8

Transmission

8.1 Introduction

The distances involved in transmission vary from that of a short cable between adjacent units to communication anywhere on earth via data networks or radio communication. This chapter must consider a correspondingly wide range of possibilities. The importance of direct digital interconnection between audio devices was realized early, and numerous incompatible (and now obsolete) methods were developed by various manufacturers until standardization was reached in the shape of the AES/EBU digital audio interface for professional equipment and the SPDIF interface for consumer equipment. These standards were extended to produce the MADI standard for multi-channel interconnects. All of these work on uncompressed PCM audio.

As digital audio and computers continue to converge, computer networks are also being used for audio purposes. Audio may be transmitted on networks such as Ethernet, ISDN, ATM and Internet. Here compression may or may not be used, and non-real-time transmission may also be found according to economic pressures.

Digital audio is now being broadcast in its own right as DAB, alongside traditional analog television as NICAM digital audio and as MPEG or AC-3 coded signals in digital television broadcasts. Many of the systems described here rely upon coding principles described in Chapters 6 and 7.

Whatever the transmission medium, one universal requirement is a reliable synchronization system. In PCM systems, synchronization of the sampling rate between sources is necessary for mixing. In packet-based networks, synchronization allows the original sampling rate to be established at the receiver despite the intermittent transfer of a real packet systems. In digital television systems, synchronization between vision and sound is a further requirement.

8.2 Introduction to AES/EBU interface

The AES/EBU digital audio interface, originally published in 1985,¹ was proposed to embrace all the functions of existing formats in one standard. The goal was to ensure interconnection of professional digital audio equipment irrespective of origin. The EBU ratified the AES proposal with the proviso that the optional transformer coupling was made mandatory and led to the term AES/EBU interface, also called EBU/AES in some European countries. The contribution of the BBC to the development of the interface must be mentioned here. Alongside the professional format, Sony and Philips developed a similar format now known as SPDIF (Sony Philips Digital Interface) intended for consumer use. This offers different facilities to suit the application, yet retains sufficient compatibility with the professional interface so that, for many purposes, consumer and professional machines can be connected together.^2,3

The AES concerns itself with professional audio and accordingly has had little to do with the consumer interface. Thus the recommendations to standards bodies such as the IEC (International Electrotechnical Commission) regarding the professional interface came primarily through the AES whereas the consumer interface input was primarily from industry, although based on AES professional proposals. The IEC and various national standards bodies naturally tended to combine the two into one standard such as IEC 958⁴ which refers to the professional interface and the consumer interface. This process has been charted by Finger.⁵

Understandably with so many standards relating to the same subject differences in interpretation arise leading to confusion in what should or should not be implemented, and indeed what the interface should be called. This chapter will refer generically to the professional interface as the AES/EBU interface and the consumer interface as SPDIF.

Getting the best results out of the AES/EBU interface, or indeed any digital interface, requires some care. Section 13.9 treats this subject in some detail

8.3 The electrical interface

During the standardization process it was considered desirable to be able to use existing analog audio cabling for digital transmission. Existing professional analog signals use nominally 600 impedance balanced line screened signalling, with one cable per audio channel, or in some cases one twisted pair per channel with a common screen. The 600 standard came from telephony where long distances are involved in comparison with electrical audio wavelengths. The distances likely to be found within a studio complex are short compared to audio electrical wavelengths and as a result at audio frequency the impedance of cable is high and the 600 ohm figure is that of the source and termination. Such a cable has a different impedance at the frequencies used for digital audio, around 110 .

If a single serial channel is to be used, the interconnect has to be self-clocking and self-synchronizing, i.e. the single signal must carry enough information to allow the boundaries between individual bits, words and blocks to be detected reliably. To fulfil these requirements, the AES/EBU and SPDIF interfaces use FM channel code (see Chapter 6) which is DC-free, strongly self-clocking and capable of working with a changing sampling rate. Synchronization of deserialization is achieved by violating the usual encoding rules.

The use of FM means that the channel frequency is the same as the bit rate when sending data ones. Tests showed that in typical analog audio cabling installations, sufficient bandwidth was available to convey two digital audio channels in one twisted pair. The standard driver and receiver chips for RS-422A⁶ data communication (or the equivalent CCITTV.11) are employed for professional use, but work by the BBC⁷ suggested that equalization and transformer coupling were desirable for longer cable runs, particularly if several twisted pairs occupy a common shield. Successful transmission up to 350 m has been achieved with these techniques.⁸ Figure 8.1 shows the standard configuration. The output impedance of the drivers will be about 110 ohms, and the impedance of the cable used should be similar at the frequencies of interest. The driver was specified in AES-3–1985 to produce between 3 and 10 V peak-to-peak into such an impedance but this was changed to between 2 and 7 volts in AES-3–1992 to better reflect the characteristics of actual RS-422 driver chips.

Figure 8.1    Recommended electrical circuit for use with the standard two-channel interface.

Figure 8.2    The minimum eye pattern acceptable for correct decoding of standard two-channel data.

The original receiver impedance was set at a high 250 , with the intention that up to four receivers could be driven from one source. This was found to be inadvisable because of reflections caused by impedance mismatches and AES-3–1992 is now a point-to-point interface with source, cable and load impedance all set at 110 . Whilst analog audio cabling was adequate for digital signalling, cable manufacturers have subsequently developed cables which are more appropriate for new digital installations, having lower loss factors allowing greater transmission distances.

In Figure 8.2, the specification of the receiver is shown in terms of the minimum eye pattern (see Chapter 6) which can be detected without error. It will be noted that the voltage of 200 mV specifies the height of the eye opening at a width of half a channel bit period. The actual signal amplitude will need to be larger than this, and even larger if the signal contains noise. Figure 8.3 shows the recommended equalization characteristic which can be applied to signals received over long lines.

As an adequate connector in the shape of the XLR is already in wide service, the connector made to IEC 268 Part 12 has been adopted for digital audio use. Effectively, existing analog audio cables having XLR connectors can be used without alteration for digital connections. The AES/EBU standard does, however, require that suitable labelling should be used so that it is clear that the connections on a particular unit are digital. Whilst the XLR connector was never designed to have constant impedance in the megaHertz range, it is capable of towing an outside broadcast vehicle without unlatching.

The need to drive long cables does not generally arise in the domestic environment, and so a low-impedance balanced signal was not considered necessary. The electrical interface of the consumer format uses a 0.5 V peak single-ended signal, which can be conveyed down conventional audio-grade coaxial cable connected with RCA ‘phono’ plugs. Figure 8.4 shows the resulting consumer interface as specified by IEC 958.

Figure 8.3    EQ characteristic recommended by the AES to improve reception in the case of long lines.

Figure 8.4    The consumer electrical interface.

There is the additional possibility⁹ of a professional interface using coaxial cable and BNC connectors for distances of around 1000 m. This is simply the AES/EBU protocol but with a 75 coaxial cable carrying a one-volt signal so that it can be handled by analog video distribution amplifiers. Impedance converting transformers are commercially available allowing balanced 110 ohm to unbalanced 75 matching.

8.4 Frame structure

In Figure 8.5 the basic structure of the professional and consumer formats can be seen. One subframe consists of 32 bit-cells, of which four will be used by a synchronizing pattern. Subframes from the two audio channels, A and B, alternate on a time-division basis. Up to 24-bit sample wordlength can be used, which should cater for all conceivable future developments, but normally 20-bit maximum length samples will be available with four auxiliary data bits, which can be used for a voice-grade channel in a professional application. In a consumer DAT machine, subcode can be transmitted in bits 4–11, and the sixteen-bit audio in bits 12–27.

Figure 8.5    The basic subframe structure of the AES/EBU format. Sample can be 20 bits with four auxiliary bits, or 24 bits. LSB is transmitted first.

Preceding formats sent the most significant bit first. Since this was the order in which bits were available in successive approximation convertors it has become a de-facto standard for inter-chip transmission inside equipment. In contrast, this format sends the least significant bit first. One advantage of this approach is that simple serial arithmetic is then possible on the samples because the carries produced by the operation on a given bit can be delayed by one bit period and then included in the operation on the next higher-order bit. There is additional complication, however, if it is proposed to build adaptors from one of the manufacturers’ formats to the new format because of the word reversal. This problem is a temporary issue, as new machines are designed from the outset to have the standard connections.

The format specifies that audio data must be in two’s complement coding. Whilst pure binary could accept various alignments of different wordlengths with only a level change, this is not true of two’s complement. If different wordlengths are used, the MSBs must always be in the same bit position otherwise the polarity will be misinterpreted. Thus the MSB has to be in bit 27 irrespective of wordlength. Shorter words are leading zero filled up to the 20-bit capacity. The channel status data included from AES-3–1992 signalling of the actual audio wordlength used so that receiving devices could adjust the digital dithering level needed to shorten a received word which is too long or pack samples onto a disk more efficiently.

Four status bits accompany each subframe. The validity flag will be reset if the associated sample is reliable. Whilst there have been many aspirations regarding what the V bit could be used for, in practice a single bit cannot specify much, and if combined with other V bits to make a word, the time resolution is lost. AES-3–1992 described the V bit as indicating that the information in the associated subframe is ‘suitable for conversion to an analog signal’. Thus it might be reset if the interface was being used for non-audio data as is done, for example, in CD-I players.

The parity bit produces even parity over the subframe, such that the total number of ones in the subframe is even. This allows for simple detection of an odd number of bits in error, but its main purpose is that it makes successive sync patterns have the same polarity, which can be used to improve the probability of detection of sync. The user and channel-status bits are discussed later.

Two of the subframes described above make one frame, which repeats at the sampling rate in use. The first subframe will contain the sample from channel A, or from the left channel in stereo working. The second subframe will contain the sample from channel B, or the right channel in stereo. At 48 kHz, the bit rate will be 3.072 MHz, but as the sampling rate can vary, the clock rate will vary in proportion.

In order to separate the audio channels on receipt the synchronizing patterns for the two subframes are different as Figure 8.6 shows. These sync patterns begin with a run length of 1.5 bits which violates the FM channel coding rules and so cannot occur due to any data combination. The type of sync pattern is denoted by the position of the second transition which can be 0.5, 1.0 or 1.5 bits away from the first. The third transition is designed to make the sync patterns DC-free.

The channel status and user bits in each subframe form serial data streams with one bit of each per audio channel per frame. The channel status bits are given a block structure and synchronized every 192 frames, which at 48 kHz gives a block rate of 250 Hz, corresponding to a period of four milliseconds. In order to synchronize the channel-status blocks, the channel A sync pattern is replaced for one frame only by a third sync pattern which is also shown in Figure 8.6. The AES standard refers to these as X, Y and Z whereas IEC 958 calls them M, W and B. As stated, there is a parity bit in each subframe, which means that the binary level at the end of a subframe will always be the same as at the beginning. Since the sync patterns have the same characteristic, the effect is that sync patterns always have the same polarity and the receiver can use that information to reject noise. The polarity of transmission is not specified, and indeed an accidental inversion in a twisted pair is of no consequence, since it is only the transition that is of importance, not the direction.

Figure 8.6    Three different preambles (X, Y and Z) are used to synchronize a receiver at the starts of subframes.

8.5 Talkback in auxiliary data

When 24-bit resolution is not required, which is most of the time, the four auxiliary bits can be used to provide talkback.

This was proposed by broadcasters¹⁰ to allow voice coordination between studios as well as program exchange on the same cables. Twelve-bit samples of the talkback signal are taken at one third the main sampling rate. Each twelve-bit sample is then split into three nibbles (half a byte, for gastronomers) which can be sent in the auxiliary data slot of three successive samples in the same audio channel. As there are 192 nibbles per channel status block period, there will be exactly 64 talkback samples in that period. The reassembly of the nibbles can be synchronized by the channel status sync pattern as shown in Figure 8.7. Channel status byte 2 reflects the use of auxiliary data in this way.

Figure 8.7    The coordination signal is of a lower bit rate than the main audio and thus may be inserted in the auxiliary nibble of the interface subframe, taking three subframes per coordination sample.

8.6 Professional channel status

In the both the professional and consumer formats, the sequence of channel-status bits over 192 subframes builds up a 24-byte channel-status block. However, the contents of the channel status data are completely different between the two applications. The professional channel status structure is shown in Figure 8.8. Byte 0 determines the use of emphasis and the sampling rate, with details in Figure 8.9. Byte 1 determines the channel usage mode, i.e. whether the data transmitted are a stereo pair, two unrelated mono signals or a single mono signal, and details the user bit handling. Figure 8.10 gives details. Byte 2 determines wordlength as in Figure 8.11. This was made more comprehensive in AES-3–1992. Byte 3 is applicable only to multichannel applications. Byte 4 indicates the suitability of the signal as a sampling rate reference and will be discussed in more detail later in this chapter.

Figure 8.8    Overall format of the professional channel status block.

Figure 8.9    The first byte of the channel-status information in the AES/EBU standard deals primarily with emphasis and sampling-rate control.

Figure 8.10    Format of byte 1 of professional channel status.

Figure 8.11    Format of byte 2 of professional channel status.

There are two slots of four bytes each which are used for alphanumeric source and destination codes. These can be used for routing. The bytes contain seven-bit ASCII characters (printable characters only) sent LSB first with the eighth bit set to zero acording to AES-3–1992. The destination code can be used to operate an automatic router, and the source code will allow the origin of the audio and other remarks to be displayed at the destination.

Bytes 14–17 convey a 32-bit sample address which increments every channel status frame. It effectively numbers the samples in a relative manner from an arbitrary starting point. Bytes 18–21 convey a similar number, but this is a time-of-day count, which starts from zero at midnight. As many digital audio devices do not have real-time clocks built in, this cannot be relied upon.

AES-3–92 specified that the time-of-day bytes should convey the real time at which a recording was made, making it rather like timecode. There are enough combinations in 32 bits to allow a sample count over 24 hours at 48 kHz. The sample count has the advantage that it is universal and independent of local supply frequency. In theory if the sampling rate is known, conventional hours, minutes, seconds, frames timecode can be calculated from the sample count, but in practice it is a lengthy computation and users have proposed alternative formats in which the data from EBU or SMPTE timecode are transmitted directly in these bytes. Some of these proposals are in service as de-facto standards.

Figure 8.12    Byte 22 of channel status indicates if some of the information in the block is unreliable.

The penultimate byte contains four flags which indicate that certain sections of the channel-status information are unreliable (see Figure 8.12). This allows the transmission of an incomplete channel-status block where the entire structure is not needed or where the information is not available. For example, setting bit 5 to a logical one would mean that no origin or destination data would be interpreted by the receiver, and so it need not be sent.

The final byte in the message is a CRCC which converts the entire channel-status block into a codeword (see Chapter 7). The channel status message takes 4 ms at 48 kHz and in this time a router could have switched to another signal source. This would damage the transmission, but will also result in a CRCC failure so the corrupt block is not used. Error correction is not necessary, as the channel status data are either stationary, i.e. they stay the same, or change at a predictable rate, e.g. timecode. Stationary data will only change at the receiver if a good CRCC is obtained.

8.7 Consumer channel status

For consumer use, a different version of the channel-status specification is used. As new products come along, the consumer subcode expands its scope.

Figure 8.13 shows that the serial data bits are assembled into twelve words of sixteen bits each. In the general format, the first six bits of the first word form a control code, and the next two bits permit a mode select for future expansion. At the moment only mode zero is standardized, and the three remaining codes are reserved.

Figure 8.13    The general format of the consumer version of channel status. Bit 0 has the same meaning as in the professional format for compatibility. Bits 6–7 determine the consumer format mode, and presently only mode 0 is defined (see Figure 8.14).

Figure 8.14 shows the bit allocations for mode zero. In addition to the control bits, there are a category code, a simplified version of the AES/ EBU source field, a field which specifies the audio channel number for multichannel working, a sampling-rate field, and a sampling-rate tolerance field.

Originally the consumer format was incompatible with the professional format, since bit zero of channel status would be set to a one by a four-channel consumer machine, and this would confuse a professional receiver because bit zero specifies professional format. The EBU proposed to the IEC that the four-channel bit be moved to bit 5 of the consumer format, so that bit zero would always then be zero. This proposal is incorporated into the bit definitions of Figures 8.13 and 8.14.

The category code specifies the type of equipment which is transmitting, and its characteristics. In fact each category of device can output one of two category codes, depending on whether bit 15 is or is not set. Bit 15 is the ‘L-bit’ and indicates whether the signal is from an original recording (0) or from a first-generation copy (1) as part of the SCMS (Serial Copying Management System) first implemented to resolve the stalemate over the sale of consumer DAT machines. In conjunction with the copyright flag, a receiving device can determine whether to allow or disallow recording. There were originally four categories; general purpose, two-channel CD player, two-channel PCM adaptor and two-channel digital tape recorder (DAT), but the list has now extended as Figure 8.14 shows.

Figure 8.14    In consumer mode 0, the significance of the first two sixteen-bit channel-status words is shown here. The category codes are expanded in Tables 8.1 and 8.2.

Table 8.1 illustrates the format of the subframes in the general-purpose category. When used with CD players, Table 8.2 applies. In this application, the extensive subcode data of a CD recording (see Chapter 12) can be conveyed down the interface. In every CD sync block, there are twelve audio samples, and eight bits of subcode, P–W. The P flag is not transmitted, since it is solely positioning information for the player; thus only Q–W are sent. Since the interface can carry one user bit for every sample, there is surplus capacity in the user-bit channel for subcode. A CD subcode block is built up over 98 sync blocks, and has a repetition rate of 75 Hz. The start of the subcode data in the user bitstream will be seen in Figure 8.15 to be denoted by a minimum of sixteen zeros, followed by a start bit which is always set to one. Immediately after the start bit, the receiver expects to see seven subcode bits, Q–W. Following these, another start bit and another seven bits may follow immediately, or a space of up to eight zeros may be left before the next start bit. This sequence repeats 98 times, when another sync pattern will be expected. The ability to leave zeros between the subcode symbols simplifies the handling of the disparity between user bit capacity and subcode bit rate. Figure 8.16 shows a representative example of a transmission from a CD player.

Figure 8.15    In CD, one subcode block is built up over 98 sync blocks. In this period there will be 1176 audio samples, and so there are 1176 user bits available to carry the subcode. There is insufficient subcode information to fill this capacity, and zero packing is used.

In a PCM adaptor, there is no subcode, and the only ancillary information available from the recording consists of copy-protect and emphasis bits. In other respects, the format is the same as the general-purpose format.

When a DAT player is used with the interface, the user bits carry several items of information.¹¹ Once per drum revolution, the user bit in one subframe is raised when that subframe contains the first sample of the interleave block (see Chapter 9). This can be used to synchronize several DAT machines together for editing purposes. Immediately following the sync bit, start ID will be transmitted when the player has found the code on the tape. This must be asserted for 300 ± 30 drum revolutions, or about 10 seconds. In the third bit position the skip ID is transmitted when the player detects a skip command on the tape. This indicates that the player will go into fast forward until it detects the next start ID. The skip ID must be transmitted for 33 ± 3 drum rotations. Finally DAT supports an end-of-skip command which terminates a skip when it is detected. This allows jump editing (see Chapter 11) to omit short sections of the recording. DAT can also transmit the track number (TNO) of the track being played down the user bitstream.

Figure 8.16    Compact Disc subcode transmitted in user bits of serial interface.

8.8 User bits

The user channel consists of one bit per audio channel per sample period. Unlike channel status, which only has a 192-bit frame structure, the user channel can have a flexible frame length. Figure 8.10 showed how byte 1 of the channel status frame describes the state of the user channel. Many professional devices do not use the user channel at all and would set the all-zeros code. If the user channel frame has the same length as the channel status frame then code 0001 can be set. One user channel format which is standardized is the data packet scheme of AES18–1992.^12,13 This was developed from proposals to employ the user channel for labelling in an asynchronous format.¹⁴ A computer industry standard protocol known as HDLC (High-level Data Link Control)¹⁵ is employed in order to take advantage of readily available integrated circuits.

The frame length of the user channel can be conveniently made equal to the frame period of an associated device. For example, it may be locked to Film, TV or DAT frames. The frame length may vary in NTSC as there are not an integer number of samples in a frame.

8.9 MADI – Multi-channel audio digital interface

Whilst the AES/EBU digital interface excels for the interconnection of stereo equipment, it is at a disadvantage when a large number of channels is required. MADI¹⁶ was designed specifically to address the requirement for digital connection between multitrack recorders and mixing consoles by a working group set up jointly by Sony, Mitsubishi, Neve and SSL.

The standard provides for 56 simultaneous digital audio channels which are conveyed point-to-point on a single 75 coaxial cable fitted with BNC connectors (as used for analog video) along with a separate synchronization signal. A distance of at least 50 m can be achieved.

Essentially MADI takes the subframe structure of the AES/EBU interface and multiplexes 56 of these into one sample period rather than the original two. Clearly this will result in a considerable bit rate, and the FM channel code of the AES/EBU standard would require excessive bandwidth. A more efficient code is used for MADI. In the AES/EBU interface the data rate is proportional to the sampling rate in use. Losses will be greater at the higher bit rate of MADI, and the use of a variable bit rate in the channel would make the task of achieving optimum equalization difficult. Instead the data bit rate is made a constant 100 megabits per second, irrespective of sampling rate. At lower sampling rates, the audio data are padded out to maintain the channel rate.

The MADI standard is effectively a superset of the AES/EBU interface in that the subframe data content is identical. This means that a number of separate AES/EBU signals can be fed into a MADI channel and recovered in their entirety on reception. The only caution required with such an application is that all channels must have the same synchronized sampling rate. The primary application of MADI is to multitrack recorders, and in these machines the sampling rates of all tracks are intrinsically synchronous. When the replay speed of such machines is varied, the sampling rate of all channels will change by the same amount, so they will remain synchronous.

At one extreme, MADI will accept a 32 kHz recorder playing 12½ per cent slow, and at the other extreme a 48 kHz recorder playing 12½ per cent fast. This is almost a factor of 2:1. Figure 8.17 shows some typical MADI configurations.

Figure 8.17    Some typical MADI applications. In (a) a large number of two-channel digital signals are multiplexed into the MADI cable to achieve economy of cabling. Note the separate timing signal. In (b) a pair of MADI links is necessary to connect a recorder to a mixing console. A third MADI link could be used to feed microphones into the desk from remote convertors.

8.10 MADI data transmission

The data transmission of MADI is made using a group code, where groups of four data bits are represented by groups of five channel bits. Four bits have sixteen combinations, whereas five bits have 32 combinations. Clearly only 16 out of these 32 are necessary to convey all possible data. It is then possible to use some of the remaining patterns when it is required to pad out the data rate. The padding symbols will not correspond to a valid data symbol and so they can be recognized and thrown away on reception. A further use of this coding technique is that the 16 patterns of 5 bits which represent real data are chosen to be those which will have the best balance between high and low states, so that DC offsets at the receiver can be minimized. Chapter 6 discussed the coding rules of 4/5 in MADI. The 4/5 code adopted is the same one used for a computer transmission format known as FDDI, so that existing hardware can be used.

8.11 MADI frame structure

Figure 8.18(a) shows the frame structure of MADI. In one sample period, 56 time slots are available, and these each contain eight 4/5 symbols, corresponding to 32 data bits or 40 channel bits. Depending on the sampling rate in use, more or less padding symbols will need to be inserted in the frame to maintain a constant channel bit rate. Since the receiver does not interpret the padding symbols as data, it is effectively blind to them, and so there is considerable latitude in the allowable positions of the padding. Figure 8.18(b) shows some possibilities. The padding must not be inserted within a channel, only between channels, but the channels need not necessarily be separated by padding. At one extreme, all channels can be butted together, followed by a large padding area, or the channels can be evenly spaced throughout the frame. Although this sounds rather vague, it is intended to allow freedom in the design of associated hardware. Multitrack recorders generally have some form of internal multiplexed data bus, and these have various architectures and protocols. The timing flexibility allows an existing bus timing structure to be connected to a MADI link with the minimum of hardware. Since the channels can be inserted at a variety of places within the frame, the necessity of a separate synchronizing link between transmitter and receiver becomes clear.

Figure 8.18    In (a) all 56 channels are sent in numerical order serially during the sample period. For simplicity no padding symbols are shown here. In (b) the use of padding symbols is illustrated. These are necessary to maintain the channel bit rate at 125 M bits/s irrespective of the sample rate in use. Padding can be inserted flexibly, but it must only be placed between the channels.

8.12 MADI Audio channel format

Figure 8.19 shows the MADI channel format, which should be compared with the AES/EBU subframe shown in Figure 8.5. The last 28 bits are identical, and differences are only apparent in the synchronizing area. In order to remain transparent to an AES/EBU signal, which can contain two audio channels, MADI must tell the receiver whether a particular channel contains the A leg or B leg, and when the AES/EBU channel status block sync occurs. Bits 2 and 3 perform these functions. As the 56 channels of MADI follow one another in numerical order, it is necessary to identify channel zero so that the channels are not mixed up. This is the function of bit 0, which is set in channel zero and reset in all other channels. Finally bit 1 is set to indicate an active channel, for the case when less than 56 channels are being fed down the link. Active channels have bit 1 set, and must be consecutive starting at channel zero. Inactive channels have all bits set to zero, and must follow the active channels.

Figure 8.19    The MADI channel data are shown here. The last 28 bits are identical in every way to the AES/EBU interface, but the synchronizing in the first four bits differs. There is a frame sync bit to identify channel 0, and a channel active bit. The A/B leg of a possible AES/EBU input to MADI is conveyed, as is the channel status block sync.

8.13 Fibre-optic interfacing

Whereas a parallel bus is ideal for a distributed multichannel system, for a point-to-point connection, the use of fibre optics is feasible, particularly as distance increases. An optical fibre is simply a glass filament which is encased in such a way that light is constrained to travel along it. Transmission is achieved by modulating the power of an LED or small laser coupled to the fibre. A phototransistor converts the received light back to an electrical signal.

Optical fibres have numerous advantages over electrical cabling. The bandwidth available is staggering. Optical fibres neither generate, nor are prone to, electromagnetic interference and, as they are insulators, ground loops cannot occur.¹⁷ The disadvantage of optical fibres is that the terminations of the fibre where transmitters and receivers are attached suffer optical losses, and while these can be compensated in point-to-point links, the use of a bus structure is not really feasible. Fibre-optic links are already in service in digital audio mixing consoles.¹⁸ The fibre implementation by Toshiba and known as the TOSLink is popular in consumer products, and the protocol is identical to the consumer electrical format.

8.14 Synchronizing

When digital audio signals are to be assembled from a variety of sources, either for mixing down or for transmission through a TDM (time-division multiplexing) system, the samples from each source must be synchronized to one another in both frequency and phase. The source of samples must be fed with a reference sampling rate from some central generator, and will return samples at that rate. The same will be true if digital audio is being used in conjunction with VTRs. As the scanner speed and hence the audio block rate is locked to video, it follows that the audio sampling rate must be locked to video. Such a technique has been used since the earliest days of television in order to allow vision mixing, but now that audio is conveyed in discrete samples, these too must be genlocked to a reference for most production purposes.

AES11–1991¹⁹ documented standards for digital audio synchronization and requires professional equipment to be able to genlock either to a separate reference input or to the sampling rate of an AES/EBU input.

Figure 8.20    The timing accuracy required in AES/EBU signals with respect to a reference (a). Inputs over the range shown at (b) must be accepted, whereas outputs must be closer in timing to the reference as shown at (c).

As the interface uses serial transmission, a shift register is required in order to return the samples to parallel format within equipment. The shift register is generally buffered with a parallel loading latch which allows some freedom in the exact time at which the latch is read with respect to the serial input timing. Accordingly the standard defines synchronism as an identical sampling rate, but with no requirement for a precise phase relationship. Figure 8.20 shows the timing tolerances allowed. The beginning of a frame (the frame edge) is defined as the leading edge of the X preamble. A device which is genlocked must correctly decode an input whose frame edges are within ± 25 per cent of the sample period. This is quite a generous margin, and corresponds to the timing shift due to putting about a kilometre of cable in series with a signal. In order to prevent tolerance build-up when passing through several devices in series, the output timing must be held within ± 5 per cent of the sample period.

The reference signal may be an AES/EBU signal carrying program material, or it may carry muted audio samples; the so-called digital audio silence signal. Alternatively it may just contain the sync patterns. The accuracy of the reference is specified in bits 0 and 1 of byte 4 of channel status (see Figure 8.8). Two zeros indicates the signal is not reference grade (but some equipment may still be able to lock to it). 01 indicates a Grade 1 reference signal which is ±1 ppm accurate, whereas 10 indicates a Grade 2 reference signal which is ±10 ppm accurate. Clearly devices which are intended to lock to one of these references must have an appropriate phase-locked-loop capture range.

In addition to the AES/EBU synchronization approach, some older equipment carries a word clock input which accepts a TTL level square wave at the sampling frequency. This is the reference clock of the old Sony SDIF-2 interface.

Modern digital audio devices may also have a video input for synchronizing purposes. Video syncs (with or without picture) may be input, and a phase-locked loop will multiply the video frequency by an appropriate factor to produce a synchronous audio sampling clock.

8.15 Asynchronous operation

In practical situations, genlocking is not always possible. In a satellite transmission, it is not really practicable to genlock a studio complex halfway around the world to another. Outside broadcasts may be required to generate their own master timing for the same reason. When genlock is not achieved, there will be a slow slippage of sample phase between source and destination due to such factors as drift in timing generators. This phase slippage will be corrected by a synchronizer, which is intended to work with frequencies that are nominally the same. It should be contrasted with the sampling-rate convertor which can work at arbitrary but generally greater frequency relationships. Although a sampling-rate convertor can act as a synchronizer, it is a very expensive way of doing the job. A synchronizer can be thought of as a lower-cost version of a sampling-rate convertor which is constrained in the rate difference it can accept.

In one implementation of a digital audio synchronizer,²⁰ memory is used as a timebase corrector. Samples are written into the memory with the frequency and phase of the source and, when the memory is half-full, samples are read out with the frequency and phase of the destination. Clearly if there is a net rate difference, the memory will either fill up or empty over a period of time, and in order to recentre the address relationship, it will be necessary to jump the read address. This will cause samples to be omitted or repeated, depending on the relationship of source rate to destination rate, and would be audible on program material. The solution is to detect pauses or low-level passages and permit jumping only at such times. The process is illustrated in Figure 8.21. An alternative to address jumping is to undertake sampling-rate conversion for a short period (Figure 8.22) in order to slip the input/ output relationship by one sample.²¹ If this is done when the signal level is low, short wordlength logic can be used. However, now that sampling rate convertors are available as a low-cost single chip, these solutions are found less often in hardware, although they may be used in software-controlled processes.

Figure 8.21    In jump synchronizing, input samples are subjected to a varying delay to align them with output timing. Eventually the sample relationship is forced to jump to prevent delay building up. As shown here, this results in several samples being repeated, and can only be undertaken during program pauses, or at very low audio levels. If the input rate exceeds the output rate, some samples will be lost.

Figure 8.22    An alternative synchronizing process is to use a short period of interpolation in order to regulate the delay in the synchronizer.

The difficulty of synchronizing unlocked sources is eased when the frequency difference is small. This is one reason behind the clock accuracy standards for AES/EBU timing generators.²²

8.16 Routing

Routing is the process of directing signals betwen a large number of devices so that any one can be connected to any other. The principle of a router is not dissimilar to that of a telephone exchange. In analog routers, there is the potential for quality loss due to the switching element. Digital routers are attractive because they need introduce no loss whatsoever. In addition, the switching is performed on a binary signal and therefore the cost can be lower. Routers can be either cross-point or time-division multiplexed.

In a BBC proposal²³ the 28 data-bit structure of the AES/EBU subframe has been turned sideways, with one conductor allocated to each bit. Since the maximum transition rate of the AES/EBU interface is 64 times the sampling rate it follows that, in the parallel implementation, 64 channels could be time-multiplexed into one sample period within the same bandwidth. The necessary signals are illustrated in Figure 8.23. In order to separate the channels on reception, there are six address lines which convey a binary pattern corresponding to the audio channel number of the sample in that time slot. The receiver simply routes the samples according to the attached address. Such a point-to-point system does, however, neglect the potential of the system for more complex use. The bus cable can loop through several different items of equipment, each of which is programmed so that it transmits samples during a different set of time slots from the others. Since all transmissions are available at all receivers, it is only necessary to detect a given address to latch samples from any channel. If two devices decode the same address, the same audio channel will be available at two destinations.

Figure 8.23    Time-division multiplexed 64 channel audio bus proposed by BBC.

In such a system, channel reassignment is easy. If the audio channels are transmitted in address sequence, it is only necessary to change the addresses which the receiving channels recognize, and a given input channel will emerge from a different output channel. Since the address recognition circuitry is already present in a TDM system, the functionality of a 64 64 point channel-assignment patchboard has been achieved with no extra hardware. The only constraint in the use of TDM systems is that all channels must have synchronized sampling rates. In multitrack recorders this occurs naturally because all the channels are locked by the tape format. With analog inputs it is a simple matter to drive all convertors from a common clock.

Given that the MADI interface uses TDM, it is also possible to perform routing functions using MADI-based hardware.

For asynchronous systems, or where several sampling rates are found simultaneously, a cross-point type of channel-assignment matrix will be necessary, using AES/EBU signals. In such a device, the switching can be performed by logic gates at low cost, and in the digital domain there is, of course, no quality degradation.

8.17 Networks

In the most general sense a network is means of communication between a large number of places. According to this definition the Post Office is a network, as are parcel and courier companies. This type of network delivers physical objects. If, however, we restrict the delivery to information only the result is a telecommunications network. The telephone system is a good example of a telecommunications network because it displays most of the characteristics of later networks.

It is fundamental in a network that any port can communicate with any other port. Figure 8.24 shows a primitive three-port network. Clearly each port must select one or other of the remaining ports in a trivial switching system. However, if it were attempted to redraw Figure 8.24 with one hundred ports, each one would need a 99-way switch and the number of wires needed would be phenomenal. Another approach is needed.

Figure 8.24    Switching is simple with a small number of ports.

Figure 8.25    An exchange or switch can connect any input to any output, but extra switching is needed to support more than one connection.

Figure 8.25 shows that the common solution is to have an exchange, also known as a router, hub or switch, which is connected to every port by a single cable. In this case when a port wishes to communicate with another, it instructs the switch to make the connection. The complexity of the switch varies with its performance. The minimal case may be to install a single input selector and a single output selector. This allows any port to communicate with any other, but only one at a time. If more simultaneous communications are needed, further switching is needed. The extreme case is where every possible pair of ports can communicate simultaneously.

The amount of switching logic needed to implement the extreme case is phenomenal and in practice it is unlikely to be needed. One fundamental property of networks is that they are seldom implemented with the extreme case supported. There will be an economic decision made balancing the number of simultaneous communications with the equipment cost. Most of the time the user will be unaware that this limit exists, until there is a statistically abnormal condition which causes more than the usual number of nodes to attempt communication.

The phrase ‘the switchboard was jammed’ has passed into the language and stayed there despite the fact that manual switchboards are only seen in museums. This is a characteristic of networks. They generally only work up to a certain throughput and then there are problems. This doesn’t mean that networks aren’t useful, far from it. What it means is that with care, networks can be very useful, but without care they can be a nightmare.

There are two key factors to get right in a network. The first is that it must have enough throughput, bandwidth or connectivity to handle the anticipated usage and the second is that a priority system or algorithm is chosen which has appropriate behaviour during overload. These two characteristics are quite different, but often come as a pair in a network corresponding to a particular standard.

Where each device is individually cabled, the result is a radial network shown in Figure 8.26(a). It is not necessary to have one cable per device and several devices can co-exist on a single cable if some form of multiplexing is used. This might be time-division multiplexing (TDM) or frequency division multiplexing (FDM). In TDM, shown in Figure 8.26(b), the time axis is divided into steps which may or may not be equal in length. In Ethernet, for example, these are called frames. During each time step or frame a pair of nodes have exclusive use of the cable. At the end of the time step another pair of nodes can communicate. Rapidly switching between steps gives the illusion of simultaneous transfer between several pairs of nodes. In FDM, simultaneous transfer is possible because each message occupies a different band of frequencies in the cable. Each node has to ‘tune’ to the correct signal. In practice it is possible to combine FDM and TDM. Each frequency band can be time multiplexed in some applications.

Figure 8.26    (a) Radial installations need a lot of cabling. Time-division multiplexing, where transfers occur during different time frames, reduces this requirement (b).

Data networks originated to serve the requirements of computers and it is a simple fact that most computer processes don’t need to be performed in real time or indeed at a particular time at all. Networks tend to reflect that background as many of them, particularly the older ones, are asynchronous.

Asynchronous means that the time taken to deliver a given quantity of data is unknown. A TDM system may chop the data into several different transfers and each transfer may experience delay according to what other transfers the system is engaged in. Ethernet and most storage system buses are asynchronous. For broadcasting purposes an asynchronous delivery system is no use at all, but for copying a video data file between two storage devices an asynchronous system is perfectly adequate.

The opposite extreme is the synchronous system in which the network can guarantee a constant delivery rate and a fixed and minor delay. An AES/EBU router is a synchronous network.

In between asynchronous and synchronous networks reside the isochronous approaches. These can be thought of as sloppy synchronous networks or more rigidly controlled asynchronous networks. Both descriptions are valid. In the isochronous network there will be maximum delivery time which is not normally exceeded. The data transmission rate may vary, but if the rate has been low for any reason, it will accelerate to prevent the maximum delay being reached. Isochronous networks can deliver near-real-time performance. If a data buffer is provided at both ends, synchronous data such as AES/EBU audio can be fed through an isochronous network. The magnitude of the maximum delay determines the size of the buffer and the length of the fixed overall delay through the system. This delay is responsible for the term ‘near-real time’. ATM is an isochronous network.

These three different approaches are needed for economic reasons. Asynchronous systems are very efficient because as soon as one transfer completes, another can begin. This can only be achieved by making every device wait with its data in a buffer so that transfer can start immediately. Asynchronous sytems also make it possible for low bit rate devices to share a network with high bit rate devices. The low bit rate device will only need a small buffer and will therefore send short data blocks, whereas the high bit rate device will send long blocks. Asynchronous systems have no dificulty in handling blocks of varying size, whereas in a synchronous system this is very difficult.

Isochronous systems try to give the best of both worlds, generally by sacrificing some flexibility in block size. FireWire is an example of a network which is part isochronous and part asynchronous so that the advantages of both are available.

8.18 Introduction to NICAM 728

This system was developed by the BBC to allow the two additional high-quality digital sound channels to be carried on terrestrial television broadcasts. Performance was such that the system was adopted as the UK standard, and was recommended by the EBU to be adopted by its members, many of whom put it into service.²⁴

The introduction of stereo sound with television cannot be at the expense of incompatibility with the existing monophonic analog sound channel. In NICAM 728 an additional low-power subcarrier is positioned just above the analog sound carrier, which is retained. The relationship is shown in Figure 8.27. The power of the digital subcarrier is about one hundredth that of the main vision carrier, and so existing monophonic receivers will reject it.

Since the digital carrier is effectively shoe-horned into the gap between TV channels, it is necessary to ensure that the spectral width of the intruder is minimized to prevent interference. As a further measure, the power of the existing audio carrier is halved when the digital carrier is present.

Figure 8.27    The additional carrier needed for digital stereo sound is squeezed in between television channels as shown here. The digital carrier is of much lower power than the analog signals, and is randomized prior to transmission so that it has a broad, low-level spectrum which is less visible on the picture.

Figure 8.28 shows the stages through which the audio must pass. The audio sampling rate used is 32 kHz which offers similar bandwidth to that of an FM stereo radio broadcast. Samples are originally quantized to fourteen-bit resolution in two’s complement code. From an analog source this causes no problem, but from a professional digital source having longer wordlength and higher sampling rate it would be necessary to pass through a rate convertor, a digital equalizer to provide pre-emphasis, an optional digital compressor in the case of wide dynamic range signals and then through a truncation circuit incorporating digital dither as explained in Chapter 4.

The fourteen-bit samples are block companded to reduce data rate. During each one millisecond block, 32 samples are input from each audio channel. The magnitude of the largest sample in each channel is independently assessed, and used to determine the gain range or scale factor to be used. Every sample in each channel in a given block will then be scaled by the same amount and truncated to ten bits. An eleventh bit present on each sample combines the scale factor of the channel with parity bits for error detection. The encoding process is described as a Near Instantaneously Companded Audio Multiplex, NICAM for short. The resultant data now consists of 2 32 11 = 704 bits per block. Bit interleaving is employed to reduce the effect of burst errors.

Figure 8.28    The stage necessary to generate the digital subcarrier in NICAM 728. Audio samples are block companded to reduce the bandwidth needed.

At the beginning of each block a synchronizing byte, known as a Frame Alignment Word, is followed by five control bits and eleven additional data bits, making a total of 728 bits per frame, hence the number in the system name. As there are 1000 frames per second, the bit rate is 728 kbits/s. In the UK this is multiplied by 9 to obtain the digital carrier frequency of 6.552 MHz but some other countries use a different subcarrier spacing.

The digital carrier is phase modulated. It has four states which are 90° apart. Information is carried in the magnitude of a phase change which takes place every 18 cycles, or 2.74 μs. As there are four possible phase changes, two bits are conveyed in every change. The absolute phase has no meaning, only the changes are interpreted by the receiver. This type of modulation is known as differentially encoded quadrature phase shift keying (DQPSK), sometimes called four-phase DPSK. In order to provide consistent timing and to spread the carrier energy throughout the band irrespective of audio content, randomizing is used, except during the frame alignment word. On reception, the FAW is detected and used to synchronize the pseudo-random generator to restore the original data.

Figure 8.29 shows the general structure of a frame. Following the sync pattern or FAW is the application control field. The application control bits determine the significance of following data, which can be stereo audio, two independent mono signals, mono audio and data or data only. Control bits C₁, C₂ and C₃ have eight combinations, of which only four are currently standardized. Receivers are designed to mute audio if C₃ becomes 1.

The frame flag bit C₀ spends eight frames high then eight frames low in an endless sixteen-frame sequence which is used to synchronize changes in channel usage. In the last sixteen-frame sequence of the old application, the application control bits change to herald the new application, whereas the actual data change to the new application on the next sixteen frame sequence.

The reserve sound switching flag, C₄, is set to 1 if the analog sound being broadcast is derived from the digital stereo. This fact can be stored by the receiver and used to initiate automatic switching to analog sound in the case of loss of the digital channels.

The additional data bits AD₀ to AD₁₀ are as yet undefined, and reserved for future applications.

The remaining 704 bits in each frame may be either audio samples or data. The two channels of stereo audio are multiplexed into each frame, but multiplexing does not occur in any other case. If two mono audio channels are sent, they occupy alternate frames. Figure 8.29(a) shows a stereo frame, where the A channel is carried in odd-numbered samples, whereas Figure 8.29(b) shows a mono frame, where the M1 channel is carried in odd-numbered frames. The format for data has yet to be defined.

Figure 8.29    In (a) the block structure of a stereo signal multiplexes samples from both channels (A and B) into one block.

Figure 8.29    In mono, shown in (b), samples from one channel only occupy a given block. The diagrams here show the data before interleaving. Adjacent bits shown here actually appear at least sixteen bits apart in the data stream.

The sound/data block of NICAM 728 is in fact identical in structure to the first-level protected companded sound signal block of the MAC/ packet systems.²⁵

8.19 Audio in digital television broadcasting

Digital television broadcasting relies on the combination of a number of fundamental technologies. These are: MPEG-2 compression to reduce the bit rate, multiplexing to combine picture and sound data into a common bitstream, digital modulation schemes to reduce the RF bandwidth needed by a given bit rate and error correction to reduce the error statistics of the channel down to a value acceptable to MPEG data.

MPEG compressed video and audio are both highly sensitive to bit errors, primarily because they confuse the recognition of variable-length codes so that the decoder loses synchronization. However, MPEG is a compression and multiplexing standard and does not specify how error correction should be performed. Consequently a transmission standard must define a system which has to correct essentially all errors such that the delivery mechanism is transparent.

Essentially a transmission standard specifies all the additional steps needed to deliver an MPEG transport stream from one place to another. This transport stream will consist of a number of elementary streams of video and audio, where the audio may be coded according to MPEG audio standard or AC-3. In a system working within its capabilities, the picture and sound quality will be determined only by the performance of the compression system and not by the RF transmission channel. This is the fundamental difference between analog and digital broadcasting. In analog television broadcasting, the picture quality may be limited by composite video encoding artifacts as well as transmission artifacts such as noise and ghosting. In digital television broadcasting the picture quality is determined instead by the compression artifacts and interlace artifacts if interlace has been retained.

If the received error rate increases for any reason, once the correcting power is used up, the system will degrade rapidly as uncorrected errors enter the MPEG decoder. In practice decoders will be programmed to recognize the condition and blank or mute to avoid outputting garbage. As a result, digital receivers tend either to work well or not at all.

It is important to realize that the signal strength in a digital system does not translate directly to picture quality. A poor signal will increase the number of bit errors. Provided that this is within the capability of the error-correction system, there is no visible loss of quality. In contrast, a very powerful signal may be unusable because of similarly powerful reflections due to multipath propagation.

Whilst in one sense an MPEG transport stream is only data, it differs from generic data in that it must be presented to the viewer at a particular rate. Generic data are usually asynchronous, whereas baseband video and audio are synchronous. However, after compression and multiplexing audio and video are no longer precisely synchronous and so the term isochronous is used. This means a signal which was at one time synchronous and will be displayed synchronously, but which uses buffering at transmitter and receiver to accommodate moderate timing errors in the transmission.

Clearly another mechanism is needed so that the time axis of the original signal can be recreated on reception. The time stamp and program clock reference system of MPEG does this.

Figure 8.30 shows that the concepts involved in digital television broadcasting exist at various levels which have an independence not found in analog technology. In a given configuration a transmitter can radiate a given payload data bit rate. This represents the useful bit rate and does not include the necessary overheads needed by error correction, multiplexing or synchronizing. It is fundamental that the transmission system does not care what this payload bit rate is used for. The entire capacity may be used up by one high-definition channel, or a large number of heavily compressed channels may be carried. The details of this data usage are the domain of the transport stream. The multiplexing of transport streams is defined by the MPEG standards, but these do not define any error correction or transmission technique.

Figure 8.30    Source coder doesn’t know delivery mechanism and delivery mechanism doesn’t need to know what the data mean.

Figure 8.31    Program Specific Information helps the demultiplexer to select the required program.

At the lowest level in Figure 8.31 the source coding scheme, in this case MPEG compression, results in one or more elementary streams, each of which carries a video or audio channel. Elementary streams are multiplexed into a transport stream. The viewer then selects the desired elementary stream from the transport stream. Metadata in the transport stream ensures that when a video elementary stream is chosen, the appropriate audio elementary stream will automatically be selected.

8.20 Packets and time stamps

The video elementary stream is an endless bitstream representing pictures which take a variable length of time to transmit. Bidirection coding means that pictures are not necessarily in the correct order. Storage and transmission systems prefer discrete blocks of data and so elementary streams are packetized to form a PES (packetized elementary stream). Audio elementary streams are also packetized. A packet is shown in Figure 8.32. It begins with a header containing an unique packet start code and a code which identifies the type of data stream. Optionally the packet header also may contain one or more time stamps which are used for synchronizing the video decoder to real time and for obtaining lip-sync.

Figure 8.32    A PES packet structure is used to break up the continuous elementary stream.

Figure 8.33    Time stamps are the result of sampling a counter driven by the encoder clock.

Figure 8.33 shows that a time stamp is a sample of the state of a counter which is driven by a 90 kHz clock. This is obtained by dividing down the master 27 MHz clock of MPEG-2. This 27 MHz clock must be locked to the video frame rate and the audio sampling rate of the program concerned. There are two types of time stamp: PTS and DTS. These are abbreviations for presentation time stamp and decode time stamp. A presentation time stamp determines when the associated picture should be displayed on the screen, whereas a decode time stamp determines when it should be decoded. In bidirectional coding these times can be quite different.

Audio packets only have presentation time stamps. Clearly if lip-sync is to be obtained, the audio sampling rate of a given program must have been locked to the same master 27 MHz clock as the video and the time stamps must have come from the same counter driven by that clock.

In practice the time between input pictures is constant and so there is a certain amount of redundancy in the time stamps. Consequently PTS/ DTS need not appear in every PES packet. Time stamps can be up to 100 ms apart in transport streams. As each picture type (I, P or B) is flagged in the bitstream, the decoder can infer the PTS/DTS for every picture from the ones actually transmitted.

8.21 MPEG transport streams

The MPEG-2 transport stream is intended to be a multiplex of many TV programs with their associated sound and data channels, although a single program transport stream (SPTS) is possible. The transport stream is based upon packets of constant size so that multiplexing, adding error-correction codes and interleaving in a higher layer is eased. Figure 8.34 shows that these are always 188 bytes long.

Figure 8.34    Transport stream packets are always 188 bytes long to facilitate multiplexing and error correction.

Transport stream packets always begin with a header. The remainder of the packet carries data known as the payload. For efficiency, the normal header is relatively small, but for special purposes the header may be extended. In this case the payload gets smaller so that the overall size of the packet is unchanged. Transport stream packets should not be confused with PES packets which are larger and which vary in size. PES packets are broken up to form the payload of the transport stream packets.

The header begins with a sync byte which is an unique pattern detected by a demultiplexer. A transport stream may contain many different elementary streams and these are identified by giving each an unique thirteen-bit packet identification code or PID which is included in the header. A multiplexer seeking a particular elementary stream simply checks the PID of every packet and accepts only those which match.

In a multiplex there may be many packets from other programs in between packets of a given PID. To help the demultiplexer, the packet header contains a continuity count. This is a four-bit value which increments at each new packet having a given PID.

This approach allows statistical multiplexing as it does not matter how many or how few packets have a given PID; the demux will still find them. Statistical multiplexing has the problem that it is virtually impossible to make the sum of the input bit rates constant. Instead the multiplexer aims to make the average data bit rate slightly less than the maximum and the overall bit rate is kept constant by adding ‘stuffing’ or null packets. These packets have no meaning, but simply keep the bit rate constant. Null packets always have a PID of 8191 (all ones) and the demultiplexer discards them.

8.22 Clock references

A transport stream is a multiplex of several TV programs and these may have originated from widely different locations. It is impractical to expect all the programs in a transport stream to be genlocked and so the stream is designed from the outset to allow unlocked programs. A decoder running from a transport stream has to genlock to the encoder and the transport stream has to have a mechanism to allow this to be done independently for each program. The synchronizing mechanism is called program clock reference (PCR).

Figure 8.35 shows how the PCR system works. The goal is to re-create at the decoder a 27 MHz clock which is synchronous with that at the encoder. The encoder clock drives a 48-bit counter which continuously counts up to the maximum value before overflowing and beginning again.

A transport stream multiplexer will periodically sample the counter and place the state of the count in an extended packet header as a PCR (see Figure 8.34). The demultiplexer selects only the PIDs of the required program, and it will extract the PCRs from the packets in which they were inserted.

The PCR codes are used to control a numerically locked loop (NLL) described in section 3.16. The NLL contains a 27 MHz VCXO (voltage-controlled crystal oscillator). This is a variable-frequency oscillator based on a crystal which has a relatively small frequency range.

The VCXO drives a 48-bit counter in the same way as in the encoder. The state of the counter is compared with the contents of the PCR and the difference is used to modify the VCXO frequency. When the loop reaches lock, the decoder counter would arrive at the same value as is contained in the PCR and no change in the VCXO would then occur. In practice the transport stream packets will suffer from transmission jitter and this will create phase noise in the loop. This is removed by the loop filter so that the VCXO effectively averages a large number of phase errors.

Figure 8.35    Program or System Clock Reference codes regenerate a clock at the decoder. See text for details.

A heavily damped loop will reject jitter well, but will take a long time to lock. Lock-up time can be reduced when switching to a new program if the decoder counter is jammed to the value of the first PCR received in the new program. The loop filter may also have its time constants shortened during lock-up.

Once a synchronous 27 MHz clock is available at the decoder, this can be divided down to provide the 90 kHz clock which drives the time stamp mechanism.

The entire timebase stability of the decoder is no better than the stability of the clock derived from PCR. MPEG-2 sets standards for the maximum amount of jitter which can be present in PCRs in a real transport stream.

Clearly if the 27 MHz clock in the receiver is locked to one encoder it can only receive elementary streams encoded with that clock. If it is attempted to decode, for example, an audio stream generated from a different clock, the result will be periodic buffer overflows or underflows in the decoder. Thus MPEG defines a program in a manner which relates to timing. A program is a set of elementary streams which have been encoded with the same master clock.

8.23 Program Specific Information (PSI)

In a real transport stream, each elementary stream has a different PID, but the demultiplexer has to be told what these PIDs are and what audio belongs with what video before it can operate. This is the function of PSI which is a form of metadata. Figure 8.36 shows the structure of PSI. When a decoder powers up, it knows nothing about the incoming transport stream except that it must search for all packets with a PID of zero. PID zero is reserved for the Program Association Table (PAT). The PAT is transmitted at regular intervals and contains a list of all the programs in this transport stream. Each program is further described by its own Program Map Table (PMT) and the PIDs of of the PMTs are contained in the PAT.

Figure 8.36 also shows that the PMTs fully describe each program. The PID of the video elementary stream is defined, along with the PID(s) of the associated audio and data streams. Consequently when the viewer selects a particular program, the demultiplexer looks up the program number in the PAT, finds the right PMT and reads the audio, video and data PIDs. It then selects elementary streams having these PIDs from the transport stream and routes them to the decoders.

Figure 8.36    MPEG-2 Program Specific Information (PSI) is used to tell a demultiplexer what the transport stream contains.

Program 0 of the PAT contains the PID of the Network Information Table (NIT). This contains information about what other transport streams are available. For example, in the case of a satellite broadcast, the NIT would detail the orbital position, the polarization, carrier frequency and modulation scheme. Using the NIT a set-top box could automatically switch between transport streams.

Apart from 0 and 8191, a PID of 1 is also reserved for the Conditional Access Table (CAT). This is part of the access control mechanism needed to support pay per view or subscription viewing.

8.24 Multiplexing

A transport stream multiplexer is a complex device because of the number of functions it must perform. A fixed multiplexer will be considered first. In a fixed multiplexer, the bit rate of each of the programs must be specified so that the sum does not exceed the payload bit rate of the transport stream. The payload bit rate is the overall bit rate less the packet headers and PSI rate.

In practice the programs will not be synchronous to one another, but the transport stream must produce a constant packet rate given by the bit rate divided by 188 bytes, the packet length. Figure 8.37 shows how this is handled. Each elementary stream entering the multiplexer passes through a buffer which is divided into payload-sized areas. Note that periodically the payload area is made smaller because of the requirement to insert PCR.

Figure 8.37    A transport stream multiplexer can handle several programs which are asynchronous to one another and to the transport stream clock. See text for details.

MPEG-2 decoders also have a quantity of buffer memory. The challenge to the multiplexer is to take packets from each program in such a way that neither its own buffers nor the buffers in any decoder either overflow or underflow. This requirement is met by sending packets from all programs as evenly as possible rather than bunching together a lot of packets from one program. When the bit rates of the programs are different, the only way this can be handled is to use the buffer contents indicators. The fuller a buffer is, the more likely it should be that a packet will be read from it. Thus a buffer content arbitrator can decide which program should have a packet allocated next.

If the sum of the input bit rates is correct, the buffers should all slowly empty because the overall input bit rate has to be less than the payload bit rate. This allows for the insertion of Program Specific Information. Whilst PATs and PMTs are being transmitted, the program buffers will fill up again. The multiplexer can also fill the buffers by sending more PCRs as this reduces the payload of each packet. In the event that the multiplexer has sent enough of everything but still can’t fill a packet then it will send a null packet with a PID of 8191. Decoders will discard null packets and as they convey no useful data, the multiplexer buffers will all fill whilst null packets are being transmitted.

The use of null packets means that the bit rates of the elementary streams do not need to be synchronous with one another or with the transport stream bit rate. As each elementary stream can have its own PCR, it is not necessary for the different programs in a transport stream to be genlocked to one another; in fact they don’t even need to have the same frame rate.

This approach allows the transport stream bit rate to be accurately defined and independent of the timing of the data carried. This is important because the transport stream bit rate determines the spectrum of the transmitter and this must not vary.

In a statistical multiplexer or statmux, the bit rate allocated to each program can vary dynamically. Figure 8.38 shows that there must be tight connection between the statmux and the associated compressors. Each compressor has a buffer memory which is emptied by a demand clock from the statmux. In a normal, fixed bit rate, coder the buffer content feeds back and controls the requantizer. In statmuxing this process is less severe and only takes place if the buffer is very close to full, because the degree of coding difficulty is also fed to the statmux.

The statmux contains an arbitrator which allocates more packets to the program with the greatest coding difficulty. Thus if a particular program encounters difficult material it will produce large prediction errors and begin to fill its output buffer. As the statmux has allocated more packets to that program, more data will be read out of that buffer, preventing overflow. Of course this is only possible if the other programs in the transport stream are handling typical video.

Figure 8.38    A statistical multiplexer contains an arbitrator which allocates bit rate to each program as a function of program difficulty.

In the event that several programs encounter difficult material at once, clearly the buffer contents will rise and the requantizing mechanism will have to operate.

8.25 Introduction to DAB

Until the advent of NICAM and MAC, all sound broadcasting had been analog. The AM system is now very old indeed, and is not high fidelity by any standards, having a restricted bandwidth and suffering from noise, particularly at night. In theory, the FM system allows high quality and stereo, but in practice things are not so good. Most FM broadcast networks were planned when a radio set was a sizeable unit which usually needed an antenna for AM reception. Signal strengths were based on the assumption that a fixed FM antenna in an elevated position would be used. If such an antenna is used, reception quality is generally excellent. The forward gain of a directional antenna raises the signal above the front-end noise of the receiver and noise-free stereo is obtained. Such an antenna also rejects unwanted signal reflections.

Unfortunately, most FM receivers today are portable radios with whip antennae which have to be carefully oriented to give best reception. It is a characteristic of FM that slight mistuning causes gross distortion so an FM set is harder to tune than an AM set. In many places, nationally broadcast channels can be received on several adjacent frequencies at different strengths. Non-technical listeners tend to find all of this too much and surveys reveal that AM listening is still commonplace despite the same program being available on FM.

Figure 8.39    Multipath reception. When the direct and reflected signals are received with equal strength, nulling occurs at any frequency where the path difference results in a 180° phase shift.

Reception on car radios is at a greater disadvantage as directional antennae cannot be used. This makes reception prone to multipath problems. Figure 8.39 shows that when the direct and reflected signals are received with equal strength, nulling occurs at any frequency where the path difference results in a 180° phase shift. Effectively a comb filter is placed in series with the signal. In a moving vehicle, the path lengths change, and the comb response slides up and down the band. When a null passes through the station tuned in, a burst of noise is created. Reflections from aircraft can cause the same problem in fixed receivers.

Digital audio broadcasting (DAB), also known as digital radio, is designed to overcome the problems which beset FM radio, particularly in vehicles. Not only does it do that, it does so using less bandwidth. With increasing pressure for spectrum allocation from other services, a system using less bandwidth to give better quality is likely to be favourably received.

8.26 DAB principles

DAB relies on a number of fundamental technologies which are combined into an elegant system. Compression is employed to cut the required bandwidth. Transmission of digital data is inherently robust as the receiver has only to decide between a small number of possible states. Sophisticated modulation techniques help to eliminate multipath reception problems whilst further economizing on bandwidth. Error correction and concealment allow residual data corruption to be handled before conversion to analog at the receiver.

The system can only be realized with extremely complex logic in both transmitter and receiver, but with modern VLSI technology this can be inexpensive and reliable. In DAB, the concept of one-carrier-one-program is not used. Several programs share the same band of frequencies. Receivers will be easier to use since conventional tuning will be unnecessary. ‘Tuning’ consists of controlling the decoding process to select the desired program. Mobile receivers will automatically switch between transmitters as a journey proceeds.

Figure 8.40 shows the block diagram of a DAB transmitter. Incoming digital audio at 32 kHz is passed into the compression unit which uses the techniques described in Chapter 5 to cut the data rate to some fraction of the original. The compression unit could be at the studio end of the line to cut the cost of the link. The data for each channel are then protected against errors by the addition of redundancy. Convolutional codes described in Chapter 7 are attractive in the broadcast environment. Several such data-reduced sources are interleaved together and fed to the modulator, which may employ techniques such as randomizing which were introduced in Chapter 6.

Figure 8.40    Block diagram of a DAB transmitter. See text for details.

Figure 8.41 shows how the multiple carriers in a DAB band are allocated to different program channels on an interleaved basis. Using this technique, it will be evident that when a notch in the received spectrum occurs due to multipath cancellation this will damage a small proportion of all programs rather than a large part of one program. This is the spectral equivalent of physical interleaving on a recording medium. The result is the same in that error bursts are broken up according to the interleave structure into more manageable sizes which can be corrected with less redundancy.

Figure 8.41    Channel interleaving is used in DAB to reduce the effect of multipath notches on a given program.

A serial digital waveform has a sinx/x spectrum and when this waveform is used to phase modulate a carrier the result is a symmetrical sinx/x spectrum centred on the carrier frequency. Nulls in the spectrum appear at multiples of the phase switching rate away from the carrier. This distance is equal to 90° or one quadrant of sinx. Further carriers can be placed at spacings such that each is centred at the nulls of the others. Owing to the quadrant spacing, these carries are mutually orthogonal, hence the term orthogonal frequency division.^26,27 A number of such carriers will interleave to produce an overall spectrum which is almost rectangular as shown in Figure 8.42(a). The mathematics describing this process is exactly the same as that of the reconstruction of samples in a low-pass filter and reference should be made to Figure 4.6. Effectively sampling theory has been transformed into the frequency domain.

In practice, perfect spectral interleaving does not give sufficient immunity from multipath reception. In the time domain, a typical reflective environment turns a transmitted pulse into a pulse train extending over several microseconds.²⁸ If the bit rate is too high, the reflections from a given bit coincide with later bits, destroying the orthogonality between carriers. Reflections are opposed by the use of guard intervals in which the phase of the carrier returns to an unmodulated state for a period which is greater than the period of the reflections. Then the reflections from one transmitted phase decay during the guard interval before the next phase is transmitted.²⁹ The principle is not dissimilar to the technique of spacing transitions in a recording further apart than the expected jitter. As expected, the use of guard intervals reduces the bit rate of the carrier because for some of the time it is radiating carrier not data. A typical reduction is to around 80 per cent of the capacity without guard intervals. This capacity reduction does, however, improve the error statistics dramatically, such that much less redundancy is required in the error correction system. Thus the effective transmission rate is improved. The use of guard intervals also moves more energy from the sidebands back to the carrier. The frequency spectrum of a set of carriers is no longer perfectly flat but contains a small peak at the centre of each carrier as shown in Figure 8.42(b).

Figure 8.42    (a) When mutually orthogonal carriers are stacked in a band, the resultant spectrum is virtually flat. (b) When guard intervals are used, the spectrum contains a peak at each channel centre.

A DAB receiver must receive the set of carriers corresponding to the required program channel. Owing to the close spacing of carriers, it can only do this by performing fast Fourier transforms (FFTs) on the DAB band. If the carriers of a given program are evenly spaced, a partial FFT can be used which only detects energy at spaced frequencies and requires much less computation. This is the DAB equivalent of tuning. The selected carriers are then demodulated and combined into a single bit-stream. The error-correction codes will then be de-interleaved so that correction is possible. Corrected data then pass through the expansion part of the data reduction coder, resulting in conventional PCM audio which drives DACs.

It should be noted that in the European DVB standard, COFDM transmission is also used. In some respects, DAB is simply a form of DVB without the picture data.

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