ECE REVIEW EST | Digital Communications

A. DEFINITION & BLOCK DIAGRAM

Digital Communication covers digital transmission

and digital radio. It gradually. It gradually replaced the
traditional analog systems that use conventional AM,
FM and PM.

Digital Transmission is a transmittal of digital pulses

between two points in a communication system while
Digital Radio is a transmittal of digitally modulated
analog carrier between two points in a
communication system.

Digital Transmission Block Diagram:

Frequency Shift Keying (FSK) – frequency of the
analog carrier is varied by the digital modulating
pulse. In its simplest form, two frequencies are
generated, one corresponding to a binary zero (space)
and the other a binary one (mark).

FSK is reliable in the presence of noise since each

signal has only two possible states. FSK is also used in
high-frequency radio systems to transmit teletype
information.

FSK Transmitter
Digital Modulation Block Diagram:

FSK Waveform

B. DIGITAL MODULATION

Digital Radio or Digital Modulation is the property

that distinguishes a digital radio system from a
conventional AM, FM, or PM radio system is that in a
digital radio system the modulating and demodulated
signals are digital pulses rather than analog
waveforms
Note:
DIGITAL MODULATION TECHNIQUES

Amplitudes Shift Keying (ASK) – amplitude of the • The center or carrier frequency is shifted by a
analog carrier is varied by the digital modulating binary input data.
pulse. The modulating signals is a binary pulse stream • The FSK output shifts between two
that varies between two discrete voltage levels rather frequencies: a mark or logic 1 frequency, a
than continuously changing analog signal as in space or logic 0 frequency.
conventional AM.

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• There is a change in the output frequency Note:

each time the logic conditions of the binary
That the minimum bandwidth required propagating a
input signal changes.
signal is called the minimum Nyquist bandwidth (𝐟𝐍 ).
• Bit rate and Baud rate are equal.
When modulation is used and the double-sided
General expression of an FSK Signal output spectrum is generated, the minimum
bandwidth is called the minimum double-sided
𝐟𝐦 (𝐭) 𝚫𝛚
𝐯 (𝐭) = 𝐕𝐜 𝐜𝐨𝐬 [(𝛚𝐜 ± )] Nyquist bandwidth or the minimum IF bandwidth.
𝟐
FSK Receiver
Where:
𝐯 (𝐭) = binary FSK waveform ( V ) Minimum Shift Keying (MSK) – also known as
𝐕𝐜 = peak unmodulated carrier amplitude ( V ) Continuous – Phase Frequency Shift Keying (CPFSK). It
𝛚𝐜 = radian carrier frequency is essentially a binary FSK except that the mark and
𝐟𝐦 (𝐭) = binary digital modulating signal frequency space frequencies are synchronized with the input
binary bit rate (𝐟𝐛 ).
Modulation Index ( 𝐦𝐟 )

𝐟𝐫𝐞𝐪𝐮𝐞𝐧𝐜𝐲 𝐝𝐞𝐯𝐢𝐚𝐭𝐢𝐨𝐧 𝛅
𝐦𝐟 = =
𝐦𝐨𝐝𝐮𝐥𝐚𝐭𝐢𝐧𝐠 𝐟𝐫𝐞𝐪𝐮𝐞𝐧𝐜𝐲 𝐟𝐚

𝐟𝐦 − 𝐟𝐬
| 𝟐 | = | 𝐟𝐦 − 𝐟𝐬 |
𝐦𝐟 =
𝐟𝐛 𝟐
𝟐
Note:
Where:
Synchronous simply means that there is a precise time
𝛅 = frequency deviation ( Hz )
𝐟𝐬 = space frequency ( Hz ) relationship between the two frequencies and input
𝐟𝐦 = mark frequency ( Hz ) binary bit rate.
𝐟𝐛 = input bit rate The mark and space frequencies are selected such that
| 𝐟𝐦 −𝐟𝐬 |
= peak frequency deviation they are separated from the center frequency by an
𝟐
𝐟𝐛 exact odd multiple of one half of the bit rate.
𝟐
= fundamental frequency of the binary input signal
Mark or Space Frequencies
Modulating Frequency or Fundamental Frequency
𝐟𝐛
( 𝐟𝐚 ) 𝐟𝐦 𝐚𝐧𝐝 𝐟𝐬 = 𝐧 ( )
𝐟𝐛 𝟐
𝐟𝐚 = Where:
𝟐
𝐧 = any odd integer
Frequency Deviation ( 𝛅 ) 𝐟𝐦 𝐚𝐧𝐝 𝐟𝐬 = mark and space frequencies

If fm > fs , If fm > fs ,
This ensures that there is a smooth phase transition in
𝐟𝐦 − 𝐟𝐬 𝐟𝐬 − 𝐟𝐦 the analog output signal when in changes from a mark
𝛅= 𝛅=
𝟐 𝟐 to a space frequency or vice versa.
Minimum Bandwidth Required ( 𝐟𝐍 )
MSK Waveform
𝐟𝐍 = 𝟐𝐧𝐟𝐚

Where:
𝒏 = no. of significant sidebands ( dependent on the
modulation index similar to conventional FM )
𝐟𝐚 = modulating frequency ( Hz )

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Advantage:

Has a better BIT-ERROR Performance than the BALANCED MODULATOR – acts like a phase
conventional binary FSK for a given signal-to-noise reversing switch. Depending on the logic
ratio. condition of the digital input, the carrier is
transferred to the output either in-phase or 180°
Disadvantage:
out-of-phase with reference to carrier oscillator.
It requires synchronizing circuit and is therefore
Balanced Ring Modulator
expensive.

Gaussian Minimum-Shift Keying ( GMSK ) – is used

in the GSM cellular radio and PCS systems. The word
Gaussian refers to the shape of the filter that is used
before the modulator to reduce the transmitted
bandwidth. GMSK uses less bandwidth than
conventional FSK.

Phase Shift Keying ( PSK ) – phase of the analog

carrier is varied by the digital modulating pulse. Bandwidth of BPSK Signal
Another form of angle modulated, constant envelope
digital modulation. Similar to Phase Modulation 𝐟𝐛
𝐟𝐍 = 𝟐 ( ) = 𝐟𝐛
except that the input is a binary digital signal. When 𝟐
higher data rates are required in band-limited Output Baud Rate
channels that are capable with FSK, phase-shift keying
𝐁𝐚𝐮𝐝 𝐫𝐚𝐭𝐞 = 𝐟𝐍 = 𝐟𝐛 = 𝐢𝐧𝐩𝐮𝐭 𝐛𝐢𝐭 𝐫𝐚𝐭𝐞
is often used. The phase of each signal is compared
with that of the previous signal rather a reference BPSK Waveform
signal. This type of PSK is called delta phase-shift
keying (DPSK).

Most DPSK modems use a four-phase system called

quadrature phase-shift keying (QPSK).

1. Binary PSK (BPSK)

• Two output are possible for a single
carrier frequency.
• Another term is Phase Reversal Keying
(PRK) Note that the output signal is either in phase (0°)
• One output phase represents a logic 1 with the reference oscillator or out-of-phase by
& the other a logic 0. 180°.
• A form of suppressed carrier, square-
Binary Input Output phase
wave modulation of a continuous
0 180°
wave (CW) signal.
1 0°
BPSK Transmitter

BSPK Transmitter

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2. Quaternary Phase Shift Keying (QPSK) QPSK Receiver

In QPSK, there are four possible output phases for

a single carrier frequency. The binary input data
for QPSK are combined into groups of 2 bits called
dibits before being acted by the modulator. For
every 2 bits at the input, a single output change is
phase angle is occurring thus the output rate of
change (baud rate) is one-half of the input bit rate
(bit rate).

QPSK Modulator
Offset QPSK (OQPSK) or OFFSET-KEYED QPSK
(OKPSK)

• Is a modified QPSK where the bit

waveforms on the I and Q channels are
offset or shifted in phase from each other
by ½ of a bit time.
• Used to avoid a shift or more than 90° as
well as to avoid a double bit range which
causes the output signal to shift by 180°
• ADVANTAGE: Limited phase shift that is
imparted during modulation
Minimum Bandwidth Required ( 𝐟𝐍 ) • DISADVANTAGE: Requires a bandwidth
twice that of conventional QPSK.
𝐟𝐛 𝐟𝐛 3. Eight-Phase PSK (8-PSK)
𝐟𝐍 = 𝟐 ( )=
𝟒 𝟐
8-PSK is an M-ary technique where M = 8. With
Output Baud Rate an 8-PSK modulator, there are eight possible
𝐟𝐛 output phases. To encode eight different phases,
𝐎𝐮𝐭𝐩𝐮𝐭 𝐛𝐚𝐮𝐝 𝐫𝐚𝐭𝐞 = 𝐟𝐍 =
𝟐 the incoming bits are considered in groups of 3
bits called tribits.
M-ary Encoding ( For QPSK, 𝐌 = 𝟒 )
Minimum Bandwidth Required ( 𝐟𝐍 )
𝐍 = 𝐥𝐨𝐠 𝟐 𝐌 = 𝐥𝐨𝐠 𝟐 𝟒 = 𝟐 𝐛𝐢𝐭𝐬
𝐟𝐛 𝐟𝐛
𝐟𝐍 = 𝟐 ( )=
𝟔 𝟑
QPSK Waveform Output Baud Rate

𝐟𝐛
𝐎𝐮𝐭𝐩𝐮𝐭 𝐛𝐚𝐮𝐝 𝐫𝐚𝐭𝐞 = 𝐟𝐍 =
𝟑

M-ary Encoding (For 8-PSK, M=8)

𝐍 = 𝐥𝐨𝐠 𝟐 𝐌 = 𝐥𝐨𝐠 𝟐 𝟖 = 𝟑 𝐛𝐢𝐭𝐬

8-PSK Waveform

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8-PSK Modulator

Differential Binary PSK ( DBPSK )

DBPSK is an alternative form of digital modulation

where the binary input information is contained in the
difference between two successive signaling elements
rather than the absolute phase. The difference in the
phase of the two signaling elements determines the
logic condition of the data.

Quadrature Amplitude Modulation (QAM) – is a

form of digital modulation where the digital
information is contained in both the amplitude and
phase of the transmitted carrier hence a combination
8-PSK Receiver of ASK and PSK techniques. It is considered a hybrid
digital modulation scheme since it combines ASK and
PSK methods.

Eight QAM (8-QAM)

8-QAM is an M-ary encoding technique where M=8.

Unlike 8-PSK, the output of an 8-QAM modulator is
not a constant-amplitude signal.

Minimum Bandwidth Required ( 𝒇𝑵 )

4. Sixteen-Phase PSK ( 16-PSK )
𝐟𝐛 𝐟𝐛
𝐟𝐍 = 𝟐 ( )=
16-PSK is an M-ary encoding technique where M = 𝟔 𝟑
16. A 16-PSK modulator acts on the incoming data in
groups of 4 bits called quadbits, that is, for every four Output Baud Rate
bits at the input there is one output produced. 𝐟𝐛
𝐎𝐮𝐭𝐩𝐮𝐭 𝐛𝐚𝐮𝐝 𝐫𝐚𝐭𝐞 = 𝐟𝐍 =
𝟑
Bandwidth Required ( 𝒇𝑵 )
M-ary Encoding (For 8-QAM, M=8)
𝐟𝐛 𝐟𝐛
𝐟𝐍 = 𝟐 ( ) =
𝟖 𝟒 𝐍 = 𝐥𝐨𝐠 𝟐 𝐌 = 𝐥𝐨𝐠 𝟐 𝟖 = 𝟑 𝐛𝐢𝐭𝐬

Output Baud Rate

𝐟𝐛
𝐎𝐮𝐭𝐩𝐮𝐭 𝐛𝐚𝐮𝐝 𝐫𝐚𝐭𝐞 = 𝐟𝐍 =
𝟒

M-ary Encoding (For 16-PSK, M=16)

𝐍 = 𝐥𝐨𝐠 𝟐 𝐌 = 𝐥𝐨𝐠 𝟐 𝟏𝟔 = 𝟒 𝐛𝐢𝐭𝐬

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Sixteen QAM (16-QAM) suppressed carrier systems, methods of carrier

recovery are required.
16-QAM is an M-ary encoding technique where M =
16. The input data are acted by the modulator in Three Methods of Carrier Recovery
groups of four bits called quadbits similar to 16-PSK.
1. Squaring Loop is a common method of
Minimum Bandwidth Required ( 𝒇𝑵 ) achieving carrier recovery for BPSK.
2. Costas Loop is also called quadrature loop.
𝐟𝐛 𝐟𝐛
𝐟𝐍 = 𝟐 ( )= 3. Remodulator has the fastest acquisition time.
𝟖 𝟒
Output Baud Rate C. DIGITAL TRANSMISSION
𝐟𝐛 In Digital transmission, the information (either analog
𝐎𝐮𝐭𝐩𝐮𝐭 𝐛𝐚𝐮𝐝 𝐫𝐚𝐭𝐞 = 𝐟𝐍 =
𝟒 or digital originally) is transmitted digitally. The
following are some of these advantages.
M-ary Encoding ( For 6-QAM, M = 16 )

𝐍 = 𝐥𝐨𝐠 𝟐 𝐌 = 𝐥𝐨𝐠 𝟐 𝟏𝟔 = 𝟒 𝐛𝐢𝐭𝐬 - Noise immunity

- Ease of processing and multiplexing
- Simpler to measure and evaluate
- The use of signal regenerators rather than
signal amplifiers.

Some of the disadvantages of digital transmission

are:

- Greater bandwidth required.

- The need for ADC (transmitter side) and DAC
(receiver side) in case original information is
analog.
- Precise timing or synchronization is strictly
Bandwidth Efficiency required.

Bandwidth efficiency is the ratio of the transmission bit Pulse Modulation

rate to the minimum bandwidth required for a
Pulse modulation is used in digital transmission. Pulse
particular modulation scheme. It is used to compare
modulation is divided into two fields: Analog Pulse
the performance of one digital modulation technique
Modulation and Digital Pulse Modulation. Examples of
to the other. It is otherwise known as information
analog pulse modulation are:
density.
- Pulse Width Modulation (PWM) or Pulse
𝐭𝐫𝐚𝐧𝐬𝐦𝐢𝐬𝐬𝐢𝐨𝐧 𝐫𝐚𝐭𝐞 ( 𝐛𝐩𝐬 ) Length Modulation (PLM) or Pulse Duration
𝐁𝐖 𝐞𝐟𝐟𝐢𝐜𝐢𝐞𝐧𝐜𝐲 =
𝐦𝐢𝐧𝐢𝐦𝐮𝐦 𝐛𝐚𝐧𝐝𝐰𝐢𝐝𝐭𝐡 ( 𝐇𝐳 ) Modulation (PDM)
𝐟𝐛 - Pulse Position Modulation (PPM)
𝐁𝐖 𝐞𝐟𝐟𝐢𝐜𝐢𝐞𝐧𝐜𝐲 = ( 𝒃𝒊𝒕𝒔 / 𝒄𝒚𝒄𝒍𝒆 ) - Pulse Amplitude Modulation (PAM)
𝐟𝐍
- Pulse Frequency Modulation (PFM)
CARRIER RECOVERY

• Is the process of extracting a phase-coherent

reference carrier from a receiver signal.
• Also called phase referencing
• With PSK and QAM, the carrier is suppressed
in the balanced modulators and is not
transmitted. At the receiver, the carrier cannot
be simply tracked with standard PLL. With this

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For digital pulse modulation, the two predominant amplitude of the analog modulating signal. PAM is an
methods (with some of their variations) are: intermediate form of modulation used in either PSK,
QAM and PCM systems. Its recent designation is K3E
- Pulse Code Modulation (PCM)
(formerly P3D).
- Differential PCM (DPCM)
- Delta Modulation (DM) PAM Signal Representation
- Adaptive Delta Modulation (ADM)

Pulse Width Modulation ( PWM )

In PWM, the pulse width (active portion of the duty

cycle) is proportional to the amplitude of the analog
modulating signal. The recent designation of PWM is
L3E (formerly P3E).

PWM Signal Representation

Pulse Frequency Modulation (PFM)

Pulse frequency modulation is a pulse modulation

system in which the amplitude of the pulse is kept
constant while its period and duration are made
proportional to the modulating signal so that the duty
cycle of the pulse train remains constant. Its
designation is V3E.

PFM Signal Representation

Pulse Position Modulation (PPM)

In PPM, the position of a constant-width pulse within

a prescribed time slow is varied in accordance with the
amplitude of the analog modulating signal. The recent
designation of PPM is M3E (formerly P3F).

PPM Signal Representation Pulse Code Modulation (PCM)

In PCM, the analog signal is sampled and converted

to pulses of constant length and encoded for serial
binary transmission. The binary number varies in
accordance with the amplitude of the analog
modulating signal. Its designation is P3G.

The PCM System

Pulse Amplitude Modulation (PAM)

In PAM, the amplitude of a constant-width, constant-

position pulse is varied in accordance with the

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The Nyquist Sampling Theorem is the ratio of the minimum number of bits required to
achieve a certain dynamic range to the actual number
The Nyquist sampling theorem establishes the
of bits in the PCM code.
minimum sampling rate of PCM systems. It states that
for an analog signal to be recovered and reproduced 𝐥𝐨𝐠 ( 𝐃𝐑 + 𝟏 )
𝐂𝐨𝐝𝐢𝐧𝐠 𝐞𝐟𝐟𝐢𝐜𝐢𝐞𝐧𝐜𝐲 = 𝒙 𝟏𝟎𝟎%
accurately at the receiver, it must be sampled at least 𝐚𝐜𝐭𝐮𝐚𝐥 𝐧𝐨. 𝐨𝐟 𝐛𝐢𝐭𝐬
twice its highest frequency. This is the minimum
The Signal-to-Quantization Noise Ratio
sampling rate.
Where: The signal-to-quantization noise ratio ( SQR ) or just
𝐟𝐬 = 𝟐𝐟𝐚 𝐟𝐬 = Nyquist Sample Rate signal-to-noise ratio ( SNR ) is the ratio of the wanted
𝐟𝐚 = highest frequency to be sampled signal to the unwanted noise.
𝐌𝐢𝐧𝐢𝐦𝐮𝐦 𝐯𝐨𝐥𝐭𝐚𝐠𝐞
If the sampling rate 𝐟𝐬 is less than 𝐟𝐚 , distortion results. 𝐒𝐐𝐑 =
𝐐𝐮𝐚𝐧𝐭𝐢𝐳𝐚𝐭𝐢𝐨𝐧 𝐧𝐨𝐢𝐬𝐞 𝐯𝐨𝐥𝐭𝐚𝐠𝐞
The distortion is called aliasing or foldover distortion.
In practice, the actual sampling rate is made higher or
This is known as the “ worst-case SQR “
greater than the minimum 𝐟𝐬 .
𝐌𝐚𝐱𝐢𝐦𝐮𝐦 𝐯𝐨𝐥𝐭𝐚𝐠𝐞
𝐟𝐬 > 𝟐𝐟𝐚 𝐒𝐐𝐑 =
𝐐𝐮𝐚𝐧𝐭𝐢𝐳𝐚𝐭𝐢𝐨𝐧 𝐧𝐨𝐢𝐬𝐞 𝐯𝐨𝐥𝐭𝐚𝐠𝐞
Dynamic Range (DR)
𝐯𝟐
𝐒𝐐𝐑 (𝐝𝐁) = 𝟏𝟎𝐥𝐨𝐠 𝐑
The dynamic range is the ratio of the largest possible
𝐪𝟐
magnitude to the smallest possible magnitude that (𝟏𝟐) /𝐑
can be decoded by the digital to analog converter. Where:
𝐕𝐦𝐚𝐱 𝐕𝐦𝐚𝐱 𝐑 = resistance ( Ω )
𝐃𝐑 = =
𝐕𝐦𝐢𝐧 𝐫𝐞𝐬𝐨𝐥𝐮𝐭𝐢𝐨𝐧 𝐯 = rms signal voltage ( V )
𝐪 = quantization interval ( V )
𝐃𝐑 (𝐝𝐁) = 𝟐𝟎𝐥𝐨𝐠𝐃𝐑 𝐯𝟐
= rms signal power
𝐑
𝟐
𝐪 /𝟏𝟐
Where: = average rms quantization noise power
𝐑
𝐕𝐦𝐚𝐱 = largest possible magnitude that can be decoded
𝐕𝐦𝐢𝐧 = smallest possible magnitude that can be decoded 𝐯𝟐 𝐯
𝐒𝐐𝐑 (𝐝𝐁) = 𝟏𝟎𝐥𝐨𝐠 𝟐
= 𝟏𝟎. 𝟖 + 𝟐𝟎 𝐥𝐨𝐠
𝐪 𝐪
PCM Code Number of Bits ( )
𝟏𝟐

The PCM code is sign-magnitude code where the

Other valid expressions
most significant bit (MSB) is the sign bit and the
remaining bits are magnitude bits. It is sometimes 𝐒 𝐒 𝟑
𝐒𝐐𝐑 (𝐝𝐁) = = = 𝐌𝟐
called a folded binary code except for the sign bit. 𝐃 𝐍 𝟐
The mathematical relation between the numbers of 𝐒
(𝐝𝐁) = 𝟔𝐧 + 𝟏. 𝟖
bits required in the PCM code and the dynamic range 𝐍 ( For uniform quantizing)
is given by the expression:
Where:
𝟐𝐧 − 𝟏 ≥ 𝐃𝐑 𝐒/𝐃 = signal-to-distortion ratio
𝐒/𝐍 = signal-to-noise ratio
For minimum number of bits, 𝐌 = number of quantizing levels
𝐥𝐨𝐠 ( 𝐃𝐑 + 𝟏 ) 𝐧 = number of bits
𝟐𝐧 − 𝟏 = 𝐃𝐑 𝐧=
𝐥𝐨𝐠 𝟐
The PCM Coding Methods
Where:
𝐧 = no. of PCM bits, excluding sign bit In quantizing PAM signals into 2n levels several
𝐃𝐑 = absolute value of Dynamic Range ( not dB value ) methods are used. These are:

The Coding Efficiency - Level-at-a-time coding

- Digit-at-a-time coding
The coding efficiency is defined as the numerical
- Word-at-a-time coding
indication of how efficiently a PCM code is utilized. It
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Level-at-a-time Coding A-Law Companding

In level-at-a-time coding, the PAM signal is In A-law companding, the compression characteristic
compared to a ramp waveform while a binary counter is given by
is being advanced to a uniform rate. When the ramp
𝐀 𝐕𝐢𝐧 / 𝐕𝐦𝐚𝐱 𝐕𝐢𝐧 𝟏
waveform equals or exceeds the PAM sample, the 𝐕𝐨𝐮𝐭 = 𝐕𝐦𝐚𝐱 ;𝟎 ≤ ≤
𝟏 + 𝐥𝐧 𝐀 𝐕𝐦𝐚𝐱 𝐀
counter contains the PCM code.

Digit-at-a-time Coding 𝟏 + 𝐥𝐧 ( 𝐀 𝐕𝐢𝐧 / 𝐕𝐦𝐚𝐱 ) 𝟏 𝐕𝐢𝐧

𝐕𝐨𝐮𝐭 = 𝐕𝐦𝐚𝐱 ; ≤ ≤ 𝟏
In digit-at-a-time coding, each digit of the PCM code 𝟏 + 𝐥𝐧 𝐀 𝐀 𝐕𝐦𝐚𝐱
is determined sequentially. It is similar to a balance
DIGITAL COMPANDING
where known reference weights are used to determine
an unknown weight. In digital companding, the compression is done after
the input sample is converted to a linear PCM code
Word-at-a-time Coding
(transmitted side) and the expansion is done before
These coders are flash encoders and are more PCM encoding (receiver side).
complicated than earlier coders. They are useful for
Digitally companded PCM system
high-speed applications.

The Companding Methods

Companding in general as term implies, is the process

of compressing and expanding. This process helps the
signal and minimizes the effect of noise. There are two
prevalent methods of companding, namely:
Differential Pulse Code Modulation ( DPCM )
- Analog companding
DPCM solves the problem of redundancy (resulting to
- Digital companding
more bits transmitted) in a typical PCM-encoded
ANALOG COMPANDING speech waveform. It is designed specifically to take
advantage of the sample-to-sample redundancies.
Analog companding was popular in older PCM
In DPCM, the difference in the amplitude of two
systems. It was implemented using specially designed
successive samples is transmitted rather than the
diodes. The analog compressor is inserted between
actual sample thus fewer bits are required to be
bandpass filter and sample-and-hold circuit in the
transmitted since the range of sample differences is
transmitter. The two popular analog companding
less than range of individual samples.
methods in use are:
Delta Modulation ( DM )
- µ-Law companding (used in US and Japan)
- A-Law companding (used in Europe Delta modulation utilizes a single-bit PCM code. It
established by ITU-T) uses single bit PCM code which simply indicates + or
– in magnitude.
µ-Law Companding
Problems of Delta Modulator
In µ-law companding, the compression characteristic
is given by Slope overload
𝐕
𝐕𝐦𝐚𝐱 𝐱 𝐥𝐧 ( 𝟏 + µ 𝐕 𝐢𝐧 ) When the slope of the analog signal is greater than
𝐦𝐚𝐱
𝐕𝐨𝐮𝐭 = the delta modulator can keep, slope overload occurs.
𝐥𝐧 ( 𝟏 + µ )
This problem can be remedied by increasing the clock
Where: frequency and the magnitude of the minimum step
𝐕𝐦𝐚𝐱 = max. uncompressed analog input amplitude size.
𝐕𝐢𝐧 = amplitude of input signal at a particular instant of time

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Granular noise

Granular noise in delta modulation is similar to

quantization noise in PCM system. This occurs when
the reconstructed signal has variations that were not
present in the original signal due to the relatively
constant amplitude of original analog input. This
problem is minimized by decreasing the step size.

Adaptive Delta Modulation (ADM)

ADM is a delta modulation system where the step size

of the DAC is automatically varied depending on the
amplitude characteristics of the analog input signal.
This system is designed to more or less solve the
inherent problems of the conventional delta
modulation.

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