EECS 16B Designing Information Devices and Systems II

Fall 2018 Elad Alon and Miki Lustig Homework 4

This homework is solely for your own practice. However, everything on it is
in scope for midterm 1, and it will be assumed in lab that you have completed
the lab-related questions.
1. Mystery Microphone
You are working for Mysterious Miniature Microphone Multinational when your manager asks you to test
a batch of the company’s new microphones. You grab one of the new microphones off the shelf, use a tone
generator 1 to play pure tones of uniform amplitude at various frequencies, and measure the resultant peak-
to-peak voltages using an oscilloscope. You collect data, and then plot it (on a logarithmic scale). The plot
is shown below:

Figure 1: Frequency Response

(a) To which frequencies is the microphone most sensitive, and to which frequencies is the microphone
least sensitive?
Solution:
The microphone is most sensitive to frequencies in the range of 320 Hz to 5 kHz, and least sensitive
below ≈ 100 Hz or so.
You report these findings to your manager, who thanks you for the preliminary data and proceeds to
co-ordinate some human listener tests. In the meantime, your manager asks you to predict the effects
1 Note that soundwaves are simply sinusoids at various frequencies with some amplitude and phase. The microphone’s di-

aphragm oscillates with the sound (pressure) waves, moving the attached wire coil back and forth over an internal magnet, which
induces a current in the wire. In this way, a microphone can be modeled as a signal-dependent current source. The output current
can be converted to a voltage by simply adding a known resistor to the circuit and measuring the voltage across that resistor.

EECS 16B, Fall 2018, Homework 4 1

 of the microphone recordings on human listeners, and encourages you to start thinking more deeply
about the relationships.
(b) For testing purposes, you have a song with sub-bass (150 Hz or less), mid-range (≈ 1 kHz), and some
high frequency electronic parts (> 12 kHz). Which frequency ranges of the song would you be able to
hear easily, and which parts would you have trouble hearing? Why?
Solution:
The mid-range would be most audible since the amplitude is the highest at these frequencies. The high
frequency electronic parts are the next loudest. The sub-bass parts would be the parts you have trouble
hearing since the output amplitude is so low.
(c) After a few weeks, your manager reports back to you on the findings. Apparently, this microphone
causes some people’s voices to sound really weird, resulting in users threatening to switch to products
from a competing microphone company.
It turns out that we can design some filters to “fix” the frequency response so that the different frequen-
cies can be recorded more equally, thus avoiding distortion. Imagine that you have a few (say up to 4
or so) blocks. Each of these blocks detects a set range of frequencies, and if the signal is within this
range, it will switch on a op-amp circuit of your choice. For example, it can be configured to switch
on an op-amp filter to double the voltage for signals between 100 Hz and 200 Hz.
What ranges of signals would require such a block, and what gain would you apply to each block such
that the resulting peak-to-peak voltage is about 5 V for all frequencies?
Solution:
The output amplitude for < 100 Hz is ≈ 0.5 V, so it needs a gain of 10.
For 100-160Hz, the amplitude is ≈ 2.5 V, so it needs a gain of 2.
320-5000Hz already has an amplitude of 5V, so no gain is needed.
10000-20000Hz has an amplitude of ≈ 1.5 V, so it needs a gain of 3.33.

2. RLC Circuit
In this question, we will take a look at an electrical system described by a second order differential equations
and analyze it using the phasor

domain. Consider the circuit below, where R = 8 kΩ, L = 1 mH, C = 200 nF,
and Vs = 2 cos 2000t + π4 .

R L i(t)
+ − +
VL −
VR
+
+
Vs Vout C
−
−

(a) What are the impedances of the resistor ZR , inductor ZL , and capacitor ZC ?
Solution:
The impedance of a resistor is the same as its resistance.

ZR = 8000 Ω

EECS 16B, Fall 2018, Homework 4 2

 We can find the frequency of the circuit by looking at Vs . The form of a cosine function is A cos(ωt +
φ ), where A is the amplitude, ω is the frequency, and φ is the phase. In this case, the frequency is
2000 rad
s .
ZL = jωL = j2000 · 10−3 = j2Ω
1 1
ZC = = = − j25 · 102 Ω
jωC j2000 · 2 · 10−7

(b) Solve for Veout in phasor form.

Solution:
Converting Vs into phasor form, we have
π
Ves = |A|e jφ = 2e j 4

The circuit given is a voltage divider. Since impedances act like resistors, we can use the same equation
as that for a resistive voltage divider.

ZC π − j ∗ 2.5 ∗ 103 π − j ∗ 2500

Veout = Ves = 2e j 4 3 2
= 2e j 4
ZR + ZL + ZC 8 ∗ 10 + j ∗ 2 − j ∗ 25 ∗ 10 8000 − j ∗ 2498

We can solve for the magnitude and angle of the divider using
 
 j π − j ∗ 2500  2500
2e 4  = 2p = 0.597
 8000 − j ∗ 2498  80002 + (−2498)2
 
j π4 − j ∗ 2500 π
∠ 2e = ∠(2e j 4 ) + ∠(− j ∗ 2500) − ∠(8000 − j ∗ 2498)
8000 − j ∗ 2498
π −π
= + − atan2(−2498, 8000) = −0.4827 rad
4 2
Veout = 0.597e− j0.4827

(c) What is Vout in the time domain?

Solution: We know for Vout (t) = A cos(ωt + φ ), we have Ṽout = A(cos(φ ) + j sin(φ )) = Ae jφ . Thus,
we have A = 0.597, φ = −0.4827, which gives us Vout (t) = 0.597 cos(ωt − 0.4827)
(d) Solve for the current i(t).
Solution:

|Ves |
 
Ves j ∠Ves −∠(ZR +ZL +ZC )
ei = = e = 2.38 · 10−4 e j1.088
ZR + ZL + ZC |ZR + ZL + ZC |
Going back to the time domain:

i(t) = 2.38 · 10−4 cos(2000t + 1.088)

(e) Solve for the transfer function H(ω) = VVeout

e
s
Leave your answer in terms of R, L, C, and ω.
Solution:

EECS 16B, Fall 2018, Homework 4 3

 Looking back at part (b),
ZC
Veout = Ves
ZR + ZL + ZC
Rearranging, we get
1
Veout ZC jωC
H(ω) = = = 1
Ves ZR + ZL + ZC R + jωL + jωC

1
H(ω) =
1 + jωRC + ( jω)2 LC

3. Phasor-Domain Circuit Analysis

The analysis techniques you learned previously for resistive circuits are equally applicable for analyzing
AC circuits (circuits driven by sinusoidal inputs) in the phasor domain. In this problem, we will walk you
through the steps with a concrete example. Consider the circuit below.

iR1 R1

iL1 L1 N2 L2 iL2
N1

R2
C1
iR2 R3
ic
+ v(t)
−

The components in this circuit are given by:

Voltage source:
√
 
π
v(t) = 10 2 cos 100t −
4
Resistors:
R1 = 5 Ω, R2 = 5 Ω, R3 = 1 Ω
Inductors:
L1 = 50 mH, L2 = 20 mH
Capacitor:
C1 = 2 mF

(a) Transform the given circuit to the phasor domain (components and sources).
Solution:

EECS 16B, Fall 2018, Homework 4 4

 ZL1 = jωL = j100 × 50 × 10−3 = j5 Ω
ZL2 = jωL = j100 × 20 × 10−3 = j2 Ω
1 1
ZC = = = − j5 Ω
jωC j100 × 2 × 10−3
√ π
ve = |v|e j∠v = 10 2e− j 4

(b) Write out KCL for node N1 and N2 in the phasor domain in terms of the currents provided.
Solution:
At node 1:
iL1 + iR1 = ic

At node 2:
iR1 + iL1 + iR2 + iL2 = 0

(c) Find expressions for each current in terms of node voltages in the phasor domain. The node voltages
Ve1 and Ve2 are the voltage drops from N1 and N2 to the ground.
Solution:
We have

Ve2 − Ve1 Ve2 − Ve1 Ve1

+ =
ZL1 R1 ZC
V2 − V1 V2 − V1 V2 − ve
e e e e e V2
e
+ + + =0
R1 ZL1 R2 R3 + ZL2

Plugging in values from part (a), we get

Ve2 − Ve1 Ve2 − Ve1 Ve1

+ =
j5 5 − j5
√ −jπ
Ve2 − Ve1 Ve2 − Ve1 Ve2 − 10 2e 4 Ve2
+ + + =0
5 j5 5 1 + j2

For future parts, we want the denominators of each current to be either purely real or purely imagi-
nary. To put iL2 in this form, we can manipulate the expression by multiplying the denominator by its
conjugate: !
Ve2 1 − j2 Ve2 (1 − j2) Ve2 (1 − j2)
= =
1 + j2 1 − j2 1 − (−4) 5

Our final KCL equations at nodes N1 and N2 are

Ve2 − Ve1 Ve2 − Ve1 Ve1

+ =
j5 5 − j5
√ −jπ
V2 − V1 V2 − V1 V2 − 10 2e 4 V2 (1 − j2)
e e e e e e
+ + + =0
5 j5 5 5

EECS 16B, Fall 2018, Homework 4 5

 " #
Ve
(d) Write the equations you derived in part (c) in a matrix form, i.e., A e1 = ~b. Write out A and ~b
V2
numerically.
Solution:
From the above two equations, we have
" # " #
− 51 1
5 − j 1
5 = −0.2 0.2 − j0.2
A=
− 15 + j 15 35 − j 53 −0.2 + j0.2 0.6 − j0.6
" # " #
0 0
~b = √ π =
10 2e− j 4 2 − j2
5

(e) Solve the systems of linear equations you derived in part (d) with any method you prefer and then find
ic (t).
Solution:
The inverse of a 2 × 2 matrix is given by:
" #−1 " #
a b 1 d −b
= .
c d ad − bc −c a
" #
−6 + j3 2 − j1
A−1 =
−2 + j1 1.5 − j0.5

With that, we find " # " # "√ #

Ve1 2 − j6 40e − j1.249
= A−1~b = = √ − j0.464
Ve2 4 − j2 20e
√
Ve1 j 40 j0.322
IC = = Ve1 = e = 1.265e j0.322
− j5 5 5
Transforming IC back to time domain, we get

iC (t) = 1.265 cos(100t + 0.322)

4. Analyzing Mic Board Circuit

In this problem, we will work up to analyzing a simplified version of the mic board circuit. In lab, we will
address the minor differences between the final circuit in this problem and the actual mic board circuit.
The microphone can be modeled as a frequency-dependent current source, IMIC = k sin(ωt) + IDC , where
IMIC is the current generated by the mic (which flows from VDD to VSS), IDC is some constant current, k is
the force 2 to current conversion ratio, and ω is the signal’s frequency (in rad
s ). VDD and VSS are 5 V and
−5 V, respectively.
2 The force is exerted by the soundwaves on the mic’s diaphragm.

EECS 16B, Fall 2018, Homework 4 6

 Figure 2: Step 1. The microphone is modeled as a DC current source.

(a) DC Analysis Assume for now that k = 0 (so that we can examine just the "DC" response of the
circuit), find VOUT in terms of IDC , R1 , R2 , and R3 (Hint: You do not need to worry about Vss in your
calculations).
Solution: The current in the left branch is equal to IDC since no current flows into the op-amp.

Vin = VDD −VR1 = 5 − (IDC · R1 )

R2 R2
Vout = (1 + ) ·Vin = (1 + ) · (5 − (IMIC · R1 ))
R3 R3

(b) Now, let’s include the sinusoidal part of IMIC as well. We can model this situation as shown below,
with IMIC split into two current sources so that we can analyze the whole circuit using superposition.
Let IAC = k sin(ωt). Find and plot the function VOUT(t) .

EECS 16B, Fall 2018, Homework 4 7

Figure 3: Step 2. The microphone is modeled as the superposition of a a DC and a sinusoidal ("AC") current
source.

Solution: Doing superposition, we null each of the sources and add the results. Let’s use superposi-
tion to find Vin . Note, here when we do superposition we have 3 sources that affect Vin : VDD , IDC , and
IAC . Nulling both current sources, we see that Vin1 = VDD because there is no current flowing in our
circuit there is no change in voltage over the resistor. Nulling VDD and IAC , we get a similar expression
to part (a) except there is no 5 volt source: Vin2 = −R1 · IDC . And finally, nulling VDD and IDC , we get a
similar expression to our last one: Vin3 = −R1 · IAC
Putting these together and plugging in our expression for IAC we get:

Vin = Vin1 +Vin2 +Vin3 = 5 − R1 · (ksin(ωt) + IDC )

This then goes through a noninverting amplifier for our final answer:
R2
Vout = (1 + ) · (5 − R1 · (ksin(ωt) + IDC ))
R3

Figure 4: Vout (t) when R1 = 10 kΩ, R2 = 2040 Ω, R3 = 100 kΩ, IDC = 10 µA, k = 10−9

(c) Given that VDD = 5 V, VSS = −5 V, R1 = 10 kΩ, and IDC = 10 µA, find the maximum value of the gain G
of the noninverting amplifier circuit for which the op-amp would not need to produce voltages greater
than VDD or less than VSS (i.e, find the maximum gain G we can use without causing the op-amp to
clip).

EECS 16B, Fall 2018, Homework 4 8

 Solution: Since the signal is centered around 5 − R1 IDC = 4.9V, we know that VDD will limit the
amplitude of the signal first.
Using our expression for Vout from part (b):
VDD side:

G · (5 − R1 IDC + R1 max(k sin(ωt)) ≤ VDD

G · (5 − 104 · 10−5 + 104 k)) ≤ 5
5
G≤
4.9 + k · 104

(d) We have modified the circuit as shown below to include a high-pass filter so that the term related to
IDC is removed before we apply gain to the signal. Provide a symbolic expression for VOUT given that
that VDD0 = 5 V, VSS0 = −5 V, VDD1 = 3.3 V, VSS1 = 0 V. Show your work.

Figure 5: Step 3. Approaching the real mic board circuit. The microphone is still modeled as the superposi-
tion of a a DC and a sinusoidal ("AC") current source.

Solution: Since the high-pass filter removes the DC portion of the mic signal (the portion contributed
by IDC ), the voltage going into the noninverting terminal of AMP2 is (R1 k sin(ωt) +VBIAS , a sinusoid
centered around VBIAS . From there, the gain of the noninverting amplifier circuit is VOUT = (1 + RR34 ),
which yields:  
R3

VOUT = 1 + −R1 k sin(ωt) +VBIAS
R4

(e) We would now like to choose VBIAS so that we can get as much gain G out of the non-inverting amplifier
circuit (AMP2) as possible without causing AMP2 to clip (i.e, the output of AMP2 must stay between
0V and 3.3V). What value of VBIAS will achieve this goal? If k = 10−5 and R1 = 10 kΩ, what is the
maximum value of G you can use without having AMP2 clip?

EECS 16B, Fall 2018, Homework 4 9

 Solution: Since the sinusoidal term has zero mean, we want to put it in the middle of AMP2’s range.
In other words, we want the output of AMP2 to have a mean of 3.3−0 2 = 1.65 V. Since the non-inverting
R3
amplifier has a gain of G = 1 + R4 , in order to achieve this we need to set Vbias = 1.65
G
V

3.3V −0V
Therefore, we should choose VBIAS = 2 = 1.65V as the optimum VBIAS .

VOUT = G(−R1 k sin(ωt) +VBIAS )

1.65 V
Letting VBIAS = G :
3.3V − 1.65V = −GR1 k sin(ωt)
1.65V = −GR1 k sin(ωt)
Letting sin(ωt) = −1, its maximum value:

1.65V = GR1 k

1.65V = G(104 Ω)(10−5 A)

1.65
G= = 16.5
0.1

5. Color Organ Filter Design

In the fourth lab, we will design low-pass, band-pass, and high-pass filters for a color organ. There are red,
green, and blue LEDs. Each color will correspond to a specified frequency range of the input audio signal.
The intensity of the light emitted will correspond to the amplitude of the audio signal.

(a) First, you realize that you can build simple filters using a resistor and a capacitor. Design the first-order
passive low and high pass filters with following frequency ranges for each filter using 1 µF capacitors.
(“Passive” means that the filter does not require any power supply.)
• Low pass filter – 3-dB frequency at 2400 Hz = 2π · 2400 rad
sec
• High pass filter – 3-dB frequency at 100 Hz = 2π · 100 rad
sec
Draw the schematic-level representation of your designs and show your work finding the resistor val-
ues. Also, please mark Vin , Vout , and ground nodes in your schematic. Round your results to two
significant figures.
Solution:
i. Low-pass filter
1
f3 dB = = 2400 Hz
2πRC
Therefore, we need a 66 Ω resistor.
66 Ω
Vin Vout

1 µF

EECS 16B, Fall 2018, Homework 4 10

 ii. High-pass filter
1
f3 dB = = 100 Hz
2πRC
Therefore, we need a 1.6 kΩ resistor.
1 µF
Vin Vout

1.6 kΩ

(b) You decide to build a bandpass filter by simply cascading the first-order low-pass and high-pass filters
you designed in part (a). Connect the Vout node of your low-pass filter directly to the Vin node of your
high pass filter. The Vin of your new band-pass filter is the Vin of your old low-pass filter, and the Vout
of the new filter is the Vout of your old high-pass filter. What is HBPF , the transfer function of your
new band-pass filter? Use RL , CL , RH , and CH for low-pass filter and high-pass filter components,
respectively. Show your work.
Solution:
CH
RL
Vin Vout

CL RH

 
1 1
jωCH + RH
 
1 1 jωCL 1 + jωRH CH
+ RH k = 1 1
=
jωCH jωCL jωCL + jωCH + RH −ω 2 RH CLCH+ jω(CH +CL )
Therefore, the transfer function from Vin of the low pass filter to Vout of the low pass filter is
 
1 1
jωCH + R H k jωC L 1 + jωRH CH
HLPF =   = 2
R + 1
+R k 1 1 − ω RL RH CLCH + jω(RH CH + RLCL + RLCH )
L jωCH H jωCL

And, the transfer function from Vout of the low pass filter to Vout of the high pass filter is
jωRH CH
HHPF =
1 + jωRH CH
The overall transfer function is
jωRH CH
HBPF = HLPF · HHPF =
1 − ω 2 RL RH CLCH + jω(RH CH + RLCL + RLCH )

EECS 16B, Fall 2018, Homework 4 11

 (c) Plug the component values you found in (a) into the transfer function HBPF . Using MATLAB or
IPython, draw a Bode plot from 0.1 Hz to 1 GHz. If you use iPython, you may find the function
scipy.signal.bode useful. What are the frequencies of the poles and zeros? What is the maxi-
mum magnitude of HBPF in dB? Is that something that you want? If not, explain why not and suggest
a simple way (either adding passive or active components) to fix it.
Solution:

jω(1.6 · 10−3 )
HBPF =
1 − ω 2 (1.1 · 10−7 ) + jω(1.7 · 10−3 )
The Bode plot is as below.

There are two poles and one zero at 100 Hz, 2.4 kHz, and DC, respectively. The maximum magnitude
(around 500 Hz = 3.14 × 103 rad s ) is
 
 j(3.14 · 103 )(1.6 · 10−3 )  V
 = 0.94 = −0.52 dB
 
 1 − (3.14 · 103 )2 (1.1 · 10−7 ) + j(3.14 · 103 )(1.7 · 10−3 )  V


This is pretty similar to what we wanted. The gain, |HBPF |, is close to 0 dB at its maximum. However,
the transfer function of the bandpass filter that we likely intended to get by cascading the two filter
circuits was:

jωRH CH
Hideal BPF =
(1 + jωRH CH )(1 + jωRLCL )
jωRH CH
=
1 − ω 2 RH CH RLCL + jω(RLCL + RH CH )

EECS 16B, Fall 2018, Homework 4 12

 Therefore, in our circuit, only the jωRLCH term is added at the denominator. Because RL = 66 Ω is
small, it did not cause any significant problem in our case. jωRLCH is added because the low pass filter
is experiencing impedance loading from the high pass filter, leading to a change in HLPF . However, to
be safe, a simple solution is to place a voltage buffer between the filters as below.
Note that the ideal voltage buffer has infinite input impedance and zero output impedance. This blocks
any load effects from the following stage, and the next stage will see the op-amp output as an ideal
voltage source.

CH
−
RL Vout
Vin +

RH
CL

(d) Now that you know how to make filters and amplifiers, we can finally build a system for the color
organ circuit below. Before going into the actual schematic design, you must first set specifications for
each block. The goal of the circuit is to divide the input signal into three frequency bands and turn the
LEDs on based on the input signal’s frequency.
In this problem, assume that the mic board is a 3-pole 2-zero system. Poles are located at 10 Hz,
100 Hz, and 10 000 Hz. Zeros are at DC and 200 Hz. This means that the frequency response at the
mic board output can be modeled as follows.
 
jω 1 + ωjωz1
VMIC = KMIC    
1 + ωjωp1 1 + ωjωp2 1 + ωjωp3
where KMIC is a constant gain, ωz1 , ω p1 , ω p2 , and ω p3 are the zero and poles. Note that jω term in the
numerator denotes the zero at DC. Also note that poles are always in rad sec : for example, ω p1 = 2π ·10Hz.
The magnitude of the voltage at the mic board output is 1 V peak-to-peak at 40 Hz. (Hint: You can use
this information to calculate KMIC .)
Suppose that the three filters have transfer functions as below.
• Low pass filter
2
HLPF = jω
1 + 200π
• Band pass filter
4.54 · 10−4 jω
HBPF =   
jω jω
1 + 400π 1 + 4000π
• High pass filter
jω
8000π
HHPF = jω
1 + 8000π

EECS 16B, Fall 2018, Homework 4 13

 3
jω(1.5·10 ))
What are the phasor voltages at the output of each filter as a function of ω? To clarify, 3(1+
1+ jω(2·100)
would be a valid phasor voltage at the output of some filter. Assume that there are ideal voltage buffers
before and after each filter.

Solution:
Because we know that we have 1 Vpp at 40 Hz, we can plug 2π · 40 into ω to get KMIC .
   
j(80π) 1 + j(80π)
 
 ωz1

1 = K · 
    
j(80π) j(80π) j(80π) 

 1+ ω p1 1 + ω p2 1+ ω p3 

Therefore, K = 0.017. Finally, the phasor voltages at the output of each filter are as below.
 
jω
jω 1 + 400π
VLPF = 0.034 ·   2  
jω jω jω
1 + 20π 1 + 200π 1 + 20000π

( jω)2
VBPF = 7.72 · 10−6 ·     
jω jω jω jω
1 + 20π 1 + 200π 1 + 4000π 1 + 20000π
 
( jω)2 jω
1 + 400π
8000π
VHPF = 0.017 ·     
jω jω jω jw
1 + 20π 1 + 200π 1 + 8000π 1 + 20000π

(e) For 50 Hz, 1000 Hz, and 8000 Hz, what is the voltage gain required of each non-inverting amplifier
such that the output peak to peak voltage measured right before the 10 Ω resistor is 5 Vpp ?
Solution:
i. Low pass filter path

EECS 16B, Fall 2018, Homework 4 14

 At ω = 100π,
 
   
j100π

 j100π 1 + 400π 

|VLPF | = 0.034 ·   2    = 1.73
j100π

 1+ 20π 1 + j100π
200π
j100π 
1 + 20000π 

V
Therefore, the non-inverting amplifier gain should be 2.9 V (or 9.24 dB).
ii. Band pass filter path
At ω = 2000π,
 
 
 7.72 · 10−6 · ( j2000π)2 
|VBPF | =   = 0.27
        
 1 + j2000π 1 + j2000π 1 + j2000π 1 + j2000π 
 20π 200π 4000π 20000π 

V
Therefore, the non-inverting amplifier gain should be 18.5 V (or 25.3 dB).
iii. High pass filter path
At ω = 16000π,
   
 ( j16000π)2 j16000π
·

 0.017 8000π 1 + 400π

|VHPF | =     = 0.37
   
 1 + j16000π 1 + j16000π 1 + j16000π 1 + j16000π 
 20π 200π 8000π 20000π 

Therefore, the non-inverting amplifier gain should be 13.5 V

V (or 22.6 dB).

EECS 16B, Fall 2018, Homework 4 15