Wireless and Mobile Communication

(KEC-076)
Unit 2 Notes
Vocoder and Synthetic model of vocoders-
Vocoder is an audio processor that is used to transmit speech or voice signal in the form of
digital data. The vocoder is used as short form for voice coder. Vocoders are basically used
for digital coding of speech and voice simulation. The bitrate for available narrowband
vocoders is from 1.2 to 64 kbps.
Vocoder operates on the principle of formants. Formants are basically the meaningful
components of a speech that is generated due to the human voice.
Whenever a speech signal is transmitted, it is not needed to transmit the precise waveform. We
can simply transmit the information by which one can reconstruct that particular waveform.
This reconstructed waveform at the receiver must be similar and not identical to the waveform
actually transmitted.
Vocoder works in such a way that it first captures the characteristic element of the signal. Then
other audio signals are affected by the use of that characteristic signal. Vocoders are used
for voice synthesis. The vocoder takes two signals and creates a third signal using the spectral
information of the two input signals. It aims to emblem the amplitude and frequency
characteristic of speech signal onto the synthesis signal, while maintaining the pitch of the
speech signal.

Block diagram of synthetic speech production model for vocoders

As we can see in the above figure of speech model used in Vocoder. Here, voiced sounds are
simulated by the impulse generator, the frequency of which is equal to the fundamental
frequency of vocal cords. The noise source present in the circuit is used to simulate the
unvoiced sounds.
The position of the switch helps in determining whether the sound is voiced or unvoiced.
Then the selected signal is passed through a filter that simulates the effect of mouth, throat and
nasal passage of speaker. The filter unit then filters the input in such a way so as the required
letter is pronounced. Thus we can have a synthesised approximated speech waveform.
LPC is extensively used in case of speech and music application. LPC is an acronym
for Linear Predictive Coding. It is basically a technique to estimate future values. In simple
words we can say, by analysing two previous samples it predicts the outcome. Vocoder is
comprised of voice encoder and decoder.

Types of Vocoders
Channel Vocoders:
• Channel vocoders are parametric, frequency domain vocoders.
• The channel Vocoders was the first analysis-synthesis systems of speech demonstrated
practically.
• Channel Vocoders are frequency domain Vocoders that 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.
• The sampling is done synchronously every 10 ms to30 ms.
• Channel vocoder uses a bank of filter or digital signal processor to divide the signal
into several sub-bands.
• Along with the energy information about each band, the voiced/unvoiced decision, and
the pitch frequency for voiced speech are also transmitted.

Formant Vocoders:
• The formant vocoder is similar in concept to the channel vocoder as parametric
vocoder.
• The formant vocoder can operate at lower bit rates than the channel vocoder because it
uses fewer control signals
 • The formant vocoder attempts to transmit the positions of the peaks (formants) of the
spectral envelope, instead of sending samples
• A formant vocoder must be able to identify at least three formants for representing the
speech sounds
• Formant vocoder can reproduce speech at bit rates lower than 1200 bits/s
• It must control the intensities of the formants.

CELP coder with forward and backward approach

A block diagram of CELP analysis-by-synthesis coder is shown in the Fig below. It is called
analysis by synthesis because we encode and then decode the speech at the encoder and then
find the parameters that minimize the energy of the error signal. The CELP coder is optimized
by using a code book to find the best match for the signal the LP residual is vector quantized
typically 40 samples using 1024 code book entries. Coding requires more computation than
decoding. This method reduces complexity and required data transmission rate is achieved.
The excitation code book is a set of Random numbers where all parameters of the system are
obtained by minimizing the final mean squared error. CELP coder with two different
approaches are shown below. The code excited First LP analysis is used to estimate the vocal
system impulse response in each frame. Then the synthesized speech is generated at the
encoder by exciting the vocal system filter. The difference between the synthetic speech and
the original speech signal constitutes an error signal, which is spectrally weighted to
emphasize perceptual important frequencies and then minimized by optimizing the excitation
signal.

CELP coder (a) Forward approach (b) Backward approach

Spread Spectrum Modulation
Initially developed for military applications during II world war, that was less sensitive to
intentional interference or jamming by third parties. Spread spectrum technology has
blossomed into one of the fundamental building blocks in current and next-generation wireless
systems. Spread spectrum system can be classified into two types;
Averaging System: In this system interference reduction takes place because the interference
can be averaged over a large time interval.
Avoidance System: In this system reduction of interference occurs because the signal is made
to avoid the interference for a large fraction of time

Direct Sequence Spread Spectrum (DSSS) signal using BPSK

Direct spread spectrum (DSS) communications make signals appear wide band and noise-like.
It is this very characteristic that makes DSS signals possess the quality of Low Probability of
Intercept. DSS signals are hard to detect on narrow band equipment because the signal's
energy is spread over a bandwidth of maybe 100 times the information bandwidth. The spread
of energy over a wide band, or lower spectral power density, makes DSS signals less likely to
interfere with narrowband communications. Narrow band communications, conversely, cause
little to no interference to SS systems because the correlation receiver effectively integrates
over a very wide bandwidth to recover an SS signal. The correlator then "spreads" out a
narrow band interferer over the receiver's total detection bandwidth. Since the total integrated
signal density or SNR at the correlator's input determines whether there will be interference or
not. All SS systems have a threshold or tolerance level of interference beyond which useful
communication ceases. This tolerance or threshold is related to the SS processing gain.
Processing gain is essentially the ratio of the RF bandwidth to the information bandwidth.
 In the above diagram Input data is fed into a channel encoder Which Produces analog signal
with narrow bandwidth. Signal is further modulated using sequence of digits, spreading code
or spreading sequence Generated by pseudo noise, or pseudo-random number generator. Effect
of modulation is to increase bandwidth of signal to be transmitted. On receiving end, digit
sequence is used to demodulate the spread spectrum signal. Signal is fed into a channel
decoder to recover data.

Pseudo Noise codes in Direct Spread Spectrum-

In DSSS spreading is achieved by representing each bit of the original message sequence using
several bits. This process is achieved by using a spreading code in this case a PN sequence.
The PN sequence is XOR-ed with the message sequence.

As a case in point, consider a case where the PN sequences are produced at a frequency that is
10 times higher than the message frequency. In this case, the message frequency will be spread
by a factor of 10 after the XOR process. This step details the transmission process in a DSSS
system. At the receiver, the transmitted signal is again XOR-ed using a PN sequence that is
similar to the one originally used at the transmitter. In this way the original message sequence
can be recovered.
.

Transmitter

Receiver
The code acquisition process takes place over a finite amount of time. When longer PN
sequences are used the acquisition time may be longer. In order to appreciate this fact, a brief
description of what acquisition involves is necessary. During acquisition, the PN sequence
received from the receiver is compared with the one that is to be employed at the transmitter. If
the correlation is below a set threshold, the PN sequence at the receiver is delayed by a single
time period of the PN sequence. This process goes on and on recursively until a set threshold is
attained. Once this threshold is attained acquisition is said to have occurred.
Properties of pseudo noise codes-
Balance Property: In each period of the sequence the number of binary 1’s differ from that of
binary 0’s by atmost one digit.
Consider a PN code 0001 0011 0101 111 (7 zeros and 8 ones)
Run length property: Among the runs of 1’s and 0’s in each period, it is desirable that about
one-half of the runs of each type are of length one, one-fourth are of two, one-eight are of
length three and so on.
Number of runs=8 000(3) 1(1) 00(2) 11(2) 0(1) 1(1) 0(1) 1111(4)
Autocorrelation Property: The auto correlation function of a maximal length sequence is
periodic and binary valued.

PN code generator using linear feedback shift register and its application for data
spreading
PN sequences are usually generated using Linear Feedback Shift Registers (LFSR) based on
Galois Field arithmetic. The length of the PN sequence depends on the number of shift register
stages. If there are m shift registers used, the maximum possible PN sequence length, p is
given as in below equation:

Such a sequence is referred to as a maximal length sequence and they are obtained from
standard Galois Field Tables for the generation of maximal length sequences.
As an example consider the case of a 4-stage LFSR used to generate PN sequences. The
generator polynomial that yields the maximal length sequence in such a case is given by
equation below. For other n-stage LFSRs, the generator polynomials for maximal length
sequences can be obtained from standard generator polynomial tables.

Figure below shows the connections in a maximal length sequence 4-stage LFSR connection.

Connections in a maximal length sequence 4-stage LFSR

The initial states of X1, X2, X3 and X4 can be any value but 0000 respectively. An initial state
of 0000 would lock the output to 0. Assuming that the initial state was 1000, the output
sequence for 15 clock pulses is:
000100110101111
After 15 clock pulses the same sequence would again repeat.

Frequency-hopping spread spectrum

A pseudo-noise sequence is generated at the modulator is used in conjunction with an M-ary
FSK modulation to shift the carrier frequency of the FSK signal pseudorandomly, at the
hopping rate. Depending upon this we have two types of frequency hop.
1. Slow frequency hopping:- In which the symbol rate Rs of the MFSK signal is an integer
multiple of the hop rate Rh. That is several symbols are transmitted on each frequency hop.
2. Fast – Frequency hopping:- In which the hop rate Rh is an integral multiple of the MFSK
symbol rate Rs. That is the carrier frequency will hoop several times during the transmission of
one symbol. A common modulation format for frequency hopping system is that of M- ary
frequency – shift – keying (MFSK)
• The transmitted signal occupies a number of frequencies in time, each for a period of time Th
(=1/Rh), referred to as dwell time.
• FHSS divides the available bandwidth into N channels and hops between these channels
according to the PN sequence.
• The transmitted bandwidth is determined by the lowest and highest hop positions and by the
bandwidth per hop positionIn the spread spectrum, the information is transmitted in short
breaks of data at different carrier frequencies. In the frequency-hopping spread spectrum
(FHSS), each component is transmitted at a different carrier frequency. Conversely, multi-
carrier systems (such as OFDM) transmit multiple signals on a single carrier frequency. The
spread spectrum can be used to send independent digital data streams across a noisy channel
by assigning different ‘slots’ to each signal.
The incoming binary data are applied to an M-ary FSK modulator. The resulting modulated
wave and the output from a digital frequency synthesizer are then applied to a mixer that
consists of a multiplier followed by a band – pass filter. The filter is designed to select the sum
frequency component resulting from the multiplication process as the transmitted signal. An
‘k’ bit segments of a PN sequence drive the frequency synthesizer, which enables the carrier
frequency to hop over 2 n distinct values. Since frequency synthesizers are unable to maintain
phase coherence over successive hops, most frequency hops spread spectrum communication
system use non coherent M-ary modulation system.
In FHSS systems, the transmitted power is concentrated on one or a few carriers at a time. The
carrier frequencies are chosen in accordance with a pseudo-random sequence or hopping
sequence that changes periodically, so as to prevent long-term predictability of the carrier
frequencies used. The receiver correlates received signals against the sequence of the received
signals to determine which doesn’t interfere from noise and interference.
In the receiver the frequency hopping is first removed by mixing the received signal with the
output of a local frequency synthesizer that is synchronized with the transmitter. The resulting
output is then band pass filtered and subsequently processed by a non coherent M-ary FSK
demodulator. To implement this M-ary detector, a bank of M non coherent matched filters,
each of which is matched to one of the MFSK tones is used. By selecting the largest filtered
output, the original transmitted signal is estimated.

Time Hopping Spread Spectrum

A time hopping system is a spread spectrum system in which the period and duty cycle of a
pulsed RF carrier are varied in a pseudorandom manner under the control of a coded sequence
• Time hopped spread spectrum systems have found no commercial application to date.
However, the arrival of cheap random-access memory (RAM) and fast micro-controller chips
make time hopping a viable alternative spread spectrum technique for the future.
• Time hopping is a system in which burst signal are initiated at pseudo random rate. In this the
transmitter is switched ON and OFF by a code sequence. The main difference between a
frequency hopping and time hopping system is that in the former the transmitted frequency
changes at each code chip time in the later the frequency changes occurs only at zero/ one
transitions in the code sequence.
Transmission of THSS

Reception of THSS

Multi Carrier Modulation Technique

Multi-Carrier Modulation (MCM) is the principle of transmitting data by dividing the stream
into several bit streams, each of which has a much lower bit rate, and by using these
substreams to modulate several carriers.
Orthogonal Frequency Division Multiplexing
OFDM is a Multi-Carrier Modulation (MCM) scheme, which uses closely spaced multiple
subcarriers to transmit data. Data to be transmitted is split and transmitted using multiple
subcarriers instead of using a single carrier. The key idea is instead of transmitting at a very
high bit rate, the data is transmitted over multiple subchannels each carrying lower bit rates.
The subcarriers are chosen such that they are orthogonal to each other. This ensures that data
from one subcarrier does not interfere with the data on the other. So many bits can be packed
onto the subcarriers simultaneously that the data rate of each subcarrier’s modulation can be
much lower than that of a single-carrier architecture. Lower subcarrier data rates combined
with their orthogonality make the system much less susceptible to intersymbol interference
(ISI) as well.
To maintain orthogonality between subcarriers, the subcarriers are chosen such that they are all
integer multiples of the base frequency. If the total bandwidth of the system is B Hz. Then the
base frequency (f0) is given by B/N, where N is the number of subcarriers in the system. The
subcarriers used are f0, 2f0, 3f0 ... (N-1)f0.

Advantage of OFDM over FDM

Unlike the traditional Frequency Division Multiplexing (FDM), the OFDM does not use guard
bands to separate the various subchannels. One of the key features of OFDM is the
orthogonality of the subcarriers used to transmit data. The orthogonality of subcarriers results
in more subcarriers in a given bandwidth. This improves spectral efficiency. It also eliminates
the interference between subcarriers, often called Inter-Carrier Interference (ICI).
Zero Intersymbol Interferance

Nyquist criteria for ISI cancellation-

Nyquist proposed a condition for pulses p(t) to have zero–ISI when transmitted through a
channel with sufficient bandwidth to allow the spectrum of all the transmitted signal to pass.
Nyquist proposed that a zero–ISI pulse p(t) must satisfy the condition

A pulse that satisfies the above condition at multiples of the bit period Tb will result in zero–
ISI if the whole spectrum of that signal is received. The reason for which these zero–ISI pulses
(also called Nyquist–criterion pulses) cause no ISI is that each of these pulses at the sampling
periods is either equal to 1 at the center of pulse and zero the points other pulses are centered.
In the figure below, when ωx = 0, the spectrum becomes a rect function, and therefore the
pulse p(t) becomes the usual sinc function. For ωx = b/2, the spectrum is similar to a sinc
function but decays (drops to zero) much faster than the sinc (it extends over 2 or 3 bit periods
on each side). The expense for having a pulse that is short in time is that it requires a larger
bandwidth than the sinc function (twice as much for ωx =ω b/2). Sketch of the pulses and their
spectrum for the two extreme cases of ω x =ωb/2 and ωx = 0 are shown below
Pulse shaping Techniques
Several different types of sinc pulses can be applied to the modulated signal to implement a
pulse-shaping filter. These are the raised cosine filter, the root raised cosine filter, and the
Gaussian filter.
Raised Cosine Filter:
The raised cosine filter is one of the most common pulse-shaping filters in communications
systems. In addition, it is used to minimize intersymbol interference (ISI) by attenuating the
starting and ending portions of the symbol period. Because these portions are most susceptible
to creating interference from multi-path distortion, the shaping characteristics of the raised
cosine filter helps reduce ISI. This impulse response for this filter is given by the equation
shown below:

As the equation shows, the sinc pulse is implemented in the creation of this filter. The filter
rolloff parameter, alpha(α), can range between values of 0 and 1.
Square Root Raised Cosine Filter:
The root raised cosine filter produces a frequency response with unity gain at low frequencies
and complete at higher frequencies. It is commonly used in communications systems in pairs,
where the transmitter first applies a root raised cosine filter, and then the receiver then applies
a matched filter.
Mathematically, the raised cosine filter can be defined by the following equation:

In this equation, α is the rolloff factor, which determines the sharpness of the frequency
response. In addition, R is the number of samples per symbol. As the equation above
illustrates, the sinc pulse is used to shape the filter so that it appears with a finite frequency
response.
Gaussian Filter
The Gaussian filter is a pulse shaping technique that is typically used for frequency shift
keying (FSK) and minimum shift keying (MSK) modulation. This filter is unlike the raised
cosine and root raised cosine filters because it does not implement zero crossing points. The
impulse response for the Gaussian filter is defined by the following equation:

Diversity
Different Types Of Diversity Techniques Used In Communication System
Diversity is a method used to develop information from several signals transmitted over
independent fading paths. It exploits the random nature of radio propagation by finding
independent signal paths for communication. It is a very simple concept where if one path
undergoes a deep fade, another independent path may have a strong signal.
Space Diversity:
A method of transmission or reception, or both, in which the effects of fading are minimized
by the simultaneous use of two or more physically separated antennas, ideally separated by
one half or more wavelengths. Signals received from spatially separated antennas have
uncorrelated envelopes.
Polarization Diversity:
Polarization Diversity relies on the decorrelation of the two receive ports to achieve diversity
gain. The two receiver ports must remain cross-polarized. Polarization Diversity at a base
station does not require antenna spacing. Polarization diversity combines pairs of antennas
with orthogonal polarizations. Reflected signals can undergo polarization changes depending
on the channel. Pairing two complementary polarizations, this scheme can immunize a system
from polarization mismatches that would otherwise cause signal fade.
Frequency Diversity:
In Frequency Diversity, the same information signal is transmitted and received
simultaneously on two or more independent fading carrier frequencies. Rationale behind this
technique is that frequencies separated by more than the coherence bandwidth of the channel
will be uncorrelated and will thus not experience the same fades. The probability of
simultaneous fading will be the product of the individual fading probabilities.
Time Diversity:
In time diversity, the signal representing the same information are sent over the same channel
at different times. Time diversity repeatedly transmits information at time spacings that
exceeds the coherence time of the channel. Multiple repetition of the signal will be received
with independent fading conditions, thereby providing for diversity. A modern implementation
of time diversity involves the use of RAKE receiver for spread spectrum CDMA, where the
multipath channel provides redundancy in the transmitted message.
Diversity Combining Technique
1- Selection Diversity combining technique: In selective diversity combining, the branches
having the strongest received signal will be selected. In selective diversity method, ‘n’ number
of demodulators are used and their gains can be adjusted to give mean signal to noise ratio
(SNR) for every diversity branch. Then, the antenna signals will be sampled. Finally, the best
signal that possess good signal strength will be sent to a demodulator.

2- Threshold Diversity combining technique: In this method, the ‘n’ signals are scanned in
a proper sequence and monitored to pick a signal in the sequence which is above the preset
threshold value say ‘α’.
Then, a scanning process will be initiated for the received signals. But, the demerit of this
method is that the fading level reduction is less than the other diversity techniques. In this
method, for the received signals (m), the best signal of better strength is measured by
comparing every signal with a preset threshold value ‘α’ as shown in the figure below.

3- Maximal Ratio Combining Technique: The concept of this method is that all the branch
signals [N] are combined coherently with necessary weighting coefficients for every diversity
branch signal so that the reduction of fading will be better leading to overall improvement of
system performance.
A block diagram for this method is shown in the figure below. Unlike selection diversity, the
signals are co-phased before the addition process and for this, individual receiver and phasing
circuits are a must for all the antenna elements.
In the output, signal of maximal ratio combiner will be such that the sum of individual signal
to noise ratio (SNR) values will be equal to the SNR of output signal measured.

4- Equal Gain Combiner Technique: In the equal gain combining, all the diversity branches
are coherently added with a same weighting factor. On the other hand, this scheme also co-
phases all the diversity branches and finally adds them up. As the signals are co-phased from
all branches, they provide an equal gain factor. When compared to maximal ratio combining,
the configuration of this method is simple. By applying equal gain combining, it is convenient
for the receiver to get back the signals.

Spatial Diversity and Multiplexing in MIMO

Modern wireless communication demands constant improved spectral efficiency. More
number of users is needed to be accommodated in a given bandwidth with high quality
standards for communication. Different diversity techniques are used for it. Spatial diversity
deals with multiple number of transmitting and receiving antenna at transmitter and receiver
respectively. When same signal is transmitted or received via multiple devices, spatial
diversity is formed. There are main four types of spatial diversity: SISO (Single Input Single
Output -- No diversity) SIMO (Single Input Multiple Output) MISO (Multiple Input Single
Output) MIMO (Multiple Input Multiple output) So, different number of antenna at
transmitting end (Input side) and receiving end (Output side) forms spatial diversity systems.

Spatial Multiplexing in MIMO system-

MIMO is an multiple antennae system that shows how to receive higher data rates, wider
coverage and increased reliability all without using additional frequency spectrum. For this
MIMO exploits spatial diversity as well as spatial multiplexing. To take advantage of the
additional throughput capability, MIMO utilises several sets of antennas. In many MIMO
systems, just two are used, but there is no reason why further antennas cannot be employed
and this increases the throughput. In any case for MIMO spatial multiplexing the number of
receive antennas must be equal to or greater than the number of transmit antennas.

Spatial Multiplexing in MIMO using 3×3

To take advantage of the additional throughput offered, MIMO wireless systems utilise a
matrix mathematical approach. Data streams t1, t2, . . . tn can be transmitted from antennas 1,
2, . . . n. Then there are a variety of paths that can be used with each path having different
channel properties. To enable the receiver to be able to differentiate between the different data
streams it is necessary to use. These can be represented by the antenna properties h12,
travelling from transmit antenna one to receive antenna 2 and so forth. In this way for a three
transmit, three receive antenna system a matrix can be set up:
r1 = h11 t1 + h21 t2 + h31 t3
r2 = h12 t1 + h22 t2 + h32 t3
r3 = h13 t1 + h23 t2 + h33 t3
Where r1 = signal received at antenna 1, r2 is the signal received at antenna 2 and so forth.
In matrix format this can be represented as:
[R] = [H] x [T]
To recover the transmitted data-stream at the receiver it is necessary to perform a considerable
amount of signal processing. First the MIMO system decoder must estimate the individual
channel transfer characteristic hij to determine the channel transfer matrix. Once all of this has
been estimated, then the matrix [H] has been produced and the transmitted data streams can be
reconstructed by multiplying the received vector with the inverse of the transfer matrix.
[T] = [H]-1 x [R]
This process can be likened to the solving of a set of N linear simultaneous equations to reveal
the values of N variables.

Channel Estimation Technique

CSI based channel Estimation
CSI stands for Channel State Information, which refers to the knowledge of the current
conditions of the communication channel between the transmitter and receiver in a wireless
communication system.
• Pilot signal transmission: The transmitter periodically sends known pilot signals or
training sequences over the wireless channel.
 • Channel estimation at the receiver: The receiver processes the received pilot signals
and applies channel estimation algorithms to estimate the channel impulse response or
channel coefficients for each transmit-receive antenna pair (in the case of MIMO
systems).
• CSI feedback (for FDD systems): In frequency-division duplexing (FDD) systems,
where the uplink and downlink channels are different, the receiver needs to send the
estimated CSI back to the transmitter over a feedback channel.
• CSI acquisition (for TDD systems): In time-division duplexing (TDD) systems, where
the uplink and downlink channels are reciprocal, the transmitter can estimate the CSI
based on the uplink signals from the receiver.

Blind Channel Estimation

In this type of channel estimate pilot symbols are not included at the transmitter and the
receiver. The receiver estimates the channel without the usage of pilot symbols. To efficiеntly
estimate the channel without pilots the receivеr uses a large amount of received data to
understand the channel propеrties and estimates the channel. The benefit of this method is that
the overheаd due to pilots can be reduϲed. But the computаtional complexity increases, as to
extract the signal properties large number of received data is considered which increases
computаtional time also. This channel estimation does not give good results when compаred to
other estimation methods.

Semi-Blind Channel Estimation

“Semi Blind Channel estimation” method is the combination of Blind and Training symbols
based channel estimation. In this schеme the pilots that are considered at the transmitter are
used to estimate the chаnnel at the receiver. The channel in this methοd can be evaluated by
pilots symbols or by the combination of data and pilot symbols. After the channel is estimаted
the channel co-efficients evaluated are fed bаck to the transmitter and compared with the
original data symbols. If the fed- back symbοls matches with the original symbols then the
signal at the receiver can be deteϲted properly. If the channel coefficients are in symmеtry with
the pilot symbols then the predicted symbols at the receiver are updated further without the
utilizаtion of pilot symbols. If the signal is approximated properly that signal is used as the
reference signal for further prediction of the signаls at the receiver. The main advantage of this
scheme is that the channel behavior can be tracked efficiently.