TRANSMISSION OF INFORMATION

David Falconer and Halim Yanikomeroglu

Systems and Computer Engineering

Carleton University

1
 Topics to be Covered

l Analog and digital signals

l Power spectral density and bandwidth

l Analog to digital conversion (ADC):

l PCM (pulse code modulation)

l Digital transmission

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 Analog Signal

An analog signal is any continuous signal for which the time varying feature (variable) of the
signal is a representation of some other time varying quantity. For example, in an analog audio
signal, the instantaneous voltage of the signal varies continuously with the pressure of the
sound waves.

Analog signal differs from a digital signal, in which the continuous quantity is a representation
of a sequence of discrete values which can only take on one of a finite number of values.
[Source: Wikipedia & Google images]

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 Digital Signal

A digital signal is a signal that represents a sequence of discrete values at clock times
(discrete in amplitude & discrete in time)
[Source: Wikipedia & Google images]

4
 Detection of Analog and Digital Signals

Digital signal + noise analog signal + noise

[Source: Google images]

Fundamental question: Is detection easier in digital signaling or in analog signaling?

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 Digital and Analog Signals

Some signals (like speech and video) are inherently analog; some
(like computer data) are inherently digital.

However, both analog and digital signals can be represented and

transmitted digitally.

Advantages of digital:
» Reduced sensitivity to line noise, temp. drift, etc.
» Low cost digital VLSI for switching and transmission.
» Lower maintenance costs than analog.
» Uniformity in carrying voice, SMS, email, data, video, etc. (a bit is a bit).
» Better encryption.

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 Power Spectral Density

Power spectrum (power spectral density) describes how the

average power is distributed with respect to frequency.

Deterministic signals  Fourier transform

Random signals  Power spectral density

A statistical representation for all random signals in a particular
application

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 Power Spectrum of Analog Signals

Analog (continuous-time, continuous-amplitude) signals (like

speech) have a certain bandwidth. Their power spectrum (power
spectral density) describes how their average power is
distributed with respect to frequency.

Power
spectral
density “High-fidelity speech
(watts/Hz)

Bandwidth Telephone speech

(limited by filtering)
0 1 2 3 4 5 6 7....

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Power Spectrum of Analog Signals

Source: Wikipedia

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Power Spectrum of Digital Signals

Source: Wikipedia

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 Bandwidth

For random signals, bandwidth is determined from the power

spectral density.

Bandwidth is determined only from the +ve frequencies.

There are different bandwidth definitions

Absolute bandwidth
Y% bandwidth (for instance, 99%)
X-dB bandwidth (for instance, 3-dB)
Null-to-null bandwidth
…

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 Bandwidth

3-dB Bandwidth

Source: Google images

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Bandwidth

Digital Communications, B. Sklar

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Bandwidth

Digital Communications, B. Sklar 14

Bandwidth

Digital Communications, B. Sklar

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Bandwidth

Digital Communications, B. Sklar 16

 Sampling an Analog Signal

Sampling theorem: The original analog signal can be reconstructed if

it is sampled at a rate at least twice its bandwidth.

Reconstruction is by filtering samples with a low pass filter.

Sampling Samples Reconstruction

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 Pulse-Code Modulation (PCM)

 PCM is a method used to digitally represent sampled analog signals. It

is the standard form of digital audio in computers, Compact
Discs, digital telephony and other digital audio applications.

 PCM signal is developed by three steps: sampling, quantizing and

encoding.

 Quantizing noise is reduced by using variable sized steps. It is

independent of line length.

s(t) s(n) 011010001...

Filter

Sample at t=n Quantize Encode

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 Pulse-Code Modulation (PCM)

Sampling and
quantization of
a signal (red)
for 4-bit PCM

l The PCM process is commonly implemented on a single

integrated circuit and is generally referred to as an analog-to-
digital converter (ADC)
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 Standard PCM in Wired Telephony

• Voice circuit bandwidth is 3400 Hz.

• Sampling rate is 8 KHz (samples are 125 s apart).

• Each sample is quantized to one of 256 levels.

• Each quantized sample is coded into a 8-bit word.

• The 8-bit words are transmitted serially (one bit at a time) over a
digital transmission channel. The bit rate is 8x8,000 = 64 Kb/s.

• The bits are regenerated at digital repeaters.

• The received words are decoded back to quantized samples,

and filtered to reconstruct the analog signal.

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 Quantization

Uniform (Linear PCM: LPCM) Nonuniform

Output signal Output signal

Input signal Input signal

The more steps (levels) the less quantization noise. Nonuniform quantization
(e.g. -law) allows a larger dynamic range (important for speech).

LPMC: Uncompressed
Nonuniform quantization: Introduces compression
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 -Law Quantization and Coding

• Standardized in North America.

• Based on a logarithmic non-uniform quantizer.

• Range of amplitudes divided into 8 segments, each segment

with 16 uniformly spaced levels. Segment i is double the width
of segment i-1.

• 8 bit word: 1 bit for sign, 3 bits identify segment, 4 bits identify
level within segment.

• Can show for n-bit word, signal to quantization noise ratio is

approximately 6n-10 [dB]; e.g., 38 dB for n=8 bits.

• Most of the rest of the world uses a related logarithmic non-

uniformity, called A-law.
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 Variants of PCM (Form of Compression)

Differential PCM (DPCM) encodes the PCM values as differences

between the current and the predicted value. An algorithm
predicts the next sample based on the previous samples, and
the encoder stores only the difference between this prediction
and the actual value. If the prediction is reasonable, fewer bits
can be used to represent the same information. For audio, this
type of encoding reduces the number of bits required per
sample by about 25% compared to PCM.

Adaptive DPCM (ADPCM) is a variant of DPCM that varies the

size of the quantization step, to allow further reduction of the
required bandwidth for a given signal-to-noise ratio.

Delta Modulation is a form of DPCM which uses one bit per

sample.
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 Adaptive Differential PCM (ADPCM)

Allows coding with a lower bit rate (with same fidelity) for speech,
based on predicting the next sample; e.g., 8 or 16 or 32 Kb/s.
More circuits accommodated in the same transmission bandwidth.

Coder: Decoder:

+ Quant. +

Predictor
Predictor

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 PCM Standards

G.711 is an ITU-T standard for audio companding. It is primarily used in

telephony. The standard was released for usage in 1972. Its formal
name is Pulse Code Modulation (PCM) of voice frequencies.
G.711 uses a sampling rate of 8,000 samples per second. Non-uniform
(logarithmic) quantization with 8 bits is used to represent each sample,
resulting in a 64 kbit/s bit rate.
G.711.1 is an extension to G.711, published as ITU-T Recommendation
G.711.1 in March 2008. Its formal name is Wideband embedded
extension for G.711 pulse code modulation.
G.711.1, allows the addition of narrowband and/or wideband (16000
samples/s) enhancements, each at 25 % of the bitrate of the (included)
base G.711 bitstream, leading to data rates of 64, 80 or 96 kbit/s.
G.711.1 is compatible with G.711 at 64 kbit/s, hence an efficient
deployment in existing G.711-based voice over IP (VoIP)
infrastructures is foreseen.

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 PCM Standards

G.726 is an ITU-T ADPCM speech codec standard covering the

transmission of voice at rates of 16, 24, 32, and 40 kbit/s (1990).
The most commonly used mode is 32 kbit/s, which doubles the
usable network capacity by using half the rate of G.711. It is
primarily used on international trunks in the phone network. The
principal application of 24 and 16 kbit/s channels is for overload
channels carrying voice in digital circuit multiplication equipment
(DCME).
It also is the standard codec used in DECT wireless phone systems
and is used on some Canon cameras.
Sampling frequency 8 kHz. 16 kbit/s, 24 kbit/s, 32 kbit/s, 40 kbit/s
bit rates available.
Testing under ideal conditions yields Mean Opinion Scores of 4.30
for G.726 (32 kbit/s), compared to 4.45 for G.711 (µ-law)

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 PCM Standards

Audio Interchange File Format (AIFF) is an audio file format

standard used for storing sound data for personal computers
and other electronic audio devices.
The audio data in a standard AIFF file is uncompressed pulse-code
modulation (PCM). There is also a compressed variant of AIFF
known as AIFF-C or AIFC, with various defined compression
codecs.
Like any non-compressed, lossless format, it uses much more disk
space than MP3—about 10MB for one minute of stereo audio at
a sample rate of 44.1 kHz and a sample size of 16 bits.
Developed by Apple Inc. Initial release 21 January 1988.

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 PCM Standards

MPEG-1 or MPEG-2 Audio Layer III, more commonly referred to as MP3, is a patented
digital audio encoding format using a form of lossy data compression. It is a common
audio format for consumer audio storage, as well as a de facto standard of digital
audio compression for the transfer and playback of music on digital audio players.
Initial release: 1993.
MP3 is an audio-specific format that was designed by the Moving Picture Experts Group
(MPEG) as part of its MPEG-1 standard and later extended in MPEG-2.
The use in MP3 of a lossy compression algorithm is designed to greatly reduce the
amount of data required to represent the audio recording and still sound like a
faithful reproduction of the original uncompressed audio for most listeners. An MP3
file that is created using the setting of 128 kbit/s will result in a file that is about 1/11
the size of the CD file created from the original audio source. An MP3 file can also
be constructed at higher or lower bit rates, with higher or lower resulting quality.
The compression works by reducing accuracy of certain parts of sound that are
considered to be beyond the auditory resolution ability of most people. This method
is commonly referred to as perceptual coding. It uses psychoacoustic models to
discard or reduce precision of components less audible to human hearing, and then
records the remaining information in an efficient manner.

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 PCM Standards

MPEG-1 Audio Layer III: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224,
256, and 320 kbit/s, and the available sampling frequencies are 32, 44.1
(CD) and 48 kHz (DVD). Additional extensions were defined in MPEG-2
Audio Layer III: bit rates 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144,
160 kbit/s and sampling frequencies 16, 22.05, and 24 kHz.
A sample rate of 44.1 kHz is almost always used, because this is also used for
CD audio, the main source used for creating MP3 files. A greater variety of
bit rates are used on the Internet. The rate of 128 kbit/s is commonly used,
at a compression ratio of 11:1, offering adequate audio quality in a relatively
small space. As Internet bandwidth availability and hard drive sizes have
increased, higher bit rates up to 320 kbit/s are widespread.
Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s,
so the bitrates 128, 160, and 192 kbit/s represent compression ratios of
approximately 11:1, 9:1, and 7:1 respectively.
The bitrate limit for LPCM audio on DVD-Video is also 6.144 Mbit/s, allowing 8
channels (7.1 surround) × 48 kHz × 16-bit per sample = 6,144 kbit/s

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 T1 (DS1) Format (North America)

193 bits in 125 s

S bit (1.544 Mb/s)

24 PCM code words, each representing 1 sample

DS1 1 2 3 4 24

8 bits per code word

DS0 1 2 3 4 5 6 7 8

PCM T-Carrier Multiplexing Hierarchy

Digital Signal Designation Line rate Channels (DS0s) Line
DS0 64 kbit/s 1
DS1 1.544 Mbit/s 24 T1
DS1C 3.152 Mbit/s 48 T1C
DS2 6.312 Mbit/s 96 T2
DS3 44.736 Mbit/s 672 T3
DS4 274.176 Mbit/s 4032 T4
DS5 400.352 Mbit/s 5760 T5
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 Regenerative Repeater

Regenerative Regenerative
repeater repeater

Amplifier/ Regenerator
equalizer

Structure of a regenerative
repeater:
Timing circuit

By appropriate repeater design and inter-repeater spacing, the effect of

occasional bit errors due to noise can be controlled. Received signal quality is
essentially independent of distance.

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PCM Transmission Formats and Spectra
Time Frequency
..... 1 0 1 1 ....... Power spectra
Unipolar NRZ

0 T 2T 3T 4T -3/T -2/T -1/T 0 1/T 2/T 3/T

Bipolar NRZ

0 T 2T 3T 4T -4/T -2/T -1/T 0 1/T 2/T 4/T

 Unipolar RZ

0 T 2T 3T -4/T -1/ -2/T -1/T 0 1/T 2/T 1/ 4/T

Bandlimited Min. bandwidth

0 T 2T 3T 4T -1/2T 1/2T
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 Multilevel Transmission

1 0 1 1 0 0 0 1

Binary:
L=2

4-level:
L=4
0 T 2T 3T 4T

1
Bit rate = log2 L
T
Bandwidth proportional to 1/T for NRZ signals

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 Additive White Gaussian Noise (AWGN)

AWGN is a channel model in which the only impairment to

communication is noise.

AWGN: A linear addition of white noise with a constant spectral

density and a Gaussian distribution of amplitude. [Wiki]

The model does not account for channel impairments. However, it

produces simple and tractable mathematical models which are useful
for gaining insight into the underlying behavior of a system before
these other phenomena are considered. [Wiki]

Gaussian noise: Noise amplitude is a Gaussian distributed random

variable (central limit theorem).

White noise: An idealized noise process with a power spectral density

independent of frequency.
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 Additive White Gaussian Noise (AWGN)

lPnoise= k T B F = N0 B F

k: Boltzmann’s constant = 1.38 x 10-23 J/K

T: Temperature in degrees Kelvin (generally taken as 290oK)
N 0: Noise power spectral density (constant)
B: Bandwidth (signal bandwidth)
F: Noise figure
White noise power spectral density
SN(f)
N0 = k T = -174 dBm/Hz
N0/2
Ex: 200 KHz channel (LTE resource block)
F = 7 dB  Pnoise = -114 dBm f
Infinite total power (?)
Broadband signal  Pnoise increases
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 Bandwidth Required for Digital Transmission

l required bandwidth is approximately

(bit rate)/(log2L) for L-level transmission.

l more levels  less bandwidth, but greater sensitivity to noise.

l Examples:
» 64 Kb/s PCM requires about 64 KHz for binary transmission, 32 KHz for 4-
level transmission.
» 14.4 Kb/s modem uses a symbol rate 1/T=2400 Hz, and the equivalent of
L=32.

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 Channel Capacity

Shannon channel capacity formula:

» Highest possible transmission bit rate R, for reliable communication in a
given bandwidth W Hz, with given signal to noise ratio, SNR, is

R=Wlog2(1+SNR) bits/s
R/W = 0.332 SNR [dB] bits/s/Hz (for high SNR)

» Assumptions and qualifications:

– Gaussian distributed noise added to the signal by the channel, highly complex
modulation, coding and decoding methods.
– In typical practical situations, the above formula may be roughly modified by
dividing SNR by a factor of about 5 to 10.

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 Fundamental Limits in Digital Data Rates

5G

4G Mobile device
? for everything

Gbps 3G

Mbps 2G

Kbps 1G

bps AMPS

1980 1990 2000 2010 2020 lTime

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 Information Theory and Digital Communications

Ralph V.L. Hartley Harry Nyquist Norbert Wiener Claude Shannon

1888 – 1970 1889 – 1976 1894 – 1964 1916 – 2001

Emre Telatar Gerard J. Foschini

1964 – 1940 – 39
 Fundamental Limits in Digital Data Rates

RBS: Data rate (speed) of a wireless base station (access point)

•W: Bandwidth
•SNR: Signal-to-noise ratio at the receiver
•SE: Spectral efficiency = log2(1+SNR)
•n: Min (# of transmit antennas, # of receive antennas)

RBS = n x W x SE = n x W x log2(1+SNR)
None of the three variables (W, SE, n) scales well!

Ex 1: n = 2, W = 10 MHz, log(1+SNR) = 4  RBS = 80 Mbps

Ex 2: n = 8, W = 100 MHz, log(1+SNR) = 4.5  RBS = 3.6 Gbps

(Cellular 4th generation LTE-Advanced)

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 Fundamental Limits in Digital Data Rates

Rnetwork: Network rate

Rnetwork = K x n x W x log2(1+SNR)
•K: # of BSs in the network

Fundamental dynamics:
4 basic factors that impact network rate: K, n, W, SE

Increasing base station rate: Not easy! (neither of n, W, SE scales well)

Increasing network rate: Possible! (by adding more base stations)

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 Summary

l All information signals can be represented, switched, stored and

transmitted digitally.

l We have discussed PCM systems and their key elements:

» sampling
» quantizing
» coding
» digital transmission

l We have discussed the related concepts of:

» power spectrum density
» bandwidth
» multiplexing
» channel capacity

• We have discussed the fundamental enablers of transmission rate.

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