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Why digital?

1.1 Introduction

The applications of audio technology are numerous, but generally the goal is to reproduce sound at a later time, at another place or both. The consumer needs reasonably affordable equipment which will reproduce recordings or receive transmissions, whereas the record company or broadcaster needs equipment which can manipulate audio signals to produce recordings or programs. In this case flexibility and speed of operation are more important than first cost.

In one sense provided the sound is reproduced to an acceptable standard, the user doesn’t care how it is done. The point to be stressed is that it is the service that is needed, not the technology. People don’t want technology; instead they want the services it provides. As a result when a new technology comes along it may be adopted if the service is better in some way or if the same service is possible at lower cost or with smaller equipment. Digital audio did just that, irrevocably transforming the face of audio in a very short time for both consumer and professional alike. This is not a history book, but readers interested in the history of digital audio are referred to Chapter 8 of Magnetic Recording: The first 100 years.¹

The first techniques to be used for sound recording, transmission and processing were understandably analog. Some mechanical, electrical or magnetic parameter was caused to vary in the same way that the sound to be recorded had varied the air pressure. The voltage coming from a microphone is an analog of the air pressure (or sometimes velocity), but both vary in the same timescale; the magnetism on a tape or the deflection of a disk groove is an analog of the electrical input signal, but in recorders there is a further analog between time in the input signal and distance along the medium.

In an analog system, information is conveyed by some infinite variation of a continuous parameter such as the voltage on a wire or the strength of flux on a tape. In a recorder, distance along the medium is a further, continuous, analog of time. It does not matter at what point a recording is examined along its length, a value will be found for the recorded signal. That value can itself change with infinite resolution within the physical limits of the system.

Those characteristics are the main weakness of analog signals. Within the allowable bandwidth, any waveform is valid. If the speed of the medium is not constant, one valid waveform is changed into another valid waveform; a timebase error cannot be detected in an analog system. In addition, a voltage error simply changes one valid voltage into another; noise cannot be detected in an analog system. We might suspect noise, but how is one to know what proportion of the received voltage is noise and what is the original? If the transfer function of a system is not linear, distortion results, but the distorted waveforms are still valid; an analog system cannot detect distortion. Again we might suspect distortion, but how are we to know how much of the third harmonic energy received is due to the distortion and how much was actually present in the original signal?

It is a characteristic of analog systems that degradations cannot be separated from the original signal, so nothing can be done about them. At the end of a system a signal carries the sum of all degradations introduced in the stages through which it passed. This sets a limit to the number of stages through which a signal can be passed before it is useless. Alternatively, if many stages are envisaged, each piece of equipment must be far better than necessary so that the signal is still acceptable at the end. The equipment will naturally be more expensive.

When setting out to design any audio equipment, it is important to appreciate that the final arbiter is the human hearing system. If the audio signal is reproduced less accurately than our senses, these shortcomings will be audible, whereas if the system is more accurate than our senses, it will appear perfect even though it is not. Making the system better still is then a waste of resources. This topic will be explored in more detail in Chapters 2 and 13.

1.2 What is digital audio?

One of the vital concepts to grasp is that digital audio is simply an alternative means of carrying audio information. An ideal digital audio recorder has the same characteristics as an ideal analog recorder: both of them are totally transparent and reproduce the original applied waveform without error. One need only compare high-quality analog and digital equipment side by side with the same signals to realize how transparent modern equipment can be. Needless to say, in the real world ideal conditions seldom prevail, so analog and digital equipment both fall short of the ideal. Digital audio simply falls short of the ideal by a smaller distance than does analog and at lower cost, or, if the designer chooses, can have the same performance as analog at much lower cost.

Although there are a number of ways in which audio can be represented digitally, there is one system, known as pulse code modulation (PCM), which is in virtually universal use. Figure 1.1 shows how PCM works. Instead of being continuous, the time axis is represented in a discrete, or stepwise manner. The waveform is not carried by continuous representation, but by measurement at regular intervals. This process is called sampling and the frequency with which samples are taken is called the sampling rate or sampling frequency F_s. The sampling rate is generally fixed and is thus independent of any frequency in the signal. If every effort is made to rid the sampling clock of jitter, or time instability, every sample will be made at an exactly even time step. Clearly if there is any subsequent timebase error, the instants at which samples arrive will be changed and the effect can be detected. If samples arrive at some destination with an irregular timebase, the effect can be eliminated by storing the samples temporarily in a memory and reading them out using a stable, locally generated clock. This process is called timebase correction and all properly engineered digital audio systems must use it. Clearly timebase error is not reduced; it is totally eliminated. As a result there is little point measuring the wow and flutter of a digital recorder; it doesn’t have any. What happens is that the crystal clock in the timebase corrector measures the stability of the flutter meter. It should be stressed that sampling is an analog process. Each sample still varies infinitely as the original waveform did. Sampled analog devices are well known in audio. These are generally implemented with charge-coupled registers and are used for chorus effects in keyboards and for delay in public address systems.

Figure 1.1    In pulse code modulation (PCM) the analog waveform is measured periodically at the sampling rate. The voltage (represented here by the height) of each sample is then described by a whole number. The whole numbers are stored or transmitted rather than the waveform itself.

Those who are not familiar with digital audio often worry that sampling takes away something from a signal because it is not taking notice of what happened between the samples. This would be true in a system having infinite bandwidth, but no analog audio signal can have infinite bandwidth. All analog signal sources such as microphones, tape decks, pickup cartridges and so on have a frequency response limit, as indeed do our ears. When a signal has finite bandwidth, the rate at which it can change is limited, and the way in which it changes becomes predictable. When a waveform can only change between samples in one way, the original waveform can be reconstructed from them. A more detailed treatment of the principle will be given in Chapter 4.

Figure 1.1 also shows that each sample is also discrete, or represented in a stepwise manner. The length of the sample, which will be proportional to the voltage of the audio waveform, is represented by a whole number. This process is known as quantizing and results in an approximation, but the size of the error can be controlled until it is negligible. If, for example, we were to measure the height of humans to the nearest metre, virtually all adults would register two metres high and obvious difficulties would result. These are generally overcome by measuring height to the nearest centimetre. Clearly there is no advantage in going further and expressing our height in a whole number of millimetres or even micrometres, although no doubt some Hi-Fi enthusiasts will be able to advance reasons for doing so. The point is that an appropriate resolution can also be found for audio, and a higher figure is not beneficial. The link between audio quality and sample resolution is explored in Chapter 4.

The advantage of using whole numbers is that they are not prone to drift. If a whole number can be carried from one place to another without numerical error, it has not changed at all. By describing audio waveforms numerically, the original information has been expressed in a way which is better able to resist unwanted changes.

Essentially, digital audio carries the original waveform numerically. The number of the sample is an analog of time, and the magnitude of the sample is an analog of the pressure at the microphone. In fact the succession of samples in a digital system is actually an analog of the original waveform. This sounds like a contradiction and as a result some authorities prefer the term ‘numerical audio’ to ‘digital audio’ and in fact the French word is numérique. The term ‘digital’ is so well established that it is unlikely to change.

As both axes of the digitally represented waveform are discrete, the waveform can accurately be restored from numbers as if it were being drawn on graph paper. If we require greater accuracy, we simply choose paper with smaller squares. Clearly more numbers are then required and each one could change over a larger range.

In simple terms, the audio waveform is conveyed in a digital recorder as if the voltage had been measured at regular intervals with a digital meter and the readings had been written down on a roll of paper. The rate at which the measurements were taken and the accuracy of the meter are the only factors which determine the quality, because once a parameter is expressed as a discrete number, a series of such numbers can be conveyed unchanged. Clearly in this example the handwriting used and the grade of paper have no effect on the information. The quality is determined only by the accuracy of conversion and is independent of the quality of the signal path.

1.3 Why binary?

Humans insist on using numbers expressed to the base of ten, having evolved with that number of digits. Other number bases exist; most people are familiar with the duodecimal system which uses the dozen and the gross. The most minimal system is binary, which has only two digits, 0 and 1. BInary digiTS are universally contracted to bits. These are readily conveyed in switching circuits by an ‘on’ state and an ‘off’ state. With only two states, there is little chance of error.

In decimal systems, the digits in a number (counting from the right, or least significant end) represent ones, tens, hundreds and thousands etc. Figure 1.2 shows that in binary, the bits represent one, two, four, eight, sixteen etc. A multi-digit binary number is commonly called a word, and the number of bits in the word is called the wordlength. The right-hand bit is called the least significant bit (LSB) whereas the bit on the left-hand end of the word is called the most significant bit (MSB). Clearly more digits are required in binary than in decimal, but they are more easily handled. A word of eight bits is called a byte, which is a contraction of ‘by eight’.

The capacity of memories and storage media is measured in bytes, but to avoid large numbers, kilobytes, megabytes and gigabytes are often used. As memory addresses are themselves binary numbers, the wordlength limits the address range. The range is found by raising two to the power of the wordlength. Thus a four-bit word has sixteen combinations, and could address a memory having sixteen locations. A ten-bit word has 1024 combinations, which is close to one thousand. In digital terminology, 1K = 1024, so a kilobyte of memory contains 1024 bytes. A megabyte (1 MB) contains 1024 kilobytes and a gigabyte contains 1024 megabytes.

Figure 1.2    In a binary number, the digits represent increasing powers of two from the LSB. Also defined here are MSB and wordlength. When the wordlength is eight bits, the word is a byte. Binary numbers are used as memory addresses, and the range is defined by the address wordlength. Some examples are shown here.

In a digital audio system, the whole number representing the length of the sample is expressed in binary. The signals sent have two states, and change at predetermined times according to some stable clock. Figure 1.3 shows the consequences of this form of transmission. If the binary signal is degraded by noise, this will be rejected by the receiver, which judges the signal solely by whether it is above or below the half-way threshold, a process known as slicing. The signal will be carried in a channel with finite bandwidth, and this limits the slew rate of the signal; an ideally upright edge is made to slope. Noise added to a sloping signal can change the time at which the slicer judges that the level passed through the threshold. This effect is also eliminated when the output of the slicer is reclocked. However many stages the binary signal passes through, the information is unchange except for a delay.

Audio samples which are represented by whole numbers can reliably be carried from one place to another by such a scheme, and if the number is correctly received, there has been no loss of information en route.

There are two ways in which binary signals can be used to carry audio samples and these are shown in Figure 1.4. When each digit of the binary number is carried on a separate wire this is called parallel transmission. The state of the wires changes at the sampling rate. Using multiple wires is cumbersome, particularly where a long wordlength is in use, and a single wire can be used where successive digits from each sample are sent serially. This is the definition of pulse code modulation. Clearly the clock frequency must now be higher than the sampling rate. Whilst the transmission of audio by such a scheme is advantageous in that noise and timebase error have been eliminated, there is a penalty that a single high-quality audio channel requires around one million bits per second. Digital audio came into wide use as soon as such a data rate could be handled economically. Further applications become possible when means to reduce or compress the data rate become economic. Chapter 5 considers audio compression.

Figure 1.3    (a) A binary signal is compared with a threshold and reclocked on receipt, thus the meaning will be unchanged. (b) Jitter on a signal can appear as noise with respect to fixed timing. (c) Noise on a signal can appear as jitter when compared with a fixed threshold.

Figure 1.4    When a signal is carried in numerical form, either parallel or serial, the mechanisms of Figure 1.3 ensure that the only degradation is in the conversion processes.

1.4 Why digital?

There are two main answers to this question, and it is not possible to say which is the most important, as it will depend on one’s standpoint.

(a)	The quality of reproduction of a well-engineered digital audio system is independent of the medium and depends only on the quality of the conversion processes. If compression is used this can also affect the quality.
(b)	The conversion of audio to the digital domain allows tremendous opportunities which were denied to analog signals.

Someone who is only interested in sound quality will judge the former the most relevant. If good-quality convertors can be obtained, all the shortcomings of analog recording can be eliminated to great advantage. One’s greatest effort is expended in the design of convertors, whereas those parts of the system which handle data need only be workmanlike. Wow, flutter, particulate noise, print-through, dropouts, modulation noise, HF squashing, azimuth error, and interchannel phase errors are all eliminated.

When a digital recording is copied, the same numbers appear on the copy: it is not a dub, it is a clone. If the copy is indistinguishable from the original, there has been no generation loss. Digital recordings can be copied indefinitely without loss of quality. If you happen to be a sound engineer, this is heaven. If you are a record company executive you take another pill for blood pressure and phone your lawyer to see if you can have it stopped.

In the real world everything has a cost, and one of the greatest strengths of digital technology is low cost. If copying causes no quality loss, recorders do not need to be far better than necessary in order to withstand generation loss. They need only be adequate on the first generation whose quality is then maintained. There is no need for the great size and extravagant tape consumption of professional analog recorders. When the information to be recorded is discrete numbers, they can be packed densely on the medium without quality loss. Should some bits be in error because of noise or dropout, error correction can restore the original value. Digital recordings take up less space than analog recordings for the same or better quality. Tape costs are far less and storage costs are reduced.

Digital circuitry costs less to manufacture. Switching circuitry which handles binary can be integrated more densely than analog circuitry. More functionality can be put in the same chip. Analog circuits are built from a host of different component types which have a variety of shapes and sizes and are costly to assemble and adjust. Digital circuitry uses standardized component outlines and is easier to assemble on automated equipment. Little if any adjustment is needed.

Once audio is in the digital domain, it becomes data, and as such is indistinguishable from any other type of data. Systems and techniques developed in other industries for other purposes can be used for audio. Computer equipment is available at low cost because the volume of production is far greater than that of professional audio equipment. Disk drives and memories developed for computers can be put to use in audio products. A word processor adapted to handle audio samples becomes a workstation. There seems to be little point in waiting for a tape to wind when a disk head can access data in milliseconds. The difficulty of locating the edit point and the irrevocable nature of tape-cut editing are immediately seen as outmoded when the edit point can be located by viewing the audio waveform on a screen or by listening at any speed to audio from a memory. The edit can be simulated or previewed and trimmed before it is made permanent.

The merging of digital audio and computation is two-sided. Whilst audio may borrow RAM and hard disk technology from the computer industry, Compact Disc and DAT were borrowed back to create CD-ROM and DDS (digital data storage).

Communications networks developed to handle data can happily carry digital audio over indefinite distances without quality loss. Digital audio broadcasting (DAB) makes use of these techniques to eliminate the interference, fading and multipath reception problems of analog broadcasting. At the same time, more efficient use is made of available bandwidth. In one sense DAB is just conventional radio done with digital transmission. The listener still has to accept what the broadcaster chooses to transmit. In contrast, if the listener uses a data communication channel such as the Internet, any audio program material can in principle be accessed at any time over any distance.

Digital equipment can have self-diagnosis programs built-in. The machine points out its own failures. The days of chasing a signal with an oscilloscope are over. Even if a faulty component in a digital circuit could be located with such a primitive tool, it may be impossible to replace a chip having 60 pins soldered through a six-layer circuit board. The cost of finding the fault may be more than the board is worth. Routine, mind-numbing adjustment of analog circuits to counteract drift is no longer needed. The cost of maintenance falls. A small operation may not need maintenance staff at all; a service contract is sufficient. A larger organization will still need maintenance staff, but they will be fewer in number and their skills will be oriented more to systems than to devices.

As a result of the above, the cost of ownership of digital equipment has for some time now been less than that of analog. Debates about quality are academic; in recording and transmission, analog equipment can no longer compete economically, and it is going out of service as surely as the transistor once replaced the vacuum-tube in electronics and the turbine replaced the piston engine in commercial aviation.

1.5 Some digital audio processes outlined

Whilst digital audio is a large subject, it is not necessarily a difficult one. Every process can be broken down into smaller steps, each of which is relatively easy to follow. The main difficulty with study is to appreciate where the small steps fit in the overall picture. Subsequent chapters of this book will describe the key processes found in digital technology in some detail, whereas this chapter illustrates why these processes are necessary and shows how they are combined in various ways in real equipment. Once the general structure of digital devices is appreciated, the following chapters can be put in perspective.

Figure 1.5(a) shows a minimal digital audio system. This is no more than a point-to-point link which conveys analog audio from one place to another. It consists of a pair of convertors and hardware to serialize and deserialize the samples. There is a need for standardization in serial transmission so that various devices can be connected together. These standards for digital audio interfaces are described in Chapter 7.

Analog audio entering the system is converted in the analog-to-digital convertor (ADC) to samples which are expressed as binary numbers. A typical sample would have a wordlength of sixteen bits. The sample is loaded in parallel into a shift register which is then shifted with a clock running at sixteen times the sampling rate. The data are sent serially to the other end of the line where a slicer rejects noise picked up on the signal. Sliced data are then shifted into a receiving shift register with a bit clock. Once every sixteen bits, the shift register contains a whole sample, and this is read out by the sampling rate clock, or word clock, and sent to the digital-to-analog convertor (DAC), which converts the sample back to an analog voltage.

Following a casual study one might conclude that if the convertors were of transparent quality, the system would be ideal. Unfortunately this is incorrect. As Figure 1.3 showed, noise can change the timing of a sliced signal. Whilst this system rejects noise which threatens to change the numerical value of the samples, it is powerless to prevent noise from causing jitter in the receipt of the word clock. Noise on the word clock means that samples are not converted with a regular timebase and the impairment caused can be audible. Stated another way, analog characteristics of the interconnect are not prevented from affecting the reproduced waveform and so the system is not truly digital.

Figure 1.5    In (a) two convertors are joined by a serial link. Although simple, this system is deficient because it has no means to prevent noise on the clock lines causing jitter at the receiver. In (b) a phase-locked loop is incorporated, which filters jitter from the clock.

The jitter problem is overcome in Figure 1.5(b) by the inclusion of a phase-locked loop which is an oscillator which synchronizes itself to the average frequency of the word clock but which filters out the instantaneous jitter. The operation of a phase-locked loop is analogous to the function of the flywheel on a piston engine. The samples are then fed to the convertor with a regular spacing and the impairment is no longer audible. Chapter 4 shows why the effect occurs and deduces the remarkable clock accuracy needed for accurate conversion.

Whilst this effect is reasonably obvious, it does not guarantee that all convertors take steps to deal with it. Many outboard DACs sold on the consumer market have no phase-locked loop, and one should not be surprised that they can sound worse than the inboard convertor they are supposed to replace. In the absence of timebase correction, the sound quality of an outboard convertor can be affected by such factors as the type of data cable used and the power supply noise of the digital source. Clearly if the sound of a given DAC is affected by cable or source, it is simply not well engineered and should be rejected. Almost by definition a good remote DAC rejects noise and jitter on the digital inputs and its sound is not affected by the digital source or the analog characteristics of the cable.

1.6 The sampler

The system of Figure 1.5 is extended in Figure 1.6 by the addition of some random access memory (RAM). The operation of RAM is described in Chapter 3. What the device does is determined by the way in which the RAM address is controlled. If the RAM address increases by one every time a sample from the ADC is stored in the RAM, a recording can be made for a short period until the RAM is full. The recording can be played back by repeating the address sequence at the same clock rate but reading the memory into the DAC. The result is generally called a sampler. By running the replay clock at various rates, the pitch and duration of the reproduced sound can be altered. At a rate of one million bits per second, a megabyte of memory gives only eight seconds’ worth of recording, so clearly samplers will be restricted to a fairly short playing time.

Figure 1.6    In the digital sampler, the recording medium is a random access memory (RAM). Recording time available is short compared with other media, but access to the recording is immediate and flexible as it is controlled by addressing the RAM.

Using compression, the playing time of a RAM-based recorder can be extended. Some telephone answering machines take messages in RAM and eliminate the cassette tape. For pre-determined messages read only memory (ROM) can be used instead as it is non-volatile. Announcements in aircraft, trains and elevators are one application of such devices. RAM-based recorders are now available which can download suitably compressed audio data over the Internet. Having no moving parts these are highly portable.

1.7 The programmable delay

If the RAM is used in a different way, it can be written and read at the same time. The device then becomes an audio delay. Controlling the relationship between the addresses then changes the delay. The addresses are generated by counters which overflow to zero after they have reached a maximum count. As a result the memory space appears to be circular as shown in Figure 1.7. The read and write addresses are driven by a common clock and chase one another around the circle. If the read address follows close behind the write address, the delay is short. If it just stays ahead of the write address, the maximum delay is reached. Programmable delays are useful in TV studios where they allow audio to be aligned with video which has been delayed in various processes. They can also be used in auditoria to align the sound from various loudspeakers.

Figure 1.7    If the memory address is arranged to come from a counter which overflows, the memory can be made to appear circular. The write address then rotates endlessly, overwriting previous data once per revolution. The read address can follow the write address by a variable distance (not exceeding one revolution) and so a variable delay takes place between reading and writing.

One of the earliest digital audio products was a delay unit of the type shown here which was used to delay the signal leading to a vinyl disk cutter. The cutter control system could use the input to the delay to obtain advance warning of a loud passage and increase the groove pitch accordingly.

1.8 Time compression

When samples are converted, the ADC must run at a constant clock rate and it outputs an unbroken stream of samples. Time compression allows the sample stream to be broken into blocks for convenient handling.

Figure 1.8 shows an ADC feeding a pair of RAMs. When one is being written by the ADC, the other can be read, and vice versa. As soon as the first RAM is full, the ADC output switched to the input of the other RAM so that there is no loss of samples. The first RAM can then be read at a higher clock rate than the sampling rate. As a result the RAM is read in less time than it took to write it, and the output from the system then pauses until the second RAM is full. The samples are now time compressed. Instead of being an unbroken stream which is difficult to handle, the samples are now arranged in blocks with convenient pauses in between them. In these pauses numerous processes can take place. A rotary head recorder might switch heads; a hard disk might move to another track. On a tape recording, the time compression of the audio samples allows time for synchronizing patterns, subcode and error correction words to be recorded.

In digital audio recorders based on video cassette recorders (VCRs) time compression allows the continuous audio samples to be placed in blocks in the unblanked parts of the video waveform, separated by synchronizing pulses.

Subsequently, any time compression can be reversed by time expansion. Samples are written into a RAM at the incoming clock rate, but read out at the standard sampling rate. Unless there is a design fault, time compression is totally inaudible. In a recorder, the time expansion stage can be combined with the timebase correction stage so that speed variations in the medium can be eliminated at the same time. The use of time compression is universal in digital audio recording. In general the instantaneous data rate at the medium is not the same as the rate at the convertors, although clearly the average rate must be the same.

Figure 1.8    In time compression, the unbroken real-time stream of samples from an ADC is broken up into discrete blocks. This is accomplished by the configuration shown here. Samples are written into one RAM at the sampling rate by the write clock. When the first RAM is full, the switches change over, and writing continues into the second RAM whilst the first is read using a higher-frequency clock. The RAM is read faster than it was written and so all the data will be output before the other RAM is full. This opens spaces in the data flow which are used as described in the text.

Another application of time compression is to allow more than one channel of audio to be carried in a single channel. If, for example, audio samples are time compressed by a factor of two, it is possible to carry samples from a stereo source in one cable. In digital video recorders both audio and video data are time compressed so that they can share the same heads and tape tracks.

1.9 Synchronization

Transfer of samples between digital audio devices in real time is only possible if both use a common sampling rate and they are synchronized. A digital audio recorder must be able to synchronize to the sampling rate of a digital input in order to record the samples. It is frequently necessary for such a recorder to be able to play back locked to an external sampling rate reference so that it can be connected to, for example, a digital mixer. The process is already common in video systems but now extends to digital audio. Chapter 8 describes a digital audio reference signal (DARS).

Figure 1.9 shows how the external reference locking process works. The timebase expansion is controlled by the external reference which becomes the read clock for the RAM and so determines the rate at which the RAM address changes. In the case of a digital tape deck, the write clock for the RAM would be proportional to the tape speed. If the tape is going too fast, the write address will catch up with the read address in the memory, whereas if the tape is going too slow the read address will catch up with the write address. The tape speed is controlled by subtracting the read address from the write address. The address difference is used to control the tape speed. Thus if the tape speed is too high, the memory will fill faster than it is being emptied, and the address difference will grow larger than normal. This slows down the tape.

Thus in a digital recorder the speed of the medium is constantly changing to keep the data rate correct. Clearly this is inaudible as properly engineered timebase correction totally isolates any instabilities on the medium from the data fed to the convertor.

In multitrack recorders, the various tracks can be synchronized to sample accuracy so that no timing errors can exist between the tracks. Extra transports can be slaved to the first to the same degree of accuracy if more tracks are required. In stereo recorders image shift due to phase errors is eliminated.

Figure 1.9    In a recorder using time compression, the samples can be returned to a continuous stream using RAM as a timebase corrector (TBC). The long-term data rate has to be the same on the input and output of the TBC or it will lose data. This is accomplished by comparing the read and write addresses and using the difference to control the tape speed. In this way the tape speed will automatically adjust to provide data as fast as the reference clock takes it from the TBC.

In order to replay without a reference, perhaps to provide an analog output, a digital recorder generates a sampling clock locally by means of a crystal oscillator. Provision will be made on professional machines to switch between internal and external references.

1.10 Error correction and concealment

As anyone familiar with analog recording will know, magnetic tape is an imperfect medium. It suffers from noise and dropouts, which in analog recording are audible. In a digital recording of binary data, a bit is either correct or wrong, with no intermediate stage. Small amounts of noise are rejected, but inevitably, infrequent noise impulses cause some individual bits to be in error. Dropouts cause a larger number of bits in one place to be in error. An error of this kind is called a burst error. Whatever the medium and whatever the nature of the mechanism responsible, data are either recovered correctly, or suffer some combination of bit errors and burst errors. In Compact Disc and DVD, random errors can be caused by imperfections in the moulding process, whereas burst errors are due to contamination or scratching of the disc surface.

The audibility of a bit error depends upon which bit of the sample is involved. If the LSB of one sample was in error in a loud passage of music, the effect would be totally masked and no-one could detect it. Conversely, if the MSB of one sample was in error in a quiet passage, noone could fail to notice the resulting loud transient. Clearly a means is needed to render errors from the medium inaudible. This is the purpose of error correction.

In binary, a bit has only two states. If it is wrong, it is only necessary to reverse the state and it must be right. Thus the correction process is trivial and perfect. The main difficulty is in reliably identifying the bits which are in error. This is done by coding the data by adding redundant bits. Adding redundancy is not confined to digital technology, airliners have several engines and cars have twin braking systems. Clearly the more failures which have to be handled, the more redundancy is needed. If a four-engined airliner is designed to fly normally with one engine failed, three of the engines have enough power to reach cruise speed, and the fourth one is redundant. The amount of redundancy is equal to the amount of failure which can be handled. In the case of the failure of two engines, the plane can still fly, but it must slow down; this is graceful degradation. Clearly the chances of a two-engine failure on the same flight are remote.

In digital audio, the amount of error which can be corrected is proportional to the amount of redundancy, and it will be shown in Chapter 7 that within this limit, the samples are returned to exactly their original value. Consequently corrected samples are audibly indistinguishable from the originals. If the amount of error exceeds the amount of redundancy, correction is not possible, and, in order to allow graceful degradation, concealment will be used. Concealment is a process where the value of a missing sample is estimated from those nearby. The estimated sample value is not necessarily exactly the same as the original, and so under some circumstances concealment can be audible, especially if it is frequent. However, in a well-designed system, concealments occur with negligible frequency unless there is an actual fault or problem.

Concealment is made possible by rearranging or shuffling the sample sequence prior to recording. This is shown in Figure 1.10 where odd-numbered samples are separated from even-numbered samples prior to recording. The odd and even sets of samples may be recorded in different places, so that an uncorrectable burst error only affects one set. On replay, the samples are recombined into their natural sequence, and the error is now split up so that it results in every other sample being lost. The waveform is now described half as often, but can still be reproduced with some loss of accuracy. This is better than not being reproduced at all even if it is not perfect. Almost all digital recorders use such an odd/even shuffle for concealment. Clearly if any errors are fully correctable, the shuffle is a waste of time; it is only needed if correction is not possible.

Figure 1.10    In cases where the error correction is inadequate, concealment can be used provided that the samples have been ordered appropriately in the recording. Odd and even samples are recorded in different places as shown here. As a result an uncorrectable error causes incorrect samples to occur singly, between correct samples. In the example shown, sample 8 is incorrect, but samples 7 and 9 are unaffected and an approximation to the value of sample 8 can be had by taking the average value of the two. This interpolated value is substituted for the incorrect value.

In high-density recorders, more data are lost in a given sized dropout. Adding redundancy equal to the size of a dropout to every code is inefficient. Figure 1.11 shows that the efficiency of the system can be raised using interleaving. Sequential samples from the ADC are assembled into codes, but these are not recorded in their natural sequence. A number of sequential codes are assembled along rows in a memory. When the memory is full, it is copied to the medium by reading down columns. On replay, the samples need to be de-interleaved to return them to their natural sequence. This is done by writing samples from tape into a memory in columns, and when it is full, the memory is read in rows. Samples read from the memory are now in their original sequence so there is no effect on the recording. However, if a burst error occurs on the medium, it will damage sequential samples in a vertical direction in the de-interleave memory. When the memory is read, a single large error is broken down into a number of small errors whose size is exactly equal to the correcting power of the codes and the correction is performed with maximum efficiency.

Figure 1.11    In interleaving, samples are recorded out of their normal sequence by taking columns from a memory which was filled in rows. On replay the process must be reversed. This puts the samples back in their regular sequence, but breaks up burst errors into many smaller errors which are more efficiently corrected. Interleaving and de-interleaving cause delay.

The interleave, de-interleave, time compression and timebase correction processes cause delay and this is evident in the time taken before audio emerges after starting a digital machine. Confidence replay takes place later than the distance between record and replay heads would indicate. In stationary head recorders, confidence replay may be about one tenth of a second behind the input. Synchronous recording requires new techniques to overcome the effect of the delays.

The presence of an error-correction system means that the audio quality is independent of the tape/head quality within limits. There is no point in trying to assess the health of a machine by listening to it, as this will not reveal whether the error rate is normal or within a whisker of failure. The only useful procedure is to monitor the frequency with which errors are being corrected, and to compare it with normal figures. Professional digital audio equipment should have an error rate display.

Some people claim to be able to hear error correction and misguidedly conclude that the above theory is flawed. Not all digital audio machines are properly engineered, however, and if the DAC shares a common power supply with the error- correction logic, a burst of errors will raise the current taken by the logic, which in turn loads the power supply and interferes with the operation of the DAC. The effect is harder to eliminate in small battery-powered machines where space for screening and decoupling components is difficult to find, but it is only a matter of good engineering; there is no flaw in the theory.

1.11 Channel coding

In most recorders used for storing digital information, the medium carries a track which reproduces a single waveform. The audio samples have to be recorded serially, one bit at a time. Some media, such as CD, only have one track, so it must be totally self-contained. Other media, such as digital compact cassette (DCC) have many parallel tracks. At high recording densities, physical tolerances cause phase shifts, or timing errors, between parallel tracks and so it is not possible to read them in parallel. Each track must still be self-contained until the replayed signal has been timebase corrected.

Recording data serially is not as simple as connecting the serial output of a shift register to the head. In digital audio, a common sample value is all zeros, as this corresponds to silence. If a shift register is loaded with all zeros and shifted out serially, the output stays at a constant low level, and no events are recorded on the track. On replay there is nothing to indicate how many zeros were present, or even how fast to move the medium. Clearly serialized raw data cannot be recorded directly, it has to be modulated into a waveform which contains an embedded clock irrespective of the values of the bits in the samples. On replay a circuit called a data separator can lock to the embedded clock and use it to count and separate strings of identical bits.

The process of modulating serial data to make it self-clocking is called channel coding. Channel coding also shapes the spectrum of the serialized waveform to make it more efficient. With a good channel code, more data can be stored on a given medium. Spectrum shaping is used in optical disks to prevent the data from interfering with the focus and tracking servos, and in DAT to allow rerecording without erase heads.

Channel coding is also needed to broadcast digital information where shaping of the spectrum is an obvious requirement to avoid interference with other services. NICAM TV sound, digital video broadcasting (DVB) and digital audio broadcasting (DAB) rely on it.

All the techniques of channel coding are covered in detail in Chapter 6 and digital broadcasting is considered in Chapter 9.

1.12 Compression

The human hearing system comprises not only the physical organs but also processes taking place within the brain. One of purposes of the subconscious processing is to limit the amount of information presented to the conscious mind, to prevent stress and to make everyday life safer and easier. Chapter 2 shows how auditory masking selects only the most important frequencies from the spectrum applied to the ear. Compression takes advantage of this process to reduce the amount of data needed to carry sound of a given subjective quality. The data-reduction process mimics the operation of the hearing mechanism as there is little point in recording information only for the ear to discard it. Compression is explained in detail in Chapter 5.

Compression is essential for services such as DAB and DVB where the bandwidth needed to broadcast regular PCM would be excessive. It can be used to reduce consumption of the medium in consumer recorders such as DCC and MiniDisc. Reduction to around one quarter or one fifth of the PCM data rate with small loss of quality is possible with high-quality compression systems. Greater compression factors inevitably result in quality loss which may be acceptable for certain applications such as communications but not for quality music reproduction.

The output of a compressor is called an elementary stream. This is still binary data, but it is no longer regular PCM, so it cannot be fed to a normal DAC without passing through a matching decoder which provides a conventional PCM output. Compressed data are more sensitive to bit errors than PCM data and concealment is more complex to implement.

There are numerous proprietary compression algorithms, and each needs the appropriate decoder to return to PCM. The combination of a compressor and a decoder is called a codec. The performance of a codec is tested on a single pass, as it would be for use in DAB or in a single-generation recording. The same performance is not necessarily obtained if codecs are cascaded, particularly if they are of different types. If an equalization step is performed on audio which has been through a codec, artifacts may be raised above the masking threshold. As a result, compression may not be suitable for the recording of original material prior to post-production.

1.13 Hard disk recorders

The hard disk stores data on concentric tracks which it accesses by moving the head radially. Clearly while the head is moving it cannot transfer data. Using time compression, a hard disk drive can be made into an audio recorder with the addition of a certain amount of memory.

Figure 1.12 shows the principle. The instantaneous data rate of the disk drive is far in excess of the sampling rate at the convertor, and so a large time-compression factor can be used. The disk drive can read a block of data from disk, and place it in the timebase corrector in a fraction of the real time it represents in the audio waveform. As the timebase corrector read address steadily advances through the memory, the disk drive has time to move the heads to another track before the memory runs out of data. When there is sufficient space in the memory for another block, the drive is commanded to read, and fills up the space. Although the data transfer at the medium is highly discontinuous, the buffer memory provides an unbroken stream of samples to the DAC and so continuous audio is obtained.

Figure 1.12    In a hard disk recorder, a large-capacity memory is used as a buffer or timebase corrector between the convertors and the disk. The memory allows the convertors to run constantly despite the interruptions in disk transfer caused by the head moving between tracks.

Recording is performed by using the memory to assemble samples until the contents of one disk block is available. This is then transferred to disk at high data rate. The drive can then reposition the head before the next block is available in memory.

An advantage of hard disks is that access to the audio is much quicker than with tape, as all the data are available within the time taken to move the head. This speeds up editing considerably. As hard disks offer so much to digital audio, the entirety of Chapter 10 is devoted to them.

The use of compression allows the recording time of a disk to be extended considerably. This technique is often used in personal computers or organizers to allow them to function as a recorder.

1.14 The PCM adaptor

The PCM adaptor was an early solution to recording the wide bandwidth of PCM audio before high density digital recording developed. The video recorder offered sufficient bandwidth at moderate tape consumption. Whilst they were a breakthrough at the time of their introduction, by modern standards PCM adaptors are crude and obsolescent. Figure 1.13 shows the essential components of a digital audio recorder using this technique. Input analog audio is converted to digital and time compressed to fit into the parts of the video waveform which are not blanked. Time-compressed samples are then odd–even shuffled to allow concealment. Next, redundancy is added and the data are interleaved for recording. The data are serialized and set on the active line of the video signal as black and white levels shown in Figure 1.14. The video is sent to the recorder, where the analog FM modulator switches between two frequencies representing the black and white levels, a system called frequency shift keying (FSK). This takes the place of the channel coder in a conventional digital recorder.

Figure 1.13    Block diagrams of PCM adaptor. Note the dub connection needed for producing a digital copy between two VCRs.

On replay the FM demodulator of the video recorder acts to return the FSK recording to the black/white video waveform which is sent to the PCM adaptor. The PCM adaptor extracts a clock from the video sync pulses and uses it to separate the serially recorded bits. Error correction is performed after de-interleaving, unless the errors are too great, in which case concealment is used after the de-shuffle. The samples are then returned to the standard sampling rate by the timebase expansion process, which also eliminates any speed variations from the recorder. They can then be converted back to the analog domain.

Figure 1.14    Typical line of video from PCM-1610. The control bit conveys the setting of the pre-emphasis switch or the sampling rate depending on position in the frame. The bits are separated using only the timing information in the sync pulses.

In order to synchronize playback to a reference and to simplify the circuitry, a whole number of samples is recorded on each unblanked line. The common sampling rate of 44.1 kHz is obtained by recording three samples per line on 245 active lines at 60 Hz. The sampling rate is thus locked to the video sync frequencies and the tape is made to move at the correct speed by sending the video recorder syncs which are generated in the PCM adaptor.

1.15 An open-reel digital recorder

Figure 1.15 shows the block diagram of a machine of this type. Analog inputs are converted to the digital domain by convertors. Clearly there will be one convertor for every audio channel to be recorded. Unlike an analog machine, there is not necessarily one tape track per audio channel. In stereo machines the two channels of audio samples may be distributed over a number of tracks each in order to reduce the tape speed and extend the playing time.

Figure 1.15    Block diagram of one channel of a stationary-head digital audio recorder. See text for details of the function of each block. Note the connection from the timebase corrector to the capstan motor so that the tape is played at such a speed that the TBC memory neither underflows nor overflows.

The samples from the convertor will be separated into odd and even for concealment purposes, and usually one set of samples wil be delayed with respect to the other before recording. The continous stream of samples from the convertor will be broken into blocks by time compression prior to recording. Time compression allows the insertion of edit gaps, addresses and redundancy into the data stream. An interleaving process is also necessary to reorder the samples prior to recording. As explained above, the subsequent de-interleaving breaks up the effects of burst errors on replay.

The result of the processes so far is still raw data, and these will need to be channel coded before they can be recorded on the medium. On replay a data separator reverses the channel coding to give the original raw data with the addition of some errors. Following de-interleave, the errors are reduced in size and are more readily correctable. The memory required for de-interleave may double as the timebase correction memory, so that variations in the speed of the tape are rendered undetectable. Any errors which are beyond the power of the correction system will be concealed after the odd–even shift is reversed. Following conversion in the DAC an analog output emerges.

On replay a digital recorder works rather differently from an analog recorder, which simply drives the tape at constant speed. In contrast, a digital recorder drives the tape at constant sampling rate. The timebase corrector works by reading samples out to the convertor at constant frequency. This reference frequency comes typically from a crystal oscillator. If the tape goes too fast, the memory will be written faster than it is being read, and will eventually overflow. Conversely, if the tape goes too slow, the memory will become exhausted of data. In order to avoid these problems, the speed of the tape is controlled by the quantity of data in the memory. If the memory is filling up, the tape slows down, if the memory is becoming empty, the tape speeds up. As a result, the tape will be driven at whatever speed is necessary to obtain the correct sampling rate.

1.16 Rotary head digital recorders

The rotary head recorder borrows technology from videorecorders. Rotary heads have a number of advantages which will be detailed in Chapter 9. One of these is extremely high packing density: the number of data bits which can be recorded in a given space. In a digital audio recorder packing density directly translates into the playing time available for a given size of the medium.

Figure 1.16    In a rotary-head recorder, the helical tape path around a rotating head results in a series of diagonal or slanting tracks across the tape. Time compression is used to create gaps in the recorded data which coincide with the switching between tracks.

In a rotary head recorder, the heads are mounted in a revolving drum and the tape is wrapped around the surface of the drum in a helix as can be seen in Figure 1.16. The helical tape path results in the heads traversing the tape in a series of diagonal or slanting tracks. The space between the tracks is controlled not by head design but by the speed of the tape and in modern recorders this space is reduced to zero with corresponding improvement in packing density.

The added complexity of the rotating heads and the circuitry necessary to control them is offset by the improvement in density. These techniques are detailed in Chapter 8. The discontinuous tracks of the rotary head recorder are naturally compatible with time compressed data. As Figure 1.16 illustrates, the audio samples are time compressed into blocks each of which can be contained in one slant track.

In a machine such as DAT (rotary-head digital audio tape) there are two heads mounted on opposite sides of the drum. One rotation of the drum lays down two tracks. Effective concealment can be had by recording odd-numbered samples on one track of the pair and even-numbered samples on the other.

As can be seen from the block diagram shown in Figure 1.17, a rotary head recorder contains the same basic steps as any digital audio recorder. The record side needs ADCs, time compression, the addition of redundancy for error correction, and channel coding. On replay the channel coding is reversed by the data separator, errors are broken up by the de-interleave process and corrected or concealed, and the time compression and any fluctuations from the transport are removed by timebase correction. The corrected, time stable, samples are then fed to the DAC.

Figure 1.17    Block diagram of DAT

1.17 Digital Compact Cassette

Digital Compact Cassette (DCC) is a consumer digital audio recorder using compression. Although the convertors at either end of the machine work with PCM data, these data are not directly recorded, but are reduced to one quarter of their normal rate by processing. This allows a reasonable tape consumption similar to that achieved by a rotary head recorder. In a sense the complexity of the rotary head transport has been exchanged for the electronic complexity of the compression and expansion circuitry.

Figure 1.18    In DCC audio and auxiliary data are recorded on nine parallel tracks along each side of the tape as shown in (a). The replay head shown in (b) carries magnetic poles which register with one set of nine tracks. At the end of the tape, the replay head rotates 180° and plays a further nine tracks on the other side of the tape. The replay head also contains a pair of analog audio magnetic circuits which will be swung into place if an analog cassette is to be played.

Figure 1.18 shows that DCC uses stationary heads in a conventional tape transport which can also play analog cassettes. Data are distributed over eight parallel tracks which occupy half the width of the tape. At the end of the tape the head rotates and plays the other eight tracks in reverse. The advantage of the conventional approach with linear tracks is that tape duplication can be carried out at high speed. This makes DCC attractive to record companies.

Owing to the low frequencies recorded, DCC has to use active heads which actually measure the flux on the tape. These magneto-resistive heads are more complex than conventional inductive heads, and have only recently become economic as manufacturing techniques have been developed. DCC is treated in detail in Chapter 9.

As was introduced in section 1.12, compression relies on the phenomenon of auditory masking and this may effectively restrict DCC to being a consumer format. It will be seen from Figure 1.19 that the compression unit adjacent to the input is complemented by the expansion unit or decoder prior to the DAC.

1.18 Digital audio broadcasting

Digital audio broadcasting operates by modulating the transmitter with audio data instead of an analog waveform. Analog modulation works reasonably well for fixed reception sites where a decent directional antenna can be erected at a selected location, but has serious short-comings for mobile reception where there is no control over the location and a large directional antenna is out of the question. The greatest drawback of broadcasting is multipath reception, where the direct signal is received along with delayed echoes from large reflecting bodies such as high-rise buildings. At certain wavelengths the reflection is received antiphase to the direct signal, and cancellation takes place which causes a notch in the received spectrum. In an analog system loss of the signal is inevitable.

Figure 1.19    In DCC, the PCM data from the convertors are reduced to one-quarter of the original rate prior to distribution over eight tape tracks (plus an auxiliary data track). This allows a slow linear tape speed which can only be read with an MR head. The compression unit is mirrored by the decoder on replay.

In DAB, several digital audio broadcasts are merged into one transmission which is wider than the multipath notches. The data from the different signals are distributed uniformly within the channel so that a notch removes a small part of each channel instead of all of one. Sufficient data are received to allow error correction to re-create the missing values.

A DAB receiver actually receives the entire transmission and the process of ‘tuning in’ the desired channel is now performed by selecting the appropriate data channel for conversion to analog, making a DAB receiver easier to operate.

DAB resists multipath reception to permit mobile reception and the improvement to reception in car radios is dramatic. The data rate of PCM audio is too great to allow it to be economic for DAB. Compression is essential and this is detailed in Chapter 5.

1.19 Audio in PCs

Whilst the quality digital audio permits in undeniable, the potential of digital audio may turn out to be more important in the long term. Once audio becomes data, there is tremendous freedom to store and process it in computer-related equipment. The restrictions of analog technology are no longer applicable, yet we often needlessly build restrictions into equipment by making a digital replica of an analog system. The analog system evolved to operate within the restrictions imposed by the technology. To take the same system and merely digitize it is to miss the point.

A good example of missing the point was the development of the stereo quarter-inch digital audio tape recorder with open reels. Open-reel tape is sub-optimal for high-density digital recording because it is unprotected from contamination. The recorded wavelengths must be kept reasonably long or the reliability will be poor. Thus the tape consumption of these machines was excessive and more efficient cassette technologies such as DAT proved to have lower purchase cost and running costs as well as being a fraction of the size and weight. The speed and flexibility with which editing could be carried out by hard disk systems took away any remaining advantage. Quarter-inch digital tape found itself trapped between DAT and hard disks and passed into history because it was the wrong approach.

Part of the problem of missed opportunity is that traditionally, professional audio equipment manufacturers have specialized in one area leaving users to assemble systems from several suppliers. Mixer manufacturers may have no expertise in recording. Tape recorder manufacturers may have no knowledge of disk drives.

In contrast, computer companies have always taken a systems view and configure disks, tapes, RAM, processors and communications links as necessary to meet a given requirement. Now that audio is another form of data, this approach is being used to solve audio problems.

Small notebook computers are increasingly available with microphones and audio convertors so that they can act as dictating machines. A personal computer with high-quality audio convertors, compression algorithms and sufficient disk storage becomes an audio recorder. The recording levels and the timer are displayed on screen and soft keys become the rewind, record, etc. controls for the virtual recorder. The recordings can be edited to sample accuracy on disk, with displays of the waveforms in the area of the in and out points on screen. Once edited, the audio data can be sent anywhere in the world using telephone modems and data networks. The PC can be programmed to dial the destination itself at a selected time. At the same time as sending the audio, text files can be sent, along with images from a CCD camera. Without digital technology such a device would be unthinkable.

The market for such devices may well be captured by those with digital backgrounds, but not necessarily in audio. Computer, calculator and other consumer electronics manufacturers have the wider view of the potential of digital techniques.

Digital also blurs the distinction between consumer and professional equipment. In the traditional analog audio world, professional equipment sounded better but cost a lot more than consumer equipment. Now that digital technology is here, the sound quality is determined by the convertors. Once converted, the audio is data. If a bit can only convey whether it is one or zero, how does it know if it is a professional bit or a consumer bit? What is a professional disk drive? The cost of a digital product is a function not of its complexity, but of the volume to be sold. Professional equipment may be forced to use chip sets and transports designed for the volume market because the cost of designing an alternative is prohibitive. A professional machine may be a consumer machine in a stronger box with XLRs instead of phono sockets and PPM level meters. It may be that there will be little room for traditional professional audio manufacturers in the long term.

1.20 Networks

The conventional analog routing structure used in professional installations was simply replicated in the digital domain by the AES/EBU digital audio interface. However, using computer data approaches digital audio routing can also be achieved using networks, interconnecting a number of file servers which store the audio data with workstations from which the recordings can be manipulated. No dedicated audio routeing hardware is required. Chapter 8 considers how data networks operate.

Reference

1.	Daniel, E.D., Mee, C.D. and Clark, M.H. (eds), Magnetic Recording: The first 100 years, Piscataway: IEEE Press (1999)