digital communication basic

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Digital Communication

Introduction

In Figure 1 an overview of the typical functional elements of a digital communication system are shown.

Figure 1: Digital Communication System

The information source generates particular symbols at a particular rate. The source encoder translates these

symbols in sequences of 0's and 1's. The channel encoder is oriented towards translating sequences of 0's

and 1's to other sequences of 0's and 1's, to realize high transmission reliability and efficiency. The modulator

accepts streams of 0's and 1's, and converts them to electrical waveforms suitable for transmission.

The communication channel provides the electrical connection between the source and destination. It has a

finite bandwidth, and the waveform transmitted suffers from amplitude distortion and phase distortion. In

addition to distortion, power is decreased due to attenuation of the channel. Finally, the waveform is

corrupted by unwanted electrical signals, referred to as noise. The primary objective of a communication

system is to suppress the bad effects of noise as much as possible.

The inverse process takes place at the destination side. The demodulator converts the electrical waveforms to

sequences of 0's and 1's, the channel decoder translates the sequence of 0's and 1's to the original sequence

of 0's and 1's. It also performs error correction and clock recovery. The source decoder finally translates the

sequence of 0's and 1's into symbols.

The gray boxes in the following figure show how the elements are grouped in a digital audio configuration, in

this case a CD player and a DA-convertor.

Information sourceThe main digital source for consumer purpose is the CD-player. This section will present a short summary

about the history and principles of a CD-player. It is derived from a document of Grant M. Erickson, e.g. see

A Fundamental Introduction to the Compact Disc Player.
http://www.tc.umn.edu/~erick205/Papers/paper.html


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The history of the CD player

As staggering as the release of the compact disc player was in 1982, the technology and theories which

allowed it to be born were long in development. In 1841, the great mathematician Augustin-Louis Cauchy

first proposes the sampling theorem. Nearly 80 years later J.R. Carson publishes a mathematical analysis of

time sampling in communications. In a 1928 lecture at the American Institute of Electrical Engineers Harry

Nyquist provides proof of the sampling theorem in "Certain Topics in Telegraph Transmission Theory". In

1937, A. Reeves proposes pulse code wave modulation (PCM). In 1948, John Bordeen, William Shockley,and Walter Brattain invent the bipolar junction transistor at Bell Labs--compact digital circuitry is a reality.

Two years later, in 1950 Richard W. Hamming publishes significant work on error correction and detection

codes. In 1958 C.H. Townes and A.L. Shawlow invent the laser. In 1960 R.C. Bose publishes binary group

error correction codes. That same year I.S. Reed and G. Solomon publish error correction codes to be used

in the CD player 22 years later. Also early computer music experiments take place at Bell Labs. Fifteen years

before consumers see the first player, NHK Technical Research Institute publicly demonstrates a PCM digital

audio recorder with a 30 kHz sampling rate and 12-bit resolution. Two years later, Sony Corporation

demonstrates a PCM digital audio recorder with a 47.25 kHz sampling rate and 13-bit resolution. A

hemisphere away, Dutch physicist Klaas Compaan uses a glass disc to store black and white holographicimages using frequency modulation at Philips Laboratories. Four years later, in 1973 Philips engineers begin

to contemplate an audio application for their "video" disc system. A prototype disc with a 44 kHz sampling

rate is run through a 14-bit digital-to-analog converter and exhibits a signal-to-noise (S/N) ratio of 80 dB in

monaural. Now a research frontier, Mitsubishi, Sony, and Hitachi all demonstrate digital audio discs at the

Tokyo Audio Fair in 1977. One year later, Philips joins with its recording subsidiary Polygram Records to

develop a worldwide digital audio standard. In March 1979, Philips demonstrates a prototype compact disc

player in Europe. Sony joins the Philips/Polygram coalition after Matsushita declines. In June of 1980, the

coalition formally proposes their CD standard. A year later in 1981, Sharp successfully mass produces the

semiconductor laser. This step was crucial to delivering a consumer product. In Fall of 1982 nearly 150 years

of work comes to fruition and Sony and Philips introduce their respective players to consumer in Europe. The

following spring, the player is introduced in the United States.

The specifications for the compact disc and compact disc players were jointly developed by Sony, Philips,

and Polygram as mentioned previously. This specification is contained in their standards document referred to

as the Red Book. Some of these specifications are summarized in Table 1, 2 and 3.

Table 1: Disc format

Playing time 74 minutes, 33 seconds maximum

Rotation 1.2 - 1.4 m/sec. (constant linear velocity)

Track pitch 1.6 m

Diameter 120 mm

Thickness 1.2 mm


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Center hole diameter 15 mm

Recording area 46 mm - 117 mm

Signal area 50 mm - 116 mm

Material Any acceptable medium with a refraction index of 1.55

Minimum pit length 0.833 m (1.2 m/sec) to 0.972 m (1.4 m/sec)

Maximum pit length 3.05 m (1.2 m/sec) to 3.56 m (1.4 m/sec)

Pit depth ~0.11 m

Pit width ~0.5 m

Table 2: Optical system

Standard wavelength 780 nm

Focal depth 2 m

Table 3: Signal format

Number of channels 2 channels (4 channel recording possible)

Quantization 16-bit linear


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Quantizing timing Concurrent for all channels

Sampling frequency 44.1 kHz

Channel bit rate 4.3218 Mb/sec

Data bit rate 2.0338 Mb/sec

Error correction code Cross Interleave Reed-Solomon Code (with 25% redundancy)

Modulation system Eight-to-fourteen Modulation (EFM)

The compact disc player contains two main subsystems: the audio data processing system and the

servo/control system. The servo, control, and display system orchestrate the mechanical operation of the

player and include such items as the spindle motor, auto-tracking, lens focus, and the user interface. The

audio data processing section covers all other player processes.

Source encoding/decoding

Digital data provided by digital sources like CD, DAB, DAT or DCC is transmitted by using the SPDIF

consumer standard, derived from the AES/EBU professional standard. Using this standard, every sample is

transmitted as a 32-bit word (called a subframe). Two subframes make one frame (64 bits total), which

repeats at the sampling rate in use. A block contains 192 frames. At 48 kHz the bit rate will be 3.072 MHz,

at 44.1 kHz the bit rate will be 2.8224 MHz. The first subframe will contain the sample from channel A, the

second frame will contain the sample from channel B.

The basic SPDIF and AES/EBU subframe structure can be found in the following figure:

Figure 2: SPDIF subframe structure

The first 4 bits in the preamble are used for synchronization. There are 3 different sync-patterns (called B, M,

W), but they can appear in different forms, depending on the value of the last cell of the previous subframe

(see the section about channel encoding/decoding for more information). Preamble B marks a wordcontaining data for channel A (left) at the start of a block. Preamble M marks a word with data for channel A

that isn't at the start of a block. Preamble W marks a word containing data for channel B (right, for stereo).


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Figure 3: Block structure

There is room for 24 audio bits (4-27) in a subframe, but normally 20 audio bits are available (8-27). In case

of CD only 16 bits are used. Bits that are not used, are defined to be zero in SPDIF (in contrary to for

instance I2S, where the LSB is repeated, resulting in less transitions in a serial interface, and hence less power

dissipation). As audio data must be in 2-complements code, and different word lengths may be used, the

MSB must always be at the same place. SPDIF sends the MSB first.

Four status bits (28-31) accompany each subframe. Bit 28 (V) is the validity flag, which indicates whether the

data received is suitable for conversion to an analog signal (e.g. to indicate non-audio data, like CD-I or CD-

ROM players, which can be used for muting the audio outputs). Bit 31, the parity bit (P), produces even

parity (total number of ones is even), which is meant for error correction at the destination. The user bit (bit

29, U) and channel bits (bit 20, C) are collected to form user and channel status blocks.

The sequence of channel bits over 192 subframes (or one block) builds up a 24 byte (192 bit) channel status

block. The preamble is used to denote the start of a channel status block (see the section about channel

encoding/decoding for more information). The structure of a channel status block is shown in the following

figure.

Figure 4: Channel status block structure

First, the serial data is assembled into 12 words of 16 bits each. The first 6 bits of the first word form a

control word. The following 2 bits permit a mode select for future expansion. Currently only mode 0 is

standardized (and the corresponding bit allocation shown in the figure).


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Control word: bit 0 0 = consumer, 1 = professional

bit 1 0 = normal, 1 = digital data

bit 2 0 = copy prohibit, 1 = copy permit

bit 3 0 = no pre-emphasis, 1 = pre-emphasis

bit 4 reserved

bit 5 0 = 2CH, 1 = 4CH

Mode: bit 6,7 00 = mode 0 (01 10 11 = reserved)

Category code: bit 8, 15 00000000 = general format

10000000 = CD player

11000000 = DAT player

bit 15 0 = original recording

1 = first-generation copy (SCMS)

Source no.: bit 16 - 19 0000 = don't care

0001 = source 1

0010 = source 2

...

Channel no.: bit 20 - 23 0000 = don't care

1000 = A (left channel for stereo)

0100 = B (right channel for stereo)

1100 = C

...

Sampling rate: bit 24 - 27 0000 = 44.1 kHz


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0100 = 48 kHz

1100 = 32 kHz

remaining codes reserved

Clock accuracy: bit 28, 29 00 = normal accuracy

10 = high accuracy

01 = variable speed

In case of a CD player (category code 10000000), the audio sample words in the subframe structure

consists of 16 bit words, MSB in position 8, bits 24 - 27 in the channel status block are 0000, indicating a

44.1 kHz sampling frequency, and bits 28,29 are set according to the source accuracy. The two channel

format (00000000) leaves room for a maximum of 20 bits/sample, and various sample frequencies.

W.r.t. possible sample rates and pre-emphasis the following observations can be made. CD uses 44.1kHz,

with the possibility of 50/15 s pre-emphasis. DAB uses 48 kHz, without pre-emphasis. DCC and DAT

support 32, 44.1, 48 kHz with the possibility for 50/15 s pre-emphasis. ISO/MPEG supports 6 sampling

rates (16, 22.05, 24, 32, 44.1, 48 kHz) with 2 possible pre-emphasis-types (not simultaneously): 50/15 s

and the so-called CCITT J.17. CD-i only uses 44.1 kHz (also for Full-Motion Video with MPEG), with the

possibility for 50/15 s pre-emphasis. The whole 50/15 s pre-emphasis matter is a gift from Sony, who in

the early digital period weren't capable of building reasonable DA-convertors. The CCIT standard originates

from the telecommunication environment.

The user code bits can be used by the manufacturer at will. They are used in blocks of 1176 bits before

which a sync-word of 16 "0"-bits is transmitted.

Channel encoding/decoding

Transmitting a digital signal by simply serializing the data is not a good strategy, since it is difficult to estimate

the data rate of the source at the destination, and hence it is difficult to separate the received bits and to

recover an accurate jitter-free clock.

PDIF interfaces use FM channel coding (also known as Manchester coding or bi-phase mark coding), which

is DC free, strongly self-clocking and capable of working with a changing sampling rate. The use of FM

means that the channel frequency is the same as the bit rate when sending data ones, because FM takes care

that a state transition takes place at each bit. A logical 1 is represented by a second transition. Hence the

clock signal can be recovered, independent of the data.

Table 4: Biphase mark coding examples

data Biphase mark equivalent

0000 00110011


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0111 00101010

0101 00101101

The sync-pattern deliberately violates the Manchester coding, so that the beginning of a block is easy todetect. The bit patterns in the preamble always contain more than two 0's or 1's in a row.

Table 5: Preamble contents

Sync pattern

Contents

(bit 31="0")

Contents

(bit 31="1")

B 11101000 00010111

M 11100010 00011101

W 11100100 00011011

Modulation/demodulation

The electrical interface of the consumer format as specified by IEC 958 is shown in the following figure.

Figure 5: Electrical SPDIF interface

It uses a 0.5 Vtt signal, and a frequency of almost 6 MHz1, which can be conveyed down conventional

audio-grade coaxial cable connected with RCA phono plugs.

A slicer can be used to convert the analog waveform to a binary output. The signal voltage is compared to a

particular voltage, called the threshold, and if the signal voltage is above the threshold the comparator outputsa high level, and a low level otherwise. If the waveform would contain a DC component, the average level of

the waveform would raise, and hence this would seriously affect the slicing process. Hence it is clearly not

possible to serialize arbitrary data in a shift-register for direct transmission. The manchester coding process
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takes care that on average the number of 0's is equal to the number of 1's, making slicing possible.

Communication channel

The communication channel consists of a coax-cable, connecting the source and destination. The waveform

transmitted through the cable is affected by noise, baseline wander, intersymbol inerference and imperfect

equalization, resulting in time uncertainties in the position of the edges, resulting in jitter when a slicer is used to

recover the clock. An eye-pattern can be used to observe the characteristics of the communication channel.

In Figure 6 it is shown how particular waveforms (left hand side) are typically displayed using an oscilloscope

(right hand side) triggering on such a waveform.

Figure 6: Eye diagrams (unlimited bandwidth ideal, limited bandwidth ideal, distorted)

In Figure 7 the typical characteristics derived from such an eye-pattern are shown. Noise closes the eye in a

vertical direction, whereas jitter closes the eye in a horizontal direction.

Figure 7: Eye diagram characteristics

In [Watk94] the following minimum eye pattern (see [Shan85]) acceptable for correct decoding of data for a


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biphase-marked data stream in the AES/EBU standard is given (see Figure 8). In this figure Tnom denotes

the half of a biphase symbol period, and Tmin = 0.5 * Tnom . Because in the biphase-marked digital data

stream a data bit is represented by two transitions, the jitter margin can only be a half bit (in the absence of

noise) to be able to reconstruct the data, which is about 80ns. The minimum height of the eye should be

200mV.

As error correcting methods are incorporated inside the SPDIF interface, data errors because of data

corruption can be detected very easily. In normal conditions, data corruption caused by jitter or noise ishardly an issue.

Figure 8: SPDIF eye pattern

Cable impedance

The coax-cable has a so called impedance,which will be explained in the remainder of this section.

Transmission can be modelled as electromagnetic energy travelling from one place to another. Depending on

the frequency of the of the electromagnetic energy, different characteristics can be observed when it is

tranported by conductors.

An electric model of a conductor can be found in Figure 9.

Figure 9: Electrical model of coax cable

In a nutshell the cable capacitance Cresults from the fact that there exists an electric field between the two

conductors given a potential difference between them, determined by the geometry of the space between the

conductors and the nature of the dielectric. The cable inductance L results from the fact that any current-

carrying conductor produces a magnetic field. A cable (with the exception of superconductors) always has a

certain amount of resistance R . And finally, the insultor will cause signal losses modelled by G .

For an infinite length of the network model we find:


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The equation does not depend on the length, because these quantities are distributed. No matter how short

the length of the cable is, it still can be modelled as above. The cable impedance is only a concern when the

cable is a fraction of the wavelength of the signal in question.

At low frequencies electromagnetic energy is called electricity. Electricity is transported completely inside

conductors. At DC an inductor acts as a short circuit, and a capacitance acts as an open circuit. The

resistance is determined by the cross-sectional area of the conductor. The insulation has no effect on the

ability of the conductor to pass current.

As frequency rises, inductors display an increasing impedance with frequency and capacitors show falling

impedance. Electromagnetic energy becomes increasingly desperate to leave the conductor. This results in the

skin-effect, where current only flows in outside layer of the inductor, causing the resistance to rise. As the

energy is starting to leave the conductor, the geometry of the space between them is becoming more

important. This determines the impedance. A change of impedance causes reflections in the energy flow,

hence constant impedance cables with fixed inductor spacing are necessary. As frequency rises even further,

the energy travels less in the conductors and more in the insulation between them. The composition in the

insulators is becoming more important, and they begin to be called dielectrics. Poor dielectrics like PVC

absorb high-frequency energy, and attenuate the signal.

As the transition rate is within an average of a few megabytes per second, and only moderate cable lengths

will be used in practice (hence R and G can be ignored to a certain extent), the main losses of the cable will

be determined byL and C :

Notice that the frequency dependency is gone. Also notice that the result does not depend on the cable length

(when ignoring cable losses), because the quantities are distributed along the cable (you can't isolate the C-

part or L-part from a cable). Also notice that the bandwidth of the signal should be narrow, and the transition

of a sequence of many identical bits should therefore be avoided by modifying the spectrum of the signal by

application of a suitable channel encoding.

So, if the cable length is a fair fraction of the wavelength of the signal in question, the cable can be considered

as a transmission line, and pulses travel down it as current loops. If a positive pulse is launched across the

line, it will charge the dielectric in the direction from the driver to the receiver, resulting in a leading edge.

When the driver ends the pulse, the dielectric will discharge leading to a falling edge, which bears the same

current as the leading edge. There is thus a loop of current rolling along the line as if it were a Caterpillartractor.

Transmission lines which transport energy in this way have a characteristic impedance because of the interplay

of inductance along the conductors with the capacitance. One consequence is that a correct termination or

matching is required between the line and both the driver and the receiver to make the rolling energy roll

straight out of the line into the load. If the impedance is mismatched, reflections will result.

The main concern with digital cables is of the junction that forms at the major connection. If the connectors

are not clean or the cables are not connected in a right manner, a diode effect can be observed, resulting in a

slow rise time and a fast fall time.

1. More exactly, one frame contains 64 bits which are sent during one clock period. This results in a bit-rate

digital communication basic

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