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.
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 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.
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) 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.
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.
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.
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.
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 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.
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) 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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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 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.
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.
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 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.
SPICE MODEL of uPC24A12HF in SPICE PARK. English Version is http://www.spicepark.net. Japanese Version is http://www.spicepark.com by Bee Technologies.
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.
SPICE MODEL of uPC24A15HF in SPICE PARK. English Version is http://www.spicepark.net. Japanese Version is http://www.spicepark.com by Bee Technologies.
This document summarizes the modeling parameters and performance of a voltage regulator component. It describes the manufacturer, part number, and key electrical parameters represented in the PSpice model. Simulation results show the input-output voltage differential is within 0.1% of measured, and ripple rejection ratio is within 0.9% of measured. The output characteristic comparison shows simulation within 0.65% of measured.
The document is an assignment on operational amplifiers submitted by Sarah Krystelle P. Cauan to her instructor Engr. Grace Ramones. It contains information on:
1) The basic characteristics and idealized parameters of operational amplifiers such as infinite open loop gain and bandwidth.
2) Common op-amp circuit configurations including the inverting amplifier, non-inverting amplifier, and transresistance amplifier.
3) Equations for calculating the closed-loop gain of inverting and non-inverting amplifiers.
4) Descriptions of how feedback controls the gain in each type of circuit.
The document summarizes different classes of amplifiers, including power amplifiers, class A amplifiers, and class B amplifiers. It describes the main function of power amplifiers as delivering power to the load. It then discusses the single-ended class A amplifier circuit and its low efficiency of less than 30%. Finally, it introduces the class B push-pull amplifier circuit which uses two complementary transistors to conduct alternating half cycles, improving efficiency to around 70%.
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.
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.
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.
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.
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.
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.
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 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.
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.
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 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.
SPICE MODEL of uPC24A12HF in SPICE PARK. English Version is http://www.spicepark.net. Japanese Version is http://www.spicepark.com by Bee Technologies.
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.
SPICE MODEL of uPC24A15HF in SPICE PARK. English Version is http://www.spicepark.net. Japanese Version is http://www.spicepark.com by Bee Technologies.
This document summarizes the modeling parameters and performance of a voltage regulator component. It describes the manufacturer, part number, and key electrical parameters represented in the PSpice model. Simulation results show the input-output voltage differential is within 0.1% of measured, and ripple rejection ratio is within 0.9% of measured. The output characteristic comparison shows simulation within 0.65% of measured.
The document is an assignment on operational amplifiers submitted by Sarah Krystelle P. Cauan to her instructor Engr. Grace Ramones. It contains information on:
1) The basic characteristics and idealized parameters of operational amplifiers such as infinite open loop gain and bandwidth.
2) Common op-amp circuit configurations including the inverting amplifier, non-inverting amplifier, and transresistance amplifier.
3) Equations for calculating the closed-loop gain of inverting and non-inverting amplifiers.
4) Descriptions of how feedback controls the gain in each type of circuit.
The document summarizes different classes of amplifiers, including power amplifiers, class A amplifiers, and class B amplifiers. It describes the main function of power amplifiers as delivering power to the load. It then discusses the single-ended class A amplifier circuit and its low efficiency of less than 30%. Finally, it introduces the class B push-pull amplifier circuit which uses two complementary transistors to conduct alternating half cycles, improving efficiency to around 70%.
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.
Yes, this is what is expected for a two-pole filter. A two-pole filter rolls off at -40 dB per decade.
Step 8 Measure the phase angle at the cutoff frequency (fc) and record it on the curve
plot.
Phase angle at fc = -89.999°
Question: What was the expected phase angle at the cutoff frequency for a two-pole
filter?
The expected phase angle at the cutoff frequency for a two-pole filter is -90°.
High-Pass Active Filter
Step 9 Open circuit file FIG 3-2. Make sure that the Bode plotter settings are the
same as for the low-pass filter.
Step
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
This document provides information about operational amplifiers including:
1. It describes the characteristics and components of an operational amplifier, including very high gain, very high input impedance, and very low output impedance.
2. It explains the closed-loop operation of an operational amplifier using negative feedback, and how this stabilizes the circuit and determines the gain based on resistor values.
3. It gives examples of applications including a summing amplifier circuit that adds multiple input voltages together to produce an output voltage equal to the summed input voltages but opposite in polarity.
The document discusses different classes of amplifiers - A, B, AB, C, D, and E - based on their conduction angle.
Class A amplifiers have a conduction angle of 360 degrees, meaning the amplifying device remains on all the time. They are simple but very inefficient. Class B amplifiers have a conduction angle of 180 degrees and use two devices in a push-pull configuration to amplify opposite halves of the signal cycle, improving efficiency but introducing crossover distortion. Class AB amplifiers operate between class A and B to reduce crossover distortion. The document provides details on the characteristics and applications of different amplifier classes.
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.
Operational amplifiers are linear devices that can perform mathematical operations like addition, subtraction, integration and differentiation. They have ideal characteristics such as infinite input impedance, zero output impedance, and infinite gain. Common op-amp circuits include the inverting amplifier, non-inverting amplifier, summing amplifier, differential amplifier, integrator, and differentiator. The integrator produces an output voltage proportional to the integral of the input over time, while the differentiator produces an output proportional to the rate of change of the input voltage.
This document discusses frequency modulation (FM) and provides details about:
1) FM can be used for both analog and digital data transmission by varying the instantaneous frequency of a carrier wave.
2) In analog FM the carrier frequency varies continuously, while in digital FM it shifts abruptly between discrete frequency states.
3) FM bandwidth depends on the modulation index, with higher indices resulting in wider bandwidth signals classified as wideband FM.
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.
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 types of amplitude modulation (AM) used in radio communications, including double sideband (DSB), single sideband (SSB), and vestigial sideband modulation. It explains that AM involves modulating a high frequency carrier signal with lower frequency audio/voice signals in order to transmit electromagnetic radiation. The power in an AM signal varies according to the audio signal rather than the carrier power remaining constant. The modulation index is defined as the ratio of the audio signal amplitude to the carrier amplitude.
This document discusses types of amplitude modulation including:
- Double sideband full carrier (DSB-FC) which transmits both sidebands and the carrier.
- Double sideband suppressed carrier (DSB-SC) which transmits both sidebands but suppresses the carrier.
- Single sideband suppressed carrier (SSB-SC) which transmits either the upper or lower sideband and suppresses the carrier.
It also discusses power utilization in amplitude modulation, noting that only 33% of transmitted power is used to carry information in the sidebands, while the rest is wasted in the carrier. Finally, it defines modulation index as the ratio of modulation signal amplitude to carrier amplitude, with
The document outlines the duties and responsibilities of officers in a student organization. The Treasurer is responsible for collecting dues, keeping financial records, paying bills, and presenting financial reports. The Auditor assists the Treasurer and reviews the financial reports. The PRO is responsible for announcements and communicating with members and advisers. Officers serve one-year terms from July to June. Vacancies will trigger a special election within a month. Amendments require a two-thirds member vote.
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.
This document provides an overview of cellular network technology. It discusses key concepts such as how a cellular network divides geographic coverage into cells served by base stations, allowing frequencies to be reused across cells. It also summarizes techniques for distinguishing signals like frequency division multiple access (FDMA) and code division multiple access (CDMA). The document concludes with explanations of frequency reuse patterns, directional antenna use, broadcast messaging, paging, and handovers as mobile devices move between cells.
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Objective5
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)
Tagasa, Jerald A. September 20, 2011
Signal 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.
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
5. 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
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
6. 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
Figure 23–1 Pulse-Code Modulation (PCM)
7. 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
Question: How did the measure sampling frequency compare with
the frequency of the sampling frequency generator?
Both frequency have difference of 50 kHz.
8. How did the sampling frequency compare with the analog input
frequency? Was it more than twice the analog input frequency?
The sampling frequency is 20 times higher. 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 and the DAC staircase output?
Both outputs are both in 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 output became analog after filtering. Yes it is
close representation.
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.
9. 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
Question: 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=105.26Hz
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?
The new sampling frequency is 5 times the analog input.
10. 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
Question: 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 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?
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.
11. 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?
They are the same. For sampling frequency of 15.037 kHz,
it is expected to have same outputs.
12. CONCLUSION:
I conclude, that analog signal can be digitize for digital
communication. One way is the PCM. ADC and DAC are used for
encoding and decoding of PCM.
The ADC provides the sampling frequency. The sampling
frequency is inversely proportional to the sampling time of the
DAC output. The staircase output is the output generated by the
DAC. It is digital signal like the sampling pulse.
The filter frequency is the frequency of the analog input
frequency. The cutoff frequency is inversely proportional to the
capacitance and remain constant as the sampling frequency changes.