Goel Institute of Technology & Management

Department of Electrical and Electronics Engineering

Subject: Wireless and Mobile Communication (REC 085)
Faculty: Ranjan Das Bagh
Tutorial Sheet 3 based on MCQ
1. The higher the bit rate, the more speech channels can be compressed within a given
bandwidth.
a) True
b) False
Answer: b
Explanation: The lower the bit rate at which the coder can deliver toll quality speech, the
more speech channels can be compressed within a given bandwidth. Thus, manufacturers
are continuously in search of speech coders that provide toll quality speech at lower bit
rates.

2. Which of the following are two types of speech coders?

a) Waveform coders and source coders
b) Active coders and passive coders
c) Direst coders and indirect coders
d) Time and frequency coders
Answer: a
Explanation: Speech coders can be categorised into waveform coders and source coders.
Waveform coders can further be categorised into time domain and frequency domain.
Source coders can be classified into linear predictive coders and vocoders.

3. Waveform coders has _______ complexity and achieves _______ economy in transmission bit
rate.
a) Maximum, moderate
b) Maximum, high
c) Minimal, moderate
d) Minimal, high
Answer: c
Explanation: Waveform coders have minimal complexity. This class of coders achieves only
moderate economy in transmission bit rate. They are designed to be source independent
and hence code equally well a variety of signals.

4. Vocoders has _______ complexity and achieves _______ economy in transmission bit rate.
a) Maximum, moderate
b) Maximum, high
c) Minimal, moderate
d) Minimal, high
 Answer: b
Explanation: Vocoders achieve very high economy in transmission bit rate. They are in
general more complex. They are based on using a priori knowledge about the signal to be
coded, and for this reason, they are signal specific.

5. Which of the following is not a property that is utilized in coder design?

a) Non zero autocorrelation between successive speech signals
b) Non flat nature of speech signal
c) Quasiperiodicity of voiced speech signals
d) Uniform probability distribution of speech amplitude
Answer: d
Explanation: Speech waveforms have a number of useful properties that can be exploited
when designing efficient coders. They are non uniform probability distribution of speech
amplitude, non-zero autocorrelation between successive speech samples, the nonflat
nature of the speech spectra and quasi periodicity of voiced speech signals.

6. Speech waveforms are _______

a) Bandlimited
b) Bandpass
c) High pass
d) Infinite bandwidth
Answer: a
Explanation: The most basic property of speech waveforms that are exploited by all speech
coders is that they are bandlimited. A finite bandwidth means that it can be time-
discretized at a finite rate and reconstructed complexity from its samples.

7. Which of the following is not a property of pdf of speech signals?

a) Non uniformity
b) Very high probability of non-zero amplitudes
c) Significant probability of very high amplitudes
d) Increasing function of amplitudes between these extremes
Answer: d
Explanation: There is a non-uniform probability distribution of speech amplitude. The pdf
of a speech signal is in general characterized by a very high probability of non-zero
amplitudes, a significant probability of very high amplitudes, and a monotonically
decreasing function of amplitudes between these extremes.

8. Power spectral density of speech is flat.

a) True
b) False
Answer: b
Explanation: There is a nonflat characteristic in power spectral density of speech. It makes
it possible to obtain significant compression by coding speech in the frequency domain.

9. Vocoders analyse the speech signals at ______

a) Transmitter
 b) Receiver
c) Channel
d) IF Filter
Answer: a
Explanation: Vocoders are a class of speech coding systems. They analyse the speech
signal at the transmitter. And then transmit the parameters derived from the analysis.

10. Vocoders __________ the voice at the receiver.

a) Analyse
b) Synthesize
c) Modulate
d) Evaluate
Answer: b
Explanation: Vocoders synthesize the voice at the receiver. All vocoder systems attempt to
model the speech generation process as a dynamic system and try to quantify certain
physical constraints of the system.

11. Vocoders are simple than the waveform coders.

a) True
b) False
Answer: b
Explanation: Vocoders are much more complex than the waveform coders. They can
achieve very high economy in transmission bit rate but are less robust.

12. Which of the following is not a vocoding system?

a) Linear predictive coder
b) Channel vocoder
c) Waveform coder
d) Formant vocoder
Answer: c
Explanation: Waveform coder is not a vocoding system. LPC (linear predictive coding) is the
most popular vocoding system. Other vocoding systems are channel vocoder, formant
vocoder, cepstrum vocoder etc.

13. Which of the following pronunciations lead to voiced sound?

a) ‘f’
b) ‘s’
c) ‘sh’
d) ‘m’
Answer: d
Explanation: Voiced sounds are ‘m’, ‘n’ and ‘v’ pronounciations. They are a result of
quasiperiodic vibrations of the vocal chord.

14. Speech signal can be categorised in _____ and ______

a) Voiced, unvoiced
b) Active, passive
 c) Direct, indirect
d) Balanced, unbalanced
Answer: a
Explanation: Speech signal is of two types, voiced and unvoiced. Voiced sound is a result of
quasiperiodic vibrations of the vocal chord. Unvoiced signals are fricatives produced by
turbulent air flow through a constriction.

15. Channel vocoders are the time domain vocoders.

a) True
b) False
Answer: b
Explanation: Channel vocoders are frequency domain vocoders. They determine the
envelope of the speech signal for a number of frequency bands and then sample, encode
and multiplex these samples with the encoded outputs of the other filters.

16. ________ is often called the formant of the speech signal.

a) Pitch frequency
b) Voice pitch
c) Pole frequency
d) Central frequency
Answer: c
Explanation: The pole frequencies correspond to the resonant frequencies of the vocal
tract. They are often called the formants of the speech signal. For adult speakers, the
formants are centered around 500 Hz, 1500 Hz, 2500 Hz and 3500 Hz.

17. Formant vocoders use large number of control signals.

a) True
b) False
Answer: b
Explanation: Formant vocoders use fewer control signals. Therefore, formant vocoders can
operate at lower bit rates than the channel vocoder. Instead of transmitting the power
spectrum envelope, formant vocoders attempt to transmit the position of peaks of spectral
envelope.

18. Cepstrum vocoder uses __________

a) Wavelet transform
b) Inverse wavelet transform
c) Cosine transform
d) Inverse Fourier transform
Answer: d
Explanation: Cepstrum vocoders use inverse Fourier transform. It separates the excitation
and vocal tract spectrum by Fourier transforming spectrum to produce the cepstrum of
the signal.
19. Linear predictive coders belong to _______ domain class of vocoders.
a) Time
b) Frequency
c) Direct
 d) Indirect
Answer: a
Explanation: Linear predictive vocoders belong to the time domain class of vocoders. This
class of vocoders attempts to extract the significant features of the speech from the time
waveform.

20. Linear predictive coders are computationally simple.

a) True
b) False
Answer: b
Explanation: Linear predictive coders are computationally intensive. But, they are the most
popular among the class of low bit vocoders. With LPC, it is possible to transmit good
quality voice at 4.8 kbps and poorer quality voice at even lower rates.

21. Linear predictive coding system models the vocal tract as __________ linear filter.
a) Pole and zero
b) All zero
c) All pole
d) No pole
Answer: c
Explanation: The linear predictive coding system models the vocal tract as an all pole linear
filter. The excitation to this filter is either a pulse at the pitch frequency or random white
noise depending on whether the speech segment is voiced or unvoiced.

22. Linear predictive vocoders use __________ to estimate present sample.

a) Weighted sum of past samples
b) Multiplication of past samples
c) One past sample
d) Do not use past samples
Answer: a
Explanation: The linear predictive coder uses a weighted sum of p past samples. Using this
technique, the current sample can be written as linear sum of the immediately precoding
samples.

23. Which of the following LPC uses code book?

a) Multiple excited LPC
b) Residual excited LPC
c) LPC Vocoders
d) Code excited LPC
Answer: d
Explanation: Code excited LPC uses code book. In this method, the coder and decoder
have a predetermined code book of stochastic (zero mean white Gaussian) excitation
signals.

24. How many past samples are used by linear predictive coders to estimate present sample?
a) 100-150
b) 10-15
 c) 1
d) 1000-1100
Answer: b
Explanation: LPCs uses weighted sum of past p samples to estimate the present samples.
The number of past samples used by linear predictive coders ranges from 10 to 15.

25. Which of the non-linear transform is generally used to improve the coding of reflection
coefficient?
a) Long area ratio transform
b) Mutual information
c) Least square
d) Interpolation
Answer: a
Explanation: Long area ratio (LAR) transform is generally used to improve the coding of
reflection coefficient. This non linear transformation reduces the sensitivity of reflection
coefficients to quantization errors. LAR performs an inverse hyperbolic tangent mapping of
reflection coefficients.

26. Which of the following LPC uses two sources at the receiver?
a) Multiple excited LPC
b) Residual excited LPC
c) LPC Vocoders
d) Code excited LPC
Answer: c
Explanation: LPC vocoder uses two sources at the receiver, one of white noise and the
other with a series of pulses at the current pitch rate. The selection of either of these
excitation methods is based on voiced/unvoiced decision made at the transmitter.

27. Which of the following LPC produces a buzzy twang in the synthesized speech?
a) Multiple excited LPC
b) Residual excited LPC
c) LPC Vocoders
d) Code excited LPC
Answer: c
Explanation: LPC vocoder requires that the transmitter extract pitch frequency information
which is often very difficult. Moreover, the phase coherence between the harmonic
components of the excitation pulse tends to produce a buzzy twang in the synthesized
speech.

28. Linear predictive coders are computationally simple.

a) True
b) False
Answer: b
Explanation: Linear predictive coders are computationally intensive. But, they are the most
popular among the class of low bit vocoders. With LPC, it is possible to transmit good
quality voice at 4.8 kbps and poorer quality voice at even lower rates.
29. Linear predictive coding system models the vocal tract as __________ linear filter.
a) Pole and zero
b) All zero
c) All pole
d) No pole
Answer: c
Explanation: The linear predictive coding system models the vocal tract as an all pole linear
filter. The excitation to this filter is either a pulse at the pitch frequency or random white
noise depending on whether the speech segment is voiced or unvoiced.

30. Linear predictive vocoders use __________ to estimate present sample.

a) Weighted sum of past samples
b) Multiplication of past samples
c) One past sample
d) Do not use past samples
Answer: a
Explanation: The linear predictive coder uses a weighted sum of p past samples. Using this
technique, the current sample can be written as linear sum of the immediately precoding
samples.

31. Which of the following LPC uses code book?

a) Multiple excited LPC
b) Residual excited LPC
c) LPC Vocoders
d) Code excited LPC
Answer: d
Explanation: Code excited LPC uses code book. In this method, the coder and decoder
have a predetermined code book of stochastic (zero mean white Gaussian) excitation
signals.

32. Which of the non-linear transform is generally used to improve the coding of reflection
coefficient?
a) Long area ratio transform
b) Mutual information
c) Least square
d) Interpolation
Answer: a
Explanation: Long area ratio (LAR) transform is generally used to improve the coding of
reflection coefficient. This non linear transformation reduces the sensitivity of reflection
coefficients to quantization errors. LAR performs an inverse hyperbolic tangent mapping of
reflection coefficients.

33. Which of the following LPC uses two sources at the receiver?
a) Multiple excited LPC
b) Residual excited LPC
c) LPC Vocoders
d) Code excited LPC
Answer: c
 Explanation: LPC vocoder uses two sources at the receiver, one of white noise and the
other with a series of pulses at the current pitch rate. The selection of either of these
excitation methods is based on voiced/unvoiced decision made at the transmitter.

34. Which of the following LPC produces a buzzy twang in the synthesized speech?
a) Multiple excited LPC
b) Residual excited LPC
c) LPC Vocoders
d) Code excited LPC
Answer: c
Explanation: LPC vocoder requires that the transmitter extract pitch frequency information
which is often very difficult. Moreover, the phase coherence between the harmonic
components of the excitation pulse tends to produce a buzzy twang in the synthesized
speech.

35. Equalization techniques can be categorised into _______ and ______ techniques.
a) Linear, non linear
b) Active, passive
c) Direct, indirect
d) Slow, fast
Answer: a
Explanation: Equalization techniques can be classified into linear and non linear
techniques. These categories are determined from how the output of an adaptive
equalizer is used for subsequent control of the equalizer.

36. Which of the following is not a non-linear equalization technique?

a) Decision feedback equalization
b) Maximum likelihood symbol detection
c) Minimum square error detection
d) Maximum likelihood sequence detection
Answer: c
Explanation: Decision feedback equalization, maximum likelihood symbol detection and
maximum likelihood sequence detection offers non-linear equalization. They offer
improvements over linear equalization techniques and are used in most 2G and 3G
systems.

37. Equalization techniques can be categorised into _______ and ______ techniques.
a) Linear, non linear
b) Active, passive
c) Direct, indirect
d) Slow, fast
Answer: a
Explanation: Equalization techniques can be classified into linear and non linear
techniques. These categories are determined from how the output of an adaptive
equalizer is used for subsequent control of the equalizer.

38. Which of the following factor could not determine the performance of algorithm?
a) Structural properties
 b) Rate of convergence
c) Computational complexity
d) Numerical properties
Answer: a
Explanation: The performance of an algorithm is determined by various factors. These
factors are rate of convergence, computational complexity and numerical properties. The
performance of algorithm does not depend on structural properties.

39. Rate of convergence is defined by __________ of algorithm.

a) Time span
b) Number of iterations
c) Accuracy
d) Complexity
Answer: b
Explanation: Rate of convergence is required as number of iterations required for the
algorithm to converge close enough to the optimum solution. It enables the algorithm to
track statistical variations when operating in non stationary environment.

40. Computational complexity is a measure of ________

a) Time
b) Number of iterations
c) Number of operations
d) Accuracy
Answer: c
Explanation: Computational complexity is the number of operations required to make one
complete iteration of the algorithm. It helps in comparing the performance with other
algorithms.

41. Choice of equalizer structure and its algorithm is not dependent on ________
a) Cost of computing platform
b) Power budget
c) Radio propagation characteristics
d) Statistical distribution of transmitted power
Answer: d
Explanation: The cost of the computing platform, the power budget and the radio
propagation characteristics dominate the choice of an equalizer structure and its
algorithm. Battery drain at the subscriber unit is also a paramount consideration.

42. Coherence time is dependent on the choice of the algorithm and corresponding rate of
convergence.
a) True
b) False
Answer: a
Explanation: The choice of algorithm and its corresponding rate of convergence depends
on the channel data rate and coherence time. The speed of the mobile unit determines the
channel fading rate and the Doppler spread, which is directly related to coherence time of
the channel.
43. Which of the following is not an algorithm for equalizer?
a) Zero forcing algorithm
b) Least mean square algorithm
c) Recursive least square algorithm
d) Mean square error algorithm
Answer: d
Explanation: Three classic equalizer algorithm are zero forcing (ZF) algorithm, least mean
squares (LMS) algorithm and recursive least squares (RLS) algorithm. They offer
fundamental insight into algorithm design and operation.

44. Which of the following is a drawback of zero forcing algorithm?

a) Long training sequence
b) Amplification of noise
c) Not suitable for static channels
d) Non zero ISI
Answer: b
Explanation: The zero forcing algorithm has the disadvantage that the inverse filter may
excessively amplify noise at frequencies where the folded channel spectrum has high
attenuation.

45. Zero forcing algorithm performs well for wireless links.

a) True
b) False
Answer: b
Explanation: ZF is not often used in wireless links as it neglects the effect of noise
altogether. However, it performs well for static channels with high SNR, such as local wired
telephone links.

46. LMS equalizer minimizes __________

a) Computational complexity
b) Cost
c) Mean square error
d) Power density of output signal
Answer: c
Explanation: LMS equalizer is a robust equalizer. It is used to minimize mean square error
(MSE) between the desired equalizer output and the actual equalizer output.

47. For N symbol inputs, LMS algorithm requires ______ operations per iterations.
a) 2N
b) N+1
c) 2N+1
d) N2
Answer: c
Explanation: The LMS algorithm is the simplest algorithm. For N symbol inputs, it requires
only 2N+1 operations per iteration.

48. Stochastic gradient algorithm is also called ________

a) Zero forcing algorithm
 b) Least mean square algorithm
c) Recursive least square algorithm
d) Mean square error algorithm
Answer: b
Explanation: The minimization of the MSE is carried out recursively, and it can be
performed by the use of stochastic gradient algorithm. This more commonly called the
least mean square (LMS) algorithm.

49. Convergence rate of LMS is fast.

a) True
b) False
Answer: b
Explanation: The convergence rate of the LMS algorithm is slow. It is slow due to the fact
that it uses only one parameter i.e. step size that control the adaptation rate.

50. Which of the following does not hold true for RLS algorithms?
a) Complex
b) Adaptive signal processing
c) Slow convergence rate
d) Powerful
Answer: c
Explanation: Recursive least square (RLS) algorithm uses fast convergence rate as opposed
to LMS algorithms. They are powerful, albeit complex, adaptive signal processing
techniques which significantly improves the convergence of adaptive equalizer.

51. Which of the following algorithm uses simple programming?

a) LMS Gradient DFE
b) FTF algorithm
c) Fast Kalman DFE
d) Gradient Lattice DFE
Answer: a
Explanation: Advantages of LMS gradient DFE algorithm are low computational complexity
and simple programming. While fast tranversal filter (FTF) algorithm, Fast Kalman DFE and
gradient lattice DFE uses complex programming.

52. Small scale fades are characterized by ____________ amplitude fluctuations.

a) Large
b) Small
c) Rapid
d) Slow
Answer: c
Explanation: Small scale fades are characterized by deep and rapid fluctuations. They
occur as the mobile system moves over distances of just a few wavelengths. These fades
are caused by multiple reflections from the surrounding in the vicinity of the mobile.

53. ____________ is used to prevent deep fade for rapidly varying channel.
a) Modulation
b) Demodulation
 c) Macroscopic diversity technique
d) Microscopic diversity technique
Answer: d
Explanation: In order to prevent deep fades from occurring, microscopic diversity
techniques can exploit the rapidly changing signal. By selecting the best signal at all times,
a receiver can mitigate small scale fading effects.

54. Large scale fading can be mitigated with the help of _________
a) Modulation
b) Demodulation
c) Macroscopic diversity technique
d) Microscopic diversity technique
Answer: c
Explanation: Large scale fading is mitigated with macroscopic diversity techniques. It is
done by selecting a base station which is not shadowed when others are, the mobile can
improve substantially the average signal to noise ratio.

55. Space diversity s also known as ________

a) Antenna diversity
b) Time diversity
c) Frequency diversity
d) Polarization diversity
Answer: a
Explanation: Space diversity is also known as antenna diversity. It is one of the popular
forms of diversity used in wireless communications. Signals received from the spatially
separated antenna on the mobile would have essentially uncorrelated envelopes for
antenna separation.

56. Which of the following is not a category of space diversity technique?

a) Selection diversity
b) Time diversity
c) Feedback diversity
d) Equal gain diversity
Answer: b
Explanation: Space diversity reception methods can be classified into four categories. They
are selection diversity, feedback diversity, maximal ratio combining and equal gain
diversity.

57. In selection diversity, the gain of each diversity branch provides different SNR.
a) True
b) False
Answer: b
Explanation: Selection diversity uses m demodulators to provide m diversity branches.
Their gain is adjusted to provide the same average SNR for each branch.

58. Polarization diversity uses the ________ as the diversity element.

a) Modulation index
b) Carrier frequency
 c) Reflection coefficient
d) Coherence time
Answer: c
Explanation: Decorrelation of the signal in each polarization is caused by multiple
reflections in the channel between mobile and base station antenna. Reflection coefficient
for each polarization is different, which results in different amplitudes and phases for each
reflection.

59. Which of the factor does not determine the correlation coefficient?
a) Polarization angle
b) Cross polarization discrimination
c) Offset angle from the main beam direction
d) Coherence time
Answer: d
Explanation: The correlation coefficient is determined by three factors, polarization angle,
offset angle from the main beam direction of the diversity antenna, and the cross
polarization discrimination. The correlation coefficient generally becomes higher as offset
angle becomes large.

60. Frequency diversity is implemented by transmitting information on more than one

___________
a) Carrier frequency
b) Amplitude
c) Phase
d) Modulation scheme
Answer: a
Explanation: Frequency diversity is implemented by transmitting information on more than
one carrier frequency. Frequency diversity is often employed in microwave line of sight
links which carry several channels in frequency division multiplex mode.

61. Frequency diversity uses ________ as a diversity element.

a) Correlation coefficient
b) Coherence time
c) Coherence bandwidth
d) SNR
Answer: c
Explanation: The rationale behind the frequency diversity is that frequencies separated by
more than the coherence bandwidth of the channel will be uncorrelated. Thus, they will
not experience the same fade.

62. Frequency diversity is good for low traffic conditions.

a) True
b) False
Answer: b
Explanation: Frequency diversity is not good for low traffic conditions. This technique has a
disadvantage that it not only requires spare bandwidth but also requires that there be as
many receivers as there are channels used for frequency diversity. However, for critical
traffic, the expense may be justified.
63. Time diversity repeatedly transmits information at time spacings that exceed ___________
a) Coherence bandwidth
b) Dwell time
c) Run time
d) Coherence time
Answer: d
Explanation: Time diversity repeatedly transmits information at time spacings that exceed
coherence time of the channel. Thus, multiple repetitions of the signal will be received with
independent fading conditions, thereby providing for diversity.

64. In maximal ratio combining, the output SNR is equal to __________

a) Mean of all individual SNRs
b) Maximum of all SNRs
c) Sum of individual SNR
d) Minimum of all SNRs
Answer: c
Explanation: Maximal ratio combining produces an output SNR equal to the sum of the
individual SNRs. Thus, it has the advantage of producing an output with an acceptable SNR
even when none of the individual signals are themselves acceptable.

65. In CDMA spread spectrum systems, chip rate is less than the bandwidth of the channel.
a) True
b) False
Answer: b
Explanation: In CDMA spread spectrum systems, the chip rate is typically much greater
than the flat fading bandwidth of the channel. Whereas conventional modulation
techniques require an equalizer to undo intersymbol interference between adjacent
channels.

66. The ability of the block code to correct errors is a function of __________
a) Number of parity bits
b) Number of information bits
c) Number of code bits
d) Code distance
Answer: d
Explanation: The ability of a block code to correct errors is a function of the code distance.
Block codes can be used to improve the performance of communication systems when
other means of improvement are impractical.

67. In systematic codes, parity bits are appended at the __________

a) Beginning
b) End
c) End
d) Odd places
 Answer: b
Explanation: A systematic code is one in which the parity bits are appended to the end of
the information bits. For an (n,k) code, the first k bits are identical to the information bits,
and the remaining (n-k) bits of each code word are linear combinations of k information
bits.

68. Which of the following is not an example of block code?

a) Hamming code
b) Cyclic code
c) Convolution code
d) BCH codes
Answer: c
Explanation: Hamming codes, cyclic codes and BCH codes are the example of block codes.
Convolution code does not come in the category of block code. Some other examples of
block codes are Reed Solomon codes, Golay codes and Hadamard codes.

69. Which of the following code is a class of non-binary BCH?

a) Hamming code
b) Hadamard code
c) Golay code
d) Reed Solomon codes
Answer: d
Explanation: The most important and most common class of non binary is the family of
codes known as Reed Solomon codes. BCH codes are among the most popular block codes
that exist for a wide range of rates, achieve significant coding gains.

70. Which of the following linear codes achieve largest possible minimum distance?
a) Hamming code
b) Hadamard code
c) Golay code
d) Reed Solomon codes
Answer: d
Explanation: RS codes achieve the largest possible minimum distance, d min of any linear
code. They are non-binary codes which are capable of correcting errors that appears in
bursts.

71. CDPD stands for ___________

a) Cellular Digital Packet Data
b) Cellular Decoded Packet Data
c) Cellular Demodulated Packet Data
d) Cellular Decoded Plane Data
Answer: a
Explanation: CDPD (Cellular Digital Packet Data) is a wide area mobile data service which
uses unused bandwidth. It was mostly used in AMPS phones. Reed Solomon code in US
CDPD uses m=6 bits per code symbol.