WCN - Unit 1 - 2 PDF
WCN - Unit 1 - 2 PDF
WCN - Unit 1 - 2 PDF
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The ability to communicate with people on the move has evolved remarkably
since Guglielmo Marconi first demonstrated radio's ability to provide
continuous contact with ships sailing the English channel. That was in 1897,
and since then new wireless communications have been enthusiastically adopted
by people throughout the world.
Reflection occurs when the Electromagnetic waves fall on objects which are
much greater than the wavelength of the travelling wave. Diffraction is a
phenomena occurring when the wave interacts with a surface having sharp
irregularities. Scattering occurs when the medium through the wave is travelling
contains objects which are much smaller than the wavelength of the EM wave.
Most cellular radio systems operate in urban areas where there is no direct line-
of-sight path between the transmitter and the receiver, and where the presence
of high rise buildings causes severe diffraction loss. Due to multiple reflections
from various objects, the signal travels along different paths of different
lengths.
Due to the this multipath propogation of the signal , we will get many signal
replicas instead of one. These multipath component of the signal will phase
difference due to the difference in distances travelled by them, and hence the
sum of the component will have weakened strength as out of phase lead to
cancel each other. This phenomena is called MULTIPATH FADING .
Propagation Model : Models which predict the signal strength for arbitrary
transmitter receiver distances for wireless channels are termed as propagation
models.
In figure 1 the rapid fluctuation of received power is small scale fading and
average power is large scale .
Fig : 1 Small Scale and Large scale Fading
In small-scale fading, the received signal power may vary by as much as three
or four orders of magnitude (30 or 40 dB) when the receiver is moved by only a
fraction of a wavelength. As the mobile moves away from the transmitter over
much larger distances, the local average received signal will gradually decrease,
and it is this local average signal level that is predicted by large-scale
propagation models.
.........(1)
where Pt is the transmitted power, Pr(d) is the received power, Gt is the
transmitter
antenna gain, Gr is the receiver antenna gain, d is the Tx-Rx separation and L is
the system loss factor and is generally 1 for wireless communication.. The gain
of the antenna is related to the effective aperture of the antenna which in turn is
dependent upon the physical size of the antenna as given below
G = 4πAe/λ2
The effective aperture Ae is related to the physical size of the antenna, and
λ is related to the carrier frequency by
λ = c/f
where c is speed of light in meters/s and f is carrier frequency of transmitted
signal in Hz
The path loss, representing the attenuation suffered by the signal as it travels
through the wireless channel is given by the difference of the transmitted and
received power in dB and is expressed as:
PL(dB) = 10log Pt / Pr
The Friis free space model is only a valid predictor for for values of d which are
in the far-field of the 'transmitting antenna. The far-field, or Fraunhofer region,
of a transmitting antenna is defined as the region beyond the far field distance
df, which is related to the largest linear dimension of the transmitter antenna
aperture and the carrier wavelength. The Fraunhofer distance is given by
df = 2D2/ λ
The received power Pr(d) at any distance d > d0, may be related to Pr at d0.
Example 1
Solution
Given:
Transmitter power Pt, = 50 W
Carrier frequency fc, = 900 MHz
= 1/3 m
The received power in watts and dBm is calculate for distance 100m assuming
Gt , Gr and L =1 as
*******************************************************
In a mobile radio channel, a single direct path between the base station and
a mobile is seldom the only physical means for propagation, and hence the free
space propagation model is in most cases inaccurate.
The 2-ray ground reflection model shown in Figure 2 where two signal
components are considered one direct Line of sight and other is a reflected
component of the signal. At the receiver the two signals reaching with different
path length ( Distance ) and thus having different phases associated with them.
Here the ground is considered as reflecting object.
The total received E-field, ETOT, is then a result of the direct line-of-sight
component, ELOS , and the ground reflected component, Eg. Referring to Figure
2, ht is the height of the transmitter and hr is the height of the receiver.
If E0 is the free space E-field (in units of V/m) at a reference distance d0 from
the transmitter, then for d > d0, the free space propagating E-field is given by
where E (d, t) = E0d0/d represents the envelope of the E-field at d meters from
the transmitter. and cos wct is tranmitted signal . after travelling distance d the
time delay of the signal is d/c, so delay difference will be( t-d/c ) .
Two propagating waves arrive at the receiver: the direct wave that travels a
distance d'; and the reflected wave that travels a distance d''.
The E-field due to the line-of-sight component at the receiver can be expressed
as
and the E-fleld for the ground reflected wave, which has a propagation distance
of d", can be expressed as
The resultant E-field, is the vector sum of ELOS and Eg and the resultant total
E-field envelope is given by
similarly
d''2 = (ht +hr) 2 + d2
However, when T-R separation distance is very large compared to (ht + hr),
then
by taylors approximation
= d[ {1+ 1/2 (ht2+ hr2 +2hthr)/d2}- {1+ 1/2 (ht2+ hr2 - 2hthr)/d2}
=d[ 1+ 1/2 ht2/ d2 +1/2 hr 2/ d2 + 1/2 *2ht hr / d2 - 1 - 1/2 ht2/ d2 -1/2 hr2/ d2 +
1/2 *2ht hr /d 2}
= d/ d2 [ 2hthr]
= 2hthr/d
.....(3)
....(4)
and the time difference
When d is very large, then Δ becomes very small and therefore ELOS and Eg are
virtually identical with only phase difference,i.e.,
Then
Figure 4
we get
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Okumura Model
Okumura's model is one of the most widely used models for sighal prediction
in urban areas. This model is applicable for frequencies in the range 150
MHz to 1920 MHz (although it is typically extrapolated up to 3000 MHz) and
distances of 1 km to 100 km. It can be used for base station antenna heights
ranging from 30 m to 1000 m.
Okumura developed a set of curves giving the median attenuation relative
to free space (Amu), in an urban area over a quasi-smooth terrain with a base
station
effective antenna height (hte) of 200 m and a mobile antenna height (hre) of
3m
where L50 is the 50th percentile (i.e., median) value of propagation path loss, LF
is the free space propagation loss, Amu (f,d) is the median attenuation relative to
free
space, G(hte) is the base station antenna height gain factor, G(hre) is the
mobile antenna height gain factor, and GAREA is the gain due to the type of
environment.
The plots for Amu (f,d) and GAREA are shown in figure 5 and 6.
Figure 5 : Amu( f,d)
Figure 6 : GAREA
Example 2 :
Find the median path loss using Okumura's model for d = 50 km, = 100, hte =
100m , hre = 10m in a suburban environment. If the base station transmitter
radiates an power of I kW at a carrier frequency of 900 MHz, find the power at
the receiver.
Solution
Hata Model
The standard formula for empirical path loss in urban areas under the Hata
model is
The parameters in this model are same as in the Okumura model,and a(hr) is a
correction factor for the mobile antenna height based on the size of coverage
area.For
small to medium sized cities this factor is given by
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(2) Speed of the mobile – The relative motion between the base station and the
mobile results in random frequency modulation due to different doppler shifts
on each of the multipath components.
Doppler Shift
where Δt is the time required for the mobile to travel from A to B, and θ is
assumed to be the same at points A and B since the source is assumed to be very
far away. The phase change in the received signal due to the difference in path
lengths is therefore
The mean excess delay is the first moment of the power delay profile and is
defined to be
where ak is the amplitude, τk is the excess delay and P(τk) is the power of the
individual multipath signals.
The mean square excess delay spread is defined as
Since the rms delay spread is the square root of the second central moment of
the
power delay profile, it can be written as
The maximum excess delay (X dB) of the power delay profile is defined to be
the time delay during which multipath energy falls to X dB below the
maximum.
Figure 7: Example of an indoor power delay profile; nns delay spread, mean
excess delay, maximum excess delay (10 dB).
Coherence Bandwidth
Example 3 :
Calculate the mean excess delay, rms delay spread, and the maximum excess
delay (10 dE) for the multipath profile given in the figure below. Estimate the
50% coherence bandwidth of the channel
Solution
The second moment for the given power delay profile can be calculated
as
Coherence time: this is a statistical measure of the time duration over which
the channel impulse response is almost invariant. When channel behaves like
this, it is said to be slow faded. Essentially it is the minimum time duration over
which two received signals are affected differently.
For an example, if the coherence time is considered to be the bandwidth over
which the time correlation is above 0.5, then
it can be approximated as
The type of fading experienced by the signal through a mobile channel depends
on the relation between the signal parameters (bandwidth, symbol period) and
the channel parameters (rms delay spread and Doppler spread). Hence we have
four different types of fading. There are two types of fading due to the time
dispersive nature of the channel.
Flat Fading
Such types of fading occurs when the bandwidth of the transmitted signal is less
than the coherence bandwidth of the channel. Equivalently if the symbol period
of the signal is more than the rms delay spread of the channel, then the fading is
flat fading.
So we can say that flat fading occurs when
BS < BC
where BS is the signal bandwidth and BC is the coherence bandwidth. Also
TS > στ
where TS is the symbol period and στ is the rms delay spread. And in such a
case,
mobile channel has a constant gain and linear phase response over its bandwidth
Fast Fading
In a fast fading channel, the channel impulse response changes rapidly within
the symbol duration of the signal. Due to Doppler spreading, signal undergoes
frequency dispersion leading to distortion. Therefore a signal undergoes fast
fading if
TS > TC
BD is doppler spread
Slow Fading
In such a channel, the rate of the change of the channel impulse response is
much less than the transmitted signal. We can consider a slow faded channel a
channel in which channel is almost constant over atleast one symbol duration.
Hence
TS < TC
and
BS > BD
We observe that the velocity of the user plays an important role in deciding
whether the signal experiences fast or slow fading.
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************************************
Equalization
ISI has been identified as one of the major obstacles to high speed data
transmission over mobile radio channels. If the modulation bandwidth exceeds
the coherence bandwidth of the radio channel (i.e., frequency selective fading),
modulation pulses are spread in time, causing ISI. An equalizer at the front end
of a receiver compensates for the average range of expected channel amplitude
and delay characteristics. As the mobile fading channels are random and time
varying, equalizers must track the time-varying characteristics of the mobile
channel and therefore should be time varying or adaptive.
Tracking Mode:
• When the training sequence is finished the filter coefficients are near optimal.
• Immediately following the training sequence, user data is sent.
• When the data of the users are received, the adaptive algorithms of the
equalizer
tracks the changing channel.
• As a result, the adaptive equalizer continuously changes the filter
characteristics
over time.
Figure 8 Block diagram of a simplified communications system using
an adaptive equalizer at the receiver
where d(t) is the transmitted signal, h(t) is the combined impulse response of the
transmitter,channel and the RF/IF section of the receiver and nb (t) denotes the
baseband noise.
If the impulse response of the equalizer is heq (t), the output of the equalizer is
However, the desired output of the equalizer is d(t) which is the original source
data.
Assuming nb (t)=0, we can write y(t) = d(t), which in turn stems the following
equation:
The main goal of any equalization process is to satisfy this equation optimally.
In frequency domain it can be written as
which indicates that an equalizer is actually an inverse filter of the channel. If
the channel is frequency selective, the equalizer enhances the frequency
components with small amplitudes and attenuates the strong frequencies in the
received frequency spectrum in order to provide a flat, composite received
frequency response and linear phase response .
The basic structure of an adaptive filter is shown in Figure This filter is called
the transversal filter, and in this case has N delay elements, N+1 taps and N+1
tunable complex multipliers, called weights. These weights are updated
continuously by an adaptive algorithm. In the figure the subscript k represents
discrete time index. The adaptive algorithm is controlled by the error signal ek.
The error signal is derived by comparing the output of the equalizer, with some
signal dk which is replica of transmitted signal. The adaptive algorithm uses ek.
to minimize the cost function and uses the equalizer weights in such a manner
that it minimizes the cost function iteratively.
Linear Equalizer
Nonlinear Equalization
Three very effective nonlinear methods have been developed which offer
improvements over linear equalization techniques.
1. Decision Feedback Equalization (DFE)
2. Maximum Likelihood Symbol Detection
3. Maximum Likelihood Sequence Estimation (MLSE)
where c, and y, are tap gains and the inputs, respectively, to the forward filter,
F1 are tap gains for the feedback filter
Three classic equalizer algorithms are discussed below. These include the
zero forcing (ZF) algorithm, the least mean squares (LMS) algorithm, and the
recursive least squares (RLS) algorithm.
LMS Algorithm
A more robust equalizer is the LMS equalizer where the criterion used is
the minimization of the mean square error (MSE) between the desired equalizer
output and the actual equalizer output.
The LMS algorithm seeks to minimize the mean square error given in
equation