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The final exam for eecs 145m: microcomputer interfacing lab held at the university of california, berkeley in spring 2001. The exam covers topics such as transparent latches, sample and hold amplifiers, successive approximation a/d converters, flash a/d converters, frequency aliasing, spectral leakage, computer-controlled system design for a/d converter testing, fft analysis, and loudspeaker impulse response measurement.
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NAME (please print)
STUDENT (SID) NUMBER
College of Engineering Electrical Engineering and Computer Sciences Berkeley
Total of top 4 Lab Grades
Total of top 4 Question Sections
Lab Participation
Mid-Term #
Mid-Term #
Final Exam
Total Course Grade
_______________ (400 max)
_______________ (100 max)
_______________ (100 max)
_______________ (100 max)
_______________ (100 max)
_______________ (200 max)
_______________ (1000 max)
Answer the questions on the following pages completely, but as concisely as possible. The exam is to be taken closed book. Use the reverse side of the exam sheets if you need more space. Calculators are OK. In answering the problems, you are not limited to the particular equipment you used in the laboratory exercises.
Partial credit can only be given if you show your work.
FINAL EXAM GRADE :
1 __________ (30 max) 2 __________ (70 max)
3 __________ (50 max) 4 __________ (50 max)
TOTAL __________ (200 max)
Problem 1 (total 30 points):
Briefly describe the essential differences between the following pairs of terms:
1 a. (10 points) Transparent latch vs. sample and hold amplifier
1 b. (10 points) Successive Approximation A/D converter vs. Flash A/D converter
1c. (10 points) Frequency aliasing vs. spectral leakage.
2 b. (10 points) List the steps your program must do to measure the first transition voltage V(0,1) of the first A/D converter (pseudocode is OK, so long as the logic is clear).
2 c. (10 points) How would you determine the maximum absolute accuracy error of the A/D? (Explain the procedure in steps or with a flow diagram.)
2 d. (10 points) How would you determine the maximum linearity error?
2 e. (10 points) How would you determine the maximum differential linearity error?
2 f. (10 points) With what accuracy could this system measure the quantities in parts b. , c. , and d. in units of 1 LSB of the A/D?
3. e (4 points) What is the FFT coefficient with the highest index and to what frequency does it correspond?
3. f (4 points) What is the FFT coefficient that corresponds to the highest frequency and what is that frequency?
3. g (6 points) You sample a 4,000 Hz s i n e w a v e with the system and take the FFT. What FFT coefficients should be non-zero?
3. h (6 points) You sample a 4,000 Hz symmetric square wave with the system and take the FFT. What FFT coefficients should be non-zero? (symmetric means 50% high, 50% low)
3. i (6 points) You sample a 4,002 Hz s i n e w a v e with the system and take the FFT. What FFT coefficients should be non-zero?
3. j (6 points) You sample two sinewave signals, one with a frequency of 4,000 Hz and another at a nearby frequency and 10 times smaller in magnitude. How closely can the frequency of the second signal approach 4,000 Hz and still be resolved in the FFT coefficients as a separate peak?
3.k (2 points) How would you change the system to reduce the answer to the previous question 3.j by a factor of two?
3. l (6 points) You sample a sinewave of frequency 2 18 – 84,000 Hz = 178,144 Hz and take the FFT. What FFT coefficients should be non-zero? How does the magnitude of the largest FFT coefficient compare with that you would get if you sampled an 84,000 Hz sinewave?
4 b (10 points) Using the Fourier convolution theorem in the continuous time and frequency domains, what acoustic output waveform d(t) would result from using an arbitrary waveform a(t) as the input to a loudspeaker whose impulse response is c(t)?
4 c (10.points) In the continuous time and frequency domains, determine a function b(t) so that if the convolution of a(t) and b(t) is used as the input to a loudspeaker with impulse response c(t), the loudspeaker will produce an acoustic waveform that is close to a(t)
4 d (10.points) How would you apply the result of part 4c to a digitized version of a(t) (sampled at 100 kHz) to correct for the response of the loudspeaker and produce an acoustic waveform that is close to the digitized version of a(t).