Art of Digital Audio

John Watkinson

• Editorial: Routledge
• Año: 2013
• ISBN:
  9781136117091
• Edición: 3
• Categoría: Technology & Engineering
• Categoría: Acoustical Engineering

Described as "the most comprehensive book on digital audio to date", it is widely acclaimed as an industry "bible". Covering the very latest developments in digital audio technology, it provides an thorough introduction to the theory as well as acting as an authoritative and comprehensive professional reference source. Everything you need is here from the fundamental principles to the latest applications, written in an award-winning style with clear explanations from first principles. New material covered includes internet audio, PC audio technology, DVD, MPEG audio compression, digital audio broadcasting and audio networks.Whether you are in the field of audio engineering, sound recording, music technology, broadcasting and communications media or audio design and installation, this book has it all. Written by a leading international audio specialist, who conducts professional seminars and workshops around the world, the book has been road tested for many years by professional seminar attendees and students to ensure their needs are taken into account, and all the right information is covered.This new edition now includes: Internet audio
PC Audio technology
DVD
MPEG Audio compression
Digital Audio Broadcasting
Audio networksDigital audio professionals will find everything they need here, from the fundamental principles to the latest applications, written in an award-winning style with clear explanations from first principles.John Watkinson is an international consultant in audio, video and data recording. He is a Fellow of the AES, a member of the British Computer Society and a chartered information systems practitioner. He presents lectures, seminars, conference papers and training courses worldwide. He is the author of many other Focal Press books, including: the Kraszna-Krausz award winning MPEG-2; The Art of Digital Audio; An Introduction to Digital Video; The Art of Sound Reproduction; An Introduction to Digital Audio; TV Fundamentals and Audio for Television. He is also co-author, with Francis Rumsey, of The Digital Interface Handbook, and contributor to the Loudspeaker and Headphone Handbook, 3rd edition.

1

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 thr...