Introduction to Vocoder

Definition: 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.

A voice model is used to simulate voice. As speech contains a sequence of voiced and
unvoiced sounds, this is the basis for the operation of a voice model.
Before proceeding further, it is better to first understand what is voiced and unvoiced sounds.

Voice sounds are basically the sounds generated by vibrations of the vocal cords.

On contrary, the sound produced at the pronunciation of the letters such as ‘s’, ‘p’ or ‘f’ is
known as unvoiced sounds. Unvoiced sounds are generated by expelling air through lips
and teeth.

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. Let us now discuss the operation of
each in detail-

Voice Encoder

The figure given below shows the block diagram of voice encoder-
The frequency spectrum of the speech signal (200Hz – 3200Hz) is divided into 15 frequency
ranges by using 15 Bandpass filter(BPF) each having bandwidth range of 200Hz. The output
of BPF acts as input for the rectifier unit.

Here, the signal is rectified and filtered so as to produce a dc voltage. This generated dc
voltage is proportional to the amplitude of AC signal present at the output of the filter.

The input of the frequency discriminator is the speech signal. Frequency discriminator unit is
followed by a Low pass filter(LPF) of 20Hz. This LPF generates a dc voltage proportional
to the voice frequency. The frequency represents nothing else than the pitch of the voice.

This dc voltage also indicates whether the speech is voiced or unvoiced.

Now, the output at all the LPF’s is dc voltage which is sampled, multiplexed and A/D
converted. So, we have a digital equivalent of the speech signal at the output of the encoder.
This encoded voice signal consists of frequency component from 200Hz to 3200Hz,
information regarding the pitch of the speech and whether it is voiced or unvoiced.

Voice Decoder

The digital voice signal generated by the voice encoder is firstly decoded. Then voice
decoder using a speech synthesizer produces voice signal at its output. It generally generates
an approximate voice signal.

The block diagram of voice decoder section is shown below-

The demultiplexer and DAC section convert the received encoded signal back to its analog
form. Here, a balanced modulator(BM)-filter combination is used in correspondence to
rectifier-filter combination at the encoder. The carrier to this BM is either the output of noise
generator or pulse generator. But this depends on the position of the switch.

However, the switch position is decided by the decoder. It is so because when the voiced
signal is received, the switch connects the pulse generator output to the input of all the BM.

Similarly, when an unvoiced signal is received, the switch connects noise generator output to
the input of all the BM. But, the position of the switch totally depends on the decision of
decoder.

Only certain BM will provide the output if the received signal is voiced. This totally depends
on the frequency component of the received signal. But we can get output from all the BM if
the received signal is unvoiced. The adder will thus add up all the analog signal and produce
voice or speech output.

Speech transmission using Vocoder is helpful but it is a disadvantageous technique. This is so

because it leads to degradation in speech quality.

Types of Vocoders

Classification of Vocoders:

 Channel Vocoders
 Formant Vocoders
 Cepstrum Vocoders
 Voice-Excited Vocoders

Channel 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.
 Thesampling is done synchronously every 10 ms to30 ms.
 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.

  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
 It must control the intensities of the formants.
 Formant vocoder can reproduce speech at bit rates lower than 1200 bits/s.

Cepstrum Vocoders:

 The Cepstrum vocoder separates the excitation and vocal tract spectrum by inverse
Fourier transforming of the log magnitude spectrum to produce the Cepstrum of
the signal.
 The low frequency coefficients in the cepstrum correspond tothe vocal tract spectral
envelope
 The high frequency excitation coefficients forming a periodic pulse train at
multiples of the sampling period
 Linear filtering is performed to separate the vocal tract cepstral coefficients from the
excitation coefficients
 In the receiver, the vocal tract cepstral coefficients are Fourier transformed to
produce the vocal tract impulse response.
 By convolving this impulse response with a synthetic excitation signal (random
noise or periodic pulse train), the original speech is reconstructed

Voice-Excited Vocoder:

 Voice-excited Vocoders eliminate the need for pitch extraction and voicing
detection operation.
 This system uses a hybrid combination of PCM transmission for the low frequency
band of speech, combined with channel vocoding of higher frequency bands.
 A pitch signal is generated at the synthesizer by rectifying, band pass filtering, and
clipping the baseband signal.
 Voice excited vocoder have been designed for operation at 7200 bits/s to 9600 bits/s
Hybrid spread spectrum systems
The use of hybrid techniques attempt to capitalize upon the advantages of a particular method
while avoiding the disadvantages.
DS, on one hand, suffers heavily from the near-far effect, which makes this technique hard to
apply to systems without the ability of power control. On the other hand, its implementation
is inexpensive.
The PN code generators are easy to implement and the spreading operation itself can be
simply performed by XOR ports.
FH effectively suppresses the near-far effect and reduces the need for power control.
However, implementation ofthe (fast) hopping frequency synthesizer required for a
reasonable spreading gain is more problematic in terms of higher silicon cost and increased
power consumption.
Applying both techniques allows for combining their advantages while reducing the
disadvantages. This results in a reasonable near-far resistance at an acceptable hardware cost.
Many different hybrid combinations are possible, some of which are PN/FH, PN/TH, FH/TH
and PN/FH/TH.
While designing a hybrid system, the designer should decide FFH or SFH is to be applied.
FFH increases the cost of the frequency synthesizer but provides more protection against the
near-far effect.
SFH combines a less expensive synthesizer with a poor near-far rejection and the need for a
more powerful error-correction scheme (several symbols are lost during a hit jamming).

Multicarrier Modulation
Multicarrier modulation is a form of signal waveform that uses multiple normally close spaced
carriers in a block to carry the information.

Multicarrier modulation, MCM is a technique for transmitting data by sending the data over
multiple carriers which are normally close spaced.

Multicarrier modulation has several advantages including resilience to interference, resilience

to narrow band fading and multipath effects.

As a result, multicarrier modulation techniques are widely used for data transmission as it is
able to provide an effective signal waveform which is spectrally efficient and resilient to the
real world environment.
One form of multicarrier modulation is OFDM
Multicarrier modulation basics

Multicarrier modulation operates by dividing the data stream to be transmitted into a number
of lower data rate data streams. Each of the lower data rate streams is then used to modulate
an individual carrier.

When the overall transmission is received, the receiver has to then re-assembles the overall
data stream from those received on the individual carriers.

It is possible to use a variety of different techniques for multicarrier transmissions. Each form
of MCM has its own advantages and can be sued in different applications.

Development of MCM

The history of multicarrier modulation can said to have been started by military users. The
first MCM were military HF radio links in the late 1950s and early 1960s. Here several
channels were sued to overcome the effects of fading.

Originally the concept of MCM required the use of several channels that were separated from
each other by the use of steep sided filters of they were close spaced. In this way, interference
from the different channels could be eliminated.

However, multicarrier modulation systems first became widely used with the introduction of
broadcasting systems such as DAB digital radio and DVB, Digital Video Broadcasting which
used OFDM, orthogonal frequency division multiplexing. OFDM used processing power
within the receiver and orthogonality between the carriers to ensure no interference was
present.

Diversity techniques and spatial multiplexing

The wireless communication environment is very hostile. The signal transmitted over a
wireless communication link is susceptible to fading (severe fluctuations in signal level), co-
channel interference, dispersion effects in time and frequency, path loss effect, etc. On top of
these woes, the limited availability of bandwidth poses a significant challenge to a designer in
designing a system that provides higher spectral efficiency and higher quality of link
availability at low cost.

Multiple antenna systems are the current trend in many of the wireless technologies that is
essential for their performance (you will even see it in your future hard disk drives as Two
Dimensional Magnetic Recording (TDMR) technology). Multiple Input Multiple Output
systems (MIMO) improve the spectral efficiency and offers high quality links when
compared to traditional Single Input Single Output (SISO) systems. Many theoretical
studies and communication system design experimentations on MIMO systems demonstrated
a great improvement in performance of such systems.
Techniques for improving performance

Spatial Multiplexing techniques , example – BLAST yields increased data rates in wireless
communication links. Fading can be mitigated by employing receiver and transmit diversity
(Alamouti Scheme , Tarokh et. al) , there by improving the reliability of the transmission
link. Improved coverage can be effected by employing coherent combining techniques –
which gives array gain and increases the signal to noise ratio of the system. The goals of a
wireless communication system are conflicting and a clear balance of the goals is needed for
maximizing the performance of the system.
The following text concentrates on two of the above mentioned techniques – diversity and
spatial multiplexing.

MIMO classification with respect to antenna configuration

In MIMO jargon, communication systems are broadly categorized into four categories with
respect to number of antennas in the transmitter and the receiver, as listed below.

● SISO – Single Input Single Output system – 1 Tx antenna , 1 Rx antenna

● SIMO – Single Input Multiple Output system – 1 Tx antenna, Rx antennas ( )
● MISO – Multiple Input Single Output system – Tx antennas, 1 Rx antenna ( )
● MIMO – Multiple Input Multiple Output system – Tx antennas, Rx antennas (
)
Diversity and Spatial-Multiplexing

Apart from the antenna configurations, there are two flavors of MIMO with respect to how
data is transmitted across the given channel. Existence of multiple antennas in a system,
means existence of different propagation paths. Aiming at improving the reliability of the
system, we may choose to send same data across the different propagation (spatial) paths.
This is called spatial diversity or simply diversity. Aiming at improving the data rate of the
system, we may choose to place different portions of the data on different propagation paths
(spatial-multiplexing). These two systems are listed below.
● MIMO – implemented using diversity techniques – provides diversity gain – Aimed
at improving the reliability
● MIMO – implemented using spatial-multiplexing techniques – provides degrees of
freedom or multiplexing gain – Aimed at improving the data rate of the system.
Diversity:

As indicated, two fundamental resources available for a MIMO system are diversity and
degrees of freedom. Let’s see what these terms mean

In diversity techniques, same information is sent across independent fading channels to

combat fading. When multiple copies of the same data are sent across independently fading
channels, the amount of fade suffered by each copy of the data will be different. This
guarantees that at-least one of the copy will suffer less fading compared to rest of the copies.
Thus, the chance of properly receiving the transmitted data increases. In effect, this improves
the reliability of the entire system. This also reduces the co-channel interference significantly.
This technique is referred as inducing a “spatial diversity” in the communication system.
Consider a SISO system where a data stream [1, 0, 1, 1, 1] is transmitted through a channel
with deep fades. Due to the variations in the channel quality, the data stream may get lost or
severely corrupted that the receiver cannot recover.The solution to combat the rapid channel
variations is to add independent fading channel by increasing the number of transmitter
antennas or receiver antennas or the both.
The SISO antenna configuration will not provide any diversity as there is no parallel link.
Thus the diversity is indicated as (0).

Single Input Single Output (SISO) system – no diversity

Instead of transmitting with single antenna and receiving with single antenna (as in SISO),
let’s increase the number of receiving antennas by one more count. In this Single Input
Multiple Output (SIMO) antenna system, two copies of the same data are put on two different
channels having independent fading characteristics. Even if one of the link fails to deliver the
data, the chances of proper delivery of the data across the other link is very high. Thus,
additional fading channels increase the reliability of the overall transmission – this
improvement in reliability translates into performance improvement – measured as diversity
gain. For a system with transmitter antennas and receiver antennas, the maximum
number of diversity paths is . In the following configuration, the total number of
diversity path created is .
 Single Input Multiple Output Channel with diversity
In this way, more diversity paths can be created by adding multiple antennas at transmitter or
receiver or both. The following figure illustrates a MIMO system with number of
diversity paths equal to .
 MIMO system with diversity
Spatial Multiplexing:

In spatial multiplexing, each spatial channel carries independent information, there by

increasing the data rate of the system. This can be compared to Orthogonal Frequency
Division Multiplexing (OFDM) technique, where, different frequency subchannels carry
different parts of the modulated data. But in spatial multiplexing, if the scattering by the
environment is rich enough, several independent subchannels are created in the same
allocated bandwidth. Thus the multiplexing gain comes at no additional cost on bandwidth or
power. The multiplexing gain is also referred as degrees of freedom with reference to signal
space constellation [2]. The number of degrees of freedom in a multiple antenna
configuration is equal to , where is the number of transmit antennas and
is the number of receive antennas. The degrees of freedom in a MIMO configuration
governs the overall capacity of the system.
Following figure illustrates the difference between diversity and spatial multiplexing. In the
transmit diversity technique shown below, same information is sent across different
independent spatial channels by placing them on three different transmit antennas. Here, the
diversity gain is 3 (assuming MISO configuration) and multiplexing gain is 0.
In the spatial multiplexing technique, each bit of the data stream (independent information) is
multiplexed on three different spatial channels thereby increasing the data rate. Here, the
diversity gain is 0 and the multiplexing gain is 3 (assuming MIMO configuration).
 MIMO system – Diversity Vs spatial multiplexing
Exploiting diversity and degree of freedom:

As seen above, in a MIMO system with rich scattering environment (independent

uncorrelated spatial paths), space time codes are designed to exploit following two resources.
SPREAD SPECTRUM MODULATION

Introduction:

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.

Problem of radio transmission

Narrow band can be wiped out due to interference. To disrupt the communication,
the adversary needs to do two things,
(a) to detect that a transmission is taking place and
(b) to transmit a jamming signal which is designed to confuse the receiver.

Solution

A spread spectrum system is therefore designed to make these tasks as difficult

as possible.
Firstly, the transmitted signal should be difficult to detect by an adversary/jammer,i.e.,
the signal should have a low probability of intercept (LPI).
Secondly, the signal should be difficult to disturb with a jamming signal, i.e., the
transmitted signal should possess an anti-jamming (AJ) property

Remedy

spread the narrow band signal into a broad band to protect against interference

In a digital communication system the primary resources are Bandwidth and

Power. The study of digital communication system deals with efficient utilization of
these two resources, but there are situations where it is necessary to sacrifice their
efficient utilization in order to meet certain other design objectives.

For example to provide a form of secure communication (i.e. the transmitted

signal is not easily detected or recognized by unwanted listeners) the bandwidth of the
transmitted signal is increased in excess of the minimum bandwidth necessary to
transmit it. This requirement is catered by a technique known as “Spread Spectrum
Modulation”.

The primary advantage of a Spread – Spectrum communication system is its ability to

reject ‘Interference’ whether it be the unintentional or the intentional interference.
 The definition of Spread – Spectrum modulation may be stated in two parts.

1. Spread Spectrum is a mean of transmission in which the data sequence

occupies a BW (Bandwidth) in excess of the minimum BW necessary to transmit it.

2. The Spectrum Spreading is accomplished before transmission through the use of a code
that is independent of the data sequence. The Same code is used in the receiver to
despread the received signal so that the original data sequence may be recovered.

Fig. Block diagram for spread spectrum communication

Fig: Spread spectrum technique.

b(t) = Data Sequence to be transmitted (Narrow Band);

c(t) = Wide Band code ;
s(t) = c(t) * b(t) – (wide Band)
 Fig: Spectrum of signal before & after spreading

PSUEDO-NOISE SEQUENCE:

Generation of PN sequence:

Fig: Maximum-length sequence generator for n=3

A feedback shift register is said to be Linear when the feedback logic consists of
entirely mod-2-address (Ex-or gates). In such a case, the zero state is not permitted.
The period of a PN sequence produced by a linear feedback shift register with ‘n’ flip
flops cannot exceed 2n-1.
When the period is exactly 2 n-1, the PN sequence is called a ‘maximum length
sequence’ or ‘m-sequence’.
Example1: Consider the linear feedback shift register shown in above figure

Involve three flip-flops. The input so is equal to the mod-2 sum of S1 and S3. If
the initial state of the shift register is 100. Then the succession of states will be as
follows.
100,110,011,011,101,010,001,100 . . . . . .

The output sequence (output S3) is therefore. 00111010 . . . . . Which repeats itself with
period 23–1 = 7 (n=3). Maximal length codes are commonly used PN codes In binary
shift register, the maximum length sequence is

N = 2m-1

chips, where m is the number of stages of flip-flops in the shift register.

At each clock pulse

• Contents of register shifts one bit right.

• Contents of required stages are modulo 2 added and fed back to input.

Fig: Initial stages of Shift registers 1000

Let initial status of shift register be 1000

Properties of PN Sequence

Randomness of PN sequence is tested by following properties

1. Balance property

2. Run length property

3. Autocorrelation property

1. Balance property

In each Period of the sequence , number of binary ones differ from binary zeros by
at most one digit.
Consider output of shift register 0 0 0 1 0 0 1 1 0 1 0 1 1 1 1
Seven zeros and eight ones -meets balance condition.

2. Run length property

Among the runs of ones and zeros in each period, it is desirable that about one
half the runs of each type are of length 1, one- fourth are of length 2 and one-eighth
are of length 3 and so-on.
 Consider output of shift register

Number of runs =8

0 0 0 1 0 0 1 1 0 1 0 1 1 1 1

3 1 2 2 1 1 1 4

3. Auto correlation property

Auto correlation function of a maximal length sequence is periodic and binary

valued. Autocorrelation sequence of binary sequence in polar format is given by

𝑁
1
𝑅𝑐 𝑘 = 𝑐𝑛 𝑐𝑛−𝑘
𝑁
𝑛=1

Where N is length or the period of the sequence, k is the lag of auto correlation function.
1 𝑖𝑓 𝑘 = 1𝑁
𝑅𝑐 𝑘 = 1
− 𝑖𝑓 𝑘 ≠ 1𝑁
𝑁

Where 1 is any Integer. We can also state the auto correlation function is
1
𝑅𝑐 𝑘 =
𝑁

{ No. of agreements – No. of disagreements in comparison of one full period }

Consider output of shift register for l=1

1 1
𝑅𝑐 𝑘 = 7−8 = −
15 15
Yields PN autocorrelation as

Range of PN Sequence Lengths

Length 0f Shift Register, m PN Sequence Length,

7 127
8 255
9 511
10 1023
11 2047
12 4095
13 8191
17 131071
19 524287

Notion of Spread Spectrum:

An important attribute of Spread Spectrum modulation is that it can provide protection

against externally generated interfacing signals with finite power. Protection against jamming
(interfacing) waveforms is provided by purposely making the information – bearing signal
occupy a BW far in excess of the minimum BW necessary to transmit it. This has the effect
of making the transmitted signal a noise like appearance so as to blend into the background.
Therefore Spread Spectrum is a method of ‘camouflaging’ the information – bearing signal.
 Let { bK} denotes a binary data sequence.

{ cK } denotes a PN sequence.

b(t) and c(t) denotes their NRZ polar representation respectively.

The desired modulation is achieved by applying the data signal b(t) and PN signal
c(t) to a product modulator or multiplier. If the message signal b(t) is narrowband and
the PN sequence signal c(t) is wide band, the product signal m(t) is also wide band. The
PN sequence performs the role of a ‘Spreading Code”.
For base band transmission, the product signal m(t) represents the transmitted
signal. Therefore m(t) = c(t).b(t)
The received signal r(t) consists of the transmitted signal m(t) plus an additive
interference noise n(t), Hence
r(t) = m(t) + n(t)

= c(t).b(t) + n(t)
 To recover the original message signal b(t), the received signal r(t) is applied to a
demodulator that consists of a multiplier followed by an integrator and a decision device.
The multiplier is supplied with a locally generated PN sequence that is exact replica of that
used in the transmitter. The multiplier output is given by

Z(t) = r(t).c(t)

= [b(t) * c(t) + n(t)] c(t) = c2(t).b(t) + c(t).n(t)

The data signal b(t) is multiplied twice by the PN signal c(t), where as unwanted
signal n(t) is multiplied only once. But c2(t) = 1, hence the above equation reduces to

Z(t) = b(t) + c(t).n(t)

Now the data component b(t) is narrowband, where as the spurious component
c(t)n(t) is wide band. Hence by applying the multiplier output to a base band (low pass)filter
most of the power in the spurious component c(t)n(t) is filtered out. Thus the effect of the
interference n(t) is thus significantly reduced at the receiver output.

The integration is carried out for the bit interval 0 ≤ t ≤ Tb to provide the sample value
V. Finally, a decision is made by the receiver.
 If V > Threshold Value ‘0’, say binary symbol ‘1’ If V < Threshold Value ‘0’, say
binary symbol ‘0’

Direct – Sequence Spread Spectrum with coherent binary Phase shift

Keying:-
 To provide band pass transmission, the base band data sequence is multiplied by a
Carrier by means of shift keying. Normally binary phase shift keying (PSK) is used because
of its advantages. The transmitter first converts the incoming binary data sequence {b k}
into an NRZ waveform b(t), which is followed by two stages of modulation.
The first stage consists of a multiplier with data signal b(t) and the PN signal c(t)as
inputs. The output of multiplier is m(t) is a wideband signal. Thus a narrow – band data
sequence is transformed into a noise like wide band signal.
The second stage consists of a binary Phase Shift Keying (PSK) modulator.
Which converts base band signal m(t) into band pass signal x(t). The transmitted signal x(t) is
thus a direct – sequence spread binary PSK signal. The phase modulation θ(t) of
x(t) has one of the two values ‘0’ and ‘π’ (180 o) depending upon the polarity of the
message signal b(t) and PN signal c(t) at time t.
Polarity of PN & Polarity of PN signal both +, + or - - Phase ‘0’

Polarity of PN & Polarity of PN signal both +, - or - + Phase ‘π’

The receiver consists of two stages of demodulation.

In the first stage the received signal y(t) and a locally generated carrier are
applied to a coherent detector (a product modulator followed by a low pass filter), Which
converts band pass signal into base band signal.
The second stage of demodulation performs Spectrum despreading by
multiplying the output of low-pass filter by a locally generated replica of the PN signal
c(t), followed by integration over a bit interval Tb and finally a decision device is used to
get binary sequence.
 Fig : Direct Sequence Spread Spectrum Example

Fig : Direct Sequence Spread Spectrum Using BPSK Example

Signal Space Dimensionality and Processing Gain

 Fundamental issue in SS systems is how much protection spreading can

provide against interference.
 SS technique distribute low dimensional signal into large dimensional signal
space (hide the signal).
 Jammer has only one option; to jam the entire space with fixed total power or
to jam portion of signal space with large power.

Consider set of orthonormal basis functions;

2
𝜑𝑘 𝑡 = cos 2𝛱 𝑓𝑐 𝑡 𝑘𝑇𝑐 ≤ 𝑡 ≤ 𝑡 + 1 𝑇𝑐
𝑇𝑐
0 𝑂𝑡𝑕𝑒𝑟𝑤𝑖𝑠𝑒

2
𝜑𝑘 𝑡 = sin 2𝛱 𝑓𝑐 𝑡 𝑘𝑇𝑐 ≤ 𝑡 ≤ 𝑡 + 1 𝑇𝑐
𝑇𝑐
0 𝑂𝑡𝑕𝑒𝑟𝑤𝑖𝑠𝑒 = 0,1 … … … … … 𝑁 − 1

Where Tc is chip duration, N is number of chips per bit.

Transmitted signal x(t) for the interval of an information bit is

𝑥 𝑡 = 𝑐(𝑡)𝑠(𝑡)

2
𝜑𝑘 𝑡 = ± c(t)cos 2𝛱 𝑓𝑐 𝑡
𝑇𝑐
N−1
𝐸𝑏
𝜑𝑘 𝑡 = ± ck φk (t) 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑁
k=0
where, Eb is signal energy per bit.

PN Code sequence { c0, c1, ……cN-1} with ck= + 1, Transmitted signal x(t) is
therefore N dimensional and requires N orthonormal functions to represent it. j(t)
represent interfering signal (jammer). As said jammer tries to places all its available
energy in exactly same N dimension signal space. But jammer has no knowledge
of signal phase. Hence tries to place equal energy in two phase coordinates that is
cosine and sine. As per that jammer can be represented as
 𝑁−1 𝑁−1

𝑗 𝑡 = 𝑗𝑘 𝜑𝑘 𝑡 + 𝑗𝑘 𝜑𝑘 𝑡 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑘=0 𝑘=0

Where

𝑇𝑏
𝑗𝑘 = 𝑗 𝑡 𝜑𝑘 𝑡 𝑘 = 0,1, … … 𝑁 − 1
0

𝑇𝑏
𝑗𝑘 = 𝑗 𝑡 𝜑𝑘 𝑡 𝑘 = 0,1, … … 𝑁 − 1
0
Thus j(t) is 2N dimensional, twice the dimension as that of x(t).

Average interference power of j(t)

𝑇𝑏 𝑁−1 𝑁−1
1 1 1
𝐽= 𝑗 2 𝑡 𝑑𝑡 = 𝑗𝑘 2 + 𝑗𝑘 2
𝑇𝑏 0 𝑇𝑏 𝑇𝑏
𝑘=0 𝑘=0

as jammer places equal energy in two phase coordinates , hence

𝑁−1 𝑁−1

𝑗𝑘 2 = 𝑗𝑘 2
𝑘=0 𝑘=0

𝑁−1
2
𝐽= 𝑗𝑘 2
𝑇𝑏
𝑘=0
To evaluate system performance we calculate SNR at input and output of DS/BPSK
receiver. The coherent receiver input is u(t) =s(t) + c(t)j(t) and using this u(t), output at
coherent receiver

Tb
2
𝑉= u(t) cos 2𝛱 𝑓𝑐 𝑡 𝑑𝑡 = 𝑉𝑠 + 𝑉𝑐𝑗
𝑇𝑏 0

Where vs is despread component of BPSK and vcj of spread interference.

Tb
2
𝑉𝑠 = s(t) cos 2𝛱 𝑓𝑐 𝑡 𝑑𝑡
𝑇𝑏 0
 Tb
2
𝑉𝑐𝑗 = c t j(t) cos 2𝛱 𝑓𝑐 𝑡 𝑑𝑡
𝑇𝑏 0

Consider despread BPSK signal s(t)

2𝐸𝑏
𝑠(𝑡) = ± cos 2𝛱 𝑓𝑐 𝑡 𝑑𝑡 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑇𝑏

Where + sign is for symbol 1

- sign for symbol 0.

If carrier frequency is integer multiple of 1 / Tb , we have 𝑉𝑠 = ± 𝐸𝑏

Consider spread interference component vcj, here c(t) is considered in sequence form
{ c0, c1, ……cN-1}

N−1 𝑇𝑏 N−1
𝑇𝑐 𝑇𝑐
𝑉𝑐𝑗 = Ck 𝑗 𝑡 𝜑𝑘 𝑡 𝑑𝑡 = Ck 𝑗𝑘
𝑇𝑏 0 𝑇𝑏
k=0 k=0

With Ck treated as independent identical random variables with both symbols having
equal probabilities
1
𝑃 𝐶𝑘 = 1 = 𝑃 𝐶𝑘 = −1 =
2
Expected value of Random variable vcj is zero, for fixed k we have

1 1
𝐸 𝑐𝑘 𝑗𝑘 |𝑗𝑘 = 𝑗𝑘 𝑃 𝐶𝑘 = 1 − 𝑝 𝐶𝑘 = −1 = 𝑗𝑘 − 𝑗𝑘 = 0
2 2

And Variance

𝑁−1
1 𝐽𝑇𝑐
𝑉𝑎𝑟 𝑉𝑐𝑗 |𝑗 = 𝑗𝑘 2 =
𝑁 2
𝑘=0

Spread factor N = Tb/Tc

Output signal to noise ratio is

2𝐸𝑏
(𝑆𝑁𝑅)𝑜 =
𝐽 𝑇𝑐
 The average signal power at receiver input is Eb/Tb hence input SNR
𝐸𝑏 𝑇𝑏
(𝑆𝑁𝑅)𝑖 =
𝐽
2𝑇𝑏
(𝑆𝑁𝑅)0 = (𝑆𝑁𝑅)𝑖
𝑇𝑐

Expressing SNR in decibels

10𝑙𝑜𝑔10 (𝑆𝑁𝑅)0 = 10𝑙𝑜𝑔10 (𝑆𝑁𝑅)𝑖 + 3 + 10𝑙𝑜𝑔10 𝑃𝐺 , 𝑑𝐵

𝑇𝑏
Where 𝑃𝐺 = 𝑇𝑐
3db term on right side accounts for gain in SNR due to coherent detection. Last term
accounts for gain in SNR by use of spread spectrum. PG is called Processing Gain.

1
1. Bit rate of binary data entering the transmitter input is 𝑅𝑏 = 𝑇𝑏
 2. The bandwidth of PN sequence c(t) , of main lobe is Wc
1
𝑊𝐶 =
𝑇𝑐
𝑊𝑐
𝑃𝐺 =
𝑅𝑏
Probability of error

To calculate probability of error, we consider output component v of coherent

detector as sample value of random variable

𝑉 = ± 𝐸𝑏 + 𝑉𝑐𝑗

Eb is signal energy per bit and Vcj is noise component

Decision rule is, if detector output exceeds a threshold of zero volts; received bit is
symbol 1 else decision is favored for zero.

• Average probability of error Pe is nothing but conditional probability which

depends on random variable Vcj.
• As a result receiver makes decision in favor of symbol 1 when symbol 0
transmitted and vice versa
• Random variable Vcj is sum of N such random variables. Hence for
Large N it can assume Gaussian distribution .
• As mean and variance has already been discussed , zero mean and variance
JTc/2

Probability of error can be calculated from simple formula for DS/BPSK system

1 𝐸𝑏
𝑃𝑒 ≅ 𝑒𝑟𝑓𝑐
2 𝐽𝑇𝑐
Antijam Characteristics

Consider error probability of BPSK

1 𝐸𝑏
𝑃𝑒 = 𝑒𝑟𝑓𝑐
2 𝑁0

Comparing both probabilities;

𝑁0 𝐽𝑇𝑐
=
2 2
Since bit energy Eb =PTb , P= average signal power.

We can express bit energy to noise density ratio as

 𝐸𝑏 𝑇𝑏 𝑃
=
𝑁0 𝑇𝑐 𝐽
Or
𝐽 𝑃𝐺
=
𝑃 𝐸𝑏 𝑁0

The ratio J/P is termed jamming margin. Jamming Margin is expressed in decibels as

𝐸𝑏
𝑗𝑎𝑚𝑚𝑖𝑛𝑔 𝑚𝑎𝑟𝑔𝑖𝑛 𝑑𝐵 = 𝑃𝑟𝑜𝑐𝑒𝑠𝑠𝑖𝑛𝑔 𝑔𝑎𝑖𝑛 𝑑𝐵 − 10 𝑙𝑜𝑔10
𝑁0 𝑚𝑖𝑛

𝐸𝑏
Where is minimum bit energy to noise ration needed to support a prescribed
𝑁0
average probability of error.

Example1

A pseudo random sequence is generated using a feed back shift register of

length m=4. The chip rate is 107 chips per second. Find the following
a) PN sequence length b) Chip duration of PN sequence c) PN sequence
period

Solution

a) Length of PN sequence N = 2m-1= 24-1 =15

b) Chip duration Tc = 1/chip rate =1/107 = 0.1µ sec
c) PN sequence period T = NTc
= 15 x 0.1µ sec = 1.5µ sec

Example2

A direct sequence spread binary phase shift keying system uses a feedback
shift register of length 19 for the generation of PN sequence. Calculate the
processing gain of the system.

Solution

Given length of shift register = m =19

Therefore length of PN sequence N = 2m - 1
= 219 - 1
Processing gain PG = Tb/Tc =N in db =10log10N = 10 log10 (219) = 57db
Example3

A Spread spectrum communication system has the following parameters.

Information bit duration Tb = 1.024 msecs and PN chip duration of 1µsecs. The
average probability of error of system is not to exceed 10 -5. calculate a) Length of
shift register b) Processing gain c) jamming margin

Solution

Processing gain PG =N= Tb/Tc =1024

corresponding length of shift register m = 10
In case of coherent BPSK For Probability of error 10-5. [Referring to error function table]
Eb/N0 =10.8
Therefore jamming margin
𝐸𝑏
𝑗𝑎𝑚𝑚𝑖𝑛𝑔 𝑚𝑎𝑟𝑔𝑖𝑛 𝑑𝐵 = 𝑃𝑟𝑜𝑐𝑒𝑠𝑠𝑖𝑛𝑔 𝑔𝑎𝑖𝑛 𝑑𝐵 − 10 𝑙𝑜𝑔10
𝑁0 𝑚𝑖𝑛

𝐸𝑏
𝑗𝑎𝑚𝑚𝑖𝑛𝑔 𝑚𝑎𝑟𝑔𝑖𝑛 𝑑𝐵 = 10 𝑙𝑜𝑔10 𝑃𝐺𝑑𝐵 − 10 𝑙𝑜𝑔10
𝑁0 𝑚𝑖𝑛

𝑗𝑎𝑚𝑚𝑖𝑛𝑔 𝑚𝑎𝑟𝑔𝑖𝑛 𝑑𝐵 = 10 𝑙𝑜𝑔10 1024 − 10 𝑙𝑜𝑔10 10.8

𝑗𝑎𝑚𝑚𝑖𝑛𝑔 𝑚𝑎𝑟𝑔𝑖𝑛 𝑑𝐵 = 30.10 − 10.33 = 19.8 𝑑𝐵

Frequency – Hop Spread Spectrum:

In a frequency – hop Spread – Spectrum technique, the spectrum of data

modulated carrier is widened by changing the carrier frequency in a pseudo – random
manner. The type of spread – spectrum in which the carrier hops randomly form one
frequency to another is called Frequency – Hop (FH) Spread Spectrum.

Since frequency hopping does not covers the entire spread spectrum
instantaneously. We are led to consider the rate at which the hop occurs. 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).
 Slow frequency hopping:-

Fig Shows the block diagram of an FH / MFSK transmitter, which involves

frequency modulation followed by mixing.
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.

Fig :- Frequency hop spread transmitter

 Fig :- Frequency hop spread receiver

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.

An individual FH / MFSK tone of shortest duration is referred as a chip. The chip

rate Rc for an FH / MFSK system is defined by
Rc = Max(Rh,Rs)

Where Rh is the hop rate and Rs is Symbol Rate

In a slow rate frequency hopping multiple symbols are transmitted per hop.
Hence each symbol of a slow FH / MFSK signal is a chip. The bit rate Rb of the
incoming binary data. The symbol rate Rs of the MFSK signal, the chip rate Rc and the
hop rate Rn are related by
Rc = Rs = Rb /k ≥ Rh

where k= log2M

Fast frequency hopping:-

A fast FH / MFSK system differs from a slow FH / MFSK system in that

there are multiple hops per m-ary symbol. Hence in a fast FH / MFSK system each hop
is a chip.
 Fast Frequency Hopping Slow Frequency Hopping

Several frequency hops Several modulation symbols per

Per modulation hop

Shortest uninterrupted waveform Shortest uninterrupted waveform in

in the system is that of hop the system is that of data symbol

Chip duration =hop duration Chip duration=bit duration.

The following figure illustrates the variation of the frequency of a slow FH/MFSK
signal with time for one complete period of the PN sequence. The period of the PN
sequence is 2 4-1 = 15.
The FH/MFSK signal has the following parameters:
Number of bits per MFSK symbol K = 2. Number of MFSK tones M = 2 K = 4
Length of PN segment per hop k = 3; Total number of frequency hops 2k = 8

Fig. Slow frequency hopping

 The following figure illustrates the variation of the transmitted frequency of a fast
FH/MFSK signal with time.
The signal has the following parameters:
Number of bits per MFSK symbol K = 2. Number of MFSK tones M = 2 K = 4
Length of PN segment per hop k = 3; Total number of frequency hops 2k = 8

Fig. Fast frequency hopping

FHSS Performance Considerations:

• Typically large number of frequencies used

– Improved resistance to jamming

Code Division Multiple Access (CDMA):

• Multiplexing Technique used with spread spectrum

• Start with data signal rate D
– Called bit data rate
• Break each bit into k chips according to fixed pattern specific to each user
– User’s code
• New channel has chip data rate kD chips per second
• E.g. k=6, three users (A,B,C) communicating with base receiver R
• Code for A = <1,-1,-1,1,-1,1>
• Code for B = <1,1,-1,-1,1,1>
• Code for C = <1,1,-1,1,1,-1>
CDMA Example:

•Consider A communicating with base

•Base knows A’s code
•Assume communication already synchronized
•A wants to send a 1
– Send chip pattern <1,-1,-1,1,-1,1>
• A’s code
• A wants to send 0
– Send chip[ pattern <-1,1,1,-1,1,-1>
• Complement of A’s code
• Decoder ignores other sources when using A’s code to decode
– Orthogonal codes
–
CDMA for DSSS:
• n users each using different orthogonal PN sequence
• Modulate each users data stream
– Using BPSK
• Multiply by spreading code of user
CDMA in a DSSS Environment:
 49

Combining Techniques of Diversity

If we want to get benefit from diversity technique then we must need to combine some
diversity technique to get advantage. Therefore, diversity combining concepts are described
in this chapter, in section 5.2, we described Maximal-ratio Combining (MRC), in section 5.3,
we described Equal-gain Combining (EGC) and in section 4.4, we described Selection
Combining (SC). Above three combining systems are our main focuses and applied in
experiments to improve performance in wireless communication systems. Block diagram of
combining methods are drawn in this chapter. Switched combining method, Periodic
combining method, Phase-sweeping methods are described shortly and their diagrams also
drawn in this chapter.

1 Concepts of Diversity Combining Techniques:

It is important to combine the uncorrelated faded signals which were obtained from the
diversity branches to get proper diversity benefit. The combing system should be in such a
manner that improves the performance of the communication system. Diversity combing also
increases the signal-to-noise ratio (SNR) or the power of received signal. Mainly, the
combining should be applied in reception; however it is also possible to apply in transmission.
There are many diversity combining methods available but only three of them are going to be
discussed here.

 Maximal ratio combining (MRC)

 Equal gain combining (EGC)
 Selection combining (SC)

The combining processes which use to combine multiple diversity branches in the reception,
has two classes such as post-detection combing and pre-detection combining. The signals
from diversity branches are combined coherently before detection in pre-detection combining.
However, signals are detected individually before combining in post-detection. The
performance of communication system is the same for both combining techniques for
coherent detection. However, the performance of communication system is better by using
pre-detection combining for non-coherent detection. It does mean that there is no effect in
performance by the type of combining procedure for the coherent modulation case. The post-
detection combining is not complex in non-coherent detection, results very common in use.
There is a difference in system performance when used pre-detection combining and post-
detection combining for non-coherent detection such as frequency modulation (FM)
discriminator or differential detection schemes. Moreover, the terms pre-detection and post-
 50

detection are also indicates the time of combining means when the combining is performed,
before or after the hard decision.

Squire-law non-coherent combining is employed frequently in diversity reception when non-

coherent modulation methods are used. The demodulator outputs of all diversity branches are
squired and summed to form a decision variable when used squire-law pre-detection
combining. The system performance is decreased in non-coherent combining comparing to
coherent combining and the degradation is called combining loss.

2 Maximal Ratio Combining (MRC):

This is a very useful combining process to combat channel fading. This is the best combining
process which achieves the best performance improvement comparing to other methods. The
MRC is a commonly used combining method to improve performance in a noise limited
communication systems where the AWGN and the fading are independent amongst the
diversity branches. But the MRC employment needs summing circuits, weighting and co-
phasing. In the MRC combining technique, the signals from different diversity branches are
co-phased and weighted before summing or combining. The weights have to be chosen as
proportional to the respective signals level for maximizing the combined carrier-to-noise ratio
(CNR). The applied weighting to the diversity branches has to be adjusted according to the
SNR. For maximizing the SNR and minimizing the probability of error at the output
combiner, signals of diversity branch is weighted before making sum with others by a
factor, . Here is noise variance of diversity branch and is the complex

conjugate of channel gain [1]. As a result the phase-shifts are compensated in the diversity
channels and the signals coming from strong diversity branches which has low level noise, are
weighted more comparing to the signals from the weak branches with high level of noise. The
term in weighting can be neglected conditioning that has equal value for all d. Then
the realization of the combiner needs the estimation of gains in complex channel and it does
not need any estimation of the power of noise.

It is feasible to employ MRC in transmission process of transmit diversity. But in this case the
transmitter should get proper feedback information about the sub-channels state between
single receive antenna and multiple transmit antennas. However, it is not feasible to weight
transmissions from multiple antennas optimally for every receiving antenna, in a combined
transmit-receive diversity channel. Moreover, if interference is limited in a communication
system, then there is a scheme which combines the diversity branches in order to maximize
the signal-to-interference-plus-noise ratio may allow much better performance than MRC
provides. The assumption is valid for spatially white Gaussian noise if we can observe noise
power at the receiver where just thermal noise is accounted. If we use the same type antenna
elements then the thermal noise power is uncorrelated and equal for each branch.
 51

Ant

 Rx
Ant
Det
Det

Rx

Picture 12: Maximal-ratio combining (MRC)

3: Equal-gain Combining (EGC):

MRC is the most ideal diversity combining but the scheme requires very expensive design at
receiver circuit to adjust the gain in every branch. It needs an appropriate tracking for the
complex fading, which very difficult to achieve practically. However, by using a simple phase
lock summing circuit, it is very easy to implement an equal gain combining. The EGC is
similar to MRC with an exception to omit the weighting circuits. The performance
improvement is little bit lower in EGC than MRC because there is a chance to combine the
signals with interference and noise, with the signals in high quality which are interference and
noise free. EGC’s normal procedure is coherently combined the individual signal branch but it
non-coherently combine some noise components according to following figure:

Q C
Q A
A
B
B
t
t
C

(1) Non-coherent branch signal. (2) Coherent combining

Picture 13: Concept of coherent combining, complex signal diagram illustration.

The EGC can employ in the reception of diversity with coherent modulation. The envelope
gains of diversity channels are neglected in EGC and the diversity branches are combined
 52

here with equal weights but conjugate phase. The structure of equal-gain combining (EGC) is
as following since there is no envelope gain estimation of the channel.

 Rx

Det
Det
Rx

Picture 14: Equal gain combining (EGC)

4 Selection Combining (SC):

MRC and EGC are not suitable for very high frequency (VHF), ultra high frequency (UHF) or
mobile radio applications. Realization of a co-phasing circuit with precise and stable tracking
performance is not easy in a frequently changing, multipath fading and random-phase
environment. SC is more suitable comparing to MRC and EGC in mobile radio application
because of simple implementation procedure. In SC, the diversity branch which has the
highest signal level has to be selected. Therefore, the main algorithm of this method is on the
base of principle to select the signal amongst the all signals at the receiver end. If there is even
a fast multipath fading environment, the stable operation easily can be achieved. It is
experimentally proved that the performance improvement achieved by the selection
combining is just little lower than performance improved achieved by an ideal MRC. As a
result the SC is the most used diversity technique in wireless communication.

The general form of selection combining is to monitor all the diversity branches and select the
best one (the one which has the highest SNR) for detection. Therefore we can say that SC is
not a combining method but a selection procedure at the available diversity. However,
measuring SNR is quite difficult because the system has to select it in a very short time. But
selecting the branch with the highest SNR is similar to select the branch with highest received
power when average power of noise is the same on each branch. Therefore, it is practical to
select the branch which has the largest signal composition, noise and interference. If there is
an availability of feedback information about the channel state of the diversity branch the
selection combining also can be used in transmission.
 53

Rx

Comp Det

Rx

Figure: Selection combining (SC)

5 Switched Combining (SWC):

It is impractical to monitor the all diversity branches in selection combining. In addition, if we
want to monitor the signals continuously then we need the same number of receivers and
branches. Therefore, the form of switched combining is used to implement selection
combining. According to the figure (a), switching from branch to branch occurs when the
signal level falls under threshold. The value of threshold is fixed under a small area but the
value is not the best necessarily over the total service area. As a result the threshold needs to
be set frequently according to the movement of vehicle fig (b). It is very important to
determine the optimal switching threshold in SWC. If the value of threshold is very high, then
the rate of undesirable switching transient increases. However, if the threshold is very low
then the diversity gain is also very low. The switching of switch combining can be performed
periodically in the case of frequency hopping systems.

Performance improvement obtained by the switching method leys on the value of threshold
selection, the delay of time that creates from the loop of feedback of monitoring estimation,
switching and decision. Moreover, phase transients and envelope of a carrier can reduce the
improvement of performance. In the system of angle modulation, for example, GSM, the
phase transient is responsible to create errors in detection stream of data. In this case, a pre-
detection band pass filter may be used to remove envelope transients.

Ant
Switch
Ant
Rx Det

Comp
Fixed
Threshold

(a)
 54

Ant
Switch
Ant
Rx Det

Comp Estimation

(b)

Picture 15: Switching combining methods with fixed threshold (a) and variable threshold (b).

6 Periodic Switching Method:

In a simple switching method, the diversity branches are selected periodically by a
conventional, free-running oscillator. This procedure is useful in comparably large deviational
and low-speed frequency modulation systems which includes phase transients creates by
switching can be diminished. The only selectable parameter switching rate can be chosen as
twice the height of the bit rate of signal. As a result the signal of the better branch can select
per signaling period. The performance can be improved as the same amount as it does at
conventional switching method by using FM discriminator which follows a suitable low-pass
filter (LPF). However, performance improvement may reduce in adjust-channel area because
this channel spectrum may be folded into desired channel band by periodic switching in the
pre-detection radio frequency stage. So we can see an overlap here which can be solved by
rising selectivity of the adjust-channel at the receiver.

Ant
Switch
Ant
Rx Det

 Oscillator

Picture 16: Periodic switching method

 55

7 Phase Sweeping Method:

Phase sweeping method is another version of switching method which uses a single receiver.
In phase sweeping method sweeping rate is more than twice the highest frequency of
modulation signal. But we can gain the same diversity improvement which we achieve by the
periodic switching method. The phase sweeping method is as like mode-averaging method
where spaced antennas are used with electrically scanned directional patterns. On the other
hand, Phase sweeping method may be applicable to Digital Phase Shift Keying (DPSK) and
FM systems.

Ant
Ant

 Rx Det

Ant


Sweep signal

Picture 17: Phase sweeping combining.

Channel estimation
In all communication systems, data is transferred from source to the destination in form of signals. These
signals traverse different medium which can be wired or wireless. Copper wires or fibre cables are two
examples of wired medium while air is a wireless medium. These mediums are also called channel. When
a signal passes from channel, it is distorted from the noise or from other signals traversing that same
medium. This means that when signal is received at its destination, it could have errors. So, in order to
remove the noise and distortion effects of channel from the received signal, channel’s properties have to
be found out. The process of figuring out channel characteristics is called Channel Estimation.

Sender
Medium Receiver

Input signal
Received Signal

Channel estimation process consists of multiple steps. First a mathematical model is created of
the channel. Then a signal which is known by both sender and receiver is transmitted over the
channel.
When the receiver receives the signal, it is of course distorted and contains noise from the
channel, but the receiver also knows the original signal, thus it can compare the original signal
and received signal to extract the properties of channel and the noises added to the sent signal in
the channel.

To put is in 3 main steps:

1. Mathematical model for channel is created. This model correlates sent and received signal
using channel matrix.
2. A signal known by both sender and receiver is sent by sender over the channel.
3. Receiver compares the received signal with original signal and figures out the values in
channel matrix.

Note: the signal that is sent and is known by both sender and receiver is usually called reference
signal or pilot signal.

Channel Estimation in Single Input Single Output System

Let’s consider a single input single output (SISO) system. As channel can affect different
frequency signals differently, so channel estimation have to be done for each frequency channel.
Depending on the number of channels it could be complex and resource consuming so often
channel estimation is done for fewer channels and estimates of rest of the channels are
interpolated from the computed estimates.
We consider 3 frequencies (in our example) from 𝑓 1 to 𝑓 3 from all the available frequencies. And this
will mean that reference signal or pilot signals will be 𝑥 (𝑓 1 ), 𝑥 (𝑓 2 ), 𝑎𝑛𝑑 𝑥 (𝑓 3 ). Reference signals are
complex signals (having I/Q form of data). For the remaining frequency channels, we can interpolate the
channel characteristics via the ones that are calculated.

When these reference signals are received at destination, they contain distortions and noise, and they are
represented by 𝑦(𝑓1 ), 𝑦(𝑓2 ), 𝑎𝑛𝑑 𝑦(𝑓3 ). Now to represent received signals 𝑦(𝑓) in terms of 𝑥(𝑓),channel
function is required for that specific frequency. It can be given by ℎ(𝑓).

Therefore, the relation of reference signal, received signal, and channel function can be represented by
the correlation function.

Which can be written as

𝑦(𝑓1 ) = ℎ(𝑓1 ) · 𝑥(𝑓1 )

𝑦(𝑓2 ) = ℎ(𝑓2 ) · 𝑥(𝑓2 )
𝑦(𝑓3 ) = ℎ(𝑓3 ) · 𝑥(𝑓3 )
From these equations, 𝑥(𝑓) 𝑎𝑛𝑑 𝑦(𝑓) are known thus ℎ(𝑓) can be calculated.

ℎ(𝑓1 ) = 𝑦(𝑓1 ) · 𝑥 𝐻 (𝑓1 )

ℎ(𝑓2 ) = 𝑦(𝑓2 ) · 𝑥 𝐻 (𝑓2 )

ℎ(𝑓3 ) = 𝑦(𝑓3 ) · 𝑥 𝐻 (𝑓3 )

Here 𝑥 𝐻 (𝑓) is the Hermitian of 𝑥(𝑓).

Now since, only 3 frequencies had been considered and channel characteristics are estimated
for only those frequencies, channel properties for rest of the frequencies can be estimated via
interpolation of already known characteristics.

Channel function in these equations represents channel distortion. Noise is also added to the
distorted signals, therefore actual equations of the received signal look like following:
𝑦(𝑓1 ) = ℎ(𝑓1 ) · 𝑥(𝑓1 ) + 𝑛(𝑓1 )
𝑦(𝑓2 ) = ℎ(𝑓2 ) · 𝑥(𝑓2 ) + 𝑛(𝑓2 )
𝑦(𝑓3 ) = ℎ(𝑓3 ) · 𝑥(𝑓3 ) + 𝑛(𝑓3 )
Similar to how channel function was estimated, theoretically noise can also be estimated by using
averaged channel estimate ℎ̅(𝑓).
 𝑛 ( 𝑓 1) = 𝑦 ( 𝑓 1) − ℎ̅( 𝑓 1) · 𝑥 ( 𝑓 1)

𝑛 ( 𝑓 2) = 𝑦 ( 𝑓 2) − ℎ̅( 𝑓 2) · 𝑥 ( 𝑓 2)

𝑛(𝑓3 ) = 𝑦(𝑓3 ) − ℎ̅(𝑓3 ) · 𝑥(𝑓3 )

But this provides with absolute values of noise and due to continuously variations in noise in a channel,
absolute values of noise are not beneficial in channel estimation. What’s beneficial is the estimated
function of noise which encompasses and models noise variations too. For that, there are different
algorithms and methods. One of them, which is implemented in srsLTE (Open-source LTE
implementation on SDR) is to subtract averaged channel estimate from actual channel estimate.

𝑛(𝑓1 ) = ℎ(𝑓1 ) − ℎ̅(𝑓1 )

𝑛(𝑓2 ) = ℎ(𝑓2 ) − ℎ̅(𝑓2 )

𝑛(𝑓3 ) = ℎ(𝑓3 ) − ℎ̅(𝑓3 )

Channel Estimation in Multiple Input Multiple Output (MIMO) system

In a Multiple input multiple output system, there are multiple transmission endpoints and multiple
receptions endpoints.
Receiver
Sender
ℎ11 𝑦1
𝑥1

ℎ12

ℎ21

ℎ22 𝑦2
𝑥2

In MIMO system the process of channel estimation remains the same except now there are two signals
received from a single source. This means that two paths in the medium were used, one path per signal.
Therefore, to compute the final signal 𝑦(𝑓) for each frequency, both received signals have to be
considered. This results in formation of the matrix of received signals.
𝑦1 ℎ ℎ12 𝑥1 𝑛1
[𝑦 ] = [ 11 ] [𝑥 ] + [𝑛 ]
2 ℎ21 ℎ22 2 2

Similar to SISO system, Hermitian of input matrix 𝑥 can be taken to estimate channel matrix ℎ.
And noise matrix 𝑛 can also be calculated in the similar manner via matrix operations.
 QUESTIONS FOR PRACTISE
Q.1. Define constraint length in convolutional codes?
Q.2. What is pseudo noise sequence?
Q.3. What is direct sequence spread spectrum modulation
Q.4. What is frequency hap spread spectrum modulation?
Q.5. What is processing gain?
Q.6. What is jamming margin ?
Q.7. When is the PN sequence called as maximal length sequence?
Q.8. What is meant by processing gain of DS spread spectrum system?
Q.9. What is the period of the maximal length sequence generated using 3 bit shift register.
Q.10. Define frequency hopping.
Q.11. What are the Advantages of DS-SS system
Q.12. What are the Disadvantages of DS-SS system.
Q.13. What are the Advantages of FH-SS System
Q.14. What are the Disadvantages of FH-SS System
Q.15. Define synchronization in Spread Spectrum Systems
Q.16. Comparison between DS-SS and FH-SS
Q.17. What are the Application of Direct Sequence Spread Spectrum 18. State the balance
property of random binary sequence.
Q.18. Mention about the run property.
Q.19. What is called jamming effect.
Q.20. What is Anti jamming ?
Q.21. What is slow and fast frequency hopping.