1	Input	Output
	STR	4	Output	Input
Common	Com	5	--	--
Other	RI	9	Output	Input

The DB-9 connector is occasionally found on smaller RS-232 lab equipment. It is compact, yet has enough pins for the core set of serial pins (with one pin extra).

NOTE:

The DB-9 pin numbers for transmit and receive (3 and 2) are opposite of those on the DB-25 connector (2 and 3). Be careful of this difference when you are determining if a device is DTE or DCE.

The DB-25 connector is the standard RS-232 connector, with enough pins to cover all the signals specified in the standard. Table 2 shows only the core set of pins that are used for most RS-232 interfaces.

**Figure 4**

TABLE 2
Function	Signal	PIN	DTE	DCE
Data	TxD	2	Output	Input
Data	RxD	3	Input	Output
Handshake	RTS	4	Output	Input
	CTS	5	Input	Output
	DSR	6	Input	Output
	DCD	8	Input	Output
	STR	20	Output	Input
Common	Com	7	--	--

Software Overview

Use the VIs and functions located on the Functions>>All Functions>>Instrument I/O>>Serial palette for serial port communication.

You used some of the VISA functions on this palette for GPIB communication. The VISA Write and VISA Read functions work with any type of instrument communication and are the same whether you are doing GPIB or serial communication. However, because serial communication requires you to configure extra parameters, you must start the serial port communication with the VISA Configure Serial Port VI.

The VISA Configure Serial Port VI initializes the port identified by VISA resource name to the specified settings. timeout sets the timeout value for the serial communication. baud rate, data bits, parity, and flow control specify those specific serial port parameters. The error in and error out clusters maintain the error conditions for this VI.

EXAMPLE 1

Figure 5 shows how to send the identification query command *IDN? to the instrument connected to the COM2 serial port. The VISA Configure Serial Port VI opens communication with COM2 and sets it to 9,600 baud, eight data bits, odd parity, one stop bit, and XON/XOFF software handshaking. Then the VISA Write function sends the command. The VISA Read function reads back up to 200 bytes into the read buffer, and the Simple Error Handler VI checks the error condition.

**Figure 5**

NOTE:

The VIs and functions located on the Functions>>All Functions>>Instrument I/O>>Serial palette are also used for parallel port communication. You specify the VISA resource name as being one of the LPT ports. For example, you can use MAX to determine that LPT1 has a VISA resource name of ASRL10::INSTR.

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Simple Data Logger VI

2010-09-25T07:42:00.001-07:00

Summary: In this exercise, the objective is to learn to use a LabVIEW Measurement File. Complete the following steps to modify the Measurement Averaging VI to log the acquired data to a LabVIEW Measurement File. Create another VI that reads the data file.

EXERCISE 1

1.A) Simple Data Logger Block Diagram

Open the Measurement Averaging VI located in the C:\Exercises\LabVIEW Basics I directory.
Modify the block diagram to log the acquired data as shown in Figure 1.
Figure 1

Place the Write LabVIEW Measurement File Express VI, located on the Functions>>Output palette, on the block diagram. This Express VI stores the data acquired from the DAQ device. In the Configure Write LabVIEW Measurement File dialog box that appears, set the following options:
1. Set the Action to Ask user to choose file for the filename.
2. Set the Segment Headers to One header only to provide a header for all of the data. The header contains information about the sampling rate and the time when the sample was taken.
3. Set X Value Columns to One column per channel to provide a table of data that can be read by any spreadsheet editor or an ASCII text file editor.
4. Set the Delimiter to Tab to make it easy for a spreadsheet editor to determine where a column of data starts in the file.
5. Click the OK button to close the dialog box.
Place the Merge Errors VI, located on the Functions>>All Functions>>Time & Dialog palette, on the block diagram. It is important to catch errors with both DAQ and file I/O, and because the code has a parallel structure it is necessary to merge the errors from all of the parallel operations to determine if the code is functioning properly.
Select File>>Save As to save the VI as Simple Data Logger.vi in the C:\Exercises\LabVIEW Basics I directory.
Run the VI. A filename prompt appears. Name the file logger.lvm in the C:\Exercises\LabVIEW Basics I directory.
Stop and close the VI.

1.B) Simple Data Reader Front Panel

Open a blank VI and build the front panel shown in Figure 2 by placing a waveform chart, located on the Controls>>Graph Indicators palette, on the front panel.

**Figure 2**

1.C) Block Diagram

Build the block diagram shown in Figure 3.
Figure 3
1. Place the Read LabVIEW Measurement File Express VI, located on the Functions>>Input palette, on the block diagram. Because this Express VI reads data located in a LabVIEW measurement file one data point at a time, it must be placed in a loop. In the Configure Read LabVIEW Measurement Filedialog box that appears, set the following options:
  - In the Action section, place a checkmark in the Ask user to choose file checkbox.
  - Set the Segment Size to Retrieve segments of original size so that all the data stored in the file is retrieved.
  - Set Time Stamps to Relative to start of measurement. Because the dynamic data type stores information about the signal timing, this setting aligns the data with the time of the measurement.
  - In the Generic Text File section, remove the checkmark from the Read generic text files checkbox because the data is stored in a LabVIEW measurement file.
  - Click the OK button to close the dialog box.
2. Place the Unbundle by Name function, located on the Functions>>All Functions>>Cluster palette, on the block diagram.
3. Place the Or function, located on the Functions>>Arithmetic & Comparison>>Express Boolean palette, on the block diagram.
4. Wire the EOF? output of the Read LabVIEW Measurement File function to the Or function. Wire the status output of the error cluster to the second input of the Or function. This stops the While Loop when the entire LabVIEW Measurement File has been read or when an error occurs.
Save the VI as Simple Data Reader.vi in the C:\Exercises\LabVIEW Basics I directory.
Display the front panel, and run the VI. In the filename prompt that appears, select the logger.lvm file that you created earlier.
The data that was stored in the LabVIEW Measurement File appears in the waveform chart.
NOTE:
You might need to rescale or autoscale the y-axis of the waveform chart to display the data.
Close the Simple Data Reader VI.

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Digital Communication Experiments

2010-09-25T07:41:00.001-07:00

Summary: In this module, some digital communication experiments are performed using the NI-ELVIS workbench and the virtual exeriments available in the LanVIEW software.

EXPERIMENT NO: 1

ASK Generation

The method of generating the ASK waveform is given in figure 1.1, along with its corresponding output in figure 1.2.

**Figure 1**

Figure 1.1: The circuit used to generate the ASK waveform

**Figure 2**

Figure 1.2: The message signal with its ASK waveform

EXPERIMENT NO: 2

Sample-and-hold

The method of generating the sample and hold out is given in figure 2.1, along with its corresponding output in figure 2.2. Incidentally, LF-398 is a sample and hold IC.

**Figure 3**

Figure 2.1: The circuit used to generate Sample-and-hold output

**Figure 4**

Figure 2.2: The message signal with its corresponding Sample-and-hold output

EXPERIMENT NO: 3

BPSK

The ‘binary phase shift keying’ (BPSK), output can be obtained using the circuit of figure 3.1. An example output is contained in figure 3.2.

**Figure 5**

Figure 3.1: The circuit used to generate BPSK output

**Figure 6**

Figure 3.2: The binary message signal with its corresponding BPSK output

EXPERIMENT NO: 4

FSK

The ‘binary frequency shift keying’ (BFSK), output can be obtained using the circuit of figure 4.1. An example output is contained in figure 4.2.

**Figure 7**

Figure 4.1: The circuit used to generate FSK output

**Figure 8**

Figure 4.2: The binary message signal with its corresponding BFSK output

EXPERIMENT NO: 5

Natural sampling

The method of generating the ‘natural sampled’ signal is given in figure 5.1, while figure 5.2 has the output.

**Figure 9**

Figure 5.1: The circuit used to generate ‘Natural-sampled’ waveform

**Figure 10**

Figure 5.2: The message signal with its corresponding natural sampled output

EXPERIMENT NO: 6

Flat-top sampling

The method of generating the ‘flat-top-sampled’ signal is given in figure 6.1, while figure 6.2 has a typical input and corresponding output.

**Figure 11**

Figure 6.1: The circuit used to generate ‘flat-top sampling’

**Figure 12**

Figure 6.2: The message signal with its corresponding Flat-top sampled output

EXPERIMENT NO: 7

PAM-TDM

The method of generating the two channel pulse-amplitude-modulation time-division-multiplexed (PAM-TDM) output is given in figure 7.1, while a typical output is contained in figure 3, with figure 2 having the two message signals. A straight forward extension to include more analog channels is possible.

**Figure 13**

Figure 7.1: The circuit used to generate two channel PAM-TDM

**Figure 14**

Figure 7.2: The two message signals to be time division multiplexed.

**Figure 15**

Figure 7.3: The carrier, and the corresponding PAM-TDM waveform

EXPERIMENT NO: 8

QPSK

The blocks used for the generation of quadrature phase shift keying (QPSK), output are given in figure 8.1, while the signals at various stages are contained in figures 8.2 through 8.4.

**Figure 16**

Figure 8.1: The block diagram used to generate QPSK output

**Figure 17**

Figure 8.2: The binary inputs to the BPSK modulators of figure 1 (the signals b₁(t) and b₂(t))

**Figure 18**

Figure 8.3: The two inputs to the adder of figure 1 (the signals s₁(t) and s₂(t))

**Figure 19**

Figure 8.4: The two BPSK waveforms along with the corresponding QPSK waveform

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Acquiring a Signal VI

2010-09-25T07:40:00.001-07:00

Summary: Explore the LabVIEW enviroment by creating a VI that generates a signal and displays it on the front panel.

EXERCISE 1

In the following exercise, you will build a VI that generates a signal and displays that signal in a graph. LabVIEW provides templates containing information from which you can build a VI. These templates help you get started with LabVIEW.

Complete the following steps to create a VI that generates a signal and displays it on the front panel.

Launch LabVIEW.
In the LabVIEW dialog box that appears, shown in Figure 1, click the New button to display the New dialog box.

Figure 1
Select VI from Template>>Tutorial (Getting Started)>>Generate and Display in the Create new list. This template VI generates and displays a signal. Notice that previews of the template VI appear in the Front panel preview and the Block diagram preview sections. The Figure 2 shows the New dialog box and theGenerate and Display template VI.

Figure 2
Click the OK button to open the template. You also can double-click the name of the template VI in the Create new list to open the template.
Examine the front panel of the VI. The user interface, or front panel, appears with a gray background and includes controls and indicators. The title bar of the front panel indicates that this window is the front panel for the Generate and Display VI.

NOTE:
If the front panel is not visible, you can display the front panel by selecting Window>>Show Front Panel.
Examine the block diagram of the VI. The block diagram appears with a white background and includes VIs and structures that control the front panel objects. The title bar of the block diagram indicates that this window is the block diagram for the Generate and Display VI.

NOTE:
If the block diagram is not visible, you can display the block diagram by selecting Window>>Show Block Diagram.
On the front panel toolbar, click the Run button, shown at left. Notice that a sine wave appears on the graph.
Stop the VI by clicking the Stop button, shown at left, on the front panel.

1.A) Adding a Control to the Front Panel

Controls on the front panel simulate the input devices on a physical instrument and supply data to the block diagram of the VI. Many physical instruments have knobs you can turn to change an input value. Complete the following steps to add a knob control to the front panel.

TIP:

Throughout these exercises, you can undo recent edits by selecting Edit>>Undo or pressing the Ctrl-Z keys.

If the Controls palette is not visible on the front panel, select Window>>Show Controls Palette to display it.
Move the cursor over the icons on the Controls palette to locate the Numeric Controls palette. Notice that when you move the cursor over icons on the Controlspalette, the name of that subpalette appears in the gray space above all the icons on the palette. When you idle the cursor over any icon on any palette, the full name of the subpalette, control, or indicator appears.
Click the Numeric Controls icon to access the Numeric Controls palette.
Select the knob control on the Numeric Controls palette and place it on the front panel to the left of the waveform graph. You will use this knob in a later exercise to control the amplitude of a signal.
Select File>>Save As and save this VI as Acquiring a Signal.vi in the C:\Exercises\LabVIEW Basics I directory.

NOTE:
Save all the VIs you edit or create in this course in the C:\Exercises\LabVIEW Basics I directory.

1.B) Changing the Signal Type

The block diagram has a blue icon labeled Simulate Signal. This icon represents the Simulate Signal Express VI. The Simulate Signal Express VI simulates a sine wave by default. Complete the following steps to change this signal to a sawtooth wave.

Display the block diagram by selecting Window>>Show Block Diagram or by clicking the block diagram. Notice the Simulate Signal Express VI, shown in Media 5. An Express VI is a component of the block diagram that you can configure to perform common measurement tasks. The Simulate Signal Express VI simulates a signal based on the configuration that you specify.
Right-click the Simulate Signal Express VI and select Properties from the shortcut menu to display the Configure Simulate Signal dialog box.
Select Sawtooth from the Signal type pull-down menu. Notice that the waveform on the graph in the Result Preview section changes to a sawtooth wave. TheConfigure Simulate Signal dialog box should appear similar to Figure 3.

Figure 3
Click the OK button to apply the current configuration and close the Configure Simulate Signal dialog box.
Move the cursor over the down arrows at the bottom of the Simulate Signal Express VI.
When a double-headed arrow appears, shown at left, click and drag the border of the Express VI until the Amplitude input appears. Notice how you expanded the Simulate Signal Express VI to display a new input. Because the Amplitude input appears on the block diagram, you can configure the amplitude of the sawtooth wave on the block diagram. In Figure 3, notice how Amplitude is an option in the Configure Simulate Signal dialog box. When inputs, such asAmplitude, appear on the block diagram and in the configuration dialog box, you can configure the inputs in either location.

1.C) Wiring Objects on the Block Diagram

To use the knob control to change the amplitude of the signal, you must connect the two objects on the block diagram. Complete the following steps to wire the knob to theAmplitude input on the Simulate Signal Express VI.

Move the cursor over the Knob terminal, shown at left, until the Positioning tool appears. Notice how the cursor becomes an arrow, or the Positioningtool, shown in Media 9. Use the Positioning tool to select, position, and resize objects.
Click the Knob terminal to select it, then drag the terminal to the left of the Simulate Signal Express VI. Make sure the Knob terminal is inside the loop, shown in Media 10. The terminals are representations of front panel controls and indicators. Terminals are entry and exit ports that exchange information between the front panel and block diagram.
Deselect the Knob terminal by clicking a blank space on the block diagram.
Move the cursor over the arrow of the Knob terminal, shown in Media 11. Notice how the cursor becomes a wire spool, or the Wiring tool, shown at left. Use the Wiring tool to wire objects together on the block diagram.

NOTE:
The cursor does not switch to another tool while an object is selected.
When the Wiring tool appears, click the arrow and then click the Amplitude input of the Simulate Signal Express VI, shown in Media 13, to wire the two objects together. Notice that a wire appears and connects the two objects. Data flows along this wire from the terminal to the Express VI.
Select File>>Save to save this VI.

1.D) Running the VI

Running a VI executes your solution. Complete the following steps to run the Acquiring a Signal VI.

Display the front panel by selecting Window>>Show Front Panel or by clicking the front panel.

TIP:
Press the Ctrl-E keys to switch from the front panel to the block diagram or from the block diagram to the front panel.
Click the Run button.
Move the cursor over the knob control. Notice how the cursor becomes a hand, or the Operating tool, shown at left. Use the Operating tool to change the value of a control or select the text within a control.
Using the Operating tool, turn the knob to adjust the amplitude of the sawtooth wave. Notice how the amplitude of the sawtooth wave changes as you turn the knob. Also notice that the y-axis on the graph autoscales to account for the change in amplitude. To indicate that the VI is running, the Run button changes to a darkened arrow, shown at left. You cannot edit the front panel or block diagram while the VI runs.
Click the Stop button, shown at left, to stop the VI.

NOTE: Although Abort Execution button looks like a stop button, the Abort Execution button does not always properly close the VI. National Instruments recommends stopping your VIs using the Stop button on the front panel. Use the Abort Execution button only when errors prevent you from terminating the application using the Stop button.

1.E) Modifying the Signal

Complete the following steps to add scaling to the signal and display the results in the graph on the front panel.

On the block diagram, use the Positioning tool to double-click the wire that connects the Simulate Signal Express VI to theWaveform Graph terminal shown in Media 17.
Press the Delete key to delete this wire.
If the Functions palette is not visible on the block diagram, select Window>>Show Functions Palette to display it.
Select the Scaling and Mapping Express VI, shown at left, on the Arithmetic & Comparison palette and place it on the block diagram inside the loop between the Simulate Signal Express VI and the Waveform Graph terminal. If there is no room between the Express VI and the terminal, move theWaveform Graph terminal to the right. Notice that the Configure Scaling and Mapping dialog box automatically opens when you place the Express VI on the block diagram.
Define the value of the scaling factor by entering 10 in the Slope (m) text box. The Configure Scaling and Mapping dialog box should appear similar to Figure 4.

Figure 4
Click the OK button to apply the current configuration and close the Configure Scaling and Mapping dialog box.
Move the cursor over the arrow on the Sawtooth output of the Simulate Signal Express VI.
When the Wiring tool appears, click the arrow and then click the arrow on the Signals input of the Scaling and Mapping Express VI, shown in Media 20, to wire the two objects together.
Using the Wiring tool, wire the Scaled Signals output of the Scaling and Mapping Express VI to the Waveform Graph terminal. Notice the wires connecting the Express VIs and terminals. The arrows on the Express VIs and terminals indicate the direction that the data flows along these wires. The block diagram should appear similar toFigure 5.

Figure 5

NOTE:
The terminals in the block diagram are set to display as icons. To display a terminal as a data type on the block diagram, right-click the terminal and selectView As Icon from the shortcut menu to remove the checkmark.
Select File>>Save to save this VI.

1.F) Displaying Two Signals on the Graph

To compare the signal generated by the Simulate Signal Express VI and the signal modified by the Scaling and Mapping Express VI on the same graph, use the Merge Signals function. Complete the following steps to display two signals on the same graph.

Move the cursor over the arrow on the Sawtooth output of the Simulate Signal Express VI.
Using the Wiring tool, wire the Sawtooth output to the Waveform Graph terminal. The Merge Signals function, shown in Media 22, appears where the two wires connect. This function takes the two separate signals and combines them so that both can be displayed on the same graph. The block diagram should appear similar toFigure 6.

Figure 6
Select File>>Save to save this VI. You also can press the Ctrl-S keys to save a VI.
Return to the front panel, run the VI, and turn the knob control. Notice that the graph plots the sawtooth wave and the scaled signal. Also notice that the maximum value on the y-axis automatically changes to be 10 times the knob value. This scaling occurs because you set the slope to 10 in the Scaling and Mapping Express VI.
Click the Stop button.

1.G) Customizing the Knob

The knob control changes the amplitude of the sawtooth wave so labeling it Amplitude accurately describes the function of the knob. Complete the following steps to customize the appearance of a control on the front panel.

Right-click the knob and select Properties from the shortcut menu to display the Knob Properties dialog box.
In the Label section on the Appearance tab, delete the label Knob, and type Amplitude in the text box. The Knob Properties dialog box should appear similar toFigure 7.

Figure 7
Click the Scale tab and, in the Scale Range section, change the maximum value to 5.0. Notice how the knob on the front panel instantly updates to reflect these changes.
Click the OK button to apply the current configuration and close the Knob Properties dialog box.
Save this VI.

TIP:
As you build VIs, you can experiment with different properties and configurations. You also can add and delete objects. Remember, you can undo recent edits by selecting Edit>>Undo or pressing the Ctrl-Z keys.
Experiment with other properties of the knob by using the Knob Properties dialog box. For example, try changing the colors for the Marker Text Color by clicking the color box located on the Scale tab.
Click the Cancel button to avoid applying the changes you made while experimenting. If you want to keep the changes you made, click the OK button.

1.H) Customizing the Waveform Graph

The waveform graph indicator displays the two signals. To indicate which plot is the scaled signal and which is the simulated signal, you customize the plots. Complete the following steps to customize the appearance of an indicator on the front panel.

Move the cursor over the top of the plot legend on the waveform graph. Notice that while there are two plots on the graph, the plot legend displays only one plot.
When a double-headed arrow appears, shown in Figure 8, click and drag the border of the plot legend until the second plot name appears.

Figure 8
Right-click the waveform graph and select Properties from the shortcut menu to display the Graph Properties dialog box.
On the Plots tab, select Sawtooth from the pull-down menu. Click the Line Color color box to display the color picker. Select a new line color.
Select Sawtooth (Scaled) from the pull-down menu.
Place a checkmark in the Don't use waveform names for plot names checkbox.
In the Name text box, delete the current label and change the name of this plot to Scaled Sawtooth.
Click the OK button to apply the current configuration and close the Graph Properties dialog box. Notice how the plot color on the front panel changes.
Experiment with other properties of the graph by using the Graph Properties dialog box. For example, try disabling the autoscale feature located on the Scales tab.
Click the Cancel button to avoid applying the changes you made while experimenting. If you want to keep the changes you made, click the OK button.
Save and close this VI.

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Intersymbol Interference (ISI) and the Eye Diagram

2010-09-25T07:39:00.001-07:00

Summary: This project studies intersymbol interference (ISI) in an intuitive way by using a LabVIEW VI to simulate a pulse transmitter, finite bandwidth channel, and received signaling waveform. Rectangular pulses are considered first to demonstrate the ISI problem, and then two alternative pulse shapes are explored as a way to minimize ISI. The eye diagram is also introduced in this project as a visual aid to present the time-domain signaling waveform to promote understanding of the ISI phenomenon.

Links

[hidelinks]

Supplemental links

Introduction to ISI

TABLE 1
	This module refers to LabVIEW, a software development environment that features a graphical programming language. Please see the LabVIEW QuickStart Guide module for tutorials and documentation that will help you:
	• Apply LabVIEW to Audio Signal Processing
	• Get started with LabVIEW
	• Obtain a fully-functional evaluation edition of LabVIEW

NOTE:

Summary

This project studies intersymbol interference (ISI) in an intuitive way by using a LabVIEW VI to simulate a pulse transmitter, finite bandwidth channel, and received signaling waveform. Rectangular pulses are considered first to demonstrate the ISI problem, and then two alternative pulse shapes are explored as a way to minimize ISI. The eye diagram is also introduced in this project as a visual aid to present the time-domain signaling waveform to promote understanding of the ISI phenomenon.

Objectives

Understand the root cause of intersymbol interference (ISI)
Explain the significance of the sinc pulse and raised cosine pulse as a means to eliminate ISI
Understand the construction of an eye diagram
Be able to measure performance metrics (peak ISI, noise margin, jitter, and timing sensitivity) directly from the eye diagram

Deliverables

Summary write-up of your results
Any plots or diagrams requested

NOTE:

You can easily export LabVIEW front-panel waveform plots directly to your report. Right-click on the waveform indicator and choose "Export Simplified Image."

Setup

LabVIEW 8.5 or later version

Textbook Linkages

Refer to the following textbooks for additional background on the project activities of this module; see the "References" section below for publication details:

Carlson, Crilly, and Rutledge -- Ch 11
Couch -- Ch 3
Haykin -- Ch 4
Haykin and Moher -- Ch 6
Proakis and Salehi (FCS) -- Ch 9
Proakis and Salehi (CSE) -- Ch 8
Stern and Mahmoud -- Ch 4

Prerequisite Modules

Introduction

Introductory digital logic courses present digital waveforms as essentially rectangular pulses. Indeed, the internal signals of a digital integrated circuit ideally exist at one of two voltage levels (high and low), with minimal time spent changing from one state to the other. Waveform displays from digital circuit simulators further emphasize the two-level rectangular shape of ideal digital signals.

Rectangular pulses are not ideal for transmission through communication links, however, since communication channels always restrict the bandwidth available between the transmitter and the receiver. Rectangular signaling pulses contain significant spectral energy across a wide frequency range due to the step-like transition between levels, and yet most communication systems do not allocate nearly enough bandwidth to faithfully transmit these abrupt changes. Passing a rectangular pulse through a limited-bandwidth channel distorts the pulse by "smearing" it -- that is, the pulse stretches out in time. The transmitter sends a series of pulses to convey the message, therefore this time smearing causes interference between adjacent time slots (or bit slots). This intersymbol interference (abbreviated ISI) adds extraneous signal energy at the exact moments when a receiver's bit sampler decides whether a received bit should be called a logic "1" or a logic "0." ISI is not the same as additive random noise, but plays a similar role by reducing the noise margin, i.e., the room for error before the receiver's bit sampler makes an error.

This project studies intersymbol interference in an intuitive way by using a LabVIEW VI to simulate a pulse transmitter, finite bandwidth channel, and received signaling waveform. Rectangular pulses are considered first to demonstrate the ISI problem, and then two alternative pulse shapes are explored as a way to minimize ISI.

The eye diagram is also introduced in this project as a visual aid to present the time-domain signaling waveform to promote understanding of the ISI phenomenon. The eye diagram also reveals other key performance measures such as noise margin, timing jitter, and timing sensitivity.

ISI_and_EyeDiagram.vi

Download the LabVIEW VI ISI_and_EyeDiagram.vi, an interactive tool to study various pulse shapes as they pass through a band-limited channel.

Open the VI which starts running automatically, and then view the Figure 1 screencast video for a short orientation tour of the VI.

**Figure 1:** [video] Orientation tour of the "ISI_and_EyeDiagram.vi" LabVIEW VI

Rectangle Pulse Shape

Restore the front panel controls of "ISI_and_EyeDiagram.vi" to their default values by selecting "Edit | Reinitialize Values to Defaults."

Set the symbols control to 1 to produce a single rectangular pulse. The channel bandwidth should already be set to its maximum value of 0.49, which corresponds to essentially unlimited bandwidth. Note that this VI uses normalized frequency, therefore the sampling frequency corresponds to 1 and the Nyquist frequency is 0.5.

Compare the "transmitted waveform" and the "received waveform" plots in the lower-right front panel. How well does the received pulse match the transmitted pulse? Also, to what extent does the received pulse "spill out" of its designated time slot?

Decrease the channel bandwidth until you begin to observe noticeable pulse shape distortion. At what bandwidth does this occur?

Continue decreasing the channel bandwidth. What effects do you begin to observe?

Make a series of plots that show the progressive degradation of the rectangular pulse shape as the channel bandwidth is restricted. Right-click on the plot and choose "Export Simplified Image" to copy the graph to the clipboard for pasting into your report. Be sure to indicate the channel bandwidth for each plot.

Sinc Pulse Shape

Restore the front panel controls of "ISI_and_EyeDiagram.vi" to their default values.

Set the symbols control to 1 to produce a single pulse, and set the bandwidth control to 0.02. The received pulse should show noticeable distortion.

Now set the pulse shape control to "Sinc." How much distortion is evident at the receiver? How much lower can you restrict the bandwidth while still preserving the basic sinc waveform shape?

The sinc function's ability to maintain its basic shape through a restricted channel bandwidth is important, but its true significance extends beyond this fact, as explored in the next section.

Multiple Pulses

A transmitter converts a message, or sequence of bits, into a series of analog pulses to create the signaling waveform. A receiver recovers the bitstream by periodically sampling the signaling waveform and comparing the sample to a threshold value to decide "1" or "0." Sinc-shaped pulse do not interfere with adjacent bit slots, provided that the bit slots are sampled at the correct instant in time.

To see this, reinitialize the front panel control values to their default settings, choose the "Sinc" pulse shape, and choose 2 symbols. Look carefully at the transmitted and received pulses on the lower-left front panel plots. The white trace shows the first pulse in the sequence, while the red trace shows the second pulse in the sequence. The first pulse has an amplitude of +1, while the second pulse has an amplitude of -1, corresponding to a bit sequence "10"; refer to the message bitstream indicator to confirm that the first bit is T (green LED indicator active) and the second bit is F (inactive LED indicator).

The waveform plots on the lower-right front panel show the actual transmitted and received waveforms, which superimpose (i.e., add) the individual pulses together. The plots on the lower-left front panel illustrate the contribution of each individual pulse.

Look carefully at the time 450 samples, in which the second (red) pulse is at its most negative value. What is the value of the first (white) pulse at this instant? Hopefully you can see that it iszero, indicating that the first (white) pulse produces zero interference at the instant the second (red) pulse attains its maximum absolute value.

Now, increase the number of symbols to 3, and also to higher values. Study the waveforms to convince yourself that even though a single sinc pulse extends over many bit time intervals, the contribution of all adjacent pulses is always zero at the moment that a given sinc pulse attains its maximum absolute value. Therefore, the sinc pulse shape achieves zero ISI when properly sampled.

Make a series of plots from the "received pulses" waveform display and explain your understanding of the sinc pulse shape and its ability to achieve zero ISI.

Restrict the channel bandwidth to 0.02, and confirm that the sinc pulses remain essentially unchanged. Now, set the pulse shape to "Rectangular." Set the symbols control to 2 and study the lower-left front panel plots. Identify where the second (red) pulse attains its maximum absolute value. How much interference is present from the first (white) pulse?

Make a series of plots from the "received pulses" waveform display and explain your understanding of the rectangle pulse shape and its susceptibility to intersymbol interference.

Eye Diagram

Study the transmitted and received waveform plots on the lower-right front panel as you increase the number of symbols to 40, and try this for the two pulse shapes considered so far. Also try varying the channel bandwidth. The received signaling waveform is reasonably easy to understand for rectangle pulses, but is rather difficult to interpret when sinc pulses are used. For example, you should find that you can easily correlate the "message bitstream" sequence with the high and low regions in the received waveform when rectangular pulse shapes are transmitted; the correlation is more difficult when sinc pulses are used.

The eye diagram provides a powerful visualization tool to interpret the behavior of the received waveform regardless of the pulse shape. The Figure 2 screencast video continues the discussion by explaining the eye diagram plot on the upper-right front panel of "ISI_and_EyeDiagram." Follow along with video, matching the front panel controls of "ISI_and_EyeDiagram.vi" to those of the video.

**Figure 2:** [video] Explanation of the eye diagram plot in the "ISI_and_EyeDiagram.vi" LabVIEW VI

The eye diagram reveals important performance metrics for a communication link, including noise margin, ISI, timing sensitivity, and zero-crossing jitter. In addition, the eye diagram shows the optimum sampling time for the receiver to regenerate a bitstream from the received signaling waveform. View the Figure 3 screencast video to learn how to measure these performance metrics and how to determine the optimum sampling time.

**Figure 3:** [video] Measuring noise margin, ISI, timing sensitivity, zero-crossing jitter, and optimum sampling time using an eye diagram

Eye Diagram Measurements

Figure 4 illustrates a generic eye pattern superimposed on a measured eye diagram plot and summarizes the definition of the various performance metrics discussed earlier. Use these definitions for the following measurements.

**Figure 4:** Generic eye pattern and definition of performance metrics

Rectangle Pulse

Restore the front panel controls of "ISI_and_EyeDiagram.vi" to their default values, and set the symbols control to 40. Vary the channel bandwidth and observe its effect on the eye diagram plot, and then set the channel bandwidth to 0.05. Increase the eye diagram start time to 245 samples to center the eye in the plot window.

Export the eye diagram plot to a piece of paper, and then use the eye diagram cursor as a tool to measure the following (show and label the relevant distances you measured on your hardcopy plot):

Optimum sampling time; report this as the number of samples from the nearest zero crossing
Peak ISI
Zero crossing jitter; report this as the maximum variation in time samples
Noise margin

Sinc Pulse

Ensure that the front panel controls of "ISI_and_EyeDiagram.vi" are the same as in the previous step, and then select the "Sinc" pulse shape. Adjust the eye diagram start time andtime span to maximize the number of displayed bit intervals and also to avoid the initial startup transient that causes lines to cross through the center of the eye; also make adjustments to place the maximum eye opening at the center of the plot window.

As in the previous step, export the eye diagram plot to a piece of paper, and then use the eye diagram cursor as a tool to measure the following (show and label the relevant distances you measured on your hardcopy plot):

Peak ISI
ISI at the optimum sampling time
Zero crossing jitter; report this as the maximum variation in time samples
Noise margin
Timing error sensitivity; report this in terms of time samples

Raised Cosine Pulse

Keep the front panel controls of "ISI_and_EyeDiagram.vi" at the same settings you used for the previous "Sinc" pulse measurements, and then select the "Raised Cosine" pulse shape. You should expect to see the maximum eye opening remain centered in the eye diagram plot.

As in the previous steps, export the eye diagram plot to a piece of paper, and then use the eye diagram cursor as a tool to measure the the same five metrics as for the "Sinc" pulse. Show and label the relevant distances you measured on your hardcopy plot.

Compare your results for the raised cosine pulse and the sinc pulse. What appears to be advantageous about the raised cosine pulse shape?

See the video screencast in pam_RaisedCosinePulse.vi for more background about the raised cosine pulse, the most widely-used pulse shape in digital communication systems.

Noise

Add random channel noise to the received waveform by moving the noise standard deviation control away from zero. Note how the eye pattern begins to close as the noise level increases.

Report the noise standard deviation value at which the eye just begins to close completely for each of the three pulse shapes. Make hardcopy plots of the eye diagram for each value that you report.

Channel Filter

"ISI_and_EyeDiagram.vi" uses a 10th-order IIR lowpass filter to model the limited-bandwidth channel. This type of filter is fairly realistic, and produces delay distortion, an effect caused by the filter's nonlinear phase response. Delay distortion causes the various signal frequency components to arrive at the receiver at different times, and can severely distort the originally-transmitted pulse shape.

Engage the FIR pushbutton control to select a linear-phase FIR filter that does not introduce delay distortion.

Final Comments

The seed front-panel control in the "Pulses" section primes the random number generator that creates the message bitstream. Feel free to try other seed values to produce other bit sequences for the message.

"ISI_and_EyeDiagram.vi" uses two programming techniques that you may wish to investigate further, namely, an event structure and property nodes. Type Ctrl+E to show the block diagram window, and observe the event structure within the while-loop structure. The event structure only "fires" when a front-panel control value changes; the CPU does not do any work except to instantly respond to front-panel activity. The event structure makes the VI very responsive to user input, but does not burden the CPU as would a plain while-loop.

The property nodes allow front panel controls and indicators to be programmatically controlled. For example, changing the value of the samples front-panel control causes the upper limit ofstart time to automatically change to the same value.

References

Carlson, A. Bruce, Paul B. Crilly, and Janet C. Rutledge, "Communication Systems," 4th ed., McGraw-Hill, 2002. ISBN-13: 978-0-07-011127-1
Couch, Leon W. II, "Digital and Analog Communication Systems," 7th ed., Pearson Prentice Hall, 2007. ISBN-10: 0-13-142492-0
Haykin, Simon. "Communication Systems," 4th ed., Wiley, 2001. ISBN-10: 0-471-17869-1
Haykin, Simon, and Michael Moher, "Introduction to Analog and Digital Communication Systems," 2nd ed., Wiley, 2007. ISBN-13: 978-0-471-43222-7
Proakis, John G., and Masoud Salehi, "Fundamentals of Communication Systems," Pearson Prentice Hall, 2005. ISBN-10: 0-13-147135-X
Proakis, John G., and Masoud Salehi, "Communication Systems Engineering," 2nd ed., Pearson Prentice Hall, 2002. ISBN-10: 0-13-061793-8
Stern, Harold P.E., and Samy A. Mahmoud, "Communication Systems," Pearson Prentice Hall, 2004. ISBN-10: 0-13-040268-0

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Reading and Writing Audio Files in LabVIEW

2010-09-25T07:38:00.001-07:00

Summary: Learn how to use LabVIEW to retrieve an audio signal from a WAV-format file, and how to save an audio signal that you have created to a WAV-format file.

Overview

LabVIEW offers a variety of ways to read and write audio files in WAV format. After completing this module you will be able to use the Simple Read and Simple Write subVIs located in the “Programming | Graphics & Sound | Sound | Files” palette to retrieve an audio signal as a 1-D array from a .wav file, and also to save a 1-D array that represents an audio signal to a .wav file. Additional points covered include scaling your audio signal to have a maximum absolute value of one before saving as an audio file, and how to create a two-channel (stereo) audio file.

Retrieve an Audio Signal from a .wav File

The Sound File Simple Read subVI accepts a filename for an audio file in .wav format and returns a waveform data type. You can read mono or stereo files, and also determine information such as the audio signal’s sampling frequency and its total number of samples.

The following video screencast shows how to use Simple Read to retrieve the audio signal as an array data type that can be used as a signal input for your own VIs. You may want to start LabVIEW now, then follow along to create your own version of the VI pictured below. If so, you will need the two audio files referenced in the screencast: tone-noise.wav and left-right.wav.

**Figure 1:** [video] LabVIEW Techniques: Retrieve an audio signal from a .wav file

Save an Audio Signal to a .wav File

The Sound File Simple Write subVI accepts a signal in waveform data type and a filename and stores the signal as a .wav file. You can set the number of bits per sample (16 is recommended for best fidelity). You must ensure that your signal values lie in the range -1 to +1, otherwise other applications may not be able to read your .wav file properly.

The next video screencast shows how to use Simple Write to save an existing 1-D array that represents an audio signal to a .wav file. The screencast also covers some not-so-obvious data-type transformations required to successfully create your .wav file. You may want to start LabVIEW now, then follow along to create your own version of the VI pictured below. If so, you will need the audio file referenced in the screencast: tone-noise.wav.

**Figure 2:** [video] LabVIEW Techniques: Save an audio signal to a .wav file

Scale Your Audio Before Saving to a .wav File

As mentioned earlier, you must ensure that all of the samples in your signal lie in the range -1 to +1. It is easy to create signals that exceed this range, especially when you add multiple signals together.

The next screencast illustrates how Quick Scale is an easy way to scale your 1-D array such that the maximum absolute value is always one; Quick Scale is located in the “Signal Processing | Sig Operation” palette, and should be used as the last step before converting to the waveform data type.

**Figure 3:** [video] LabVIEW Techniques: Scale audio signal to +/- 1 range before saving to a .wav file

Create a Two-Channel (Stereo) .wav File

Creating a stereo .wav file requires that you assemble an array of waveforms, one for the left channel and the other for the right channel. Watch the next screencast video to learn how.

**Figure 4:** [video] LabVIEW Techniques: Save a stereo (two-channel) audio signal to a .wav file

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Instrumenting a Circuit Simulation in National Instruments Multisim

2010-09-25T07:37:00.001-07:00

Summary: This module provides a brief introduction to instrumenting a circuit simulation in National Instruments Multisim. Multisim is fully featured SPICE schematic capture and simulation tool used for circuit design, simulation, and in combination with Ultiboard for layout of custom PCBs.

Instrumenting a Simulation in MultiSim

Virtual instruments are components in Multisim which model real-world bench-top instruments. Examples of virtual instruments in Multisim include Oscilloscopes, Function Generators, Network Analyzers, and Bode Plotters.

Virtual instruments provide designers with an easy and intuitive method for interacting with their circuits as they would in a testing or prototyping phase.

In addition to the provided virtual instruments, designers with familiar with National Instruments LabVIEW can create their own custom instruments from scratch. For example, one could create a custom noise generator to model electromagnetic interference.

Custom virtual instruments written in NI LabVIEW can also acquire real-world data and use the data to drive simulations, and can send generated data to analog output hardware, allowing simulated data to control real-world devices. The LabVIEW development suite is required to create custom LabVIEW instruments, but not necessary to run existing LabVIEW based instruments.

To place a virtual instrument, select the desired instrument from the Instruments toolbar (Figure 1). To view the front panel of the instrument, double-click on the instrument icon. Make connections to the terminals of the instrument icon as you would to any other component.

Multisim also provides designers with simulated benchtop instruments. These are instruments such as the Tektronix TDS 2024 Oscilloscope. They will look and operate exactly according to the manufacturers’ user manuals.

**Figure 1:** Instruments Toolbar

A single circuit can have multiple instruments attached to it, including (for most tiers) multiple instances of the same instrument. In addition, each circuit window can have its own set of instruments. Each instance of any instrument is configured and connected independently.

An overview of the more common instruments is included in this section. For more detailed information on how to use each instrument, consult the Multisim User Guide or helpfile.

Multimeter

Use the multimeter to measure AC or DC voltage or current, and resistance or decibel loss between two nodes in a circuit. The multimeter is auto-ranging, so a measurement range does not need to be specified. Its internal resistance and current are preset to near-ideal values, which can be changed.

**Figure 2:** Multimeter Schematic Symbol

**Figure 3:** Multimeter Front Panel

Function Generator

The function generator is a voltage source that supplies sine, triangular or square waves. It provides a convenient and realistic way to supply stimulus signals to a circuit. The waveform can be changed and its frequency, amplitude, duty cycle and DC offset can be controlled. The function generator's frequency range is great enough to produce conventional AC as well as audio- and radio-frequency signals.

The function generator has three terminals through which waveforms can be applied to a circuit. The common center terminal provides a reference level for the signal.

**Figure 4:** Function Generator Schematic Symbol

**Figure 5:** Function Generator Front Panel

Oscilloscopes

Multisim includes several varieties of oscilloscopes. All of the scopes can be controlled like real-world oscilloscopes. Horizontal timing and vertical voltages parameters can be adjusted. Triggering levels, and types are also selectable. Data from the Multisim-exclusive oscilloscopes is available in the Grapher after the interactive simulation has completed by selecting View/Grapher.

Multisim provides the following oscilloscopes:

2-Channel.
4-Channel.
Agilent 54622D Mixed Signal Oscilloscope.
Tektronix TDS 2024 Four Channel Digital Storage Oscilloscope.

**Figure 6:** Oscilloscope Schematic Symbol

**Figure 7:** Oscilloscope Front Panel

**Figure 8:** Tektronix Oscilloscope Schematic Diagram

**Figure 9:** Tektronix Oscilloscope Front Panel

Bode Plotter

The bode plotter displays the relative phase or magnitude response of an input and output signal. This is especially useful when characterizing the bandwidth of filter circuits.

**Figure 10:** Function Generator Schematic Symbol

**Figure 11:** Bode Plotter Front Panel

Spectrum Analyzer

The spectrum analyzer is used to measure amplitude versus frequency. This instrument is capable of measuring a signal's power and frequency components, and helps determine the existence of harmonics in the signal.

The spectrum analyzer displays its measurements in the frequency domain rather than the time domain. Usually the reference frame in signal analysis is time. In that case, an oscilloscope is used to show the instantaneous value as a function of time. Sometimes a sine waveform is expected but the signal, rather than being a pure sinusoidal, has a harmonic on it. As a result, it is not possible to measure the waveform's level. If the same signal is displayed on a spectrum analyzer, its amplitude is displayed, along with its frequency components, that is, its fundamental frequency and any harmonics it may contain.

**Figure 12:** Spectrum Analyzer Schematic Symbol

**Figure 13:** Spectrum Analyzer Front Panel

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Spatial Frequency Analysis

2010-09-25T07:34:00.001-07:00

Summary: This module discusses how the Spatial Transform is accomplished.

Aliasing in the Spatial Frequency Domain

Avoiding Spatial Aliasing

As we saw was the case in the time domain, a phenomenon known as aliasing can occur in the frequency domain if signals are not sampled at high enough rate. We have the same sort of considerations to take into account when we want to analyze the spectrum of the spatial frequency as well. As was discussed in the introduction, the Nyquest equivalent of the sampling rate is 1/2 of the minimum wavelength. This comes about from the relationship between speed, frequency and wavelength, which was discussed in the introduction as well. The figure below demonstrates the effects of aliasing in the spatial domain; it looks identical to filtering the time domain except that instead of the x-axis being related to pi/T it is now pi/d, where d is the distance between sensors. So, if we bandlimit our signal in temporal frequency, so that we can sample as two times the maximum temporal frequency, and if we design the sensors so that half of the minimum wavelength is greater than distance between sensors, we can avoid aliasing in both time and space!

**Figure 1:** Spatial Aliasing

Spatial Frequency Transform

Introduction to the Spatial Frequency Transform

Analogous to the DFT, is the sampled and windowed spatial equivalent, which is what we used to be able to filter our signal in frequency. The reason we want the information in the spatial frequency or wavenumber domain is because it is directly correlated to the angle the signal is coming from relative to the ULA. The spatial DFT is computed as the FFT of the first FFT. The first FFT represents the time domain frequency response and the second FFT represents the wavenumber response. This seems strange this would work, but let's explore this a little more fully. Let's look at theoretical example.

EXAMPLE 1: Mentally Visualizing the Spatial Frequency Transform

The 2-D Transform

Consider a box filled with numbers. The box is labeled on one edge time and on the other edge space. The first FFT we are taking is to obtain the temporal frequencies, so this would be like looking at a row along the box and taking the FFT of the numbers going across, while the spatial FFT would be calculated by looking at the numbers going down the columns. This is done repeatedly on each row and column, so the first FFT would go across each row, while the 2nd one would go down each column. This is easier to comprehend with a picture like the one below.

**Figure 2:** Visualization of mapping a signal into Spatial & Temporal Frequencies

SFT with Sinusoids

Since we were interested in detecting sinusoids, it would be interesting to consider what this kind of "double" Fourier Transform would do to a sinusoid. From our list of Fourier Transforms we know that the FFT of a sinusoid will give us a delta function shifted by the frequency of the sinusoid. We then see that the FFT of a delta function is 1, which would mean that we get the equivalent of white noise in spatial frequency! Fortunately, this is not exactly how the spatial FFT works. We are basically taking the FFT across one set of vectors followed by the FFT down the columns of those vectors, we are NOT taking the FFT(FFT(f(x,t)). So, when we accomplish this sort of arrangement on our signal, f(x,t), we get:

**Figure 3:** Spatial FFT of a Sinusoid

A sinc function!

Spatial Domain Filtering

Just as we are able to filter signals in temporal frequency, we can filter signals in spatial frequency. In fact, the way we accomplished the direction detecting algorithm in labview used a graph very similiar as the one above and then looking for the largest magnitude part of the signal. Once, this value is known, quick computation can then find the angle that signal came from! Ta da! We're done! Well, sort of.

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Creating a SubVI from Sections of a VI

2010-09-25T07:33:00.001-07:00

You can simplify the block diagram of a VI by converting sections of the block diagram into subVIs. Convert a section of a VI into a subVI by using the Positioning tool to select the section of the block diagram you want to reuse and selecting Edit>>Create SubVI. An icon for the new subVI replaces the selected section of the block diagram. LabVIEW creates controls and indicators for the new subVI and wires the subVI to the existing wires. Figure 1 shows how to convert a selection into a subVI.

**Figure 1**

The new subVI uses a default pattern for the connector pane and a default icon. Double-click the subVI to edit the connector pane and icon, and to save the subVI.

NOTE:

Do not select more than 28 objects to create a subVI because 28 is the maximum number of connections on a connector pane.

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Accessing Previous Loop Data

2010-09-25T07:32:00.001-07:00

When programming with loops, you often need to access data from previous iterations of the loop. For example, you may have a VI that reads the temperature and displays it on a graph. If you want to display a running average of the temperature as well, you need to use data generated in previous iterations. Two ways of accessing this data include the shift register and theFeedback Node.

Shift Registers

Use shift registers on For Loops and While Loops to transfer values from one loop iteration to the next. Shift registers are similar to static variables in text-based programming languages.

A shift register appears as a pair of terminals, shown in Media 1, directly opposite each other on the vertical sides of the loop border. The right terminal contains an up arrow and stores data on the completion of an iteration. LabVIEW transfers the data connected to the right side of the register to the next iteration. Create a shift register by right-clicking the left or right border of a loop and selecting Add Shift Register from the shortcut menu.

A shift register transfers any data type and automatically changes to the data type of the first object wired to the shift register. The data you wire to the terminals of each shift register must be the same type.

To initialize a shift register, wire any value from outside the loop to the left terminal. If you do not initialize the shift register, the loop uses the value written to the shift register when the loop last executed or the default value for the data type if the loop has never executed.

Use a loop with an uninitialized shift register to run a VI repeatedly so that each time the VI runs, the initial output of the shift register is the last value from the previous execution. Use an uninitialized shift register to preserve state information between subsequent executions of a VI. After the loop executes, the last value stored in the shift register remains at the right terminal. If you wire the right terminal outside the loop, the wire transfers the last value stored in the shift register.

You can add more than one shift register to a loop. If you have multiple operations within a loop, use multiple shift registers to store the data values from those different processes in the structure. The block diagram in Figure 1 uses two initialized shift registers.

**Figure 1**

Stacked Shift Registers

To create a stacked shift register, right-click the left terminal and select Add Element from the shortcut menu. Stacked shift registers let you access data from previous loop iterations. Stacked shift registers remember values from previous iterations and carry those values to the next iterations.

Stacked shift registers, as shown in Figure 2, can only occur on the left side of the loop because the right terminal only transfers the data generated from the current iteration to the next iteration.

**Figure 2**

If you add two more elements to the left terminal, values from the last three iterations carry over to the next iteration, with the most recent iteration value stored in the top shift register. The second terminal stores the data passed to it from the previous iteration, and the bottom terminal stores data from two iterations ago.

Feedback Nodes

The Feedback Node, shown in Media 4, appears automatically in a For Loop or While Loop if you wire the output of a subVI, function, or group of subVIs and functions to the input of that same VI, function, or group. Like a shift register, the Feedback Node stores data when the loop completes an iteration, sends that value to the next iteration of the loop, and transfers any data type. Use the Feedback Node to avoid unnecessarily long wires in loops. The Feedback Node arrow indicates in which direction the data flows along the wire.

You also can select the Feedback Node on the Structures palette and place it inside a For Loop or While Loop. If you place the Feedback Node on the wire before you branch the wire that connects the data to the tunnel, the Feedback Node passes each value to the tunnel. If you place the Feedback Node on the wire after you branch the wire that connects data to the tunnel, the Feedback Node passes each value back to the input of the VI or function and then passes the last value to the tunnel. The Accessing Previous Data VIexercise contains an example of this behavior.

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Waveform Transfers

2010-09-25T07:31:00.001-07:00

Summary: In this lesson, you will learn about waveform transfers.

Many instruments return a waveform as an ASCII string or a binary string. Assuming the same waveform, a binary string transfer is faster and requires less memory than an ASCII string transfer. Binary encoding requires fewer bytes than ASCII encoding.

ASCII Waveforms

As an example, consider a waveform composed of 1,024 points, each point having a value between 0 and 255. Using ASCII encoding, you would need a maximum of 4 bytes to represent each point (a maximum of 3 bytes for the value of the point and 1 byte for the separator, such as a comma). You would need a maximum of 4,096 ( 4×1,024) bytes plus any header and trailer bytes to represent the waveform as an ASCII string. Figure 1 is an ASCII waveform string.

**Figure 1**

You can use the Extract Numbers VI located in the C:\Exercises\LabVIEW Basics I directory to convert an ASCII waveform into a numeric array, as follows. This VI outputs the waveform as a double precision array.

**Figure 2**

Binary Waveforms Encoded as 1-Byte Integers

The same waveform using binary encoding requires only 1,024 bytes ( 1×1,024) plus any header and trailer bytes to be represented as a binary string. Using binary encoding, you need only 1 byte to represent the point, assuming each point is an unsigned 8-bit integer. Figure 3 is a binary waveform string.

**Figure 3**

Converting the binary string to a numeric array is a little more complex. You must convert the string to an integer array. You can do this by using the String To Byte Array function located on the Functions>>All Functions>>String>>String/Array/Path Conversion palette. You must remove all header and trailer information from the string before you can convert it to an array. Otherwise, this information also is converted.

**Figure 4**

Binary Waveforms Encoded as 2-Byte Integers

If each point in the binary waveform string is encoded as a 2-byte integer, it is easier and much faster to use the Type Cast function located on the Functions>>All Functions>>Advanced>>Data Manipulation palette. Refer to the LabVIEW Basics II: Development Course Manual for more information about type casting.

For example, consider a GPIB oscilloscope that transfers waveform data in binary notation. The waveform is composed of 1,024 data points. Each data point is a 2-byte signed integer. Therefore, the entire waveform is composed of 2,048 bytes. In Figure 5, the waveform has a 4-byte header DATA and a 2-byte trailer--a carriage return followed by a linefeed.

**Figure 5**

The block diagram in Figure 6 shows how you can use the Type Cast function to cast the binary waveform string into an array of 16-bit integers.

**Figure 6**

You might need to use the Swap Bytes function located on the Functions>>All Functions>>Advanced>>Data Manipulation palette to swap the most significant 8 bits and the least significant 8 bits for every element. Remember, the GPIB is an 8-bit bus. It can transfer only one byte at a time. If the instrument first sends the low byte and then the high byte, you do not need to use the Swap Bytes function.

In the example, you needed to use the Swap Bytes function because the instrument sent the most significant byte first. Because the most significant byte is received first, it is placed in a lower memory location than the least significant byte sent after the most significant byte.

**Figure 7**

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Voltage Monitor VI

2010-09-25T07:30:00.001-07:00

Summary: In this exercise, the objective is to build a VI that uses the DevSim instrument driver VIs to acquire and plot voltages. Complete the following steps to build a VI that acquires a DC voltage measurement from the NI Instrument Simulator once every second and plots it in a waveform chart until you click a button. As each value is acquired, the VI compares it with the previous minimum and maximum values. The VI calculates and displays the minimum and maximum values continuously on the front panel.

Front Panel

Select File>>New, then select Template>>Frameworks>>Single Loop Application to open the Single Loop Application template VI.
Build the front panel shown in Figure 1.
Figure 1

Use the following guidelines to help you construct the front panel.
- Place a VISA resource name control, located on the Controls>>All Controls>>I/O palette, on the front panel.
- Set the x-axis scale of the waveform chart to show incremental values.

Block Diagram

Build the block diagram shown in Figure 2.
Figure 2
1. Create two shift registers by right-clicking the right or left border of the loop and selecting Add Shift Register from the shortcut menu.
2. Place the NI DEVSIM Initialize VI, located on the Functions>>Input>>Instrument Drivers>>NI Device Simulator palette, on the block diagram. This VI opens communication between LabVIEW and the NI Instrument Simulator.
  - Right-click the ID Query input and select Create>>Constant from the shortcut menu. Use the Operating tool to change the constant to a False value.
  - Wire the Boolean constant to the Reset input.
3. Place the NI DEVSIM Multimeter Configuration VI, located on the Functions>>Input>>Instrument Drivers>>NI Device Simulator>>Configuration palette, on the block diagram. This VI configures the range of voltage measurements that the NI Instrument Simulator generates. The default is 0.0 to 10.0 V DC.
4. Place the NI DEVSIM Measure DC Voltage VI, located on the Functions>>Input>>Instrument Drivers>>NI Device Simulator>>Data palette, on the block diagram. This VI returns a simulated voltage measurement from the NI Instrument Simulator.
5. Place the NI DEVSIM Close VI, located on the Functions>>Input>>Instrument Drivers>>NI Device Simulator palette, on the block diagram. This VI ends communication between LabVIEW and the NI Instrument Simulator.
6. Place the Max & Min function, located on the Functions>>All Functions>>Comparison palette, on the block diagram. Use two of these functions to check the current voltage against the minimum and maximum values stored in the shift registers.
7. Place the Simple Error Handler VI, located on the Functions>>All Functions>>Time & Dialog palette, on the block diagram. This VI displays a dialog box if an error occurs and displays the error information.
8. Place the Unbundle by Name function, located on the Functions>>All Functions>>Cluster palette, on the block diagram. This function accepts status from the error cluster.
9. Place the Or function, located on the Functions>>Arithmetic & Comparison>>Express Boolean palette, on the block diagram. This function controls when the While Loop ends. If there is an error or you click the Stop button, the While Loop stops.
10. Set the wait for the Time Delay Express VI to 1 second.
11. Wire the block diagram as shown in Figure 2.
  NOTE:
  You do not need to wire every terminal for each node. Wire only the necessary inputs for each node, such as instrument descriptor, VISA session, and error I/O.
Save the VI as Voltage Monitor.vi in the C:\Exercises\LabVIEW Basics I directory.
Make sure the NI Instrument Simulator is powered on.
Display the front panel and run the VI. The LEDs alternate between Listen and Talk as LabVIEW communicates with the GPIB instrument once a second to get a simulated voltage reading. This voltage displays on the chart, and the minimum and maximum values update accordingly.
Stop and close the VI.

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Using Instrument Driver VIs

2010-09-25T07:29:00.001-07:00

Summary: In this lesson, you will learn how to use instrument driver VIs.

Instrument drivers are developed with a specific instrument in mind and eliminate the need for the user to know the exact IEEE 488.2 commands that the instrument is expecting.

Components of an Instrument Driver

All instrument drivers in the library have the same basic VI hierarchy. The hierarchy, sequence of VIs, and error handling are the same as those used in other areas of I/O in LabVIEW, such as file I/O, DAQ, TCP/IP, and so on. Refer to the File I/O VIs and Functions section of the Strings and File I/O lesson, for more information about error handling.

Figure 1 shows the hierarchy of an instrument driver.

**Figure 1**

The high-level functions are built from the low-level functions. For the most control over the instrument, use the low-level functions. The high-level functions are easy to use and have soft front panels that resemble the instrument. Instrument drivers have VIs in the following categories:

Initialize: Initializes the communication channel to the instrument. This VI also can perform an identification query and reset operation, and it can perform any necessary actions to place the instrument in its default power-on state or other specified state.
Configuration: Configures the instrument to perform operations, such as setting up the trigger rate.
Action/Status: Contains two types of VIs. Action VIs cause the instrument to initiate or terminate test and measurement operations. Status VIs obtain the current status of the instrument or the status of pending operations. An example of an action VI is Acquire Single Shot. An example of a status VI is Query Transfer Pending.
Data: Transfers data to or from the instrument, such as reading a measured waveform from the instrument or downloading a waveform to the instrument.
Utility: Performs a wide variety of functions, such as reset, self-test, error query, and revision query.
Close: Terminates the communication channel to the instrument and deallocates the resources for that instrument.

All National Instruments instrument drivers are required to implement the following functions: initialize, close, reset, self-test, revision query, error query, anderror message.

Application Examples

LabVIEW also includes application example VIs that show how to use the component VIs to perform common tasks. Typically, this includes configuring, triggering, and returning measurements from an instrument. An application example VI does not initialize or close the instrument driver. These VIs are not intended to be a soft front panel for the instrument but rather to demonstrate some instrument driver capabilities and guide you in developing your own VI.

Inputs and Outputs of Instrument Driver VIs

Just as all instrument drivers share a common VI hierarchy, they also share common inputs and outputs.

Resource Name or Instrument Descriptor

When you initialize the communication channel to an instrument, you must know the resource name or instrument descriptor. A resource is an instrument or interface, and the instrument descriptor is the exact name and location of a resource in the following format:


     Interface Type[board index]::Address::INSTR

Optional parameters are shown in square brackets [ ]. For example, GPIB::2::INSTR is the instrument descriptor for a GPIB instrument at address 2.

The VISA resource name control located on the Controls>>All Controls>>I/O palette is similar to the DAQ channel name control, but it is specifically used for instrument control. Refer to the VISA section for more information about VISA.

You can use MAX to determine what resources and instrument addresses are available, as you did in GPIB Configuration with MAX when you assigned a VISA alias of devsim to the NI Instrument Simulator. The alias makes it easier to communicate with instruments because you no longer need to memorize which interface and address each instrument uses. You can use the alias in the VISA resource name control instead of the instrument descriptor. For example, you can type devsim instead of GPIB::2::INSTR.

VISA Sessions

After you initialize an instrument, the Initialize VI returns a VISA session number. The VISA session is a connection or link to a resource, such as the instrument. You do not need to display this value. However, each time you communicate with that device, you must wire the VISA session input on the instrument driver VIs. After you finish communicating with the instrument, you use the Close VI to close all references or resources for the instrument.

EXAMPLE 1: Example Instrument Driver Application

The block diagram in Figure 2 initializes the instrument with the devsim alias, uses a configuration VI to select a waveform, uses two data VIs to read the waveform and the waveform scaling information, closes the instrument, and checks the error status. Every application that uses an instrument driver has a similar sequence of events.

**Figure 2**