Digital Communications: Part I: Source Encoding

DIGITAL COMMUNICATIONS
Part I: Source Encoding

2000 Bijan Mobasseri 2
Why digital?
Ease of signal generation
Regenerative repeating capability
Increased noise immunity
Lower hardware cost
Ease of computer/communication
integration
Basic block diagram
Info
source
Source
encoder
Channel
encoder
Digital
modulation
Output
transducer
Source
decoder
Channel
decoder
Digital
demod
CH
channel
Some definitions
Information source
Raw data:voice, audio
Source encoder:converts analog info to a binary
bitstream
Channel encoder:map bitstream to a pulse
pattern
Digital modulator: RF carrier modulation of
bits or bauds
A bit of history
Foundation of digital communication is the
work of Nyquist(1924)
Problem:how to telegraph fastest on a
channel of bandwidth W?
Ironically, the original model for
communications was digital! (Morse code)
First telegraph link was established between
Baltimore and Washington in 1844
Nyquist theorem
Nyquist theorem, still standing today, says
that over a channel of bandwidth W, we can
signal fastest with no interference at a rate
no more than 2W
Any faster and we will get intersymbol
interference
He further proved that the pulse shape that
achieves this rate is a sinc
Signaling too fast
Here is what might happen when signaling
exceeds Nyquists rate
Transmittted bitstream
Received bitstream
Pulse smearing could have been avoided if pulses
had more separation, I.e. bitrate reduced
Shannon channel capacity
Claude Shannon, a Bell Labs
Mathematician, proved in 1948 that a
communication channel is fundamentally
speed-limited. This limit is given by
C=Wlog
2
(1+P/N
o
W) bits/sec
Where W is channels bandwidth, P signal
power and N
o
is noise spectral density

Implications of channel capacity
If data rate is kept below channel capacity,
R<C, then t is theoretically possible to
achieve error-free transmission
If data rate exceeds channel capacity, error-
free transmission is no longer possible
First step toward digital comm:
sampling theorem
Main question: can a finite number of
samples of a continuous wave be enough to
represent the information? OR
Can you tell what the original signal was
below?
How to fill in the blanks?
Could you have guessed this? Is there a
unique signal connecting the samples?
Sampling schemes
There are at least 3 sampling schemes
Ideal
Flat-top
Sample and hold
Ideal sampling
Ideal sampling refers to the type of samples
taken. Here, we are talking about impulse
like(zero width) samples.
T
s

Ideal sampler
Multiply the continuous signal g(t) with a
train of impulses
g(t)
Eo(t-nT
s
)
g
o
(t)=Eg(nT
s
) o(t-nT
s
)
T
s

Key question
What is the proper sampling rate to allow
for a perfect reconstruction of the signal
from its samples?
To answer this question, we need to know
how g(t) and g
o
(t) are related?
Spectrum of g
o
(t)
g
o
(t) is given by the following product
g
o
(t)=g(t)Eo(t-nT
s
)
Taking Fourier transform
G
o
(f)= G(f)*{f
s
Eo(f-nf
s
)
Graphical rendition of this convolution
follows next
Expanding the convolution
We can exchange convolution and
summation
G
o
(f)=G(f)*{f
s
Eo(f-nf
s
)= f
s
E {G(f)* o (f-nf
s
)}
Each convolution shifts G(f) to f= nf
s

G(f)* o (f-nf
s
)}
nf
s
G(f)
G
o
(f):final result
Spectrum of the sampled signal is then
given by

This is simply the replication of the original
continuous signal at multiples of sampling
rate
G
o
(f)=f
s
E {G(f-nf
s
)
Showing the spectrum of g
o
(t)
Each term of the convolution is the original
spectrum shifted to a multiple of sampling
frequency
G(f)
G
o
(f)
f
s
2f
s

f
s

Recovering the original signal
It is possible to recover the original
spectrum by lowpass filtering the sampled
signal

G
o
(f)
f
s
2f
s

f
s

LPF
W
W -W
Nyquist sampling rate
In order to cleanly extract baseband
(original) spectrum, we need sufficient
separation with the adjacent sidebands
Min. separation can be found as follows
G
o
(f)
f
s

W
f
s

f
s
-w>W
f
s
>2W
Sampling below Nyquist:
aliasing
If signal is sampled below its Nyquist rate,
spectral folding, or aliasing, occurs.
f
s
<2W
Lowpass filtering will not recover
the baseband spectrum intact as a
result of spectral folding
Sample-and-hold
A practical way of sampling a signal is
sample-and-hold operation. Here is the
idea:signal is sampled and its value held
until the next sample
Issues
Here are the questions we need to answer:
What is the sampling rate now?
Can the message be recovered?
What price do we pay for going with a practical
approach?
Modeling sample-and-hold
The result of sample-and-hold can be
simulated by writing the sampled signal as
s(t)=Em(nT
s
)h(t-nT
s
)
Where h(t) is a basic square pulse and m(t)
is the baseband message

This is a square pulse h(t) scaled by signal
Sample at that point, ie m(nT
s
)h(t-nT
s
)
T
s

A systems view
It is possible to come up with a system that
does sample-and-hold.
X h(t)
T
s

T
s

Each impulse generates a square pulse,h(t), at the output.
Outputs are also spaced by Ts this we have a sample-and-
hold signal
h(t)
Ideal sampling
Message reconstruction
Key question: can we go back to the
original signal after sample-and-hold ?
This question can be answered in the
frequency domain
Spectrum of the sample-and-hold
signal
Sample-and-hold signal is generated by
passing an ideally sampled signal, m
o
(t),
through a filter h(t). Therefore, we can write
s(t)= m
o
(t)*h(t)
or
S(f)= M
o
(f)H(f)
what we have available
Contains message M(f)
Known( it is a sinc)
Is message recoverable?
Lets look at the individual components of
S(f). From ideal sampling results
M
o
(f)=f
s
EM(f-kf
s
)
M
o
(f)
Problems with message recovery
The problem here is we dont have access to
M
o
(f). If we did, it would be like ideal
sampling
What we do have access to is S(f)
S(f)= M
o
(f)H(f)
We therefore have a distorted version of an
ideally sampled signal
Example message
Lets show what is happening. Assume a
message spectrum that is flat as follows
W -W
M(f)
M
o
(f)
fs 2fs
Sample-and-hold spectrum
We dont see M
o
(f). We see M
o
(f)H(f).
Since h(t) was a square pulse of width T
s
,
H(f) is sinc(fT
s
) .
M
o
(f).
H(f)
f
f
1/Ts=fs
W
Distortion potential
The original analog message is in the
lowpass term of M
o
(f)
H(f) through the product M
o
(f)H(f) causes
a distortion of this term.
Lowpass filtering of the sample-and-hold
signal will only recover a distorted message
Illustrating distortion
H(f)
f
2fs
1/Ts=fs
W
M
o
(f)
Sample and hold signal.
If lowpass filtered, the original
Message is not recovered
want to recover this
What is actually recovered
fs
How to control distortion?
In order to minimize the effect of H(f) on
reconstruction, we must make H(f) as flat as
possible in the message bandwidth(-W,W)
What does it mean? It means move the first
zero crossing to the right by increasing the
sampling rate, or decreasing pulse width
Does it make sense?

The narrower the pulse, hence higher
sampling rate, the more accurate you can
capture signal variations
Variation on sample-and-hold
Contrast the two following arrangements
sample period
and pulse width
are not the same
t
Ts
How does this affect
reconstruction?
The only thing that will change is h(t) and
hence H(f)
H(f)
f
f
1/t
W
M
o
(f)
Sample and hold signal.
If lowpass filtered, the original
Message is not recovered
want to recover this
What is actually recovered
different zero crossing
How to improve reconstruction?
Again, we need to flatten out H(f) within (-
W,W). and the way to do it is to use
narrower pulses (smaller t)
Sample-and-hold converges to
ideal sampling
If reducing the pulse width of h(t) is a good
idea, why not take it to the limit and make
them zero?
We can do that in which case sample-and-
hold collapses to ideal sampling(impulses
are zero width pulses)
Pulse Code Modulation
Filtering, Sampling, Quantization and
Encoding
Elements of PCM Transmitter
Encoder consists of 5 pieces

Transmission path
Continuous
message
LPF Sampler Quantizer Encoder
Regenerative
repeater
Regenerative
repeater
Quantization
Quantization is the process of taking
continuous samples and converting them to
a finite set of discrete levels
1.2
1.52
.86
-0.41
?
Defining a quantizer
Quantizer is defined by its input/output
characteristics; continuous values in,
discrete values out
in
out
in
out
Midtread type Midrise type
Output remains constant
Even as input varies over a range
Quantization noise/error
Quantizer clearly discards some
information. Question is how much error is
committed?
q(m)
Message(m) Quantized message (v)
Error=q=m-v
Illustrating quantization error

Sampled
quantized
Quantization error
v1
v2
v3
A
A quantizer step size
More on A
A Controls how fine samples are quantized.
Equivalently, A controls quantization error.
To determine A we need to know two
parameters
Number of quantization levels
Dynamic range of the signal
A for a uniform quantizer
Let sample values lie in the range ( -m
max
,
+m
max
). We also want to have exactly L
levels at the output of the quantizer. Simple
math tells us
max
min
L levels
A=2m
max
/L
Quantization error bounds
Quantization error is bounded by half the
step size
Level 2
Level 1
Error q
A
|q|<A/2
Error q
Statistics of q
Quantization error is random. It can be
positive or negative with equal
probability.
This is an example of a uniformly
distributed random variable.
q
Density function f(q)
A/2
-A/2
1/A
Quantization noise power
Any uniformly distributed random variable
in the range (-a/2 to a/2) has an average
power(variance) given by a
2
/12.
Here, quantization noise range is A,
therefore
o
2
q
= A
2
/12
Signal-to-quantization noise
Leaving aside random noise, there is always
a finite quantization noise.
Let the original continuous signal have
power P=<m
2
(t)> and quantization noise
variance(power) o
2
q
(SNR)
q
=P/ o
2
q
=12P/ A
2
Substituting for A
We have related step size to signal dynamic
range and number of quantization levels

Therefore, signal to quantization noise(sqnr)
sqnr=(SNR)q=[3P/m
2
max
]L
2

A=2m
max
/L
Example
Let m(t)=cos(2tf
m
t). What is the signal to
quantization noise ratio(sqnr) for a 256-
level quantizer
Average message power P is 0.5, therefore
sqnr=(3x0.5/1)256
2
=98304~50dB
Nonuniform quantizer
Uniform quantization is a fantasy. Reason is
that signal amplitude is not equally spread
out. It occupies mostly low amplitude levels
Solution:nonuniform intervals
Quantize fine where amplitudes spend most
of their time
Implementing nonuniform
quantization:companding
Signal is first processed through a nonlinear
device that stretches low amplitudes and
compresses large amplitudes
input
output
Low amplitudes stretched
Large amplitudes pressed
A-law and -law
There are two companding curves, A-law
and -law. Both are very similar
Each has an adjustment parameter that
controls the degree of companding (slope of
the curve)
Following companding, a uniform
quantization is used
Encoder
Quantizer outputs are merely levels. We
need to convert them to a bitstream to finish
the A/D operation
There are many ways of doing this
Natural coding
Gray coding
Natural coding
How many bits does it take to represent L-
levels? The answer is
n=log
2
L bits/sample
Natural coding is a simple decimal to binary
conversion
0000
1001
2010
3011
.
7111
Encoder output(3 bits per sample Quantizer levels(8)
Gray coding
Here is the problem with natural coding: if
levels 2(010) and 1(001) are mistaken, then
we suffer two bit errors
We want an encoding scheme that assigns
code words to adjacent levels that differ in
at most one bit location
Gray coding example
Take a 4-bit quantizer (16 levels). Adjacent
levels differ by juts one bit

0 0 0 0 1
1 0 0 0 0
2 0 1 0 0
3 0 1 0 1
4 1 1 0 1
.
Quantizer word size
Knowing n, we can refer to n-bit quantizers
For example, if L=256 with n=8bits/sample
We are then looking at an 8-bit quantizer
Interaction between sqnr and
bit/sample
Converting sqnr to dB provides a different
insight. Take 10log
10
(sqnr)
sqnr=kL
2
where k=[3P/m
2
max
]
In dB
(sqnr)
dB
=o+20logL= o+20log2
n
(sqnr)
dB
= o+6n dB
sqnr varies linearly with
bits/sample
What we just saw says higher sqnr is
achieved by increasing n(bits/sample).
Question then is, what keeps us from doing
that for ever thus getting arbitrarily large
sqnrs?
Cost factor
We can increase number of bits/sample
hence quantization levels but at a cost
The cost is in increased bandwidth but why?
One clue is that as we go to finer
quantization, levels become tightly packed
and difficult to discern at the receiver hence
higher error rates. There is also a bandwidth
cost

Basis for finding PCM bandwidth
Nyquist said in a channel with transmission
bandwidth B
T
, we can transmit at most 2B
T

pulses per second:
R(pulses/second)<2B
T
(Hz)
Or
B
T
(Hz)>R/2(pulses/second)
Transmission over phone lines
Analog phone lines are limited to 4KHz in
bandwidth, what is the fastest pulse rate
possible?
R<2BT=2x4000=8000 pulses/sec
Thats it? Modems do a bit faster than this!
One way to raise this rate is to stuff each
pulse with multiple bits. More on that later
Accomodating a digital source
A source is generating a million bits/sec.
What is the minimum required transmission
bandwidth.
B
T
>R/2=10
6
/2=500 KHz
PCM bit rate
The bit rate at the output of encoder is
simply the following product
R(bits/sec)=n(bits/sample)xf
s
(samples/sec)
R=nf
s
bits/sec

1 0 1 1 0 1
quantized
Encoded at 5 bits/sample
PCM bandwidth
But we know sampling frequency is 2W.
Substituting f
s
=2W in R=n f
s

R=2nW (bits/sec)
We also had B
T
>R/2. Replacing R we get
B
T
>nW

Comments on PCM bandwidth
We have established a lower bound(min) on
the required bandwidth.
The cost of doing PCM is the large required
bandwidth. The way we can measure it is
Bandwidth expansion quantified by
B
T
/W>n (bits/sample)
Bandwidth expansion factor
Similar to FM, there is a bandwidth
expansion factor relative to baseband, i.e.
|=B
T
/W>n
Lets say we have 8 bits/sample meaning it
takes , at a minimum, 8 times more than
baseband bandwidth to do PCM
PCM bandwidth example
Want to transmit voice (~4KHz ) using an
8-bit PCM. How much bandwidth is
needed?
We know W=4KHz, fs=8 KHz and n=8.
B
T
>nW=8x4000=32KHz
This is the minimum PCM bandwidth under
ideal conditions. Ideal has to do with
pulse shape used
Bandwidth-power exchange
We said using finer quantization (more
bits/sample) enhances sqnr because
(sqnr)
dB
= o+6n dB
At the same time we showed bandwidth
increases linearly with n. So we have a
trade-off

sqnr improvement
Lets say we increase n by 1 from 8 to 9
bits/sample. As result, sqnr increases by 6
dB
sqnr= o+6x8= o+48
sqnr= o+6x9= o+54

+6dB
Bandwidth increase
Going from n= 8 bits/sample, to 9
bits/sample, min. bandwidth rises from 8W
to 9W.
If message bandwidth is 4 KHz, then
B
T
=32 KHz for n=8
B
T
=36 KHz for n=9
+4 KHz or 12.5% increase
Is it worth it?
Lets look at the trade-off:
Cost in increased bandwidth:12.5%
Benefit in increased sqnr: 6dB
Every 3 dB means a doubling of the sqnr
ratio. So we have quadrupled sqnr by
paying 12.5% more in bandwidth
Another way to look at the
exchange
We provided 12.5% more bandwidth and
ended up with 6 dB more sqnr.
If we are satisfied with the sqnr we have,
we can dial back transmitted power by 6 dB
and suffer no loss in sqnr
In other words, we have exchanged
bandwidth for lower power
Similarity with FM
PCM and FM are examples of wideband
modulation. All such modulations provide
bandwidth-power exchange but at different
rates. Recall |=B
T
/W
FM.SNR~|
2
PCM..SNR~2
2|

Much more sensitive to beta,
Better exchnage
Complete PCM system design
Want to transmit voice with average power
of 1/2 watt and peak amplitude 1 volt using
256 level quantizer. Find
sqnr
Bit rate
PCM bandwidth
Signal to quantization noise
We had
sqnr=[3P/m
2
max
]L
2
We have L=256, P=1/2 and m
max
=1.

sqnr=98304~50 dB

PCM bitrate
Bit rate is given by
R=2nW (bits/sec)=2x8x4000=64 Kb/sec
This rate is a standard PCM voice channel
This is why we can have 56K transmission
over the digital portion of telephone
network which can accomodating 64
Kb/sec.
PCM bandwidth
We can really talk about minimum
bandwidth given by
B
T
|
min
=nW=8x4000=32 KHz
In other words, we need a minimum of 32
KHz bandwidth to transmit 64 KB/sec of
data.
Realistic PCM bandwidth
Rule of thumb to find the required
bandwidth for digital data is that
bandwidth=bit rate
B
T
=R
So for 64 KB/sec we need 64 KHz of
bandwidth
One hertz per bit
Differential PCM
Concept of differential encoding is of great
importance in communications
The underlying idea is not to look at
samples individually but to look at past
values as well.
Often, samples change very little thus a
substantial compression can be achieved
Why differential?
Lets say we have a DC signal and blindly
go about PCM-encoding it. Is it smart?

Clearly not. What we have failed to realize
is that samples dont change. We can send
the first sample and tell the receiver that the
rest are the same
Definition of differential
encoding
We can therefore say that in differential
encoding, what is recorded and ultimately
transmitted is the change in sample
amplitudes not their absolute values
We should send only what is NEW.

Where is the saving?
Consider the following two situations

The right samples are adjacent sample
differences with much smaller dynamic
range requiring fewer quantization levels
2
1.6
0.8
1.6
2 2
1.6
2
2
-0.8
-0.4
0
0.4
-0.4
0.4
0.8
Implementation of
DPCM:prediction
At the heart of DPCM is the idea of
prediction
Based on n-1 previous samples, encoder
generates an estimate of the nth sample.
Since the nth sample is known, prediction
error can be found. This error is then
transmitted
Illustrating prediction
Here is what is happening at the transmitter
Past samples(already sent)
To be trasmited
Prediction of the
Current sample
Prediction error
Only Prediction error is sent
What does the receiver do?
Receiver has the identical prediction
algorithm available to it. It has also
received all previous samples so it can make
a prediction of its own
Transmitter helps out by supplying the
prediction error which is then used by the
receiver to update the predicted value
Interesting speculation
What if our power of prediction was
perfect? In other words, what if we could
predict the next sample with no error?.
What kind of communication system would
be looking at?
Prediction error
Let m(t) be the message and Ts sample
interval, then prediction error is given

e(nT
s
) = m(nT
s
) m nT
s
( )
Prediction error
Prediction filter
Prediction is normally done using a
weighted sum of N previous samples

The quality of prediction depends on the
good choice of weights w
i

m nT
s
( ) = w
i
m n i ( )T
s
( )
i =1
N

Finding the optimum filter
How do you find the best weights?
Obviously, we need to minimize the
prediction error. This is done statistically

Choose a set of weights that gives the
lowest (on average) prediction error
over w
Min
e
2
nT
s
( )
{ }
Prediction gain
Prediction provides an SNR improvement
by a factor called prediction gain
G
p
=
o
M
2
o
e
2
=
message power
prediction error power
How much gain?
On average, this gain is about 4-11 dB.
Recall that 6 dB of SNR gain can be
exchanged for 1 bit per sample
At 8000 samples/sec(for speech) we can
save 1 to 2 bits per sample thus saving 8-16
Kb/sec.
DPCM encoder

Prediction error is used to correct the
estimate in time for the next round of
prediction
+
quantizer encoder
+
N-tap
prediction
Prediction error
Prediction
-
+
Prediction
error
Updated prediction
Input sample
Delta modulation (DM)
DM is actually a very simplified form of
DPCM
In DM, prediction of the next sample is
simply the previous sample
Estimate of
Prediction error
DM encoder-diagram
+
1-bit
quantizer
+
Delay Ts
Prediction error(A)
Prediction
-
+
Prediction
error
Updated prediction
Input sample
in
out
A
-A
DM encoder operation
Prediction error generates A at the output
of quantizer
If error is positive, it means prediction is
below sample value in which case the
estimate is updated by + A for the next step
Slope overload effect
Signal rises faster than prediction: A too
small
samples
Ts
initial estimate
A
predictions
Steady state: granular noise
Prediction can track the signal; prediction
error small
Two drops to reach the signal
A
Shortcomings of DM
It is clearly the prediction stage that is
lacking
Samples must be closely taken to insure that
previous-sample prediction algorithm is
reasonably accurate
This means higher sample rates
Multiplexing
Concurrent communications calls for some
form of multiplexing. There are 3 categories
FDMA(frequency division multiple access)
TDMA(time division multiple access)
CDMA(code division multiple access)
All 3 enjoy a healthy presence in the
communications market
FDMA
In FDM, multiple users can be on at the
same time by placing them in orthogonal
frequency bands
guardband
user 1 user 2 user N
TOTAL BANDWIDTH
FDMA example:AMPS
AMPS, wireless analog standard, is a good
example
Reverse link(mobile-to-base): 824-849MHz
Forward link: 869-894 MHz
channel bandwidth:30 KHz
total # channels: 833
Modulation: FM, peak deviation 12.5 KHz

TDMA
Where FDMA is primarily an analog
standard, TDMA and CDMA are for digital
communication
In TDMA, each user is assigned a time
slot, as opposed to a frequency slot in
FDMA

Basic idea behind TDMA
Take the following 3 digital lines

frame
TDM-PCM
quantizer and
encoder
quantizer and
encoder
channel
decoder
TDM-PAM
TDM-PCM(bits)
lpf
lpf
Parameters of TDM-PCM
A TDM-PCM line multiplexing M users is
characterized by the following parameters
data rate(bit or pulse rate)
bandwidth
TDM-PCM Data rate
Here is what we have
M users
Each sampled at Nyquist rate
Each sample PCMd into n bit words
Total bit rate then is
R=M(users)xf
s
(samples /sec/user)xn(bits/sec)
=nMf
s
bits sec
TDM-PCM bandwidth
Recall Nyquist bandwidth. Given R pulses
per second, we need at least R/2 Hz.
In reality we need more (depending on the
pulse shape) so
B
T
=R=nMf
s
Hz
T1 line
Best known of all TDM schemes is AT&Ts
T1 line
T1 line multiplexes 24 voice
channels(4KHz) into one single bitstream
running at the rate of 1.544 Mb/sec. Lets
see how
T1 line facts
Each of the 24 voice lines are sampled at 8
KHz
Each sample is then encoded into 8 bits
A frame consists of 24 samples, one from
each line
Some data bits are preempted for control
and supervisory signaling
T1 line structure:
all frames except 1,7,13,19...

1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8
channel 1
channel 2
channel 24
FRAME(repeats)
information bits (8-bits per sample)
Inserting non-data bits
In addition to data, we need slots for
signaling bits (on-hook/off hook, charging)
Every 6th frame (1,7,13,19..) is selected and
the least significant bit per channel is
replaced by a signaling bit
1 2 3 4 5 6 7 1 2 3 4 5 6 7 1 2 3 4 5 6 7
channel 1
channel 2
channel 24
Framing bit
Timing is of utmost significance in T1. We
MUST be able to know where the beginning
of each frame is
At the end of each frame a single bit is
added to help with frame identification
1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8
channel 1
channel 2
channel 24
information bits (8-bits per sample)
F
T1 frame length
How long is one frame?One revolution
generates frame
24
sampled at 8KHz
rotates at 8000 revs/sec.
frame length=1/8000=
125 microseconds
T1 bit rate per frame
Data rate
8x24=192 bits per frame
Framing bit rate
1 bit per frame
Total per frame
193 bits/frame

Total T1 bit rate
We know there are 8000 frames a sec. and
there are 193 bits per frame. Therefore

T1 rate=193x8000=1.544 Mb/sec
Signaling rate component
Not all 1.544 Mb/sec is data. In every 6th
frame, we replace 24 data bits by signaling
bits. Therefore
signaling rate=
(8000 frames/sec)(1/6)(24 bits)=32 Kbits/sec
TDM hierarchy
It is possible to build upon T1 as follows
1st level
multiplexer
24
64 kb/sec
2nd level
multiplexer
3rd level
multiplexer
DS-0
DS-1
DS-2
DS-3
DS-1:
1.544 MB/sec
DS-2:
6.312 Mb/sec
DS-3:
44.736 Mb/sec
7 lines
Recommended problems
6.2
6.15
6.17

Digital Communications: Part I: Source Encoding

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Digital Communications: Part I: Source Encoding

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DIGITAL COMMUNICATIONS

Part I: Source Encoding

2000 Bijan Mobasseri 96

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