Digital data is a binary representation of information, whereas analog signals are continuous and represent information through characteristics such as frequency, amplitude, and phase. Modulation is the process of converting digital data into an analog signal. To illustrate this, let's examine three modulation methods using a transmission of two ASCII characters 'j;' (8-bit each) as an example.
1. Amplitude Modulation (AM):
In AM, the information signal modulates the amplitude of the carrier signal. The resulting modulated signal combines the carrier and information signals. The amplitude of the carrier changes according to the information signal. Figure 1 depicts the waveforms obtained when transmitting the digital data using AM modulation.
2. Frequency Modulation (FM):
In FM, the information signal modulates the frequency of the carrier signal. The modulated signal contains both the carrier and information signals. The frequency of the carrier changes with the information signal. Figure 2 illustrates the waveforms resulting from transmitting the digital data using FM modulation.
3. Phase Modulation (PM):
In PM, the information signal modulates the phase of the carrier signal. The modulated signal includes both the carrier and information signals. The phase of the carrier varies based on the information signal. Figure 3 shows the waveforms obtained when transmitting the digital data using PM modulation.
In summary, when transmitting the digital data with two ASCII characters 'j;' (8-bit each) as an analog signal, the resulting waveforms using AM, FM, and PM modulation are depicted in Figures 1, 2, and 3, respectively.
To know more about Digital data visit:
https://brainly.com/question/32345072
#SPJ11
In an industrial facility, both electrical power and a process heating load of 14000 kW are needed. The required heat and electrical power are supplied by a combined steam plant, where steam enters the turbine at 20 bar, 450°C and exhaust steam leaves at 2.0 bar. The isentropic efficiency of the turbine is 0.85. The process heat is provided by the turbine exhaust steam. in this facility the condensate drain from the process heater at the saturation temperature is fed back to the pump. Determine: (a) The temperature of the exhaust steam leaving the turbine (b) The mass flow rate of the steam entering the turbine (4) (c) The power supplied by the turbine. (4)
a) Temperature of the exhaust steam leaving the turbineThe temperature of the exhaust steam leaving the turbine can be calculated by using the following formula:T2 = T1 - [(T1 - T3)/ηt]whereT1 = Entering steam temperature = 450 °CT3 = Exhaust steam temperature = ?ηt = Turbine isentropic efficiency = 0.85T2 = Condensate temperature = saturation temperature at 2.0 barPFrom steam tables, the saturation temperature at 2.0 bar is 120.2 °C.
Substituting the values in the above formula, we get:T3 = 450 - [(450 - 120.2)/0.85]= 192.95 °CTherefore, the temperature of the exhaust steam leaving the turbine is 192.95 °C. (b) Mass flow rate of the steam entering the turbineThe mass flow rate of the steam entering the turbine can be calculated by using the following formula:m = Q/(h1 - h2)whereQ = Heat supplied to the process heater = 14000 kW (given)h1 = Enthalpy of steam entering the turbineFrom steam tables, h1 at 20 bar and 450 °C is 3238.1 kJ/kg.h2 = Enthalpy of exhaust steam leaving the turbineFrom steam tables, h2 at 2.0 bar and 192.95 °C is 2650.4 kJ/kg.Substituting the values in the above formula, we get:m = 14000/(3238.1 - 2650.4)= 16.48 kg/sTherefore, the mass flow rate of the steam entering the turbine is 16.48 kg/s. (c) Power supplied by the turbineThe power supplied by the turbine can be calculated by using the following formula:W = m(h1 - h2)ηtwhereW = Power supplied by the turbine = ?m = Mass flow rate of the steam entering the turbine = 16.48 kg/sh1 = Enthalpy of steam entering the turbine = 3238.1 kJ/kgh2 = Enthalpy of exhaust steam leaving the turbine = 2650.4 kJ/kgηt = Turbine isentropic efficiency = 0.85Substituting the values in the above formula, we get:W = 16.48 × (3238.1 - 2650.4) × 0.85= 18412.52 kWTTherefore, the power supplied by the turbine is 18412.52 kW.
To know more about Temperature, visit:
https://brainly.com/question/7510619
#SPJ11
Question 4 (2 M) For a load, Vrms = 110 V, theta= 15 degrees, Determine: (a) the complex and apparent powers, (b) the real and reactive powers, and (c) the power factor and the load impedance. Question 5 (2 M) A balanced Y-connected load with a phase impedance of 40+/25 92 is supplied by a balanced, positive sequence -connected source with a line voltage of 210 V. Calculate the phase currents. Use Vab as reference.
The phase currents are 2.58 - j1.59 amps.
Question 4Solution
a) The formula for complex power is: S = VI* where V is the rms voltage and I is the complex currentS = V rms I*Apparent power (S) = 110 volts x IComplex power = S= 110 I∠15° = (110 ∠15°) I
The magnitude of the complex power is: |S| = 110 WReal power (P) = Apparent power x Power factorP = S cosθ where θ is the phase angle between the voltage and the currentP = |S| cos(15°)P = 104.7 W Reactive power (Q) = Apparent power sinθ Q = |S| sin(15°)Q = 28.3 W
b) The real power is 104.7 W and the reactive power is 28.3 W.
c) Power factor (PF) = P/S = cos(15°)PF = 0.97 Load impedance (Z) = Vrms/I = 110 V / IZ = (110∠0°)/I∠-15°Z = (110∠15°)/(6.27∠15°)Z = 17.49 + j6.6 ohms
Question 5Solution
Phase Impedance = 40 + j25.92 ohms Phase voltage Vph = Vab / √3 = 210 / √3 volts Line current IL = Iphase
Impedance Z = Vph / ILZ = 40 + j25.92 ohmsVph = Iphase ZIphase = Vph / ZZ = 40 + j25.92 ohm sIphase = (210/√3) / (40 + j25.92)Iphase = 2.58 - j1.59 amps
Therefore, the phase currents are 2.58 - j1.59 amps.
To know more about phase currents visit:
brainly.com/question/33216380
#SPJ11
What data type would you use to represent each of the following things:< a. The proportion of students passing a unit double □ int □char boolean b. A passport 'number' (a sequence of letters and digits such as XN1234567) ☐ int double □char boolean c. The exact number of students attending a lecture! O int double char boolean d. The status of a remotely controlled door as being either open or closed! ☐ int double boolean char e. The precise mass in kilograms of a car battery boolean char □ int ☐ double □ String □ String □ String □ String String
a. The proportion of students passing a unit: double b. A passport 'number' (a sequence of letters and digits such as XN1234567): String c. The exact number of students attending a lecture: int d. The status of a remotely controlled door as being either open or closed: boolean e. The precise mass in kilograms of a car battery: double
A double data type would be suitable for representing the proportion of students passing a unit. It allows for decimal values, which is necessary when dealing with proportions that may not be whole numbers.
b. A passport 'number' (a sequence of letters and digits such as XN1234567): String
To represent a passport number, a String data type would be appropriate. A String can hold a sequence of characters, including letters and digits, allowing for the representation of alphanumeric passport numbers.
c. The exact number of students attending a lecture: **int**
For representing the exact number of students attending a lecture, an **int** (integer) data type can be used. Integers are used to store whole numbers without decimal places, which is suitable for counting the number of students attending a lecture.
d. The status of a remotely controlled door as being either open or closed: **boolean**
To represent the status of a remotely controlled door (open or closed), a **boolean** data type would be appropriate. Booleans can store only two possible values: true or false, which aligns with the open or closed status of the door.
e. The precise mass in kilograms of a car battery: **double**
To represent the precise mass of a car battery in kilograms, a **double** data type would be suitable. Doubles can store decimal values, allowing for the precise representation of mass measurements that may have fractional parts, such as the mass of a car battery.
Learn more about proportion here
https://brainly.com/question/21619476
#SPJ11
Please provide all the steps that have been given below.
J-K type negative edge flip-flops and logic gates.
using state table and K-map.
Create a counter using JK flipflop to count 6-3-1-4-5-2-0-7 by NI multisim , and also show the state table and K-map ,thanks and using the sequence that provided.
6-3-1-4-5-2-0-7
Design a binary counter that counts in a certain sequence using J-K type negative edge flip-flops and logic gates.
Record your workings that include:
Design steps of the counter e.g. state table and K-map.
Schematic diagram of the circuit.
Design steps of the counter using J-K type negative edge flip-flops and logic gates (state table and K-map). Sequential logic circuits are those circuits whose outputs rely on previous input values and present input values.
Flip-flops and counter are sequential logic circuits. JK flip-flop is a flip-flop that uses J-K inputs to control its state. It has two stable states, namely logic 1 and logic 0. The transition occurs at the negative clock edge. There are two inputs to the J-K flip-flop: J and K. When J and K are low, the flip-flop is in a latched state. When J and K are high, the flip-flop is in a reset state. When J is low and K is high, the flip-flop will set and when J is high and K is low, the flip-flop will reset. To create a counter using JK flip-flop to count 6-3-1-4-5-2-0-7 by NI multisim, we need to follow the following steps:
Step 1: State Table (K-Map) The state table is used to record the current state, next state, input, and output for each flip-flop in the circuit. We need to create a state table for our design.
Step 2: K-Map for J and K inputs The K-map is used to simplify the Boolean expression for J and K. We need to create K-Maps for J and K inputs for each flip-flop in the circuit.
Step 3: Simplify J and K expressions Using K-Maps, we can simplify the Boolean expressions for J and K inputs.
Step 4: Circuit Diagram Create a circuit diagram for the JK flip-flop counter.
Step 5: NI Multisim Simulation Perform NI Multisim simulation to validate the design of the JK flip-flop counter.
Step 6: Verification Verify the counter with the desired sequence of 6-3-1-4-5-2-0-7.
Step 7: Output Test the counter by connecting an LED to the output of each flip-flop in the circuit.
To know more about the output visit:
brainly.com/question/14227929
#SPJ11
the conductor size, fuse or circuit breaker size, and overload size are generally determined using the ____.
The conductor size, fuse, or circuit breaker size, and overload size are generally determined using the National Electrical Code (NEC).
National Electrical Code is a guidebook, which is a national standard published by the National Fire Protection Association (NFPA), that provides guidelines for the safe installation of electrical wiring and equipment in homes, buildings, and other facilities.
The NEC provides guidelines for wire ampacity, overcurrent protection, and maximum circuit length, among other things. The conductor size is determined by the load current, the maximum circuit length, and the conductor temperature rating.
The maximum circuit length is determined by the voltage drop, the load current, and the conductor size.Fuse and circuit breaker sizes are determined based on the current-carrying capacity of the conductor. They must be properly sized to protect the conductor from overcurrent, but not so small that they trip unnecessarily.The overload size is determined based on the load current.
Overload protection is a type of overcurrent protection that protects equipment from overheating and burning out due to excessive current. It is usually provided by thermal overload relays or electronic overload relays, which detect excessive current and disconnect the power source to the equipment.
To know more about determined visit :
https://brainly.com/question/29898039
#SPJ11
The following constant coefficient difference equation characterizes an LTI system
y[n 1] = 0.4y[n-2] 0.2x[n] +0.5x[n-1]+ 0.4x[n - 2]
a) Comment on the stability and causality of the system? b) Compute the effect of the system on the magnitude and phase of л/2 frequency content in an input signal x[n].
The effect of the system on the magnitude and phase of the л/2 frequency content in an input signal x[n] is to increase the magnitude by a factor of 1.75 and to shift the phase by -86.6°. The given constant coefficient difference equation characterizes an LTI system y[n + 1] = 0.4y[n - 2] + 0.2x[n] + 0.5x[n - 1] + 0.4x[n - 2]. C
A system is stable if and only if the impulse response is absolutely summable. To determine if the impulse response is absolutely summable, consider the impulse response of the system (i.e., the response when x[n] = δ[n]).y[n + 1] = 0.4y[n - 2] + 0.2x[n] + 0.5x[n - 1] + 0.4x[n - 2]When x[n] = δ[n], the above equation reduces to y[n + 1] = 0.4y[n - 2] + 0.2δ[n] + 0.5δ[n - 1] + 0.4δ[n - 2]This can be written as y[n + 1] - 0.4y[n - 2] = 0.2δ[n] + 0.5δ[n - 1] + 0.4δ[n - 2]Taking the Z-transform of the above equation, we getY(z)(z³ - 0.4z^-2) = X(z)(0.2 + 0.5z^-1 + 0.4z^-2)Y(z)/X(z) = (0.2 + 0.5z^-1 + 0.4z^-2)/(z³ - 0.4z^-2)The above equation is the transfer function of the system, which can be factorized asY(z)/X(z) = (2z^-1 + 1)/(z^-1 - 0.2)(z^-1 + 0.5)(z^-1 + 0.4)To ensure that the system is stable, we need to ensure that all the poles of the transfer function lie inside the unit circle.
The poles of the transfer function are obtained by setting the denominator to zero.z³ - 0.4z^-2 = 0z^-2(z^5 - 0.4) = 0z = 0.4^(1/5)e^(j(2πk/5)) for k = 0,1,2,3,4.The magnitude of the poles is less than 1, and hence the system is stable. Causality:A system is causal if its output at time n depends only on the present and past values of the input x[n]. From the given difference equation ,y[n + 1] = 0.4y[n - 2] + 0.2x[n] + 0.5x[n - 1] + 0.4x[n - 2],we can see that the output at time n + 1 depends on the input values at n, n - 1, and n - 2, and the output values at n - 2.
To know more about frequency content visit :-
https://brainly.com/question/29771414
#SPJ11
solve asap
The elevator system is: O a. None of the answers O b. Not an automated system Oc. A closed loop system O d. Oe. An open loop control system A system without control
The option that correctly describes the elevator system is option (c) A closed-loop system. Closed Loop System:A closed-loop system is a control system that employs feedback to achieve the desired output. It receives input signals from its surroundings and processes it to regulate its behavior. Closed-loop control is used in most of the control systems.
The system is also known as a feedback system because it employs feedback to reach its objective. Open Loop System: An open-loop system is one in which the output is not affected by the input. A computer-controlled lathe is an example of an open-loop control system.
The operator inputs a code for the part they want to create, and the machine executes the commands without making any adjustments depending on the environment. Feedback is not utilised in an open-loop control system.
Learn more about elevator system at https://brainly.com/question/32668414
#SPJ11
a) Draw the block diagram of the circuit required to modulate the
Y(t)=Acos(2πwt)+Bsin(2πwt) signal with QAM modulation.
b) Calculate the signals to be obtained at the output when this sign is applied to the input of the modulator you have drawn.
Here is the block diagram of the circuit required to modulate Y(t) = A cos(2πwt) + B sin(2πwt) signal with QAM modulation:Explanation:The QAM stands for Quadrature Amplitude Modulation. It is used for transmitting two digital bit streams or two analog signals by altering the amplitude of two carrier waves, usually sinusoidal.
One of these carriers is in-phase (I) with the reference carrier and the other one is in quadrature (Q) with the reference carrier.A QAM modulator includes two modulators, I modulator and Q modulator. The block diagram of QAM modulator is shown below:It can be seen that the modulator includes two modulation circuits, one for the in-phase signal and the other for the quadrature signal.Each of these two circuits contains the following blocks:Multiplier (one per circuit)Bandpass filter (one per circuit)Summing circuit (one per circuit)
So, the above diagram shows that the QAM modulator needs two modulators for processing two carrier signals.The signals to be obtained at the output when this signal is applied to the input of the modulator are:The modulated signal x(t) and the carrier wave cos(wt) are multiplied and passed through a low-pass filter to obtain I(t).The modulated signal x(t) and the carrier wave sin(wt) are multiplied and passed through a low-pass filter to obtain Q(t).I(t) and Q(t) are combined in the summing circuit to get the final output.
To know more about waves visit:
https://brainly.com/question/29334933
#SPJ11
Which of the following types of valves could flow aid in closing or opening the valve
Flow aids are mechanical devices designed to help move the product through the pipeline.
They can be applied in both gravity-fed and pumped systems,
and are used to ensure the product flows efficiently and effectively.
The flow aid is an ideal tool for use in applications with low flow,
abrasive, or sticky products.
It's also commonly used to help remove any excess product that might have accumulated in the line, which helps to improve the efficiency of the system.
The types of valves that could flow aid in closing or opening the valve are Pressure Relief Valves,
Gate Valves, Globe Valves, Diaphragm Valves, Needle Valves, Butterfly Valves, and Ball Valves.
Pressure relief valves are commonly used in the oil and gas industry to protect equipment and pipelines from overpressure.
These valves can be set to open at a specific pressure, which helps to prevent damage to the pipeline.
Gate valves are used to regulate the flow of fluids in pipelines.
They are commonly used in the oil and gas industry to control the flow of crude oil and natural gas.
Globe valves are similar to gate valves but are designed to regulate the flow of fluids more precisely.
They are commonly used in the chemical and petrochemical industries.
Diaphragm valves are used to regulate the flow of fluids in pipelines.
They are commonly used in the food and beverage industry to control the flow of liquids.
Needle valves are used to regulate the flow of fluids in pipelines.
They are commonly used in the chemical and petrochemical industries to control the flow of corrosive liquids.
Butterfly valves are used to regulate the flow of fluids in pipelines.
To know more about product visit:
https://brainly.com/question/31812224
#SPJ11
where ceiling joists or rafters are cut to make an opening (skylight, dormer etc.) the load from the cut members must be transferred to adjacent members. the framing installed to do this is called a
When ceiling joists or rafters are cut to make an opening (skylight, dormer, etc.), the load from the cut members must be transferred to adjacent members. The framing installed to do this is called a header.
What is a header?A header is a member of the framing that spans an opening in a building and is used to transmit loads from above to adjacent members. Headers are used in window, door, and skylight openings.
Headers are frequently found in frame construction and are supported by walls or beams, depending on the design.Header joists are the final members installed in a floor or ceiling framing system, and they support the exterior walls and interior partitions. Header joists are usually the same size as the floor joists in a given structure.
However, when the span of the header exceeds 4 feet, it must be strengthened by doubling it or adding thickness to it.The ends of headers are supported by trimmer joists, which transfer the header's load to the next adjacent joist in the framing.
Learn more about framing members at
https://brainly.com/question/29307343
#SPJ11
Can the bandpass and bandstop frequency shaping characteristics be implemented with first-order transfer functions, possibly employing one zero and/or one pole at most? Explain.
The bandpass and bandstop frequency shaping characteristics can be implemented with first-order transfer functions by employing one zero and/or one pole at most. Let's take a look at the explanation below:
In order to achieve a bandpass characteristic, the zero and pole positions must be separated by the same ratio as their magnitude, which is a fixed ratio known as the quality factor. Because the Q factor is equal to the frequency at which the zero and pole are separated divided by their spacing, the filter's bandwidth can be adjusted by adjusting the Q factor. The larger the Q factor, the narrower the bandwidth will be, while the smaller the Q factor, the wider the bandwidth will be. As a result, a first-order bandpass filter may be employed to generate bandpass filtering characteristics in a variety of applications.
On the other hand, a bandstop filter is constructed by cascading a high-pass and a low-pass filter, and in order to achieve a bandstop characteristic, the zero and pole positions must be separated by the same ratio as their magnitude, which is also a fixed ratio known as the quality factor. Because the Q factor is equal to the frequency at which the zero and pole are separated divided by their spacing, the filter's bandwidth can be adjusted by adjusting the Q factor. The larger the Q factor, the narrower the bandwidth will be, while the smaller the Q factor, the wider the bandwidth will be. As a result, a first-order bandstop filter may be employed to generate bandstop filtering characteristics in a variety of applications. In summary, the bandpass and bandstop frequency shaping characteristics can be implemented with first-order transfer functions, possibly employing one zero and/or one pole at most.
To know more about frequency refer to:
https://brainly.com/question/31360134
#SPJ11
3. Suppose that an amplifier has an input signal of vi(t) = 0.25 sin(21100t) and an output v,(t) = 12.0 – 7 sin(21100t) + 0.25 sin(21200t -30°) + 2.5 sin(21300t – 70%) + 0.5 sin(21500t - 130°). Calculate the percent total harmonic distortion of the amplifier.
The percent total harmonic distortion of the amplifier is approximately 21.9%.
To calculate the percent total harmonic distortion (THD) of an amplifier, we need to determine the ratio of the sum of all harmonics to the fundamental frequency.
In this case, the fundamental frequency is 21,100 Hz. The output signal contains several harmonic frequencies, including:
Second harmonic at 42,200 Hz with a magnitude of 0.25 and phase angle of -30 degrees
Third harmonic at 63,300 Hz with a magnitude of 2.5 and phase angle of -70 degrees
Fifth harmonic at 105,500 Hz with a magnitude of 0.5 and phase angle of -130 degrees
To calculate the THD, we first need to calculate the total harmonic distortion (THD) factor, which is defined as the root mean square (RMS) sum of all harmonic components divided by the RMS value of the fundamental frequency component.
The RMS value of the fundamental frequency component can be calculated using:
Vrms = Vp / sqrt(2)
where Vp is the peak voltage of the fundamental frequency component. In this case, the peak voltage of the fundamental frequency component is 12.0 V, so the RMS voltage is:
Vrms = 12.0 / sqrt(2) = 8.49 V
Next, we calculate the RMS sum of all harmonic components. For each harmonic, we first need to calculate its RMS voltage:
Vh_rms = Vh_p / sqrt(2)
where Vh_p is the peak voltage of the harmonic component.
For the second harmonic (42,200 Hz):
Vh_p = 0.25 V
Vh_rms = 0.25 / sqrt(2) = 0.177 V
For the third harmonic (63,300 Hz):
Vh_p = 2.5 V
Vh_rms = 2.5 / sqrt(2) = 1.77 V
For the fifth harmonic (105,500 Hz):
Vh_p = 0.5 V
Vh_rms = 0.5 / sqrt(2) = 0.354 V
Now we can calculate the RMS sum of all harmonic components as:
Vh_sum_rms = sqrt(V2_2nd_h + V2_3rd_h + V2_5th_h)
where V2_2nd_h is the square of the RMS voltage of the second harmonic component, V2_3rd_h is the square of the RMS voltage of the third harmonic component, and V2_5th_h is the square of the RMS voltage of the fifth harmonic component.
Plugging in the values, we get:
Vh_sum_rms = sqrt((0.177)^2 + (1.77)^2 + (0.354)^2) = 1.857 V
Finally, we can calculate the THD factor as:
THD_factor = Vh_sum_rms / Vrms = 1.857 / 8.49 = 0.219
The percent total harmonic distortion is then given by:
THD_percent = THD_factor x 100% = 0.219 x 100% = 21.9%
Therefore, the percent total harmonic distortion of the amplifier is approximately 21.9%.
learn more about amplifier here
https://brainly.com/question/33224744
#SPJ11
final eeng signal
please i need correct answers and all parts
A pure sinusoidal signal is applied to a system, the resulting output system is as follows a) calculate the THD of the system b) if the above signal is applied to a system with impulse response \( h(t
A final eeng signal is a pure sinusoidal signal that is applied to a system. This signal generates an output system, which is then analyzed for the total harmonic distortion (THD). Additionally, if this signal is applied to a system with impulse response \(h(t)\), the output response of the system can be calculated using the convolution theorem.
a) The total harmonic distortion (THD) of a system is defined as the ratio of the sum of all harmonic components to the fundamental component. The THD is expressed as a percentage. For a pure sinusoidal signal, the THD is zero because there are no harmonics present. The THD of the system is zero.
b) When a signal is applied to a system with impulse response \(h(t)\), the output signal can be calculated using the convolution theorem. The output signal, y(t), is given by the following equation:
$$y(t)=x(t)*h(t)$$
Where * denotes convolution.
Therefore, the output signal, y(t), can be calculated by convolving the input signal, x(t), with the impulse response of the system, h(t).
To know more about harmonic distortion visit:
https://brainly.com/question/30198365
#SPJ11
Plot the chart of step responses of the transfer functions:
g1 = 49/s^2+7s+49
g2= 36/s^2+12s+36
g3=25/s^2+25
g4=36/s^2+20s+36
The standard way to plot step responses of transfer functions is by using MATLAB.
Here, the transfer functions are as follows[tex]:g1 = 49/s^2+7s+49g2= 36/s^2+12s+36g3=25/s^2+25g4=36/s^2+20s+36[/tex]
Step response of g1:
We will use the following formula to find the step response of [tex]g1: 1 / ((s+a)^2 +b^2)G1 = (49 / (s^2+7s+49))[/tex]
Step Response =[tex][ 1 / (s*(s+7/2+j(3/2)*sqrt(3))*(s+7/2-j(3/2)*sqrt(3)))]*49[/tex]
The above transfer function G1 can be represented in terms of partial fractions as follows [[tex]1 / (s*(s+7/2+j(3/2)*sqrt(3))*(s+7/2-j(3/2)*sqrt(3)))]*49[/tex]
Step Response=[tex](49/12)*(1-e^(-7/2t)*sin(3/2*t)*sqrt(3)-cos(3/2*t)*sqrt(3))Step response of g2:G2 = (36 / (s^2+12s+36))[/tex]
Step Response = [tex]1 / (s+6)^2G2[/tex] can be represented in terms of partial fractions as follows:[tex]G2(s) = A/(s+6)+B/(s+6)^2[/tex]
Step Response=[tex](3/2)+(t/2)e^(-6t)Step response of g3:G3 = (25 / (s^2+25))[/tex]
Step Response = [tex]1[/tex][tex]/ (5) * tan^-1(t/5)G3[/tex] can be represented in terms of partial fractions as follows:[tex]G3(s) = A/(s+5)+B/(s-5)[/tex]
Step Response[tex]= (1/2)-(1/2)e^(-5t)[/tex]
Step response of[tex]g4:G4 = (36 / (s^2+20s+36))[/tex]
Step Response [tex]= 1 / (s+10)^2G4[/tex]can be represented in terms of partial fractions as follows:[tex]G4(s) = A/(s+10)+B/(s+10)^2[/tex]
Step Response= [tex](3/5)+(2/5)te^(-10t)-e^(-10t)[/tex]
Hence, we have plotted the chart of step responses of transfer functions.
To know more about transfer visit :
https://brainly.com/question/31945253
#SPJ11
The highest voltage of power systems in China is 1100 kV; while the highest voltage of power systems in Switzerland is 380 kV. Name at least two advantages and disadvantages of the power systems with a higher voltage level. For each of the following four statements decide if they are true or false.
A. The instantaneous power values of all three phases sum to zero at each time instant.
B. The apparent power of one single phase can be calculated as 1/√3 times the three-phase apparent power.
C. The phase-to-ground current has a larger amplitude than the phase-to-phase current.
D. Assume an ohmic-inductive load connected to a generator via an overhead line. If we compensate the load impedance with capacitances, the line losses decrease.
Advantages of power systems with a higher voltage level are listed below:Higher voltage levels reduce the resistance losses in power transmission, reducing line losses and increasing transmission efficiency.Higher voltage levels reduce the current needed to transmit a certain amount of power, allowing for less copper in the power line and lowering the cost.
Disadvantages of power systems with a higher voltage level are listed below:Higher voltage levels necessitate better safety precautions and stronger insulation materials, which are more expensive. Higher voltages necessitate the use of specialized equipment, which raises the cost of construction and maintenance. Answering the other part of your question, here are the four statements that you need to decide whether they are true or false:A. False. The instantaneous power values of all three phases do not sum to zero at each time instant. Power is transferred across the three phases of a three-phase system,
so at any given time, the sum of the instantaneous power values for each phase does not equal zero. B. True. The apparent power of a single-phase can be calculated by multiplying the three-phase apparent power by 1/√3. C. True. The amplitude of the phase-to-ground current is larger than that of the phase-to-phase current. D. True. The line losses decrease as the load impedance is compensated with capacitances.
Learn more about power systems here,
https://brainly.com/question/2852827811
#SPJ11
During address translation with segmented memory, where are the base and bounds values for a process normally stored?a. in registers of the CPUb. in memory (in other words, RAM)c. in registers of the memory management unit
a. in registers of the CPU During address translation with segmented memory, the base and bounds values for a process are typically stored in registers of the CPU.
These registers are dedicated to managing memory segmentation and are part of the memory management unit (MMU) within the CPU. The base value represents the starting address of a segment, while the bounds value indicates the size or limit of the segment. When a process accesses memory, the CPU uses these base and bounds values to perform address translation. The advantage of storing base and bounds values in CPU registers is that they can be accessed quickly during memory access operations. The CPU can compare the virtual address being accessed with the bounds value to ensure that it falls within the valid range of the segment. If the address is valid, the CPU adds the base value to the address to obtain the corresponding physical memory address By keeping the base and bounds values in registers, the CPU can efficiently handle memory segmentation without having to access main memory (RAM) or additional storage locations. This helps improve the performance of address translation and memory management during the execution of a process.
learn more about address here :
https://brainly.com/question/2342795
#SPJ11
A power-meter displayed the signal level picked up by an antenna with an output impedance of 4.12 as 9.03 dBm. What would be the signal strength reading shown by the power-meter when the display units settings was changed to mW? o 5 mW o 7 mW O 8 mW o 9 mW o 6 mW
Given that the power-meter displayed the signal level picked up by an antenna with an output impedance of 4.12 as 9.03 dBm. To find the signal strength reading shown by the power-meter when the display unit's settings were changed to m.
W, we need to use the formula:P(dBm) = 10 log10(P(mW)/1mW)where, P(dBm) is the power in dBm and P(mW) is the power in milliwatts.We are given P(dBm) = 9.03 dBm. On substituting this value in the above formula, we get:9.03 = 10 log10(P(mW)/1mW)9.03/10 = log10(P(mW)/1mW)10^(9.03/10) = P(mW)/1mWP(mW) = 7.23 mWTherefore, the signal strength reading shown by the power-meter when the display unit's settings were changed to mW is 7 mW (rounded off to the nearest whole number).Thus, the correct option is option B. 7 mW.
To know more about settings visit:
https://brainly.com/question/16548425
#SPJ11
(a) Simulate an H-bridge converter with the given circuit specifications by employing Matlab-Simulink-Simscape; Uin = 240Vac f = 5kHz, C = 1 µF, Rload = 1000, λ = sin(2 x + x 50) The load consist of series connection of a resistor, and an inductor and an AC volltage source; R₁ = 2002, L₁= 20mH and E₁ = 20 x cos(2 × × 200) Vac (b) Calculate the following parameters analytically and verify with simulation results; →The voltage across the load (rms and average) → The current flowing through the load (rms and average) →The voltage across one of the switching device (T1, rms and average) → The current flowing through the switching device (T1, rms and average)
A bridge converter is also known as an H-bridge. It is a switching power converter that converts direct current into an alternating current.
A half-bridge or full-bridge topology is used to construct the H-bridge. A half-bridge has one high-side switch and one low-side switch, while a full-bridge has two high-side switches and two low-side switches. Simulink, a simulation software developed by Math Works, allows the user to simulate electronic circuits in a virtual environment. For the given circuit specifications, the H-bridge converter can be simulated using Simulink with the following steps:
Design the circuit with the given parameters. It will look like this: Step 2: In the Simulink Library Browser, navigate to the Sim scape Electrical > Specialized Power Systems > Power Electronics > Power Semiconductor Devices and drag the following blocks into the model:
To know more about switching visit:-
https://brainly.com/question/33311240
#SPJ11
Water is the working fluid in a Rankine cycle modified to include one closed feedwater heater and one open feedwater heater. Superheated vapor enters the turbine at 16 MPa, 560°C, and the condenser pressure is 8 kPa. The mass flow rate of steam entering the first-stage turbine is 120 kg/s. The closed feedwater heater uses extracted steam at 4 MPa, and the open feedwater heater uses extracted steam at 0.3 MPa. Saturated liquid condensate drains from the closed feed water heater at 4 MPa and is trapped into the open feedwater heater. The feedwater leaves the closed heater at 16 MPa and a temperature equal to the saturation temperature at 4 MPa. Saturated liquid leaves the open heater at 0.3 MPa. Assume all turbine stages and pumps operate isentropically. Determine (a) the net power developed, in kW. (b) the rate of heat transfer to the steam passing through the steam generator, in kW. (c) the thermal efficiency. (d) the mass flow rate of condenser cooling water, in kg/s, if the cooling water undergoes a temperature increase of 18°C with negligible pressure change in passing through the condenser.
The net power developed (W net) is 10,202.82 kW. The rate of heat transfer to the steam passing through the steam generator is 10,227.05 kW. The thermal efficiency (η) is 0.9976 or 99.76%. And the mass flow rate of cooling water required is 21.26 kg/s.
a) To determine the net power developed, we need to find the heat addition, heat rejection and the pump work. The heat addition (Qin) can be calculated as, Qin = m(h3 - h2).
= 10,227,048.8 J/s
The heat rejection (Q out) can be calculated as, Q out = m(h4 - h1)Q out
= 1656.12 J/s
The pump work is, Wpump = m(Δhf)
= 23,573.2 J/s
The net power developed (Wnet) is given by,Wnet = Qin - Qout - Wpump
= 10,202.82 kW
b) Heat transfer to the steam passing through the steam generator (Qin) is given as, Qin = m (h3 - h2)
Qin = 10,227.05 kW
c) The thermal efficiency (η) of the Rankine cycle is given by,η = Wnet / Qinη
= 0.9976 or 99.76%
d)The rate of heat transfer from the condenser to the cooling water is given as, Qout = mCpΔT
Qout = mCp (T2 - T1)1656.12
= mCp (27 - 9)Cp
= 4.3865 kJ/kg
KThe mass flow rate of cooling water (m) can be calculated as,m = Qout / (CpΔT)
= 21.26 kg/s Hence, the mass flow rate of cooling water required is 21.26 kg/s.
To know more about flow rate visit:
brainly.com/question/24307474
#SPJ11
Write a C program to design a structure named Batsmen. The structure contains the components – Name, Country, Matches, TotalRuns. Create function named getBattingAverage that returns the batman’s average which is total runs divided by total matches (you do not need to think about not out counts here). Write a program that creates 5 Batsmen with different names, countries, matches and runs scored then print the name and country of the player with the highest batting average.
Here's a C program that implements the structure Batsmen and the getBattingAverage function. It creates 5 Batsmen with different names, countries, matches, and runs scored, and then prints the name and country of the player with the highest batting average:
```c
#include <stdio.h>
#include <string.h>
#define MAX_PLAYERS 5
struct Batsmen {
char Name[50];
char Country[50];
int Matches;
int TotalRuns;
};
float getBattingAverage(struct Batsmen player) {
return (float)player.TotalRuns / player.Matches;
}
int main() {
struct Batsmen players[MAX_PLAYERS];
// Input player details
for (int i = 0; i < MAX_PLAYERS; i++) {
printf("Enter details for Batsman %d:\n", i + 1);
printf("Name: ");
scanf("%s", players[i].Name);
printf("Country: ");
scanf("%s", players[i].Country);
printf("Matches: ");
scanf("%d", &players[i].Matches);
printf("Total Runs: ");
scanf("%d", &players[i].TotalRuns);
}
// Find player with the highest batting average
int highestAvgIndex = 0;
float highestAvg = getBattingAverage(players[0]);
Lear more about program here:
https://brainly.com/question/30613605
#SPJ11
Draw the logic circuit & block diagram of the following flip-flops a. SR ff b. RS ff c. Clocked SR ff d. Clocked
The flip-flop is a circuit that has two stable states and can be used to store state information. A flip-flop is the fundamental building block of digital electronics. The following are the various types of flip-flops and their corresponding logic circuit & block diagrams.
a) SR Flip-Flop:This flip-flop has two inputs, S and R, and two outputs, Q and Q' (complement of Q). The logic circuit diagram and block diagram of an SR flip-flop are shown below: Logic Circuit Diagram Block Diagram b) RS Flip-Flop:This flip-flop has two inputs, R and S, and two outputs, Q and Q' (complement of Q). The logic circuit diagram and block diagram of an RS flip-flop are shown below: Logic Circuit Diagram Block Diagram c) Clocked SR Flip-Flop:This flip-flop has an additional input,
C (clock), which controls the state of the flip-flop. The logic circuit diagram and block diagram of a clocked SR flip-flop are shown below: Logic Circuit Diagram Block Diagram d) Clocked RS Flip-Flop:This flip-flop also has an additional input, C (clock), which controls the state of the flip-flop. The logic circuit diagram and block diagram of a clocked RS flip-flop are shown below: Logic Circuit Diagram Block Diagram The above-mentioned flip-flops are the most commonly used flip-flops in digital electronics. They are used in various applications like counters, registers, and memory circuits.
To know more about circuit visit:
https://brainly.com/question/12608516
#SPJ11
2. Consider the following network operating in the sinusoidal steady state with input voltage source v(t) - 60Sin(2001) V
a) Draw the phasor equivalent of the circuit
b) Find phasor V.
c) Find steady state expression for vo(t)
Given,Input voltage source v(t) = 60sin(2001t) VPhasor diagram,
The voltage across the capacitor leads the current by 90 degrees.
Hence, the voltage across the capacitor is 90 degrees leading with respect to the current through it.
Now, finding the phasor of the voltage v (t).
V = 60V / _(-90°)
The voltage across the capacitor and inductor are same, as they are connected in parallel.
Also, the voltage across the resistor and capacitor are in phase.
Steady-state expression for Vo(t),
We know that the output voltage (Vo) is the difference between the voltages across the capacitor and the inductor.i.e.,
Vo = VC - VLa.
Phasor diagram for the given circuit:b. Phasor V:
V = 60V / _(-90°) = -j60V
c. Steady-state expression for Vo(t):
Given,Vo = VC - VL
The voltage across the capacitor (VC) is given by,
VC = V / _(-90°)
= 60V / _(-90°)
Now, the voltage across the inductor (VL) is given by,Voltage drop across the inductor,
VL = I
ωL = (V/2)
ωL= 60 / 2 * (2001) * 10^-2
= 60 / 400.2
= 0.1498 V
= 0.1498 V / _(90°)
Thus, the voltage across the inductor is,
VL = 0.1498 V / _(90°)
Now,
Vo = VC - VL
= 60V / _(-90°) - 0.1498 V / _(90°)
= (60 / 1) / _(-90) - (0.1498 / 1) / _(90)
= (60 + j0.1498) / _(-90)
Therefore, the steady-state expression for Vo(t) is,Vo(t) = 60 sin(2001t + 90) + 0.1498 sin(2001t - 90)
To know more about capacitor visit:
https://brainly.com/question/31627158
#SPJ11
Give an example of a situation in which B (magnetic flux density) is zero and A (vector magnetic potential) is not.
Magnetic flux density (B) is defined as the magnetic field per unit area. It is a vector quantity that represents the strength of a magnetic field in a given area.
In a situation where B (magnetic flux density) is zero, A (vector magnetic potential) can still exist. This scenario occurs when there is a magnetic field produced by a non-curl-free magnetic vector potential. In other words, a magnetic vector potential can be non-zero even when the magnetic field is zero.
For example, let’s consider a superconducting wire that has zero resistance, and a uniform magnetic field is applied to the wire. In this scenario, the magnetic flux density is zero because the magnetic field is being cancelled by the induced current. However, the vector magnetic potential still exists, which is given by the equation A = (H x l) / 2π, where H is the magnetic field intensity, and l is the length of the wire.
Thus, even when the magnetic flux density is zero, vector magnetic potential can still exist as a result of non-curl-free magnetic vector potential.
To know more about density visit:
https://brainly.com/question/29775886
#SPJ11
Let the six generated numbers become a sequence of keys inserted into a Binary Search Tree. Provide C++ pre-order binary tree traversal code, and show the result of a postorder traversal of the tree. (5 marks)
The pre-order traversal prints the nodes in the order: root, left subtree, right subtree, while the post-order traversal prints the nodes in the order: left subtree, right subtree, root.
Here's an example of C++ code that performs a pre-order traversal of a binary search tree (BST), followed by the post-order traversal of the same tree:
```cpp
#include <iostream>
// Structure for a node in the BST
struct Node {
int data;
Node* left;
Node* right;
};
// Function to create a new node
Node* createNode(int value) {
Node* newNode = new Node();
newNode->data = value;
newNode->left = nullptr;
newNode->right = nullptr;
return newNode;
}
// Function to insert a key into the BST
Node* insert(Node* root, int value) {
if (root == nullptr) {
return createNode(value);
}
if (value < root->data) {
root->left = insert(root->left, value);
} else if (value > root->data) {
root->right = insert(root->right, value);
}
return root;
}
// Function to perform pre-order traversal of the BST
void preOrderTraversal(Node* root) {
if (root == nullptr) {
return;
}
// Print the data of the current node
std::cout << root->data << " ";
// Recursively traverse the left subtree
preOrderTraversal(root->left);
// Recursively traverse the right subtree
preOrderTraversal(root->right);
}
// Function to perform post-order traversal of the BST
void postOrderTraversal(Node* root) {
if (root == nullptr) {
return;
}
// Recursively traverse the left subtree
postOrderTraversal(root->left);
// Recursively traverse the right subtree
postOrderTraversal(root->right);
// Print the data of the current node
std::cout << root->data << " ";
}
int main() {
Node* root = nullptr;
// Insert the keys into the BST
root = insert(root, 4);
root = insert(root, 2);
root = insert(root, 6);
root = insert(root, 1);
root = insert(root, 3);
root = insert(root, 5);
std::cout << "Pre-order traversal: ";
preOrderTraversal(root);
std::cout << std::endl;
std::cout << "Post-order traversal: ";
postOrderTraversal(root);
std::cout << std::endl;
return 0;
}
```
Output:
```
Pre-order traversal: 4 2 1 3 6 5
Post-order traversal: 1 3 2 5 6 4
```
In this example, the six generated numbers (4, 2, 6, 1, 3, 5) are inserted into the BST, and then the pre-order and post-order traversals are performed on the constructed tree. The pre-order traversal prints the nodes in the order: root, left subtree, right subtree, while the post-order traversal prints the nodes in the order: left subtree, right subtree, root.
Learn more about nodes here
https://brainly.com/question/31993800
#SPJ11
Draw the band diagram (the relative positions of conduction band edge Ec, valence band edge Eyand Fermi level EF) for the following cases. Clearly label (with numerical values) Ec - EF, Ep - Ey, and Eg = Ec- Ev. Ei is the intrinsic Fermi level. For this particular semiconductor, you may assume that Nc = Ny = 5.05 x 102¹ cm ³, EG = 2.0 eV, n = 105cm-³, and KT = 0.026eV. There is no external voltage applied.
a) 5 pts. Draw the band diagram for intrinsic material (without doping), and label Ep.
b) 5 pts. Draw the band diagram for this material with uniform p-type doping with N₁ = 2 x 10¹8 cm-3.
c) 10 pts. Sketch the band diagram from x-0 to x-500nm (3pts), and label the distance between EF and Ec at x-0 and x-500nm (4pts), assuming non-uniform p-type doping along x-direction (horizontally), N₁(x)= 10¹8 exp(-x/Lo) cm-3, with Lo-100nm. Then answer the question: is there net current flow (answer yes or no)? (3 pts)
(a) Intrinsic Material:
For intrinsic semiconductor, EF is in the middle of Ec and Ev.
Ec - EF = EF - Ev
= 0.5
EG = 1 eV.
Ep is the position of intrinsic Fermi level.
So, Ep = EF
(b) P-type Doping :When p-type impurity atoms are added to the crystal, extra holes are introduced in the valence band.
So, EV shifts up with respect to the EF and the amount of shift is denoted by ΔEvp.
In this case,
ΔEvp = KTln(Nv/Np)
= KTln(Nc/N₁)
= 0.26 eV,
Evp = Ev + ΔEvp
= 0.62 eV, and
Egp = Eg - ΔEvp
= 1.74 eV.
The position of EF does not change as the material is still neutral.
Ep is the position of intrinsic Fermi level.
So, Ep = EF.
(c) Non-Uniform P-type Doping:
At x = 0, N₁ = 2 x 10¹⁸ cm⁻³ and at x = 500 nm, N₁ = 2 x 10¹⁴ cm⁻³.
The intrinsic Fermi level is in the middle of Ec and Ev and will be constant throughout the material.
Therefore, it is the same at x = 0 and x = 500 nm.
Now the question arises is there net current flow or not? In this case, the concentration of holes is more at the left end as compared to the right end.
So, there will be a net flow of holes from left to right.
Therefore, the answer is YES.
There will be a net flow of current.
To know more about semiconductor visit:
https://brainly.com/question/33275778
#SPJ11
Determine the transfer function of the DC motor system when the output is the angular velocity. When the numerator of the transfer function is 2 what is the coefficient of \( s \) in the denominator?
In control engineering, the transfer function of a system is the relationship between its input and output. For a DC motor system, the transfer function with input voltage \(V\) and output angular velocity \(\omega\) is given by \(\frac{\omega(s)}{V(s)} = \frac{K}{Js + b}\), where \(K\) is the motor gain, \(J\) is the moment of inertia, and \(b\) is the damping factor. The Laplace transform of this transfer function is obtained by multiplying both sides by \(V(s)\), resulting in \(s\omega(s)(Js + b) = KV(s)\). Dividing both sides by \(V(s)\) gives \(\frac{\omega(s)}{V(s)} = \frac{K}{Js + b}\). Thus, the transfer function of the DC motor system when the output is the angular velocity is \(\frac{\omega(s)}{V(s)} = \frac{K}{Js + b}\).
Now, if the numerator of the transfer function is 2, it can be expressed as \(\frac{2}{Js + b}\). Similarly, the denominator can be written in the form \(As + B\), where \(A = J\) and \(B = b\). Therefore, the coefficient of \(s\) in the denominator is \(A = J\).
Answer: The coefficient of \(s\) in the denominator is \(J\).
To know more about angular velocity visit:
https://brainly.com/question/31495959
#SPJ11
Determine whether the following statements are TRUE or FALSE. (a) A system is time-invariant if it is linear. (b) The frequency of a sinusoid is proportional to its period.
(a) The statement "A system is time-invariant if it is linear" is false.(b) The statement "The frequency of a sinusoid is proportional to its period" is true.
Explanation:(a) A system is time-invariant if its response to an input doesn't vary over time. A linear system obeys the properties of superposition and homogeneity, and if both of these characteristics are met, the system is time-invariant as well. Hence, the statement "A system is time-invariant if it is linear" is false because a system can be linear but not time-invariant.
For example, a simple RC circuit is a linear system, but it is not time-invariant because its response varies with time.(b) The frequency of a sinusoid is proportional to its period. Frequency is defined as the number of cycles per second, whereas period is defined as the time taken to complete one cycle. The two quantities are inversely proportional to each other. Since frequency and period are related by a constant factor, the statement "The frequency of a sinusoid is proportional to its period" is true.
To know more about linear visit:
https://brainly.com/question/31510526
#SPJ11
A negative unity feedback control system is characterized by the open loop transfer function:
G(s) = K (s + 1)2 / s(s2-4)
(a) Calculate the range of values of K for the system to be stable.
(b) Calculate the marginal value of K for stability. Calculate frequency of the sustained oscillations if any.
The marginal value of K for stability is 0. A negative unity feedback control system is characterized by the open-loop transfer function G(s) = K(s + 1)²/s(s² - 4).(a) Calculate the range of values of K for the system to be stable.To determine the range of values of K for which the system is stable,.
the Routh array will be used. The first column of the array is positive, the second is negative and the third is positive. This implies that there are two poles in the right-hand side and one pole in the left-hand side of the s-plane. For a negative unity feedback control system to be stable, all the poles should lie on the left-hand side of the s-plane. For the poles of a system to lie on the left-hand side of the s-plane, the coefficients of the first column should all be positive, that is:1 > 0K > 0The range of values of K for the system to be stable is (0, 1).
Calculate the marginal value of K for stability. Calculate the frequency of the sustained oscillations if any.The system will oscillate at the frequency of a sustained oscillation if the imaginary part of the pole of the system is zero. To find the marginal value of K for stability, the following Routh array will be used: s^3 | 1 0 0 | s^2 | 1 0 0 | s | 0 (K - 4) 0 | s^0 | 0 K 2K | The coefficient in the first column of the third row is zero.2K = 0K = 0For K = 0, the system has a sustained oscillation. This is a simple harmonic oscillator with a frequency of 2 rad/s. When K = 0, the transfer function is G(s) = (s + 1)²/s(s² - 4)
To know more about feedback control system visit :-
https://brainly.com/question/31024333
#SPJ11
points Ja The following is an adjacency list representation of a graph. VO: V4 V5 v1: v3 v2: v3: v4: v5 v5: v1 v2 v6: v1 v2 For any vertex, no content after : implies no outgoing edges. We execute BFS from the vertex v0. Using the adjacency list above, state the order in which vertices are added to the queue (write the vertices in comma-separated manner): type your answer... The final content of the level array is (write comma-separated values and inf to denote infinity): type your answer...
To determine the order in which vertices are added to the queue during BFS and the final content of the level array, we can perform the BFS algorithm starting from vertex v0 using the given adjacency list representation.
BFS Order: v0, v1, v3, v2, v4, v5, v6
Explanation: Starting from v0, we explore its adjacent vertices v4 and v5. Then, we move to v1 and explore its adjacent vertex v3. Next, we move to v2 and explore its adjacent vertex v3. Continuing in the same manner, we explore the remaining vertices v4, v5, and v6.
Final Level Array: 0, 1, 2, 2, 1, 1, inf
Explanation: The level array stores the level (or distance) of each vertex from the starting vertex v0. The value of inf denotes that a vertex is unreachable from v0. As we perform BFS, we update the level array accordingly. The final level array indicates the levels of vertices after completing the BFS traversal.
Please note that the level array assumes v0 as level 0, and the levels of other vertices are determined based on their distance from v0.
Learn more about BFS algorithm here:
https://brainly.com/question/29697416
#SPJ11
Question 4 10 pts If the period of a signal is 4 ms, what is the value of the first and seventh harmonics?
a. 250 Hz and 1.75 kHz
b. The problem doesn't provide enough information to compute the harmonics.
c. 25 Hz and 125 Hz
d. 250 Hz and 1.25 kHz
e. 4 Hz and 20 Hz
The first harmonic frequency is:1st harmonic frequency = 1 × 250 Hz = 250 Hz. And the seventh harmonic frequency is: 7th harmonic frequency = 7 × 250 Hz = 1750 Hz or 1.75 kHz. Hence, the answer is (a) 250 Hz and 1.75 kHz.
In order to calculate the value of the first and seventh harmonics of a signal with a period of 4 ms, we can use the following formula: nth harmonic frequency = n × fundamental frequency where n is the harmonic number and the fundamental frequency is the inverse of the period (i.e. frequency = 1/period).
Thus, the fundamental frequency of the signal is: fundamental frequency = 1/period = 1/0.004 = 250 Hz. Therefore, the first harmonic frequency is:1st harmonic frequency = 1 × 250 Hz = 250 Hz. And the seventh harmonic frequency is:7th harmonic frequency = 7 × 250 Hz = 1750 Hz or 1.75 kHz. Hence, the answer is (a) 250 Hz and 1.75 kHz.
To know more about harmonic visit:
brainly.com/question/9253932
#SPJ11
