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Digital Communication
Digital Communication
Introduction
In Figure 1 an overview of the typical functional elements of a digital communication system are shown.
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 source
The 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.
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Playing time
Rotation
Track pitch
1.6 m
Diameter
120 mm
Thickness
1.2 mm
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Digital Communication
Recording area
46 mm - 117 mm
Signal area
50 mm - 116 mm
Material
Pit depth
~0.11 m
Pit width
~0.5 m
Focal depth
2m
Quantization
16-bit linear
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Quantizing timing
Sampling frequency
44.1 kHz
4.3218 Mb/sec
2.0338 Mb/sec
Error correction code Cross Interleave Reed-Solomon Code (with 25% redundancy)
Modulation system
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:
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 word containing 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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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 CDROM 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.
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
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 ...
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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
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 to detect. The bit patterns in the preamble always contain more than two 0's or 1's in a row.
Table 5: Preamble contents
Contents
11101000
00010111
11100010
00011101
11100100
00011011
Modulation/demodulation
The electrical interface of the consumer format as specified by IEC 958 is shown in the following figure.
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 outputs a 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.
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 is hardly an issue.
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.
In a nutshell the cable capacitance C results 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 currentcarrying 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 by L 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 Cpart 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 Caterpillar tractor. 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
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of 2.8224Mbit/s (44.1kHz), 3.072 Mbit/s (48kHz), or 2.048Mbit/s (32kHz), which after bi-phase mark coding is doubled.
Copyright 2001, Marc Heijligers and the DAC group - All rights reserved.
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