Introduction

Not available.

Paper Number:   Rye-001    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Blesser, Barry A.

Affiliation:   Blesser Associates, Raymond, NH

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Elementary and Basic Aspects of Digital Audio

This first paper of the conference on digital audio is intended for those who prefer a basic introduction. The goal of the conference is to bring all participants to a high level, but we begin at the beginning. Some will feel free to sip this paper or else read it as entertainment.: Digital audio has only a few basic ideas that may be unfamiliar. All of the power of digital audio derives from these few ideas. The problem becomes large only when we wish to implement very high performance systems performing very complex tasks. Begin mastering digital audio with an open mind, and you will find that it is really quite simple.

Paper Number:   Rye-002    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Blesser, Barry A.

Affiliation:   Blesser Associates, Raymond, NH

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The Promise of Digital Audio

The promise of digital audio based on its singular advantages in the areas of permanence and fidelity are presented. Emphasis is placed on considering digital recording as a new medium, not just a new technology. Detailed aspects of the points of advantage will be disclosed. To provide balance, disadvantages, most of which will be overcome with time, will also be reviewed. The final part of the paper includes an elementary discussion of several aspects of digital recording and transmission.

Paper Number:   Rye-003    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Stockham, Jr., Thomas G.

Affiliation:   Soundstream, Inc., Salt Lake City, UT

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Recent Progress in Digital Audio Technology

This paper reports on recent progress in digital audio technology, including improvements in analog-to-digital (A/D) and digital-to-analog (D/A) converters, magnetic-recording-head design, modulation schemes, error-correction schemes, and editing processes. The improvements are remarkable and have contributed greatly to cost reduction, better sound quality, reliable higher packing density, and increased versatility. Newly developed digital audio systems that include professional recorders, satellite broadcasting, and digital audio disks are described. Large-scale integrated (LSI) circuits developed for consumer systems and their impact on digital audio are also reviewed.

Paper Number:   Rye-004    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Doi, Toshi T.

Affiliation:   Sony Corporation, Tokyo, Japan

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Advanced Analog-to-Digital Conversion and Filtering: Data Conversion

The quality of all digital audio systems is limited by those degradations introduced in the analog function of conversion to and from the digital domain. Anti-aliasing and anti-image filters are a difficult technology when the specifications correspond to 16-bit conversion systems. The various theoretical and practical limitations of the different filter structures will be reviewed with reference to internal distortion, passband ripple, stopband attenuation, and phase nonlinearity. The different classes of filters are: biquadratic cascade, and ladder structures with either passive or active components. Some of the newer theory results will be presented.: Analog-to-digital and digital-to-analog technology has made significant advances during the last decade. There are now several different classes of converters in the analog-to-digital family: flash, successive approximation, successive approximation by residual expansion, and dual slope count algorithms. Some manufacturers are now using self-calibration modes internally. Because almost none of the so-called 16-bit converters will achieve 95 dB signal-to-noise ratio, it is important to understand the nature of device limitations.

Paper Number:   Rye-005    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Blesser, Barry A.

Affiliation:   Blesser Associates, Raymond, NH

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A Monolithic 16-Bit D/A Conversion System for Digital Audio

A monolithic 16-bit conversion system including a high-speed digital-to-analog (D/A) converter and a phase-linear 96-element digital low-pass filter operating at four times the sampling frequency is described. In the D/A converter a special technique, called dynamic element matching, is used to obtain high-Z accuracy and stability. The linearity of the converter is maintained at 1/2 LSB over a long time and over a large temperature range. Only a simple third-order analog Bessel filter is needed to remove the high-frequency components of the output signal. The total system has a dynamic image of 96 dB with a full scale distortion level at -90 dB. Pass-band ripple is within 0.15 dB and phase linearity within 0.5 degrees. All specifications are valid over the 20-kHz audio band.

Paper Number:   Rye-006    Conference:   1st International Conference: Digital Audio (June 1982)

Authors:   van de Plassche, R. J.; Dijkmans, E. C.

Affiliation:   Philips Research Laboratories, 5600 JA Eindhoven, The Netherlands

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Progress and Pitfalls Associated with Scientific Measures of Auditory Acuity

Psychophysical data concerning the functioning of the ear and auditory portions of the nervous system have been obtained using a large variety of stimuli in tasks which require that people either detect, discriminate, identify, scale, or locate sounds. The myriad measures of auditory acuity or functioning obtained in such tasks have been used to infer how well people can process auditory information.: We now know that there can be great disparities between measures obtained across seemingly similar tasks and that we must be capable of distinguishing how well people typically do perform from how well they can perform in controlled or -optimal- situations.: An effort is made to illustrate and to assess the importance of several factors that determine obtained measures of auditory acuity including individual differences, practice effects, musical training, attention, and the paradigm used to collect the data.: The purpose of the discussion is to acquaint audio engineers with a portion of a large body of knowledge that appears to be applicable and useful to those who desire to use human judgments to evaluate high-fidelity systems and components.

Paper Number:   Rye-007    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Trahiotis, Constantine

Affiliation:   University of Illinois, Departments of Speech and Hearing Science and Psychology, Champaign, IL

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Digital Audio Impairments and Measures

In order to ensure the satisfactory performance of digital audio apparatus it is necessary to pay particular attention to providing a dynamic range which is sufficient for the audio signals, and to minimizing the errors and distortions which cause impairment. Coding range requirements for digital audio signals, and the audible impairments resulting from errors in the analog-digital conversion processes, from program-modulated noise and from bit errors have been studied by the BBC, and these are discussed in the paper.

Paper Number:   Rye-008    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Gilchrist, Neil H.

Affiliation:   British Broadcasting Corporation, Tadworth, Surrey, United Kingdom

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Digital Techniques for Changing the Sampling Rate of a Signal

Digital signals are often available in a wide variety of formats and with different sampling rates. : For maximum flexibility in processing such signals it is important to be able to digitally change the sampling rate of an incoming signal to almost any desired rate. The theory is reviewed, and some practical implementations of digital systems are discussed which can change the sampling rate of a signal. The Nyquist sampling theorem is the basis for all sampling rate conversion techniques. It is shown how a simple, straightforward application of the sampling theorem leads to the canonical system for sampling rate conversion. Considerations in the implementation of a sampling rate converter, including structures, filter designs, ad cascading techniques, are briefly discussed. Finally an example is given of a signal processing system based on sampling rate conversion.

Paper Number:   Rye-009    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Rabiner, L. R.

Affiliation:   Bell Laboratories, Murray Hill, NJ

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Digital Sampling Frequency Conversion

Many applications in digital audio raise the issue of changing the sampling frequency of a digital audio signal without otherwise modifying its audio content or its pitch. Economics and signal quality make it intuitively appealing to look for a purely digital solution. The requirements for such a solution are severe. For generality, arbitrary sampling frequencies should be tolerated, including a prior unknown sampling frequencies which may drift with time. In addition, the signal quality of 16-bit and possibly even 24-bit digital audio must be preserved in conversion. Finally, automatic bandwidth adjustment as a function of the sampling frequencies is a natural requirement if aliasing effects are to be prevented. A new approach to sampling frequency conversion, based exclusively on digital signal processing and meeting all the above requirements, is presented.

Paper Number:   Rye-010    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Lagadec, Roger

Affiliation:   Willi Studer, Regensdorf, Switzerland

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The State of the Art in High-Density Magnetic Recording

Direct magnetic recording of audio results in the most efficient use of the recording-channel capacity from an information theoretic point of view - far more efficient than has been achieved by digital, video, or telemetry recorders. The limitations of baseband recording, however, have dictated a move to digital recording to remove the problems of nonlinearity, phase distortion, multiple generation noise, and so on. Unfortunately digital recording currently results in a large increase in the amount of tape needed relative to baseband recording. However, recent developments in the magnetic recording art will improve that situation and, in fact, will actually require less area of tape per second than baseband recorders.: Perpendicular recording and vector field recording (with isotropic particles and microgap heads) have permitted very large increases in lineal density and signal-to-noise ratio. By narrowing the tracks of such systems to spend some of the surplus signal-to-noise ratio, densities comparable to those of optical recorders are projected. An FM audio system utilizing isotropic recording at normal cassette speed is described. The advent of high bit densities at low cost makes very simple and effective error-correcting codes attractive.

Paper Number:   Rye-011    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Lemke, James U.

Affiliation:   Eastman Kodak Company, San Diego, CA

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Tape Formats and Multitrack Formats

The tape format of a digital audio recorder is directly related to its capabilities for error correction, editing, and punch-in and punch-out. A digital recorder can be classified as a stationary-head or rotary-head machine. Various tape formats, limited by the physical constraints of the recording mechanism, are described.:

Paper Number:   Rye-012    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Tanaka, Kuminaro

Affiliation:   Mitsubishi Electric Corporation, Hyogo, Japan

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EFM-The Modulation Method for the Compact Disc Digital Audio System

The modulation method employed in the Compact Disc CD) digital audio system codeveloped by Philips N.V. (Eindhoven, The Netherlands) and Sony Corporation (Tokyo, Japan) is described. This method, called eight-to-fourteen (EFM), is an 8 (data bit) ( 14 (channel bit) conversion block code with a space of 3 channel bits for every converted 14 channel bits which is used to connect the blocks. These 3 channel bits, called merging bits, are selectable, enabling the suppression of the low-frequency contents of the frequency spectrum.: First some of the major conditions are listed which are required of the modulation method used for recording/reproducing digital audio signals on an optical disk. The various parameters of EFM as a modulation method are explained in the second part, proving the suitability of EFM for optical disks. An actual example explains the method in detail; a frequency spectrum is also given to enhance understanding.: EFM is well matched with the error-correction method CIRC employed in the CD. The combination of these two methods plays an important role in stably reproducing a 2-channel 16-bit audio signal on a 12-cm-diameter optical disk for more than playing time, single sided.

Paper Number:   Rye-013    Conference:   1st International Conference: Digital Audio (June 1982)

Authors:   Ogawa, Hiroshi; Immink, Kees A.

Affiliations:   Sony Corporation, Audio Technology Center, Tokyo, Japan ; Philips Research Laboratories, Eindhoven, The Netherlands    (See document for exact affiliation information.)

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Error Correcting Codes for Digital Audio

The word -code- has many meanings. Some people refer to computer programs as codes; others refer to magnetic modulation schemes or analog-to-digital conversion formats as codes. However, from our point of view a code embodies a methodology for inserting digital redundancy into a digital data stream. The purpose of this redundancy is to provide some protection against errors or garbles which may occur after the encoding.

Paper Number:   Rye-014    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Berlekamp, E. R.

Affiliation:   Cyclotomics, Berkeley, CA

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Coding and Interleaving for Correcting Burst and Random Errors in Recording Media

Both high-density magnetic and optical recording media give rise to bit errors which are bursty in nature. Such bursts are often moderately short (less than a few hundred bits) but occasionally much longer (several thousand bits). Modeling such a channel by a three-state Markov chain (with different parameters depending on the recording medium), various techniques to virtually eliminate errors are investigated. Results are given for specific examples involving both magnetic and optical media. A practical implementation of these techniques is described, which uses a convolutional encoder and sequential decoder implemented on a single LSI chip. For digital audio applications this LSI encoder-decoder, together with a few general-purpose memories and support chips, can provide a close approximation to the theoretical performance of the analytical model in the space of a small printed-circuit card.

Paper Number:   Rye-015    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Viterbi, Andrew J.

Affiliation:   M/A-COM LINKAB1T, Inc., San Diego, CA

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Error Correction for Digital Audio Recordings

Error correction is one of the key technologies in the field of digital audio. The basis of error correction is discussed sing a -supermarket shopping model,- and various criteria are shown for the application of error-correcting codes to digital audio systems. The details of code schemes are explained, such as the EIAJ format for recorders using video cassettes, the DASH format for professional stationary-head recorders, and the Compact Disc (CD).: Cross interleaving is a unique method of combining two codes by interleaving delay and proved very efficient in performance as well as in hardware design. Therefore the method is applied to the DASH format, the CD format, and many other systems.

Paper Number:   Rye-016    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Doi, Toshi T.

Affiliation:   Sony Corporation, Tokyo, Japan

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CIRC-The Error-Correcting Code for the Compact Disc Digital Audio System

Recently N.V. Philips and the Sony Corporation agreed on the standardization of their Compact Disc (CD) digital audio systems. The error-correcting code CIRC (cross interleave Reed-Solomon code) for the system permits highly efficient detection and correction for burst as well as random errors. The code format defined for CIRC has built-in flexibility to accommodate three or four audio channels.: The requirements for error correction in a digital audio disk are explained, as are the ways in which they are met. The correction performance for random as well as burst errors is specified for several candidate decoding strategies. The application of CIRC and the well-matched EFM (eight-to-fourteen modulation) method enables the CD system to record stereo sound in 16-bit accuracy for more than 1hr practically free of errors.

Paper Number:   Rye-017    Conference:   1st International Conference: Digital Audio (June 1982)

Authors:   Vries, Lodewijk B.; Odaka, Kentaro

Affiliations:   Philips Research Laboratories, Eindhoven, The Netherlands ; Sony Corporation, Tokyo, Japan    (See document for exact affiliation information.)

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Compact Disc (CD) Mastering-An Industrial Process

Compact Disc (CD) mastering is a process in which digital audio and subcode information is encoded into the standard CD format and recorded on a disk surface. The information is contained in pits of discretely varying lengths arranged in a spiral.: The disk-mastering process lies between tape mastering and replication. It involves the application of thin photoresistant layers onto glass substrates, encoding and recording the audio and subcode information, and developing and testing to generate the required pit dimensions (pit geometry).: The parameters influencing the pit geometry and other quality parameters of masters are many, and the process requires a specific philosophy and discipline to be performed industrially. This philosophy and the resulting equipment, operating requirements, quality control, and test methods are described.

Paper Number:   Rye-018    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Verkaik, Willem

Affiliation:   Philips Electro-Acoustics Division, Optical Disc Mastering, Eindhoven, The Netherlands

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Manufacturing Technology of the Compact Disc

First, we describe the structure of the disk, and then review the production process, from master-tape editing to final disk finishing. Finally, some signal characteristics will be pointed out, which should be carefully monitored.

Paper Number:   Rye-019    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Miyaoka, Senri

Affiliation:   Disc Development Division, Sony Corporation, Tokyo, Japan

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The Audio Signal Processor: The Next Step in Digital Audio

With the advent of the Compact Disc and the proliferation of digital audio tape recorders it would appear that digital audio will soon be available on a very broad scale. It seems only reasonable, then, to expect that the next step will be to digitize the production of the sound that these media carry. Simulation of the music production equipment currently in use requires a great deal of signal processing power that is not readily available on commercial computers at this time, but is now available on an experimental basis at reasonably competitive cost in a few places around the world through the use of special-purpose digital audio signal processors. In general terms the problem of digital audio processing is discussed, and some suggestions are prescribed for the builders of such devices.

Paper Number:   Rye-020    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Moorer, James A.

Affiliation:   Lucasfilm Ltd, San Rafael, CA

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BBC Digital Audio-A Decade of On-Air Operation

The BBC has unique experience in the benefits and problems of digital audio, having used digital systems in the normal distribution of sound signals from London to various transmitters around the British Isles continuously for a period of some ten years. The BBC also evolved a digital stereo tape recorder in 1971, followed by a multitrack recorder and a working digital sound control desk in the late seventies. Experiments in how the broadcast transmission of digital audio is affected by difficult reception conditions have also been conducted. At a time when the BBC as well as other users and manufacturers of audio equipment are contemplating lager scale excursions into the use of digital techniques, it is appropriate to discuss our experience in the subjective and objective evaluation of such systems during their design, acceptance, and continuing use. Some of the impairments which may arise are not disclosed by conventional distortion measuring techniques and although they may only be detected subjectively on a limited variety of program material under ideal listening conditions, if uncorrected they may lead to the generalized criticism of digital sound which exists in some areas. It is also important to consider the repercussions on associated analog components of the introduction of digital processes into parts of the audio chain, such as the performance of analog limiters, the limitations of existing level indicating meters, and even the criteria for acoustic noise levels in studios.

Paper Number:   Rye-021    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Stripp, D.

Affiliation:   BBC, London, United Kingdom

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Processing Systems for the Digital Audio Studio

The potential immaculate quality of digital recording and processing is readily recognized by the audio industry. Furthermore, the inherent remote-control capability of digital processing opens up new opportunities for improved ergonomic design of control consoles. For the audio industry to take advantage of these opportunities, a variety of design challenges must be conquered.

Paper Number:   Rye-022    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Jones, M. H.

Affiliation:   Neve Electronics International Limited, Royston, Hertfordshire, United Kingdom

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Digital Equalization of Audio Signals

This paper examines the use of digital filtering as a method of correcting of performance of audio systems, especially in applications where the desired correction cannot be known in advance, such as room equalization and the restoration of historic recordings. First, the goals of correction are examined in light of differing listener expectations and recording methods. The requirements for objective accuracy in sound reproduction are analyzed, and presently available technology evaluated. Net, analog and digital equalization methods are compared. Analog equalization is shown to be inherently problematic; the frequent need for high-frequency attenuation in sound reinforcement is explained as an aspect of the performance of analog band-pass equalizers. Digital filtering is briefly reviewed; time-domain specification of such filters allows equalization to be applied separately to early and later segments of an elctroacoustic system's impulse response, that is, its direct and reverberant fields. Finally, two digital signal processing methods applicable to equalization are reviewed, the fast Fourier transform method, already extensively applied, and a new system employing adaptive linear prediction and real-time convolution.

Paper Number:   Rye-023    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Berkovitz, Robert

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Digital Synthesis of Natural and Unnatural Sounds

The sampling theorem guarantees theoretically that any sound which can be heard by the human ear can be synthesized from digital samples. Today, computers have actually produced a great variety of interesting and powerful timbres. These range from close imitations of musical instruments to sounds never heard before. An analysis-synthesis technique has been developed which involves analyzing natural sounds and synthesizing approximations to these sounds using simplified models. This work has led not only to rich synthesized sounds, but also to a fundamental understanding of the important factors in natural sounds. Unnatural sounds include pitch and rhythm paradoxes in which the apparent pitch or tempo of a sound can simultaneously increase or decrease; timbres with nonharmonic overtones which preserve some perceptions of classic harmony while rejecting others; and sounds in which the timbre depends on unusual ways in which the spectrum of the sound changes during a note. Examples of various synthesized sounds are included.

Paper Number:   Rye-024    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Mathews, Max V.

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Computer Systems and Languages for Audio Research

While most of the professional audio industry is preoccupied with the reproduction of recorded music created by natural instruments, there is a more profound application: creation of original music with digital signal processing. The historic limitations of natural air and string resonances can be overcome by the use of computer sound synthesis. Although computers can generate any sound that can be specified in point-by-point fashion, creative exploration of new timbres and new musical effects requires fabrication of new signal-processing structures at the level of software programming. For both musician and engineer, replacing an instrument by a terminal - or an analog pot by a subroutine - can be very disturbing, particularly if the modes of human-machine interaction are orthogonal to the task. A composer-oriented software system is described which affords intuitive yet flexible control over the most recent methods of digital audio processing.

Paper Number:   Rye-025    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Vercoe, Barry

Affiliation:   Massachusetts Institute of Technology, Experimental Music Studio, Cambridge, MA

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High-Quality Picture Transmission in a Digital Audio System

A high-quality digital video signal is transmitted with low bit rate in a digital audio system. Several digital audio systems have been developed for professional and consumer use. The transmission rate of most two-channel digital recorders is approximately 2 Mbit/s. For digital television, systems have been worked out by CCIR, SMPTE, and EBU. Digital video tape recorders are also under development; their transmission rate is about 300 Mbit/s using a component television signal. Since the ratio of video to audio rates is more than 100, it is difficult for digital audio recording media to capably record a television video signal. A new signal format for still-frame picture transmission is presented. The advantages and disadvantages of composite-signal and component-signal methods have been studied. The format can be extended to partial moving pictures, computer-generate moving graphics, and still pictures for high-definition television (1125 lines).

Paper Number:   Rye-026    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Takahashi, Nobuaki

Affiliation:   JVC (Victor Company of Japan, Ltd.), Kanagawa, Japan

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Conference Speakers

Not available.

Paper Number:   Rye-027    Conference:   1st International Conference: Digital Audio (June 1982)

Author:   Blesser, Barry A.

Affiliation:   Blesser Associates, Raymond, NH

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