1810377135

Atikur Rahman
Assignment on
Waveform Coding Techniques

# What is sampling Process and need of sampling rate:

In the sampling process, a continuous-time signal is converted into a discrete time signal by
measuring at periodic instants of time.
Proper selection of the sampling rate is very important because it determines how uniquely the
samples would represent the original signal.

# State and explain sampling theorem.

A bandwidth limited signal o finite energy g(t) that has no frequency components higher than W
hertz is completely described by specifying the values of the signal instants of time separated by 1/2W
seconds apart.
Ts ≤ 1/2W

A band limited signal of finite energy that has no frequency components higher than W hertz is
completely recovered from knowledge of its sample taken at the rate of 2W samples per second.
fs ≥ 2W

Proof Sampling Theorem:

There are two parts-

(I) Representation of g(t) in terms of its samples.
(II) Representation of g(t) from its samples.

Part (I):
Representation of g(t) in its samples g(nTs)

Step 1:
The sampled signal gδ(t) given as,
∞
gδ(t) = ∑ g (t ) δ (t −n Ts)
n=−∞

In the above equation δ(t-n Ts) indicates the sampled placed at ±Ts, ±2Ts, ±3Ts and so on.
Figure: (a) Analog signal (b) Discrete-time signal

Step 2:
Taking Fourier Transform of both sides of the equation (1)
∞
FT (gδ (t))=FT ( ∑ g (t ) δ (t−n Ts))
n=−∞

∞
Gδ (f )=G(f ) f s ∑ δ ( f −mf s )
m=−∞

∞
=f s ∑ G (f )δ (f −mf s) ………………. (2)
m=−∞

∞
=f s ∑ G (f −mf s ) …. By shifting property of impulse function.
m=−∞

Another form of Gδ(t) and G(f) is

∞
Gδ (f )= ∑ g (nT s )exp (− j 2 π n T s f ) …… (3)
m=−∞

and
∞
G(f )= ∑ g(t )exp (− j 2 π n f t) …….. (3a)
m=−∞
Figure: (a) Spectrum of signal g(t) (b) Spectrum of sampled signal gδ(t) for a sampling rate fs = 2W
(c) Ideal amplitude response of reconstruction filter

Suppose that, we choose the sampling period Ts = 1/2W and putting in equation (3)
∞
Gδ (f )= ∑ g (n/ 2 W ) exp(− j 2 π n f /ω )
n=−∞

Putting fs = 2W in equation 2(a), we have

G(f) = (1/2W) Gδ(f) for -ω < f < ω ……..(4)

∞
1
=
2W
∑ g(n /2 W )exp (− j π nF/ ω ) …… (5)
m=−∞

Part II:
Reconstruction of g(t) from its samples. Take the inverse Fourier transform of both side of
equation 3(a).
∞
g( f )= ∫ G (f )exp(2 π ft)df
−∞

ω ∞
1
=∫ ∑ g (n/2 W ) exp(− j π nf / ω )exp (2 π ft) df
−ω 2W m=−∞

∞ ω
1
= ∑ g (n /2 W ) ∫ exp[ j 2 π f (t−n/ 2 ω )] df
2 W −ω
m=−∞

∞
sin(2 π ω t−n π )
= ∑ g (n /2 W )
(2 π ω t −n π )
m=−∞

∞
g(t )= ∑ g (n/2 W ) sin(2 ω t −n)......... [as sin cx=sin π x / π x ]
n=−∞
# Quadrature Sampling of Band-Pass Signals:

In this scheme the band pass signal is split into two components one is in-phase component and
other is quadrature component. These two components will be low-pass so signals and are sampled
separately. This form of sampling is called Quadrature Sampling.

Let g(t) be a band pass signal of bandwidth 2W, centered around the frequency, fc (fc>W)

In phase component = gi(t) and quadrature component = gq(t)

The band pass signal g(t) can be expressed as

g(t) = gi(t) cos(2πfct) – gq(t) sin(2πfct).

The gi(t) and gq(t) signal are low-pass signals, having band limited to (-W<f<W)

#Reconstruction of a massage process from its samples:

#What is aliasing error:

Let {g(n/fs)} denote the sequence obtained by sampling an arbitrary signal g(t) at the rate fs
samples per second. Let gi(t) denote the signal reconstructed from this sequence by interpolation.
That is
∞
gi (t)= ∑ g(n / f s )sin c ( f s t−n)........( I )
n=−∞
equation (1) which is obtained from,
∞
g(t )= ∑ g (n/ f s )sin c (2 ω t−n) using fs in phase of 2ω.
n=−∞
The absolute error,

Є = | g(t) – gi(t)| is called aliasing error ………. (II)

#Bound on Aliasing Error:

Let {g(n/fs)} denote the sequence obtained by sampling an arbitrary signal g(t) at the rate fs
samples per second. Let gi(t) denote the signal reconstructed from this sequence by interpolation.
That is
∞
gi (t)= ∑ g(n / f s )sin c ( f s t−n)........( I )
n=−∞
The absolute error,

Є = | g(t) – gi(t)| ………. (II)

is called aliasing error

The Inverse Fourier transform of the spectrum G(f) is given as,
∞
g(t )= ∫ G(f )exp ( j 2 π ft )dt ……….(III)
−∞

or equivalently

∞ (m+1/2) f s

g(t )= ∑ ∫ G( f ) exp( j 2 π ft )dt ………...(IV)

m=−∞ (m−1/2) f s

Using poisons formula written in the form

∞
∑ G(f −mf s)=(1 /f s) g(n / f s )exp(− j 2 π f / f s ) ……..(V)
m=−∞

∞ +f s /2

so, g(t )= ∑ ∫ G(f −mf s )exp(2 π ft)df ……….(VI)

m=−∞ −f s /2

change f to (f-mfs )
(m+1/ 2)f s
∞
gi (t)= ∑ exp (− j 2 π mf s t) ∫ G(f )exp ( j 2 π ft )dt ……...(VII)
m=−∞ (m−1/ 2)f s

# Discuss signal distortion in sampling:

=> In deriving sampling theorem, we assumed that the signal is strictly band-limited.
=> The signal g(t) must have infinite duration for its spectrum to be strictly band-limited.
=> In practice, we have to work with a finite segment of the signal, in which case the spectrum
cannot strictly band-limited.
=> Consider a signal g(t) whose spectrum G(f) decreases with increasing frequency f without limit
as shown in figure bellow .
=> The spectrum Gδ(f) of the discrete-time signal gδ(f) and an infinite number of frequency-shifted
replicas of it. The replicas of G(f) are shifted in frequency by multiplies of sampling rate fs. Two
replicas of G(f) are shown in bellow.
#Draw the model for evaluating signal to distortion ratio.

#Explain the basic elements of a PCM System.

PCM is a digital representation of an analog signal that takes samples of amplitude of the
analog signal that takes samples of amplitude of the analog signal of regular intervals.
1. low-pass filter:
This filter eliminates the high frequency components present in the input signal, which is
greater than the highest frequency of the massage signal, to avoid error massage signal.

2. Sampling:
The incoming massage signal is sampled with a train of narrow rectangular pulses. In order to
ensure perfects reconstruction of the massage at the receiver, the sampling rate must be greater than
twice the highest frequency component W of the massage wave.

3. Quantizing:
Quantization is representing the samples values of the amplitudes by a finite sets or levels,
which means converting a continuous-amplitude samples into a discrete time signal.

4. Encoding:
A process t translate the discrete set of samples values to a more appropriate form of signal .
Best suited for transmission over a line or radio path.

5. Regeneration:

to control the effect of noise and distortion while passing through a channel and increase the signal
strength.

6. Decoding:
The decoder circuit decodes the pulses coded waveform to reproduce the original signal. This
circuit acts as the demodulator.

7. Reconstruction:
Decoder output is passed through a low-pass reconstruction filter whose cut-off frequency equal
to message bandwidth.

8. Multiplexing:
Different message sources are multiplexed by time division.

9. Synchronization:
Timing operations at the receiver must follow closely the corresponding operations at the
transmitter. Synchronization pulse or frame the transmitted along with code elements.
#What are the major source of PCM System?

The performance of a PCM system is influenced by two major sources of noise that are
independent.
(I) Channel noise, which may be introduced anywhere along the transmission path.
(II) Quantizing noise, which is introduced in the transmitter and is carried along to the
receiver output.

# Probability of error of PCM receiver.

Consider a binary-encoder PCM wave s(t) that uses the nonreturn-to-zero unipolar format.
When symbol 1 is sent, s(t) equals s1(t) defined by,

s 1 (t)=
√ E max
Tb
0≤t≤Tb …………………… (I)

where, Tb is the bit duration & Emax is the maximum or peak signal energy.

When symbol 0 is sent, s(t) equals s2(t) defined by

s2(t) = 0 0≤t≤Tb …………………… (II)

The transmitted PCM wave s(t) defined by

s(t) = s1(t) + s2(t)

The receiver PCM wave x(t) defined by,

x(t) = s(t) + w(t) w(t) = channel noise

where, w(t) is modeled as additive white Gaussian noise (AWGN)

For and AWGN channel, the optimum receiver uses a matched filter output is sampled at time (t-Tb),
where Tb is the bit duration. The resulting sample value is compared with a threshold by means of
decide device.

If the threshold is exceeded the symbol is 1, otherwise 0.

There is only one basis function of unit energy,

ϕ 1=
√ 1
Tb
0≤t≤Tb

the transmitted waveform s1(t) in terms of ϕ1(t) as follows,

s 1 (t)=√ E max ϕ (t) 0≤t≤Tb

Signal space for on-off PCM system that is one-dimensional and with two message points.
As shown in figure bellow,
Tb

s 11=∫ s 1 (t) ϕ 1 (t )dt =√ E max ϕ (t ) ……………..(III)

0¿

Tb

s 12=∫ s 1 (t) ϕ 1 (t )dt =0 …………….(IV)

0¿
The received signal point is calculated by sampling the matched filter output at time t=Tb
we thus write,
Tb

x 1=∫ x (t) ϕ 1 (t )dt ………………..(V)

0¿

When symbol 0 is sent we express the conditional probability of error Pe(t) as,

1
P e (0 )=( )erfc (
2
1
2 √ E max
N0
)

When given that symbol 1 was sent, we may express the conditional probability of error Pe(t) as,

1
P e (1)=( ) erfc(
2 2 N0 √
1 E max
)

We find that the average probability of error in the receiver is given by,

P e =P o P e (0)+ P 1 P e (1)

Since Pe(1) = Pe(0) and Po + P1 = 1, we obtain

Pe = Pe(1)= Pe(0)

or,
1
P e = erfc(
2 2 No√
1 E max
)

E max
represent the peak signal energy-to-noise spectrum density ratio.
No

The peak signal energy Emax may be written as,

Emax = Pmax Tb

where, Pmax is the maximum or peak signal power, Tb is the bit duration.

E max
Hence, we may expressed the ratio as,
No

E max P max
=
No No
Tb
 No
where is average noise power contained in a transmission bandwidth equal to the bit rate.
Tb