This document describes an experiment on pulse code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC). The objectives are to demonstrate PCM encoding and decoding, show how the ADC sampling rate relates to analog signal frequency, and examine the effect of low-pass filtering on the DAC output. The experiment involves using an 8-bit ADC to sample an analog signal and an 8-bit DAC to reconstruct the signal, with a low-pass filter to smooth the DAC output.
1) The experiment demonstrated pulse code modulation (PCM) encoding using an analog-to-digital converter (ADC) and decoding using a digital-to-analog converter (DAC).
2) The sampling frequency determined by the pulse generator was measured to be 250 kHz, which was higher than twice the analog input frequency, satisfying the Nyquist criterion.
3) After low-pass filtering the DAC output, the waveform was smoothed into a close approximation of the original analog input signal.
The document describes an experiment demonstrating pulse code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC). The experiment showed how the ADC sampling rate must be at least twice the analog signal frequency to avoid aliasing. It also showed that a low-pass filter can smooth the DAC's staircase output into a representation of the original analog signal. The conclusions were that PCM can digitize analog signals for digital communication, with ADC and DAC performing the encoding and decoding, and that the filter output retains the analog input frequency regardless of the sampling rate.
SIGNAL SPECTRA EXPERIMENT 2 - FINALS (for PULA)Sarah Krystelle
This document describes an experiment conducted on a class B push-pull power amplifier. The objectives were to determine the dc and ac load lines, observe crossover distortion, measure maximum output voltage and power, and calculate efficiency. The circuit diagram and theory of operation for a class B push-pull amplifier are provided. Key steps in the procedure involve using simulations and equipment to analyze the input/output waveforms, dc bias voltages, and performance metrics.
1. The experiment demonstrated pulse-code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC).
2. The DAC output had a staircase-like waveform that was smoothed into an analog signal by a low-pass filter.
3. The sampling frequency determined by the pulse generator affected the time between samples but did not change the cutoff frequency of the filter or the output frequency, which matched the input analog signal frequency.
This document describes an experiment on Fourier theory involving the time domain and frequency domain. The objectives are to generate square and triangular waves from Fourier series, examine the difference between time and frequency domain plots, and analyze periodic pulses with different duty cycles in both domains while varying a low-pass filter's cutoff frequency. Procedures generate waves using function generators and measure them on an oscilloscope and spectrum analyzer while eliminating harmonics. The document explains Fourier analysis and how signals can be represented by sine/cosine waves of different frequencies and amplitudes in the frequency domain.
This document describes an experiment to demonstrate pulse-code modulation (PCM) using an analog-to-digital converter (ADC) and a digital-to-analog converter (DAC). The objectives are to encode and decode analog signals using PCM and demonstrate how the sampling rate affects the reproduction of analog signals. The experiment uses an 8-bit ADC to sample an analog input signal and convert it to an 8-bit digital code. The digital output is then converted back to an analog signal using an 8-bit DAC. A low-pass filter is used to smooth the staircase output of the DAC into a representation of the original analog input signal.
This document describes an experiment on Fourier theory and its applications in signal processing. The objectives are to: 1) Produce a square wave from sine waves of different frequencies and amplitudes using Fourier theory. 2) Produce a triangular wave similarly using cosine waves. 3) Examine the difference between time domain and frequency domain representations of signals. 4) Analyze periodic pulses with different duty cycles in both domains. 5) Examine the effect of low-pass filtering on pulses as the cutoff frequency varies. The experiment uses function generators, oscilloscopes, spectrum analyzers, and op-amps to generate and analyze signals.
The document summarizes an experiment on characterizing a class A power amplifier. Key steps include:
1) Determining the operating point (Q-point) on the DC load line. 2) Drawing the AC load line and ensuring the Q-point is centered. 3) Measuring the maximum undistorted output voltage and input voltage to calculate voltage gain. The measured gain is compared to theoretical calculations accounting for resistances. Unbypassed emitter resistance reduces gain and stability.
1) The experiment demonstrated pulse code modulation (PCM) encoding using an analog-to-digital converter (ADC) and decoding using a digital-to-analog converter (DAC).
2) The sampling frequency determined by the pulse generator was measured to be 250 kHz, which was higher than twice the analog input frequency, satisfying the Nyquist criterion.
3) After low-pass filtering the DAC output, the waveform was smoothed into a close approximation of the original analog input signal.
The document describes an experiment demonstrating pulse code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC). The experiment showed how the ADC sampling rate must be at least twice the analog signal frequency to avoid aliasing. It also showed that a low-pass filter can smooth the DAC's staircase output into a representation of the original analog signal. The conclusions were that PCM can digitize analog signals for digital communication, with ADC and DAC performing the encoding and decoding, and that the filter output retains the analog input frequency regardless of the sampling rate.
SIGNAL SPECTRA EXPERIMENT 2 - FINALS (for PULA)Sarah Krystelle
This document describes an experiment conducted on a class B push-pull power amplifier. The objectives were to determine the dc and ac load lines, observe crossover distortion, measure maximum output voltage and power, and calculate efficiency. The circuit diagram and theory of operation for a class B push-pull amplifier are provided. Key steps in the procedure involve using simulations and equipment to analyze the input/output waveforms, dc bias voltages, and performance metrics.
1. The experiment demonstrated pulse-code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC).
2. The DAC output had a staircase-like waveform that was smoothed into an analog signal by a low-pass filter.
3. The sampling frequency determined by the pulse generator affected the time between samples but did not change the cutoff frequency of the filter or the output frequency, which matched the input analog signal frequency.
This document describes an experiment on Fourier theory involving the time domain and frequency domain. The objectives are to generate square and triangular waves from Fourier series, examine the difference between time and frequency domain plots, and analyze periodic pulses with different duty cycles in both domains while varying a low-pass filter's cutoff frequency. Procedures generate waves using function generators and measure them on an oscilloscope and spectrum analyzer while eliminating harmonics. The document explains Fourier analysis and how signals can be represented by sine/cosine waves of different frequencies and amplitudes in the frequency domain.
This document describes an experiment to demonstrate pulse-code modulation (PCM) using an analog-to-digital converter (ADC) and a digital-to-analog converter (DAC). The objectives are to encode and decode analog signals using PCM and demonstrate how the sampling rate affects the reproduction of analog signals. The experiment uses an 8-bit ADC to sample an analog input signal and convert it to an 8-bit digital code. The digital output is then converted back to an analog signal using an 8-bit DAC. A low-pass filter is used to smooth the staircase output of the DAC into a representation of the original analog input signal.
This document describes an experiment on Fourier theory and its applications in signal processing. The objectives are to: 1) Produce a square wave from sine waves of different frequencies and amplitudes using Fourier theory. 2) Produce a triangular wave similarly using cosine waves. 3) Examine the difference between time domain and frequency domain representations of signals. 4) Analyze periodic pulses with different duty cycles in both domains. 5) Examine the effect of low-pass filtering on pulses as the cutoff frequency varies. The experiment uses function generators, oscilloscopes, spectrum analyzers, and op-amps to generate and analyze signals.
The document summarizes an experiment on characterizing a class A power amplifier. Key steps include:
1) Determining the operating point (Q-point) on the DC load line. 2) Drawing the AC load line and ensuring the Q-point is centered. 3) Measuring the maximum undistorted output voltage and input voltage to calculate voltage gain. The measured gain is compared to theoretical calculations accounting for resistances. Unbypassed emitter resistance reduces gain and stability.
The document discusses generating square and triangular waves using Fourier series of sine and cosine waves. It also examines signals in the time and frequency domains. Key points:
1) A square wave can be produced from a series of sine waves at different frequencies and amplitudes, with the fundamental and odd harmonics present.
2) A triangular wave results from a series of cosine waves, with the fundamental and odd harmonics.
3) Signals can be viewed in the time domain as voltage over time, or in the frequency domain as the amplitude of sine/cosine waves at different frequencies.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for AGDON)Sarah Krystelle
This experiment analyzed the operation of a class A power amplifier. Key findings include:
1) The initial operating point (Q-point) was not centered on the AC load line, resulting in output clipping.
2) Adjusting the emitter resistance centered the Q-point on the AC load line, eliminating clipping and increasing the maximum undistorted output voltage.
3) A class A amplifier has low efficiency due to conduction over the entire input cycle, but provides the most linear amplification.
SIGNAL SPECTRA EXPERIMENT 2 - FINALS (for CAUAN)Sarah Krystelle
This document describes Experiment #2 on a class B push-pull power amplifier. The objectives are to determine the dc and ac load lines, observe crossover distortion, measure voltage gain, output power, and efficiency. Sample computations are provided for voltage gain, output power, input power, and efficiency. The theory section describes class B push-pull amplifiers and how biasing the transistors slightly above cutoff can eliminate crossover distortion. Procedures are outlined to simulate and measure the amplifier's input, output, voltage gain, power output, and efficiency.
This document describes experiments performed to characterize active band-pass and band-stop filters, including plotting the gain-frequency response curves to determine cutoff frequencies and bandwidth, calculating quality factors and center frequencies, and comparing measured and expected voltage gains. Procedures are provided to implement and analyze a multiple-feedback band-pass filter and a two-pole Sallen-Key notch filter using op-amps and passive components.
This document describes an experiment to demonstrate pulse code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC). The objectives are to encode and decode analog signals using PCM and observe the effects of sampling rate and low-pass filtering. An 8-bit ADC samples an analog input and converts it to an 8-bit digital code. The DAC then reconverts the digital code to an analog output. Low-pass filtering the DAC output smooths the steps to approximate the original analog signal.
1. The document describes an experiment on Fourier theory and how signals can be represented in both the time domain and frequency domain. Square waves and triangular waves are generated from a series of sine and cosine waves (Fourier series) and plotted in both domains.
2. Low-pass filters are used to remove higher harmonics from signals. This distorts the original waveshape as more harmonics are removed. The bandwidth needed to transmit pulses with minimal distortion depends on the duty cycle.
3. Objectives include learning how square and triangular waves can be produced from Fourier series, comparing time and frequency domain plots, and examining how duty cycle and filtering affect pulses in both domains.
1. The document describes an experiment on Fourier theory involving the generation of square waves and triangular waves from a series of sine and cosine waves at different frequencies and amplitudes.
2. Key findings include that a square wave can be produced from odd harmonics of a fundamental sine wave, while a triangular wave can be produced from odd harmonic cosine waves. Eliminating harmonics distorts the output wave shape.
3. The time domain shows voltage over time, while the frequency domain shows amplitude by frequency using a Fourier series. Filtering affects the frequency spectrum and output wave shape.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for ABDON)Sarah Krystelle
The document describes Experiment #1 on a class A power amplifier. Key points:
1. The operating point (Q-point) of the amplifier was initially not centered on the AC load line, causing distortion. Adjusting the emitter resistor centered the Q-point.
2. With the centered Q-point, the maximum undistorted output voltage increased. The expected and measured output voltages matched closely.
3. A class A amplifier has low efficiency due to conduction over the full input cycle, but provides an undistorted output waveform.
The document describes experiments conducted to analyze the characteristics of active band-pass and band-stop filters. Specifically, it discusses plotting the gain-frequency response curves and determining the center frequency, bandwidth, quality factor, and phase shift for both types of filters. Sample computations are provided for an active band-pass filter to calculate the actual voltage gain, expected voltage gain, center frequency, quality factor, and percentage differences between measured and expected values. The objectives, theory, materials used, and procedures for the experiments are also outlined.
This document describes an experiment to characterize active band-pass and band-stop filters. The objectives are to determine the gain-frequency response, center frequency, bandwidth, quality factor, and phase response. For the band-pass filter, the measured and calculated results for center frequency, gain, bandwidth, and quality factor agree to within 5%. For the band-stop filter, the measured and calculated results for center frequency, gain, bandwidth, and quality factor agree to within 1%. The phase response of the band-pass filter shows the output is approximately 180 degrees out of phase with the input at the center frequency.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for PULA)Sarah Krystelle
The document describes Experiment #1 on a class A power amplifier. It involves determining the operating point (Q-point) on the DC and AC load lines, measuring the voltage gain, maximum undistorted output, and efficiency. The student is to perform steps such as calculating voltages/currents, drawing load lines, measuring gain, and adjusting the emitter resistance to center the Q-point on the AC load line. Objectives include analyzing the amplifier's DC and AC characteristics, measuring linearity and maximum output before clipping occurs.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for CAUAN)Sarah Krystelle
This document describes an experiment conducted on a Class B push-pull power amplifier. The experiment involves determining the operating point on the DC and AC load lines, centering the operating point on the AC load line, measuring the voltage gain, maximum undistorted output power, and efficiency of the amplifier. Objectives of the experiment include locating the operating point, drawing load lines, measuring voltage gain, output power, and efficiency. Components used include a transistor, resistors, capacitors, a power supply, function generator, oscilloscope and multimeter. Calculations are shown for determining load lines, voltage gain, output power and efficiency. Results are recorded for undistorted output voltage and input voltage.
This document describes an experiment on Fourier theory involving the generation of square waves and triangular waves from a series of sine and cosine waves. Key points:
1. Square and triangular waves were generated on an oscilloscope from Fourier series of sine/cosine waves at different frequencies and amplitudes.
2. Measurements showed the fundamental frequency of the generated waves matched the frequency of the individual sine/cosine waves.
3. Removing higher harmonic waves caused the generated waves to become more sinusoidal, demonstrating the role of harmonics in shaping the waveform.
This document describes an experiment on Fourier theory involving the time domain and frequency domain. It explains how square waves and triangular waves can be produced from a series of sine/cosine waves at different frequencies and amplitudes. The experiment uses function generators, an oscilloscope, and spectrum analyzer to generate and observe waveforms in both the time and frequency domains. It also examines how filtering affects periodic pulses with varying duty cycles.
1. The experiment demonstrated pulse code modulation (PCM) encoding using an analog-to-digital converter (ADC) and PCM decoding using a digital-to-analog converter (DAC).
2. The sampling frequency determined by the ADC sampling rate generator affected the time between samples on the DAC output. A lower sampling frequency resulted in a longer time between samples and vice versa.
3. The filter cutoff frequency was unaffected by changes to the sampling frequency generator. It was determined solely by the filter capacitor value. With a higher capacitor value, the cutoff frequency decreased.
This document describes an experiment involving active band-pass and band-stop filters. The objectives are to determine the gain-frequency response, quality factor, bandwidth, and phase shift of these filters. The experiment uses op-amps, capacitors, and resistors to build a multiple feedback band-pass filter and a two-pole Sallen-Key notch (band-stop) filter. Equations are provided to calculate the center frequency, bandwidth, quality factor, and voltage gain of the filters based on their circuit component values. The procedures involve simulating the filters and measuring their gain-frequency responses to determine these characteristics and compare them to theoretical calculations.
This document discusses Fourier theory and how it can be used to represent non-sinusoidal signals as a combination of sinusoidal waves of different frequencies and amplitudes. It provides examples of how square waves and triangular waves can be produced by adding together sine and cosine waves. The document also discusses the difference between analyzing signals in the time domain versus the frequency domain and how these representations provide different insights. Finally, it discusses how Fourier analysis can be used to understand the bandwidth requirements to transmit digital pulses accurately.
This document describes an experiment on Fourier theory involving the time and frequency domains. The objectives are to: 1) Produce a square wave from sine waves of different frequencies and amplitudes using Fourier series; 2) Produce a triangular wave from cosine waves using Fourier series; 3) Examine the difference between time and frequency domain plots; 4) Examine periodic pulses with different duty cycles in both domains; and 5) Examine the effect of low-pass filtering on pulses. Circuits are provided to generate square and triangular waves from Fourier series components for analysis on an oscilloscope and spectrum analyzer.
This document describes an experiment to characterize active band-pass and band-stop filters. Key steps include plotting the gain-frequency response curves for each filter, measuring the center frequency, bandwidth, and quality factor, and comparing these measured values to theoretical calculations based on the circuit components. For the band-pass filter, all measured values agreed closely with calculations. For the band-stop filter, the center frequency matched calculations to within 0.5% and other measured values matched well. The conclusion is that active filters perform similarly to passive filters in allowing or blocking certain frequency bands.
This document describes an experiment to characterize active low-pass and high-pass filters. The objectives were to determine the cutoff frequencies, gain-frequency responses, and roll-offs of second-order low-pass and high-pass filters. The experiments involved plotting the gain-frequency and phase-frequency responses of the filters using a function generator, oscilloscope, and op-amps. The measured cutoff frequencies and roll-offs matched the expected values based on the circuit components. However, when higher frequencies approached the op-amp's bandwidth limit, the high-pass filter response became band-pass-like due to the active element limitation. In conclusion, active filters are suitable for low-frequency applications where the op-
The document discusses types of amplitude modulation including double sideband amplitude modulation (DSB-AM), double sideband suppressed carrier (DSBSC), double sideband reduced carrier (DSBRC), and single sideband modulation. It also discusses power in amplitude modulation and how only 33% of total power transmitted contains useful information. Modulation index is defined as a measurement of how much a carrier wave is modulated by another signal.
This document outlines an experiment to analyze the gain, phase, and cutoff frequency responses of first-order passive low-pass and high-pass RC filters. The objectives are to plot the gain and phase responses of the filters, determine how the cutoff frequency is affected by the R and C component values, and answer related questions at various steps of the experiment.
The document discusses generating square and triangular waves using Fourier series of sine and cosine waves. It also examines signals in the time and frequency domains. Key points:
1) A square wave can be produced from a series of sine waves at different frequencies and amplitudes, with the fundamental and odd harmonics present.
2) A triangular wave results from a series of cosine waves, with the fundamental and odd harmonics.
3) Signals can be viewed in the time domain as voltage over time, or in the frequency domain as the amplitude of sine/cosine waves at different frequencies.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for AGDON)Sarah Krystelle
This experiment analyzed the operation of a class A power amplifier. Key findings include:
1) The initial operating point (Q-point) was not centered on the AC load line, resulting in output clipping.
2) Adjusting the emitter resistance centered the Q-point on the AC load line, eliminating clipping and increasing the maximum undistorted output voltage.
3) A class A amplifier has low efficiency due to conduction over the entire input cycle, but provides the most linear amplification.
SIGNAL SPECTRA EXPERIMENT 2 - FINALS (for CAUAN)Sarah Krystelle
This document describes Experiment #2 on a class B push-pull power amplifier. The objectives are to determine the dc and ac load lines, observe crossover distortion, measure voltage gain, output power, and efficiency. Sample computations are provided for voltage gain, output power, input power, and efficiency. The theory section describes class B push-pull amplifiers and how biasing the transistors slightly above cutoff can eliminate crossover distortion. Procedures are outlined to simulate and measure the amplifier's input, output, voltage gain, power output, and efficiency.
This document describes experiments performed to characterize active band-pass and band-stop filters, including plotting the gain-frequency response curves to determine cutoff frequencies and bandwidth, calculating quality factors and center frequencies, and comparing measured and expected voltage gains. Procedures are provided to implement and analyze a multiple-feedback band-pass filter and a two-pole Sallen-Key notch filter using op-amps and passive components.
This document describes an experiment to demonstrate pulse code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC). The objectives are to encode and decode analog signals using PCM and observe the effects of sampling rate and low-pass filtering. An 8-bit ADC samples an analog input and converts it to an 8-bit digital code. The DAC then reconverts the digital code to an analog output. Low-pass filtering the DAC output smooths the steps to approximate the original analog signal.
1. The document describes an experiment on Fourier theory and how signals can be represented in both the time domain and frequency domain. Square waves and triangular waves are generated from a series of sine and cosine waves (Fourier series) and plotted in both domains.
2. Low-pass filters are used to remove higher harmonics from signals. This distorts the original waveshape as more harmonics are removed. The bandwidth needed to transmit pulses with minimal distortion depends on the duty cycle.
3. Objectives include learning how square and triangular waves can be produced from Fourier series, comparing time and frequency domain plots, and examining how duty cycle and filtering affect pulses in both domains.
1. The document describes an experiment on Fourier theory involving the generation of square waves and triangular waves from a series of sine and cosine waves at different frequencies and amplitudes.
2. Key findings include that a square wave can be produced from odd harmonics of a fundamental sine wave, while a triangular wave can be produced from odd harmonic cosine waves. Eliminating harmonics distorts the output wave shape.
3. The time domain shows voltage over time, while the frequency domain shows amplitude by frequency using a Fourier series. Filtering affects the frequency spectrum and output wave shape.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for ABDON)Sarah Krystelle
The document describes Experiment #1 on a class A power amplifier. Key points:
1. The operating point (Q-point) of the amplifier was initially not centered on the AC load line, causing distortion. Adjusting the emitter resistor centered the Q-point.
2. With the centered Q-point, the maximum undistorted output voltage increased. The expected and measured output voltages matched closely.
3. A class A amplifier has low efficiency due to conduction over the full input cycle, but provides an undistorted output waveform.
The document describes experiments conducted to analyze the characteristics of active band-pass and band-stop filters. Specifically, it discusses plotting the gain-frequency response curves and determining the center frequency, bandwidth, quality factor, and phase shift for both types of filters. Sample computations are provided for an active band-pass filter to calculate the actual voltage gain, expected voltage gain, center frequency, quality factor, and percentage differences between measured and expected values. The objectives, theory, materials used, and procedures for the experiments are also outlined.
This document describes an experiment to characterize active band-pass and band-stop filters. The objectives are to determine the gain-frequency response, center frequency, bandwidth, quality factor, and phase response. For the band-pass filter, the measured and calculated results for center frequency, gain, bandwidth, and quality factor agree to within 5%. For the band-stop filter, the measured and calculated results for center frequency, gain, bandwidth, and quality factor agree to within 1%. The phase response of the band-pass filter shows the output is approximately 180 degrees out of phase with the input at the center frequency.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for PULA)Sarah Krystelle
The document describes Experiment #1 on a class A power amplifier. It involves determining the operating point (Q-point) on the DC and AC load lines, measuring the voltage gain, maximum undistorted output, and efficiency. The student is to perform steps such as calculating voltages/currents, drawing load lines, measuring gain, and adjusting the emitter resistance to center the Q-point on the AC load line. Objectives include analyzing the amplifier's DC and AC characteristics, measuring linearity and maximum output before clipping occurs.
SIGNAL SPECTRA EXPERIMENT 1 - FINALS (for CAUAN)Sarah Krystelle
This document describes an experiment conducted on a Class B push-pull power amplifier. The experiment involves determining the operating point on the DC and AC load lines, centering the operating point on the AC load line, measuring the voltage gain, maximum undistorted output power, and efficiency of the amplifier. Objectives of the experiment include locating the operating point, drawing load lines, measuring voltage gain, output power, and efficiency. Components used include a transistor, resistors, capacitors, a power supply, function generator, oscilloscope and multimeter. Calculations are shown for determining load lines, voltage gain, output power and efficiency. Results are recorded for undistorted output voltage and input voltage.
This document describes an experiment on Fourier theory involving the generation of square waves and triangular waves from a series of sine and cosine waves. Key points:
1. Square and triangular waves were generated on an oscilloscope from Fourier series of sine/cosine waves at different frequencies and amplitudes.
2. Measurements showed the fundamental frequency of the generated waves matched the frequency of the individual sine/cosine waves.
3. Removing higher harmonic waves caused the generated waves to become more sinusoidal, demonstrating the role of harmonics in shaping the waveform.
This document describes an experiment on Fourier theory involving the time domain and frequency domain. It explains how square waves and triangular waves can be produced from a series of sine/cosine waves at different frequencies and amplitudes. The experiment uses function generators, an oscilloscope, and spectrum analyzer to generate and observe waveforms in both the time and frequency domains. It also examines how filtering affects periodic pulses with varying duty cycles.
1. The experiment demonstrated pulse code modulation (PCM) encoding using an analog-to-digital converter (ADC) and PCM decoding using a digital-to-analog converter (DAC).
2. The sampling frequency determined by the ADC sampling rate generator affected the time between samples on the DAC output. A lower sampling frequency resulted in a longer time between samples and vice versa.
3. The filter cutoff frequency was unaffected by changes to the sampling frequency generator. It was determined solely by the filter capacitor value. With a higher capacitor value, the cutoff frequency decreased.
This document describes an experiment involving active band-pass and band-stop filters. The objectives are to determine the gain-frequency response, quality factor, bandwidth, and phase shift of these filters. The experiment uses op-amps, capacitors, and resistors to build a multiple feedback band-pass filter and a two-pole Sallen-Key notch (band-stop) filter. Equations are provided to calculate the center frequency, bandwidth, quality factor, and voltage gain of the filters based on their circuit component values. The procedures involve simulating the filters and measuring their gain-frequency responses to determine these characteristics and compare them to theoretical calculations.
This document discusses Fourier theory and how it can be used to represent non-sinusoidal signals as a combination of sinusoidal waves of different frequencies and amplitudes. It provides examples of how square waves and triangular waves can be produced by adding together sine and cosine waves. The document also discusses the difference between analyzing signals in the time domain versus the frequency domain and how these representations provide different insights. Finally, it discusses how Fourier analysis can be used to understand the bandwidth requirements to transmit digital pulses accurately.
This document describes an experiment on Fourier theory involving the time and frequency domains. The objectives are to: 1) Produce a square wave from sine waves of different frequencies and amplitudes using Fourier series; 2) Produce a triangular wave from cosine waves using Fourier series; 3) Examine the difference between time and frequency domain plots; 4) Examine periodic pulses with different duty cycles in both domains; and 5) Examine the effect of low-pass filtering on pulses. Circuits are provided to generate square and triangular waves from Fourier series components for analysis on an oscilloscope and spectrum analyzer.
This document describes an experiment to characterize active band-pass and band-stop filters. Key steps include plotting the gain-frequency response curves for each filter, measuring the center frequency, bandwidth, and quality factor, and comparing these measured values to theoretical calculations based on the circuit components. For the band-pass filter, all measured values agreed closely with calculations. For the band-stop filter, the center frequency matched calculations to within 0.5% and other measured values matched well. The conclusion is that active filters perform similarly to passive filters in allowing or blocking certain frequency bands.
This document describes an experiment to characterize active low-pass and high-pass filters. The objectives were to determine the cutoff frequencies, gain-frequency responses, and roll-offs of second-order low-pass and high-pass filters. The experiments involved plotting the gain-frequency and phase-frequency responses of the filters using a function generator, oscilloscope, and op-amps. The measured cutoff frequencies and roll-offs matched the expected values based on the circuit components. However, when higher frequencies approached the op-amp's bandwidth limit, the high-pass filter response became band-pass-like due to the active element limitation. In conclusion, active filters are suitable for low-frequency applications where the op-
The document discusses types of amplitude modulation including double sideband amplitude modulation (DSB-AM), double sideband suppressed carrier (DSBSC), double sideband reduced carrier (DSBRC), and single sideband modulation. It also discusses power in amplitude modulation and how only 33% of total power transmitted contains useful information. Modulation index is defined as a measurement of how much a carrier wave is modulated by another signal.
This document outlines an experiment to analyze the gain, phase, and cutoff frequency responses of first-order passive low-pass and high-pass RC filters. The objectives are to plot the gain and phase responses of the filters, determine how the cutoff frequency is affected by the R and C component values, and answer related questions at various steps of the experiment.
This document provides information about different classes of amplifiers:
1. Class A amplifiers have the transistor conducting during the entire cycle of the input signal, providing minimal distortion but lower efficiency.
2. Class B amplifiers only conduct during half of the input signal cycle, improving efficiency but introducing crossover distortion.
3. Class AB amplifiers reduce crossover distortion by adding a small bias current, keeping transistors slightly on during both halves of the cycle.
This document appears to be an assignment submission for a communications course. It was submitted by Berverlyn B. Agdon on July 11, 2011 to Eng'r. Grace Ramones for Assignment #3 on the topic of frequency modulation. The document relates to studies at the National College of Science and Technology located in Dasmariñas City, Cavite, Philippines.
The document announces the officers of the Mathematics Society at the National College of Science and Technology for the 2011-2012 academic year. It lists the names and contact information of the president, vice presidents, secretary, treasurer, auditor, public relations officer, and year representatives. It also informs the Office of Student Affairs that the Mathematics Society intends to continue its functions and services to students. Finally, it outlines the Mathematics Society's student development program for 2011-2012, which includes events like elections, tutorials, competitions, seminars, and inter-school olympiads.
This document discusses frequency modulation (FM) principles and advantages. It provides details on:
1) How FM works by varying the carrier frequency, not amplitude, in proportion to the modulating signal to transmit information.
2) The benefits of FM include resilience to noise and interference, making it suitable for high-quality audio broadcasts. It also allows modulation at low transmitter power stages using efficient non-linear amplifiers.
3) Additional topics covered are phase modulation which indirectly produces FM, frequency deviation rates and amounts, and sidebands generated during modulation.
1) The document describes an experiment on pulse-code modulation (PCM) using an analog-to-digital converter (ADC) and a digital-to-analog converter (DAC).
2) The objectives are to demonstrate PCM encoding and decoding, examine how sampling rate relates to analog signal frequency, and study the effect of low-pass filtering on the DAC output.
3) The conclusion is that PCM converts analog voltages to binary codes using an ADC for encoding and a DAC for decoding, and the sampling rate must be at least twice the highest analog signal frequency to retain all information.
This document discusses different classes of power amplifiers, including class A, class B, class AB, and push-pull amplifiers. It provides details on the operating principles, biasing, power efficiency, and output characteristics of each type. Key points include: Class A amplifiers have output current flowing for the full input cycle, leading to low efficiency. Class B amplifiers only conduct for half the input cycle. Class AB provides a small amount of bias to increase conduction. Push-pull amplifiers use two transistors connected out of phase to increase power and gain.
This document describes an experiment on low-pass and high-pass filters. It includes the theoretical background of different filter types and how to analyze them using Bode plots. The experiment uses resistor-capacitor filters to study low-pass and high-pass behavior. For the low-pass filter, the cutoff frequency is measured from the Bode plot and matches the calculated value based on the component values. Changing the resistor changes the cutoff frequency as expected, while maintaining the same roll-off rate.
This document describes an experiment on passive low-pass and high-pass filters. The objectives are to analyze the gain-frequency and phase-frequency responses of first-order RC filters and determine how component values affect cutoff frequency. Low-pass and high-pass RC filters are modeled in simulation software. For both filters, the cutoff frequency, gain, and phase responses are measured from Bode plots and compared to theoretical values. The results show the cutoff frequency changes as expected when resistance or capacitance values are altered.
This document provides information on operational amplifiers (op amps) including:
1) It defines an op amp as a high-performance dc amplifying circuit containing transistors that has features like high gain, high input resistance, and low output resistance.
2) It discusses the history and development of op amps from early bipolar transistor designs to modern CMOS and BiFET technologies.
3) It describes common op amp circuit configurations like inverting and non-inverting amplifiers, comparators, summing amplifiers, integrators, and voltage followers. Circuit diagrams and explanations of their theory and operation are provided.
This document outlines objectives and procedures for analyzing low-pass and high-pass filters. It includes plotting gain and phase responses, determining cutoff frequencies, and observing how component values affect cutoff frequency. Key points are:
- Low-pass filters pass low frequencies and reject high frequencies, with output dropping 20dB/decade above cutoff.
- High-pass filters pass high frequencies and reject low frequencies, with output dropping 20dB/decade below cutoff.
- Cutoff frequency is where output drops 3dB and is calculated from component values.
- Phase shifts from 0 to -90 degrees for low-pass, and 0 to 90 degrees for high-pass, being 45 degrees at cutoff.
The document provides details about operational amplifiers including:
1. Operational amplifiers are high-gain amplifiers used to perform computing or transfer functions like filtering. They have very high input impedance and low output impedance.
2. Common op-amp configurations include inverting and non-inverting amplifiers, comparators, integrators, differentiators, and more.
3. Op-amps can be used to simulate components like inductors through circuits like the inductance gyrator.
1. Microwave remote sensing uses radar and radiometers to measure Earth's surface.
2. Radar is unaffected by clouds and can image day/night, detecting variations in surface roughness and moisture. Radiometers measure microwave brightness temperature related to kinetic temperature and emissivity.
3. Key applications include radar altimeters to measure ocean topography, scatterometers to estimate wind speed over oceans, and synthetic aperture radar for fine-scale surface mapping.
This document discusses techniques for measuring amplitude modulation (AM) and frequency modulation (FM) using a spectrum analyzer. It describes how to use a spectrum analyzer to measure the degree of AM by viewing the modulated signal in both the time and frequency domains. The modulation index m can be calculated from measurements of the peak, minimum and carrier amplitudes. Very low levels of AM below 1% can be measured using markers to determine amplitude ratios. Frequency modulation measurements using a spectrum analyzer are also discussed.
This document describes an experiment involving active low-pass and high-pass filters. The objectives are to: plot the gain-frequency response and determine the cutoff frequency of a second-order low-pass active filter; plot the gain-frequency response and determine the cutoff frequency of a second-order high-pass active filter; determine the roll-off in dB per decade for a second-order filter; and plot the phase-frequency response of a second-order filter. The procedures involve using an op-amp, capacitors, and resistors to build second-order low-pass and high-pass Sallen-Key Butterworth filters. Key measurements and calculations are made to analyze the gain-frequency response and determine the cutoff
The dB gain at the 3dB point is:
1.006 dB
The frequency at the 3dB point is: 100 Hz
Step 6 Calculate the cutoff frequency (fc) based on the frequency at the 3dB point.
fc = 100 Hz
Question: How does the calculated cutoff frequency in Step 6 compare with the expected
cutoff frequency based on the circuit component values?
The calculated cutoff frequency in Step 6 is equal to the expected cutoff
frequency based on the circuit component values, which is 100 Hz.
Step 7 Determine the roll-off in dB/decade based on the Bode plot.
Roll-off = -40 dB/decade
1. The document describes experiments performed to analyze the frequency response of second-order low-pass and high-pass active filters. Bode plots were generated to determine the cutoff frequencies and rolloff slopes.
2. For the low-pass filter, the measured cutoff frequency and rolloff matched expectations for a two-pole filter. However, when higher component values were used, the cutoff exceeded the op-amp's bandwidth.
3. Similarly, the high-pass filter response matched predictions except at high frequencies, where it resembled a bandpass response due to the op-amp's bandwidth limitation.
This document describes an experiment to demonstrate pulse code modulation (PCM) using an analog-to-digital converter (ADC) and digital-to-analog converter (DAC). The experiment will encode an analog input signal using an 8-bit ADC and decode it back to an analog output using an 8-bit DAC. The sampling rate of the ADC will determine how well high frequency components of the original analog signal are represented. Passing the DAC's staircase output through a low-pass filter will smooth it into a representation that is closer to the original analog waveform.
1. The document describes the design of a dynamo-speedometer system that uses the rotational power of a bicycle to determine its speed.
2. The system works by converting the AC voltage output of the dynamo into a regulated DC voltage, then measuring the frequency of the AC signal to determine the rotational speed.
3. The speed is determined either by a digital circuit that counts pulses and displays the speed on 7-segment displays, or by a microcontroller that uses timers to measure frequency and displays the speed on an LCD screen.
This document provides information on the 74F08 integrated circuit, which is a quad two-input AND gate. It includes specifications on propagation delay, supply current, pin configurations, logic diagram, input/output loading, and absolute maximum ratings. The 74F08 is available in commercial and industrial temperature ranges from -40°C to +85°C.
The buck converter simulation example evaluates the switching waveforms and power switch voltages and currents. The specifications include a voltage output of 5V from an input voltage ranging from 7-40V. Inductor and capacitor values are selected to be 330uH and 330uF respectively. Simulation results are obtained for the switching waveforms, power switch voltages and currents using the average models with analysis directives to skip the breakpoints for a 10ms transient simulation.
The document presents information on digital to analog conversion (DAC). It discusses the basic concept of DAC, where a digital input is converted to a proportional analog output. It then describes two common types of DAC - the weighted resistor DAC and R-2R ladder DAC. Applications of DACs are also highlighted, such as in digital audio, function generators, and motor controllers. The document provides details on the circuit design and output calculation for both weighted resistor and R-2R ladder DACs. It concludes that the R-2R ladder DAC only requires two resistor values but has slower conversion than the weighted resistor DAC.
The Arduino is described as being low cost, easy to use, open source and compatible with multiple platforms. The initial labs focus on basics like blinking an LED and interfacing with the serial port. Later labs introduce communicating with GPS devices and integrating multiple devices. The document outlines various common electronic components that can be interfaced with Arduino like displays, sensors and more. It also defines some common terms used and provides instructions for setting up the Arduino software and board. Contact information is provided for further queries.
A digital to analog converter (DAC) accepts a binary input and produces a proportional analog output signal. A 4-bit DAC has 4 digital inputs representing the 4 bits, with the most significant bit (MSB) as d0 and least significant bit (LSB) as d3. The output voltage v0 is plotted against all possible 16 input combinations. An inverted R/2R ladder DAC uses identical resistors and voltage scaling instead of resistor scaling and a common reference used in a binary-weighted resistor DAC. It uses additional series resistors between nodes for voltage dropping. In a 3-bit R/2R ladder DAC, the binary input 001 connects switches to ground or the inverting op
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The document describes an 8-bit pipelined analog-to-digital converter (ADC) with a selectable resolution of 5-8 bits. The ADC was fabricated in a 0.13-micron CMOS process and achieves an effective number of bits of 6.10 in 8-bit mode with a 162 MHz input signal. Key aspects of the ADC include double sampling to relax amplifier settling times, redundant sign digit correction to compensate comparator offsets, and a two-stage op-amp design to provide sufficient gain and signal headroom given the low 1.2V supply voltage. Measured performance meets the requirements for medium resolution and sampling rate ADCs in modern synthetic aperture radar systems.
This document discusses digital to analog converters (DACs). It begins by defining analog and digital signals and what a DAC is. It then describes two common types of DACs: (1) weighted resistor DACs, which use a series of weighted resistors to convert digital codes to analog voltages; and (2) R-2R ladder DACs, which only require two resistor values and are easier to implement accurately. The document concludes by listing some applications of DACs such as digital audio players, signal generators, and motor controllers.
This document describes a simple 0-5V digital voltmeter circuit using an 8051 microcontroller. The circuit uses an ADC0804 analog-to-digital converter to convert the input voltage to a digital value which is then displayed on a 7-segment display. The program controls the ADC to get a digital reading, manipulates the value to display it on the display properly, and multiplexes the display digits by activating the display driver transistors at different times.
1. Analog to digital converters (ADCs) sample analog signals and convert them into digital words. This allows analog signals from sensors to be processed digitally.
2. The conversion process has two steps - quantization breaks down the analog value into discrete levels, and encoding assigns a digital code to each level. For example, a 3-bit ADC of a 0-10V signal quantizes it into 8 levels separated by 1.25V and encodes each with a 3-bit binary code.
3. There are several types of ADCs including flash, successive approximation, delta-sigma, and dual slope. Flash ADCs are fastest but most expensive, while successive approximation and dual slope ADCs are slower
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The document is a presentation on digital to analog conversion (DAC) submitted by three students to their lecturer. It provides an overview of DAC, including definitions and applications. It describes the operation of two common DAC types: weighted resistor DAC and R-2R ladder DAC. For each, it explains how the output analog voltage is determined from the digital input and compares their advantages and disadvantages.
The document summarizes a high-side power switch component. It includes a block diagram showing an N-channel MOSFET power transistor with input, enable, and output pins. It describes the characteristics of the input pin, enable pin, on-state resistance, and turn on/off timing. Simulation results are shown comparing the component's behavior to measurements for various operating conditions.
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This document summarizes the modeling of a Power MOSFET transistor. It includes:
1) Details of the MOSFET part number, manufacturer, and remarks on the model.
2) Descriptions and parameters for the Pspice model of the MOSFET.
3) Results of circuit simulations characterizing the MOSFET's transconductance, Vgs-Id curve, gate charge, switching time and other properties.
4) Comparisons of the simulation results to manufacturer measurement data showing good agreement.
Digital voltmeter using 89c51 microcontrollerSaylee joshi
Voltmeter is a voltage measuring instrument.
We can measure the potential difference between any two points in an electrical network using voltmeter.
There are two types of voltmeter as analog voltmeter and digital voltmeter.
Analog voltmeter moves pointer on a scale but it has some limitations like accuracy of few percent of full scale.
Digital voltmeter can display numerical value of voltage on a display by use of analog to digital converter (ADC).
All the data processing and manipulating is in digital form, so it is very essential to use ADC.
We have used ADC0804 analog-to-digital converter IC. The range of input voltage is 0-15V. Here the input voltage should be DC voltage so as to get the steady output on the LCD display.
If you give the AC voltage as an input, it will display continuously running numbers as the nature of AC voltage.
SIGNAL SPECTRA EXPERIMENT AMPLITUDE MODULATIONSarah Krystelle
1. The document describes an experiment on amplitude modulation (AM) involving modulating a carrier signal with different modulation indexes and frequencies.
2. Key objectives are to demonstrate AM signals in the time and frequency domains, determine modulation indexes and bandwidths, and compare side frequency levels.
3. Amplitude modulation varies the amplitude of a carrier signal based on an information-carrying modulating signal. This generates sidebands above and below the carrier frequency.
SIGNAL SPECTRA EXPERIMENT AMPLITUDE MODULATION COPY 2Sarah Krystelle
This experiment demonstrates amplitude modulation (AM) using a circuit that multiplies a carrier signal with a modulating signal and adds the results.
1. The experiment showed AM signals in the time and frequency domains for different modulation indexes. In the time domain, the envelope matched the modulating signal.
2. For 100% modulation, the sideband voltages were half the carrier voltage, matching expectations. The bandwidth matched the modulating frequency.
3. Reducing the modulating signal amplitude to 0.5 V resulted in a modulation index near 50%, close to the expected value, demonstrating the circuit can produce AM signals.
This document describes an experiment on amplitude modulation. The objectives are to demonstrate AM in the time and frequency domains, determine modulation index from plots, and examine how modulation index affects sideband levels. The experiment uses a circuit to multiply a carrier and modulating signal, producing an AM carrier viewed on an oscilloscope in the time domain and a spectrum analyzer in the frequency domain. For a modulation index of 1, the sideband voltage is half the carrier voltage as expected. Changing the modulating signal amplitude produces a lower modulation index as seen in the modulated carrier plot.
1. The document describes an experiment on amplitude modulation (AM) that demonstrates AM in the time and frequency domains for different modulation indexes and modulating frequencies.
2. Key objectives are to observe the modulation index, sideband frequencies, bandwidth, and power distribution between the carrier and sidebands for AM signals.
3. The experiment uses a circuit that multiplies a carrier signal with a modulating signal to generate an AM signal, which is then observed on an oscilloscope in the time domain and a spectrum analyzer in the frequency domain.
The document describes an experiment on amplitude modulation (AM). The objectives are to demonstrate AM signals in the time and frequency domains for different modulation indexes and frequencies. Key aspects covered include modulation index, sideband frequencies, bandwidth, and power distribution between the carrier and sidebands. The experiment uses function generators, an oscilloscope, and spectrum analyzer to generate and analyze AM signals.
1. The document describes an experiment on amplitude modulation (AM) that aims to demonstrate AM in the time and frequency domains for different modulation indexes and frequencies.
2. Key objectives are to determine modulation index, side frequency levels, signal bandwidth, and effects of complex modulation.
3. AM involves varying the amplitude of a carrier wave using a modulating signal, generating sidebands above and below the carrier frequency. The bandwidth occupied depends on the modulating signal frequencies.
1) The document describes an experiment on amplitude modulation (AM) involving demonstrating AM signals in the time and frequency domains for different modulation indexes and frequencies.
2) Key aspects of AM are discussed, including how the modulation index is defined and relates to percent modulation. Modulation indexes above 1 cause overmodulation and distortion.
3) AM generates sidebands above and below the carrier frequency by the modulating frequency. The bandwidth occupied depends on the highest modulating frequency components.
This document describes an experiment on amplitude modulation. The objectives are to demonstrate AM in the time and frequency domains for different modulation indexes and frequencies. The experiment uses a circuit to mathematically multiply a carrier signal with a modulating signal. Key findings include:
- For a 5 kHz modulating signal, the modulation index was 100% and sideband frequencies were 5 kHz from the 100 kHz carrier.
- Reducing the modulating signal to 0.5 V reduced the modulation index to 51%, as expected based on the signal amplitudes.
- Sideband voltage levels were half the carrier voltage for 100% modulation, matching theoretical calculations.
This experiment examines amplitude modulation (AM) using a circuit that mathematically multiplies a carrier signal and a modulating signal.
When the modulating signal amplitude is 1 V, the modulation index is 100% based on both calculation and observation of the modulated carrier waveform. The frequency spectrum shows sidebands separated from the carrier by the modulating frequency.
Reducing the modulating signal to 0.5 V yields a modulation index of 50% as expected. Overall the experiment demonstrates the generation of an AM signal and measurement of modulation index from the signal waveform and spectrum.
This document describes an experiment on amplitude modulation (AM). The objectives are to demonstrate AM in the time and frequency domains, determine modulation index and bandwidth, and examine how sideband power depends on modulation index. The experiment uses a circuit to mathematically multiply a carrier and modulating signal. Measurements are made on an oscilloscope in the time domain and a spectrum analyzer in the frequency domain. Results show the expected relationships between carrier, sideband frequencies and voltages, and how modulation index impacts bandwidth and sideband power. Changing the modulating signal amplitude alters the measured modulation index as expected.
This document discusses Fourier analysis of signals in the time and frequency domains. It explains that any non-sinusoidal periodic signal can be represented as a sum of sinusoidal waves of different frequencies and amplitudes. Signals are normally expressed in the time domain but Fourier theory allows expressing them in the frequency domain. The frequency spectrum reveals the bandwidth needed to transmit the signal with minimal distortion. Fourier analysis is useful for analyzing digital pulses, and the duty cycle of a periodic pulse train affects its frequency spectrum. Sample circuits are provided to generate square and triangular waves using Fourier series approximations.
1. The document describes experiments on representing non-sinusoidal signals as a sum of sinusoidal waves using Fourier analysis and examining signals in both the time and frequency domains.
2. It involves generating square and triangular waves from Fourier series of sine and cosine waves and observing the effects of removing harmonics on the output waveform.
3. The experiments aim to demonstrate the differences between time and frequency domain representations and determine the bandwidth required to transmit periodic pulses with minimal distortion.
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Objectives
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)
Balane, Maycen M. September 20, 2011
Signal Spectra and Signal Processing/BSECE 41A1 Score:
Engr. Grace Ramones
Instructor
2. Objectives:
1. Demonstrate PCM encoding using an analog-to-digital converter (ADC).
2. Demonstrate PCM encoding using an digital-to-analog converter (DAC)
3. Demonstrate how the ADC sampling rate is related to the analog signal frequency.
4. Demonstrate the effect of low-pass filtering on the decoder (DAC) output.
4. Data Sheet:
Materials
One ac signal generator
One pulse generator
One dual-trace oscilloscope
One dc power supply
One ADC0801 A/D converter (ADC)
One DAC0808 (1401) D/A converter (DAC)
Two SPDT switches
One 100 nF capacitor
Resistors: 100 Ω, 10 kΩ
Theory
Electronic communications is the transmission and reception of information over a communications
channel using electronic circuits. Information is defined as knowledge or intelligence such as audio
voice or music, video, or digital data. Often the information id unsuitable for transmission in its
original form and must be converted to a form that is suitable for the communications system.
When the communications system is digital, analog signals must be converted into digital form prior
to transmission.
The most widely used technique for digitizing is the analog information signals for 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 levels into a sequence of binary codes, with each binary number that is proportional to the
magnitude of the voltage level sampled. Translating analog voltages into binary codes is called A/D
conversion, digitizing, or encoding. The device used 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 convert each analog voltage into
its binary code. During the ADC conversion time, the analog input voltage must remain constant.
The conversion time for most modern A/D converters is short enough so that the analog input
voltage will not change during the conversion time. For high-frequency information signals, the
analog voltage will change during the conversion time, introducing an error called 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 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 of sampling,
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 conversion time.
The rate at which the analog input voltage is sampled is called the sampling rate. The ADC
conversion time puts a limit on the sampling rate because the next sample cannot be read until the
5. previous conversion time is complete. The sampling rate is important because it determines the
highest analog signal frequency that can be sampled. In order to retain the high-frequency
information in the analog signal acting sampled, a sufficient number of samples must be taken so
that all of the voltage changes in the waveform are adequately represented. Because a modern ADC
has a very short conversion time, a high sampling rate is possible resulting in better reproduction of
high0frequency analog signals. Nyquist frequency is equal to twice the highest analog signal
frequency component. Although theoretically analog signal can be sampled at the Nyquist
frequency, in practice the sampling rate is usually higher, depending on the application and other
factors such as channel bandwidth and cost limitations.
In a PCM system, the binary codes generated by the ADC are converted into serial pulses and
transmitted over the communications medium, or channel, to the PCM receiver one bit at a time. At
the receiver, the serial pulses are converted back to the original sequence of parallel binary codes.
This sequence of binary codes is reconverted into a series of analog voltage levels in a D/A converter
(DAC), often called a decoder. In a properly designed system, these analog voltage levels should be
close to the analog voltage levels sampled at the transmitter. Because the sequence of binary codes
applied to the DAC input represent a series of dc voltage levels, the output of the DAC has a
staircase (step) characteristic. Therefore, the resulting DAC output voltage waveshape is only an
approximation to the original analog voltage waveshape at the transmitter. These steps can be
smoothed out into an analog voltage variation by passing the DAC output through a low-pass filter
with a cutoff frequency that is higher than the highest-frequency component in the analog
information signal. The low-pass filter changes the steps into a smooth curve by eliminating many of
the harmonic frequency. If the sampling rate at the transmitter is high enough, the low-pass filter
output should be a good representation of the original analog signal.
In this experiment, pulse code modulation (encoding) and demodulation (decoding) will be
demonstrated using an 8-bit ADC feeding an 8-bit DAC, as shown in Figure 2-1. This ADC will convert
each of the sampled analog voltages into 8-bit binary code as that represent binary numbers
proportional to the magnitude of the sampled analog voltages. The sampling frequency generator,
connected to the start-of conversion (SOC) terminal on the ADC, will start conversion at the
beginning of each sampling pulse. Therefore, the frequency of the sampling frequency generator will
determine the sampling frequency (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 to the output (OE)
terminal on the ADC will keep the digital output connected to the digital bus. The DAC will convert
these digital codes back to the sampled analog voltage levels. This will result in a staircase output,
which will follow the original analog voltage variations. The staircase output of the DAC feeds of a
low-pass filter, which will produce a smooth output curve that should be a close approximation 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 and capacitor C determine the cutoff frequency (fC) of the low-
pass filter, which is determined from the equation
6. Figure 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
In an actual PCM system, the ADC output would be transmitted to serial format over a transmission
line to the receiver and converted back to parallel format before being applied to the DAC input. In
Figure 23-1, the ADC output is connected 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 µs
Step 2 Calculate the sampling frequency (fS) based on the time between samples (TS)
fS = 250 kHz
7. Question: How did the measure sampling frequency compare with the frequency of the sampling
frequency generator?
It is almost equal. 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?
The sampling frequency is more than 20 times of the analog input frequency. Yes it is more than
twice the analog input frequency.
How did the sampling frequency compare with the Nyquist frequency?
It is 2π or 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 and the DAC staircase
output?
They are both digital.
Step 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 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 oscilloscope
Questions: What happened to the DAC output after filtering? Is the filter output waveshape a close
representation of the analog input waveshape?
The DAC output became analog. Yes, it is a close representation of the analog input. The
DAC lags the input waveshape.
Step 5 Calculate the cutoff frequency (fC) of the low-pass filter.
fC = 15.915 kHz
Question: How does the filter cutoff frequency compare with the analog input frequency?
They have difference of approximately 6 kHz.
Step 6 Change the filter capacitor (C) to 20 nF and calculate the new cutoff frequency (fC).
fC = 79.577 kHz
Step 7 Bring down the oscilloscope enlargement and run the simulation to completion 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 = 9.5µs
8. Question: How does the time between the samples in Step 8 compare with the time between the
samples in Step 1?
It doubles.
Step 9 Calculate the new sampling frequency (fS) based on the time between the samples (TS)
in Step 8?
fS=105.26Hz
Question: How does the new sampling frequency compare with the analog input frequency?
It is 10 times the analog input frequency.
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 = 19µs
Question: 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 kHz
Question: 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=10kHz
Question: 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.
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 = 66.5µs
9. Question: How does the time between samples in Step 15 compare with the time between samples
in Step 11?
It is 3.5 times more 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 kHz
Question: 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?
The Nyquist frequency is 6.28 times larger than the sampling 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=10kHz
Question: How does the frequency of the filter output compare with the frequency of the analog
input? Was this expected based on the sampling frequency?
It is the same. Yes, it is expected.
10. CONCLUSION:
I conclude that ADC and DAC can be use for Pulse Code Modulation. The output waveform
produced was a staircase wave. However, the low-pass filter output is like the input analog signal. The
ADC sampling rate affects the frequency of the sampling signal. As the ADC sampling rate increases, the
frequency of the sampling signal also increases. On the other hand, the filter frequency was not affected
by the rate of the sampling generator from the ADC. The analog frequency is the same as the frequency
of the filter. The filter’s cutoff frequency is inversely proportional to the capacitor, as the capacitor
increases, the cutoff frequency decreases. The Nyquist frequency is always 6.28 times larger than the
sampling frequency.