Telecommunications

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TELECOMMUNICATIONS

SYSTEMS AND TECHNOLOGY
PART 2-1
Telecommunications
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DIGITAL TRANSMISSION

PULSE

CODE

MODULATION
Telecommunications
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PRIMARY ADVANTAGE OF DIGITAL TRANSMISSION
OVER ANALOG TRANSMISSION IS NOISE IMMUNITY.
A DIGITAL SIGNAL IS A DISCRETE SIGNAL INSTEAD
OF A CONTINUOUS (ANALOG) SIGNAL. IT IS AN EASY
DETERMINATION FOR THE RECEIVER TO DISTINGUISH
A 0 FROM A 1

DIGITAL SYSTEMS USE SIGNAL REGENERATION
(REPEATERS) INSTEAD OF AMPLIFIERS. NOISE DOES
NOT PROPAGATE PAST THE REPEATER. SNR REMAINS
THE SAME AS THE ORIGINAL TRANSMITTED SIGNAL.
THUS, DIGITAL SIGNALS CAN BE TRANSPORTED
LONGER DISTANCES THAN ANALOG SIGNALS
DIGITAL TRANSMISSION
ADVANTAGES
Telecommunications
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NOISE
ELECTRICAL NOISE IS DEFINED AS ANY
UNDESIRABLE ELECTRICAL ENERGY THAT FALLS
WITHIN THE PASSBAND OF THE SIGNAL
SIGNAL WITHOUT NOISE
SIGNAL WITH NOISE
Telecommunications
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SIGNAL-TO-NOISE
POWER RATIO
S/N IS THE RATIO OF THE SIGNAL POWER LEVEL TO
THE NOISE POWER LEVEL
N
S
P
P
N
S
=
WATTS
WATTS
N
S
P
P
dB
N
S
log 10 ) ( =
S/N (DESIRABLE)
CIRCUIT/
SYSTEM
INPUT OUTPUT
N
S
V
V
dB
N
S
log 20 ) ( =
S/N Input
S/N Output
Factor Noise =
Best Case = 1
Telecommunications
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DIGITAL SIGNALS ARE BETTER SUITED TO
PROCESSING (USING DSP TECHNIQUES)
Voice Processing (MP3), Video Processing (MPEG)

DIGITAL SIGNALS ARE BETTER SUITED TO
MULTIPLEXING (USING TDM TECHNIQUES)

DIGITAL SIGNALS ARE BETTER SUITED FOR
STORAGE (MEMORY DEVICES)

THE TRANSMISSION RATE (THROUGHPUT = BPS)
CAN BE EASILY CHANGED TO ADAPT TO DIFFERENT
ENVIRONMENTS (SPEED MATCHING FEATURE) AND TO
INTERFACE WITH VARIOUS PERIPHERALS.
DIGITAL TRANSMISSION
ADVANTAGES
Telecommunications
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PULSE MODULATED SIGNALS SUCH AS:

PAM (PULSE AMPLITUDE MODULATION)
PWM (PULSE WIDTH MODULATION)
PPM (PULSE POSITION MODULATION)
PCM (PULSE CODE MODULATION)

DESPITE THE TERM MODULATION, THE ABOVE SIGNALS
ARE BASEBAND CODING SCHEMES AND THEY YIELD
BASEBAND SIGNALS. THE TERM PCM IS A MISNOMER
AS IT IS NOT REALLY A MODULATION FORM AS IN
AM, PM, FM


BASEBAND COMMUNICATION
Telecommunications
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BASEBAND SIGNALS HAVE SIZABLE POWER AT LOW
FREQUENCIES

BASEBAND SIGNALS CANNOT BE TRANSMITTED
OVER A RADIO LINK (FREE SPACE)

BASEBAND SIGNALS ARE SUITABLE FOR TRANSMISSION
OVER COPPER (PAIR OF WIRES, COAXIAL CABLE) OR
GLASS (FIBER). EXAMPLES:

LOCAL TELEPHONE COMMUNICATION
SHORT-HAUL PCM COMMUNICATION (BETWEEN
LOCAL EXCHANGES)
BASEBAND COMMUNICATION
Telecommunications
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B [Hz]
t
B
i

t
s

CIRCUIT SWITCHING
(FIXED CONNECTION PATH)
PACKET SWITCHING
(VARYING CONNECTION PATH)

TRANSMITTING DIGITAL DATA

Telecommunications
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PAM (PULSE AMPLITUDE MODULATION). THE
AMPLITUDE OF A TRAIN OF PULSE IS VARIED
ACCORDING TO THE AMPLITUDE OF THE ANALOG
SIGNAL (MODULATING SIGNAL)

PCM (PULSE CODE MODULATION). THE
ANALOG SIGNAL MODULATES A TRAIN OF PULSE
(PAM). IN EFFECT THE ANALOG SIGNAL IS SAMPLED
AND THE SAMPLES ARE CODED TO A BINARY VALUE
WHICH IS A FUNCTION OF THE AMPLITUDE OF THE
SAMPLED ANALOG SIGNAL
Digitizing Analog Speech
PAM & PCM
PCM = PAM + QUANTIZATION + COMPANDING
Telecommunications
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PULSE CODE MODULATION
PCM IS A TYPE OF CODING THAT IS CALLED
"WAVEFORM" CODING BECAUSE IT CREATES A
CODED FORM OF THE ORIGINAL VOICE WAVEFORM.
PCM IS A WAVEFORM CODING METHOD DEFINED
IN THE ITU-T G.711 SPECIFICATION.

http://www.itu.int/home/index.html
PCM WAS DEVELOPED BY BELL LABS IN 1937
PCM (ADPCM) IS THE PREFERRED METHOD OF
COMMUNICATION WITHIN THE PSTN
Telecommunications
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SINGLE CHANNEL SIMPLEX PCM TRANSMISSION SYSTEM
Quantization
Companding
Telecommunications
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PULSE CODE MODULATION
1. FILTERING STEP (ADDRESSES ALIASING ISSUES)

2. SAMPLING STEP (SAMPLE & HOLD PAM Signal)

3. DIGITIZING STEP (A/D CONVERSION BINARY VALUES)

QUANTIZATION
UNIFORM (LINEAR) QUANTIZATION PROCESS
ISSUE: QUANTIZATION DISTORTION (NOISE)

COMPANDING (COMPRESSING/EXPANDING)
COMBINES LOGARITHMIC & UNIFORM
QUANTIZATION
A-LAW
-LAW
Telecommunications
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FILTERING STEP
Frequency [Hz]
Amplitude
300
4000
Filter
Telecommunications
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FILTERING STEP



MOST OF THE ENERGY OF SPOKEN LANGUAGE IS
SOMEWHERE BETWEEN 200 OR 300 HERTZ AND ABOUT
3300 OR 3400 HERTZ:
KHZ B 3 ~
A GENERAL DECISION HAS BEEN MADE TO MAKE:
KHZ B 4 =
Telecommunications
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FILTERING STEP



THIS BANDLIMITING FILTER IS USED TO PREVENT
ALIASING DISTORTION (FOLDOVER DISTORTION).

THIS BANDLIMITING FILTER IS THUS CALLED AN
ANTIALIASING FILTER OR A ANTIFOLDOVER FILTER

THE ALIASING DISTORTION MIGHT OCCUR IN THE
SAMPLING STEP OF THE PCM PROCESS. THIS
DISTORTION IS A FUNCTION OF THE ANALOG VOICE
SIGNAL BANDWITH (B) AND THE VOICE SAMPLING
RATE ( f
s
). THIS DISTORTION OCCURS WHEN:
B fs 2 <
Telecommunications
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SAMPLING STEP



Telecommunications
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SAMPLING STEP



SAMPLE THE FILTERED INPUT SIGNAL AT A CONSTANT
SAMPLING FREQUENCY. ACCOMPLISHED BY USING A
PROCESS CALLED PULSE AMPLITUDE MODULATION
(PAM), THIS STEP USES THE ORIGINAL ANALOG SIGNAL
TO MODULATE THE AMPLITUDE OF A PULSE TRAIN
THAT HAS A CONSTANT AMPLITUDE AND FREQUENCY
THE PULSE TRAIN IS CHANGING AT A CONSTANT RATE
CALLED THE SAMPLING FREQUENCY
s f
FOR THE ADC TO ACCURATELY CONVERT THE
SIGNAL TO A BINARY CODE, THE SIGNAL NEEDS
TO BE RELATIVELY CONSTANT, THUS,
Telecommunications
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SAMPLING STEP



NATURAL SAMPLING (SAMPLE)


FLAT-TOP SAMPLING (SAMPLE AND HOLD)

MOST COMMON METHOD USED IN PCM
TWO TECHNIQUES
Telecommunications
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NATURAL SAMPLING
m(t)
s(t)
m(t)s(t)
PAM
Telecommunications
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FLAT-TOP SAMPLING
PAM
Telecommunications
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FLAT-TOP SAMPLING
Acquisition
time
LM398 SAMPLE&HOLD
DEVICE
KEEP
SHORT
sample
hold
Telecommunications
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(Coder/Decoder)
(Filtered)
Telecommunications
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SAMPLING RATE
(NYQUIST CRITERION)



NYQUIST (BELL LABS) SAMPLING THEOREM STATES
THAT FOR A SAMPLE TO BE REPRODUCED
ACCURATELY AT THE RECEIVER, THE SAMPLING RATE
IS AT LEAST TWICE THAT OF THE HIGHEST FREQUENCY
COMPONENT
B fs 2 >
f
S
= SAMPLING FREQUENCY

B = BANDWIDTH OF ORIGINAL ANALOG VOICE
SIGNAL
Telecommunications
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SAMPLING RATE
(NYQUIST CRITERION)



IF THE CRITERION
B fs 2 <
ALIASING OR FOLDOVER DISTORTION OCCURS
IS NOT MET
Telecommunications
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ALIASING DISTORTION



THE SAMPLE AND HOLD CIRCUIT IS AN AM SWITCHING
MODULATOR. IT IS A NON-LINEAR DEVICE WITH TWO
INPUTS:

1. THE SAMPLING TRAIN OF PULSE

2. THE INPUT ANALOG VOICE


CONSEQUENTLY, NON-LINEAR MIXING OCCURS
BETWEEN THESE TWO SIGNALS (HETERODYNING).
INTERMODULATION DISTORTION (IMD) OCCURS
Telecommunications
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SAMPLE & HOLD FUNCTIONALITY
) (t m
) (t s
) ( ) ( t s t m
MULTIPLIER
MODULATOR
EFFECT
NON-LINEAR MIXER
ANALOG
INPUT
SAMPLING
CLOCK
PAM
Telecommunications
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ALIASING DISTORTION



Input
Voice
signal
Sampling
signal
IMD signals
NO ALIASING
ALIASING
Telecommunications
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FILTERING STEP
Frequency [Hz]
Amplitude
fl
fh
Unwanted Signal noise Distortion
fs fs - fa
Telecommunications
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ALIASING DISTORTION
EXAMPLE
GIVEN A PCM SYSTEM (B = 4kHZ). THUS, SAMPLING
FREQUENCY IS 8 KHz (1 SAMPLE EVERY 1/8000 secs =
1 SAMPLE EVERY 125 seconds).
WHAT HAPPENS IF A 5 KHZ SIGNAL WERE ALLOWED
TO ENTER THE SAMPLE AND HOLD CIRCUIT (noise)?
NOTE THAT SINCE THE HIGHEST FREQUENCY
COMPONENT IS 5 KHZ:
B fs 2 <
ALIASING DISTORTION WILL OCCUR
Telecommunications
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ALIASING DISTORTION
a s f f
HIGHEST FREQUENCY
COMPONENT
QUANTIZING
NOISE
Telecommunications
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DIGITIZING STEP



THE PROCESS OF DIGITIZING ANALOG VOICE SIGNALS
IS CALLED PULSE CODE MODULATION (PCM). THE
ONLY DIFFERENCE BETWEEN PAM AND PCM IS THAT
PCM TAKES THE PROCESS ONE STEP FURTHER BY
ENCODING EACH ANALOG SAMPLE USING BINARY
CODE WORDS. BASICALLY, PCM HAS AN ANALOG-TO-
DIGITAL CONVERTER ON THE SOURCE SIDE (ADC) AND
A DIGITAL-TO-ANALOG CONVERTER (DAC) ON THE
DESTINATION SIDE.
Telecommunications
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LINEAR A/D CONVERTERS



CIRCUIT THAT GENERATES A BINARY OUTPUT CODE
THAT IS PROPORTIONAL TO AN ANALOG INPUT
VOLTAGE

TO MAINTAIN THE CONVERSION SPEED INDEPENDENT
OF THE ANALOG INPUT VOLTAGE VALUE, THE ADC
USES SUCCESSIVE APPROXIMATION (SA) TECHNIQUE

CONVERSION SPEED IS A FUNCTION OF THE NUMBER
OF BITS USED IN THE CONVERSION (i.e. 8-BIT ADC
NEEDS ONLY 8 COMPARISONS)
Telecommunications
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LINEAR A/D CONVERTER



8-BIT
SA REGISTER
AND CONTROL
8-BIT
DAC
8-BIT BUS
+
-
CK
V
IN

V
OUT

EXAMPLE:
ADC0804 (NS)
CC
CS
ANALOG
IN
DIGITAL
OUT
Telecommunications
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DIGITIZING STEP



THE INPUT ANALOG SIGNAL HAS A MAXIMUM
VOLTAGE LEVEL CALLED V
MAX

THE OUTPUT BINARY CODE HAS N BITS THUS WITH:
2
N
CODES

THE INPUT VOLTAGE RANGE IS THEN PARTITIONED
IN MANY STEPS EACH WITH SIZE:
N
MAX V
2
STEP SIZE =
RESOLUTION
Telecommunications
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DIGITIZING STEP



THE RESOLUTION OF AN ADC REPRESENTS THE
MAGNITUDE OF THE MINIMUM STEP SIZE. IT IS AN
INDICATOR OF HOW SMALL OF A VOLTAGE LEVEL CAN
BE DETECTED

THE SMALLER THE STEP SIZE, THE BETTER THE
RESOLUTION (GREATER), THE MORE ACCURATELY THE
QUANTIZATION LEVEL WILL REPRESENT THE ACTUAL
ANALOG SAMPLE

FOR A GIVEN INPUT VOLTAGE RANGE, THE GREATER
THE NUMBER OF BITS (N), THE BETTER THE
RESOLUTION (THE LARGER THE BITS/CODE)
Telecommunications
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DIGITIZING STEP



EXAMPLE
WITH A MAXIMUM INPUT ANALOG VOLTAGE OF 2.56 v
(ASSUMING ALL POSITIVE LEVELS) AND AN 8-BIT ADC
(ALL 8 BITS REPRESENT MAGNITUDE BITS):

NUMBER OF STEPS = 256

STEP SIZE (QUANTIZATION RANGE) = 2.56v / 256 = 10 mV

RESOLUTION = 10 mV
Telecommunications
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LINEAR QUANTIZATION



ALL QUANTIZATION INTERVALS ARE EQUALLY
SPACED (UNIFORM QUANTIZATION) THROUGHOUT THE
DYNAMIC RANGE OF THE INPUT ANALOG SIGNAL. EACH
QUANTIZATION LEVEL IS ASSIGNED A DISCRETE VALUE
IN THE FORM OF A BINARY CODE WORD.

EACH INPUT SAMPLE IS ASSIGNED A QUANTIZATION
LEVEL THAT IS CLOSEST TO ITS AMPLITUDE HEIGHT. IF
AN INPUT SAMPLE IS NOT ASSIGNED A QUANTIZATION
LEVEL THAT MATCHES ITS ACTUAL HEIGHT, THEN AN
ERROR IS INTRODUCED INTO THE PCM PROCESS. THIS
ERROR IS CALLED QUANTIZATION DISTORTION,
QUANTIZATION NOISE.
Telecommunications
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LINEAR QUANTIZATION



EXAMPLE:


1 = POSITIVE
0 = NEGATIVE
+2v
-1v +3v
2 BITS
SIGN BIT
8 quantization
Level (2 . 2
2
)
Telecommunications
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LINEAR QUANTIZATION
+1
-1
0
Range of standard
values (V)
15 : 1111
14: 1110
13: 1101
12: 1100
11: 1011
10: 1010
9: 1001
8: 1000
7: 0111
6: 0110
5: 0101
4: 0100
3: 0011
2: 0010
1: 0001
0: 0000
Coded v alues
Figure 2.4: quantization and resulting coding
using 16 quantizing steps
8 9 10 1112 13 12 11 10.. 6 .........
FIXED STEP SIZES
1000
1001
1010
1011
1100
1101
1110
1111
0111
0000
0110
0101
0100
0011
0010
0001
16 QUANTIZING STEPS
(8 POSITIVE, 8 NEGATIVE)
MSB
1 = +
MSB
0 = -
QUANTIZATION
DISTORTION
2s Complement
16 quantization
Level (2 . 2
3
)
Telecommunications
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QUANTIZATION NOISE



Telecommunications
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QUANTIZATION NOISE



QUANTIZATION NOISE IS EQUIVALENT TO THE
RANDOM NOISE THAT IMPACTS THE SIGNAL-TO-NOISE
RATIO (SNR) OF A VOICE SIGNAL. THE HIGHER THE
SNR, THE BETTER THE VOICE QUALITY.

QUANTIZATION NOISE REDUCES THE SNR OF A
SIGNAL. AN INCREASE IN QUANTIZATION NOISE
DEGRADES THE QUALITY OF A VOICE SIGNAL (SIGNAL
DEGRADATION)

IMPROVING QUANTIZATION NOISE IMPLIES A
LARGER NUMBER OF BITS (N) THUS AFFECTING
BANDWITH USAGE (UNDESIRABLE)
Telecommunications
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LINEAR QUANTIZATION



USES EQUAL QUANTIZATION INTERVALS
THROUGHOUT THE ENTIRE DYNAMIC RANGE OF AN
INPUT ANALOG SIGNAL (FOR LOW & HIGH ENERGY
SIGNALS)

LOW LEVEL SIGNALS HAVE A SMALL SNR, HIGH
LEVEL SIGNALS HAVE A LARGE SNR

MOST VOICE SIGNALS ARE OF LOW LEVELS. THUS,
VERY INEFFICIENT WAY OF DIGITIZING VOICE
SIGNALS

TO IMPROVE VOICE QUALITY AT LOWER SIGNAL
LEVELS, USE A NON-UNIFORM (NON-LINEAR)
QUANTIZATION PROCESS CALLED COMPANDING
Telecommunications
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NON-LINEAR QUANTIZATION



THE VOLTAGE RANGE BETWEEN THE LOWEST
LEVEL AND THE HIGHEST LEVEL IS DIVIDED INTO
SEGMENTS IN A NON-LINEAR MANNER
LOGARITHMIC

THE LOWER THE VOLTAGE LEVELS, THE SMALLER
THE RANGE OF A SEGMENT. THE RANGE OF A
SEGMENT GETS LARGER FOR HIGHER VOLTAGE
LEVELS

THE NUMBER OF STEPS FOR EACH SEGMENT IS THE
SAME
Telecommunications
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COMPANDING



DURING THE COMPANDING PROCESS, INPUT
ANALOG SIGNAL SAMPLES ARE COMPRESSED INTO
LOGARITHMIC SEGMENTS AND THEN EACH
SEGMENT IS QUANTIZED AND CODED USING
UNIFORM QUANTIZATION.
COMPANDING (COMPRESSION AND EXPANSION)
INCREASES SNR PERFORMANCE (MINIMIZED
QUANTIZATION NOISE) WHILE KEEPING THE
NUMBER OF BITS USED FOR QUANTIZATION
CONSTANT.
Telecommunications
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NON-LINEAR QUANTIZATION



LINEAR QUANTIZATION
NON-LINEAR QUANTIZATION
Telecommunications
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NON-LINEAR QUANTIZATION



0
0.5
1.5
3.0
5.0
Voltage levels
Segment #
1
2
3
4
5
6
7
8
Figure2_5: Nonlinear quantization using 8 segments with each
segment assigned two steps (two coded words)
SEGMENTS
Fixed Number
Of Steps Per
Segments
Companding
with
Uniform
Quantization
Telecommunications
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COMPANDING





A-LAW (EUROPE)
-LAW (NORTH AMERICA (USA), JAPAN)
Telecommunications
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-LAW ENCODING



) 1 ln(
) 1 ln(

+
+
=
Input
Compressed
V
V
V
Input
=AMPLITUDE OF INPUT

= PARAMETER TO DEFINE AMOUNT OF COMPRESSION

V
Compressed
= COMPRESSED OUTPUT AMPLITUDE
= 255 DS-1 systems
Telecommunications
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-LAW ENCODING



DS-1
= 255
Telecommunications
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-LAW DECODING



V
Input
=AMPLITUDE OF INPUT

= PARAMETER TO DEFINE AMOUNT OF COMPRESSION

V
Decompressed
= DECOMPRESSED OUTPUT AMPLITUDE

1 ) 1 ( +
=
Input
ed Decompress
V
V
= 255 DS-1 systems
Telecommunications
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-LAW DECODING



DS-1
= 255
Telecommunications
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255 CHARACTERISTICS



Telecommunications
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A-LAW ENCODING



A
V A
V
IN
Compressed
ln 1+
=
V
INPUT
=AMPLITUDE OF INPUT

A = PARAMETER TO DEFINE AMOUNT OF COMPRESSION

V
Compressed
= COMPRESSED OUTPUT AMPLITUDE
A
V A
V
IN
Compressed
ln 1
) ln( 1
+
+
=
A
VIN
1
0 s s
1
1
s s IN V
A
Telecommunications
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A-LAW ENCODING



A = 87.6
Telecommunications
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A-LAW DECODING



V
Input
=AMPLITUDE OF INPUT

A = PARAMETER TO DEFINE AMOUNT OF COMPRESSION

V
Decompressed
= DECOMPRESSED OUTPUT AMPLITUDE
A
V A
V
IN
ed Decompress
) ln 1 ( +
=
A
e
V
VIN A
ed Decompress
1 ) ln 1 ( +
=
A
VIN
1
0 s s
1
1
s s IN V
A
Telecommunications
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A-LAW DECODING



A = 87.6
Telecommunications
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A-LAW, u-LAW COMPANDING

Both are implemented using 8-bit code words (256 levels)
and provide a bit rate of 64 kbps (2 x 4 kHz x 8 bits)
Both break a dynamic range into a total of 16 segments: 8 positive
and 8 negative segments (3-bit segment identifier)
Each Segment is twice the length of the preceding one
Uniform quantization is used within each segment
(4-bit quantization level identifier)
Both use a similar approach to coding the 8-bit word:
First (MSB) identifies polarity
Bits 2,3,4 identify segment
Final 4 bits identify the quantization level
An international connection should use A-law, u to A conversion is
the responsibility of the u-law country
Telecommunications
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u-LAW PCM CODE FORMAT
S D C B A
Sign
Segment
Number
Level
Value
Figure 2.8: PCM Code Example
7-SEGMENTS
1 = +
0 = -
16 QUANTIZATION
LEVELS
3-bit Segment
Identifier
4-bit Quantization
Intervals
Telecommunications
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DELTA MODULATION

ADAPTIVE DELTA MODULATION

DIFFERENTIAL PCM (DPCM)

ADAPTIVE DIFFERENTIAL PCM (ADPCM)
OTHER MODULATION TECHNIQUES
Telecommunications
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DELTA MODULATION
ATTEMPTS TO MINIMIZE CODE SIZE (BITS/CODE)
WHILE MINIMIZING EFFECTS OF QUANTIZING NOISE

SAMPLE FAST ENOUGH SO THAT EACH SAMPLE DOES
NOT VARY MORE THAN 1 QUANTIZING STEP

TRANSMIT A SINGLE BIT THAT INDICATES WHETHER
THE CURRENT SAMPLE IS LARGER BY A SINGLE
STEP (XMIT A 1) OR SMALLER THAN THE PREVIOUS
ONE BY A SINGLE STEP (XMIT A 0)

DELTA MODULATION USEFUL ONLY FOR SIGNALS THAT
CHANGE SLOWLY (OTHERWISE GET SLOPE OVERLOAD)
Telecommunications
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DELTA MODULATION
IDEAL OPERATION
Telecommunications
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DELTA MODULATOR
Staircase
Waveform
Generator
Comparator
S
G
S > G 1
S < G 0
Telecommunications
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DELTA DEMODULATOR
Telecommunications
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DELTA MODULATION
SLOPE OVERLOAD
HARD TO TRACK
Telecommunications
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ADAPTIVE DELTA MODULATION
THE STEP SIZE OF THE DAC IS VARIED AS A
FUNCTION OF THE AMPLITUDE CHARACTERISTICS
OF THE ANALOG SIGNAL
INCREASING THE SAMPLING RATE IS NOT A DESIRABLE
SOLUTION BECAUSE IT INCREASES THE BANDWIDTH
REQUIREMENT
RESOLVING THE SLOPE OVERLOAD PROBLEM:
Telecommunications
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ADAPTIVE DELTA MODULATION (ADM)
In ADM, long string of 1s & 0s indicate that the staircase
generator is trying to catch up with the original signal and
the modulator will the increase the step size
Telecommunications
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DIFFERENTIAL PULSE CODE MODULATION
(DPCM)
During the PCM process, the differences between input sample
signals are minimal. Differential PCM (DPCM) is designed to
calculate this difference and then transmit this small difference
signal instead of the entire input sample signal.

Since the difference between input samples is less than an
entire input sample, the number of bits required for transmission
is reduced, allowing for a reduction in the throughput required to
transmit voice signals. Using DPCM can reduce the bit rate of
voice transmission down to 32Kbps (2:1 compression), 16 Kbps
(4:1 compression), 8 Kbps(8:1 compression largest voice quality
degradation)
Telecommunications
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ADAPTIVE DPCM
(ADPCM)
Adaptive DPCM (ADPCM) is a waveform coding method
defined in the ITU-T G.726 specification.
ADPCM adapts the quantization levels to the difference signal
that was generated during the DPCM process.
If the difference signal is low, ADPCM lowers the size of the
quantization levels. If the difference signal is high, ADPCM
decreases the size of the quantization levels.
ADPCM adapts the quantization level to the size of the input
difference signal, generating an SNR that is uniform throughout
the dynamic range of the difference signal.
Telecommunications
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CODEC (CODER/DECODER)
A CODEC IS AN LSI CHIP DESIGNED FOR USE IN THE
TELECOMMUNICATIONS INDUSTRY IMPLEMENTING
THE G.711 PCM RECOMMENDATION. IT IS USED IN
APPLICATIONS SUCH AS:

PRIVATE BRANCH EXCHANGES (PBXs)
MULTIPOINT CONFERENCING UNITS (MCUs)
CENTRAL OFFICE SWITCHES (COs)
DIGITAL PHONES
PCM-TDM CARRIERS

SAMPLING, ADC, DAC, COMPANDING

MANUFACTURERS: TI, AGERE, MOTOROLA, NATIONAL
SEMICONDUCTORS, ETC
Telecommunications
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COMBO CHIPS
LSI DEVICE THAT COMBINES THE CODEC AND
FILTER FUNCTIONS IN THE SAME DEVICE:

BANDPASS LIMITING: NOISE /ALIASING REJECTION
ENCODING/DECODING OF VOICE SIGNALS
ENCODING/DECODING OF CALL PROGRESS TONES
ENCODING/DECODING OF SIGNALING
ENCODING/DECODING OF SUPERVISION SIGNALS
DIGITAL COMPANDING

FIXED-DATA-RATE MODE: TRANSMIT & RECEIVE
CLOCKS ARE EQUAL (CLKX = CLKR).

VARIABLE-DATA-RATE MODE: CLKX != CLKR
Telecommunications
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COMBO CHIPS
TRANSMITTER
RECEIVER
Telecommunications
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COMBO CHIPS
SINGLE
CHANNEL
PCM
SYSTEM
Telecommunications
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COMBO CHIPS
Telecommunications
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COMBO CHIPS
24-CHANNELS
PCM
SYSTEM
Telecommunications
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COMBO CHIPS
Telecommunications
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URLS
http://www.ti.com/sc/docs/psheets/abstract/apps/spra349.htm

A-LAW , -LAW COMPANDING USING THE
TMS320C2XX DSP
MATLAB APPLICATIONS IN PCM:

http://engineering.eng.rowan.edu/~shreek/spring01/ecomms/tutorial.html

VOICEBOX Speech Processing Toolbox for MATLAB

http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/voicebox.html