Ch03pulse Modulation

Chapter 3: Pulse Modulation
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2000, John Wiley & Sons, Inc.
Haykin/Communication Systems, 4th Ed
1
Chapter 3

Pulse Modulation
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2
Chapter Outline
- Sampling: is basic to all forms of pulse modulation.
- Pulse-amplitude modulation (PAM): is the simplest
form of modulation.
- Quantization: when combined with sampling,
permits to digitize analog signals.
- Pulse-code modulation (PCM): is the standard
method used to transmit analog signals by digital
means.
- Time-division multiplexing: provides for the time
sharing by a common channel.
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Chapter Outline (Continued)
- Digital multiplexers: combines many slow bit
streams into a single faster stream.
- Other forms of PCM: delta modulation (DM) and
differential PCM (DPCM).
- Linear prediction: is a basic form of encoding
analog message signals at low bit rates as in
DPCM.
- Adaptive forms of DPCM and DM.
- The MPEG-1/audio coding standard: is a transpa-
rent, perceptually loss-less compression system for
audio signals.

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Sampling Process
Let g(t) be a finite-energy band-limited signal with bandwidth W.
- The ideal sampled signal g
o
(t) obtained from g(t) is
=
o
=
o
t =
= o =
n
s s
m
s s
n
s s
nfT j nT g
mf f G f f G kT t nT g t g
) 2 exp( ) (
) ( ) ( ) ( ) ( ) (
where T
s
is the sampling period and f
s
=1/T
s
is the sampling frequency (rate).
- From the expression of G
o
(f), it is clear that sampling in time produces periodicity
in frequency. Also G
o
(f) can be represented as the discrete-time Fourier transform
(DTFT) of the samples g(kT
s
).
- The DTFT may be viewed as a complex Fourier series representation of the period
frequency function G
o
(f), where g(kT
s
) play the role of Fourier coefficients.
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5
Sampling Process (Continued)
- If T
s
=1/2W (or f
s
=2W), then G
o
(f) takes the form
=
=
=
o
+ =
|
.
|
\
|
t
|
.
|
\
|
=
0
) ( ) (
exp
2
) (
m
m
s s s
n
mf f G f f G f
W
nf j
W
n
g f G

- The second equation shows that with the choice of f
s
=2W, the spectrum G(f) may be
separated from other periods of G
o
(f), which is essential for the reconstruction
process.
- Under the two conditions: G(f)=0 for |f|>W and f
s
=2W, G(f) can be written as
W f W
W
nf j
W
n
g
W
W f W f G
W
f G
n
< <
|
.
|
\
|
t
|
.
|
\
|
=
< < =
=
o
, exp
2 2
1
), (
2
1
) (
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6
Sampling Process (Continued)
- This latter expression means that G(f) is uniquely determined by using the discrete-
time Fourier transform (DTFT) of the samples g(n/2W), taken at a sampling rate
f
s
=2W.
- This is true as long as f
s
>2W.
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7
Figure 3.1
The sampling process. (a) Analog signal.
(b) Instantaneously sampled version of
the analog signal.
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Figure 3.2
(a) Spectrum of a strictly band-limited
signal g(t). (b) Spectrum of the sampled
version of g(t) for a sampling period T
s
=
1/2 W.
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9
Signal Reconstruction
- In case T
s
=1/2W, the reconstruction of g(t) can be performed using inverse FT

}
}

}

< <
|
.
|
\
|
=
t t
t t
|
.
|
\
|
=
(
|
.
|
\
|
t
|
.
|
\
|
=
(
|
.
|
\
|
t
|
.
|
\
|
t
|
.
|
\
|
=
t =
n
n
n
W
W
W
W
n
n Wt
W
n
g
n Wt
n Wt
W
n
g
df
W
n
t f j
W W
n
g
df
W
n
t f j
W
nf j
W
n
g
W
df ft j f G t g
t - ), 2 ( c sin
2
) 2 (
) 2 sin(
2
2
2 exp
2
1
2
2
2 exp exp
2 2
1
) 2 exp( ) ( ) (
where sinc(2Wt-n) is called the interpolation function.
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Signal Reconstruction (Contd)
- As can be seen g(t) can be reconstructed from its samples g(n/2W), provided that
the sampling frequency (rate) f
s
>2W.
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11
Sampling Theorem
The sampling theorem for finite-energy band-limited signals can be stated in two equivalent
parts related to the transmitter and receiver of a pulse modulation system:
1. A finite-energy band-limited signal with bandwidth W Hertz, is completely described in
terms of its samples taken at a rate f
s
=1/T
s
>2W samples per second.
2. A finite-energy band-limited signal with bandwidth W Hertz, may be completely recovered
(reconstructed) from its samples taken at a rate f
s
=1/T
s
>2W samples per second.
Note: The minimum sampling rate f
s
=2W is called Nyquist rate and its reciprocal (inverse)
T
s
=1/2W is called Nyquist interval.
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Aliasing Effect
- In practice a message signal is not strictly band-limited. This means that the sampling rate
f
s
=2W may result in an overlapping between the different periods of G
o
(f) as shown by
Figure 3.3.
- This overlapping is called aliasing effect.
- Aliasing effect can be eliminated by using an anti-aliasing filter prior to sampling and using
a sampling rate slightly higher than Nyquist rate (f
s
=2W). This is shown in Figure 3.4.
g(t) g(kT
s
)
Anti-aliasing
Filter
Sampler
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Figure 3.3
(a) Spectrum of a signal. (b) Spectrum of an
undersampled version of the signal exhibiting the aliasing
phenomenon.
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Figure 3.4
(a) Anti-alias filtered
spectrum of an
information-bearing
signal. (b) Spectrum
of instantaneously
sampled version of
the signal, assuming
the use of a sampling
rate greater than the
Nyquist rate.
(c) Magnitude
response of
reconstruction filter.
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Example
1. In telephone system, the bandwidth of the voice signal (message signal) is limited
to W=3.1 kHz and the universal sampling frequency is f
s
= 8kHz > 2W.
2. Determine the Nyquist rates and Nyquist intervals used to sample the following
signals:
x(t)=sinc(200t), y(t)=sinc
2
(200t).
To determine the sampling rate of a signal, we have to know its frequency
spectrum. For x(t) and y(t), this may be done using Fourier transform.
|
.
|
\
|
=
200 200
1
) (
f
rect f X
,

>
<
|
|
.
|
\
|

=
200 , 0
200 ,
200
1
200
1
) (
f
f
f
f Y
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Example (Continued)
From X(f) and Y(f), it is clear that for x(t) the maximum frequency (bandwidth) W=100Hz and
for y(t) the maximum frequency (bandwidth) W=200Hz
X(f) Y(f)
1/200 1/200
f(Hz) f(Hz)
-100 0 100 -200 0 200
For x(t), f
s
=2W=200Hz and T
s
=1/f
s
=5ms.
For y(t), f
s
=2W=400Hz and T
s
=1/f
s
=2.5ms.
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Pulse-Amplitude Modulation
- Ideal sampling, as seen before, uses instantaneous Dirac pulses. This cannot be
realized (generated) in practice.
- A practical way to perform sampling is to use pulses with finite non-zero width
such as rectangular pulse.
- The result of sampling with rectangular pulses is pulse-amplitude modulation
(PAM). This is shown in Figure 3.5.
- The sampling theorem is still valid when a rectangular pulse instead of Dirac pulse
is used for sampling.
- The resulting signal is given by a rectangular pulse train but with amplitudes that
are varied in proportion to the corresponding sample values.
- For a message signal m(t), PAM signal is expressed as

=
=
n
s s
nT t h nT m t s ) ( ) ( ) (
,
where T
s
is the sampling interval and h(t) is a rectangular pulse of duration T ,
given by Figure 3.6a.
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Figure 3.5
Flat-top samples, representing an analog
signal.
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19
PAM (Continued)
- From previous (ideal sampling) section, the instantaneous sampled version of m(t)
is given by

=
o
o =
n
s s
nT t nT m t m ) ( ) ( ) (
- One can show that
=
o
= =
n
s s
nT t h nT m t h t m t s ) ( ) ( ) ( * ) ( ) ( , where * denotes the
convolution operation.
- In the frequency domain, we have S(f)=M
o
(f)H(f)=

=

k
s s
f H kf f M f ) ( ) (
, where
S(f)=FT[s(t)], M
o
(f)=FT[m
o
(t)], and H(f)=FT[h(t)]. FT[.] stands for Fourier
transform.
- How do we recover m(t) from PAM signal s(t) whose Fourier transform S(f)?
- From the expression of S(f), it is clear that the spectrum of the sampled PAM s(t) is
a repetition of the spectrum of m(t) reshaped with the frequency response
H(f)=Tsinc(fT)exp(-jtfT) of the sampling rectangular pulse.
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PAM (Continued)
- If f
s
>2W, then the output of a reconstruction filter will be M(f)H(f). This output is
equivalent to passing m(t) through another low-pass filter of frequency response
H(f).
- From Figure 3.6 we see that by using rectangular pulses to generate PAM signals,
we have introduced amplitude distortion and phase delay. This effect is called
aperture effect. The amplitude distortion effect can be removed by connecting an
equalizer in cascade with the reconstruction filter. The frequency response of the
equalizer should have a magnitude of the form

) sin( ) ( c sin
1
) (
1
fT
f
fT T f H t
t
= =
- Usually if the duty cycle T/T
s
is very small, then no equalization is needed.
- The noise performance of PAM systems can never be better than base-band
signal transmission. In practice PAM is used as intermediate step for other form
of modulation or time multiplexing.
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Figure 3.6
(a) Rectangular pulse h(t). (b) Spectrum H(f), made up
of the magnitude |H(f)|, and phase arg[H(f)].
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Figure 3.7
System for recovering message signal
m(t) from PAM signal s(t).
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Other Forms of Pulse
Modulation
Two other parameters of a pulse may be used to convey (carry) information. These are
duration and position.
- The resulting modulation schemes are pulse duration (width) modulation (PDM or PWM)
and pulse position modulation (PPM). PDM and PPM waveforms are shown in Figure 3.8.
- The generation of PDM and PPM waveforms is performed using the following circuit
Technique for generating PDM and PPM signals
PAM
Generator
oro
Clock
Mono-
stable
K
+
+
Triangle
Gnerator
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24
Figure 3.8
Illustrating two
different forms of
pulse-time
modulation for the
case of a sinusoidal
modulating wave.
(a) Modulating wave.
(b) Pulse carrier.
(c) PDM wave.
(d) PPM wave.
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Bandwidth-Noise Trade-off
- In the context of noise performance, a PPM system is the optimum form of analog
pulse modulation. PPM and FM have similar noise performance.
- Both PPM and FM have figure of merit (FOM) proportional to (B
T
)
2
.
- In terms of trade-off of increased B
T
for improved noise performance, the best we
can do with continuous-wave (CW) modulation and analog pulse modulation
systems is to follow a square law.
- Can we do better than a square law?
- The answer is yes, and the digital pulse modulation is the technique to have better
law for improvement.
- Specifically in pulse-code modulation (PCM), the message signal m(t) is
represented in discrete form both in time and amplitude.
- Digital pulse modulation techniques require sampling and quantization be
performed on the base-band signal m(t).
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Quantization Process
- Quantization is a process that transforms the continuous sample amplitude m(nT
s
)
of a message signal m(t) at time t=nT
s
into a discrete amplitude v(nT
s
) taken from a
finite set of possible discrete levels.
- In our study we assume that the quantization process is memoryless and
instantaneous, which means that the quantization at time t=nT
s
is not affected by
earlier or later samples of the message signal.
- Figure 3.10 shows the input/output characteristics of two types of quantizers.
- The relationship between the input and output of the quantizer is represented as
v=g(m).
- Quantizers can be of uniform or non-uniform type. In the former type, the levels of
the quantized amplitude are uniformly spaced. However, in the latter type the levels
are not.
- Since the quantization is an approximation process, it results into an approximation
error called quantization noise. Figure 3.11 illustrates the quantized signal and the
corresponding error (noise).
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Quantization Process (Contd)
- The quantization noise (error) is random and therefore for a specific sample it can
be represented by a random variable Q of sample value q. This sample value of
quantization error is given by: q=m-v.
- If the signal m(t) has an amplitude with dynamic range [-m
max
,m
max
], then for a
uniform quantization the step size of the quantizer is: A=2m
max
/L, where L is the
total number of quantization levels.
- According to Figure 3.10, the quantization error Q will have its sample values
bounded by -A/2sqsA/2.
- If A is sufficiently small (which means L is sufficiently large), we can assume that
Q is a uniformly distributed random variable with PDF

A s s A A
=
otherwise , 0
2 / 2 / , / 1
) (
q
q f
Q
- With this PDF, the mean and variance of Q are zero and
12
) (
2
2 2
A
= = o Q E
Q
,
respectively.
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Quantization Process (Contd)
- Typically the number of quantization levels L is chosen as power of two, i.e L=2
R
,
or equivalently R=log
2
(L), where R represents the number of bits used to encode
each quantization level. This gives a quantization step size A=2m
max
/2
R
and a
variance of quantization noise
R
Q
m
2 2
max
2
2
2
3
1
12

=
A
= o .
- The performance of the quantizer is measured using the output SNR defined as

R
m
Q
m
o
m
P P
SNR
2
2
max
2
2
2
) (
|
|
.
|
\
|
=
o
=
- This expression shows that the SNR at the quantizer output increases exponentially
with the number of bits R. And since increasing R requires a proportionate increase
in the channel (transmission) bandwidth B
T
, one can see that digital pulse
modulation follows an exponential law for noise performance.
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Figure 3.9
Description of a memoryless quantizer.
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Figure 3.10
Two types of quantization: (a) midtread and (b)
midrise.
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Figure 3.11
Illustration of the quantization process. (Adapted from
Bennett, 1948, with permission of AT&T.)
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Figure 3.12
Illustrating the partitioning of the dynamic
range A s m s A of a message signal
m(t) into a set of L cells.
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33
Example
Let m(t)=A
m
cos(2tf
m
t).

2 /
2
m m
A P =
, m
max
=A
m
, and
R
m Q
A
2 2 2
2
3
1

= o
) 2 (
2
3
3 / 2
2 /
) (
2
2 2
2
R
R
m
m
o
A
A
SNR = =

In terms of decibels, (SNR)
o
becomes
10log
10
(SNR)
o
= 1.8 + 6R
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Example (Continued)
Table 3.1 Signal-to-(quantization) noise ratio for different number of
quantization levels for sinusoidal modulation.
Number of Levels L Number of
bits/sample R
(SNR)
o
in dB
32 5 31.8
64 6 37.8
128 7 43.8
256 8 49.8
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Pulse-Code Modulation
- In PCM, a message signal is represented by a sequence of coded pulses, obtained
from representing the signal in discrete form in both time an amplitude.
- As shown by Figure 3.13 the basic operations performed at the transmitter are:
sampling, quantization, and encoding.
- The basic operations in the receiver are regeneration of impaired signals, decoding
and reconstruction.
- Regeneration also occurs during the route of transmission.
- In PCM of voice signals, non-uniform quantization is used to allow smaller
quantization step sizes form smaller amplitudes and larger step sizes for larger
amplitudes so that the (SNR)
o
remains quasi-constant for all levels of amplitudes.
- To use non-uniform quantization, the message signal is passed through a
compressor, then a uniform quantization is applied to the compressed signal. At
the receiver an expander circuit is used to undo the effect of the compressor.
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Pulse-Code Modulation (Contd)
- For the compression, two laws are adopted: the -law in North America and the A-
law in Europe.
-law:
) 1 ( log
) 1 ( log
+
+
=
e
e
m
v
A-law:
s s
+
+
s s
+
=
1
1

log 1
) ( log 1
1
0
log 1
m
A
,
A
m A
A
m ,
A
m A
v
e
e
e
- The characteristics of these two laws are shown in Figure 3.14. The typical values
used in practice are: =255 and A=87.6.
- After quantization the different quantized levels have to be represented in a form
suitable for transmission. This is done via an encoding process.
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37

Figure 3.13
The basic elements of a PCM system.
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38

Figure 3.14
Compression laws. (a) -law. (b) A-law.
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39
- Each discrete level is represented by a code element (symbol). In a binary code,
we have two symbols only, representing two levels of the signal.
- The electrical representation of a code is done by assigning a waveform (or a
pulse) to each symbol as shown in Figure 3.15 for the binary case.
- Some well-known line codes that can be used for the electrical representation of
a binary data stream, are: (a) Unipolar NRZ. (b) Polar NRZ signaling. (c)
Unipolar RZ signaling. (d) Bipolar RZ signaling. (e) Split-phase or Manchester
code. NRZ: non-return to zero. RZ: return to zero.
- These lines codes have some interesting spectral properties shown in Figure 3.16.
- Differential encoding method is used to encode information in terms of signal
transitions. In practice a transition designates symbol 0, while no transition
designates symbol 1 (see Figure 3.17). The original binary information is
recovered simply by comparing the polarity of adjacent binary symbols to
establish whether or not a transition has occurred.
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Figure 3.15
Line codes for the
electrical representations
of binary data.
(a) Unipolar NRZ
signaling. (b) Polar NRZ
signaling.
(c) Unipolar RZ signaling.
(d) Bipolar RZ signaling.
(e) Split-phase or
Manchester code.
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Figure 3.16a
Power spectra of line codes: (a) Unipolar NRZ signal.
The frequency is normalized with respect to the bit rate
1/T
b
, and the average power is normalized to unity.
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Figure 3.16b
Power spectra of line codes: (b) Polar NRZ signal.
1/T
b
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43
Figure 3.16c
Power spectra of line codes: (c) Unipolar RZ signal.
1/T
b
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Figure 3.16d
Power spectra of line codes: (d) Bipolar RZ signal.
1/T
b
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Figure 3.16e
Power spectra of line codes: (e) Manchester-encoded signal.
The frequency is normalized with respect to the bit rate 1/T
b
,
and the average power is normalized to unity.
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Regeneration
- A regenerative repeater (see Figure 3.18) consists of (1) an equalizer, (2) a timing circuit,
and (3) a decision-making device. The equalizer is used to undo the effect of the
transmission channel to get back the pulses in their original shape before transmission. The
timing circuit is used to recover the clock of the transmitted symbols (pulses), which is then
used in the decision-making process. The function of the decision-making device is to detect
the different pulses based on some threshold information.
- The purpose of a regenerative repeater is to clean the PCM signal during its transmission
through a channel.
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Noise in PCM Systems
- The performance of a PCM system is influenced by two noise sources: (1) channel
noise and (2) quantization noise.
- The main effect of channel noise is to introduce bit errors into the received signal.
The presence of this noise can be measured in terms of probability of symbol error
or bit error rate (BER).
- The effect of channel noise can be made practically negligible by using high signal
energy-to-noise density ratio through short spacing between regenerative repeaters.
- In the absence of channel noise, quantization noise is acting alone. Since
quantization noise is under the designer's control, it can be made negligible by
increasing the number of levels L and selecting a compressor-expander
(compander) pair that is matched to the message signal characteristics.
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Figure 3.17
(a) Original binary data. (b) Differentially
encoded data, assuming reference bit 1.
(c) Waveform of differentially encoded
data using unipolar NRZ signaling.
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Figure 3.18
Block diagram of regenerative repeater.
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50
Time-Division Multiplexing
- This technique combines time-domain samples from different message signals
(sampled at the same rate) and transmits them together across the same channel.
- The multiplexing is performed using a commutator (switch) as shown in Figure
3.19. At the receiver a decommutator (switch) is used in synchronism with the
commutator to demultiplex the data.
- TDM system is very sensitive to symbol dispersion, that is, to variation of
amplitude with frequency or lack of proportionality of phase with frequency.
This problem may be solved through equalization of both magnitude and phase.
- One of the methods used to synchronize the operations of multiplexing and
demultiplexing is to organize the mutiplexed stream of data as frames with a
special pattern. The pattern is known to the receiver and can be detected very
easily.
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Figure 3.19
Block diagram of TDM system.
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52
Digital Multiplexers
- This type of multiplexers is used to combine digital signals at different bit rates
such as voice, video, audio, and computer data.
- There are two groups of digital multiplexers: One group uses low bit-rate data
streams and the other is for high bit-rate data streams.
- The first group requires the use of modems.
- The second group of digital multiplexers forms a part of the data transmission
service provided by telecommunication carriers such as AT&T.
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Figure 3.20
Conceptual diagram of multiplexing-
demultiplexing.
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54
Digital Multiplexers (Continued)
- The hierarchy starts at 64 kb/s, which corresponds to PCM representation of voice signal.
- One bit stream with bit rate 64 kb/s is called digital signal 0 (DS0).
- The first-level hierarchy combines 24 DS0 into one single DS1 at a rate 1.544 Mb/s.
- The second-level hierarchy combines 4 DS1 into one single DS2 at a rate 6.312 Mb/s.
- The third-level hierarchy combines 7 DS2 into one single DS3 at a rate 44.736 Mb/s.
- The fourth-level hierarchy combines 6 DS3 into one single DS4 at a rate 274.176 Mb/s.
- The five-level hierarchy combines 2 DS4 into one single DS5 at a rate 560.160 Mb/s.
- The bit rate produced by each multiplexer is slightly higher than the total rate of the
multiplexed signals because of bit stuffing.
- The multiplexed signal must include some form of framing so that its individual
components can be identified at the receiver.
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Digital Multiplexers (Continued)
- The bit rate of interleaved signals must be locked to a common clock.
- The multiplexer has to handle small variations in the bit rates of the incoming digital signal
due to the propagation delay, by using a technique known as bit stuffing. At the
demultiplexer, the stuffed bits must be removed from the multiplexed signal. This requires a
method that can be used to identify the stuffed bits.
- AT&T M12 multiplexer is designed to combine 4 DS1 bit streams into 1 DS2. The format of
DS2 is given by Figure 3.21. One frame of DS2 consists of 6 48-bit words separated by 6
control bits. The 48-bit words are obtained by multiplexing 12 bits from each incoming DS1.
A DS2 frame consists of 24 control bits. Overall we have 3 types of control bits: F for the
overall frame, M for subframes, and C for stuffing indication.
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Figure 3.21
Signal format of AT&T M12 multiplexer.
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57
Virtues and Limitations of PCM
- The most important advantages of PCM are:
1. Robustness to channel noise and interference.
2. Efficient regeneration of the coded signal along the channel path.
3. Efficient exchange between B
T
and SNR.
4. Uniform format for different kind of base-band signals.
5. Flexible TDM.
6. Secure communication through the use of special modulation schemes of
encryption.
- These advantages are obtained at the cost of more complexity and increased B
T
.
- With cost-effective implementations, the cost issue no longer a problem of
concern.
- With the availability of wide-band communication channels and the use of
sophisticated data compression techniques, the large bandwidth is not a serious
problem.
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Delta Modulation (DM)
- In DM, the message signal is over-sampled to purposely increase correlation
between adjacent samples.
- The DM provides a staircase approximation to the message signal m(t) as shown in
Figure 3.22.
- the difference e[nT
s
]=m[nT
s
]-m
q
[(n-1)T
s
] is quantized into only two levels A.
- The error e[nT
s
] is quantized to give
e
q
= Asgn(e[nT
s
]).
The quantity e
q
is then used to compute the new
quantized level
m
q
[nT
s
]=m
q
[(n-1)T
s
]+e
q
[nT
s
]
- In DM the quantization levels are represented by two symbols: 0 for -A and 1 for
+A. In fact the coding process is performed on e
q
.
- The main advantage of DM is its simplicity as shown by Figure 3.23.
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Delta Modulation (Contd)
- The transmitter of a DM system (Figure 3.23a) is given by a comparator, a one-bit quantizer,
an accumulator, and an encoder.
- The receiver of a DM system (Figure 3.23b) is given by a decoder, an accumulator, and a low-
pass filter.
- DM is subject to two types of quantization error: Slope overload distortion and granular noise
(see Figure 3.24).
- Slope overload distortion is due to the fact that the staircase approximation m
q
(t) can't follow
closely the actual curve of the message signal m(t). In order for m
q
(t) to follow closely m(t), it
is required that

dt
t dm
T
s
) (
max >
A
be satisfied. Otherwise, step-size A is too small for the staircase approximation m
q
(t) to follow
m(t).
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60
Delta Modulation (Contd)
- In contrast to slope-overload distortion, granular noise occurs when A is too large relative to
the local slope characteristics of m(t). granular noise is similar to quantization noise in PCM.
- It seems that a large A is needed for rapid variations of m(t) to reduce the slope-overload
distortion and a small A is needed for slowly varying m(t) to reduce the granular noise. The
optimum A can only be a compromise between the two cases.
- To satisfy both cases, an adaptive DM is needed, where the step size A can be adjusted in
accordance with the input signal m(t).
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61
Figure 3.22
Illustration of delta modulation.
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62
Figure 3.23
DM system.
(a) Transmitter.
(b) Receiver.
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63
Figure 3.24
Illustration of the two different forms of
quantization error in delta modulation.
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64
Delta-Sigma Modulation
- The quantizer input of a DM transmitter may be viewed as an approximation to
the derivative of m(t). This leads to an accumulation of noise in the demodulated
signal.
- This drawback can be overcome by integrating m(t) prior to delta modulation.
- The use of integration has the following beneficial effects
1. The low frequency content of m(t) is pre-emphasized.
2. Correlation between adjacent samples of the delta modulator input is increased
which tends to reduce the variance of the error signal at the quantizer input.
3. Design of the receiver is simplified
- A DM scheme that includes an integrator at its input is called delta-sigma
modulation (D-EM). Figure 3.25 shows two equivalent versions of D-EM
system.
- The receiver of D-EM system consists simply of a low-pass filter.
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65
Figure 3.25
Two equivalent versions of delta-sigma modulation system.
7/25/2014
66
Linear Prediction
- Linear prediction is a signal processing function performed by a finite-duration impulse
response (FIR) discrete-time filter, as shown by Figure 3.26.
- Linear prediction consists of estimating the current sample of a signal from a certain
number of previous samples. This is always possible when the signal samples are
correlated.
- Linear predictor involves the use of three functional blocks (see Figure 3.26): (1) a set of
delay units, (2) a set of multipliers, and (3) a set of adders.
- For a linear predictor, the output is given by
=
=
p
k
k
k n x w n x
1
) ( ) (
,
where p is the prediction order, w
k
's are the predictor coefficients.
- Since the linear prediction is an estimation, it results into an error called prediction error
and given by:
) ( ) ( ) ( n x n x n e =
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67
Linear Prediction (Continued)
- The objective of designing a linear predictor (FIR filter) is to choose the coefficients
w=(w
1
, w
2
, , w
p
)
t
so as to minimize the mean-square error (MSE) criterion
J(w)=E[e
2
(n)]
- The result of minimizing J(w) with respect to w is
x opt x
r w R =
where R
x
is a Toeplitz matrix whose entries are given by
R
x
[i,j]=R
x
[|i-j|]=E[x(k-i)x(k-j)], 0si,jsp-1, w
opt
is the optimum coefficients vector in the
MSE sense, and r
x
is a px1 vector given by r
x
=(R
x
(1), R
x
(2), , R
x
(p))
t
.
- The corresponding minimum mean square error (MMSE) is given by
x x
t
x x
J r R r
1 2
min

o =
- It is common in practice to use an adaptive algorithm to perform the linear prediction, as
shown by Figure 3.27. The coefficients w are estimated using a gradient-based algorithm
called linear mean square (LMS) algorithm
) (
) ( ) 1 (
k e
k k
x w w + =
+
where k indicates the iteration number, is the adaptation step size, x=(x(k-1), x(k-2),
, x(k-p))
t
, and e(k) is the prediction error, given by
). ( ) ( ) ( k x k x k e =
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68
Linear Prediction (Continued)
- The LMS algorithm is a stochastic adaptive filtering algorithm, which starting from an
initial vector w
(0)
, seeks to find the minimum point of J(w) by following a zig-zag path. In
fact the exact minimum is never found because the algorithm keeps wandering around it
in a random fashion.
- The usefulness of the LMS algorithm resides in its tracking property since it can track the
minimum even for nonstationary signals such as speech signals, where the optimum
prediction coefficients w vary with time.
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69
Figure 3.26
Block diagram of a linear prediction filter
of order p.
7/25/2014
70
Figure 3.27
Block diagram illustrating the linear
adaptive prediction process.
7/25/2014
71
Differential PCM (DPCM)
- Voice and video signals represented in PCM exhibit high correlation, which means that
PCM signals contain redundant information. The result is an inefficient coding.
- By removing the PCM information redundancy a more efficient coded signal may be
obtained. This is done using DPCM.
- In DPCM a linear prediction is performed on samples of a message signal m(kT
s
)=m(k),
then the prediction error ) ( ) ( ) ( k m k m k e = is computed and fed to a quantizer to obtain
the quantized value e
q
(k)=e(k)+q(k), as shown by Figure 3.28a. q(k) is the quantization
error.
- The input of the linear predictor of Figure 3.28a is
) ( ) ( ) ( ) ( ) ( k q k m k e k m k m
q q
+ = + =
,
which represents a quantized version of the input sample m(k).
- If the prediction is well performed, then the variance of e(k) will be much smaller than
the variance of m(k), which results into a smaller number of levels to quantize e(k).
- The receiver as given by Figure 3.28b, consists of a decoder which produces e
q
(k), that is
added to the output of a prediction filter identical to the one used in the transmitter. The
result is the quantized message signal m
q
(k).
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72
Differential PCM (Continued)
- DPCM includes DM as a special case, where the prediction filter is a simple delay
element. Simply put, DM is a one-bit version of DPCM.
- The problem of slope-overload distortion may also arise in DPCM, whenever the slope of
the message signal changes too rapidly for the prediction filter to track it.
- The noise performance of DPCM is measured, as in other digital modulation systems, by
the output signal-to-quantization noise, given by
Q p
Q
E
E
M
Q
M
o
SNR G SNR ) ( ) (
2
2
2
2 2
2
=
|
|
.
|
\
|
o
o
|
|
.
|
\
|
o
o
=
o
o
=
where
2 2 2
and , ,
E Q M
o o o
are the variances of m(k), q(k), and e(k), respectively.
- The factor G
p
is the processing gain produced by the DPCM quantization scheme. When
G
p
>1, which is the case most of the time, it represents the gain in SNR obtained by using
DPCM compared to PCM.
- The receiver as given by Figure 3.28b, consists of a decoder which produces e
q
(k), that is
added to the output of a prediction filter identical to the one used in the transmitter. The
result is the quantized message signal m
q
(k).
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73
Figure 3.28
DPCM system.
(a) Transmitter.
(b) Receiver.
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74
Adaptive DPCM
- In PCM, the standard bit rate is 64 kbits/s. The aim of all the variants of PCM is to reduce
the number of bits used in the encoding process by removing redundancies.
- Adaptive DPCM (ADPCM) is a scheme that permits the coding of speech (voice) signals
at 32 kbits/s through the combined use of adaptive quantization and adaptive prediction.
- Adaptive quantization refers to a quantizer that operates with a time-varying step-size
) ( ) ( k k
M
o | = A
and adaptive prediction filter refers to a filter with time-varying
coefficients. | is a constant and
) ( k
M
o
is an estimate of the standard deviation of m(k).
- In ADPCM adaptive quantization can be performed using adaptive quantization with
forward estimation (AQF) or adaptive quantization with backward estimation (AQB).
- In ADPCM adaptive prediction can be performed using adaptive prediction with forward
estimation (APF) or adaptive prediction with backward estimation (APB).
- AQF and APF use unquantized samples of the input message signal to estimate o
M
and the
predictor coefficients w, respectively.
- AQB and APB use quantized samples of the input message signal to estimate o
M
and the
predictor coefficients w, respectively.
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75
Adaptive DPCM (Continued)
- Both AQF and APF suffer from the same disadvantages, which are the buffering, the side
(extra) information to be transmitted, and the delay. But by using AQB and APB these
disadvantages are eliminated.
- Figure 3.29 shows the AQB scheme and Figure 3.30 shows the APB scheme.
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76
Figure 3.29
Adaptive quantization with backward
estimation (AQB).
7/25/2014
77
Figure 3.30
Adaptive prediction with backward estimation (APB).
7/25/2014
78
Computer Experiment:
Adaptive delta Modulation
- Adaptive delta modulation (ADM) is a modification of DM, in which the step size is
adapted to the slope (variation) of the message signal.
- If successive errors are of opposite polarity, then the delta modulator is operating in the
granular mode; in such a case it is advantageous to use reduced step size.
- If successive errors are of the same polarity, then the delta modulator is operating in its
slope-overload mode; in this case, the step size should be increased.
- The algorithm used for adaptive DM with step size increase/decrease of 50% is
A(k)=|A(k-1)|(m
q
(k)+0.5m
q
(k-1))/m
q
(k), if A(k-1)>A
min
A(k)=A
min
, if A(k-1)<A
min
where if A(k) is the step size at iteration k and m
q
(k) is the one-bit quantizer output that
is equal to 1.
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79
Figure 3.31
Adaptive delta modulation system: (a) Transmitter. (b)
Receiver.
7/25/2014
80
Figure 3.32
Waveforms resulting from the computer
experiment on delta modulation: (a) Linear delta
modulation. (b) Adaptive delta modulation.
7/25/2014
81
MPEG/Audio Coding System
- For speech (voice) signal we have efficient coding schemes such as ADPCM because a
speech production model is available. Unfortunately, nothing similar exists for audio
signal.
- MPEG-1/audio coding standard is a lossy compression system that is used for audio
signals. It is capable of achieving transparent, perceptually lossless compression of
stereophonic audio signals at high sampling rate.
- MPEG stands for Motion Picture Experts Group.
- The MPEG-1/audio coding standard achieves such a performance by exploiting two
psycho-acoustic characteristics of the human auditory system: (1) Critical bands and (2)
auditory masking.
- Critical Bands: The human auditory system (with bandwidth up to 20 kHz) may be
modeled as a band-pass filter bank, consisting of 25 overlapping bands called critical
bands with bandwidths less than 100 Hz for low frequencies up to 5 kHz for high
frequencies.
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82
MPEG/Audio Coding System
(Continued)
- Auditory masking: A strong signal can mask a weak signal if they are in the same critical
band and the latter lies below a masking threshold (see Figure 3.33).
- From Figure 3.33, as long as the quantization noise level lies below the minimum masking
threshold, the quantization noise is inaudible.
- The transmitter and receiver of MPEG/audio coding system are given by Figure 3.34a and
Figure 3.34b, respectively. In the transmitter, the function of time-to-frequency mapping
network is to decompose the input audio signal into multiple sub-bands with frequency
resolutions closer to the partitions between the critical bands.
- The function of the psycho-acoustic model is to analyze the spectral content of the input
audio signal so as to compute the mask level for each sub-band. This will help the
quantizer allocating the appropriate number of bits to each quantized sample.
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83
Figure 3.33
Illustrating the definitions of masking threshold and related
parameters. The high-level signal (masker) lies inside the
darker-shaded critical band, hence the masking is more
effective in this band than in the neighboring band shown in
lighter shading. (Adapted from Noll (1998) with permission of
the CRC Press.)
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84
Figure 3.34
MPEG/Audio coding system. (a) Transmitter. (b)
Receiver.
7/25/2014
85
Figure P3.5
7/25/2014
86
Figure P3.22
7/25/2014
87
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