Most Latest Wireless & Mobile Communications MCQs – Speech Coding ( Wireless & Mobile Communications ) MCQs
Latest Wireless & Mobile Communications MCQs
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Wireless & Mobile Communications MCQs – Speech Coding ( Wireless & Mobile Communications ) MCQs
The most occurred mcqs of Speech Coding ( ) in past papers. Past papers of Speech Coding ( Wireless & Mobile Communications ) Mcqs. Past papers of Speech Coding ( Wireless & Mobile Communications ) Mcqs . Mcqs are the necessary part of any competitive / job related exams. The Mcqs having specific numbers in any written test. It is therefore everyone have to learn / remember the related Speech Coding ( Wireless & Mobile Communications ) Mcqs. The Important series of Speech Coding ( Wireless & Mobile Communications ) Mcqs are given below:
Characteristics of Speech Signals
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 quasiperiodicity 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. Auto correlation function measures______ between samples of a speech signal as a function of _______
a) Similarity, frequency
b) Dissimilarity, time
c) Similarity, time
d) Dissimilarity, frequency
Answer: c
Explanation: The autocorrelation function (ACF) gives a quantitative measure of the closeness or similarity between samples of a speech signal as a function of their time separation. In every sample of speech, there is a large component that is easily predicted from the values of the previous samples.
9. 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.
10. Spectral flatness measure is the ratio of ______ and _____
a) Variance, Geometric mean
b) Geometric Mean, Variance
c) Arithmetic mean, geometric mean
d) Geometric mean, arithmetic mean
Answer: c
Explanation: Spectral flatness measure is defined as ratio of arithmetic to geometric mean of the samples of the PSD taken at uniform intervals in frequency. Spectral flatness measure is a qualitative measure of the theoretical maximum coding gain that can be obtained by exploiting the nonflat characteristics of speech spectra.
11. Low frequency signals contribute very little to the total speech signals.
a) True
b) False
Answer: b
Explanation: Lon term averaged PSD’s of speech show that high frequency signals contribute very little to the total speech energy. However, high frequency components are insignificant in energy, they are very important carriers of speech information.
Quantization Techniques
1. Quantization is a process of mapping a ________ range of amplitude of a signal into a finite set of __________ amplitudes.
a) Continuous, discrete
b) Discrete, continuous
c) Discrete, discrete
d) Continuous, continuous
Answer: a
Explanation: Quantization is a process of mapping a continuous range of amplitude of a signal into a finite set of discrete amplitudes. Quantizers are the devices that remove the irrelevancies in the signal.
2. For a n bit quantizer, number of levels is equal to __________
a) n
b) 2n
c) n2
d) 2n
Answer: b
Explanation: A quantizer that uses n bits can have M = 2n discrete amplitude levels. Amplitude quantization is an important step in any speech coding process, and it determines to a great extent the overall distortion as well as bit rate necessary to represent each waveform.
3. Distortion produced by quantization is directly proportion to ________ and inversely proportional to ______
a) Number of levels, step size
b) Amplitude of the signal, step size
c) Square of step size, number of levels
d) Amplitude of the signal, number of levels
Answer: c
Explanation: The distortion introduced by any quantization operation is directly proportional to the square of the step size. It is inversely proportional to the number of levels for a given amplitude range.
4. Signal to quantization noise ratio of a PCM encoder is given by _________
a) 6.02n + α
b) 3n + 6.02α
c) n + 6.02α
d) 6.02n + α
Answer: d
Explanation: Signal to quantization noise (SQNR) ratio of a PCM encoder is related to the number of bits, n used for encoding. It follows the following relation, SQNR= 6.02n + α. Here, α = 4.77 dB for peak SQNR and α = 0 dB for average SQNR.
5. Companding technique used in the US is called ____________
a) μ law
b) A law
c) Hybrid companding
d) Direct companding
Answer: a
Explanation: Companding technique known as μ law is used in US. In Europe, A law companding technique is used.
6. An adaptive quantizer varies its ___________ in accordance to input speech signal power.
a) Level
b) Step size
c) Amplitude
d) Frequency
Answer: b
Explanation: An adaptive quantizer varies its step size in accordance to the input speech signal power. Its characteristics shrink and expand in time like an accordion.
7. Shannon predicted that better performance can be achieved by coding one sample at a time.
a) True
b) False
Answer: b
Explanation: Shannon predicted that better performance can be achieved by coding many samples at a time instead of one sample at a time.
8. ________ is a delayed decision coding technique.
a) Adaptive quantization
b) Uniform quantization
c) Vector quantization
d) Non-uniform quantization
Answer: c
Explanation: Vector (VQ) quantization is a delayed decision coding technique. It maps a group of input samples called a vector to a code book index. A code book is set up consisting of a finite set of vectors covering the entire anticipated range of values.
9. The characteristics of compressor in μ-law companding are _________
a) Continuous in nature
b) Logarithmic in nature
c) Linear in nature
d) Discrete in nature
Answer: a
Explanation: In the μ law companding, the compressor characteristic is continuous. It approximates the linear dependence on x for low input signals and a logarithmic one for high input signals.
10. What is the sequence of operations in PCM?
a) Sampling, quantizing, encoding
b) Quantizing, encoding, sampling
c) Quantizing, sampling, encoding
d) None of the mentioned
Answer: a
Explanation: Sequence of operation in PCM is sampling, quantization and encoding. Sampling and quantizing operations transform an analogue signal to a digital signal. The quantizing and encoding operations are usually performed in the same circuit at the transmitter, which is called an Analogue to Digital Converter (ADC). At the receiver end the decoding operation converts the pulse back into an analogue voltage in a Digital to Analogue Converter (DAC).
11. In Adaptive Delta Modulation, the slope error reduces and ___________
a) Quantization error decreases
b) Quantization error increases
c) Quantization error remains same
d) None of the mentioned
Answer: b
Explanation: ADM reduces slope error, at the expense of increasing quantizing error. This error can be reduced by using a low-pass filter.
Frequency Domain Coding of Speech
1. Frequency domain coders divides the speech signal into __________
a) A set of frequency components
b) A set of different amplitudes
c) A set of time delays
d) A set of phase components
Answer: a
Explanation: In the class of frequency domain coders, the speech signal is divided into a set of frequency components. Each frequency component is quantized and encoded separately.
2. In frequency domain coding of speech, the number of bits used to encode each frequency component is constant.
a) True
b) False
Answer: b
Explanation: Frequency domain coders have an advantage that number of bits used to encode each frequency component can be dynamically varied. They can also be shared among different bands.
3. Quantization is a ________ process.
a) Linear
b) Direct
c) Non-linear
d) Indirect
Answer: c
Explanation: Quantization is a non-linear process. It produces distortion products that are typically broad in spectrum.
4. Sub band coding codes the short time transform of a windowed signal.
a) True
b) False
Answer: b
Explanation: It is function of block transform coding. However, a sub band coder divides the speech signal into many smaller sub bands and encodes each sub band separately according to some perceptual criterion.
Equalization, Diversity And Channel Coding MCQs
5. Which of the following is one of the most frequently used transform in speech coding?
a) Fourier transform
b) Wavelet transform
c) Shearlet transform
d) Discrete cosine transform
Answer: d
Explanation: DCT (discrete cosine transform) is one of the most attractive and frequently used transforms for speech coding. Fast algorithms developed for computing the DCT in a computationally efficient manner are used.
6. What does ATC stands for in speech coders?
a) Automatic transform code
b) Air traffic controller
c) Active thermal convection
d) Adaptive transform coding
Answer: d
Explanation: In speech coding, ATC stands for adaptive transform coding. It is form of frequency domain coder that encodes the speech at bit rates in the range of 9.6 kbps and 20 kbps.
7. Waveform coders and Vocoders are the types of ____________
a) Speech coders
b) Modulation technique
c) Frequency translation methods
d) Channel allocation for transmission
Answer: a
Explanation: Speech coders can be classified into waveform coders and Vocoders. Waveform coders convert the analog signal into digital signal. Vocoders exploit the special properties of speech signal to reduce the bit rate.
8. The type of frequency domain coding that divides the speech signal into sub bands is _____
a) Waveform coding
b) Vocoders
c) Block transform coding
d) Sub-band coding
Answer: d
Explanation: Sub band coding (SBC) is a method where the speech signal is subdivided into several frequency bands and each band is digitally encoded separately. The audible frequency spectrum 20Hz-20 KHz is divided into frequency sub-bands using a bank of finite impulse response (FIR) filter and output of each filter is sampled and encoded.
9. Speech coders are categorized on the basis of __________
a) Signal compression techniques
b) Frequency of signal
c) Bandwidth of the signal
d) Phase of the signal
Answer: a
Explanation: Speech coders are categorised on the basis of signal compression techniques. Speech coding is an art of compressing and then encoding speech signals.
10. The speech coding technique that is dependent on the prior knowledge of the signal is __________
a) Waveform coders
b) Vocoders
c) Sub band coding
d) Block transform
Answer: b
Explanation: Vocoders are dependent on the prior knowledge of the signals. They capture the characteristic elements of audio signal and then uses this characteristic signal to affect other audio signals.
Vocoders
1. 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.
2. 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.
3. 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.
4. 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.
5. 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.
6. 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.
7. Channel vocoders are the time domain vocoders.
a) True
b) False
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.
7. 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.
8. ________ 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.’
9. 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.
10. 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.
Linear Predictive Coders
1. 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.
2. 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.
3. 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.
4. 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.
5. 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.
6. 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.
7. 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.
8. 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.
9. 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.
10. The problem of buzzy twang in synthesized speech is mitigated by multipulse excited LPC or code excited LPC.
a) True
b) False
Answer: a
Explanation: LPC vocoder produces buzzy twang in the synthesized speech due to phase coherence between the harmonic components of the excitation pulses. This problem is mitigated by multipulse excited or code excited LPC.
11. Multipulse excited LPC requires pitch detection.
a) True
b) False
Answer: b
Explanation: Multipulse excited LPC does not require pitch detection and the prediction residual is better approximated by several pulses per pitch period. This is the reason for better speech quality.
Speech Codecs
1. The choice of speech coder does not depend on cell size used.
a) True
b) False
Answer: b
Explanation: The choice of speech coder depends on the cell size used. When the cell size is sufficiently small such that high spectral efficiency is achieved through frequency reuse, it may be sufficient to use a simple high rate speech codec.
2. Which of the following is an important factor in determining spectral efficiency of the system?
a) Multiple access technique
b) Cell size
c) Modulation technique
d) Vocoder
Answer: a
Explanation: The type of multiple access technique used is an important factor in determining the spectral efficiency of the system. It strongly influences the choice of speech codec.
3. The type of modulation does not affect the choice of speech codec.
a) True
b) False
Answer: b
Explanation: The type of modulation employed has a considerable impact on the choice of speech codec. Using bandwidth efficient modulation scheme can lower the bit rate reduction requirements on the speech codec and vice versa.
4. Which of the following is the name of original speech coder used in the pan European digital cellular standard GSM?
a) Multipulse excited codec
b) Residual excited codec
c) Regular pulse excited long term prediction
d) Code excited codec
Answer: c
Explanation: The original speech coder used in the pan European digital cellular standard GSM goes by a rather grandiose name of regular pulse excited long term prediction (RPE-LTP) codec. This codec has a bit rate of 13 kbps.
5. Which of the following is true for baseband RELP codec?
a) Good quality of speech, low complexity
b) Good quality of speech, high complexity
c) Bad quality of speech, low complexity
d) Bad quality of speech, high complexity
Answer: a
Explanation: The advantage of baseband RELP codec is that it provides good quality speech at low complexity. The speech quality is sometimes limited due to tonal noise introduced by the process of high frequency generation.
6. Which of the following is true for MPE-LTP codec?
a) Good quality of speech, low complexity
b) Good quality of speech, high complexity
c) Bad quality of speech, low complexity
d) Bad quality of speech, high complexity
Answer: b
Explanation: The MPE-LTP technique produces excellent speech quality at high complexity. It is not much affected by bit errors present in the channel.
7. RPE-LTP codec combines the advantage of RELP codec and CELP codec.
a) True
b) False
Answer: b
Explanation: The RPE-LTP codec combines the advantages of the earlier French proposed RELP codec with those of the multipulse excited long term prediction (MPE-LTP) codec proposed by Germany.
8. Which of the following codec is used by IS-136?
a) Residual Excited Linear Predictive Coders
b) Multipulse Excited LPC
c) LPC Vocoders
d) Vector sum excited LPC
Answer: d
Explanation: The US digital cellular system, IS-136 uses a vector sum excited linear predictive coder (VSELP). This coder operates at a raw data rate of 7950 bits/s and a total data rate of 13 kbps after channel coding.
9. VSELP speech coder is a variant of ___________
a) CELP
b) MPE_LTP
c) RELP
d) RPE-LTP
Answer: a
Explanation: The VSELP speech coder is a variant of the CELP type vocoders. The code books in the VSELP encoder are organised with a predefined structure such that a brute-force search is avoided.
10. Which of the following is true for VSELP?
a) Low speech quality, modest computational complexity, robust to channel errors
b) Highest speech quality, low computational complexity, channel errors
c) Highest speech quality, high computational complexity, robust to channel errors
d) Highest speech quality, modest computational complexity, robust to channel errors
Answer: d
Explanation: VSELP speech coder is designed to accomplish the three goals of highest speech quality, modest computational complexity and robustness to channel errors. The code books used by VSELP impart high speech quality and increased robustness to channel errors.
11. What is DAM in speech coding system?
a) Diagnostic Acceptability Measure
b) Digital Acceptability Measure
c) Diagnostic Accessibility Measure
d) Digital Accessibility Measure
Answer: a
Explanation: The diagnostic acceptability measure is used in speech coding system. It is used for evaluation of acceptability of speech coding systems.
12. ________ exaggerates the bit errors originally received at the base station.
a) Non linear transformation
b) Tandem signalling
c) Large cell size
d) Complex vocoders
Answer: b
Explanation: Tandem signalling tends to exaggerate the bit errors originally received at the base station. Tandem signalling is difficult to protect against but is an important evaluation criterion in the evaluation of speech coders.