Your SlideShare is downloading. ×
Objective4
Objective4
Objective4
Objective4
Objective4
Objective4
Objective4
Objective4
Objective4
Objective4
Objective4
Upcoming SlideShare
Loading in...5
×

Thanks for flagging this SlideShare!

Oops! An error has occurred.

×
Saving this for later? Get the SlideShare app to save on your phone or tablet. Read anywhere, anytime – even offline.
Text the download link to your phone
Standard text messaging rates apply

Objective4

523

Published on

Published in: Business, Technology
0 Comments
0 Likes
Statistics
Notes
  • Be the first to comment

  • Be the first to like this

No Downloads
Views
Total Views
523
On Slideshare
0
From Embeds
0
Number of Embeds
0
Actions
Shares
0
Downloads
2
Comments
0
Likes
0
Embeds 0
No embeds

Report content
Flagged as inappropriate Flag as inappropriate
Flag as inappropriate

Select your reason for flagging this presentation as inappropriate.

Cancel
No notes for slide

Transcript

  • 1. NATIONAL COLLEGE OF SCIENCE & TECHNOLOGY Amafel Bldg. Aguinaldo Highway Dasmariñas City, Cavite EXPERIMENT 2 DIGITAL COMMUNICATION OF ANALOG DATA USING PULSE-CODE MODULATION (PCM)Pula, Rolando A. September 20, 2011Signal Spectra and Signal Processing/BSECE 41A1 Score: Engr. Grace Ramones Instructor
  • 2. Objectives: Demonstrate PCM encoding using an analog-to-digital converter (ADC). Demonstrate PCM encoding using an digital-to-analog converter (DAC) Demonstrate how the ADC sampling rate is related to the analog signal frequency. Demonstrate the effect of low-pass filtering on the decoder (DAC) output.
  • 3. Sample Computation Step2 Step 6 Step 9 Step 12 Step 14 Step 16 Step 18
  • 4. Data Sheet:MaterialsOne ac signal generatorOne pulse generatorOne dual-trace oscilloscopeOne dc power supplyOne ADC0801 A/D converter (ADC)One DAC0808 (1401) D/A converter (DAC)Two SPDT switchesOne 100 nF capacitorResistors: 100 Ω, 10 kΩTheoryElectronic communications is the transmission and reception of informationover a communications channel using electronic circuits. Information is definedas knowledge or intelligence such as audio voice or music, video, or digitaldata. Often the information id unsuitable for transmission in its original form andmust be converted to a form that is suitable for the communications system.When the communications system is digital, analog signals must be convertedinto digital form prior to transmission.The most widely used technique for digitizing is the analog information signalsfor transmission on a digital communications system is pulse-code modulation(PCM), which we will be studied in this experiment. Pulse-code modulation(PCM) consists of the conversion of a series of sampled analog voltage levelsinto a sequence of binary codes, with each binary number that is proportionalto the magnitude of the voltage level sampled. Translating analog voltagesinto binary codes is called A/D conversion, digitizing, or encoding. The deviceused to perform this conversion process called an A/D converter, or ADC.An ADC requires a conversion time, in which is the time required to converteach analog voltage into its binary code. During the ADC conversion time, theanalog input voltage must remain constant. The conversion time for mostmodern A/D converters is short enough so that the analog input voltage will notchange during the conversion time. For high-frequency information signals, theanalog voltage will change during the conversion time, introducing an errorcalled an aperture error. In this case a sample and hold amplifier (S/H amplifier)will be required at the input of the ADC. The S/H amplifier accepts the input
  • 5. and passes it through to the ADC input unchanged during the sample mode.During the hold mode, the sampled analog voltage is stored at the instant ofsampling, making the output of the S/H amplifier a fixed dc voltage level.Therefore, the ADC input will be a fixed dc voltage during the ADC conversiontime.The rate at which the analog input voltage is sampled is called the samplingrate. The ADC conversion time puts a limit on the sampling rate because thenext sample cannot be read until the previous conversion time is complete. Thesampling rate is important because it determines the highest analog signalfrequency that can be sampled. In order to retain the high-frequencyinformation in the analog signal acting sampled, a sufficient number of samplesmust be taken so that all of the voltage changes in the waveform areadequately represented. Because a modern ADC has a very short conversiontime, a high sampling rate is possible resulting in better reproduction ofhigh0frequency analog signals. Nyquist frequency is equal to twice the highestanalog signal frequency component. Although theoretically analog signal canbe sampled at the Nyquist frequency, in practice the sampling rate is usuallyhigher, depending on the application and other factors such as channelbandwidth and cost limitations.In a PCM system, the binary codes generated by the ADC are converted intoserial pulses and transmitted over the communications medium, or channel, tothe PCM receiver one bit at a time. At the receiver, the serial pulses areconverted back to the original sequence of parallel binary codes. Thissequence of binary codes is reconverted into a series of analog voltage levelsin a D/A converter (DAC), often called a decoder. In a properly designedsystem, these analog voltage levels should be close to the analog voltagelevels sampled at the transmitter. Because the sequence of binary codesapplied to the DAC input represent a series of dc voltage levels, the output ofthe DAC has a staircase (step) characteristic. Therefore, the resulting DACoutput voltage waveshape is only an approximation to the original analogvoltage waveshape at the transmitter. These steps can be smoothed out intoan analog voltage variation by passing the DAC output through a low-passfilter with a cutoff frequency that is higher than the highest-frequencycomponent in the analog information signal. The low-pass filter changes thesteps into a smooth curve by eliminating many of the harmonic frequency. Ifthe sampling rate at the transmitter is high enough, the low-pass filter outputshould be a good representation of the original analog signal.
  • 6. In this experiment, pulse code modulation (encoding) and demodulation(decoding) will be demonstrated using an 8-bit ADC feeding an 8-bit DAC, asshown in Figure 2-1. This ADC will convert each of the sampled analog voltagesinto 8-bit binary code as that represent binary numbers proportional to themagnitude of the sampled analog voltages. The sampling frequencygenerator, connected to the start-of conversion (SOC) terminal on the ADC, willstart conversion at the beginning of each sampling pulse. Therefore, thefrequency of the sampling frequency generator will determine the samplingfrequency (sampling rate) of the ADC. The 5 volts connected to the VREF+terminal of the ADC sets the voltage range to 0-5 V. The 5 volts connected tothe output (OE) terminal on the ADC will keep the digital output connected tothe digital bus. The DAC will convert these digital codes back to the sampledanalog voltage levels. This will result in a staircase output, which will follow theoriginal analog voltage variations. The staircase output of the DAC feeds of alow-pass filter, which will produce a smooth output curve that should be a closeapproximation to the original analog input curve. The 5 volts connected to the+ terminal of the DAC sets the voltage range 0-5 V. The values of resistor R andcapacitor C determine the cutoff frequency (fC) of the low-pass filter, which isdetermined from the equationFigure 23–1 Pulse-Code Modulation (PCM) XSC2 G T A B C D S1 VCC Key = A 5V U1 Vin D0 S2 D1 V2 D2 D3 Key = B 2 Vpk D4 10kHz D5 0° Vref+ D6 Vref- D7 SOC VCC OE EOC 5V D0 D1 D2 D3 D4 D5 D6 D7 ADC V1 Vref+ R1 VDAC8 Output 5V -0V Vref- 100Ω 200kHz U2 R2 10kΩ C1 100nF
  • 7. In an actual PCM system, the ADC output would be transmitted to serial formatover a transmission line to the receiver and converted back to parallel formatbefore being applied to the DAC input. In Figure 23-1, the ADC output isconnected to the DAC input by the digital bus for demonstration purposes only.PROCEDURE:Step 1 Open circuit file FIG 23-1. Bring down the oscilloscope enlargement. Make sure that the following settings are selected. Time base (Scale = 20 µs/Div, Xpos = 0 Y/T), Ch A(Scale 2 V/Div, Ypos = 0, DC) Ch B (Scale = 2 V/Div, Ypos = 0, DC), Trigger (Pos edge, Level = 0, Auto). Run the simulation to completion. (Wait for the simulation to begin). You have plotted the analog input signal (red) and the DAC output (blue) on the oscilloscope. Measure the time between samples (TS) on the DAC output curve plot. TS = 4 µsStep 2 Calculate the sampling frequency (fS) based on the time between samples (TS) fS = 250 kHzQuestion: How did the measure sampling frequency compare with thefrequency of the sampling frequency generator? The difference is 50 kHz.How did the sampling frequency compare with the analog input frequency?Was it more than twice the analog input frequency? It is 20 times the analog input frequency. It is more than twice the analog input frequency.How did the sampling frequency compare with the Nyquist frequency? The Nyquist frequency is higher. Nyquist is 6.28 times more than the sampling frequency.Step 3 Click the arrow in the circuit window and press the A key to change Switch A to the sampling generator output. Change the oscilloscope time base to 10 µs/Div. Run the simulation for one oscilloscope screen display, and then pause the simulation. You are plotting the sampling generator (red) and the DAC output (blue).Question: What is the relationship between the sampling generator output andthe DAC staircase output? Both outputs are both in digitalStep 4 Change the oscilloscope time base scale to 20 µs/Div. Click the arrow in the circuit window and press the A key to change Switch A to the
  • 8. analog input. Press the B key to change the Switch B to Filter Output. Bring down the oscilloscope enlargement and run the simulation to completion. You are plotting the analog input (red) and the low-pass filter output (blue) on the oscilloscopeQuestions: What happened to the DAC output after filtering? Is the filter outputwaveshape a close representation of the analog input waveshape? The output became analog. Yes.Step 5 Calculate the cutoff frequency (fC) of the low-pass filter. fC = 15.915 kHzQuestion: How does the filter cutoff frequency compare with the analog inputfrequency? They have difference of approximately 6 kHz.Step 6 Change the filter capacitor (C) to 20 nF and calculate the new cutofffrequency (fC). fC = 79.577 kHzStep 7 Bring down the oscilloscope enlargement and run the simulation tocompletion again.Question: How did the new filter output compare with the previous filter output?Explain. It is almost the same.Step 8 Change the filter capacitor (C) back to 100 nF. Change the Switch B back to the DAC output. Change the frequency of the sampling frequency generator to 100 kHz. Bring down the oscilloscope enlargement and run the simulation to completion. You are plotting the analog input (red) and the DAC output (blue) on the oscilloscope screen. Measure the time between the samples (TS) on the DAC output curve plot (blue) TS = 9.5µsQuestion: How does the time between the samples in Step 8 compare with the time between the samples in Step 1? The time between the samples in Step 8 doubles.Step 9 Calculate the new sampling frequency (fS) based on the time between the samples (TS) in Step 8? fS=105.26HzQuestion: How does the new sampling frequency compare with the analog input frequency? It is 10 times the analog input frequency.
  • 9. Step 10 Click the arrow in the circuit window and change the Switch B to the filter output. Bring down the oscilloscope enlargement and run the simulation again.Question: How does the curve plot in Step 10 compare with the curve plot in Step 4 at the higher sampling frequency? Is the curve as smooth as in Step 4? Explain why. Yes, they are the same. It is as smooth as in Step 4. Nothing changed. It does not affect the filter.Step 11 Change the frequency of the sampling frequency generator to 50 kHz and change Switch B back to the DAC output. Bring down the oscilloscope enlargement and run the simulation to completion. Measure the time between samples (TS) on the DAC output curve plot (blue). TS = 19µsQuestion: How does the time between samples in Step 11 compare with the time between the samples in Step 8? It doubles.Step 12 Calculate the new sampling frequency (fS) based on the time between samples (TS) in Step 11. fS=52.631 kHzQuestion: How does the new sampling frequency compare with the analog input frequency? It is 5 times the analog input.Step 13 Click the arrow in the circuit window and change the Switch B to the filter output. Bring down the oscilloscope enlargement and run the simulation to completion again.Question: How does the curve plot in Step 13 compare with the curve plot in Step 10 at the higher sampling frequency? Is the curve as smooth as in Step 10? Explain why. Yes, nothing changed. The frequency of the sampling generator does not affect the filter.Step 14 Calculate the frequency of the filter output (f) based on the period for one cycle (T). T=10kHzQuestion: How does the frequency of the filter output compare with the frequency of the analog input? Was this expected based on the sampling frequency? Explain why. It is the same. Yes, it is expected.
  • 10. Step 15 Change the frequency of the sampling frequency generator to 15 kHz and change Switch B back to the DAC output. Bring down the oscilloscope enlargement and run the simulation to completion. Measure the time between samples (TS) on the DAC output curve plot (blue) TS = 66.5µsQuestion: How does the time between samples in Step 15 compare with the time between samples in Step 11? It is 3.5 times higher than the time in Step 11.Step 16 Calculate the new sampling frequency (fS) based on the time between samples (TS) in Step 15. fS=15.037 kHzQuestion: How does the new sampling frequency compare with the analog input frequency? It is 5 kHz greater than the analog input frequency.How does the new sampling frequency compare with the Nyquist frequency? It is 6.28 times smaller than the Nyquist frequency.Step 17 Click the arrow in the circuit window and change the Switch B to the filter output. Bring down the oscilloscope enlargement and run the simulation to completion again.Question: How does the curve plot in Step 17 compare with the curve plot in Step 13 at the higher sampling frequency? They are the same.Step 18 Calculate the frequency of the filter output (f) based on the time period for one cycle (T). f=10kHzQuestion: How does the frequency of the filter output compare with the frequency of the analog input? Was this expected based on the sampling frequency? They are the same. Yes it is expected for that sampling frequency.
  • 11. CONCLUSION: I can able to say that the Analog to Digital and Digital to Analog Converterscan be use as Pulse Code Modulation encoder and decoder. Based on the circuit performed, DAC output is a staircase while the filteroutput is analog. The staircase output sampling time is inversely proportional to thesampling frequency while the filter frequency is always equal with the analog inputfrequency. The cutoff frequency of the filter is inversely proportional to thecapacitance. Lastly, the Nyquist is always 6.28 times higher than the DAC outputfrequency.

×