NBSIR 88-3725

Evaluation of A Copy Prevention Method

for Digital Audio Tape Systems

NATIONAL ENGINEERING LABORATORY

NATIONAL BUREAU OF STANDARDS

B.A. Bell and G.N. Stenbakken

Electronic Instrumentation arid Metrology Group

D.R. Flynn, D.J. Evans, E.D. Burnett, and V. Nedzelnitsky

Acoustic Measurements Group

K.R. Eberhardt

Statistical Engineering Division

February 1988

U.S. Department of Commerce

C. William Verity , Secretary

National Bureau of Standards,

Ernest Ambler, Director
.
EXECUTIVE SUMMARY

Digital audio tape (DAT) systems have all the attractive audio qualities now

associated with compact disc technology plus the ability to make very high

quality copies of copies. Concern over loss of intellectual property rights

has produced proposals for legislation requiring that DAT systems sold in the

United States be fitted with a copy prevention decoder such that suitably

coded material could not be copied by the DAT recorder. The CBS Records

system, the basis for the proposals, removes a narrow band of frequencies from

the audio signal - termed putting a "notch” in the frequency spectrum of the

signal

.

In considering this proposal, several committees of the Congress asked the

National Bureau of Standards (NBS) three questions about the copy prevention

system:

1. Does the copy prevention system achieve its purpose to prevent digital

audio taping machines from recording?

2. Does the system diminish the quality of the prerecorded material into

which the notch is inserted?

3. Can the system be bypassed, and, if so, how easily?

The National Bureau of Standards received from CBS Records descriptive

material, specifications, circuit diagrams, and operating encoding and

recording/decoding devices for the tests. The results and conclusions of the

NBS tests apply to these devices. Suggestions for recorded material to be

used in listening tests were received from several sources.

NBS conducted a series of laboratory studies to understand the copy prevention

system and to exercise it in several ways to obtain answers to question 1 above. The NBS also constructed and evaluated several circuits designed to

bypass the system (question 3) . With the aid of an expert in acoustical

evaluations on contract from a university, listening studies were conducted

using subjects drawn from the local section of the Audio Engineering Society,

NBS staff, and a few local audiophiles and musicians. These listening studies

were carried out to answer question 2 above.

The NBS findings are as follows:

1. Answer to question 1

- "Does the system achieve its purpose?":

NBS Conclusion : The system does not achieve its stated purpose.

The system does prevent the copying of notched material much of the

time. However, for about half of the recorded tracks studied, the

system exhibited false negatives : i.e., notched material was

nonetheless recorded. In addition, the system also exhibited false

positive behavior, i.e., the system failed to record unnotched

material. NBS studied 502 tracks on 54 compact discs and found

false positives for 16 tracks on 10 discs.

iii
2 . Answer to question 2 - "Does the system diminish the quality of the

prerecorded material?":

NBS Conclusion : The system's encoder alters the original electrical

signal. For some listeners for some selections, this results in a discernible difference between prerecorded notched and unnotched

material

.

NBS interpreted this question both in terms of objective electrical

measurements and in terms of whether or not listeners can detect a difference between notched and unnotched material in carefully

controlled listening tests. Electrical measurements showed that the

encoded signal is degraded relative to the signal as it was prior to

copy prevention coding, degraded in terms of adding a deep notch in

signal amplitudes near 3840 Hz, and degraded in terms of changes in

signal phase relationships near 3840 Hz. In a double-blind series

of listening tests, 87 subjects listened to a prerecorded tape of

short selections from 24 different sources - 20 compact discs and

four selections produced on a keyboard synthesizer. The results

show that, although the effects of the encoder are fairly subtle for

some musical selections, there are some selections for which the

subjects detected differences between notched and unnotched

material

.

In a second series, 15 experienced listeners worked with 10

selections presented on parallel tape tracks such that the subject

could switch back and forth from notched to unnotched material. For

2 of the 10 selections the encoded version was correctly identified

12 out of 15 times; these results are statistically significant.

3. Answer to question 3 - "Can the system be bypassed, and, if so, how

easily?"

:

NBS Conclusion : The copy prevention system can be bypassed easily.

This question was interpreted by NBS as asking whether or not a DAT

recorder can be made to record notched prerecorded material. It is

not possible to restore all of the actual information removed by the

encoding process. NBS engineers designed and constructed several

electronic circuits for implementing five different methods to

circumvent or defeat the copy prevention system by the use of

external signal conditioning. All five methods succeeded in

bypassing the copy prevention system. The circuits are simple and

easy to construct. The cost would be of the order of $100.

iv
Table of Contents

Executive Summary i List of Figures V

List of Tables X

1. Introduction 1

2. Description and Characterization of the System 3

2.1 Encoder: Operation 3

2.2 Encoder: Physical Measurements 6

2.3 Decoder: Operation 18

2.4 Decoder: Physical Measurements 22

3. Does the System Achieve Its Purpose? 44

3.1 Interpretation of the Question 44

3.2 Decoder False Positives 44

3.3 System False Negatives 46

4. Does the System Diminish the Quality of the Prerecorded Material? 49

4.1 Interpretation of the Question 49

4.2 Effect on the Performance Specifications of an Audio System 49

4.3 Subjective Response Testing Procedure 49

4.4 Selection and Characterization of Recorded Material 51

4.5 Results of Listening Tests 53

5. Can the System be Bypassed, and if so, How Easily? 65

5.1 Interpretation of the Question 65

5.2 Description of Defeat Methods 6 5

5.3 Ease of Implementation 70

5.4 Audibility of Defeat Methods 71

V
6 . Summary of Results 72

6.1 Does the System Achieve Its Purpose?.. 72

6.2 Does the System Diminish the Quality of the Prerecorded Material? .. 73

6.3 Can the System be Bypassed, and if so, How Easily? 74

Acknowledgements 76

Appendices

Appendix A: Congressional Committee Letters A.1

Appendix B: Compact Discs Used in False Positive and False Negative

Studies B.1

Appendix C: Contractor's Report C.1

Appendix D: Sponsors' Lists of Recorded Material D.1

Appendix E: Description of Recorded Material for Listening Tests E.1

Appendix F: Preparation of Recorded Material F.1

Appendix G: Audio Systems Used for Listening Tests G.1

Appendix H: Listener Instructions H.1

Appendix I: Schematic Diagrams of Defeat Circuits 1.1

VI
List of Figures

Figure 1 Conceptual block diagram of the major elements contained in the

Encoder, as determined in the NBS study 4

Figure 2 Amplitude- frequency characteristic for the encoding notch filter

in the left channel of Encoder 006 7

Figure 3 Phase characteristic for the encoding notch filter in the left

channel of Encoder 006 7

Figure 4 Amplitude- frequency characteristic of the encoding notch filters

in Encoder 006 8

Figure 5 Phase characteristic of the encoding notch filters in Encoder

006 9

Figure 6 Amplitude -frequency characteristic of the encoding notch filters

in Encoder 012 9

Figure 7 Phase characteristic of the encoding notch filters in Encoder

012 10

Figure 8 Impulse response of the left- channel notch filter for Encoder

006 11

Figure 9 Amplitude -frequency characteristic of the on-notch and

off-notch filters in Encoder 006 12

Figure 10 Amplitude -frequency characteristic of the on-notch and

off -notch filters in Encoder 012 12

Figure 11 The Encoder state and the voltage levels at the output of the

on-notch and off-notch filters for selection "Prokofiev" 15

Figure 12 The Encoder state and the voltage levels at the output of the

on-notch and off-notch filters for Track 3 of "Drumming" by

Steve Reich, Electra/Nonesuch 9079170-2 16

Figure 13 Amplitude -frequency characteristic for the left channel of

Encoder 006 with the encoding notch filter out of the circuit 17

Figure 14 Phase characteristic for the left channel of Encoder 006 with

the encoding notch filter out of the circuit 17

Figure 15 Basic diagram of the decoder and recorder electronics in the

CBS-supplied DAT Recorder/Decoder 18

Figure 16 Block diagram of the major circuits contained in the Decoder

section (within the dashed outline) of the DAT Recorder/Decoder,

as determined in the NBS study 20

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Figure 17 Amplitude response for the three bandpass filters in the Decoder

versus the effective input signal frequency 23

Figure 18 Amplitude -frequency response of the anti-aliasing filter and

automatic gain control (AGC) circuits in the Decoder 24

Figure 19 Difference level as a function of the lower sideband level

(upper sideband level constant at about -43 dBV) that causes the

POSSIBILITY AND RECORD INHIBIT states in the Decoder 26

Figure 20 Difference level as a function of the lower sideband level

(upper sideband level constant at about -23 dBV) that causes

the POSSIBILITY AND RECORD INHIBIT states in the Decoder 28

Figure 21 Difference level as a function of the upper sideband level . (lower sideband level constant at about -43 dBV) that causes the

POSSIBILITY AND RECORD INHIBIT states in the Decoder 29

Figure 22 Difference level as a function of the upper sideband level

(lower sideband level constant at about -23 dBV) that causes the

POSSIBILITY AND RECORD INHIBIT states in the Decoder 30

Figure 23 Summary of the RECORD INHIBIT activity observed for two different

amplitudes of sawtooth (ramp), triangle, and square-wave input

signals applied to the Decoder 32

Figure 24 Frequency spectrum (in green) of a 486 Hz (fundamental) square

wave input signal superimposed on the frequency response

characteristics of the three bandpass filters in the Decoder 33

Figure 25 Frequency spectrum (in green) of a 337 Hz (fundamental) sawtooth

(ramp) waveform superimposed on the frequency response

characteristics of the three bandpass filters in the Decoder 34

Figure 26 Frequency spectrum (in green) of a 486 Hz (fundamental) sawtooth

(ramp) waveform superimposed on the frequency response

characteristics of the three bandpass filter in the Decoder 35

Figure 27 Frequency spectrum (in green) of an 870 Hz (fundamental) sawtooth

(ramp) waveform superimposed on the frequency response

characteristics of the three bandpass filters in the Decoder 36

Figure 28 Frequency spectra of an 841 Hz (in red), an 870 Hz (in green), and

a 901 Hz (in blue) sawtooth (ramp) waveform superimposed on the

frequency response characteristics of the three bandpass filters

in the Decoder 37

Figure 29 The equipment setup for running unencoded music tests 4 0

Figure 30 The equipment setup for running encoded music tests 41
Figure 31 Color spectrographs of music with naturally-occurring notches

which caused False Positives 42

Figure 32 Color spectrographs of direct (left) and encoded (right)

material, for the serial listening study, from Debussy, Barber,

Beethoven chord, and Beethoven chorus selections 55

Figure 33 Color spectrographs of direct (upper) and encoded (lower)

material, for the parallel listening study, from Bizet

selection 5 6

Figure 34 "Signal detection" plot for n = 61 earphone listeners 60

Figure 35 "Signal detection" plot for n = 23 loudspeaker listeners 61

Figure 36 Schematic plot of responses for listening subjects 62

Figure 37 Defeat method 1 that uses a bandpass -filtered noise source

added to the encoded input signal 66

Figure 38 Frequency spectra of: (1) a broadband noise input signal

containing the encoding notch (green spectrum) , (2) the added

bandpass filtered noise source (blue spectrum), and (3) the

resultant output signal (red spectrum) 67

Figure 39 Defeat method 2 that uses a 1.9 kHz component of the input

signal doubled to match the 3.8 kHz encoding notch frequency 68

Figure 40 Defeat method 3, which is similar to method 1, and uses the

amplitude of the 2.7 kHz component of the input signal to

modulate the bandpass -filtered noise 68

Figure 41 Defeat method 4, which is similar to method 2, and uses the

amplitude of the 2.7 kHz component of the input signal to

modulate the frequency doubled 1.9 kHz component of the input

signal 69

Figure 42 Defeat method 5, similar to method 4, and uses the amplitude of

the 3.5 kHz and 4.3 kHz components of the input signal to

provide the modulation 70

Appendices

Figure C-l Distribution of same-different test scores over 24 selections,

including four synthesizer selections C.10

Figure C-2 Comparison between same -different test scores over 24

selections, including four synthesizer selections plotted as stars C.11

Figure C-3 Distribution of same -different test scores over listeners for

the 24 musical selections C.16

ix
Figure F-l Equipment setup used for transfer from a compact disc to digital

audio tape F.3

Figure F-2 Equipment setup used for making direct and encoded digital

tapes F.4

Figure F-3 Equipment setup used for digital editing F.4

Figure F-4 Equipment setup for digital transfer to multi-channel tape

recorder F.12

Figure F-5 Equipment setup for making direct and encoded tapes on the

multi-channel recorder F.13

Figure G-l Equipment setup used for audio playback in serial listening

tests G.2

Figure G-2 Equipment setup used for audio playback in parallel listening

tests G.3

Figure G-3 Amplitude- frequency characteristics of the left and right

earphones of a typical headset G.5

Figure G-4 Phase characteristics of the left and right earphones of a typical headset G.6

Figure G-5 Amplitude- frequency characteristics of the two loudspeakers,

as measured in the large NBS anechoic chamber G.8

Figure G-6 Phase characteristics of the two loudspeakers, as measured in

the large NBS anechoic chamber G.9

Figure G-7 The room used for the serial listening tests. G.11

Figure G-8 Diffusing panels at the back of the room used for the serial

listening tests G.12

Figure G-9 Impulse response of the two loudspeakers as measured in the

listening room G.14

Figure G-10 Amplitude -frequency characteristic of the two loudspeakers as determined in the listening room by FFT analysis of time record

waveforms G.15
List of Tables

Table 1 Logic map for Encoder 006 13

Table 2 Logic map for Encoder 012 13

Table 3 The four types of possible test cases 39

Table 4 False positive data 45

Table 5 False negative data 47

Table 6 Average score summary for all listeners in the serial

listening tests 57

Table 7 Subject responses for all listeners in the parallel listening

tests 64

Table 8 Audibility of defeat methods 71

Appendices

Table B-l Compact discs used in false positive and false negative

studies B.1

Table C-l Score summary by selection for earphone listeners... C.8

Table C-2 Score summary by selection for loudspeaker listeners C.9

Table C-3 Score summary by selection for earphone listeners C.13

Table C-4 Score summary by selection for loudspeaker listeners C.15

Table C-5 Median split analysis across earphone listeners (n = 61) and

musical selections (n = 20) C.15

Table C-6 Identification of encoded channel in the parallel- test

format C.19

Table E-l Musical selections for listening tests E.3

Table 1-1 Circuits used in each of the defeat methods 1.1

XI
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1.

INTRODUCTION

Digital encoding of sound is causing a revolution in the world's audio

industry. Digital sound systems have lower noise, less distortion, and a more

even frequency response than conventional analog sound systems. Moreover,

successive rerecordings of digital material produce negligible degradation of

quality from generation to generation. The compact disc (CD) player is the

first mass marketed hi-fi product to exploit this digital technology. The CD

player commands a rapidly expanding market share. However, only playback

versions are currently available. This situation is about to change with the

introduction of digital audio tape (DAT) recorder/players . These promise to

have all the qualities of the CD systems plus the ability to make very high

quality copies of copies.

There is a concern that the quality of rerecording from DAT systems may cause

financial harm to recording artists and companies by making it relatively easy

for copies to be made and sold or used by others who might otherwise buy

additional tapes and discs. This situation is seen as a violation of

intellectual property rights. Not everyone agrees; there are those who feel

that home copying for personal use should be permitted and only copying for

profit should be barred. The controversy is similar to the one that occurred

a few years ago over video taping and rerecording.

The Administration proposed in 1987 that Congress enact legislation requiring

that DAT systems sold in the United States be fitted with a copy prevention

decoder such that suitably "coded" material could not be copied by the DAT

recorder. Bills were introduced in the House (H.R. 1384) and the Senate

(S. 506). In the CBS Records system, which is the basis for the proposals,

the "code" is achieved by removing a narrow band of frequencies from the audio

signal - referred to as putting a "notch" in the signal's frequency spectrum.

The copy prevention is accomplished by adding to the DAT system a decoder

circuit for detecting the notch and shutting down the recorder.

Hearings were held at which the opposition to the proposal claimed that

serious degradation of recorded sound would result from notching by the CBS

method. Opponents also say that the CBS system may produce false positives,

i.e.

,

refusals to record unnotched material, and that the copy prevention

system could in any case be easily defeated or circumvented by a determined

pirate and even by home electronics buffs. The proponents strongly disagree.

The responsible Congressional committees decided to ask the National Bureau of

Standards (NBS) to study the CBS copy prevention system and provide answers to

the following questions (see Appendix A)

:

1. Does the copy prevention system achieve its purpose to prevent

digital audio taping machines from recording?

2. Does the system diminish the quality of the prerecorded

material into which the notch is inserted?

3. Can the system be bypassed, and if so, how easily?

1
The two groups representing the opposing views - the Home Recording Rights

Coalition and the Recording Industry Association of America, Inc. - agreed to

fund the NBS study. The Bureau requested descriptive material,

specifications, circuit diagrams, and operating devices. CBS Records agreed

to supply these items but officially requested that the Bureau not reveal

information identified by CBS as company proprietary. NBS notified both

parties that NBS would hold confidential information (the schematic diagrams

and detailed specifications) supplied and identified as proprietary by CBS,

but would publish all information developed by the NBS staff in the course of

the study. Inevitably, the NBS measurements describe some of the subjects

included in the CBS information. The Bureau also stated that it planned to

subcontract the necessary listening studies to a university or firm

specializing in psychoacoustic measurements.

The reader should keep firmly in mind that the National Bureau of Standards

agreed to evaluate a specific implementation of the CBS Records copy

prevention system. The Bureau did not undertake to study variations in

(statistical) lots of devices supplied by CBS nor did NBS undertake efforts to

improve upon the devices and information provided by CBS. In short, NBS

worked with what was provided. Altogether, CBS sent two recorders/decoders

and two encoders to NBS. The first recorder/decoder was defective at the

outset and so the second unit was shipped. The second encoder was requested

by NBS to speed up the evaluation. As will be discussed below, differences

between the two encoders were noted but not addressed.

This report has been written to provide the answers to the three questions

listed above and to supply the necessary supporting documentation of the

laboratory work, while at the same time revealing a minimum of information

identified by CBS as company proprietary. This situation may make complete

understanding of some aspects a little more difficult than would otherwise be

the case.

NOTE: Certain commercial equipment, instruments, or components are identified in this report in order to specify adequately the

experimental procedure. Such identification does not imply

recommendation or endorsement by the National Bureau of Standards,

nor does it imply that the equipment or components are necessarily

the best available for the purpose.

2
2. DESCRIPTION AND CHARACTERIZATION OF THE COPY- PREVENTION SYSTEM

CBS Records made available to the National Bureau of Standards (NBS) two copyprevention encoders that were designed and built for use by recording

companies to encode, or "notch," recorded material and two Sony rotary-head

digital audio tape recorders (R-DAT) , each equipped with a decoder that was

designed and built by CBS Records to detect the presence of encoding.

Sections 2.1 and 2.2 describe the operation of the encoder and the physical

measurements that were made in order to characterize its performance. In

summary, the encoder consists of a narrow band-reject filter that removes

signal components near a nominal center frequency of 3840 Hz. The encoder was

found to be dynamic, with logic circuitry that determines whether to encode or not encode, depending upon the signal levels near 3840 and 2715 Hz.

Sections 2.3 and 2.4 describe the operation of the decoder and the physical

measurements that were made in order to characterize its performance . In

summary, the decoder scans the input signal components that pass through

filters having center frequencies near 3300, 3800, and 4300 Hz, with these

center frequencies being varied by about ±5 percent, to determine whether or not the signal is encoded. Since these frequencies vary in a manner not

correlated with the incoming signal, the behavior of the decoder/recorder will

not be exactly reproducible upon repeated tests with a given passage of

recorded material, thus requiring a statistical approach to testing these

circuits

.

2.1 Encoder: Operation

The encoders had the serial numbers CCE-006 and CCE-012. In this report, the

first encoder supplied by CBS Records is hereafter referred to as "Encoder 006" and the second encoder sent to NBS is hereafter referred to as "Encoder 012." When it is not necessary to distinquish between the two specimens of encoder that were provided, they are referred to simply as "the

Encoder .

The Encoder provided to NBS is an analog device that filters the signal in

each stereo channel so as to remove energy over a narrow band of frequencies.

The front panel controls on the Encoder include a switch that enables the

Encoder to be totally bypassed and another switch that disables the action of

the Encoder but causes the signals for the two channels to pass through the

unity gain amplifiers of the Encoder.

Conceptually, the Encoder operates as shown in figure 1. The signal for each

channel passes through unity gain amplifiers (not shown) to a summing circuit

("SUMMER," at the top and bottom of figure 1). In parallel, each signal

passes through a narrow bandpass filter and then to a solid-state switch that

can be enabled to direct the filtered signal for each channel to the

corresponding summing circuit. If the switch is "closed," each summing

circuit subtracts the bandpass-filtered signal from its corresponding direct

signal and passes the resultant signal on to the output of the Encoder. Thus,

when the solid-state switches are closed, the Encoder acts as a dual-channel

notch filter that removes energy from the audio signals over the range of

3
frequencies transmitted by the bandpass filters. This pair of notch filters

is hereafter referred to in the singular as the encoding notch filter or simply as the encoding filter. When the solid-state switches are open, the

signal passes through the Encoder without being "notched."

SUMMER

SUMMER

LEFT

OUTPUT

RIGHT

OUTPUT

Figure 1 Conceptual block diagram of the major elements contained in the

Encoder, as determined in the NBS study.

The operation of the solid-state switches shown in figure 1 is controlled by

additional filters and logic circuits that operate as follows. The signals

from the left and right channels are summed by a third summing circuit, shown

at the center left of figure 1. This monaural signal is passed through a bandpass filter, hereafter referred to as the "on-notch filter," with a center

frequency that is nominally the same as that of the encoding notch filter.

The monaural signal is also passed through another band pass filter, hereafter

referred to as the "off-notch filter," with a center frequency nominally onehalf octave below that of the encoding notch filter. The outputs of the on- notch filter and the off-notch filter are each rectified, detected, and passed

on to logic circuitry that activates or deactivates the two-channel encoding

notch filter, depending upon the signal levels from the on-notch and off -notch

filters. Note that the on-notch and off-notch filters never operate directly

upon the signals going to the output of the Encoder.

4
The encoding notch filter removes energy at frequencies in the vicinity of

3840 Hz. In musical notation, this frequency falls between B-flaty and B 7 , the highest B-flat and B on the standard piano keyboard, in the equally- tempered chromatic scale. (The subscript notation used is that which assigns to

Cq a frequency which corresponds roughly to the lowest audible pitch; in this

scheme, the seventh octave extends from C 7 with a frequency of 2093.0 Hz

through B7 with a frequency of 3951.1 Hz. The next note above B 7 is Cg, which

is four octaves above C4 , commonly known as middle C.) The level reduction

near the center frequency of 3840 Hz is more than 80 dB, corresponding to a reduction to less than 0.01 percent of the original amplitude. At this degree

of reduction, the signal effectively may be considered to be totally removed.

At frequencies away from the center of the encoding notch filter, the reduction is less, becoming 3 dB (a 3-dB decrease represents a reduction in amplitude to 71 percent of the original amplitude) at about 110 Hz above and below

the center frequency, i.e., at nominally 3730 and 3950 Hz. These two frequencies are very close to those of the musical notes, B-flaty and B 7

. The

off-notch filter has a center frequency that is near to 2715 Hz, the geometrical mean of the frequencies corresponding to the musical notes E 7 and F7

.

It should be noted that although 3840 Hz is a high frequency in terms of the

musical scale, it is not at all above the region of significance to hearing.

In fact, for young adults with normal hearing, the ear is most sensitive at

frequencies near 3840 Hz.

Not many musical instruments are capable of producing notes that have fundamental frequencies falling at or above the center frequency of the encoding

notch filter. However, when performing this study, we did observe fundamental

notes in the encoding frequency region for piano, piccolo, and bells. Many

organs and synthesizers can also produce such fundamental notes.

Frequencies near 3840 Hz have significance for the perception of musical

timbre because overtones of notes falling lower on the musical scale may occur

in this region. Also, transient sounds, such as those produced by percussion

instruments, will have energy over a broad range of frequencies, frequently

including those in the encoding region. Instruments played slightly off

pitch, or using a scale based on notes other than the 12 tones of the "western" scale will have many components in the encoding region. Synthesizers

can produce notes of any frequency whatsoever, and several examples of com- ponents with frequencies "sliding" though the encoding region were noted.

The use of vibrato, or the production of trills, with voice or instruments,

especially stringed instruments, will cause energy which otherwise would be

below or above the center frequency of the encoding notch filter (e.g., at the

notes B-flaty and B 7 ) to shift into the encoding region. This effect was

frequently observed in this study.

As indicated above, the Encoder contains logic circuitry which switches the

encoding notch filter in and out of the circuit, depending upon the signal

levels through the on-notch and off -notch filters. Presumably, this interruption of the notching operation is effected to prevent notching of audio signals when the encoding might be more noticeable. The results of NBS measurements of the influence of the on-notch and off-notch signal levels on the

encoding operation are presented in section 2.2. However, in order to

describe the operation of the Encoder, the results of these measurements are

5
qualitatively described here. The default operation of the Encoder is to

engage the encoding notch filter, thus encoding both audio channels. However,

if the off-notch signal is below a certain critical level and the on-notch

signal is above a particular critical level, the logic circuitry will cause

the Encoder to remove the encoding notch filter from both channels. When the

off-notch signal and the on-notch signal are each above their respective

critical levels, the action of the Encoder is dependent upon both of these

signal levels.

In summary, the Encoder places a notch in both audio channels, with the center

frequency of the notch occurring approximately midway between the musical

notes of B-flaty and B 7 , with there being some attenuation of the signal at

these notes. Encoding does not occur at all times -- an attempt is made in

the Encoder to eliminate encoding when it would produce effects that are more

likely to be noticed.

2.2 Encoder: Physical Measurements.

This section describes the physical measurements that were made in order to

characterize the Encoder and obtain quantitative information concerning its

operation and the effects of the Encoder on the signals passing through it.

Encoding Notch Filter

The frequency characteristics, both amplitude and phase, of both channels of

the encoding notch filter were measured for both Encoder 006 and Encoder 012

using a Hewlett-Packard (HP) Model 3562A Dynamic Signal Analyzer. Because of

the dynamic nature of the Encoder -- the encoding notch filter switching in

and out depending upon the signal levels from the on-notch and off-notch filters -- special steps had to be taken to force the encoding notch filter to be

switched in while its physical characteristics were being measured.

Figures 2 and 3 show the amplitude -frequency characteristic and the phase

characteristic, respectively, of the encoding notch filter for the left channel of Encoder 006. These data, shown over the frequency range from 0 to 5

kHz, were obtained using, as an input signal, noise which was pre-shaped to

have a dip in its spectrum in the region around 3840 Hz, thus causing encoding

to occur at all times. The amplitude -frequency characteristic is seen to be

quite flat except for the deep notch centered nominally near 3840 Hz; cursors

are included at the frequencies corresponding to the musical notes B-flat7 and

B 7 , so as to help the reader to follow the different frequency scales used for

the abscissa in subsequent plots. The phase characteristic is seen to deviate

considerably from an ideal flat characteristic in the region around 3840 Hz.

Note that this phase plot, as well as others to follow, "wraps" when the phase

passes through ±180 deg, e.g., a phase of +200 deg is shown plotted at 200 - 360 = -160 deg.

The dynamic range of the analyzer is about 80 dB for this type of measurement

using a random noise signal. Since the encoding filter introduces more than

80 dB attenuation near its center frequency, methods using random noise as the

source may not give an accurate description of the filter characteristics in

the region of its maximum attenuation.

6
Figure

Figure

B

—f1 3.t -7 By

-30

CDo

<

CD -60

-90

1

rf

2 3

FREQUENCY (kHz)

2 Amplitude -frequency characteristic for the encoding notch filter in

the left channel of Encoder 006.

3 Phase characteristic for the encoding notch filter in the left channel of Encoder 006. (The small circle represents the cursor of the

analyzer being at 3.84 kHz.)

7
To determine accurately the amplitude -frequency and phase response characteristics of the encoding notch filter over its full dynamic range, the

swept-sine mode of the signal analyzer was used. A limited frequency range of

3690 Hz to 3990 Hz was used, and a pilot tone at 2740 Hz was used to force

encoding at all times. A rejection filter, tuned to that same frequency of

2740 Hz, was placed in the signal path to the analyzer in order to prevent

overload from the pilot tone. A frequency response of this rejection filter

was first obtained with the Encoder switched to the "OUT" position. This

response was stored in the memory of the analyzer. Then the Encoder was

switched to the "IN" position, and the frequency response was again obtained.

The division of the complex values of this second response by those of the

first response yields the desired amplitude -frequency and phase characteristics of the encoding filter. These procedures were repeated, so that the

responses for both the left and right channels of the Encoder were obtained.

The amplitude -frequency and phase characteristics over the frequency region

from 3690 to 3990 are shown in figures 4 and 5 for Encoder 006 and in figures

6 and 7 for Encoder 012. These figures have marked on them the frequencies

for the notes of B-flaty and B 7 , as defined for the equal tempered chromatic

scale. Note that the center frequency of the encoding notch for each channel

of Encoder 012 is closely centered between the frequencies of these two notes

and each note is attenuated by about 3 dB. The encoding notch for each channel of Encoder 006 is centered at a somewhat lower frequency, and attenuates

B-flaty by about 5 dB, and B 7 by about 2 dB. For both channels of both

B—flat 7 0

-30

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1 3.69 3.78 3.84 3.90 3.99

FREQUENCY (kHz)

Figure 4 Amplitude -frequency characteristics of the encoding notch filters in

Encoder 006. The solid and dashed curves correspond to the left and

right channels, respectively.

8
LD

FREQUENCY (kHz)

Figure 5 Phase characteristics of the encoding notch filters in Encoder 006.

The upper and lower plots correspond to the left and right channels,

respectively.

FREQUENCY (kHz)

Figure 6 Amplitude -frequency characteristics of the encoding notch filters in

Encoder 012. The solid and dashed curves correspond to the left and

right channels, respectively.

9
Figure 7 Phase characteristics of the encoding notch filters in Encoder 012.

The upper and lower plots correspond to the left and right channels,

respectively.

Encoders, the attenuation near the center of the notch-filter passband is in

excess of 80 dB and the phase characteristic deviates drastically from a flat,

uniform phase response, exhibiting a discontinuity in the frequency region of

the notch.

Since a narrowband filter can "ring" when excited by a transient signal, measurements were made to determine the transient response of the encoding notch

filter. The signal analyzer employed for these tests has the capability of

determining an impulse response by calculating the inverse Fourier transform

of the frequency response. This calculated impulse response is shown in

figure 8 for the left channel of Encoder 006. The ringing which is seen after

the initial impulse consists of two frequencies in the vicinity of 4 kHz. The

beating between the two frequencies causes the ringing to appear to decrease

and increase in amplitude several times before dying out completely. The

impulse response of the left channel of Encoder 012 was also measured and

found to be essentially identical.

In order to be certain that the ringing observed was not an artifact of the

filters in the signal analyzer rather than of the encoding notch filter,

direct time-domain measurements were also made. The ringing behavior observed

in these measurements was similar to that shown in figure 8.

10
Figure 8 Impulse response, on a linear scale, of the left-channel notch

filter for Encoder 006 as obtained from the inverse Fourier transform of the frequency response.

Logic circuitry

On-notch and off-notch filters These filters (see figure 1) were characterized by placing a 1 -megohm probe at their outputs and obtaining their

frequency responses using the swept-sine mode of the signal analyzer. The

amplitude -frequency characteristics of these filters for Encoders 006 and 012

are shown in figures 9 and 10, respectively. The frequencies for the musical

notes B-flaty and B 7 are shown for the on-notch filter. The frequencies for

the musical notes E 7 and F7 are shown for the off -notch filter. Notice that

the maximum level from the on-notch filter of Encoder 012 is several decibels

lower than that of Encoder 006. This reduced output caused Encoder 012 to

keep the encoding filter engaged more of the time than does Encoder 006.

Switching The signal levels that cause the encoding filter to be switched in

or out were determined using two sine-wave generators, one adjusted to produce

a tone at the frequency of maximum response of the on-notch filter of the

Encoder under test and the other set to the frequency of maximum response for

the off-notch filter. The summed output of these frequency synthesizers was

fed to both the left and right inputs of the Encoder. The input level of the

on-notch tone was adjusted to a particular value, e.g., -3 dBV (where here and

elsewhere in this report, "dBV" denotes an rms -voltage level, relative to 1 V

rms , expressed in decibels). The level of the off-notch tone was set at that

same value and then slowly reduced in 1 dB increments until encoding ceased.

The level of the off -notch tone was then reduced several decibels further, and

then slowly increased until encoding resumed. A new level was selected for

the on-notch frequency, and the process was repeated.

11
FREQUENCY (kHz)

Figure 9 Amplitude -frequency characteristic of the on-notch (solid curve) and

off-notch (dashed curve) filters in Encoder 006.

Ey F7 B-flat7 B7

Figure 10 Amplitude- frequency characteristic of the on-notch (solid curve)

and off-notch (dashed curve) filters in Encoder 012.

12
Tables 1 and 2 show the results of these measurements of the switching action

for Encoder 006 and 012, respectively. These values were not completely

stable, particularly for Encoder 006 where the results varied with time over a range of 1 dB. or more. The reduced output from the on-notch filter of Encoder

012 as compared to that of Encoder 006 is reflected in the difference between

the on-notch and off-notch action levels in the two encoders.

Table 1 . Logic map for Encoder 006

On-notch voltage

level at input

Off-notch voltage level at input

required to change encoding state

to start encoding to stop encoding

-3 -4 -8

-13 -15 -19

-23 -25 -29

-33 -32 -41

-38 -42 -52

-40 Always encodes

Table 2. Logic map for Encoder 012

On-notch voltage

level at input

Off-notch voltage level at input

required to change encoding state

to start encoding to stop encoding

7 2 -6

-3 -13 -16

-13 -23 -26

-24 -36 -40

-27 Always encodes

Note: the use of the term "dBm" has been avoided in these tables since the

quantity measured was voltage, not electrical power. Since the term

"dBm" has been used in the audio industry to denote voltage level

referenced to 1 milliwatt of average electrical power dissipated in a purely resistive 600-0 electrical load, the following is included for the

reader's information. The rms voltage corresponding to 1 milliwatt of

average electrical power dissipated in a purely resistive 600-ohm

electrical load is approximately 0.775 V rms or, in logarithmic terms,

-2.2 dBV

.

13
During the course of these measurements of the switching action, it was noted

that the encoding notch filters were switched in and out over a period of some

milliseconds so that abrupt switching transients did not occur.

Figure 11 shows the voltage levels at the outputs of the on-notch and

off-notch filters when a 20-second sample of Prokofiev's "Alexander Nevsky"

(see Appendix E for a description of music used in this study) was played

through Encoder 012. The presence of an orange line at the bottom of the

figure indicates that the encoding notch filter was engaged; when no orange

line is present, the Encoder had dynamically removed the notch filter from the

signal path. Figure 12 is a similar plot for a sample of glockenspiel music

passed through Encoder 006, showing a change from continuous encoding to

intermittent encoding at a time when there was an obvious change in the

musical key. As can be seen, each of the Encoders could change states

frequently within a short period of time. Because of the higher voltage

levels from the on-notch filter in Encoder 006, this unit frequently would

switch the encoding notch filter out under conditions when Encoder 012 would

continue to encode.

It was noted that the Encoder action was not always repeatable; immediate

repetition of a sample of music did not always lead to the Encoder changing

state at the same time in the music.

Dynamic range. Frequency Response, and Distortion

With the encoding switch in the "OUT" position so that the left- and right

channel signals passed through the unity gain amplifiers of the Encoder but

the encoding notch filter was not engaged, the dynamic range of the Encoder

was measured and found to be in excess of 105 dB.

The frequency characteristic of each Encoder was measured with the the encoding switch in the "OUT" position. Figures 13 and 14 show the measured

amplitude -frequency characteristic and the phase characteristic, respectively,

for the left channel of Encoder 006, as determined over the frequency range

from 125 Hz to 100 kHz using the HP 3562A analyzer with its random noise

source. The root-mean- square input voltage for this measurement was 0.77 V.

The shape of the response may be taken as typical of both channels of both

Encoders. Switching the Encoder "IN" affects the response only in the region

near 3.84 kHz, as indicated in figures 2 through 7.

Harmonic distortion of the Encoders, with the encoding switch in the "OUT"

position, was checked by using an input signal from a low-distortion oscillator and the HP 3562A analyzer in the THD mode. The results, for all combinations of frequency and levels which were tried, indicated that total

harmonic distortion was more than 80 dB below the level of the fundamental

frequency

.

14
LLJ

(9P) 13A31

Figure

11.

The

Encoder

state

and

the

voltage

levels

at

the

output

of

the

on-notch

and

off-notch

filters

for

Prokofiev

selection

(see

Appendix

E

for

a

more

complete

description

of

this

selection)

.
ON-NOTCH

LEVEL

(BLUE)

OFF-NOTCH

LEVEL

(GREEN)

ENCODING

(RED)

O O O O

CM CO

(SP) 13A31

16

Figure

12.

The

Encoder

state

and

the

voltage

levels

at

the

output

of

the

on-notch

and

off-notch

filters

for

a

part

of

Track

3

of

"Drumming"

by

Steve

Reich

(

Electra/Nonesuch

9079170-2).
PHASE

(deg)

a>

GAIN

(dB)

13 Amplitude -frequency characteristic for the left channel of

Encoder 006 with the encoding notch filter out of the circuit.

FREQUENCY (kHz)

Figure 14 Phase characteristic for the left channel of Encoder 006 with

the encoding notch filter out of the circuit.

17
2.3 Decoder: Operation

A general description of the operation of the CBS -supplied DAT Recorder,

modified to contain a "copy prevention code" scanner or Decoder, is given in

this section. Details about the performance of the DAT Recorder/Decoder are

provided in section 2.4.

The purpose of the Decoder is to detect the presence of the copy prevention

code contained in the incoming audio signals. This information or code is the

removal of a narrow frequency band of signal amplitudes from the original,

unencoded signal, creating a "notch" in the frequency spectrum. As

implemented for the NBS tests, the Decoder portion of the DAT Recorder is

simply an adjunct circuit to the other recorder electronics, as illustrated in

figure 15. The left and right channel signal lines coming into the recorder

are connected in parallel to go to the Decoder circuits.

If the presence of a notch is detected by the Decoder, it activates a yellow

light-emitting diode (LED), mounted on the rear of the DAT Recorder, to

indicate the possibility of the presence of a copy prevention notch in the

signal. The Decoder then continues to monitor the presence of the notch for a period of approximately 13 to 15 seconds. If the notch in the signal persists

during this interval, the Decoder then activates a red LED to indicate that a RECORD INHIBIT state exists, and this signal is used to disable the other DAT

Recorder electronics.

DAT Recorder and Decoder

Figure 15 Basic diagram of the decoder and recorder electronics in the

CBS -supplied DAT Recorder/Decoder.

18
The Decoder operates by passing the audio input signal through three bandpass

filters, creating three separate signals having levels proportional to the

amplitude of the input signal frequencies contained in the narrow passband of

each filter. The filters are tuned to the notch center frequency and to

frequencies slightly above and below the notch center frequency. The

frequency bands above and below the notch band are referred to as the upper

and lower sidebands, respectively. By comparing signal levels corresponding

to the upper and lower sideband filters with the signal corresponding to the

notch bandpass filter, the Decoder detects when the level in the notch

frequency passband is less than the level in either of the two sidebands,

indicative of a notch present in the frequency spectrum of the incoming audio

signal. A more detailed description of the operation of the Decoder is given

below.

In order for the Decoder to operate when the incoming signal has its frequency

spectrum shifted (due to variations in the encoding process or the original

source material), the Decoder has a built-in scanner which provides for the

detection of a notch which is 5 to 10 percent above or below the nominal

center frequency. The scanning time to cover this variation was found to be

approximately 35 seconds.

The Decoder must differentiate between a naturally- occurring notch in the

spectrum of the original audio input signal and one due to the encoding

process. To make this distinction, NBS found that the Decoder implements

several criteria. One requirement is the time during which the notch must be

sustained. As mentioned above, this time is about 13 to 15 seconds. Another

criterion is a minimum level of the audio input signal in the upper and lower

sidebands just above and below the notch. The level of the sum of the signals

in these two sidebands must be above a lower threshold limit in order to

activate the comparison process in the Decoder, i.e., a notch cannot be

detected simply because the input signal level in the notch region is too low.

This low-level threshold was measured to be about -51 dBV. Finally, as described above, the level of the audio signal in the notch frequency band is

compared to the level of the signal in the sidebands. For the Decoder to

detect (and indicate) the presence of an encoded signal, the notch band level

must be lower than the level of the lower sideband. As will be described in

section 2.4, physical measurements made on the Decoder showed considerable

variation in the required relative levels of the notch and sideband signals in

order for this criterion to be met.

Figure 16 on the following page is a block diagram of the Decoder circuitry

contained in the DAT Recorder/Decoder. The incoming audio signals (both left

and right channels of a stereo input) are first summed together and then

passed through a low-pass, anti-aliasing filter which produces a combined

audio signal free of high-frequency components. The combined signal is then

fed into an automatic gain control (AGC) circuit. The function of the AGC

circuit is to keep the level of its output signal within a narrow range, while

its input signal level varies over a wide dynamic range. Next, the AGC output

signal is modulated using a carrier frequency of about 10 kHz. This

modulation shifts the frequency spectrum of the combined audio signal to

frequency bands above and below the modulation frequency. The analog

19
DAT Recorder and Decoder

INPUT SIGNALS

Figure 16 Block diagram of the major circuits contained in the Decoder

section (within the dashed outline) of the DAT Recorder/Decoder, as determined in the NBS study. These circuits were implemented on an

ancillary circuit board, separate from the other recorder

electronics and tape drive.

discrimination circuits following the modulator are tuned to make use of the

audio signal spectrum that lies below the modulation frequency. The

frequency- shifted signal is then split into three signal paths, as indicated

in figure 16, and fed into three separate bandpass filters. Each filter is

tuned to one of three center frequencies, corresponding to the upper sideband,

lower sideband, and notch band. The modulation frequency is slowly varied by

the scan controller circuit so that the bandpass filters sample a varying

frequency range of the incoming audio signal.

Following the bandpass filters, the resultant narrowband signals, centered

about their corresponding bandpass filter center frequency, are then converted

to dc signals (by a rectifier circuit and low-pass filter). Thus, the audio

input signal is decomposed into three dc signal levels, each proportional to

20
j

the amplitude of the input signal frequencies contained in the particular

passband (notch band, upper and lower sidebands) . These dc signals are next

processed by either averaging or integrating the three signals-, depending on

the state of the stop -scan/time -constant control line, i.e., whether or not a notch in the input signal has been detected by the subsequent analog

comparator and digital logic circuits. If a notch has not been detected, the

Decoder is in the scanning or "search" mode. If a notch has been detected,

the Decoder is in the integrate mode.

In the scanning mode, the three dc signals are averaged with circuits having

short time constants (100 to 300 ms). The smaller of the upper or lower

sideband signals is switch selected by another sideband selector comparator

circuit, not shown in this diagram. The upper or lower sideband averaged

signal is first attenuated, and then compared by the analog comparator with

the averaged notch signal. The attenuator essentially provides the difference

factor (or offset) by which the averaged notch signal must be less than the

averaged sideband signal in order for the comparator to indicate the presence

of a notch in the spectrum of the input signal. When the level of the

averaged dc signal from the notch band filter falls sufficiently below the

level of the averaged dc signal of the selected sideband, the comparator

output line changes, activating the rear panel yellow LED, and indicating the

"POSSIBILITY" state. The change in output from the analog comparator also

causes the 15 -second timer to start its timeout cycle.

One of the outputs from the 15 -second timer is the stop -scan/time -constant

control line so that upon changing its state, the scan control circuit stops

the frequency sweep , and the averaging circuits are switched to true

integrators. The Decoder is then in the integrate mode (usually referred to as the POSSIBILITY state) . If the input signal conditions persist such that

the comparator continues to detect the presence of a notch, while the

15 -second timer completes its full 15 second timeout cycle, then another

output from the 15 -second timer activates the 25 -second timer. As soon as it

is activated, this timer changes its output to provide the "RECORD INHIBIT"

state, which activates the rear panel red LED indicator and causes the

recorder electronics to shut down for a minimum of 25 seconds. At the end of

that time, if the comparator continues to detect the presence of a notch,

another 25 second shutdown begins immediately.

21
2.4 Decoder: Physical Measurements

Frequency Response of the Three Bandpass Filters

The amplitude- frequency characteristics of the three bandpass filters used to

decompose the output signal from the frequency modulator into three separate

narrowband signals, as described in section 2.3, is critical to the

performance of the Decoder. It is the shape of these filters, and their

relative position on the frequency scale, that determines the spectrum and

amount of energy from the input signal that is required to cause the

POSSIBILITY and RECORD INHIBIT states of the Decoder to occur.

Figure 17 is a plot of the amplitude response of the three bandpass filters in

the Decoder versus the effective input signal frequency. These

characteristics were measured by a spectrum analyzer with the recorder output

from the spectrum analyzer used to generate the corresponding plots. The

notch bandpass filter is indicated in green, the upper sideband filter in

blue, and lower sideband filter in red. These filters are tuned to capture

signal frequencies that are below the modulation frequency, which was measured

and found to vary from 9.9 to 10.5 kHz. The corresponding input signal

frequency has been derived from the measured filter frequency by subtracting

the measured filter frequency from the nominal 10.2 kHz modulation frequency.

These are high Q filters (Q = -80) with 3 dB bandwidths of about 50 Hz. The

skirts of these filters begin to overlap at rejection levels that are 30 dB or more down from the peaks. The non- ideal ripple in the skirts below these

levels has negligible effect on the Decoder performance. The three bandpass

filters of the Decoder have excellent narrowband rejection characteristics

that select distinctive frequency segments of the incoming signal to be

selected for purposes of notch encoding discrimination. Note that the peak of

the upper sideband filter is a few dB lower than that of the notch and lower

sideband filters. This attenuation appears to compensate for the gain

emphasis of the input filter and AGC characteristic in the frequency region of

the upper sideband, as described below.

Frequency Response of the Anti-Aliasing Filter and AGC Circuit

As described in section 2.3, the operation of the Decoder includes the use of

an anti-aliasing filter, following the input summing circuit, to produce a combined audio signal free of high frequency components. Measurements on the

automatic gain control (AGC) circuits following the filter showed that the AGC

output signal amplitude varied linearly with the AGC input signal amplitude.

That is , the AGC circuit of the Decoder was found to be inoperative

(examination with the aid of the circuit schematic provided by CBS showed that

a connection had been made that disabled the AGC circuits).

Figure 18 shows the plot of the amplitude- frequency characteristic of the

anti-aliasing filter and AGC circuits of the Decoder. In the notch region

from 3 kHz through 5 kHz the response is nearly flat, except around 4.5 kHz

where there is several dB of relative gain emphasis. Above 5 kHz the response

rolls off smoothly to reject high frequency components in the audio input

signal. Similarly, below 3 kHz the low frequency components of the input

22
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(9P) NIV9

FREQUENCY

(kHz)
(9P) NIV9

FREQUENCY

(kHz)

Figure

18

Amplitude

-frequency

response

of

the

anti-aliasing

filter

and

automatic

gain

control

(AGC)

circuits

in

the

Decoder:
signal are rejected, particularly those around the power line frequency, so as

to prevent any unwanted 60 Hz (and related harmonics) from affecting the

operation of the discrimination circuits.

By comparing figures 17 and 18, the total frequency characteristics of the

Decoder can be inferred. The total input signal spectrum is first attenuated

by the characteristic of figure 18, and then the signal is separated into the

three narrow frequency bands of figure 17. The frequency sweeping operation

of the scan controller effectively moves the three bandpass filter

characteristics slowly over about 300 Hz above and below the nominal frequency

position shown here. The gain emphasis of the filter/AGC characteristic in

the upper sideband region is thus compensated by the slightly greater

attenuation of the upper sideband filter.

Steady- State . Three -Tone. Sine -Wave Signals

As described above in section 2.3, the Decoder circuit responds to the

frequency components of the input signal primarily in the vicinity of the

notch band and upper and lower sideband frequencies. Consequently, the

performance of the Decoder can be characterized on a steady- state basis by

applying an input signal composed of three superimposed sinusoidal waveforms

whose frequencies are at (or near) the center of the three bandpass filters of

the Decoder, and whose amplitudes can cause the POSSIBILITY and RECORD INHIBIT

states to occur. For the purpose of these tests the scan controller was

deactivated. The three- tone, sine-wave signal had its three frequencies

adjusted to correspond to the center of the three bandpass filter

characteristics that were measured and described above. The level of the

notch frequency input was varied relative to the level of the upper and lower

sideband inputs to determine the levels at which the POSSIBILITY state was

activated and inactivated, and the level below which the RECORD INHIBIT state

occurred.

Figure 19 is a plot of the data obtained on the level differences between the

notch frequency input and the smaller of the sideband inputs for which the

POSSIBILITY (activated and inactivated) and RECORD INHIBIT states were

observed as a function of the lower sideband level. For these measurements

the upper sideband amplitude was held at approximately -43 dBV. The level

difference at which the POSSIBILITY state is activated (green curve) is about

2 to 4 dB below the level difference at which the POSSIBILITY state is

inactivated (blue curve). More important, however, is the plot of the level

difference that causes the RECORD INHIBIT state to occur (red curve) . Note

how this curve deviates from the POSSIBILITY activated state for lower

sideband levels above about -53 dBV. As the lower sideband level increases,

the difference level required to cause RECORD INHIBIT decreases from the

POSSIBILITY activated level up to a sideband level of about -23 dBV where it

drops off rapidly until, above about -16 dBV, a RECORD INHIBIT state could not

be obtained at any (low) difference level.

Analysis of the block diagram of the Decoder circuitry given in figure 16

indicates that the difference characteristic given in figure 19 should depend

essentially on the operation of the attenuator and analog comparator circuits.

As described in section 2.3, the attenuator provides the difference factor (or

25
(SP) 30N3d3ddlQ 13A31

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fe

26

LOWER

SIDEBAND

LEVEL

(dBV)
offset) that the analog comparator uses to make its threshold level

comparison. On this basis, this difference level comparison would be expected

nominally to be constant over the operating range of input levels, which would

yield a flat line difference characteristic with the POSSIBILITY activated and

RECORD INHIBIT states at the same level. NBS found that the actual difference

characteristic exhibited by the Decoder (as shown in figure 19) is more

complex, in part due to insufficient dynamic range for making the difference

level comparison (as will be described) , and to the limitations of actual

(nonideal) circuit components.

During the POSSIBILITY state the output voltage level of the integrators

slowly increases. With the lower sideband input level above about -53 dBV, it

was observed that the output of the lower sideband integrator saturates at ~14

V dc before the 15 second timeout, with a corresponding output from the

attenuator of ~3.4 V dc. Since the output voltage of the notch band

integrator does not get attenuated, it also saturates at ~14 V dc . Therefore,

whenever the notch band integrator output voltage exceeds ~3.4 V dc , the

analog comparator output changes out of the POSSIBILITY state (integrate mode)

into the scanning mode. If this change occurs before the 15 second timer

completes its timeout cycle, the RECORD INHIBIT state is not reached. With a lower sideband input level above about -53 dBV, the ~3.4 V dc maximum output

level from the attenuator is exceeded before 15 seconds, so that if the output

from the notch band integrator continues to increase above ~3.4 V dc, the

analog comparator changes state. Therefore, the difference level between the

notch signal input and the lower sideband input must be smaller than that

required to cause the POSSIBILITY activated state, in order to prevent

saturation and to cause a RECORD INHIBIT state.

Next, the same three- tone, sine-wave signal was applied to the Decoder and the

notch amplitude varied as above, only with the upper sideband amplitude held

at approximately -23 dBV. This characteristic is plotted in figure 20. It

differs considerably from the characteristic described in the previous

paragraphs. In this case, the level differences observed for obtaining the

POSSIBILITY state are more variable, and at a somewhat lower level. The

difference level required to cause RECORD INHIBIT also deviates considerably

more from the POSSIBILITY activated level with the lower sideband level above

-53 dBV. The RECORD INHIBIT state could not be obtained for any (low)

difference level when the lower sideband level was greater than about -21 dBV.

Figures 21 and 22 correspond to figures 19 and 20, respectively, and are the

plots obtained from the data taken on the level differences for the three

Decoder states (POSSIBILITY activated, POSSIBILITY inactivated, and RECORD

INHIBIT), as described above, except that the lower sideband is held constant

at -43 and -23 dBV levels while the upper sideband level is varied. Very

similar results were obtained for this symmetrical three- tone, sine-wave

testing of the opposite sideband. These results reinforce the analysis given

above of the performance of a Decoder having these level difference

characteristics

.

27
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Functional Triangle. Ramp (Sawtooth), and Square-Wave Signals

Another test of the performance of the Decoder is to apply input signals with

fundamental time -domain waveforms that may be representative of those found in

many kinds of audio input signals. For this purpose, functional triangle,

sawtooth (ramp), and square-wave input signals were utilized. These signals

are fairly rich in harmonics of the fundamental frequency of the waveform;

hence, the amplitude -frequency spectrum of these signals can be expected to

potentially satisfy the conditions required for the Decoder circuitry to

detect the presence of a notch.

In this case, a steady-state (fixed amplitude and fundamental frequency)

functional waveform is applied while the scan controller of the Decoder is

operated in the search mode (i.e., the center frequency of the frequency

modulator is varied approximately 300 Hz about its nominal value) . As with

any kind of input signal, if the POSSIBILITY state exists during the sweep

time of about 35 seconds, the scan controller stops the frequency modulator at

that point, and the Decoder begins integrating the bandpassed signals. For

these functional waveform tests, input signal levels of both relatively low

amplitude (~-30 dBV) and high amplitude (~-10 dBV) were applied. Also, the

fundamental frequency of these functional waveforms was varied from 200 Hz to

2000 Hz in (generally) 50 Hz intervals (see figure 23)

.

For triangle waveforms at the -30 dBV level, the amplitude of the spectral

content in the notch region is relatively low, and no POSSIBILITY states

occurred. However, at the -10 dBV level, the POSSIBILITY state and RECORD

INHIBIT state were observed for triangle waves at 400 Hz and at 500 Hz. The

POSSIBILITY state only was observed at 450 Hz. The RECORD INHIBIT state at

400 Hz and 500 Hz was obtained at input signal levels as low as ~-15 dBV and

~-20 dBV, respectively.

Square-wave input waveforms, which are rich in odd harmonics, caused

POSSIBILITY/RECORD INHIBIT states to occur at both the -30 dBV and -10 dBV

levels in the Decoder. For a -30 dBV square -wave signal level, RECORD INHIBIT

was obtained for inputs with fundamental frequencies from 380 through 515 Hz.

For a -10 dBV level, RECORD INHIBIT was observed with fundamental frequencies

from 325 through 500 Hz and from 600 through 675 Hz. Figure 24 shows the

frequency spectrum of a square wave input with a fundamental frequency of 486

Hz superimposed on the three bandpass filter characteristics plotted on a frequency scale from 0 through 5 kHz. Here, the 7th and 9th harmonics

(identified by numbers next to the appropriate peaks) fit very near to the

center frequency of the lower and upper sideband filters of the Decoder while

the relatively small 8th harmonic (due to the non- ideal, square-wave signal)

fits near the center of the notch passband. This spectrum is discriminated by

the Decoder as having the notch characteristic required for causing a RECORD

INHIBIT state.

For sawtooth (ramp) waveforms, which are rich in both odd and even harmonics,

the occurrence of the POSSIBILITY and RECORD INHIBIT states was also prevalent

at both the -30 dBV and -10 dBV levels. For -30 dBV inputs, RECORD INHIBIT was

observed for fundamental frequencies from 700 to 900 Hz and 1050 to 1150 Hz.

RECORD INHIBIT could also be obtained at 600 Hz for an input signal level of

31
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24

Frequency

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(in

green)

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a

486

Hz

(fundamental)

square

wave

input

signal

superimposed

on

the

frequency

response

characteristics

of

the

three

bandpass

filters

in

the

Decoder.
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25

Frequency

spectrum

(in

green)

ofa

337

Hz

(fundamental)

sawtooth

(ramp)

waveform

superimposed

on

the

frequency

response

characteristics

of

the

three

bandpass

filters

in

the

Decoder.

Note

the

position

of

the

10th

and

13th

harmonics

relative

to

the

peak

response

of

the

lower

and

upper

sideband

filters.
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Figure

26

Frequency

spectrum

(in

green)

of

a

486

Hz

(fundamental)

sawtooth

(ramp)

waveform

superimposed

on

the

frequency

response

characteristics

of

the

three

bandpass

filter

in

the

Decoder.
in

co

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Figure

27

Frequency

spectrum

(in

green)

of

an

870

Hz

(fundamental)

sawtooth

(ramp)

waveform

superimposed

on

the

frequency

response

characteristics

of

the

three

bandpass

filters

in

the

Decoder.
(gp) nivo

37

Figure

28

Frequency

spectra

of

an

841

Hz

(in

red),

an

870

Hz

(in

green),

and

a

901

Hz

(in

blue)

sawtooth

(ramp)

waveform

superimposed

on

the

frequency

response

characteristics

of

the

three

bandpass

filters

in

the

Decoder.
~-19 dBV. At the -10 dBV level, however, RECORD INHIBIT occurred for 300 and

350 Hz input frequencies, and from 600 to 1150 Hz, inclusive.

Figure 25 shows the frequency spectrum of a -10 dBV, 337 Hz sawtooth waveform

input signal superimposed on the three bandpass filter characteristics. The

sawtooth signal is rich in both even and odd harmonics of its fundamental

frequency at 337 Hz. It is apparent that the 10th and 13th harmonics of this

sawtooth signal fit very near to the center frequencies of the lower and upper

passband filters, respectively. The 11th and 12th harmonics are on either

skirt of the notch passband filter. Thus, the 11th and 12th harmonics are

highly rejected while the 10th and 13th harmonics are not. Consequently, this

input signal is discriminated as having a notch in its spectrum at the proper

frequencies, and the Decoder causes a RECORD INHIBIT state. Figure 26, on the

other hand, shows the frequency spectrum of a sawtooth signal with a 486 Hz

fundamental frequency. Here, the 7th , 8th, and 9th harmonics fit very near

to the center frequencies of the lower, notch, and upper passband filters.

All three harmonics are detected by the Decoder and discriminated as the

spectrum of a signal that does not have a notch characteristic; hence, this

signal did not cause a RECORD INHIBIT state.

The spectrum of another sawtooth (ramp) signal that caused a RECORD INHIBIT

state is shown superimposed on the three passband filter responses in figure

27. The fundamental frequency of this sawtooth waveform is 870 Hz. The 4th

and 5th harmonics fall very near to the center frequency of the lower and

upper passband frequencies of the Decoder, while there is no energy component

in the notch passband. To show the range over which this condition holds,

figure 28 illustrates the range of fundamental frequencies that the sawtooth

signal caused RECORD INHIBIT (with a fixed scanning frequency), i.e., how far

down on the skirts of the upper and lower sideband filters the 4th and 5th

harmonics caused the RECORD INHIBIT state. The red spectrum is for the

sawtooth waveform with a fundamental frequency of 841 Hz while the blue

spectrum is for the sawtooth waveform with a fundamental frequency of 901 Hz.

The 870 Hz sawtooth waveform is shown, again, in green for reference.

Actual Encoded and Unencoded Audio Signals

During normal usage, a DAT system will likely be used to record audio signals

from a wide variety of sources. For purposes of the NBS evaluation of the

CBS-supplied DAT Recorder/Decoder, the variety of possible source material was

narrowed down to commercially-available compact disc (CD) recordings of music.

All of the CDs used for these tests contained unencoded music, except for one,

which was an unencoded CD-player test disc made by CBS. This disc was used in

helping to evaluate the operation of the CD player and the interface unit

discussed below.

The table given in Appendix B contains a list of the 55 different CDs that

were used in this testing of actual music source material; number 22 is the

CBS test disc. All subsequent references to these discs will use the disc

reference number for identification purposes.

38
As described below, the Decoder was tested using both unencoded and encoded

music in order to determine whether any cases could be observed of false

positive or false negative performance, respectively. The false positive and

false negative nomenclature refers to the performance of the Decoder. During

the playing of the music into the Decoder, the on-off activity (as a function

of time) of the POSSIBILITY and RECORD INHIBIT lights (yellow and red LEDs) on

the Decoder were recorded, as described below. If the Decoder indicated a RECORD INHIBIT during the playing of any part of a musical piece, this result

was designated a "Positive" RECORD INHIBIT case. Similarly, if no indication

of a RECORD INHIBIT was made during an entire piece, this result was

designated a "Negative" RECORD INHIBIT case. Depending on whether the music

was encoded or unencoded, the results of the tests either matched what was

expected, and were designated a "True" case, or did not match, and were

designated a "False" case. Thus, if unencoded music was played into the DAT

Recorder/Decoder and the Decoder indicated no RECORD INHIBITS, this result is

referred to as a "True Negative" case. Conversely, if a RECORD INHIBIT was

indicated, this result is referred to as a "False Positive" case. For encoded

music played into the DAT Recorder/Decoder, on the other hand, a RECORD

INHIBIT result is referred to as a "True Positive" case, whereas a no RECORD

INHIBIT result is a "False Negative" case.

To summarize , table 3 defines the four types of possible test cases with

respect to the encoding of the source and the corresponding performance of the

DAT Recorder/Decoder.

Table 3

The Four Types of Possible Test Cases

Decoder

Indication

Source Material

Encoded Unencoded

RECORD INHIBIT True Positive False Positive

No RECORD INHIBIT False Negative True Negative

The test setup for the unencoded music tests is shown in figure 29. Since

both the CD player and the DAT Recorder/Decoder are designed to work with

standard unbalanced IHF (Institute of High Fidelity) signal levels, the two

(left/right) output channels of the CD player were directly connected to the

input channels of the DAT Recorder/Decoder. The CD player used a cartridge

which allowed up to ten CDs to be played automatically.

39
Figure 29 The equipment setup for running unencoded music tests

During the tests the operation of the CD player and the DAT Recorder/Decoder

was monitored and recorded on the continuous roll of an oscillographic chart

recorder. The Decoder logic signals from the yellow (POSSIBILITY) LED and the

red (RECORD INHIBIT) LED were connected via optical isolators to an interface

box and then to the chart recorder. The use of optical isolators minimized

the effects of induced noise, ground loops, etc. caused by making external

connections to the Decoder circuit. The two (left/right) audio channels and a third signal, which indicated when the CD player was changing discs, were also

connected to the interface box. This box contained electronics to combine the

POSSIBILITY and RECORD INHIBIT signals from the Decoder into one logic state

signal, which was then monitored by the chart recorder. Similarly, the three

signals from the CD player were combined and recorded on another channel.

Thus, this dual-channel recording setup allowed the operation of the Decoder

to be related to the specific disc, track, and time of occurrence within the

track for any test run. Numerous overnight test runs of several hours length

were thereby made unattended.

The setup for running tests with encoded music is shown in figure 30. Because

the CBS -supplied Encoder was designed for use in a recording studio

environment, it has relatively high level, balanced, input and output signals

(different than the IHF standard). Thus, the Encoder should not be directly

connected to the CD player and DAT Recorder/Decoder. Consequently, a commercially- available interface unit (the Match-Maker) was used to provide

compatibility between the Encoder and the other equipment.

40
Figure 30 The equipment setup for running encoded music tests

Because the Decoder scans the three narrow passband filters (see section 2.3

on the operation of the DAT Recorder/Decoder) , the results of the false

positive and false negative tests tend to vary from run to run. The reason

for this variability is that there is not a fixed relationship between the

incoming music and the modulation frequency of the scanner. Also, any notches

that are being detected are of finite duration. This finite duration has

several causes. In the case of false positives, the notches that are being

detected are those that can occur naturally in the frequency spectrum of the

music. Because of the scanning in the Decoder, these naturally- occurring

notches need not be at the nominal encoding notch frequency, but can be

anywhere from about 3.5 to 4.1 kHz. The duration of a given notch in the

spectrum is dependent upon the particular piece of music.

Two examples of music with naturally- occurring notches are shown in figure 31.

This figure shows spectrographs of two selections (disc numbers 17 and 34 from

the table given in Appendix B) which caused false positives. The spectrograph

is a three axis plot with time shown on the horizontal axis and frequency

shown on the verticle axis. For these plots, the time is about one minute and

frequency is from 0 to 6 kHz . The level within each frequency band is

indicated by color; a reference color bar shows the relative levels from 0 to

-80 dB. Spectrograph 31 A shows one minute of music from the Star Wars Proto,

starting at about one minute into the piece. Spectrograph 31 B shows the

first one minute of music from Mendelssohn's Wedding March. False positives

were observed for the Star Wars Proto at about 1:35 into the piece and at

about 0:35 into the Wedding March. Thus, for both spectrographs the false

positives occurred near the center of the time scale.

In the case of false negatives, the notches to be detected are purposely

provided by an encoder. These notches are also of varying duration. As

mentioned in section 2.1, the CBS -supplied Encoder produces notches in only

part of the encoded music. Also, as described in section 2.3, the comparison

41
, ' • . • idutf . ’ • • - ‘
FREQUENCY

(kHz)

cn

CT) o

-*

1:00I 6 ~m

MINUTES

o

MINUTES

I

I

-10

-20

-30

1

I

l

-40

-50

-60

-70

-80

-90

Encoding

Region

Figure 31 Color spectrographs of music with naturally- occurring notches which

caused False Positives. Spectrograph A shows one minute of music

from the Star Wars Proto starting at 1:00. Spectrograph B shows

the first one minute of Mendelssohn's Wedding March.

42

RELATIVE

LEVEL,

dB

process in the Decoder is inactivated if the input music levels are too low at

the sideband frequencies.

With the ~35 second scanning period of the Decoder, it can take the Decoder

from 0 to about 17 seconds to begin identifying the presence of a notch in the

incoming signal spectrum. Then, the notch must be present for an additional

13 to 15 seconds in order to cause a RECORD INHIBIT. Hence, notches that have

a duration longer than about 15 seconds will be detected by the Decoder on

some runs but not on others. The longer the duration of the notch, the more

consistently the Decoder was observed to recognize it and cause a RECORD

INHIBIT. The cases of false positive and false negative performance that were

measured during the NBS tests are summarized in sections 3.2 and 3.3 that

follow.

43
3 . DOES THE SYSTEM ACHIEVE ITS PURPOSE?

3.1 Interpretation of the Question

This question has been interpreted to mean that the copy protection code

system will achieve its purpose if it inhibits the CBS -supplied DAT

Recorder/Decoder from recording all or parts of an encoded signal and does not

inhibit the recording of unencoded signals. A signal is considered encoded if

it has passed through the Encoder; otherwise, it is unencoded.

3.2 Decoder False Positives

As described in section 2.4 above, the tests for cases of false positive

performance occurring in the Decoder were carried out by recording the

POSSIBLE and RECORD INHIBIT states of the Decoder when the DAT

Recorder/Decoder was receiving unencoded music signals. This section

summarizes the results of the test setup and measurements described in section

2.4 under Actual Encoded and Unencoded Audio Signals.

Because of the scanning operation within the Decoder, the RECORD INHIBIT

indication of the Decoder varies from one playing of a compact disc (CD) to

another playing of the same CD (see discussion in section 2.4 above).

Therefore, this behavior required playing the music pieces several times to

determine the random possible Decoder responses. The false positive testing

was done in two phases, as described below. All 54 of the music discs listed

in Appendix B were used to test the Decoder for possible false positive

operation. The Decoder responses obtained are reported both in terms of an

entire CD and in terms of the individual tracks on the discs. The 54 CDs

tested had a total of 502 tracks.

During the first phase of the false positive tests, all of the tracks on the

discs were played into the Decoder. Most of the CDs were played two or three

times; five of the 54 were played only once, and two were played more than

three times. The Decoder indicated a RECORD INHIBIT on ten of the 502 tracks.

These ten tracks were on five different CDs. Table 4 on the next page shows

which discs indicated false positives and the number of passes for each disc.

Thus, with only a few passes, about two percent of the tracks or nine percent

of the CDs had a false positive indication on at least one of the playings.

44
Table 4

False Positive Data

Disc

Number Tracks

Full Disc Run Partial Disc Runs

Number

Of Runs

False

Positive

Tracks

Track

Number

(Times)

Number

of Runs

False

Positive

Rate

3 6 6 1 4 21 10%

11 10 3 0 6 (3:30-5:30) 24 38%

6 48 25%

14 10 3 0 10 (0:00-1:30) 10 20%

10 48 8%

17 10 2 0 1 (0:00-2:30) 79 34%

1 47 11%

25 12 2 1 5 (0:00-2:30) 8 12%

5 47 62%

8 (2:30-5:00) 52 27%

8 47 43%

26 4 2 1 1 47 11%

33 16 3 4 4 21 33%

10 21 14%

11 21 0%

• 15 21 48%

34 9 3 3 1 21 29%

5 21 29%

9 21 86%

48 3 2 0 1 (15:30-18:00) 9 11%

52 11 2 0 5 (0:00-1:30) 14 29%

6 (0:00-2:00) 13 8%

45
The second phase of the false positive tests examined individual tracks or parts of tracks, using a higher number of passes than were run for the

complete CD tests of phase one. The tracks or parts of tracks were selected

based on the POSSIBILITY indication of the Decoder observed during phase one. To become a RECORD INHIBIT, the POSSIBILITY indication must last for about 15

seconds. Thus, tracks or parts of tracks which had relatively long

POSSIBILITY states of about 8 to 14 seconds, were tested further. The

selected tracks and track times are also listed in table 4, along with the

number of runs and the percent rate of false positive occurrences. With this

additional testing, false positive cases were found on a total of 16 different

tracks on ten of the CDs. Thus, over 3 percent of the total tracks tested and

18 percent of the CDs tested had cases of occasional false positives.

3.3 Decoder False Negatives

The tests for cases of false negative performance were carried out by

recording the state of the POSSIBLE and RECORD INHIBIT indications of the

Decoder while it was receiving encoded music signals. This section summarizes

the results of the test setup and measurements described above in section 2.4

under Actual Encoded and Unencoded Audio Signals.

The test signals were encoded by one of the two CBS-supplied Encoders,

designated Encoder 006 or Encoder 012 (corresponding to their serial numbers)

.

Eighteen CDs with 150 tracks were tested for false negative cases using

Encoder 006, and 32 CDs with 280 tracks were tested for false negative cases

using Encoder 012.

The false negative test data is summarized in table 5, which identifies the

discs by their reference number, and gives their associated number of tracks.

In general, each disc was run with the test setup either two or three times.

Seven of the discs (numbers 39 - 45) were tested for higher numbers of runs. For each track on a disc, the track is labeled as a False Negative Track if no RECORD INHIBIT was observed during at least one of the runs. Table 5 gives

the number of false negative tracks for each disc that was tested using the

two different encoders. If (during at least one of the runs) no RECORD

INHIBITS were observed for the entire disc, the disc was labeled as showing a disc false negative (DFN) , and the disc is so marked in table 5.

46
Table 5

False Negative Data

Disc

Number Tracks

Encoder 006 Encoder 012

Number

of Runs

False

Negative

Tracks

Number

of Runs

False

Negative

Tracks

1 9 3 6 2 1 2 5 2 4 2 2

3 6 5 DFN 4 4

4 5 2 DFN 2 3

5 7 2 6 2 2

6 12 2 DFN 2 9

7 5 2 4 2 2

8 9 - - 3 0

9 13 2 12 2 1

10 12 2 12 2 6

12 11 2 DFN 2 0

13 9 3 DFN 2 0

15 11 c . 3 4

16 8 - _ 3 0

18 8 - - 3 4

19 10 • . 3 1 20 4 - - 3 DFN

32 8 3 DFN 2 DFN

33 16 3 14 2 11

34 9 3 8 2 8

35 7 3 DFN 2 1

36 3 3 DFN 2 DFN

37 4 3 4 2 1 38 9 3 8 2 6

39 11 _ 7 8

40 8

- - 7 0

41 9

- - 7 6

42 8

_ 7 0

43 10 - - 6 0

44 10 - - 4 3

45 12 - - 4 9

46 12 - - 3 5

47
The results given in table 5 show a significant difference between the two

Encoders in the false negative data obtained. As mentioned in the description

of the encoders in section 2.1, Encoder 006 encoded less of the source

material than did Encoder 012. A consequence of this difference is that the

overall copy protection system (Encoder and DAT Recorder/Decoder) showed a much higher false negative rate when using Encoder 006 than it did when using

Encoder 012. For example, in using Encoder 006, eight DFNs out of 18 discs

tested were obtained, while in using Encoder 012 only three DFNs out of 32

discs tested were obtained. In terms of tracks, using Encoder 006, the system

had 139 false negative tracks out of 150 tracks, and, using Encoder 012, the

system had 112 false negative tracks out of 280 tracks tested. Thus, in only

a few passes, using Encoder 006, the system did not inhibit the recording of

~93 percent of the tracks at least once, and, using Encoder 012, the system

did not inhibit the recording of ~40 percent of the tracks at least once.

48
4. DOES THE SYSTEM DIMINISH THE QUALITY OF THE PRERECORDED MATERIAL?

4.1 Interpretation of the Question

This question was interpreted as asking whether the quality of the recorded

material is diminished as revealed by objective electrical measurements,

subjective listening tests, or both. These measurements and tests would apply

to the overall performance of the recording and playback system. Electrical

measurements include those parameters that are commonly used to specify the

overall performance of high-quality audio systems. Subjective tests address

whether or not discriminating listeners can detect the presence or absence of

encoding.

4.2 Effect on the Performance of an Audio System

The electrical components of a modern high-quality audio system are typically

designed and built to meet rather tight performance specifications with regard

to their frequency response. For example, a compact disc player or a good

power amplifier typically would have an amplitude -frequency characteristic

that is flat to within within ±0.5 dB over the frequency range from 20 Hz to

20 kHz. When it is specified, the phase response of a good power amplifier

would typically be flat to within ±10 deg. Studio quality recording equipment

would be expected to approach or exceed these same frequency characteristic

specifications.

As indicated in sections 2.1 and 2.2, the Encoder has a notch in its

amplitude- frequency characteristic that is more than 80 dB deep near 3840 Hz

and has a very irregular phase response in that frequency region as well.

Thus, if the Encoder were to be introduced into the music recording and

reproduction process, there would be no question but that the frequency

characteristics of the signal coming out of an amplifier in one's home would

be degraded relative to the signal that was originally recorded prior to copyprevention coding, in that the usual statements of system performance as being

within ±X dB (amplitude) and ±Y deg (phase) would no longer be met.

It is essential, however, not merely to look at physical performance specifications but to address the question of the audibility of the encoding process. That question is examined in the remainder of section 4.

4.3 Subjective Response Testing Procedure

Professor Irwin Pollack of the University of Michigan served as a consultant

to NBS on the design and implementation of the subjective listening tests that

were carried out. His report to NBS, which is reproduced as Appendix C,

discusses the philosophy and rationale behind the design of the tests and

summarizes some of the test findings.

Two separate but related series of listening tests were carried out in which

listening subjects attempted to discriminate between music which was presented

to them "directly'* (not encoded) and the same musical selection presented to

them "encoded" (having passed through one of the two Encoders described in

section 2.1). These two series of tests are briefly described below.

49
The musical selections presented to the subjects were selected from material

suggested by the project sponsors (see Appendix D) or were selected by NBS

staff. The selection and characterization of the music used in these studies

are discussed in section 4.4 and in Appendix E.

The encoded and unencoded music selections presented to the subjects were

prepared so as to be identical except for the effect of the Encoder. For both

types of material, digital signals were converted to analog, passed through

either the Encoder or a straight wire, and then converted back to digital

format. For a given musical selection, every appearance of that selection

with a given encoding state (direct or encoded) on the tape was a digital copy

of the same material so that, for example, the encoded version of a given

selection was always identically the same. The details as to how this tape

preparation was accomplished are given in Appendix F.

The material was presented to the listening subjects using a professional

quality digital audio tape recorder, audio processor, and high quality earphones or loudspeakers. The audio systems that were used are documented in

Appendix G.

Serial Listening Test

In the listening tests that are hereafter referred to as "serial listening

tests," short segments of music were presented sequentially to multiple listeners in a fixed sequence that was established by NBS staff. The music was

presented as pairs that were repeated three times in close succession, with

each member of the pair being either direct (D) or encoded material (E) ; for

example, the presentation might be as follows:

Selection 1 Selection 2

Selection 3

D E D E

EE EE

ED ED

D E

E E

E D

(Different)

(Same)

(Different)

Selection 2 D D D D D D (Same)

Selection 1 ED E D E D (Different)

The subjects were asked to decide whether the two members of the pair were the

same or were different. Over the course of the test period (approximately 2 hours) 24 musical selections were presented in all four of the possible pairings (D-D, D-E, E-D, and E-E) with, as stated above, each presentation consisting of three repeats of a given pairing. Thus, the subjects were asked to

make 96 choices as to whether the two members of a given pairing were the same * or different. For each of the four sets of 24 selections, the order in which

the selections appeared was randomized. For a given selection, the order in

which the four possible pairings appeared was also randomized.

In addition to the musical selections, the presentation tape included a narrative description of the nature of the test and instructions to the

50
subjects. The text of this narrative is included in Appendix H.

The material was presented simultaneously to up to seven subjects, five using

earphones and two using loudspeakers in a specially prepared listening room. The majority of the listening subjects consisted of volunteers who responded

to a request that was sent out to the local chapter of the Audio Engineering

Society or to an announcement that was sent to radio stations and recording

studios in the Washington-Baltimore metropolitan area. Most of these subjects

were individuals who are actively involved in music production, recording, or broadcasting. Additional subjects were selected from NBS employees, not

involved with this study, or from friends who were known to be discriminating

listeners

.

Parallel Listening Test

In the listening tests that are hereafter referred to as "parallel tests," ten

segments of music, each of about 1 minute in length, were presented to one

subject at a time, over earphones. The selections were recorded synchronously

on three parallel pairs (stereo) of tracks on a 24-track digital tape

recorder. One pair of tracks (REF) was a reference pair, where the music was

always presented directly, without encoding. On a second pair of tracks (A),

the material was either encoded or direct, while on a third pair of tracks

(B) , the material was either direct or encoded, with one of the two pairs (A or B) always being encoded material and the other being direct material. The

track pair on which encoded material was recorded was randomly assigned. Each

subject could rewind the tape recorder and switch among the three pairs. of

channels at will. His task was to attempt to identify, for each musical

selection, the track pair containing the encoded material.

The instructions to the subjects are included in Appendix H.

The subjects used for the parallel presentation study were sound recording

engineers, audio equipment designers, or musicians.

4.4 Selection and Characterization of Recorded Material

The material used for the listening tests was taken from compact discs and

from a Kurzweil Model 250 digital sampling keyboard (synthesizer) played by a professional musician in the NBS laboratory. Appendix E provides a list of

the material selected and gives a description of some of the effects that were

noted as likely to be caused by the encoding process. It is important to note

that the effects described below and in Appendix E were observed with the aid

of extensive physical measurements, which assisted the NBS staff member who

selected the material to focus his listening attention on very specific

musical passages or notes.

As indicated above, the compact discs used were chosen from lists (reproduced

in Appendix D) supplied by the Home Recording Rights Coalition (HRRC) and by

the Recording Industry Association of America. Additional material was

selected based on physical measurements and listening by NBS staff, as described below. The list furnished by the HRRC supplied tracks and times at

which audible effects had been noted by their consultant, using signal

51
processing based on his understanding of the encoding process. In the

majority of these cases, the logic used by the Encoders supplied to NBS

prevented the production of the encoding notch at these times. However, the

encoding notch was produced for many examples cited by the HRRC.

Twenty musical items from compact discs were chosen for the serial presentation tests, along with four produced by the synthesizer. The selection was

based upon real-time spectrographs of the music, and upon careful listening by

the NBS staff. The material was selected so that enough musical components

fell in the encoding region to result in a potential for detection. (The

real-time spectrographs used to aid in material selection were obtained using

an Hewlett-Packard (HP) 3565S Analyzer, and an HP 330 Computer. Spectrographs

used to document the actual material used in the listening tests were obtained

with a different HP analyzer and are discussed in Appendix E)

.

Ten of these compact disc selections were chosen for the parallel presentation

test, with the length of each selection being increased from a few seconds to

between 29 and 75 seconds.

In order to classify the possible results of encoding music, six possible

effects were hypothesized, with considerable overlapping among effects. These

hypotheses, based on the spectrographs and on extensive listening by NBS

staff, are summarized below.

1. The deletion, or significant reduction in loudness, of a particular note.

This might happen when the fundamental frequency or a predominant harmonic

of a note occurs within the band of frequencies removed by the Encoder.

2. A decrease in brightness in the timbre of notes or chords. This effect

would occur when considerable harmonic content falls in the encoding

region, but with enough energy at other frequencies that the apparent

loudness of the sound is not changed significantly. Effects 1 and 2 could

shade into each other.

3. A change in the phenomenon sometimes called "chorus” of the string section

of an orchestra. The instruments are not playing precisely in tune, and

individual players are using varying amounts and rates of vibrato. As a result, the spectrum of a note or chord will show a spread over a considerable range of frequencies for a particular harmonic, instead of the

energy being concentrated at one frequency. The encoding method under

study can remove a portion of this spread. The Encoder logic is not very

effective at preventing this sort of change in the signal, which would be

expected to be more subtle than those noted in 1 and 2. Effects 2 and 3

could overlap each other.

4. A change in the character of transient sounds. This effect would be ex- pected to be the most subtle and subjective of all effects noted. Such a change would affect the "impact" produced by percussive sounds, or result

in a slight coarsening of the overall sound.

5. Changes in the nature of a sound due to switching of the Encoder from one

52
state to another (and perhaps back again) within one note or chord due to a slight change in spectral content.

6. Amplitude modulation of a harmonic of a trill of a single instrument or voice, which extends into the encoding region or through it.

The selections of music, in general, were made in order to explore the audibility of rather specific effects upon specific passages. Even more subtle

overall quality changes may occur, but if they do, extended listening in a familiar environment may be required for evaluation.

The recorded material was characterized by obtaining spectrographs of each

musical sample. This was done for both the serial presentation material (24

short samples) and the parallel presentation material (10 extended samples).

As discussed at the end of section 2.4, the spectrographs show time on the

horizontal axis and frequency from 0 to 6 kHz on the vertical axis. Relative

signal level within each frequency band is indicated by color with the level

decreasing as the colors progress through the spectrum: white, pink, red,

yellow, green, cyan, blue, purple, and black.

Figure 32 shows the color spectrographs of the direct and encoded material for

four of the short selections used in the serial presentations. For each pair,

the first (left) spectrograph represents the direct material while the righthand spectrograph corresponds to encoded music. The arrows just below 4 kHz

indicate the encoding region. In figure 32, the Debussy selection shows that

a strong harmonic of the violins is removed. The encoder ceased notching in

the middle of the Barber selection.

Figure 33 shows the spectrographs of the direct and encoded material for the

Bizet selection used in the parallel presentations, with the lower spectrograph corresponding to encoded music. This material is seen to be encoded

most of the time, resulting in partial removal of a harmonic of a solo violin.

4.5 Results of Listening Tests

The results of the subjective listening tests show that, while the effects of

the encoder are fairly subtle for some musical selections, there are some

selections for which the subjects detected differences between direct and

encoded material. The data indicate that the ability to hear effects of the

encoder varies substantially between individual listeners and, especially,

between musical selections.

Serial Listening Test

A total of 87 subjects took the serial listening test -- 61 used earphones, 23

listened over loudspeakers, and 3 more used a "mixed" listening mode in which

they took part of the test using earphones and listened over the loudspeakers

for the remainder. Generally speaking, the earphone group achieved higher

53
discrimination scores than the loudspeaker group. The tests were not designed

to evaluate differences between earphone and loudspeaker listeners in

sensitivity to Encoder effects. Whatever differences may exist do not

preclude using both groups to test the general hypothesis of no audible

difference between direct and encoded music. In fact, both groups provide

evidence that the difference is audible.

The design of the serial listening test ensured that a subject who

persistently answered either "different" or "same" to most of the test items

could not, by so doing, achieve a high score. This is because of the balance

in the way the musical selections were presented: each selection was

presented in all four possible combinations: DD, DE, ED and EE (where D = direct, or unencoded, and E = encoded). As an extreme example, a subject who

answered "different" for every test item would get exactly half of the items

correct (the DE and ED items) and the other half wrong (DD and EE) , for a total score of exactly 50% correct.

In summarizing the test results, it is of interest to consider the average

scores (percent correct discrimination) obtained for individual musical

selections and for listeners.

Average Scores for Musical Selections

Table 6 shows the average correct discrimination scores, in percent, for each

of the 24 musical selections in the study, averaged across all 87 listening

subjects. The selection names are defined fully in Appendix E. Since each

selection was presented four times (DD, DE, ED and EE) in the serial test,

each average is based on 87 x 4 = 348 responses to the test items. Among the

348 responses there were 2 instances of missing data, where the listener did

not mark either "D" or "S" in response to a test item. These cases were

scored as incorrect.

Of the 24 selections in table 6 , eight have average scores that exceed 50% by

a statistically significant margin (p < 0.025). Statistical significance is

evaluated by calculating the probability of obtaining a score as high or higher than the observed score on the assumption that the listeners could not

hear any difference but were only guessing -- with a 50% probability of

guessing the correct answer -- on each of the 348 responses. For example,

using the binomial probability distribution, one can calculate that the

probability is 0.95 that the observed percent of correct responses will be

within ± 5.4% of 50% if 348 responses are obtained by guessing. In all, 9 of

the 24 musical selections differ from 50% by more than 5.4%. This is

substantially more deviation from 50% than could be explained by chance, since

only about 1 or 2 selections (about 5%) should fall outside the interval 50% ±

5.4% if the responses were all guesses. Of the 9 selections with scores

outside the central 0.95 probability range, 8 were above that range (i.e.

above 55.4%) and the remaining 1 was below that range (less than 44.6%).

The most pronounced deviation above 50% was obtained for the SI selection,

which was correctly discriminated 90.5% of the time in the test. This

selection consists of an F7 -B7 chord created on a music synthesizer

54
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Figure

32.

Color

spectrographs

of

direct

(left)

and

encoded

(right)

material,

the

serial

listening

study,

from

Debussy,

Barber,

Beethoven

chord,

Beethoven

chorus

selections

(see

Appendix

E

for

a

more

complete

description

of

these

selections)

.

Table 6. Average score summary for all listeners in the serial listening

tests. Percentage scores by selection.

Selection Number of Percent

Name Responses Correct

Debussy 348 52.6

Barber 348 49.7

Beethoven chord 348 55.5

Beethoven chorus 348 43.7

Bizet 348 55.5

Wilson 348 50.0

Striesand 348 49.1

Beethoven Triangle 348 54.0

McGovern 348 53.2

Weather 348 46.3

Copland 348 54.9

Bernstein 348 49.1

Pink Floyd 348 48.6

SI 348 90.5

Prokofiev 348 70.1

S2 348 54.3

Metheny 348 54.0

S3 348 55.5

S4 348 49.4

Messiaen 348 56.3

Bach 348 51.4

Respighi 348 62.9

Handel 348 49.4

Ravel 348 55.7

specifically to be an easily discernible case, and the results show clearly

that it was indeed easy to detect.

The second highest average score was obtained for the Prokofiev selection. In

this case, 70.1% of the responses for this selection correctly distinguished

whether the members of the pair presented were the "same" or "different" with

respect to encoding. The statistical significance of this result may be

described as follows: the probability of obtaining an average score of 70.1%

or more from a set of 348 responses generated by a 50-50 chance mechanism is

much less than one in a million (« 0.000001).

57
To summarize, the significance probabilities for the 8 selections with scores

above the central 0.95 probability interval were as follows:

Selection

Percent

Correct

Significance Probability

(Prob > % obs'd by chance)

Beethoven chord 55.5 0.02

Bizet 55.5 0.02

SI 90.5 « 0.000001

Prokofiev 70.1 « 0.000001

S3 55.5 0.02

Messiaen 56.3 0.009

Respighi 62.9 < 0.000001

Ravel 55.7 0.017

The Beethoven chorus, whose score fell below the 0.95 probability interval,

had a score of 43.7 percent correct, corresponding to a significance

probability of 0.009. These results, taken together, show clearly that at

least selections SI, Prokofiev, and Respighi showed discernible audible

effects of encoding in the serial tests.

Average Scores for Listeners

Another way to view the data is to consider the percent scores obtained by

each subject, averaged over the 24 selections (96 test items). As discussed

above, the 24 selections differ substantially in difficulty of detecting the

effect of the encoder. Some of the chosen musical selections were chosen with

the expectation that audible effect of the encoder would be very subtle. For

such selections, it is reasonable to expect that a typical listener would have

about a 50% average probability of giving the correct response to the

Same/Different questions in the serial listening test.

The effect of including selections having rather subtle or nonexistent audible

effects of encoding is to compress the test scores for subjects in a closer

range around 50% compared to the range of average scores for selections. This

is because the average score computed for a given subject represents an

average of the probability of correct discrimination, averaged across the 24

selections

.

For example, if half of the selections were subtle, having a 50% chance of

correct discrimination, and the other half had a 70% chance, the average score

expected for listeners would be 60%, the average of 50% and 70%. The data

suggest that at least half of the selections were subtle and that the range of

probabilities of correct responses is greater than this illustration. As

another example, suppose that 2/3 of the selections (i.e. 16 of the 24) were

subtle (with 50% probability of correct discrimination) , and that the

remaining 8 selections had probabilities spread out from 60% to 95% as follows: 60%, 65%, 70%, 75%, 80%, 85%, 90%, 95%. Then the expected score for

listeners on such a collection of selections would be 59.2%, the overall

average of all 24 probabilities. These hypothetical examples illustrate that

the result of constructing a test using a mixture of selections with subtle

58
and more noticeable effects is to reduce the average scores computed for

listeners

.

The data on average scores for earphone and loudspeaker subjects,

respectively, are shown in tables C-3 and C-4 of Appendix C. Another

presentation of these data is provided by figures 34 and 35. In these

figures, the response behavior of listeners is compared for test items

consisting of truly different pairs (DE or ED) , and test items that were truly

the same (DD or EE) . Specifically, the figures show the percentage of times

each subject replied "Different" when the test items were in fact different

(vertical axis) plotted against the percentage of times that same subject

replied "Different" when the test items were in fact the same (horizontal

axis)

.

The schematic plot shown in figure 36 illustrates some extreme possibilities.

A listening subject who never responded "Different" for either the DE and ED

pairs or for DD and EE pairs would be plotted on the diagonal line at the

point labeled "A". Similarly, a subject who always responded "Different" for

both the DE and ED pairs and also for the DD and EE pairs would be plotted on

the diagonal line at the point "B"

.

In contrast, a subject who achieved

perfect discrimination would be plotted far above the diagonal at point "C,"

which corresponds to responding "Different" 0% of the time for DD and EE items

and 100% of the time for DE and ED items

.

Subjects achieve scores that fall above the diagonal if, and only if, they

respond "Different" more often for pairs that are truly different than for

pairs that are the same. Clearly, subjects who achieved an average score of

more than 50% correct are exactly those whose scores are plotted above the

diagonal in these figures. The fact that the points in figures 34 and 35 are

somewhat spread out from left to right in the plots is evidence that the

subjects had a range of propensities to guess "Different" when they were not

sure

.

The figures reflect the fact that a majority of the subjects achieved scores

somewhat above 50% correct -- in other words, they were more likely to

describe the pair as "Different" when they were truly different than when they

were truly the same. The dashed curves in the figures enclose the range

within which scores would fall 95% of the time in the absence of an encoder

effect.^ That is, the dashed curves form a 0.95 probability interval that

would apply to the case of all responses amounting to 50-50 guesses.

Several of the subjects achieved scores outside the 0.95 probability interval,

so their performance can not be readily explained by "chance." Specifically,

there are 10 earphone subjects and 3 loudspeaker subjects whose average scores

are significantly higher than 50%, with (one-sided) significance probabilities

< 0.025. Among the 84 (=61+23) subjects represented in figures 34 and 35, it

^The dashed curve in the figures is constructed so that a plotted point,

(x,y) is inside the curve if |x-y| < 2 [ 2p ( l-p)/48

] **, where p = (x+y)/2. This

corresponds to the usual procedure for testing the hypothesis that two

proportions, x and y, are both estimates of the same probability.

59
Figure 34 "Signal detection" plot for n=61 earphone listeners. Each point

(x,y) represents one subject whose plotting position is defined as x = percentage of DD and EE test items to which the subject

answered "Different," and y = percentage of DE and ED test items

to which the subject answered "Different." Note that y is a percentage of correct answers, while x represents wrong answers.

The dashed curves enclose a region in which points would fall with

probability 0.95 if there were no audible effect of the encoder.

(The overall percentage of correct answers for a given subject is

[y + (100-x)]/2, which exceeds 50% when y > x, i.e. when the point

is above the diagonal line.)

might be expected that 2 or 3 (0.05 x 84 = 2.1) would lie outside the 0.95

probability interval by chance. The fact that the scores for 13 subjects fell

outside the interval, with all lying above the interval, supports the

conclusion that the result is not due to chance but rather to the presence of

a real average audible effect for the selections in the test.

A prominent qualitative feature of the figures is the preponderance of data

above the diagonal "chance" line. This reflects the fact that a substantial

majority of subjects scored above 50% on the tests. We now consider whether

the size of this majority could be due to chance.

60
J

Figure 35 "Signal detection plot for n=23 loudspeaker listeners. The

interpretation of this figure is the same as that for figure 35.

The statistical significance of the tendency of results to fall above the diagonal

in the figures (scores above 50% in tables C-l and C-2) can be evaluated by the

sign test, as follows. Of the 84 (=61+23) subject scores represented in figures

34 and 35, 69 of them lie above the diagonal, 7 are exactly on the diagonal (for

subjects who scored exactly 50% correct), and 8 are below the diagonal. Under the

null hypothesis of no audible encoder effect, the probability of achieving a score

above 50% would be 0.5. In that case, a calculation using the binomial

probability distribution shows that the probability of a obtaining 69 or more

scores above 50% (and 8 or less below 50%) is about 3 x 10“^, or 3 in one

trillion. Thus the data strongly suggest, through the persistent tendency to

achieve scores above 50%, that there is an audible encoder effect reflected in the

subjects' responses to the test selections.

It may be recalled that the 24 music selections chosen for the test include 4

selections that were created using a keyboard synthesizer. To investigate the

degree to which the strongly significant result of the sign test above is due to

the presence of synthesizer selections in the serial test, we computed the average

scores per subject after deleting the synthesizer selections. In this case the

average scores were based on 80 test items per subject, rather than 96 for the

61
Figure 36 Schematic plot of responses for listening subjects. Point A

corresponds to a listener who answers "Same" for all test items,

whether they are truly same or different. Point B corresponds to a listener who always answers "Different." Listeners who answer

"Different" the same percent of the time to both truly same and truly

different pairs will fall on the diagonal line. Point C corresponds

to a listener who discriminates perfectly between same and different

pairs

.

full test. For the reduced test data, consisting only of recorded music

selections, a total of 58 of the 84 earphone and speaker subjects achieved scores

greater than 50%. The number of subjects with scores of exactly 50% was 4 and the

number with scores below 50% was 22. For the total of 84 subjects, the sign test

evaluates the probability of getting 58 or more scores over 50% (and 22 or less

below 50%) assuming chance responses. The calculated probability is 0.000035 in

this case, so the observed excess of scores above 50%, compared to the number

below 50%, would happen only about 3.5 times in 100,000 assuming chance responses.

Thus, while the significance probability is somewhat less extreme than was

obtained for the full test of all 96 items, the reduced test data still show quite

strong evidence of an audible effect.

62
j

This analysis of average scores for listeners was, for convenience, based only on

the 84 subjects whose scores are plotted in figures 34 and 35, so the three

listeners who used a mixed format of earphones and loudspeakers were not included.

Since these three additional subjects all scored higher than 50% on both the full

test and also the reduced test obtained by dropping the synthesizer selections,

their inclusion in the calculations would only strengthen the evidence for an

audible effect of encoding.

Parallel Listening Test

The results of the parallel listening tests are presented in table 7. Again, the

data show that the ability to detect an audible encoder effect varies

substantially between subjects and between musical selections. Some of the

selections (e.g., Respighi) that were most consistently detected in the serial

tests were apparently harder to detect in the parallel format tests. Some

selections (e.g., Metheny) were detected more often in the parallel format than

the serial tests. The Prokofiev selection achieved consistently high scores in

both the parallel and serial tests.

To evaluate statistical significance for the listeners' scores, it can be shown

that the probability that a given subject would be able to correctly identify 9 or

10 out of 10 selections, assuming there is no audible effect and all responses are 50-50 guesses, is 0.011. So, subjects D, G, and 0 scored significantly higher

than can reasonably be accounted for by chance (p < 0.011).

Turning to the average scores for musical selections, the probability that any

given musical selection would be correctly identified 12 or more times out of 15,

assuming no audible effect, is 0.018. Since the encoded track was correctly

identified by 12 out of 15 subjects for both the Metheny and the Prokofiev

selections, these results are statistically significant at the p = 0.018 level.

(Among the four selections that had only 14 responses , the corresponding

probability of getting 10 or more correct, the highest score observed, is 0.09.)

Conducting independent significance tests repeatedly for 24 selections could be

expected, on average, to produce about 0.4 significances at the 0.018 level (0.4 = 0.018 x 24). The observed data produced results at the 0.018 significance level

for two selections, and in addition two other selections achieved scores of 11 out

of 15 correct (significance probability of 0.059), amounting to substantially more

significances than would be expected by chance. Together, these calculations show

that the test results are not well-explained by chance, even after allowing for

the fact that multiple significance tests were performed.

In summary, the data for the Metheny and Prokofiev selections, at least, show

statistically significant evidence (p < 0.02) that the encoded channel is audibly

distinguishable from the unencoded version. For the remaining selections, the

proportion of times they were correctly identified, although quite high in some

cases, is not based on a large enough sample of subjects to show a statistically

significant difference (at the p < 0.05 level) from 50%.

63
Table 7. Subject responses for all listeners in the parallel listening tests

Subj ect

Fractioi

Selection A B C D E F G H I J K L M N 0 Correct

1 . Bizet C • C C C w C C w w w w C C w 8/14

2. Streisand w C C C w C C w C C C C w C C 11/15

3. Beethoven Triangle C • C C C w C w C C w C w c c 10/14

4. Copland w w C C C w C w C w w C w w c 7/15

5. Metheny C C C C C C C C C C w w C w c 12/15

6. Prokofiev c w w C c C C C C C w C C c c 12/15

7 . Bernstein c w C C w C c w w C C c C c c 11/15

8. Messiaen c • c C w C w w C w w w w c c 7/14

9. Bach w • c C C w C w C w w C C. w c 8/14

10. Respighi w w w C C w c w w w w w w c c 5/15

Total Correct 6 2 8 10 7 5 9 3 7 5 2 6 5 7 9

C = Correct

w = Wrong

° = No Response

Average percent correct: 100 x (91/146) =62.3

64
5. CAN THE SYSTEM BE BYPASSED, AND IF SO, HOW EASILY?

5.1 Interpretation of the Question

This question is primarily one dealing with the Decoder portion of a DAT

Recorder/Decoder. Since the CBS -supplied DAT Recorder/Decoder is a prototype

unit, access to individual components may not be representative of the final

design. Therefore, only the possibilities of circumventing the Decoder by

means that do not require physical modification methods were considered, along

with their relative ease of implementation.

5.2 Description of Defeat Methods

There are basically two external signal conditioning methods of circumventing

the Decoder operation. One approach is to add electronically to the incoming

signal of the DAT Recorder an arbitrary signal with frequency components in

the notch region that serve to prevent a POSSIBILITY activated and RECORD

INHIBIT state. The total signal would then be recorded on the DAT

Recorder/Decoder, but could likely be audibly objectionable. Upon playback of

such a digital audio tape, however, the added "defeat" signal could be

subtracted electronically from the recorded signal to restore the resultant

audio signal to nearly what was input to the DAT Recorder/Decoder. That is,

the listener of the resultant signal would hear the original signal (as

encoded), plus any residual, uncompensated components of the "defeat" signal.

Because of the need for additional playback circuits and the lack of a digital

audio tape which can be copied and played without the need for special

playback circuits, NBS did not consider this approach during the evaluation of

the CBS copy prevention code system.

The second (and more preferable) external approach is to add electronically to

the incoming signal of the DAT Recorder/Decoder just enough of a signal (with

frequency components in the notch region) to compensate for what the encoding

(notch filtering) process removes from the original signal. Ideally, the

spectrum (both amplitude and phase) of the original audio signal would be

perfectly reconstructed before being input to the DAT Recorder/Decoder. Of

course, perfect reconstruction is not possible (due to noise, imperfect

electronic components, etc.), even if the encoder's notch filter

characteristics are known in great detail. However, several schemes for

filling in the notch region amplitude spectrum have been attempted during this

evaluation, and are described below.

Defeat Method 1 Figure 37 shows a basic means for adding to the encoded input signal to be

recorded another signal, such that the amplitude spectrum of the output signal

no longer contains enough of a notch in the spectrum to cause a RECORD INHIBIT

state to occur. The added signal is derived from a wideband noise source with

its output passed through a bandpass filter, centered in the notch region at

about 3.8 kHz. By adjusting the amplitude of the added signal, the notch in

the input signal spectrum is effectively filled in. This method is referred

to as Method 1. There are numerous possible electronic circuit

65
implementations of this method. Shown in Appendix I is the schematic diagram

of a noise source, a 3.8 kHz bandpass filter, and a summing amplifier

constructed at NBS for purposes of implementing Method 1.

INPUT 3 OUTPUT

SOURCE

NOISE

l\

l\ J

2 .8 kHz

BAND PASS

FILTER

Figure 37 Defeat method 1 that uses a bandpass -filtered noise source added

to the encoded input signal.

Figure 38 displays the spectrum obtained for an example of an input signal

from a broadband noise source which has been encoded with an obvious notch in

its spectrum (shown in green). This signal, of course, caused the RECORD

INHIBIT state to occur in the Decoder. The blue spectrum shown is that of the

added bandpass -filtered noise source (Method 1, as described above and with

the circuits given in Appendix I). The resultant output signal spectrum is

shown in red (offset for clarity) , and can be seen no longer to contain a notch in its spectrum. This signal, of course, did not cause RECORD INHIBIT

in the Decoder.

Defeat Method 2

Alternatively, the source for a defeat signal can be derived from the encoded

input signal itself. Figure 39 shows one method for obtaining such a defeat

signal, which is obtained by passing the input signal through a bandpass

filter an octave below the notch band, full wave rectifying the 1.9 kHz

bandpass filter output (to double the frequency) , and then filtering the

rectifier output with a 3.8 kHz bandpass filter. This defeat signal is then

added to the input signal as shown, and the resultant output (sent to the DAT

Recorder/Decoder) will have its amplitude spectrum continuously adjusted in

the notch region by the amount of energy contained in the original audio

signal at around an octave below the notch center frequency. This method is

referred to as Method 2. The assumption in this method is that if there was

an encoding notch placed in the incoming signal, then there is a likelihood

that the original, unencoded audio signal had spectral components at an octave

below the notch. Detecting the presence of these components in the input

signal can determine when (and in what amount) to add a defeat signal at the

notch band to the input signal. Shown in Appendix I is the schematic diagram

of a 1.9 kHz bandpass filter, a full wave rectifier, and the other components

constructed at NBS for purposes of implementing Method 2.

66
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Figure

38

Frequency

spectra

of:

(1)

a

broadband

noise

input

signal

containing

the

encoding

notch

(green

spectrum),

(2)

the

added

bandpass

filtered

noise

source

(blue

spectrum),

and

(3)

the

resultant

output

signal

(red

spectrum)

.
Figure 39 Defeat method 2 that uses a 1.9 kHz component of the input

signal doubled to match the 3.8 kHz encoding notch frequency.

Defeat Method 3

Figure 40 shows a variation of the basic bandpass -filtered noise source used

as the added defeat signal (see figure 37). In this case, however, rather

than continuously adding the bandpass -filtered noise source to the input, this

signal is amplitude modulated by another signal derived from the input signal

before being summed with the input signal. A dc modulation signal,

proportional to the frequency components of the input signal around 2.7 kHz,

is obtained with the use of the bandpass filter, full wave rectifier, and

lowpass filter circuits. Thus, the amount of bandpass -filtered noise (at 3.8

kHz) added to the input signal depends on the amount of energy in the input

signal spectrum around 2.7 kHz. With the presence of signal components in

this part of the input spectrum, the defeating signal in the 3.8 kHz region is

then added in a more "masked" fashion. This method is referred to as Method

3. Shown in Appendix I is the schematic diagram of a 2.7 kHz bandpass filter,

a lowpass filter, and the other components constructed at NBS for purposes of

implementing Method 3.

Figure 40 Defeat method 3, which is similar to method 1, and uses the

amplitude of the 2.7 kHz component of the input signal to

modulate the bandpass -filtered noise.

68
Defeat Method 4

Figure 41 is very similar to the circuit of figure 40, except that the source

of the defeat signal, rather than being bandpass -filtered noise, is derived

from the input signal, as discussed previously. Thus, not only is the amount

of added defeat signal dependent on the frequency components in the input

signal at an octave below the notch, this signal is also amplitude modulated

by a dc modulation signal proportional to the amplitude of the frequency

components of the input signal around 2.7 kHz. The encoding notch of the

input signal is thereby filled in dynamically by the amount of energy at two

lower frequency bands of the original audio signal. This method is referred

to as Method 4. Shown in Appendix I are all the components constructed at NBS

for purposes of implementing Method 4.

Figure 41 Defeat method 4, which is similar to method 2, and uses the

amplitude of the 2.7 kHz component of the input signal to

modulate the frequency doubled 1.9 kHz component of the input

signal

.

Defeat Method 5 A further variation of this dynamic modulation theme is provided in figure 42

where the dc modulation signal of figure 41 is derived from the amount of

input signal energy in either of the two notch sidebands, centered at about

3.5 and 4.3 kHz. The assumption in this case is that the amount of added

defeat signal (besides being dependent on having input signal components at an

octave below the notch band) should be proportional to the amplitude of the

input signal spectrum on either side of the notch band. If there are no

appreciable sideband components, then there is not likely to be either a purposeful (encoding) notch or a "natural'' notch in the amplitude spectrum of

69
the incoming signal. Of course, this modulation scheme could also be applied

to a bandpass -filtered noise source as well. This method is referred to as Method 5. Shown in Appendix I is the schematic diagram for a 3.5 kHz bandpass

filter, a 4.3 kHz bandpass filter, and all the other components constructed at

NBS for purposes of implementing Method 5.

INPUT <+>«

OUTPUT

4.3 kHz BAND PASS FILTER

A

(+)—- 1 -hS)

FULL WAVE RECTIFIER

LOW PASS FILTER

3.5 kHZ BAND PASS FILTER

1.9 kHz FULL WAVE 3.8 kHZ BAND PASS RECTIFIER BAND PASS filter filter

Figure 42 Defeat method 5, similar to method 4, and uses the amplitude of

the 3.5 kHz and 4.3 kHz components of the input signal to

provide the modulation.

5.3 Ease of Implementation

In an effort to demonstrate the ease or difficulty of bypassing or defeating

the Decoder in the CBS -supplied DAT Recorder/Decoder, several electronic

circuits have been assembled that bypass the Decoder, which are based on the

defeat methods described in section 5.2 above. As shown in Appendix I, the

ease of bypassing can be judged by the relative simplicity involved in these

circuits, and how easy they are to implement.

The basic active electronic components in all of these circuits are

semiconductor diodes and operational amplifiers. Together with passive

resistor and capacitor elements, most of these circuits can be implemented

with about a dozen commonly available, off-the-shelf parts. The power

supplies needed to operate these circuits are also available as standard

modules. It is estimated that the parts cost in small quantities of the

circuits required for each of the methods described in this report is less

than $100, including the dc power supplies.

The degree of knowledge required to conceive, design, and test the defeat

circuits shown in this report is generally that of a professionally trained

70
electrical/electronics engineer. However, the degree of knowledge required to

construct the defeat circuits described herein is that of an electronics

technician, familiar with reading circuit schematics. For the very simple

method of adding a signal from an oscillator (at or near the notch frequency)

to the encoded audio signal, in order to defeat the Decoder in a DAT recorder,

this approach and degree of knowledge is available to anyone who reads about

the subject of the proposed CBS copy prevention protection scheme described in

popular audio magazines. On the other hand, to determine just how well a given defeat method can be made to work requires test equipment, such as a spectrum analyzer, and detailed knowledge of how the Decoder circuit operates,

knowledge generally not available to a novice.

5.4 Audibility of Defeat Methods

All of the circuits shown in Appendix I were constructed for implementing the

five defeat methods described in section 5.3 above. The relative audible

effects of the defeat methods were informally evaluated by a subjective

listening test made by NBS staff. Table 8 gives a summary of the results of

this test.

The relative audibility shown in table 8 for the five defeat methods is that

observed by the NBS listening staff. No claim is made that this table is

representative of the audible effects of these five methods on an absolute

basis. However, the relative audibility that the NBS staff observed does

appear to correlate with the degree of circuit "complexity" associated with

the defeat method. However, as can be seen from the circuits shown in

Appendix I, none of the circuits implemented at NBS for the five methods

described above are considered complex. Continuously adding bandpass -filtered

noise at the notch band (Method 1), for example, is a relatively simple

method, but it causes a noticeable audible hiss, particularly during quiet

passages. The circuits for Methods 2 and 3 are of intermediate relative

complexity and produced occasional audible effects. The defeat circuits for

Methods 4 and 5 are relatively the most complex, and these provided relatively

no audible effects in the informal tests.

TABLE 8

Audibility of Defeat Methods

Defeat Method Relative Audibility

1. "Noise" Noticeable

2.

"1.9 kHz" Occasional

3.

"Noise - 2.7 kHz Mod" Occasional

4.

"1.9 kHz - 2.7 kHz Mod Inaudible

5.

"1.9 kHz - 3. 5/4. 3 kHz Mod" Inaudible

71
6. SUMMARY OF RESULTS

6.1 Does the System Achieve Its Purpose?

As stated in section 3.1 under the interpretation of this question, the copy

protection code system, as embodied by the CBS -supplied Encoder and DAT

Recorder/Decoder will achieve its purpose if the Decoder inhibits the

CBS -supplied DAT Recorder/Decoder from recording all or parts of an encoded

signal and does not inhibit the recording of unencoded signals. A signal is

considered encoded if it has passed through the Encoder; otherwise, it is

unencoded.

The evaluation of the two CBS -supplied Encoders has been described in section

2.2 (Physical Measurements) and section 4.2 (Effect on the Performance

Specifications of an Audio System) . The evaluation of the performance of the

Decoder portion of the CBS -supplied DAT Recorder/Decoder has been described in

section 2.4 (Physical Measurements), and section 3. (Does the System Achieve

Its Purpose?). As summarized below, the answer to this question on the basis

of these results is no - the system does not achieve its purpose.

Encoder Performance

The Encoder uses a narrow-band filter to "notch" the music at a nominal center

frequency of 3840 Hz, approximately midway between the highest B-flat and B of

a piano. This encoding notch filter is in excess of 80 dB deep near its

center frequency and has a bandwidth of nominally 220 Hz. The phase

characteristic of the notch filter varies drastically over the bandwidth of

the filter. The Encoder is dynamic, in that it starts and stops notching,

depending upon the signal components near 3840 Hz and near 2715 Hz. The two

Encoders that were supplied by CBS Records differ considerably as regards the

switching in and out of the encoding notch filter. One of the Encoders frequently switches the notch filter out -- this results in there being no encoding much of the time, resulting in many false negatives (see below). The

other encoder leaves the notch filter in much more of the time, potentially

resulting in increased audibility of the removal of part of the input signal.

Decoder Performance

The primary reason that the system does not achieve its purpose has to do with

the performance of the Decoder. The physical measurements made on this

Decoder have shown that it is subject to both inhibiting the recording of

unencoded signals (false positive detection) as well as not inhibiting the

recording of encoded signals (false negative detection)

.

The tests that are described in section 2.4 (under Actual Encoded and

Unencoded Audio Signals input to the Decoder) were certainly definitive in

terms of demonstrating that there are numerous cases where the Decoder failed

to detect legitimate, encoded signals from its own Encoders (false negatives)

and, more importantly, failed to accept unencoded input signals of several

varieties (false positives). Although these tests were not exhaustive,

because of the practically infinite number and kinds of possible source

material, it was observed that out of 54 different selections of unencoded

72
compact discs (CDs) , played on a standard CD player and input to the DAT

Recorder/Decoder, 10 of these CDs had tracks and sectors that consistently

caused a RECORD INHIBIT state (false positive detection) , and shut down the

DAT Recorder. These 54 CDs contained 502 tracks of a wide variety of musical

selections. False positive detection was observed on 16 of these 502 tracks.

Especially significant is the fact that organ and violin recordings

consistently produced false positive detection. Similarly, in tests with

encoded signals, the Decoder did not cause a RECORD INHIBIT (false negative

detection) at least once for 8 out of 18 CDs encoded using Encoder 006 nor for

3 out of 32 CDs encoded using Encoder 012. On a per track basis, the Decoder

made a false negative decision on 139 out of 150 tracks encoded using Encoder

006 (~93 percent false negative detection rate) and on 112 out of 280 tracks

encoded using Encoder 012 (~40 percent false negative detection rate)

.

Because of the scanning operation of the Decoder, described in sections 2.3

and 2.4, there is a variation associated with the false positive detection of

the Decoder. Consequently, as described in section 3.2, the number of false

positives observed during the tests for these occurrences varied. Depending

on the length of the time used for monitoring the input signal, and the number

of repeat monitorings, the total number of false positive detections varied.

For the NBS tests, the percentage of false positive detections ranged from 8 to 86 percent.

The other tests that are described in section 2.4 under Physical Measurements

have also helped to substantiate that the Decoder in the CBS -supplied DAT

Recorder/Decoder does not perform well in discriminating a purposeful encoding

notch in the amplitude spectrum of an input signal. Using a combination

(three -tone) sine wave input signal (with frequency components in the notch

region) , it was determined that the measured Decoder response characteristics

are very dependent on both the absolute and relative amplitudes of the notch

band and the upper and lower sideband levels of the input signal. These

relationships are shown not to require a significant notch type of

characteristic in the notch frequency region of the input signal spectrum in

order to cause a RECORD INHIBIT state under certain conditions. Other input

signal levels at the notch and upper and lower sideband frequencies may

require a very deep notch characteristic in the input signal in order to cause

a RECORD INHIBIT state. This limited discrimination characteristic is further

revealed by the NBS tests using functional triangle, sawtooth (ramp), and

square wave input signal waveforms. It has been determined that there is a range of amplitudes and (fundamental) frequencies of these waveforms which

satisfies the spectral requirements needed to cause the POSSIBILITY and RECORD

INHIBIT states in the CBS -supplied DAT Recorder/Decoder. However, these

amplitude and frequency combinations were not explored exhaustively.

6.2 Does the System Diminish the Quality of the Prerecorded Material?

If the Encoder were to be introduced into the music recording and reproduction

process, the frequency characteristics of the signal coming out of an

amplifier in one's home would be degraded relative to the signal that was originally recorded prior to copy-prevention coding. Specifically, the usual

statements of system performance as being within ±X dB (amplitude) and ±Y deg

(phase) would no longer be met.

73
In order to examine directly the audibility of the effects of the Encoder, two

separate but related series of listening tests were carried out. In each of

these tests, listeners attempted to discriminate between music which was

presented to them "directly" (not encoded) and the same musical selection presented to them "encoded." In the "serial" listening tests, 24 short (of the

order of 5 sec) segments of music were presented sequentially to multiple listeners in a fixed sequence that was established by NBS staff. Pairs

consisting of direct (D) and encoded (E) copies of each segment were presented

to the listeners, with all possible combinations, DD , DE , ED, and EE, included

for each segment in the test. The listeners were asked to identify whether

the copies in each pair presented were the same or different.

In the "parallel" listening tests, ten segments of music, each of about 1 minute in length, were presented to one subject at a time. The selections

were recorded synchronously on three parallel pairs of tracks, with one reference track-pair always being direct material, a second track-pair being

direct or encoded material, and the third track-pair being encoded or direct.

The particular track-pair containing encoded material was assigned randomly.

The subject's task was to attempt to identify, for each musical selection, the

track-pair containing the encoded material.

Generally, the results of the subjective listening tests show that the ability

to hear effects of the Encoder varies substantially among individual subjects

and, especially, among musical selections.

Of the 24 musical selections in the serial listening study, eight of these

have average scores that exceed 50 percent (i.e., pure chance) by a statistically significant margin (p < 0.025). For three of these selections,

the data showed evidence of an audible Encoder effect that was strongly

statistically significant, with the probability of obtaining such high scores

by chance being less than one in a million. Several listeners achieved scores

that exceed chance by a statistically significant margin (p < 0.025). More

significant, statistically, was the fact that 69 of 84 listeners achieved

scores higher than 50 percent on the test, a result that would occur by chance

with a probability of only 3 in a trillion.

In the parallel listening study, three listeners correctly identified the

encoded channel for at least 9 of the 10 selections, a result that is

statistically significant at the 1 percent level. For two selections, the

encoded track-pair was correctly identified often enough to achieve

statistical significance (p < 0.02). Of these selections, one was the same as the highest scoring compact disc selection in the serial listening tests,

while the other did not achieve a statistically significant score in the

serial tests. This difference suggests that a longer-term phenomenon may have

been detected in the parallel study (which used a 29 -second segment) that was not discerned in the serial test (which used a 5-second segment)

.

6.3 Can the System be Bypassed, and If So, How Easily?

Several possible methods by which the Decoder used in the CBS -supplied DAT

Recorder/Decoder can be circumvented or defeated by external signal

74
conditioning methods have been demonstrated. Five such methods are described

in section 5.2 of this report. Electronic circuits for all of these methods

were implemented during the NBS study. Besides the functional description of

these methods, the schematic diagrams of the components used in the circuits

implemented at NBS are provided in Appendix I of this report. There are certainly other possible methods besides the ones described herein for

externally circumventing the Decoder.

As shown in Appendix I, the ease of bypassing can be judged by the relative

simplicity involved in these circuits. The electronic components in all of

these circuits are very basic. Most of these circuits can be implemented with

about a dozen commonly available, off-the-shelf parts. The power supplies

needed to operate these circuits are also available as standard modules. It

is estimated that the parts cost in small quantities for each of the circuits

shown in Appendix I is approximately $100, including the dc power supplies.

The degree of knowledge required to conceive, design, and test the defeat

circuits shown in this report is generally that of a professionally trained

electrical/electronics engineer. However, the degree of knowledge required to

construct the defeat circuits described herein is that of an electronics

technician, familiar with reading circuit schematics.

To be an acceptable defeat method, the effects of the method must also not be

perceptibly audible. The relative audible effects of the five defeat methods

implemented at NBS were compared informally by subjective listening tests made

by NBS staff. Two of the five defeat methods were found to provide

practically no audible effects.

Since the question of whether the system can be defeated is one dealing with

the Decoder portion of a DAT Recorder/Decoder, and external signal

conditioning methods have been found which are practically inaudible, the

answer to this question is yes - the system can be effectively circumvented

with relative simple external electronic circuitry.

75
Acknowledgements

The authors gratefully acknowledge the many contributions made by others in the

preparation of this report. Without their help a document such as this one would not have been possible. It certainly can be said that the results

obtained in this study have been the product of a team effort.

Non-NBS

Messrs. Peter Dare and Gus Skinas , Sony Corporation of America, for the loan of

professional digital audio recording equipment and technical assistance in its

use

.

Mr. Almon H. Clegg, Denon America, Inc. and Mr. T. Shirai , Nippon Columbia Co.,

Ltd. , for providing technical information regarding the Denon DCD-3300 compact

disc player.

Mr. Farrell Becker, Audio Artistry, for consultation regarding the

acoustical treatment of the loudspeaker listening room.

Dr. Floyd E. Toole, National Research Council of Canada, for helpful details

regarding loudspeaker listening evaluations and measurements.

Ms. Lauri Jackson, Hearing Assessment Center, for determining hearing threshold

levels of the subjects in the serial listening study.

Mr. Allen Alexopulos , Hewlett-Packard Co., for provision of the software used in

the preparation of real-time color spectrographs.

NBS

Messrs. Nile M. Oldham, Donald R. Flach, Mark E. Parker, and Robert H. Palm, for

critical calibrations of instrumentation and preparation of figures and graphs.

Mr. Franklin R. Breckenridge , for technical assistance in analysis and testing,

for help in selection of musical passages for the listening tests, and for

assistance in the acoustical treatment of the loudspeaker listening room.

Ms. Susannah Schiller, for help with statistical computing and programming.

Messrs. Raymond J. Mele and F. Keith Mackley of the NEL Graphics group, for

their help in the final layout of the camera-ready copy for this report.

Mmes . Judith L. Barnard, Gail E. Trupo, and Sandra L. Carlsen, for their help in

typing the numerous drafts of this report.

Messrs. J. Franklin Mayo-Wells, George A. Sinnott, and Samuel Kramer for

critical and timely reviews of drafts of this report.

Finally, the authors would especially like to thank John W. Lyons, Director of

the NEL, for his contributions to the writing of this report, as well as his

encouragement and support to the successful completion of this project.

76
APPENDIX A

Congressional Committee Letters
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**CE J. SMITH. FLORJOA

UfO L KERMAN. CALIFORNIA

I0UCHER. VIRGINIA

si o. STAGGERS. JR. WEST VIRGINIA

||RYANT. TEXAS

[All IN L CARDIN. MARYLAND

May 21, 1987

Dr. Ernest Ambler

Director

National Bureau of Standards

Gaithersburg, MD 20899

Dear Dr. Ambler:

Our subcommittees have been considering the many issues

surrounding the advent of digital audio taping. As you may

already know, one of those issues involves a proposal to require

that a "copy-code scanner" be inserted into digital audio tape

recorders, and a corresponding "notch" into prerecorded material,

to prevent the machines from copying. A number of technical

questions about this blocking system have arisen, and the various

interested parties have agreed that the system should be submitted to an impartial entity for testing.

We therefore request that the National Bureau of Standards

perform this test, which would focus on three basic questions.

First, does the copy-code scanner system achieve its purpose to

prevent digital audio tape machines from recording? Second, does

the system diminish the quality of the prerecorded material into

which the notch is inserted? Third, can the system be bypassed,

and if so, how easily?

Counsel to our subcommittees will be available to provide

you with further information/ and to discuss the details of this

test. We also expect the interested parties to submit their

suggestions about how the test can be fairly and expeditiously

conducted

.

HAMILTON FISH. JR. NCW YORK CARLOS J. MOORHEAD. CALIFORNIA HENRY J. HYDE. EJJNCXS OAN LUNGACN. CAUFORNIA

F. JAMES SENS EXRENNER JR. WISCONSIN MU. McCOUUM. RUMBA

t. CLAY SHAW. JR. FLOAWA

GEORGE W. QEKAS. PENNSYLVANIA MICHAB. OKWINC. OHIO WILLIAM C. OANNEMEYER CALIFORNIA PATRICK L SWWDALL GEORGIA HOWARD COSLE. NORTH CAROLINA

D. FRENCH SLAUGHTER JR. VIRGINIA LAMAR S. SMITH TEXAS

31.6. £>ou0e of ftcpretfentattoetf

Committee on tfje Jubiriarp

Maitmuton. 3BC 20515-6216

(EeUpfjone: 202-225-3951

. i «rr 1 ARTHUR P. CHORES. JR

ASSOCIATE COUNSEL

ALAN f COFFEY. JR

A.1
Dr. Ernest Ambler

Page two

May 21, 1987

Please contact- Virginia Sloan, counsel to the House Subcommittee on Courts, Civil Liberties and the Administration of

Justice, at 225-3926 to begin the testing process.

Sincerely

JL

I

DENNIS DE CONCINI

Chairman

House Subcommittee on Courts

Civil Liberties and the

Administration of Justice

Senate Subcommittee on

Patents, Copyrights,

and Trademarks

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CONSUMER PROTECTION. ANO COMPETITIVENESS

June 3

Wit^njrtffn. &C 20515

3, 1987

Dr. Ernest Ambler

Director

National Bureau of Standards

Gaithersburg, Maryland 20899

Dear Dr. Ambler:

The Subconuni ttee on Commerce, Consumer Protection, and

Competitiveness is considering H.R. 1384, legislation to require

a copy-code scanner in digital audio taping devices. At the

Subcommittee's hearing on May 14, there was disagreement as to

the effect, if any, of the copy-code scanner system, developed by

CBS, on the quality of prerecorded music.

We are requesting the National Bureau of Standards to

evaluate the copy-code system. This evaluation should test

whether the system would have any adverse effect on the quality

of prerecorded sound or music. It is important that this

evaluation be performed as expeditiously as- possible since this

legislation is pending before the Subcommittee. We have asked

both the recording industry and members of the electronics

industry opposing this legislation to cooperate with you in this

test. We urge you to expeditiously consult with both sides to

ensure a fair evaluation.

It is our understanding that the House and Senate Judiciary

Committees have requested a similar evaluation. It is our hope

that a single evaluation by the National Bureau of Standards will

satisfy all requests, but we defer to your judgement on that

issue

.

We appreciate your attention to thi3 request and look forward

to the results. Could you please let us know approximately how

long it will take to complete the test?

Thank you for your cooperation.

Sincerely,

Subcommittee on Commerce,

Consumer Protection, and

Competitiveness

Subcommittee on Commerce,

Consumer Protection, and

Competitiveness

cc: Members of the Subcommittee

A.

3
'

.
APPENDIX B

Compact Discs Used in False Positive and False Negative Studies

Table B-l

Disc

Reference

Number

Disc

Identification

Code Composer/Title/Artist/Company

1 CDP 7 46001 2 Pink Floyd/The Dark Side of the Moon/

Capital

2 410987 Beethoven/Symphony No. 9/

Karajan/Berlin Philharmonic/

Deutsche Grammophon

3, 31 CD 80059 Vaughan Williams/Tallis Fantasia/

Slatkin/St. Louis Symphony Orchestra/

Telarc Digital

4 MYK 37261 Debussy/La Mer; Prelude to the Afternoon

of a Faun/Jeux Boulez/

New Philharmonia Orchestra/

CBS

5 410164 Prokofiev/Alexander Nevsky/

Chailly/Cleveland Orchestra/

London/Polygram Records

6 4000452 Vivaldi/Four Seasons/

Standage/Pinnock/English Concert/

Deutsche Grammophon

7 CD 80060 Beethoven/Symphony No. 5/

Ozawa/Boston Symphony Orchestra/

Telarc Digital

8 CK 40158 Judas Priest/Turbo/

Columbia

9 DMP CD 459 Manfredo Fest/Braziliana/

Digital Music Products

10 MK 7194 Wendy Carlos/Switched-On Bach/

CBS Masterworks

11 MCAD 6144 Back To The Future/

MCA Records

12 5E 507 2 The Cars/Candy-0/

Elektra

B.1
Table B-l (con't)

Disc

Reference

Number

Disc

Identification

Code Composer/Title/Artist/Company

13 829 179 2 The Moody Blues/The Other Side of Life/

Polygram Records

14 ARCD 8400 GTR/

Arista

15 A2 90479 Stevie Nicks/Rock A Little/

Modern Records

16 2011-2 Peter Gabriel/Security/

Geffen Records

17 CD 80094 Star Tracks/Kunzel/

Cincinnati Pops Orchestra/

Telarc Digital

18 RCD 14413 Mozart/Symphonies Nos. 40 & 41/

Chicago Symphony Orchestra/Levine/

RCA

19 412 627 2 John Williams/America , The Dream Goes On/

Philips

20 410 116 2 Dvorak/Symphony No. 9/

Chicago Symphony Orchestra/

Sir Georg Solti/

London

21 KEM 01004 Sousa/Famous Marches/

Hunsberge/Eastman Wind Ensemble/

KEM Disc

22 CD 1 CD-I Test Disc/

CBS

23 TCD 06134 Stevie Wonder/In Square Circle/

Tamla/MCA Distributing

24 5004 Equale Brass/Bacchanales/

Nimbus Records

25 MX 42258 Bach/Concertos for Violins & Strings/

Stern, Zukerman, Schneider, Gomberg/

CBS Masterworks

B.2
Table B-l (con't)

Disc

Reference

Number

Disc

Identification

Code Compos er/Title/Artis t/Company

26 MK 37848 Rodrigo/Concierto De Aranjuez/

John Williams/Fremaux/Philharmonia

Orchestra/

CBS Masterworks

27 CD 3902 The Police/Every Breath You Take/

A&M Records

28 CD 80086 Grofe/Grand Canyon Suite and

Gershwin/Catfish Row; Porge & Bess/

Erich Kunz el/Cincinnati Pops Orchestra/

Telarc Digital

29 410 025 2 Gershwin/Rhapsody in Blue and

Bernstein/West Side Story/

Los Angeles Philmarmonic/

Deutsche Grammophon

30 CD 80078 Copland/Appalachian Spring; Rodeo;

Fanfare for the Common Man/

Louis Lane/Atlanta Symphony Orchestra/

Telarc Digital

31 Same as #3

32 WD 1034 Liz Story/Unaccountable Effect/

Windham Hill Records

33 CDC 7 47395 2 A Bach Festival for Brass & Organ/

The Empire Brass and Douglas Major/

Angel/EMI Records

34 411 929 2 Bach, Widor, Albinoni, Franck,

Mendelssohn/Toccata & Fugue Great Organ

Works/London/DECCA Record

35 CD 80115 Orchestral Spectaculars/

Erich Kunzel/Cincinnati Pops Orchestra/

Telarc Digital

36 RCD 1 5303 Schubert/Arpeggione , Die Schone Muller in,

Serenade/

Galway, Moll/

RCA

B.3
Table B-l (con't)

Disc

Reference

Number

Disc

Identification

Code Composer/Title/Artist/Company

37 410 042 2 Chopin/Piano Concerto No. 2/

Davidovich/Marriner/

Philips/Polygram Records

38 CDC 747005 2 Bach/Violin Concertos for Violins and

Strings/

Mutter/Accardo/English Consort/

Angel, EMI Records

39 CK 39936 Journey/Raised on Radio/

Columbia

40 ZK 34224 Kansas/Leftovertime/

Kirshner

41 9 24088 2 Peter Gabriel/So/

Geffen Records

42 810 002 2 Rush/Signals/

Mercury

43 ZK 34929 Kansas/Point of Know Return/

Kirshner

44 9 24098 2 Ric Ocasek/This Side of Paradise/

Geffen Records

45 9 24162 2 Aerosmith/Permanent Vacation/

Geffen Records

46 CK 40092 Barbra Streisand/The Broadway Album/

Barbra Streisand

Columbia

47 400 029 2 Tchaikovsky/Symphony No. 6/

Los Angeles Philharmonic Orchestra/

Deutsche Grammophon

48 400 048 2 10 Beethoven/Violin Concerto/

Kyung Wha Chung/

DECCA/London

49 827 409 2 Pat Metheny/Watercolors/

ECM Records

B.4
Table B-l (con't)

Disc

Reference

Number

Disc

Identification

Code Composer/Title/Artist/Company

50 PCDI 5459 Miller/The Unforgettable/

Glenn Miller and His Orchestra/

RCA

51 MCAD 5570 Black Friday/

MCA Records

52 CAL 9926 Olivier Messiaen/Les Corps Glorieux,

L' Aslension/

Calliope

53 MK 42333 Wendy Carlos/Secrets of Synthesis/

CBS Inc

.

54 MK 42014 Beethoven/Symphony No. 9/

Bruno Walter/Columbia Symphony Orchestra/

CBS

55 EK 40600 Michael Jackson/Bad/

Epic

56 CDC 7 47189 2 Beethoven/Symphony No . 9/

Klemperer/Philharmonic Orchestra/

Angel, EMI Records

B.5
.

.

'
APPENDIX C

Contractor's Report

Appendix C: Contractor's Report -- Subjective Listening Tests

This research was supported by Purchase Order 43NANB800479 between the

National Bureau of Standards (NBS) and the University of Michigan, Principal

Investigator, Irwin Pollack. The contract monitor for NBS was Daniel R.

Flynn.

The opinions expressed in this Appendix are those of the contractor and do not

necessarily represent those of the National Bureau of Standards.

Subjective listening tests were carried out to examine the extent to which

listeners could discriminate between unencoded music and music that had been

encoded using the CBS Copy-Prevention Encoder. Listening materials consisted

of excerpts of music from commercially-available stereo compact discs plus,

for the serial tests described below, a few samples produced by a music

synthesizer. The music was either passed through the Encoder, or by-passed

the Encoder.

For the "serial tests," listeners attempted to identify whether repeated pairs

of sequentially-presented musical passages were selected from the same source

(either encoded or unencoded) , or from different sources (one encoded, one unencoded) . Large differences in average discrimination performance were

associated with different audio materials in the serial listening tests.

Based upon the combination of earphone and loudspeaker testing, 5 of 20

compact disc selections and 2 of 4 synthesizer examples were discriminated at

a level significantly higher than chance. The results suggest that the

expected level of discrimination is highly sensitive to the specific materials

examined. Differences among listeners tended to be smaller than differences

among materials

.

In the "parallel tests," listeners could switch at will between parallel pairs

of tracks of encoded and unencoded selections in an attempt to determine which

material differed from the unencoded material on a pair of reference tracks.

Substantially better performance was achieved in the parallel tests than in

the serial tests. Four of 15 listeners achieved discrimination levels

significantly higher than expected by chance (one listener achieving a perfect

score of 10 of 10 selections) ; and 4 of 10 selections yielded discrimination

levels significantly higher than expected by chance.

Introduction

The Congress of the United States asked the National Bureau of Standards (NBS)

to examine several aspects of the copy-prevention system developed by CBS

Records . Among the questions to be addressed by NBS was whether the quality

of recorded music is diminished by the use of the copy-prevention system. The

portion of this system that potentially could affect musical quality is the

Encoder; the Decoder is not involved in this aspect of the NBS evaluation of

the Copy-Prevention System.

C.1
In preliminary discussions, the Principal Investigator and staff members of

the NBS Acoustic Measurements Group and the NBS Statistical Engineering

Division discussed the interpretation of the Congressional directive. An

obvious test was whether or not listeners could tell when the Encoder was

operating. We call this a "chocolate-versus -vanilla" test. A possible

approach would have been to present musical selections with and without the

Encoder and to ask listeners to identify which one was produced by the

Encoder. Several NBS staff members argued strongly against that approach.

Among their reasons was that it would not be possible to give sufficient

training to the listeners to enable them to develop a proper 'taste' for

chocolate and vanilla, i.e., to appreciate encoded versus unencoded materials.

An alternative approach to the question asked by Congress was to take the

position that aural discrimination of any (non- trivial) recognizable

departure, due to the encoding process, from an original recording represents

a degradation of the quality of that recording. (An example of a trivial

departure from the original performance is a change in overall level which can

easily be compensated.) This interpretation was taken, so that in this study,

instead of asking listeners to identify the presence of the Encoder, they were

asked whether musical selections were reproduced from the same source or from

different sources, without being asked to identify the nature of the source. This decision is crucial to the tests which follow.

Assumptions

1. A broadly-based scientific examination of the Encoder would

involve a detailed examination of the effects of selected

variables on the recognition of the Encoder.

2. The discrimination of a non-trivial modification of a recorded

performance constitutes a perceived degradation in quality.

3. In the tradeoff between the length of the testing materials and the

variety of materials, it was decided to examine a wide variety of

recorded materials and relatively short musical samples.

4. In the tradeoff between professional experience and "young" ears (which

would be less likely to suffer from hearing loss) , the decision was made

not to limit the study to young subjects, but to include experienced

listeners, especially those with a background in music and in sound

recording.

5. With regard to the question of whether (1) to utilize a random selection

of music and subjects that would, in some statistical sense, be

"representative" of larger populations or (2) to make "critical"

selections of music (where the effects of the Encoder might be more

audible) and of listeners (who would be more likely to hear such

effects) , the decision was made to pick critical selections and

subj ects

.

C.2
6. It was decided not to give the subjects specific listening training as to the effects of the Encoder, prior to testing.

7. Although discrimination of the effects of the Encoder might have been

more likely if the subjects were provided with visual displays of the

results of physical measurements, it was determined that all tests would

be confined to auditory listening.

8. To ensure impartiality of testing administration, all tests would be run

' double -blind' -- without the test administrator having knowledge of the

key.

9. A difficult decision was made with respect to the loudness listening

level for the serial tests, for which the subjects were presented with a constant listening level rather than being permitted individual

adjustment to a preferred listening level.

Discussion of Assumptions , Objections, Tradeoffs

The following items correspond to the numbers of the assumptions given above:

1. Under the time and financial constraints of the Project, a broadly-based

basic scientific examination of the Encoder was deemed impossible.

2. Assumption 2 was crucial to the research. Insufficient time was

available to train listeners to hear any effects that the Encoder might

have on a wide variety of types of music. In other words, a chocolateversus -vanilla test would be impossible unless listeners could be

provided with a fuller understanding of the 'tastes' of chocolate

versus vanilla.

3. Since certain classes of music reproductions might be more sensitive

than others to the action of the Encoder, the decision was made to

utilize a range of musical selections so as to ensure adequate coverage.

Either of two extreme cases might represent an appropriate listening

test. One could present isolated individual notes in which Encoder

action was identified -- this would completely deprive the listener of

an adequate musical setting or context. One could present long musical

passages in which an isolated Encoder action occurred -- this would

place an intolerable burden on the listener's auditory memory, asking

him in effect, to "locate a needle in a giant haystack." The compromise

that was made for the serial study was to ask the listener to "locate a needle in a small haystack" by presenting selections that were nominally

5 seconds in length.

4. The original plan included using music performance and music recording

majors from local universities as listeners. The tests were delayed so that the projected starting time fell in the middle of the academic

Christmas vacation. Meanwhile, the response from the audio engineering

C.3
community was so enthusiastic that the decision was made not to return

to the university students. It was obviously important to take

advantage of the critical listening skills of experienced audio

engineers and musicians. Even more importantly, many of these

experienced professionals have devoted their professional careers toward

the improvement of music reproduction. It would not be possible to

duplicate the motivation and intrinsic interest of this group by means

of financial inducements upon a less concerned population.

Because of reports that audio engineers may suffer hearing loss in the

active frequency region of the Encoder (4 kHz) , an audiometric

examination was carried out on each of the subjects in the serial

listening tests.

5. The NBS statistician on the study pointed out that the project lacked

the time and financial resources to achieve a "representative 1

' sampling

of recorded music and of listeners. Since it made little sense to test

the performance of the Encoder when the Encoder was not engaged or to

test the Encoder on unmotivated or unskilled listeners, it was decided

to opt in the direction of a "critical" sampling of music and of

listeners

.

6. Specific training was not given for several reasons. Within the time

available for testing volunteer listeners, additional training time

would have reduced the range of examples for testing. There was concern

that to give extensive training as to the effects of the Encoder might

be construed as biasing the study. And, finally, there was reluctance

to serve up, as practice examples, the most easily detected selections.

7. In examining compact discs for Encoder action, NBS staff utilized

physical measurements, computer analyses, and visual displays to

screen musical candidates for later aural study and to identify

specific effects of the Encoder on particular musical passages.

It was found that the Encoder was often most active under

conditions where aural identification was very difficult, e.g.,

when an entire orchestra was producing music. Since music

listeners to not normally have spectral analysers available, since

it would be difficult do teach listeners to understand the

implications of spectral displays, and since to make sophisticated

instrumentation available to the listeners might be construed as biasing the study in the direction of easy detection of any

audible effects that the Encoder might have, it was decided that

the listeners would have to make do with their ears.

8. Because of the sensitive nature of the tests, test impartiality was considered to be a primary requirement. The serial tests were run by

the Principal Investigator, who did not know the answers. After the

parallel- test subjects were trained in the use of the multi-track tape

recorder, they were left alone to do their listening.

9. A single listening level was selected for the serial tests to ensure

uniform testing and to avoid unacceptably loud levels.

C.4
Limitations of the Present Study

The same-different sequential procedure is not without difficulty. A heavy

burden is placed on auditory memory. A listener may detect a large difference

with high confidence. However, when no difference is detected, the listener

is unsure whether the items are actually the same or whether there exists a difference which is indistinguishable.

The testing program did not allow for extended specialized training. There is no doubt that, with sufficient training, practice, and motivation, incredible

individual feats may be attained. Indeed, recognized critics are often

respected for the development of highly specific talents. For the purposes of

this study, however, it is surmised that moderate amounts of additional

training probably would have produced only modest improvements in performance.

Experimental Approach: Serial Tests

Listening Materials

The selection of the test materials for the listening tests is described in

section 4.4. Specific compact discs were recommended by the Home Recording

Rights Coalition and the Recording Industry Association of America. With the

aid of spectrum analyzers, computer monitoring, visual displays, and a readout of the instanteous behavior of the Encoder, significant periods of Encoder

activity could be located. Aided by this information, NBS staff then selected

examples where significant aural changes appeared to take place. In this way,

the sophisticated instrumentation served to screen potential candidates for

testing, but the final selection was based upon aural identification.

Listeners

Listeners were recruited among members of the Washington-Baltimore chapter of

the Audio , Engineering Society, among local radio stations and recording

studios, and among National Bureau of Standards employees who were

knowledgable about music.

Recording Procedure

The recording procedures for the test materials are detailed in Appendix F. In essence, the test materials were derived as follows: the first generation

consisted of a 5-10 second sample of music with manual fade-in and fade-out;

the digital sample was converted to an analog signal; that signal was both

passed through the Encoder and by-passed the Encoder; and a second generation

digital sample was obtained by reconversion of the analog signals. This

procedure ensured that both encoded and unencoded samples were treated

identically except for passage through the Encoder.

Audio Equipment and Calibration

The earphones and loudspeakers used for this study, along with their

calibration, are detailed in Appendix G. In essence, high quality

professional equipment was employed throughout the tests. Of special concern

C.5
was the selection of loudspeakers. NBS staff selected a two-way speaker

system with a crossover frequency below 2000 Hz, rather than a three-way

speaker system with a crossover frequency in the active region of the Encoder.

In this way, the phase distortion of the loudspeaker crossover network did not

interact with that of the Encoder.

Test Procedure

Listeners heard sequences of six presentations of the same musical sample. A

trial consisted of a given A-B pair repeated three times, e.g., A^B^ A2®2>

A3 B 3

. All of the A samples and all of the B samples were the same within a given trial. Across different trials, the A or B samples were equally-often

encoded (E) or unencoded (here called direct (D)). All four possibilities

were available: EE, DD, ED, and DE. The correct response was 'same' to the EE

and DD pairings and 'different' to the ED and DE pairings. Note that the

listener did not have to identify whether a sample was or was not encoded.

The intra-pair A^B^ duration was one second; the inter-pair B^A2 duration was three seconds. Each test sequence was preceded by the announcement "Here is

test pair x;" and was terminated by the announcement "Please encircle your

response in slot x„" For each musical selection, the entire sequence of

announcements, music, and pauses averaged about 56 seconds in duration. Four

practice examples were employed to introduce the test procedures. Discussion

was held after the four examples to ensure understanding of the test

procedure

.

Experimental Results: Serial Tests

Qualitative Observations

On the basis of articles in leading semi- technical audio magazines, many of

the listeners anticipated that gross and blatant changes would be introduced

by the Encoder. As dedicated professionals in the recording and musical arts,

they were obviously concerned. They were eager to hear music through the

Copy-Code system, freely volunteering time from their busy professional

careers in order to participate in the tests. Hardly any degree of financial

reward with disinterested observers could have yielded the equivalent

motivation. This point is significant because negative findings can easily be

obtained from the best-intentioned experimental design, if the listeners are poorly motivated. Regardless of the strong attitudes of the listeners, the

double-blind procedure and experimental design served to safeguard the study.

At the conclusion of the listening session, many of the listeners remarked

that the effect of the Encoder was substantially more subtle than they had

expected.

Qualitative observations such as those cited above, of course, are not

adequate substitutes for more controlled, quantitative test scores.

Nevertheless, they suggest that the effect of the Encoder can be subtle and

that quantitative tests might not demonstrate high discrimination scores.

C06
Quantitative Observations: Musical Selections

Substantial differences in discrimination performance were associated with

different musical selections. Table C-l presents the average scores over 61

listeners employing earphones for the entire set of 24 selections (20 music

and four synthesizer examples) . The range of correct identification scores is

41.4% to 91.8%. Marked with plus (+) signs are four of 24 examples yielding

an average performance which was statistically significantly better than an expected chance performance of 50%. (Statistical significance is based upon

the binomial probability distribution at the 95% confidence level; see section

4.5). Marked with a minus sign (-) is one of twenty-four examples that

yielded an average performance significantly poorer than chance. Similarly,

table C-2 presents the average scores over twenty- three listeners employing

loudspeakers for the entire set of twenty-four selections. The range of

correct identification scores is 40.2% to 87.0% with 2 of twenty-four

selections yielding scores significantly greater than expected by chance.

Figure C-l presents the cumulative distribution of average discrimination

scores over the twenty-four selections for earphone (E) and loudspeaker (S)

listeners

.

In the analysis in section 4.5, earphone and loudspeaker listeners (including

3 listeners who used earphones part of the time and loudspeakers part of the

time) were combined into a single group. As a result of the greater

statistical sensitivity of the combined group, eight selections (six compact

disc and 2 synthesizer) exceeded the expected chance level by a statistically

significant margin, and one selection was significantly lower than expected by

chance. The correlation over musical selections between scores for earphones

listening and for loudspeaker listening was moderate (r = 0.80), as shown in

figure C-2. The correlation is heavily weighted by one extreme case.

The synthesized B 7 -F7 chord yielded the highest discrimination score. Application of a statistical test for extreme cases (the Dixon test) shows

that specific example to be an atypical, or extreme, case from other examples.

This result does not mean that we can reject that example, but, rather, that

the example appears not to have been drawn from the same normal distribution

as the other scores. In partial summary, test discrimination scores for the

Encoder are strongly dependent upon the specific selection tested.

On average, the same-different test scores across selections yielded

discrimination significantly better than chance. Specifically, of 61 earphone

listeners, 45 scored better than 50 percent correct, a result expected by

chance about 1 in 10,000 times.

Among the 24 selections, the median proportion for correct responses was 0.522

for loudspeaker listening and 0.533 for earphone listening. Whereas the

median scores are only marginally higher than chance, the scores for

individual selections may demonstrate substantial discrimination.

C.7
TABLE C-l

SCORE SUMMARY BY SELECTION FOR EARPHONE

Selection Number of

Name Responses

Debussy 244

Barber 244

Beethoven chord 244

Beethoven chorus ( -

)

244

Bizet 244

Wilson 244

Streisand 244

Beethoven Triangle 244

McGovern 244

Weather 244

Copland 244

Bernstein 244

Pink Floyd 244

SI ( + ) 244

Prokofiev ( + ) 244

S2 244

Metheny 244

S3 244

S4 244

Messiaen 244

Bach 244

Respighi (+) 244

Hande

1

244

Rave 1 ( +

)

244

C.8

LISTENERS

Percent

Correct

54.1

50.4

54.5

41.4

55.7

52.9

50.8

54.9

53.3

43.9

54.5

50.4

51.6

91.8

73.4

54.1

52.5

53.3

49.2

54.9

52.5

61.1

49.2

57.0
TABLE C-2

SCORE SUMMARY BY SELECTION FOR LOUDSPEAKER LISTENERS

Selection Number of Percent

Name Responses Correct

Debussy 92 49.0

Barber 92 49.0

Beethoven chord 92 54.3

Beethoven chorus 92 47.8

Bizet 92 52.2

Wilson 92 43.5

Streisand 92 41.3

Beethoven Triangle 92 50.0

McGovern 92 51.1

Weather 92 52.2

Copland 92 54.3

Bernstein 92 44.6

Pink Floyd 92 40.2

SI (+) 92 87.0

Prokofiev 92 60.0

S2 92 55.4

Metheny 92 56.5

S3 92 60.0

S4 92 52.2

Messiaen 92 60.0

Bach 92 50.0

Respighi (+) 92 68.5

Handel 92 51.1

Ravel 92 52.2

C.9
Figure C-l Distribution of same-different test scores over 24 selections,

including four synthesizer selections. Results for earphone and

loudspeaker listening shown separately.

C.10
Speaker

scores

1.0

0 . 9

-

0 . 8

-

0. 7

-

Pearson correlation coefficient = 0.8004

0. 6

-

0. 5

-

*X

* , XX

X*V

0.4

0.4 0.5 0.6 0.7 0.8

Earphone scores

0.9 1.0

Figure C-2 Comparison between same-different test scores over 24 selections,

including four synthesizer selections plotted as stars. Results

for earphone and loudspeaker listening are plotted on separate

axes

.

C.11
Quantitative Observations: Listeners

As indicated in the last column of tables C-3 and C-4, 10 of the 61 earphone

listeners, and 3 of the 23 loudspeaker listeners, scored at a discrimination

level significantly higher than expected by chance in the serial test. The

highest discrimination score achieved was 63.5% for loudspeaker listening.

Figure C-3 presents the cumulative rank ordering of scores associated with the

individual listeners. Figures C-l and C-3 are drawn to the same scale.

Comparison between the figures shows the wider range of scores across

different selections than across different listeners.

Interaction of Selections and Listeners.

There was concern that statistical pooling of listeners and/or selections

would mask inherently good performance. A median empirical split was made

between listeners and selections, where each was divided into two groups.

Table C-5 presents mean scores with the resulting 2x2 median split for the

20 musical examples and for the earphone listeners. There was little

difference in scores between the better and poorer listeners for the more

difficult selections (47.9% and 51.6%). Both groups scored near chance.

Among the less difficult selections, however, there was a larger difference in

average performance (52.6% and 61.9%). The individual entries of table C-5

can be compared against an independence model, based simply upon the marginal

sums of the two groups of listeners and of the two groups of selections. That

model predicts the separate tabular enteries of table C-5 within 1.1 percent.

Quantitative Observations: Additional Notes

Each listener was asked to supply his or her age and years of experience, to

report any incidence of tinnitus, and to have their audiogram taken.

(Biographical information was incomplete for some listeners.) Also,

correlations were run across test items to search for fatigue and learning

effects. The results of a step-wise multiple regression analysis showed nearzero correlations between scores against age, years of experience, incidence

of tinnitus, and hearing loss. (A similar analysis of factors against hearing

loss will be reported separately by the Principal Investigator.) The nearzero correlations can be interpreted to suggest that, under the conditions of

testing employed, we would not expect large differences among other groups of

listeners. In addition, near-zero correlation with successive test items

suggests that learning and/or fatigue effects either balanced out or were not

strong.

Experimental Approach: Parallel Tests

Introduction

In the same-different serial tests of the previous sections, the listeners

heard nominally 5-second musical samples. It might be argued that a skilled

listener would better be able to identify the effects of the Encoder by

exploring longer musical selections for critical instances of his or her own

choosing. To a recording engineer, the preferred procedure is a 'hands-on'

C.12
12

34

5

6

7

8

9

10

11

12

13

14

15

16

17

18

19

20

21

22

23

24

25

26

27

28

29

30

31

32

33

34

35

36

37

38

39

40

41

42

43

TABLE C-3

SCORE SUMMARY BY SUBJECT FOR EARPHONE LISTENERS

ALL 24 SELECTIONS (96 ITEMS) USED

Station Age

ExperienceHearing

Loss

El 27 8 10

E2 42 15 15

E3 59 35 40

E4 32 12 3 E5 39 29 3

El 69 38 48

E2 30 10 8

E3 41 10 30

E4 33 8 3

E5 38 15 25

El 31 20 18

E2 25 5 18

E3 52 30 10

E4 43 30 25

E5 38 6 13

El 37 5 5 E2 38 6 13

E3 28 . 5

El 27 7 5

E2 51 . 8

E3 34 12 5

El 26 7 28

E2 23 4 10

E3 24 4 10

E4 23 3 13

El 35 13 8

E2 50 30 27

E3 22 5

E4 26 3 5

El 42 12 10

E2 39 20 15

E3 41 25 45

E4 26 5 10

E5 24 4 13

El 36 5 5 E2 47 15 . E3 35 15 8

E4 36 18 3

E5 40 # 5

El 48 10 8 E2 37 15 15

E3 23 3 3

E4 25 6 15

Number of Percent

Tinnitus Subj ect Responses Correct*

Y 1 96 60. 4(+)

N 2 96 57.,3

N 3 96 55.,2

Y 4 96 50.,0

N 5 96 58.,3

N 6 96 54.,2

N 7 96 53..1 Y 8 96 59. 4(+)

N 9 96 55.,2

N 10 96 55..2 N 11 96 52.,1

N 12 96 53..1 Y 13 96 57.,3

N 14 96 51..0

N 15 - 96 59,,4

N 16 96 51..0

Y 17 96 55..2

N 18 96 56,.3 N 19 96 47,.9

Y 20 96 59,.4

N 21 96 58 .3

Y 22 96 52,.1 N 23 96 61 • 5 (+)

N 24 96 58,• 3 (+)

Y 25 96 50 .0

N 26 96 49 .0

Y 27 96 61 5(+)

N 28 96 51 .0

N 29 96 53 .1

N 30 96 56 .3

N 31 96 52 .1 N 32 96 51 .0

Y 33 96 61 • 5 (+)

N 34 96 51 .0

N 35 96 53,.1 Y 36 96 50 .0

N 37 96 56,.2

N 38 96 54,.2

N 39 96 55,.2

N 40 96 58, 3 (+)

Y 41 96 53,.1 N 42 96 56,.2

N 43 96 54,.2

C.13
[TABLE C-3, cont'd]

44 E5 49 20 10 N 44 96 52.1

45 El 36 4 8 N 45 96 46.9

46 E2 45 6 15 N 46 96 59 . 4(+)

47 E3 65 25 23 N 47 96 51.0

48 E2 62 35 13 N 50 96 50.0

49 E3 46 27 15 N 51 96 58.3

50 E3 26 7 25 Y 55 96 55.2

51 El 36 14 8 Y 57 96 63 . 5 (+)

52 E2 34 7 15 Y 58 96 55.2

53 E3 43 7 30 N 59 96 54.2

54 El 48 11 13 N 60 96 58.3

55 E2 40 1 13 N 61 96 53.1

56 E3 24 1 3 N 62 96 49.0

57 E4 25 3 3 N 63 96 56.3

58 El 38 11 5 N 64 96 60 . 4(+)

59 E2 29 16 5 Y 65 96 53.1

60 E3 26 5 5 Y 66 96 56.3

61 E4 24 6 5 N 67 96 57.3

^Scores significantly higher than chance (p < 0.025) are marked *(+)."

C.14
TABLE C-4

SCORE SUMMARY BY SUBJECT FOR LOUDSPEAKER LISTENERS

ALL 24 SELECTIONS (96 ITEMS) USED

Row

Index Station Age

ExperienceHearing

Loss Tinnitus Subj ect

Number of

Responses

Percent

Correct

1 SI 36 15 10 N 80 96 50.0

2 S2 24 4 3 N 81 96 52.1

3 SI 39 22 30 N 82 96 58.3

4 S2 65 45 38 N 83 96 57.3

5 SI 53 30 10 N 84 96 46.9

6 S2 28 14 O N 85 96 61.5 (+)

7 SI 66 45 45 Y 86 96 61.5 (+)

8 S2 . . .

' 87 96 55.2

9 SI 27 2 5 N 88 96 51.0

10 S2 32 10 60 Y 89 96 52.1

11 SI 38 15 20 N 90 96 55.2

12 S2 33 12 5 N 91 96 52.1

13 SI 44 20 3 N 92 96 57.3

14 S2 55 12 33 Y 93 96 46.9

15 SI 25 12 3 Y 94 96 54.2

16 S2 29 12 10 N 95 96 52.1

17 SI 33 12 15 N 96 96 52.1

18 S2 23 7 5 N 97 96 50.0

19 SI 49 34 8 N 99 96 49.0

20 S2 28 7 8 N 100 96 45.8

21 S2 37 15 13 Y 101 96 61.5 (+)

22 SI 55 25 10 Y 103 96 56.3

23 S2 74 40 53 Y 104 96 50.0

^Scores significantly higher than chance (p < 0.025) are marked "(+)."

TABLE C-5

MEDIAN SPLIT ANALYSIS OF AVERAGE SCORES

ACROSS EARPHONE LISTENERS (n=61) AND MUSICAL SELECTIONS (n=20)

.

Listeners

Selections

<Median >Median

(n=10) (n=10)

<Median 47.9% 52.6%

>Median 51.6% 61.9%

C.15
x 100

ii V

CD

_c

o

o

c

o

co

* o

JQ

ZJ

CO

c

CD

o

0

CL

80 -

60 -

40 -

20 -

0

0.4

E = Earphone

S = Speaker listeners

0.6 0.7 0.8

Score (X)

0.9 1.0

Figure C-3 Distribution of same-different test scores over listeners for the

24 selections. Results for earphone and loudspeaker listening

shown separately.

C.16
approach, providing unrestricted access to the music. The tests of this

section, called parallel tests, provided much less restricted access than in

the case of the serial tests. These parallel tests also exploited differences

in quality between the two stereo channels. The Encoder's action is based

upon the average of signals in the two stereo channels. In some instances,

because of the differences between the two stereo channels, the Encoder will

operate more effectively upon one of the two stereo channels than the other.

While such cues were also available in the serial tests, they were probably

less effective because of the shorter duration of the signals.

Recording Procedure

A multi-track digital tape recorder was used, with ten selections of 29 to 75

second duration with three parallel pairs of stereo tracks: an unencoded

stereo pair of reference channel tracks and, in a scrambled order, one pair of

unencoded test tracks and one encoded pair of test tracks. The unencoded

reference track-pair was identified in advance to the observer. The observer

did not know which of the pairs of test track-pairs was encoded. The

selections encompassed the shorter 5-10 second selections of the serial tests.

The musical selections are listed in Appendix E.

Test Procedure

The task of the listener was to select which one of the two pairs of test

tracks contained encoded material, i.e., which differed from the reference

unencoded track-pair. The listener could switch at will between the reference

track-pair and the two test track-pairs. Earphone listening was employed for

all tests. Only one track-pair was audible at any instant. Time tabs were

available for repeated presentation of portions which appeared to be

especially relevant to the listener. The listener could adjust the overall

signal level to a preferred level.

The listeners were generally recording engineers, audio equipment designers,

or professional musicians.

Qualitative Observations

The listeners were encouraged to transcribe the basis for each of their

answers into a tape recorder. Some of their remarks were detailed, pointing

to the exact tape location of specific events. It is interesting to speculate

whether the cumulative experience of providing the remarks of earlier expert

listeners to later listeners would have provided an accurate consensus.

Some of the remarks reflected extraordinary insight into the operation of the

Encoder. For example, "It sounded to me that something of lower frequency was causing the filter to go in and out on an amplitude -dependent basis." Or "I

had assumed I would hear a simple timbral shift, if anything, and that the

timbral shift would be comparatively minor. Instead, in some instances, it

seems to me that the actual impact is quite pronounced and because it's

variable, it's extremely troubling." Some of the remarks also reflected

differences in stereo location.

C.17
Because some of the listeners scored significantly poorer than expected by

chance, the listeners were interviewed by telephone (at a later date) about

their understanding of the instructions. All listeners correctly responded

that their task was to identify the encoded selection, or the selection that

differed from the unencoded reference.

Quantitative Observations

The number of correct selections, out of 10, is presented for each listener

and for each selection in table C-6. By chance, a score of 10 correct of 10

selections would be expected 1 out of a 1000 times, a score of 9 or more

correct would be expected 1 out of a hundred, and a score of 8 or more out of

10 would be expected somewhat more than 5 percent of the time (p = 0.055).

Employing this latter criterion, one can say that 4 of the 15 listeners scored

significantly higher, and 1 of the listeners scored significantly lower, than

expected by chance. The average score, 62.3 percent, is substantially higher

than that achieved with the main body of serial listening tests.

Similarly, by chance, a score of 12 or more correct of 15 listeners would be

expected by chance about 2 times in a 100, and 11 or more correct of 15 would

be expected about 6 times in a 100 (p = 0.059). Employing this critereon, we can say that 4 of the 10 selections scored significantly higher than chance

response. No selection was perfectly discriminated.

Interpretation of Results

Within a wide body of results, protagonists can pick and choose selected

features which support their previously- announced positions. Protagonists can

also advocate test procedures which might support their positions.

Before commenting upon the results, let us consider what constitutes a behaviorally relevant test. For example, equipping each observer with a spectrum analyzer and visual display might have resulted in perfect

discrimination. Most neutral observers, however, might be expected to judge

such tests as behaviorally irrelevant to the aural detection of the effects of

the Encoder.

Along the same lines, a narrow bandpass filter could have screened away all

frequencies outside of the active region of the Encoder. With such a technique, the action of the Encoder might have been easily detected by aural

means alone. Most neutral observers, however, might be expected to judge such

test to be irrelevant to the aural detection of the Encoder for wide-band

music in a normal listening situation.

A test also might have been devised to take special advantage of the human

binaural system. To one ear would be fed the left unencoded channel of a musical selection. To the right ear would be fed either the right unencoded

channel of that selection or the right encoded channel. Alternatively, to the

right ear, we would feed either the left encoded channel of that selection or the left unencoded channel. Had the binaural test display yielded

substantially higher scores, would this test have qualified as a reasonable

test of the Encoder? The neutral observer might have somewhat more trouble

C.18
Table C-6 Identification of encoded channel in the parallel- test format

Subj ect

Fractio:

Selection A B C D E F G H I J K L M N 0 Correct

1 . Bizet C • C C C w C C w w w w C C w 8/14

2. Streisand w C C C w C C w C C C C w C C 11/15

3 . Beethoven Triangle C • C C C w C w C C w C w c C 10/14

4. Copland w w C C C w C w C w w C w w C 7/15

5 . Metheny C C C C C C C C C C w w C w C 12/15

6. Prokofiev C w w C c C C C C C w C C c C 12/15

7. Bernstein c w C C w C C w w C C c C c c 11/15

8. Messiaen c • C C w C w w C w w w w c c 7/14

9 . Bach w • C C C w C w c w w C C w c 8/14

10. Respighi w w w C C w C w w w w w w C c 5/15

Total Correct 6 2 8 10 7 5 9 3 7 5 2 6 5 7 9

C = Correct

w = Wrong

• = No Response

Average percent correct: 100 x (91/146) = 62.3

C.19
with rejecting this wide-band aural test procedure than the aforementioned

test procedures.

Still along the same lines, the parallel 'hands-on' test provided more degrees

of listening freedom than the serial same -different tests. Despite the gross

differences between the two test procedures, the average test scores were not

radically different from each other.

Even within the recognized limitations of the present tests, one would hope

that neutral observers would judge the present tests as behaviorally relevant

to the aural detection of the Encoder.

Where are we? I reject the extreme position that the Encoder's action is so evident and pervasive that it will be immediately recognized by

unsophisticated listeners. However one feels about the principle of the

Encoder, one must concede that its effect is extremely subtle. I also reject

the extreme position that the Encoder's action is so benevolent that it cannot

be detected. Significant discrimination was achieved in the serial tests with

several selections. Even the scores of the best discriminated example,

however, deviated substantially from perfect detection. The latter point is

important because one claim made for employing the Encoder is that it can be

by-passed in the recording studio when it "obviously" intrudes upon the music.

The results may be interpreted to follow the intermediate position of a "good

news -- bad news" scenario. On the one hand, the average level of

discrimination, especially in the serial test procedure, was only marginally

higher than chance. On the other hand, averages can be misleading.

Individual listeners and individual selections, especially in the parallel

test format, yielded substantial discrimination of the Encoder's activity.

Because of individual differences among listeners, among musical selections,

and among test procedures, and because improved encoding procedures might

ultimately be developed, I would strongly urge that original master recordings

be preserved in a unencoded mode.

Credits

The listeners in these tests must be singled out for special attention. They

were unpaid volunteers who took time from their busy careers to participate in

the tests. Their enthusiastic participation is exemplified by their complete

turnout on a Saturday in the midst of a Washington snow storm which had closed

down all government offices for a three-day weekend. It was a singular and

rewarding honor and privilege to work with such dedicated professionals.

Attachments

The attachments on the next two pages are the subject answer sheets for the

serial and parallel listening studies, respectively.

C.20
Name:

NBS COPY-CODE LISTENING PROJECT

Sequential Test

Date:

Listening Station: : Time:

S = Same D = Different

1 . S D 25. S D 49. S D 73. S D

2. S D 26. S D 50. S D 74. S D

3. S D 27. S D 51. S D 75. S D

4. S D 28. S D 52. S D 76. S D

5. S D 29. S D 53. S D 77. S D

6. S D 30. S D 54. S D 78. S D

7. S D 31. S D 55. S D 79. S D

8. S D 32. S D 56. S D CO o S D

9. S D 33. S D 57. S D 81. S D

10. s D 34. S D 58. S D 82. S D

11. s D 35. S D 59. S D 83. S D

12. s D 36. S D 60. S D 84. S D

13. s D 37. S D 61. S D 85. S D

14. s D 38. S D 62. S D 86. s D

15. s D 39. S D 63. S D 87. s D

16. s D 40. s D 64. S D 88. s D

17. s D 41. s D 65. S D 89. s D

18. s D 42. s D 66. S D 90. s D

19. s D 43. s D 67. S D 91. s D

CM o s D 44. s D 68. S D 92. s D

21. s D 45. s D 69. S D 93. s D

22. s D 46. s D 70. S D 94. s D

23. s D 47. s D 71. s D 95. s D

24. s D 48. s D 72. s D 96. s D

C.21
NBS COPY-CODE LISTENING PROJECT

PARALLEL TESTS

Name: Date:

Listening Station: Time:

•

ENCODED MATERIAL

1. A B

2. A B

3. A B

4. A B

5. A B

6. A B

7. A B

8. A B

9. A B

10. A B

C.22
APPENDIX D

Sponsor's Lists of Recorded Material

APPENDIX D: SPONSORS' LISTS OF RECORDED MATERIAL

The National Bureau of Standards invited the two sponsors to submit suggestions for recorded material to be included in the listening tests. These

tests are reproduced in this appendix.

Home Recording Rights Coalition

The first 32 selections of those below were sent to NBS on October 6, 1987.

Many of these selections were reviewed as to their suitability for inclusion

in the listening tests. The remaining selections in the HRRC list were sent

to NBS on December 10 and 11, 1987. Although these latter selections were re- ceived too late for consideration for the listening studies, a listing of them

is enclosed here for completeness.

The following text is quoted from the October 6 transmittal letter from HRRC

to NBS

:

"Attached is a list of selections from 32 compact discs which, based on screening done for HRRC, we have reason to believe illustrate the distortive effects of the CBS "copy-code" system.

"Our list of suggestions is not, by any stretch of the imagination, ex- haustive. It was compiled by David Ranada, Technical Editor of High

Fidelity Magazine, after auditions performed through a filter believed to

resemble the filter used by CBS in a Congressional hearing on April 2,

1987. The characteristics of the encoder CBS submitted to the NBS may

differ in some important respects from the April encoder --we have no way of knowing this, since NBS and CBS have declined to give us access to

the specifications of the CBS filter actually submitted to NBS for study.

Accordingly, many of these selections may not fully illustrate the audible degradation caused by the version actually given to NBS. But the

wide musical variety of the selections listed here does indicate that,

should the characteristics of the NBS encoder be made known, appropriate

musical selections of equivalent variety, probably from different passages of the discs listed here, could also be chosen to "match" its

characteristics and thereby most clearly illustrate the distortion. .

.

"The selections listed below were chosen to illustrate a principal sonic

liability of any filter fulfilling the requirements of the proposed

legislation: removal of significant and audible fundamentals or harmonics

from a musical signal. Other sonic degradations from an encoder are possible, including a highly colored noise spectrum and encoder sub-stage

overload. Knowledge of the electronic characteristics of the actual CBS

filter will allow the NBS consultant(s) to select music suitable for

testing these aspects in listening tests.

"All selections are on CD. Although prerecorded cassettes and LPs would

equally be suitable for inclusion (and could also be encoded under the

proposed legislation), CDs were chosen for their lower noise level, their

much better playback repeatability with different pressings and machines,

and the ability to perform rapid phrase repeats.

D.1
"These selections are believed to be of suitable characteristics and of

sufficient duration to reveal the audible degradation caused by sharp

band-reject filter meeting the requirements of the proposed legislation

and of sufficient bandwidth and depth to allow the design of a reliable

notch-detection scanner circuit. This is not to imply, however, that

other portions of this, or other, material, is entirely unaffected by

such filtering. .

.

"Recommended Selections

"In the list below, the first number for each selection (e.g., "5.")

refers to an entire CD. The numbers with capital letters (e.g., "5A,

5B") refer to particular tracks within that CD. This information is

followed by the track number and time reading as indicated by the readout

on the CD player. The particular material referenced is then further

described, occasionally with further notations of "obvious" or "very

obvious" (as tested by Mr. Ranada with his filter) . The selections are not in any particular order."

1. Streisand: The Broadway Album. CBS CK-40092. Pressing made

in Japan Matrix No. 1A7 73

1 -A Track 4 1:42 -1:46 note on word "time"

1-B Track 9 4:10 - 4:14 note on word "dies"

1-C Track 10 1:20 - 1:22 note on word "man"

1 -D Track 11 2:11 - 2:16 note on word "friend"

1-E Track 12 2:51 - 2:53 note on word "time" **very obvious

1-F 2:56 - 3:00 note on syllable "-day"

1-G 3:00 - 3:03 note on syllable "-where"

1-H 3:58 - 4:03 note on syllable "-how"

1-1 4:04 - 4:07 note on syllable "-day"

2. Prokofiev : Alexander Nevsky. London 410 164-2.

2-A Track 7 4:16 - 4:31 piccolo note. NB: note

disappears for a moment at 4:25 in

version

**very obvious

3. Joan Sutherland: Bel Canto Arias. London 417 253-2.

3-A Track 4 5:44 - 5:48 soprano high note

4. Villa-Lobos: Bachianas Brasileiras No. 5. London 411 730-2.

4-B Track 14 0:29 - 0:33 soprano note

4-B 1:22 - 1:24 soprano note

5. Beethovan : Symphony NO . 5 . Telarc CD-80060.

5 -A Track 2 5:16 - 5:20 unison E-flat, full orchestra

5-B Track 3 11:26 - 11:29 E-flat in violins

6. Beethoven : Symphony No . 6 . Delos D/CD-3017

6-A Track 2 11:03 - 11:07 B-flat note in violins

D.2
7. Mozart

:

Sinfonia Concertante . Deutsche Grammophon 413 461-

7-A Track 1 1:54 - 2:01 syncopated chords

7-B IV 2:23 - 2:26

7-C If 8:07 - 8:11 violin note

Mozart

:

Violin Concerto No. 1 7-D Track 5 2:06 - 2:20 violin note

7-E It 5:51 - 6:06 It 7-F II 9:08 - 9:11 IV

8. Sibelius;: Violin Concerto. RCA RCD1-7019.

8-A Track 1 5:52 - 5:54 high B-flat in violin

Prokofiev: Violin Concerto No. 2

8-8 Track 5 1:21 - 1:23 violin note

9. Copland: Fanfare for the Common Man Pro Arte CDD-102

9 -A Track 5 2:18 - 2:20 brass chord

Barber

:

Adagio for Strings

9-B Track 7 6:55 - 7:07 string chord

10. Barber

:

Adagio for Strings. Telarc CD-80059

10-A Track 4 4:57 - 5:06 string chord (same as 9-B)

11. Stravinsky: Petrouchka. CBS MK-37271

.

11 -A Track 1 35:45 - 35:48 trumpet dissonance

12. Carlos

:

Switched on Bachl. CBS MK 7194.

12-A Track 7 0:19 - 0:22

13. Beethoven: Violin Concerto. London 400 048-2.

13-A Track 2 6:27 - 6:28 violin D-sharp

14. Beethoven: Symphony No. 9. Angel CDC 7 47189 2.

14-A Track 1 0:48 - 0:52 orchestral chord

15. Beethoven: Symphony No. 9. Denon CD-7021.

15-A Track 1 0:41 - 0:48 orchestral chord (same as 14-A)

16. Stravinsky: L'Histoire du soldat. Delos D/CD 3021

16-A Track 8 0:18 - 0:22 chord

17. Elgar: Violin Concerto. Deutsche Grammophon.

17-A Track 2 7:13 - 7:22 B-flat in violin

18. The unforgettable Glenn Miller. RCA PCD1-5459.

18-A Track 4 0:22 - 0:23 telephone bell in "Pennsylvania

6- 5 -Thousand"

18-B IV 0:35 - 0:36 vv 18-C IV 1:01 - 1:02 IV 18-D It 2:13 - 2:14 vv 18-E Track 12 0:09 - 0:22 trumpet notes

D.3
18-F Track 13 3:21 3:23 last chord of "Serenade in Blue

19. Debussy

:

La Mer. Angel/ [MI CDC 7 47028 2

19 -A Track 1 2:08 - 2:12 violin chord

20. Debussy

:

La Mar

.

CBS MYK 37261

20-A Track 1 3:44 - 3:46 violin chord (same

**very obvious

as 19-A)

21. Debussy

:

La Mer. Telarc CD-80071

21 -A Track 1 3:33 - 3:36 violin chord (same asl9-A)

21 -B Track 2 0:45 -0:46 percussion notes

22, Kodaly

:

Hary Janos Suite. Angel/EMI CDC 7 47109 2

22-A Track 2 0:12 - 0:13 top piccolo note disappears

**very obvious

23. Kodaly

:

Hary Janos Suite. CBS MYK 38527

23-A Track 2 0:12 - 0:14 top piccolo note disappears (same

as 22-A)

23-B Track 2 2:06 - 2:07 piccolo notes disappear

23-C Track 6 2:51 - 2:54 If Prokofiev: Lt. Kije Suite

23-D Track 10 2:33 - 2:25 It

24. Sousa: Marches. Kem Disc 1004

24-A Track 6 2:39 - 2:40 piccolo notes disappear with

filtering during "Anchor & Star" March

24-B Track 7 1:53 - 1:56 "Stars and Stripes Forever"

24-C II 2:48 - 2:50 top piccolo note disappears with filtering

**very obvious

25. Messiaen: Turangalila Symphony. CBS M2K 42271. Disc 1.

25-A Track 2 6:20 loud chord

25-B Track 4 0:29 piccolo note altered

25-C II 1:03

26. Mahler: Symphony No. 3. CBS M2K 42403. Disc 1.

26-

A Track 15 3:12 - 3:17 orchestral chord with piccolo

**very obvious

27. Messiaen: Organ Music. Calliope CAL 9926.

27

A Track 5 3 : 20 - 3:34 organ chord **obvious

27-

B Track 8 5:17-5:37 " **obvious

28. Sousa: Stars and STripes Forever. Telarc CD-80099

28-

A Track 11 2:56 - 3:00 top piccolo note disappears (same as 24-C)

**obvious

29. Beethoven: Symphony No. 9. Deutsche Grammophon 410 987-2

29-

A Track 5 3:16 - 3:23 chord with chorus "vor Gott"

D.4
30.

Debussy: Preludes Book I No . 5. Calliope CAL 9831.

30-A Track’ 5 2:36 - 2:39 top piano note in passage

**very obvious

31.

Debussy: Preludes. Angel/EMI CDS 7 47608 8. Disc 1.

31-A Track 5 3:14 top piano note disappears with filtering

**very obvious

31-B " 3:16 - 3:17 same as 30-A

31-C Track 11 0:20 0:22 trilled piano chord

32.

Copland: Fanfare for the Common Man. Telarc CD-80078.

32-A Track 1 1:56 - 1:58 brass chord (same as 9-A)

33. Mahler: Symphony 9. CBS M2K-42033. Disc. 1.

33-

A Track 1 7:04-7:12 String and brass chords

34. Mahler: Symphony 9. CBS M3K-42200. First Movement.

34-

A 1st Mvmt. 6:37 - 6:44 Same as 33-A

35. Strauss: Ein Heldenleben. London 414-292-2.

35-A Track 1 3:54-3:58 Full orchestra chord

35-

B Track 5 5:10 - 5:30 B-flat in upper strings

36. Strauss: Ein Heldenleben. CBC SMCD5036

36-

A Track 1 3 : 00 - 3:04 String chord

36-B Track 1 4:06 - 4:11 Full orchestra chord

36-C Track 2 12:18 - 12:39 Same as 35-B

37.

Beethoven: Symphony 2. EMI/ANGEL CDC 7 47698 2.

37-A Track 2 3:19 - 3:25 B-natural in violins

37-B " 7:23 - 7:30 E-natural in violins

38.

Shostakovich: Symphony No. 10. RCA 6597-2-RC.

38-A Track 4 10:21 - 10:27 Piccolo runs altered when filtered

39. The Beatles: Rubber Soul

39--A Track 8 0:06 - 0: 08

39--B II 0:11 - 0 : 13

39--C If 0:26 - 0 : 27

39--D II 0:48 - 0: 50

39--E If 0:53 - 0: 54

Capitol CDP 7 46440 2.

Syllable "on"

If

40. The Beatles: Revolver.

40-A Track 1 1:07 - 1:10

40-B Track 4 2 : 34 - 2:54

40-C Track 7 2:19 - 2:31

Capital CDP 7 46441 2.

Syllable "oh," backup chorus

Sitar drone

Electric organ

41. The Art of Beverly Sills

41 -A Track 2 9:02 - 9:05

41 -B Track 6 5:30 - 5:32

41 -C Track 9 4:27 - 4:34

Vol . 2, EMI/ANGEL CDC 7 47332 2.

Soprano note

D.5
42. Wagner:

42-A Track B

Parsifal Prelude. EMI/ANGEL CDC 7 47255 2.

4:25 - 4:44 Wind chord

43. Verdi: La Traviata. Deutsche Grammophon DG 415 132-2.

43-A Disc 1, Track 11 0:27 - 0:29 violin chord

43-B Disc 2, Track 2 0:00 - 0:39 damage is audible on

and off during the

track, depending on

the chord being

played

44. Verdi Otello . RCA RCD2-2951

.

44-A Disc 1

,

Track 10 3:35 - 4:05 Violin Trill

44-B Disc 2, Track 12 0:00 - 0:54 Held violin note

44 -C ft 2: 55 - 3:12

44-D If 4: 03 - 4:37 Effect audible

depending on chord

45. Beethoven: Symphony No. 9. Telarc CD-80090.

45-A Track 1 0:41 - 0:50 Violin E-flat

45-B If 11:33 - 11:40 tf 45-C Track 4 10:44 - 10:50 B-flat in violins

46. Kitaro

:

My best. Gramavision 18-7016-2.

46-A Track 11 0:27 -0:30

46-B If 0:41 - 0:45

46-C ft 2:25 - 2:31

47. Britten: Ceremony of Carols. EMI/ANGEL 7 47709 2.

47-A Track 2 1:14 - 1:17 Chord, boys chorus with harp

48. Bernstein: Chichester Psalms. Deutsche Grammophon 415 965-2.

48-A Track 13 0:01 - 0:04

48-B ft 0:07 - 0:09

48-C ft 0:32 - 0:36

48-D Track 14 1:16 - 1:19

48-E ft 1:34 - 1:35

48-F ft 2:45 - 2:48

48-G fl 4:51 - 4:54

48-H Track 15 0:04 - 0:06

48-1 It 0:19 - 0:22

46-

J

If 0:48 - 0:50

48-K If 4:16 - 4:20

49. Bernstein : Symphony No. 1. Deutsche Grammophon 415 964-2.

49-A Track 2 0:54 - 0:57

49 -B It 1:41 - 1:46

49 -C II 5:12 - 5:19

50. Debussy

:

Nocturnes. Philips 400 023-2.

50-A Track 1 4:26 - 4:43 String chord

50-B It 5:07 - 5:16 It

D„6
51. The Art c>f Beverly Sills, Vol. 1. Angel CDC 47183.

51 -A Track 4 6:18 - 6:20

52. Bach: Violin Concerto No. 1. Erato ECD 75358.

52-A Track 5 3:54 - 4:01

53. Strauss

:

Lieder. Philips 416 298-2.

53 -A Track 4 1:09 - 1:12

53 -B Track 10 4:27 - 4:30

53-C Track 14 0:58 - 1:00

53-D Track 20 2:43 - 2:46

54. Strauss

:

Der Rosenkavalier . Angel CDS 7493542. Disc 1.

54-A Track 19 6:55 7:21

55. Wagner: Das Rheingold. Deutsche Grammophon 415 141-2. Disc

55-A Track 1 2:04 - 2:40 Selected notes in violin

arpeggio altered

55-B 3:48 - 4:35

56. Strauss

:

Tod und Verklaerung. Chandos CHAN 8533.

56-A Track 4 13:14 - 13:17

56-B II 13:18 - 13:24

57. Strauss

:

Don Quixote. Angel CDC 747865 2.

57-A Track 2 15:54 - 16:09

57-B II 18:01 - 18:22

57-C II 19:42 - 20:00

58. Scriabin: Poem of Ecstacy. Erato ECD 75360

58-A Track 3 18:37 - 18:42

59. Verdi: Aida. Deutsche Grammophon 410 092-2. Disc 3.

59-A Track 12 0:44 - 0:52 2nd violin B-flat harmonic

59-B It 0:55 - 1:05

59-C II 1:07 - 1:16

59-D It 1:19 - 1:26

59-E II 2:07 - 2:13

59-F It 4:50 - 5:53

60. Holst; The Planets. Deutsche Grammophon Galleria 419 475-2.

60-A Track2 1:35 - 1:54

60-B It 2:40 - 2:56

60-C II 4:55 - 5:02

61. Elgar : Cockaigne. London Jubilee 417719-2 . 61-A Track 21 4:25 - 4:33 String chord

61-B It 5:15 - 5:20

62. Bartok: Music for Strings, Percussion and Celesta. London 41

62-A Track 9 4:34 - 4:42 string chord

D.7
62-B " 4:46 - 4:55

62-C Track 10 1:52 - 2:00 Strings B-flat, left channel

62-

D Track 12 1:18 - 1:23

63. Bartok: Concerto for Orchestra. London 400 052-2

63-

A Track 1 2:36 - 2:41

64. Sibelius: Symphony No. 7. Bis BIS-311.

64-

A Track 1 16:32-16:35

64-B " 16:36 - 16:39

64-C " 18:56 - 19:20

65. Bernstein West Side

65 -A Track 1 1:53 - 65-B Track 5 0:44 - 65-C II 3:28 - 65-D Track 8 2:38 -

66. Sibelius

:

Finlandia

.

66-A Track 1 4:54 - 66-B II 5:38 - 66-C II 6:29 - 66-D II 6:50 - 66-E It 7:11 - 66-F It 8:55 -

Story. CBS CK 32603.

1:58

0:47

3:36

2:45

Angel Studio CDM-769017-2

.

5:41

5:41

6:31

6:52

8:16

9:10

67. Schumann: Symphony No. 3. Deutsche Grammophon 415 358-2.

67-A Track 1 2:57 - 3:05

67-B " 6:05 - 6:25

67-C " 9:05 - 9:24

68. Bruckner: Symphony No. 4. Angel Studio CDM- 769006- 2

.

68-A Track 1 4:08 - 4:16

68-B " 20: 01-end of movement

D.8
Recording Industry Association of America

The 35 selections listed below were sent to NBS on September 29' , 1987. Many

of these selections were reviewed as to their suitability for inclusion in the

listening tests.

Selection Artist Label

1 . Bartok: Hungarian Pictures Equale Brass Nimbus

2. Copland: Rodeo. Hoedown Atlanta Sym. Telarc

3. Mendelssohn: Sonata for Piano/

Klavier. 4.

.

Perahia CBS/Sony

4. Bach/Gounod: Ave Maria Shinozaki Devon

5. Chopin: Piano Concerto No. 2

in F minor

Davidovich Philips

6. McCartney/Jackson: Say Say Say McCartney CBS/Sony

7. Tschaikowsky : Sym. No. 6

Pathetique 3

.

L. A. Philar DG

8. Kabalevsky: con. No. 1 for

Cello & Orch. 2

Yo Yo Ma CBS/M

9. Tschaikowsky: Sym. No. 6 L.A. Philar DG

10. Sting Spirits in the Material Police (AM)

11 . Suite for Flute and Jazz Piano Rampa

1

CBS/M

12. Bach: Violin Concerto in E

maj or 1

.

Mutter/Accardo EMI

13. Haydn: Horn Concerto No. 2

in D. 2

Wallace/Thomp

.

Nimbus

14. J. Taylor: There We Are Taylor CBS

15. Rodrigo: Concierto de Aranjuez 1. J. Williams CBS/M

16. Bach: Concerto for Oboe, Violin

& Orch. 3.

Stern/Zukerman CBS/Sony

17. Schubert: Arpeggiona Sonata. 1. Galway RCA

18. Momentary Lapse of Reason Pink Floyd CBS

19. Schubert: Arpeggione Sonata. 2. Galway RCA

20. J. Zawinul : Dream Clock Weather Report CBS/Sony

21. Malcolm Arnold: Quintet. 1. Equale Brass Nimbus

22. Ellington: Rockin' in Rhythm Weather Report CBS/Sony

23. Copland: Rodeo. Corral Nocturne Atlanta Sym. Telarc

24. Beethoven Symphony No. 9 (Choral) Walter CBS/M

25. E. Jones: The Things I Used to Do S.R. Vaughn Epic

26. Haydn: Trumpet Concerto in

E flat 3.

Wallace/Thomp Nimbus

27. Schubert: Serenade Galway RCA

28. Mendellssohn: Prel. and Fugue

Op. 35, No .

1

Perahia CBS/M

29. George Strait #7 Strait MCA

30. Vivaldi Four Seasons P innock Archiv

31. Some Belouis EMI

32. In Square Circle Wonder Motown

33. Billie Holiday Story Various CBS

34. Liszt Transcendental Studies Horowitz Telarc

35. Turbo Judas Priest CBS

D.9
, - ' V . . : ‘ ' ‘ * .. .
APPENDIX E

Description of Recorded Material for Listening Tests

Appendix E: Description of Recorded Material for Listening Tests

The musical selections that were included in the serial and parallel listening

tests are identified in table E-l, along with a description of the observed

effects that led to their inclusion in the study. The first column of table

E-l includes an item number, corresponding to the order of presentation for

the first 24 trials in the serial listening study. An asterisk after the item

number indicates that the particular selection was suggested by the HRRC while

a plus sign means that the RIAA suggested that compact disc. Immediately

below the item number is given the Serial Number of the Encoder (see section

2.1) used to encode the musical selections for both the serial and the parallel listening tests. The second column provides the identification of the

compact disc. The third column indicates, as applicable, the composer, the

composition, and the performer. The fourth column provides a brief identification of the selection for use in identifying it in the color spectrographs given in section 4.4 and in this appendix, as well as in the tables in

section 4.5 and Appendix C. Column 5 indicates the particular track on the

compact disk, as well as the time period from that track -- the time interval,

in minutes and seconds, given just below the Track Number corresponds to the

short selections use in the serial listening tests while the second time

period, if any, that is enclosed in square brackets designates the portion of

the track that was used for the parallel listening tests.

The final column of table E-l indicates which, if any, of the six hypothesized

effects discussed in section 4.4 were believed to be potentially audible for

that particular selection. The text across the bottom of the block for each

selection describes the effects that were observed by NBS staff for that

selection. As mentioned in section 4.4, NBS staff utilized physical measurements and real-time spectrographs to assist in identifying these particular

selections and the potentially audible effects of the encoder. Extensive

listening by NBS staff was then performed in order to determine whether the

effects of the Encoder could be heard and to relate the audible effects, if

any, to the hypothesized effects.

After the selections were recorded to digital audio tape, in both direct and

encoded form, these tapes were used as the source material for the color

spectrographs shown on the cover of this publication, in sections 2.4 and 4.4,

and in this appendix. These spectrographs were obtained by storing the sum of

the left and right channels for each musical sample on a disc using the

"throughput" mode of an HP 3562A analyzer. The input range of the analyzer

was set so the peak level of each sample was just below overload. Then, the

data were analyzed using the Fast Fourier Transform capability of the analyzer

to obtain the power spectrum of each record, each record being 128 ms long.

Thus, there are approximately 7.8 records per second in the spectrographs.

The analyzer bandwidth was 0 to 6.25 kHz, with 801 frequency bands in the FFT

analysis. Each odd-numbered frequency band from the analysis was averaged

with the next-higher even-numbered band, reducing the 801 bands to 400. Since

only frequencies up to 6 kHz are displayed, each record contains 384 separate

frequency bands, spaced 15.62 Hz apart.

Each record from the Analyzer was read into an HP 330 computer, and processed

to transfer the complete spectrograph onto an HP PaintJet printer. The dynaE.1
mic range of compact discs is on the order of 90 dB , so this range was chosen

for display on the spectrographs, with a range of 10 dB for each of 9 colors.

Because the dynamic range of the analyzer is 80 dB , the full 90 dB cannot be

displayed for any one selection. However, the input range was set to make

optimum use of the analyzer for each selection, so 90 dB is displayed over all

samples

.

Examples of the many spectrographs that were made during the testing are illustrated on the cover of this report and in section 4.4, figures 32 and 33.

E„2
Table E-l Musical Selections for Listening Tests

Item &

Encoder

CD

identification

Title and

artist

Short

I. D.

Track no

.

and time

[ extended

time ]

hypothesized

effect(s)

1 . * CCE-006

EMI

CDC 7470282

Debussy: La Mer

Previn, London

symphony orch.

Debussy 12:09-2:12

2

A strong harmonic of the violins is removed

2 . * CCE-012

Telarc

CD80059

Barber : Adagio

for Strings

Slatkin, St.

Louis Symph.

Barber 4

4:57-5:01

2,3,5

The chord which begins at 4:57 has the upper part of the spread of a harmonic

removed. At 4:59 a new chord begins while the original chord continues. At

5:00, the sound naturally brightens slightly, although no new note is started.

This brightening causes the encoding notch filter to be switched out, causing

an increase in the level in the vicinity of 3840 Hz.

3. + CCE-012

CBS

MK42014

Beethoven: 9th

symphony

Bruno Walter,

Columbia Symph.

Orch

.

Beethoven

chord

10:38-0:42

2,3

The upper portion of the harmonic of an orchestral chord is removed

4. * CCE-006

DG

41098-2

Beethoven: 9th

symphony

Karaj an: Berlin

Philharmonic

Beethoven

chorus

53:16-3:23

2?

The spectrum of a full chorus and orchestra

encoding region, which were removed.

had minor components in the

5.

CCE-012

Denon

33C 39-7441

Bizet- Sarasete

Carmen Fantasy

Paganini Ensem.

Bizet 23

0:15-0:18

[0:15-1:10]

2,6

The lower half of the spread of a harmonic of a solo violin is removed.

E.3
[Table E-l, cont'd]

6.

CCE-012

Denon

33C 39-7441

Billy Joel:

Just the Way

You Are

Nancy Wilson

Wilson 14

0:27

2

Syllable ’Nev" of word "Never" has harmonic removed.

7. * CCE-012

CBS

CK40092

Bernstein:

Somewhere

Barb. Streisand

Streisand 11

2:56-3:01

[2:56-4:15]

2,5

Two different words have a strong harmonic removed.

8. + CCE-012CBS

MK42014

Beethoven: 9th

Symphony

Bruno Walter,

Columbia symph.

Beethoven

Triangle

4

10:56-11:01

[10:45

-11:30]

1,2

One of the partials of a triangle is completely removed

9.

CCE-012

CBS

MK42314

All the Things

You Are: Kern/

Hammerstein

Maur . McGovern

McGovern 2

3:03-3:06

2,5

A strong harmonic of the voice slides upward through encoding notch

frequency at beginning of word "are", with attendant loss of harmonic.

10. +

CCE-006

ARC/Co lumbia

CK36793

Dream Clock

Weather Report

Weather 25:39

1,2

A prominent partial of a cymbal tap was reduced in level.

11. * CCE-012

Telarc

CD80078

Copland:

Fanfare for

the Common Man

Lane, Atlanta

Symph

.

Copland 11:55

[0:53-2:03]

2,5

The brass instruments have a strong harmonic removed.

12.

CCE-006

DG

410025-2

Bernstein: Symphonic Dances

Bernstein, Los

Angeles Phil.

Berstein 11

0:41

[0:00-1:02]

2,3

A strong harmonic of high-pitched violins is removed.

E.4
[Table E-l, cont'd]

13. + CCE-006

Columbia

CK40599

Sorrow

Pink Floyd

Pink Floyd 10

0:52-1:02

2,5

A complex overloaded guitar sound has harmonic components removed.

14.

CCE-006

Synthesizer

No. 1 SI

A two-note chord of F7 and B7. The level of the B 7 is reduced.

15. * CCE-012

London

410164-2

Prokofiev:

Alex. Nevsky

Chailly, Cleveland Symph.

Prokofiev 74:16-4:21

[4

:

00 -end]

5

The Encoder is switched in and out on a sustained chord, with full orchestra

and chorus, erratically reducing a very audible component at the lower edge of

the encoding notch frequency region.

16.

CCE-012

Synthesizer

No. 2

S2

A trumpet fanfare in the key of D has harmonic components removed.

17.

CCE-012

ECM

1155

Pat Metheny

Amer. Garage

Metheny 4

0 : 00 - 0:02

[0:00-0:29]

2,4

Components in the encoding notch frequency region are removed from alternating

drum and cymbals. The cymbals originally had a minor formant at 3840 Hz.

18.

CCE-012

Synthesizer

No. 3

S3

This is a sustained E-flatg, with several other notes played consecutively.

19.

CCE-006

Synthesizer

No. 4

S4

This is a sustained chord of B4 , E 5 , and F-sharp 5 , with several other notes

being played consecutively. A partial of the E 5 is reduced.

E.5
[Table E-l, cont'd]

20. * CCE-006

CBS M2K42271,

disc 1 Messiaen:Turangalila

Salonen,

Philharmonia

Orchestra

Messiaen 4

1:03

[0:27-1:05]

1,2

The first part of a piccolo note, whose fundamental frequency falls

passband of the encoding filter, is removed.

in the

21.

CCE-012

Archiv

415 112-2

Bach: 3-part

Invention No.

2, Gilbert

harpsichord

Bach 17

0:23-0:27

[0:00-0:31]

2,4

A trill has a pronounced harmonic for each of the two notes of the trill

removed.

22.

CCE-006

Telarc

CD80085

Respighi

:

Fountains of

Rome , Lane

Atlanta Symph.

Respighi 13

1:58

[1:50-2:35]

1,2

A strong partial of a bell is markedly reduced in level

23.

CCE-006

L' oiseau-Lyre

411858-2, disc 1 Handel : MessiahNelson, Soprano

Hogwood, Acad.

Ancient Mus

.

Hande

1

11

1:58

6

A soprano trill has the center frequencies of its harmonics removed

24.

CCE-006

Telarc

CD8052

Ravel: Daphnis

et Chloe

Slatkin, St.

Louis Symph.

Ravel 3

2:07-2:10

2

The first and, especially, the third of a sequence of 3 have a prominent harmonic removed.

notes on a clarinet

* HRRC list

+ RIAA list

E.6
APPENDIX F

Preparation of Recorded Material
.

.

fen.

.

.
APPENDIX F: PREPARATION OF RECORDED MATERIAL

Equipment Used

The equipment used for making all of the recorded materials presented in both

the serial and parallel listening studies consisted of a Nagra Model IV-S

analog tape recorder, a Denon Model DCD-3300 compact disc player, a Kurzweil

Model 250 synthesizer, components of a compact disc mastering system manufactured by Sony, a Sony Model PCM- 3324 multi-channel digital audio tape recorder

with a Model RM-3310 remote control, and a digital interface box. The analog

tape recorder was used to make the original recordings of the announcements

and the tone that were a part of presenting the material to the listening

subjects. The compact disc player was used to transfer, from compact disc to

digital audio tape, the compact-disc material that was evaluated. The synthesizer was used to create and store the synthesized material that was evaluated.

The Sony disc mastering system components included two Model DMR-4000 digital

master recorders, two Model PCM-1630 digital audio processors, one Model DAE1100A digital audio editor, one Model DABK-1630 (RAR-1) printed circuit board,

one set of Model DABK-1631 digital input/output printed circuit boards (only

the input board was utilized), and several Model D-3/4-75 cassette tapes. A

more detailed description of the characteristics of these components and the

components of the multi-channel recording system follows.

The switch settings and any modifications of the printed circuit boards used

in the two audio processors were as follows. The Model DA-15 digital-toanalog printed circuit boards were modified by replacing the low-pass filters

AFL101 and AFL201 with Apogee Electronics Corporation Model 944-G anti-aliasing low-pass filters for both audio processors. The Model AD-23 analog-todigital printed circuit board installed in the audio processor used to make

the recordings was modified by replacing the low-pass filters AFL101 and

AFL201 with Apogee Electronics Corporation Model 944-G anti-aliasing low-pass

filters. (Note: all use of these printed circuit boards including recording,

playback, testing, and calibration was with the Apogee filters installed in

them.) The setting of the switches on the printed circuit boards AD-23, ENC2, SIF-1 DEC-15, and MT-16, in both audio processors were set to the factory

preset positions as given in the 4th or 5th revision of the 1st edition of the

PCM-1630 operation and maintenance manual with the exception of the EMP

(emphasis) switch on the AD-23 board and the REC MUTE (record muting) switch

on the ENC-2 board which were normally set to OFF except as needed. The

switches on the optional DABK-1630 (RAR-1) printed circuit board installed in

the audio processor used in making the recordings were set to the factory

preset positions as given in the 3rd revision of the 1st edition of the DBAK1630 operation and maintenance manual. The muting time was set for one second

on this board. (These settings are included in the interest of completeness

since the board was installed in the audio processor during the recordings.

However, the read-after-read function, available through this board was never

used since the PB MODE selector switch on the audio processor was always in

the A position) . The DI-5 digital interface printed circuit board was the

only one of the two boards included in the DBAK-1631 set of digital I/O boards

F.1
that was used. The switches on this input board were set as follows: DIP SW1

- switch 1 was OFF, i.e., set for Sony/Philips consumer formatted data, with

switches 2 through 8 ON; the manual emphasis switch (SW-2) was OFF; and the DI

SYNC switch (SW3) was ON (see the 2nd revision of the 1st edition of the DBAK1630 operation and maintenance manual) . Only one of the two audio processors

was used to make all of the digital recordings, the other was used to play

back the recorded materials during the serial presentation.

The switches of the DAE-1100A were set for serial data and word sync and a sampling frequency of 44.1 kHz (DROP FRAME OFF). The switches of the keyboard

were set as needed. Player A was the only player connection block that was

utilized. The input select switch (IN SEL) in this block was in position D

(digital input signal) when the compact disc player output signal was used and

position V (composite digital input signal) when the digital master recorder

output signal connected to PLAYER-A block was used.

The switches of the DMR-4000 digital master recorder internal printed circuit

boards DM-49, SY-37/SY37A, and TC-38 were set to the factory preset positions

as given in the 5th revision of the 1st edition of the DMR-4000 operation and

maintenance manual. These recorders were always used with the METER switch in

the R/P position, the AUX LIMITER switch OFF, the RAW OUT selector in the MAIN

position, and the SKEW control in the fixed position. The tracking control

was normally in the fixed position except as needed. All the recordings made

using these machines were done with the recorder and/or player in the PB/EE

mode of operation. Normally the synchronization of these machines was done by

the digital editor in the editing process during which time the AUX CH2

selector was in the AUDIO position. However, when it was desired to make a copy of a tape synchronously, with the time code recorded at the same time as the audio, the AUX CH2 selector switches of the player and recorder were in

the TIME CODE position with the TIME CODE selector switch in the REC/RUN

position in the case of the player and the REGEN position in the case of the

recorder (for these tapes the time code output of the player was connected

directly to the time code input of the recorder). In all cases, the mastering

recorders were used with the 75 ohm composite digital input termination ON.

The switches of the PCM- 3324 multi-channel digital recorder printed circuit

boards ENC-1, ENC-2, ENC-3, DET-1 , DET-2 , DET-3 , DIN-1, DIN-2, DIN-3, A/D-1

through A/D- 6, TBC , DCC, CLK, EDTB , MCUP, EIO, LOC , CTL, SRVO, and SYS boards

were set to the factory preset positions as given in volume 1 of the 5th re- vision of the 3rd edition of the PCM- 3324 operation and maintenance manual

except as follows. The double indication/calibration selector switch was set

to DOUBLE IND, the emphasis switches on the A/D boards were set as necessary,

the XFADE (cross fade time) control on the MCPU board was set in position F,

which corresponds to a nominal crossfade time of 1.45 ms when using a sampling

frequency of 44.1 kHz, and switch 5 of DIP SW4 on the MCPU board was set to

OFF thus defeating the AUTO REC function of analog inputs Al and A2 . (Note:

it was not verified that the adjustment of the tension controls on the SRVO

board corresponded to the factory preset adjustment values.) The 75-ohm word

sync termination switch located on the PCM- 3324 was switched ON and the

normal/external word selector on the VCLK (video clock) board was set in the

EXT WRD position. The sampling rate used in the recordings was 44.1 kHz and

the INPUT SELECT and SYNC CLOCK switches on the toggle switch (TSW) block of

F.2
the PCM-3324 were set in the DIGITAL and AUTO positions, respectively. During

the parallel listening tests, the MASTER SAFE switch located on the TSW block

was set in the ON position. The Model RM-3310 remote control unit was operated without the audio control unit connected to the system control unit and

the selector switches on the rear of the unit were set to provide 100 cue

registers

.

A custom (not mass produced) digital interface (converter) was used to interface the balanced outputs and inputs of the PCM-3324 with the unbalanced outputs and inputs of the PCM- 1630 audio processor. The circuits and components

implemented in the design of this unit were based on the circuit descriptions

for conversion between balanced RS-422 and unbalanced TTL signals given in the

August 1987 revision of the PCM-3324 product information manual. The interface

was tested using a function generator and oscilloscope prior to using it to

make recordings for the listening studies.

Two rolls (one for test signals and one for program material) of Sony Model

D-l/2-1460 digital audio master tape and a Sony Model RH-14DA empty reel were

used in making and playing the multi-channel recordings.

Serial Presentation

Copying, Encoding, and Editing Procedures

Figures F-l through F-3 illustrate the functional arrangement of the equipment

used in preparing the material for serial presentation with the exception of

the recording of the synthesized selections, announcements, and. 3840 Hz tone.

Unless otherwise noted, all cassettes used in the recording process were "prestriped" (time code was recorded on the cassettes prior to recording audio)

using the DAE-1100A digital audio editor. The digital sampling frequency used

in making all recordings was 44.1 kHz.

Figure F-l Equipment setup used for transfer from compact disc to

digital audio tape. ("D" denotes digitized audio)

F.3
A

WIRE

OR

ENCODER

DMR 4-

d_.> AUDIO D DMR

PLAYER PROCESSOR/EDITOR A RECORDER

Figure F-2 Equipment setup used for making direct and encoded digital tapes.

("D" denotes digitized audio; "A" denotes 2-channel analog audio)

DMR 4 ° >AUDIO D DMR

PLAYER PROCESSOR/EDITOR A w RECORDER

Figure F-3 Equipment setup used for digital editing. ("D" denotes digitized

audio)

When recording the synthesized selections, the balanced outputs of the synthesizer (Kurzweil Model 250) were connected to the analog inputs of the AD-23

(analog- to-digital) board installed 'in the PCM-1630 audio processor used to

make the digital recordings. The selections were recorded onto cassette

through the DAE-1100A (digital audio editor) using a DMR-4000 (digital master

recorder) . The sequencer in the Model 250 was used to playback prerecorded

sequences, or selections during these recordings. The announcements and tone

were recorded, edited and leadered using an analog tape recorder (Nagra IV-S

in the case of recording the material) . The balanced outputs of the Nagra

F.4
were connected to the PCM- 1630 and the final analog tape containing the

announcements and tone was copied onto cassette using the DAE-1100A and a DMR4000 in the same fashion as the synthesized recordings. The material that

originated on compact disc was recorded onto cassette by connecting coaxial

digital output No. 1 of the DCD-3300 (compact disc player) to the DI-5

(digital input) board installed in the PCM- 1630 used to make the recordings

(see figure F-l) . A twisted pair of shielded wires was used to connect the

DCD-3300 to the PCM-1630 with pin 3 of the XLR input connector on the PCM-1630

connected to the center conductor of the coaxial output of the DCD-3300 and

pin 2 of the XLR input connector on the PCM-1630 connected to the outer

conductor of the coaxial output (chassis) of the DCD-3300. The shield was

connected to pin 1 of the XLR input connector terminating at the PCM-1630 but

not to the shell of the XLR input connector or to the DCD-3300. The chassis

of the DCD-3300 was left isolated from ground by the input stage of the DI-5

board. Of the shielding/grounding schemes measured, this connection

arrangement provided the fastest rise time in the waveform of the data stream

and resulted in a peak-to-peak voltage greater than 0.5 volts in the digital

waveform. (Note: the DI-5 digital input board was set specifically to decode

Sony/Philips consumer formatted data, not AES/EBU formatted data. Since the

two formats are similar but not identical, there may be some commercially

available equipment that is designed to decode AES/EBU formatted data that

also accepts Sony/Philips consumer formatted data but not without some amount

of error occurring in the decoding process, e.g., incorrect recognition of em- phasis information.) The selections were recorded onto cassette through the

DAE-1100A (digital audio editor) using a DMR-4000 (digital master recorder)

.

The GAIN OFFSET fader and switches on the keyboard of the DAE-1100A were used

to create the beginning and end of each of the twenty test selections plus

four practice selections taken from compact disc. There was no overall gain

offset introduced into the recorded material during this recording process,

i.e., the gain offset was set for 0 decibels as shown by the illumination of

the LED adjacent to the gain offset fader, indicating that the gain offset

function was not in use between fading or switching the material in and out.

These recordings, plus the recordings of the synthesizer, comprised a master

cassette of the twenty- four selections used in the serial study plus the four

practice selections.

The selection master was copied three times: once with shielded cables

directly connecting the outputs (analog) of the DA- 15 digital- to -analog board

of the PCM-1630 to the inputs (analog) of the AD-23 analog- to-digital board of

the PCM-1630, once with Encoder 006 between the outputs of the DA- 15 board and

the inputs of the AD- 23 board, and once with Encoder 012 between the outputs

of the DA- 15 board and the inputs of the AD- 23 board (see figure F-2)

.

These

copies were made through the DAE-1100A with the time code output of the player

connected directly to the time code input of the recorder and 0-dB gain offset

in the editor. The recorder and player were operated synchronously with the

AUX CH2 selector switches of the player and recorder in the TIME CODE position

and the TIME CODE selector switches in the REC/RUN position in the case of the

player and the REGEN position in the case of the recorder. The time code on

the selection master was recorded onto the three copies at the same time that

the audio material was recorded so that the audio material being recorded had

the same relative position with respect to the time code track for each of the

three copies. The rms voltage levels at the output of the two channels of the

F.5
PCM-1630 were adjusted to +20.0 dB relative to J0.6 V rms (70.6 V rms corresponds to 0 dBm) using the 1-kHz, 0-dB digital reference level from track 1 of

the CBS CD-I test disc recorded onto the selection master cassette during the

same time period that the selection material was copied from compact disc.

(Note: the term "dBm" as it has been used in this appendix denotes voltage

level referenced to 1 milliwatt of average electrical power dissipated in a purely resistive 600 ohm electrical load. However, the use of the term does

not imply that the voltages that were measured were terminated by a purely

resistive load, or that they were derived from measurements of power.) The

drift observed in the PCM-1630 output levels over the duration of the time

period that the three copies were made was less than 0.1 dB . Since some of

the recorded material had peak levels equal to 0 dB (maximum digital level)

and because the gain of the Encoder was slightly greater than unity and the

AD-23 input stages exhibited a slight rise in response at low-frequencies

relative to mid-band, the gains of the analog input stages of the AD-23

analog- to-digital board were set for -1.0 dB at 1 kHz to prevent clipping the

signals during analog- to-digital conversion. (Note: since the distortion of

the input stages of the AD- 23 board was measured and found to be a function of

input level (see the following section on equipment calibration) , the rms levels of the selections were measured using a Bruel and Kjaer Model 2305

graphic level recorder and found to be at least eight decibels below the maximum peak recorded levels. Thus, at no time was the rms level of the voltage

at the input stages of the AD-23 board greater than +12 dB relative to 70.6 V

rms.) The lengths of the cables connecting the inputs and outputs of the PCM1630 during the ’’direct" recording were approximately 1 m. These cables were

connected between the outputs of the PCM-1630 and the inputs of the Encoder

during the ’’encoded" recordings. The lengths of the cables connecting the

outputs of the Encoder to the inputs of the PCM-1630 during the "encoded"

recordings were approximately 1.5m. In all cases, the shields of the cables

were connected to pin 1, but not the shell, of the connectors at each end of

the cables. The PCM-1630 and the Encoder were grounded at their respective AC

power connectors using the same AC power circuit (see the following section

for the differences in the "direct" and "encoded" noise spectra at the PCM1630 inputs with this grounding arrangement) . The Encoder was set for fixed

gain and direct (non- transformer) coupled outputs. The Encoder was switched

IN throughout the duration of each of the encoded copies. Digital emphasis

was added to the Weather and Bernstein selections at this stage of the

recording process by activating the emphasis circuitry in the input section of

the PCM-1630 with the EMP (emphasis) switch on the AD-23 analog- to-digital

board during each of the copies of these selections. (Note: the operation of

the EMP switch produced switching transients, in each of the copies, which

were removed in the next stage of the recording process. The amount of time

that the emphasis circuit was active before and after the recorded material

began and ended was also made the same for the emphasized selections at this

time.) The recordings were monitored using headphones connected to the PCM1630. These recordings, one direct and two encoded, were then copied and

edited using the DAE-1100A.

The three copies, "Direct," "E006," and "E012," were assembled into four

tapes: "DD," "DE," "ED," and "EE" using the DAE-1100A editor (see figure F-3)

.

These four tapes consisted of the four possible combinations of the pairs of

each of the twenty-four test selections repeated three times with tape DD

F.6
containing the direct/direct pairs, etc. In addition, tape DD contained the

four practice pairs presented ahead of the actual test pairs. The tapes were

assembled in the following manner. Because the material recorded on the three

copies of the selection master had the same relative position with respect to

time code, only one "play" or start point on the playback machine (EDIT- IN

point in DAE-1100A terminology) was needed for any given selection, so that

DD, DE, ED, and EE were edited using a total of twenty- four play points for

the test selections plus four play points for the practice selections. The

play points were defined 0.5 seconds before the first audible point of each of

the selections. The first "record" or start point on the recording machine

(EDIT-OUT point in DAE-1100A terminology) was arbitrarily defined for numerical convenience. The second record point was defined 0.5 seconds after the

last audible point of each of the selections, so that after the second electronic edit was complete the first presentation of the selection pair being

edited was complete with a one-second pause existing between the members of

the first pair in the presentation (0.5 seconds for the play point plus 0.5

seconds for the record point) . The third record point was defined by taking

the difference between the first and second record points to find the length

of each pair member and adding this time plus two seconds to the time defining

the second record point. This resulted in a three- second pause between the

end of the first and the beginning of the second presentation of the pair

being edited (0.5 seconds for the play point plus 2.5 seconds for the record

point) . With the length of the pair member established, the remaining record

points were defined so that there was a one -second pause between pair members

and a three-second pause between pair presentations. For each set of play and

record (EDIT-IN and EDIT-OUT) points, an electronic edit (AUTO EDIT) was

performed four times using the four tapes DD, DE, ED, and EE for recording and

two of the three tapes DIRECT, and E006 or E012 for playback (except for the

practice pairs which were presented in only one combination of pairs) . Because the editor synchronized the operation of the player and recorder during

each electronic edit and the edit points were, by definition, the same for

each of the four tapes, each selection was presented in the same way (the same pauses between pairs and pair members, beginning and ending points, etc.) for

each of the four presentations of possible combinations of pair members.

During this stage of the recording process, relative levels of the individual

selections were adjusted using the GAIN OFFSET fader in the DAE-1100A. The

level of the Streisand and Beethoven chorus selections were reduced by approximately 8 dB; the level of the Prokokiev selection was reduced approximately 7 dB; the levels of the Bach, Pink Floyd, and S2 selections were

each reduced by approximately 1 dB; the level of the Bernstein selection was raised approximately 6 dB; and the level of the Weather selection was raised

approximately 4 dB. Once the gain offset relating to each of these selections

had been established, it remained unchanged during the recording of each of

the selections onto the four tapes. These relative level adjustments

prevented excessively loud or soft playback of individual selections during

the serial presentations.

The four tapes DD, DE, ED, and EE were then edited, with the tape containing

the announcements, into two final tapes for presentation, SI and S2 (see

figure F-3)

.

The ordering and pairing of the material on these tapes cor- responded to the randomization established by the NBS Statistical Engineering

Division for Tests 1 through 96. During the editing process, the record

F.7
levels of the announcements were adjusted slightly as needed using the GAIN

OFFSET fader so that the level of the announcements corresponded more to the

level of the material following or preceding each announcement i.e., relatively loud or soft (the GAIN OFFSET fader was set to 0 dB gain as shown by the

illumination of the LED indicator adjacent to the fader indicating that the

gain offset function was not in use when recording the selections of test

material). The edit points were selected so that there was a 1.5-second pause

between the announcements introducing and concluding the test material and the

beginning and end of each of the presentations of test material. The beginning and end points of the material were those defined during the editing of

tapes DD, DE, ED, and EE i.e., the edit points were defined so that the

recorder began recording 0.5 seconds before and after the beginning and end of

the test material. The SI and S2 tapes were checked after completion to

verify that the pairing of direct and encoded material for each of the Tests

matched that established by the NBS Statistical Engineering Division and that

the pairing was consistent throughout the presentation of each Test (the HP

3565S signal processing system and 330 computer mentioned in section 4.2 were

used for this analysis)

.

Equipment Calibration.

The equipment used to calibrate and test components of the recording system

and the recording system as it was used to make the tapes for the serial

presentation were: (1) a copy of the CBS CD-I test disc for measuring CD

player performance with EIA standard signals, (2) an Audio Precision Model

System One-A dualj-channel input/output audio test system with a Zenith Model

ZWX-248-62 computer, (3) a Fluke Model 8860 digital multimeter with true rms AC voltage measurement capability, and (4) a Tektronix Model 7704 oscilloscope

with a Model 7A18 dual trace amplifier and a Model 7B70 time base.

The Audio Precision test system was calibrated by the members of the Electronic Instrumentation and Metrology Group at NBS. Unless noted otherwise, the

Audio Precision test system was used without any bandwidth control filters

resulting in a nominal measurement bandwidth of 10 Hz to 500 kHz. Thus, the

results of noise measurements, or measurements that include noise such as total harmonic distortion plus noise (THD+N) , include any ultrasonic noise

components present in the measured signal due to the digital sampling

frequency of 44.1 kHz. Unless noted otherwise, the 600 ohm input terminations

of the Audio Precision were used when making measurements with this unit. The

shields of all balanced connections to the Audio Precision were terminated at

pin one, but not the shell, of the XLR connectors at the ends of the cables

connected to the test system. The outputs of the Audio Precision were set for

50-ohm balanced operation. Only the XLR connectors on the Audio Precision

were used in connecting it to the PCM- 1630. (Note: the convention used for

the assignment of signal high and low to pins 2 and 3 of the XLR connectors

are reversed in the case of the Audio Precision test system and the Sony PCM1630 audio processor. No attempt was made to correct for this 180 degree

phase difference in the units when connecting the outputs and inputs of the

two instruments since only relative phase measurements were performed using

the Audio Precision.) All components of the Audio Precision test system were

grounded at their AC power connector when in use.

F.8
The Fluke Model 8860 digital multimeter was calibrated using a Fluke Model

5200A precision AC calibrator that in turn was calibrated by the members of

the Electronic Instrumentation and Metrology Group at NBS . The Fluke Model

8860 was used to set the gains of the analog input and output stages of the

PCM- 1630 during the recording of the Direct, E006, and E012 tapes. The

nominal AC measurement bandwidth of this instrument is 20 Hz to 300 kHz with a nominal upper frequency limit (-3 dB point) at 1 MHz. The nominal input

impedance of the 8860 is 10 megohms with an input capacitance of 70 pF or

less. The AC voltage measurements made using the Fluke 8860 were guarded with

the shield of the cable connecting the meter to the voltage source connected

to pin one, but not the shell, of the XLR connector at the voltage source and

to the guard circuitry of the digital multimeter at the Fluke 8860. The guard

circuitry was grounded at the voltage source (Fluke 8860 EXT-GD mode of

operation) . The CBS CD-I compact disc was copied onto cassette through the

DAE-1100A, without gain offset, for future use in testing the recording

system. With the compact disc player still connected to the PCM- 1630 the

outputs of the digital input board were connected directly to the inputs of

the DA-15 digital-to-analog board. The balanced analog outputs of the DA-15

board were connected directly to the 600-ohm inputs of the Audio Precision

test system and a number of measurements were performed using the test signals

on the CBS CD-I test disc. The output voltages of the DA- 15 board were

nominally adjusted to 7.75 Vrms (+20 dB re J0.6 volts rms , i.e., +20 dBm)

using the measuring voltmeter in the Audio Precision and the 1-kHz, 0-dB

reference level on track 1 of the test disc. The results of these measure^

ments were used to characterize the performance of both channels of the DA- 15

digital-to-analog board in combination with the Denon DCD-3300 as it was

connected to the PCM- 1630 during the copying of the compact disc material.

A summary of these measurement results, and the test-disc track and index num- bers associated with them, follows. The decibel terms in the linearity measurements refer to "0 dB” as the maximum (peak) digital signal level in the same sense as it is used in the description of the test signals on the test

disc. The term " 1-kHz reference" refers to the voltages present at the analog

outputs of the PCM- 1630 during the playback of the 1-kHz reference level on

track 1 of the CBS test disc. SMPTE refers to the Society of Motion Picture

and Television Engineers.

Frequency response without digital emphasis 17 to 19997 Hz: within ± 0.3 dB

(Track 6 Index 03 through Track 10 Index 01).

Frequency response with digital emphasis 125 Hz to 16 kHz: within ± 0.1 dB

(Track 12 Index 01 through Track 12 Index 05)

.

Linearity without dither 0 dB to -70 dB: within ±0.2 dB

(Track 18 Index 01 to Track 18 Index 11).

Linearity with dither -80 dB to -90 dB: within ±1 dB

(Track 19 Index 02 to Track 19 Index 03)

.

THD+N at 127, 997, and 10007 Hz: < 0.03 %

(Track 07 Index 02, Track 08 Index 01, and Track 09 Index 01)

(Note: 22-kHz band- limited measurement: < 0.02 %)

.

F.9
SMPTE intermodulation (IM) distortion: < 0.02 %

(Track 13 Index 01).

CCIF intermodulation (IM) distortion: < 0.01 %

(Track 13 Index 02)

.

Noise level re lk-Hz reference voltage: < -90 dB

(Track 04 Index 01).

Crosstalk (non-driven channel re driven) 0.125, 1, and 10 kHz: < -100 dB

(Track 02 Index 01, 02, and 04 to Track 03 Index 01, 02, and 04)

(Note: measurements were made using the CROSSTALK function in the Audio

Precision which band- limits its input signal from the non-driven channel to be

centered about the frequency of the input signal from the driven channel.)

The operation of the DMR-4000 recorders was checked by playing back the 997 Hz

harmonic distortion and the intermodulation distortion test tracks as recorded

onto cassette. In all cases, the differences in the distortion results obtained using the CD directly and the cassette copy was at least an order of

magnitude less than the results indicated above.

The DI-5 digital input board was replaced with the AD-23 analog- to-digital

input board used to make the recordings. The digital outputs of the AD-23

board were connected directly to the digital inputs of the DA- 15 board. The

balanced analog outputs of the DA- 15 board were connected directly to the 600-

ohm inputs of the Audio Precision test system and the balanced analog 50-ohm

outputs of the Audio Precision test system were connected directly to the

analog inputs of the AD- 23 board. The output voltages of the DA- 15 board were

nominally adjusted to 7.75 Vrms (+20 dB re J0.6 Vrms) and the AD- 23 analog

input stages were set for unity gain using the 1-kHz, 0-dB reference level on track 1 of the cassette copy of the CBS test disc, the measuring voltmeter in

the Audio Precision, and a 1-kHz sine wave applied to the inputs of the AD- 23

board. The results of these measurements were used to characterize the

performance of the AD-23 analog- to-digital input board in combination with the

DA- 15 digital- to-analog output board as a function of frequency and/or

amplitude. The frequency "sweeps" were actually performed at discrete

frequency intervals (typically 100 points) over the frequency range 20 Hz to

20 kHz using the SWEEP TEST function in the Audio Precision test system. The

amplitude "sweeps" were done at discrete amplitude points (typically 30

points) over a 20 decibel range using the SWEEP TEST function. Unless noted

otherwise, the voltages at the analog inputs of the PCM- 1630 during the

measurements were nominally equal to the 1-kHz reference of 7.75 Vrms (+20 dB

re JO . 6 Vrms )

.

An overall summary of the results of these measurements follows:

Frequency response 20 Hz to 20 kHz: within +0.5/- 1.0 dB

Phase response CH 2 re CH 1 20 Hz to 20 kHz: within ±3 deg

Linearity 0 dB to -70 dB re 1-kHz reference voltage: within ±0.4 dB

F.10
Linearity 0 dB to -90 dB re 1-kHz reference voltage: within ±3 dB

THD+N 20 Hz to 20 kHz at +14 dB re 0.775 Vrms (0 dBm): < 0.05 %

THD+N 20 Hz to 20 kHz at +4 dB re 0.775 Vrms (0 dBm): < 0.05 %

SMPTE IM distortion 0 dB to +14 dB re 0.775 Vrms (0 dBm): < 0.13 %

SMPTE IM distortion 0 dB to +12 dB re 0.775 Vrms (0 dBm): < 0.06 %

CCIF IM distortion 0 dB to +14 dB re 0.775 Vrms (0 dBm): < 0.04 %

(Note: the CCIF test signal was comprised of two test tones at frequencies of

11 kHz and 12 kHz combined in a 1:1 amplitude ratio.)

Noise level re 1-kHz reference voltage: < -90 dB

Crosstalk CH 1 re CH 2 (non-driven re driven) 20 Hz to 20 kHz: < -55 dB

Crosstalk CH 2 re CH 1 (non-driven re driven) 20 Hz to 20 kHz: < -70 dB

(Note: the crosstalk measurements were made using the CROSSTALK function in

the Audio Precision which band- limits its input signal from the non-driven

channel to be centered about the frequency of the input signal from the driven

channel .

)

The Fluke Model 8860 was used to establish the reference voltages present at

the outputs and inputs of the PCM-rl630 and set the gains of the analog input

stages of the PCM- 1630 to -1.0 dB during the recording of the direct and en- coded copies of the selection master. The output reference voltages were set

using the cassette-copy of the CD-I test disc to playback the 1-kHz reference

level recorded from Track 1 of the disc. The gains of the analog outputs of

the PCM- 1630 were adjusted to a level of +20.0 dB re JO . 6 Vrms (+20.0 dBm)

when terminated by the input impedances of their respective analog inputs in

parallel with the nominal 10 megohm input impedance of the digital multimeter^

When the Encoder was inserted between the inputs and outputs of the PCM-1630,

the voltage levels present at the inputs to the Encoder was measured in the

same fashion. In all cases, the variations in voltage levels present at the

PCM-1630 outputs due, for example, to variations in load impedance or drift in

the gains of the amplifiers in the PCM-1630 analog outputs were less than 0.05

dB during the direct or encoded recordings.

After the gains of the PCM-1630 outputs had been set, the noise at the inputs

of the PCM-1630 was measured as a function of frequency using the Audio

Precision test system. The noise at the inputs of the PCM-1630 was measured

three times: once with the inputs connected directly to the output stages of

the PCM-1630 and once with each of the two Encoders connected between the

inputs and outputs of the PCM-1630. The equipment was connected and grounded

as it was during recording of the direct and encoded copies of the selection

master except that the test system inputs (in this case nominally 100 kft) were

connected in parallel with the inputs of the PCM-1630. The noise was measured

at 200 points over the range 20 Hz to 100 kHz using the SWEEP TEST function in

the Audio Precision test system for each of the three cases, direct and two

encoded. For frequencies in the range 20 Hz to 100 kHz, any differences in

F.11
the noise measured at the inputs of the PCM- 1630 occurred more than 100 dB

below the reference voltage of 7.75 V rms / For frequencies measured in the

range 20 Hz to 4 kHz, the differences in the measured noise occurred more than

110 dB below the reference voltage of 7.75 V rms. Typically the differences

in the noise were less than 6 dB at any given frequency over the frequency

range measured.

Parallel Presentation

Copying, Encoding, and Editing Procedures

In preparing the material for presentation in the parallel listening study,

the selections were copied digitally from compact disc to the selection master

cassette using the same procedures as described above for the serial selections (see figure F-l). During the same time period that the selections were

copied, a number of test signals were copied from the CBS CD-I test disc for

testing purposes. The recordings of the test material were then edited with

the announcements of the start and end of the test selections using the PCM1630 audio processor and DAE-1100A digital audio editor (see figure F-3) . The

announcements and test selections were edited such that there was a two- second

pause between the announcements and the test material with the edit point (s)

occurring in the middle of the two-second pause. During this editing process,

the test signals recorded on the selection master were copied onto the same

cassette at the conclusion of the parallel listening test. At no time was

there any gain offset (as indicated by the illumination of the gain offset LED

next to the GAIN OFFSET fader) introduced into the recordings. The parallel

listening test signals and the test tracks were then copied digitally from the

parallel master cassette onto two tracks of the PCM- 3324 multi-channel recorder with the DMR-4000 player connected directly to the PCM- 1630 audio processor (see figure F-4) . (Note: the listening test signals and the test tracks

were recorded onto two separate tapes using the INSERT record mode of the PCM3324 with the connection of the recording equipment the same in each case.)

The multi-channel tapes were "prestriped" (control and time code tracks prerecorded) using the DAE-1100A editor to generate time code with the word sync of

the PCM-3324 "locked” to that of the PCM-1630 (SYNC CLOCK of the PCM-3324 set

for external reference). All recordings were performed using a 44.1-kHz

digital sampling frequency. All further recordings using the PCM-3324 were

DMR

PLAYER

AUDIO

PROCESSOR ^ CONVERTER

DIGITAL

MULTI-CHANNEL

RECORDER

Figure F-4 Equipment setup for digital transfer to multi-channel digital

audio tape. ("D" denotes digitized audio)

F.12
made at a 44.1-kHz sampling frequency with the PCM-3324 in the INSERT record

mode and the SYNC CLOCK of the PCM-3324 referenced to that of the PCM- 1630.

"Direct" and "encoded" recordings were made by connecting the analog outputs

and inputs of the PCM- 16 30 in the same manner as described for the recording

of the material used in the serial listening tests (see figure F-5) . In this

case, however, the entire test was recorded onto a set of tracks containing

direct material and a set of tracks containing encoded material using the PCM3324 to playback the initial digital recording of the listening test and

record the direct and encoded versions of the listening test. The gains of

the input stages of the AD-23 digital-to-analog board were checked and found

to be set for -1.0 dB (see the previous section Serial Presentation regarding

the reasons for this gain setting) . Direct and encoded recordings were made

of the test signals using the test-signal tape with the recording equipment

connections the same as in the case of the recordings of the listening test

except that only one Encoder was used in making these recordings. The testsignal recordings were made prior to the listening-test recordings and were

used to verify that proper channel (left and right, or CH 1 and CH 2) and

track assignments were maintained throughout the recording process. Encoder

012 was used to encode the first six selections in the test and Encoder 006

was used to encode the last four selections in the test. The AUTO PUNCH

feature of the PCM-3324 was used to transition between the Prokofiev and

Bernstein selections from material encoded using one Encoder to material

encoded using the other Encoder. The transition point ("punch in" point, in

PCM-3324 terminology) occurred during the pause between the announcement of

the end of the Prokofiev selection and the announcement of the beginning of

the Bernstein selection. To prevent the encoding of the announcements, the

Encoder was switched IN during the two -second pause between the end of each of

the announcements and the start of each of the selections and OUT during the

two-second pause between the end of each of the selections and the start of

each of the announcements The analog outputs on the PCM-3324 of the tracks

containing the initial digital recording of the test were used to monitor the

listening test when making the recordings.

Figure F-5 Equipment setup for making direct and encoded tapes on the multichannel recorder. ("D" denotes digitized audio; "A" denotes 2-

channel analog audio)

F.13
The tracks of the direct and encoded material were then copied digitally onto

three additional sets of tracks: a reference set (REF), A, and B. These

recordings were made by directly connecting the balanced digital outputs and

inputs of the tracks being used in the PCM- 3324. First the tracks containing

the direct material were copied on the REF set of tracks to insure that the

reference material had been processed in the same manner as the material

recorded on sets A and B. Then, since the majority of the direct recordings

were assigned to A, all of the direct material was copied on the A set of

tracks while, at the same time, all of the encoded material was copied onto

the B set of tracks . In order to have the direct and encoded material randomly assigned to A and B, the track assignments of A and B were then reversed

and three electronic edits were performed using the AUTO PUNCH feature of the

PCM-3324. When performing these edits, the points at which the PCM- 3324 began

and stopped recording ("punch in" and "punch out" points, in PCM-3324 terminology) occurred during pauses between announcements of the beginning and

end of selections. During these edits, the previously recorded material was replaced by the new material between the edit points. Prior to making these

recordings, the test signals were recorded on the three sets of tracks REF, A,

and B with the direct recordings assigned to REF and A and the encoded recordings assigned to B. The test signals were used to verify that proper track

assignments were maintained throughout the recording process and to test the

performance of the multi-channel recorder. The recorded material on tracks

REF, A, and B was checked after the recordings were complete to verify that

the encoding of the selections matched that of the test key and for the

absense of switching transients during the times taken the Encoder was

switched IN and OUT (the HP 3565S signal processing system and 330 computer

mentioned in section 4.2 were used to perform this analysis).

Equipment Calibration

The same instrumentation and test procedures implemented when recording the

material used in the serial study apply to recording the material used in the

parallel study except with the additional testing of the PCM-3324 multi-channel recorder and the RS-422/TTL digital interface (converter) . The digital

interface was tested using a high-frequency function generator and oscilloscope prior to use. A number of measurements were made using the PCM-3324 to

playback the digital copy of the test signals.

A summary of the results of these measurements follows:

THD+N at 997 HZ: < 0.01%

SMPTE intermodulation (IM) distortion: < 0.02 %

CCIF intermodulation (IM) distortion: < 0.01 %

Noise level re 1-kHz reference voltage: < -90 dB

Crosstalk (non-driven channel re driven) at 1 kHz: < -100 dB

(Note: measurements were made using the CROSSTALK function in the Audio Precision which band- limits the input signal from the non-driven channel centered

about the frequency of the input signal from the driven channel.)

F.14
The Fluke Model 8860 was used to establish the reference voltages present at

the outputs and inputs of the PCM- 1630 and check the gains of the analog input

stages of the PCM-1630 prior to making the direct and encoded recordings. The

output reference voltages were set using the digital multi-channel copy of the

1-kHz reference level recorded from Track 1 of the CD-I test disc. The gains

of the analog outputs of the PCM-1630 were adjusted to a level of +20.0 dB re J0.6 Vrms (corresponding to +20.0 dBm) when terminated by the input impedances

of their respective analog inputs in parallel with the nominal 10 megohm input

impedance of the digital multimeter. When the Encoder was inserted between

the inputs and outputs of the PCM-1630, the voltage levels present at the

inputs and outputs of the Encoder was measured in the same fashion. In all

cases, the variations in voltage levels present at the PCM-1630 outputs due,

for example, to variations in load impedance or drift in the gains of the

amplifiers in the PCM-1630 analog outputs were less than 0.05 dB during the

direct and encoded recordings. The differences between the reference voltage

levels measured at the inputs to the PCM-1630 prior to making the direct and

encoded recordings with the Encoder switched IN or OUT were less than 0.2 dB

.

Digital Error Generation, Correction, and Monitoring

The amount of digital error generation and correction during the playback and

recording of the tapes used in the serial and parallel listening studies was

visually monitored by watching the LED indicators on the DI-5 (digital input)

board installed in the PCM-1630 used for recording, the CRC error and MUTE

indicators on the PCM-1630 audio processors, the LED indicators on the DEC-15

(decoder) boards in the PCM-1630 audio processors, the CRC error indicators on

the TBC (time base corrector) board in the PCM- 3324, and the MUTE, HOLD, CON

(concealment) , COR (correction) , and THR (through) indicators on the DCC

(decoder control) board in the PCM-3324. Channel 1 (the factory default

channel) was selected as the channel to be monitored by the indicators on the

DCC board based on the assumption that, since it is one of the two digital

channels nearest the edges of the tape, it would be most likely to indicate

tape wear by increased error generation and correction (a digital signal af

"0" was recorded on this channel when the tapes were prestriped using the REC

MUTE function in the record mode selection section of the PCM-3324)

.

The status of the LED indicators on the DI-5 board during the copying of the

test material from compact disc to digital audio cassette was with the DI SYNC

indicator illuminated and the other indicators off. Two of the selections

used in the serial study were recorded with digital emphasis. When playing

these selections, the emphasis status indicator on the Denon Model DCD-3300

compact disc player became illuminated, which agreed with the status of the

EMPHASIS indicator on the PCM-1630. There was never any illumination of the

MUTE, ILLG (illegal), CRC (cyclic redundancy check code) error, M EMP (manual

emphasis) , or MONO indicators noticed during the time periods that digital

audio was being copied (the function of the LED indicators was checked by

observing their illumination when powering the PCM-1630).

The correction to the digital data that occurred during playback of the ma- terial recorded onto digital audio cassette was continuously monitored during

the recording process through the use of the LED indicators on the DEC- 15

board in the audio processor used to make the recordings. A decision was made

F.15
not to use the read-after-read function available through the use of the

RAR-1 board; this decision was based on the possibility that the circuitry as- sociated with the confidence head of the DMR-4000 playback machine might not

be perfectly matched to the time delay in the circuitry associated with the

record/play head and thus digital correction using the confidence head might

introduce some audible effect with respect to time offset in corrected and

uncorrected material when the test material was presented in pairs. Up to the

recording of the SI and S2 tapes associated with the serial tests, the digital

errors that occurred (apparently on an intermittent and random basis) were

totally corrected, as indicated by the DEC-15 board, with one exception when

the digital audio data were averaged (interpolated) for an instant. During

the production of the SI and S2 tapes, an intermittent technical fault in the

DMR-4000 recorder developed approximately 2.5 hours after starting the recording associated with each of the tapes SI and S2. This fault may have been due

to an intermittent problem with the SV-24 and/or CF-8 servo-control boards.

When checking the SI and S2 tapes during playback, each of them generated an excessive (continuous) amount of error correction in the DEC-15 board, starting approximately in the middle of each of the tapes (as indicated by the

continuous illumination of the CRC error indicator on the audio processor)

.

The error correction indicated by the status of the illumination of the LED

indicators on the DEC- 15 board in the audio processor was total correction

(only the "Correction", or green, LED was illuminated) during the playback of

the presentations of the test pairs. A representative of the manufacturer of

the recorder and audio processor was contacted and confirmed that even though

continuous error correction was occurring, if total error correction was

indicated by the DEC- 15 board then the errors occurring in the digital audio

data should indeed be completely (totally) corrected. A third DMR-4000

mastering recorder was shipped immediately from Sony to NBS which, upon arriving at NBS, was used to make digitally corrected copies of tapes SI and S2.

The performance of this recorder was checked prior to use with the tracks

recorded from the CBS test disc onto digital audio cassette used to measure

THD+N at 997 Hz, SMPTE IM distortion, and signal- to-noise ratio. The administrator of the serial tests monitored the status of the LED indicators on

the DEC- 15 board installed in the PCM- 1630 used during playback of the recordings prior to the completion of the corrected copies of SI and S2. When

making the digitally corrected copies of tapes SI and S2, four instances of

averaging, and four instances of averaging and/or holding were noted during

the playback of the test pairs by the brief illumination of the "Average"

(yellow) and/or "Hold" (red) LED indicators on the DEC- 15 board (no occurrences of muting, as indicted by the illumination of the MUTE indicator on the

PCM-1630, were noticed or heard). These eight points, along with the other

point at which averaging of the digital data occurred during the production of

tapes SI and S2, were analyzed using the HP 3562A analyzer to compare the time

records of the test pairs during which the averaging and/or holding was

indicated with the time records of those test pairs in the same presentation

in which total digital audio error correction occurred. A cursory analysis of

low-pass filtered signals (6.25 kHz cutoff frequency) was performed by comparing the first time records at all nine of these locations in the digitally

corrected serial tapes for both the left and right channels. (The low-pass

filtering of the time-domain analysis was a result of the anti-aliasing filters of the signal analyzer being set such that the analog input signals to

the analyzer were low-pass filtered at 6.25 kHz, effectively eliminating

F.16
frequencies above 6.25 kHz from the time record.) The results of this analysis indicated that the difference in the amplitudes of the first time record

of the partially corrected audio data as compared with the first time record

of the totally corrected audio data was less than 0.3 dB at any point in the

time record for all nine points in the test tapes in which averaging and/or

holding was noted.

The new (third) DMR-4000 was used to make the digital audio cassette recordings of the material used in the parallel listening study. The error correction during the recording and playback of the material for this study was

monitored by observing the status of the indicators on the DEC- 15 board in the

PCM- 1630, and the DCC and TBC boards in the PCM- 3324. Only total correction

was noticed during any occurrence of error generation, which occurred on an apparently intermittent and random basis. At the end and approximately the

middle of the parallel listening tests, the sensitivity of the muting in the

DCC board was set so that muting would occur when an error was continuous over

20 blocks of data (DIP SW1 switches 1 through 4 OFF -- "minimum sensitivity").

The tracks associated with REF, A, and B were listened to for the duration of

the listening test and no occurrence of audible muting was noted (at the end

of these listening sessions, the muting sensitivity setting on the DCC board

was reset to the factory preset conditions, corresponding to muting occurring

when an error is continuous over 50 blocks of data)

.

F.17
p - . • * '
APPENDIX G

Audio Systems Used for Listening Tests

APPENDIX G: AUDIO SYSTEMS USED FOR LISTENING TESTS

The potential audibility of signal processing schemes intended for copy

prevention is dependent upon the characteristics of the audio systems employed

in the presentation of material encoded by such processing. It is not

possible to address the question of audible effects upon the quality of the

recorded material without using playback systems of high quality.

Since earphones and loudspeakers affect perceived sound quality in ways that

are often quite different, and since some listeners may express a strong

preference for one or the other in critical listening sessions, audio systems

using earphones as well as loudspeakers were available for the serial

listening tests. Earphones were used for the parallel listening tests.

From the measurements that had been performed upon the Encoders and preliminary listening studies by NBS staff members, it was apparent that the principal, potentially audible effects upon material passing through the Encoder

would be attributable to the gain and phase characteristics of the encoding

filter at frequencies near 3840 Hz. It was considered essential that the

frequency response characteristics (in both amplitude and phase) of the audio

playback systems be relatively smooth functions of frequency within and near

this range. Sharp, large peaks and/or dips at these frequencies might

exaggerate or obscure the potentially audible effects of the encoding filter.

Distortion, particularly intermodulation distortion, also needed to be

acceptably representative of high quality audio systems.

Playback Systems

Functional diagrams of the playback systems used during the serial and

parallel presentations of recorded material are shown in figures G-l and G-2.

The compact disc player, synthesizer, and recording equipment used to produce

the recordings for the listening sessions are described in Appendix F. A

typical two -channel -headphone station (two-channel amplifier/polarizing

voltage supply plus headset comprising earphones and the means to hold them on

the head) and the amplifier/loudspeaker system used during the tests are described below. The DMR player (digital master recorder), multi-channel

recorder, converter (digital interface), and the audio processors are the same type used in making the recordings

.

In the case of the serial playback system, the composite digital outputs and

input of the DMR player were connected to the audio processor, and the analog

outputs of the audio processor were connected in an unbalanced configuration

to a two-channel attenuator in parallel with the inputs of the two-channel

power amplifier.

For the parallel presentation, the balanced digital outputs of the multichannel recorder were connected to a custom (not mass produced) switching

network to allow switching between pairs of tracks on the multi-channel

recorder. The balanced digital outputs of the switching network were changed

to unbalanced digital signals by the converter, and the unbalanced digital

outputs of the converter were connected to the audio processor. The analog

outputs of the audio processor were connected in an unbalanced configuration

G.1
HEADPHONE

STATIONS

51

52

LOUDSPEAKER

STATIONS

Figure G-l Equipment setup used for audio playback in serial listening tests.

("D" denotes digitized audio signal. "A” denotes 2-channel

analog audio signal.)

to a two-channel stepped attenuator which allowed the listener to adjust the

volume of the test material during playback. A remote control was provided to

allow the listener to manipulate the tape transport, which was located in a room adjacent to the room used for listening. The meter control (MT-16) board

was removed from the audio processor during the parallel testing in order to

eliminate the action of the peak indicating meters in the audio processor.

The recorder, converter, and audio processor used to present the parallel test

material were the same units used to produce the recordings for the test.

Note that both the serial and parallel tapes were made and reproduced in such

a way that at no time was the listener comparing differences introduced into

the test material by the direct and encoded material having been processed by

two different sets of analog- to-digital or digital- to-analog converters, or analog amplifiers (except those in the Encoder)

.

G.2
Figure G-2 Equipment setup used for audio playback in parallel listening

tests. ("D" denotes digitized audio signal. "A" denotes 2-

channel analog audio signal. "C" denotes control signal.)

Earphones

There are no established national or international standards for measuring the

performance of earphones intended for reproduction of high quality stereophonic recordings. The measurements reported here were performed by placing

the earphones upon an anthropometric manikin known as KEMAR (an acronym for

the Knowles Electronic Manikin for Acoustic Research) [1,2]. It is fitted

with a modified four-branch Zwislocki coupler [3] containing a microphone (B&K

Type 4134) that measures a sound pressure approximating that at the eardrum of

^•Figures in square brackets indicate the literature references at the end

of this Appendix.

G.3
a typical adult as well as the current state of the field permits. The

coupler psed for these measurements was placed in the right side of KEMAR for

measurements with the right earphones of the five headsets used in this study,

and was placed in the left side for the measurements with the left earphones.

This procedure prevented any slight differences due to normal tolerance

differences between couplers from appearing as a difference between

measurements performed on the left and right earphones of a headset. Each of

the five headsets was of the same type: Stax SR-Lambda (Professional) Electrostatic Earspeakers , with the polarizing voltage and audio driving amplifiers for each headset provided by a STAX SRM-l/MK-2 Professional Driver Unit.

Each headset/driver unit was tested as a system, i.e., the "earphone" measurements that are discussed here relate the inputs to the driver unit

(which in the listening systems are the analog outputs of the audio processor,

suitably attenuated) to the output sound pressures at the coupler microphone

in KEMAR.

A Hewlett-Packard 3562A dynamic signal analyzer [4] was used with "burst

chirp" excitation (a sinusoid rapidly swept in frequency) so that the total

chirp duration was well within the time record upon which Fast Fourier Transform processing was performed to determine the amplitude and phase

characteristics of each set of earphones as a function of frequency. Each

such characteristic is defined by data points at frequency intervals of 25 Hz.

Varying the position of a headset upon KEMAR indicated that the smoothest

amplitude -frequency characteristics at and near the notch frequency of the

encoding filter are obtained with the headset pushed as far back toward the

rear of the head as possible and placed as high on the head (although height

was not nearly as significant in effect) as possible. The characteristics of

each headphone station were measured and each station was balanced using this

position on KEMAR. This position was also recommended to listeners at the

start of their listening test sessions.

For each headset, the left and right earphone sensitivities at 800 Hz (a frequency at which relatively replicable measurements could be performed upon

KEMAR) were matched within 0.7 dB. The sensitivities of all the earphones of

four of the headsets were within 0.6 dB of each other. Including the fifth

headset (used only when necessary) , the sensitivities of all the earphones of

all headsets were within 2.4 dB.

Figure G-3 shows the amplitude -frequency characteristics of the left and right

earphones of a typical headphone station. The resonances just above 100 Hz

and the decrease in amplitude as frequency decreases below 100 Hz are considered to be influenced by an artifact due to the use of KEMAR with

earphones of this type. At least some of the differences between earphones at

frequencies above perhaps 4 kHz are attributable to an intentional slight

asymmetry between the left and right ears of KEMAR. Measurements performed

upon KEMAR at these higher frequencies may not be representative of the

performance of the earphones upon an individual adult because of differences

between the manikin and different individuals. The significant aspect of

these measurements is the smoothness of the amplitude -frequency characteristics and the excellent match between left and right earphones near 3840

Hz.

G.4
FREQUENCY (kHz)

Figure G-3 Amplitude -frequency characteristics of the left and right

earphones of a typical headset.

Figure G-4 shows the phase characteristics (relative to an arbitrarily chosen

reference) of the left and right earphones of the game headset, with the

frequencies displayed limited to the range for whi^r artifacts of measurement

are not likely to be a potentially dominant factor. Most significant are the

smoothness of the characteristics and the excellent match between earphones at

frequencies near the encoding filter notch frequency.

Intermodulation distortion was measured using signals specified by the Electronic Industries Association of America (EIA) , which are available on Track

13 of the CBS CD-I Test Disc. This track contains signals for the "Twin-Tone

(CCIF) " method, consisting of simultaneously presented sine-wave signals of

equal amplitude at 11 kHz and 12 kHz, and signals for the "SMPTE" method,

consisting of simultaneously presented sine-wave signals at 60 Hz and 7 kHz

with amplitude ratio 4:1, respectively. The signals from this track were

obtained at the analog outputs of a compact disc player (Sony CDP-610ES) that

was known from prior measurements at these outputs to produce a very low- distortion output from this track of the CD-I Test Disc. The dynamic signal

analyzer (HP 3562A) was used with its source turned off and with the "flattop" window suitable for the accurate determination of the amplitudes of sinewave signals (including distortion components) in the measured amplitudefrequency characteristics.

G.5
PHASE

(deg)

FREQUENCY (kHz)

Figure G-4 Phase characteristics of the left and right earphones of a typical

headset, relative to an arbitrary reference.

For the "Twin-Tone (CCIF)" signal, each distortion component (including the

difference component at 1 kHz) produced by each earphone of the typical

headset was at least 60 dB lower in SPL than the sine-wave signals presented

at 11 kHz and 12 kHz. Considering that each of these signals resulted in a very high SPL (at least 104) , the measured performance of the earphone system

was very good.

For the "SMPTE" signal, each distortion component was typically at least 58 dB

lower in SPL than the sine-wave signal presented at 7 kHz. Considering that

the 7-kHz signal resulted in a high SPL (e.g., 93), and that the sine-wave

signal at 60 Hz was presented to the input of the earphone system at a level

12 dB greater than the level of the 7-kHz signal, the measured earphone system

performance was again very good.

Loudspeakers and Listening Room

A decision was made not to use multiple-driver loudspeaker systems having a crossover between drivers at a frequency between 3 kHz and 5 kHz, to avoid any

confusion between possibly-audible effects of the encoding notch filter and

possibly-audible effects of the speakers' crossover networks. The Energy 22

G.6
Pro Monitor loudspeakers that were used are two-way systems with a single

crossover point, near 1500 Hz, between drivers.

Several types of measurements were performed on these loudspeakers in an

anechoic chamber and in the room used for the listening studies. Measurements

in the large NBS anechoic chamber were performed to determine loudspeaker

system amplitude -frequency and phase characteristics (both axial and nonaxial) , distortion, and the degree of matching between loudspeakers under

precisely controlled conditions. Measurements of room properties and of loudspeaker impulse response and amplitude -frequency characteristics in the room

used for listening studies provided measures of loudspeaker performance in the

listening room, as influenced by the acoustical properties of the listening

room. All measurements of loudspeaker system performance reported here include the

two-channel amplifier (McIntosh MC7270) that was used to drive the loudspeakers in the listening tests, i.e., the amplifier and loudspeakers were

considered as a unit to comprise the device for which properties were being

measured. Previous measurements indicated that this amplifier was performing

well within its manufacturer's published specifications.

Anechoic Chamber Measurements

For the anechoic chamber measurements, a loudspeaker was placed 2.7 m from the

reference microphone Bruel and Kjaer (B&K) Type 4133, followed by a B&K Type

2645 preamplifier, and B&K Type 2607 and 2606 laboratory measuring amplifiers.

The amplified output voltage of the reference microphone system was measured

with a dynamic signal analyzer (HP 3562A) , with the insert-voltage technique

used to measure and to correct for the slight departures from uniform

amplitude- frequency and phase characteristics of the preamplifier-measuring

amplifier combination.

Initial measurements of each loudspeaker system's amplitude -frequency and

phase characteristics and the impulse responses obtained from these characteristics by inverse FFT calculation, both with and without the loudspeaker

protective grille covers in place, showed that removing this cover resulted in

smoother characteristics in the frequency range of interest and a better

impulse response for each loudspeaker. NBS staff members also performed

subjective listening assessments of the loudspeaker systems in the listening

room with music reproduced from compact discs, and concluded that the systems

produced a slightly higher perceived quality of reproduction with these grille

covers removed. The grille covers were removed for all listening tests and

for all measurements reported in this study.

Figures G-5 and G-6 show the amplitude-frequency and phase characteristics,

respectively, measured on-axis, for the loudspeakers in the anechoic chamber,

using a "burst chirp" source and FFT processing in the dynamic signal

analyzer. The lowest frequency shown for these measurements is 100 Hz,

principally because the influence of typical listening rooms upon loudspeaker

response is so dominant at frequencies below a few hundred hertz that there

was little point to performing measurements under anechoic conditions at very

low frequencies. The amplitude -frequency characteristics for the left and

G.7
LU>

LU

LU

DCZ>

CO

CO

LU

DC

Q.

Q

z

IDO

CO

FREQUENCY (kHz)

Figure G-5 Amplitude- frequency characteristics of the two loudspeakers, as measured in the large NBS anechoic chamber.

right loudspeakers are within a few tenths of a decibel of each other at most

frequencies, and both are flat within +4/ -3 dB at all frequencies of

measurement, and flat within ±1 dB at frequencies from 2.7 kHz to 6,0 kHz, the

range that includes the encoding filter notch frequency. The phase

characteristics, plotted on a linear frequency scale so that a pure time delay

would correspond to a straight line, are smooth functions of frequency, and

the characteristics for the left and right loudspeakers are almost identical,

any differences probably being primarily attributable to the delay time

associated with the uncertainty (a few mm) in the positioning of the

loudspeakers in the anechoic chamber.

The amplitude -frequency and phase characteristics were also measured in the

anechoic chamber for each loudspeaker at other angles than axial. Over the

range ±45 deg relative to axial, these characteristics remained relatively

smooth functions of frequency, particularly for frequencies at and near the

notch frequency of the encoding filter.

Intermodulation distortion was measured for each loudspeaker in the anechoic

chamber using the same signal source and dynamic signal analyzer used for the

earphone measurements described previously. For the "Twin-Tone (CCIF)"

signal, each distortion component was found to be at least 56 dB lower than

G.8
FREQUENCY (kHz)

Figure G-6 Phase characteristics of the two loudspeakers, as measured in the

large NBS anechoic chamber, relative to an arbitrary reference.

the level of the closer of the tones at 11 kHz' and 12 kHz. For the " SMPTE"

signal, with the overall sound pressure level being 80 dB (about 89 dB at one meter, a distance often used in such measurements), each distortion component

was at least 40 dB lower than the level of the sine-wave signal presented at 7 kHz

.

Listening Room Measurements

The listening room had a rectangular floor plan, 4.9 m x 6.7 m, with a drop

ceiling 2 . 44 m from the floor. The floor was plush carpet and carpet pad over

a concrete slab on grade. The walls were concrete block with a hard, smooth

finish. Two windows (double glazed) on one long dimension of the room were

covered with rather lightweight curtains. The loudspeakers were placed upon

solid concrete blocks near one short wall of the room, and a rubber pad was

placed beneath the front end of each loudspeaker , so that a normal to the

front of the loudspeaker at the center of the high-frequency driver formed an angle of approximately 5 degrees to the floor. These normals then passed at

the approximate ear level of two listeners seated side by side with their ear

positions about 2.. 7 m from the line joining the front surfaces of the

loudspeakers. The listeners were centered between the two loudspeakers, so that each was as nearly equidistant from the speakers as possible.

G.9
The room was treated acoustically to be relatively sound absorptive in the end

near the loudspeakers, and diffusely reflective in the end behind the

listeners. The intent was to provide an interval of approximately 10

milliseconds after the arrival of directly radiated sound from the loudspeakers in which reflections from interior surfaces of the room would be

minimal in amplitude, so that the localization cues contained in this directly

radiated sound during the reproduction of stereo recordings would not be

obscured by these early reflections from room surfaces. Subsequent reflections were intended to be relatively diffuse, so that glaring reflections of

transient sounds and severe standing wave patterns excited by sustained notes

could be avoided.

The acoustical treatment of the room included the use of diffusing panels

("QRD Diffusors" manufactured by RPG Diffusor Systems, Inc.) and broadbandwidth sound absorptive units ( "Abffusors " from the same source;, as well

as one- inch and two- inch thick fiberglass panels. Diffusing panels were

placed along the short walls of the room, and the "Abffusors" were placed

along the long walls from the diffusors behind the loudspeakers, to the approximate position of the listeners. Either "Abffusors" or two- inch thick

fiberglass panels were mounted as ceiling tiles in positions chosen to prevent

specular reflections of directly radiated sound from the loudspeakers off the

ceiling to the listeners. A fiberglass panel was placed on the floor in front

of each loudspeaker, with the end farthest from the loudspeaker elevated by a concrete block, to reduce specular reflections from the floor. The diffusing

panels behind the loudspeakers, as well as most of the remaining wall surface

above these panels, were covered with fiberglass panels to further improve the

perceived quality of sound reproduction, and, in particular, the ability to

distinguish individual instruments and/or individual notes in rapidly ar- ticulated musical passages. Additional fiberglass panels were placed just

above the loudspeakers and within the room as needed to attempt to eliminate

flutter echoes between parallel surfaces, annoying standing wave patterns, and

potentially deleterious specular reflections. The final configuration of the

room is shown in figures G-7 and G-8.

The efficacy of the acoustical treatment was checked by using a calibrated

microphone system (B&K Type 4133 microphone, Type 2645 preamplifier, and Type

2607 measuring amplifier) and an audio spectrum analyzer (Techron Division of

Crown International, Model TEF-10) to measure the temporal pattern of directly

radiated and reflected sound, with the microphone at the listening position.

These and other measurements verified that the intended results had been

achieved.

Reverberation times were measured in octave bands using "pink noise" produced

by a sound power source (B&K Type 4205) placed between the two loudspeaker

systems so that it was equally distant from each, but 15 cm farther from the

listening position than a line joining the front faces of the loudspeakers.

The receiving microphone was placed at the listening position, at a point

between the positions of the listeners, but in the absence of the listeners.

The receiving microphone system (B&K Type 4166 microphone, Type 2639

preamplifier, Type 2607 measuring amplifier, and Type 1617 bandpass filter)

output was recorded on a level recorder (B&K Type 2305) to produce paper

strip-chart type records from which the reverberation times for various octave

G.10
G.11

Figure

G-7

The

room

used

for

the

serial

listening

tests.

The

two

subjects

are

shown

facing

the

loudspeakers

and

the

absorptive

material

used

to

minimize

reflections

from

the

diffusors

and

wall

behind

the

speakers

and

from

the

side

walls.
G.12

Figure

G-8

Diffusing

panels

at

the

back

of

the

room

used

for

the

serial

listening

tests.
bands could be determined. For the octave band centered at 125 Hz, the

reverberation time was about 0.3 second. For each of the octave bands centered

at 250 Hz through 8 kHz, the reverberation time was about 0.2 second.

The critical distance (the distance, from the sound source, at which the sound

pressure level due to directly radiated sound is equal to the sound pressure

level due to reflected sound) was determined to be about 1.8 m at frequencies

near the center frequency of the encoding notch filter. Thus, the listeners

received a substantial contribution of directly radiated sound, which is

considered helpful for careful monitoring sessions in which it is intended to

perform critical judgments that are not excessively influenced by the

particular acoustical characteristics of the listening room. An additional measurement of loudspeaker/room behavior was performed on each

loudspeaker sequentially by using a dynamic signal analyzer (HP 3561A) to

generate an audio -frequency-band- limited impulse [5]

,

which was connected to

the loudspeaker system. The signal analyzer was then triggered with the

appropriate delay parameter setting to capture the first 10 msec of the

impulse response of the loudspeaker (measured at the listening position using

the same microphone system that had been used for the measurements in the

anechoic chamber) as a low-pass-filtered (for anti-aliasing purposes)

,

digitized time record that could be displayed and plotted as an analog

waveform. This waveform (with uniform window weighting) was subjected to FFT

processing to obtain the amplitude -frequency characteristic of the

loudspeaker. Since this characteristic includes not only the direct sound

radiation from the loudspeaker, but also the reflections from room interior

surfaces occurring within the first 10 msec of the impulse, it is a measure of

the loudspeaker/room-reflection amplitude -frequency characteristic that should

be less smooth than, but essentially similar to, the corresponding anechoic

response at frequencies above about 1 kHz, where the acoustical treatment of

the room is considered particularly effective.

Figure G-9 shows the time record waveforms measured as described above for the

left and right loudspeakers in response to the source waveform (which also

exhibited a dominant initial negative peak) . Reflections are relatively small

in amplitude and are essentially negligible by the end of the time record.

Figure G-10 shows the measured amplitude- frequency characteristics obtained by

FFT analysis of the time record waveforms for the left and right loudspeakers

in the listening room. The characteristics are well-matched, flat to within

±4 dB from 2.9 kHz to 5.9 kHz, and are even better at frequencies near the

encoding notch frequency. The characteristics are, as expected, somewhat

roughened by room effects, but are comparable to the amplitude-frequency

characteristics (figure G-5) measured in the anechoic chamber at corresponding

frequencies

.

A measurement of the loudspeaker/room behavior that includes the effects of

all room reflections in response to a continuously applied broad-band signal

was also performed. Each loudspeaker was excited sequentially (i.e., first

one loudspeaker alone, then the other alone) by the "pink noise" output of a generator (GenRad 1382) of (statistically) stationary noise, and the

loudspeaker amplitude -frequency characteristic was measured in

G.13
1

>

hD

CL

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>-CO

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1

CLO

tro

0

-1

-2

-3

1

0

-1

~2

-3

10.0 13.2 16.4 19.6 22.8 26.0

TIME (ms)

Figure G-9 Impulse response of the two loudspeakers as measured in the

listening room.

one- third-octave bands using the analyzer in its synthesized one -thirdoctave-bands mode [5], with the microphone system (B&K Type 4134 microphone,

Type 2639 preamplifier, and Type 2607 measuring amplifier) determining the

sound pressure level at the listening position. Ideally, these measurements

would result in essentially "flat" (uniform with frequency)

amplitude-frequency characteristics (but note that controlled departures from

a flat characteristic are also often advocated, e.g., a characteristic that

gradually decreases in amplitude with increasing frequency to simulate the

sound received at many good seats in typical concert halls) . With the same level of noise presented sequentially to each loudspeaker, these characteristics for the left and right loudspeakers at the one- third-octave band

centered at 4 kHz (containing the notch frequency of the encoding filter) were

equal to the nearest 0.1 dB. Both characteristics were also within +3/-4 dB

of their value at 4 kHz for all one- third-octave bands centered at frequencies

from 315 Hz to 12.5 kHz, and within +7/-S.5 dB of their values at 4 kHz for

all bands from 30 Hz to 20 kHz.

The spatial uniformity of the sound field produced sequentially by each

loudspeaker within the listening region (range of possible ear positions of

G.14
Figure G-10 Amplitude- frequency characteristic of the two loudspeakers as determined in the listening room by FFT analysis of time record

waveforms

.

the listeners in the absence of the listeners) in response to "pink noise" was measured with a sound level meter and octave-band filter (B&K Type 2204 with

Type 4166 microphone and Type 1613 Octave Filter Set). Using the C-frequencyweighting network (but no additional filtering) and the "slow" exponential

averaging time on this instrument, the difference between the highest and

lowest sound pressure levels measured throughout the listening region was 0.8

dB and 1 dB for the left and right loudspeakers, respectively. Using the

octave-band- filter (but no other frequency-weighting network) and "slow"

exponential averaging time, these differences, for either the left or right

loudspeaker, at each indicated center frequency of the octave band filter,

were: 0.7 dB for the 1-kHz octave band and 0.5 dB for the 4-kHz band.

A somewhat more demanding test of spatial uniformity was performed by exciting

each loudspeaker sequentially with band- limited periodic noise, the spectrum

of which consisted of lines of essentially equal amplitude spaced 10 Hz apart

in frequency throughout the range 2 kHz to 6 kHz (the periodic noise output of

the HP 3561A analyzer served as the basic source) . This signal would excite

specific standing waves in the room much more effectively than would the pink

noise, which is random in character. Using the same sound level meter

described above with the C- frequency-weighting network and "fast" exponential

G.15
averaging time, the difference between the highest and the lowest sound

pressure levels measured throughout the listening area was 1.25 dB for each

loudspeaker

.

Care was taken to reduce the background noise levels in the listening room as much as practical. For example, the intake/exhaust air flows of the HVAG

system for the room were balanced and unnecessary gratings in the air flow

path were removed to reduce flow noise. The hot water flow to heat exchangers

in the room was turned off during the listening tests and electrical

resistance heaters were installed* The background noise level was then

measured using a microphone system (B&K Type 4166 microphone, Type 2639

preamplifier, Type 2607 measuring amplifier), with the microphone at the

listening position, and a dynamic signal analyzer (HP 3561A) . The overall Afrequency-weighted sound pressure level was 32 to 33 dB. A one- third-octaveband analysis showed that most of this noise was contributed by frequency

components below 250 Hz. Measurement of the unweighted one- third octave band

levels showed that, for bands centered at frequencies greater than or equal to

750 Hz, the measured band levels were lower than the corresponding permissible

one- third octave band levels [6] for ambient noise during audiometric testing

using earphones mounted in the MX-41/AR cushion. For example, the permissible

sound pressure level in the band centered at 4 kHz is 37 dB, and the

corresponding level measured in the listening room is 6 dB . In other words,

for frequencies near the encoding notch frequency, the listening room was

quieter than an audiometric test booth suitable for determining audiometric

thresholds of hearing with such earphones and cushions.

Measurements of the maximum C-weighted sound pressure level produced at the

listening position by each of the musical selections presented in the serial

listening tests were performed with the sound level meter set to the fast

exponential averaging time. The highest such level was 88 dB; the lowest was

64 dB.

References for Appendix G

[1] Burkhard M. D. , Sachs R. M. Anthropometric manikin for acoustic

research. J. Acoust. Soc. Am. 1975; 58:219-22

[2] Burkhard M. D. , ed. Manikin measurements. Elk Grove Village: Industrial

Research Products, Inc., 1978.

[3] American National Standard For an Occluded Ear Simulator, ANSI S3. 25-1979

(ASA 39-1979) , American Institute of Physics for the Acoustical Society

of America, New York.

[4] Atkinson E. S., Wahl G. L. Jr., Hall M. L. , Wicklund E. J., Peterson S.

K. Low-frequency analyzer combines measurement capability with modeling

and analysis tools. Hewlett-Packard Journal 1987; 38 No. 1:4-16.

[5] Epstein J. S. Versatile instrument simplifies dynamic signal analysis at

low frequencies. Hewlett-Packard Journal 1984; 35 No. 12:4-11.

G.16
[6] Criteria for Permissible Ambient Noise during Audiometric Testing, ANSI

S3. 1-1977, American Institute of Physics for the Acoustical Society of

America, New York.

G.17
.

.
APPENDIX H

Listener Instructions

APPENDIX H: LISTENER INSTRUCTIONS

Serial Presentation (presented aurally)

Thank you for your cooperation and your willingness to participate in these

tests. We hope that you will find the tests both enjoyable and instructive.

We realize that you are giving up prime family and work time to take part in

the tests. We truly appreciate your contribution.

In order to ensure impartiality of the results, your experimenter is ignorant

of the test key. When the tests are completed, we will break the code and

notify you about your performance and that of the entire group. To obtain

this report, please fill out your mailing address on the blank card provided

at the end of the testing session.

Now please enter your name, the number of your listening station, and the date

and time on your answer sheet. [wait 20 seconds]

For these tests, we either directly copied recorded music -- or we copied

recordings through an Encoder which may alter signals in the frequency range

near 3840 hertz which, musically speaking, falls between B-flaty and B 7

. The

frequency near 3840 hertz is near the top end of the piano and sounds like

this. [wait 2 seconds] This frequency is above the fundamental frequency of

most, but not all, musical instruments. Therefore, the Encoder will primarily

operate upon the harmonics of lower fundamental frequencies, and thus may

modify the timbre or coloration of the music.

In any given announced test, the same musical selection will be presented in

pairs, with the two members of each pair either being the Same (that is, both

direct or both encoded) or being Different (that is, one member of the pair

being a direct reproduction and the other member being encoded) . Each of the

pairings will be repeated three times. You will then be asked to respond by

encircling either S, for Same . or D, for Different . on your answer sheet. If

you wish to change your answer, either erase or cross out your initial

response and respond anew.

We will now run some practice pairs to acquaint you with the Encoder and the

test procedure. The differences in the practice pairs may be more obvious

than the differences within the later test pairs. Hopefully, the practice

pairs will assist you in orienting your listening.

Here is practice pair A [wait 2 seconds] . Encircle either S or D in slot A

depending upon whether you think the two members of the pair are the same or different. [wait 5 seconds]

.

[Repeat the above for practice pairs B, C, and D.]

The test administrator will now stop the tape to afford you an opportunity to

ask any questions you may have about the test procedure.

We now will begin the actual testing. Please be ready to listen to the first

test pair.

H.1
Here is test pair 1. [wait 2 seconds]. Encircle in slot 1 either S or D,

depending upon whether you think the two members of the pair are the same or different. [wait 5 seconds]

Here is test pair 2 [wait 2 seconds] . Encircle in slot 2 either S or D.

[Continued for all 96 test pairs, with three short breaks.]

Parallel Presentation (read by subjects)

Thank you for your willingness to serve as a listener in this study. You have

been asked to participate as a result of your excellent background in music

and/or audio recording. We welcome and respect your expertise in careful and

critical listening.

You have probably read in the newspapers or in the technical press that the

Congress of the United States is considering legislation concerning a system

that is intended to prevent the unauthorized copying of audio materials and

that Congress has asked the National Bureau of Standards (NBS) to evaluate

this system. This system, otherwise referred to as the "Copy-Code" system,

consists essentially of two parts: (1) an Encoder designed to encode the two

analog signals of a stereo recording by eliminating a small portion of the

audio spectrum, thereby creating a "notch" in the recorded material, and (2) a detector that is designed to detect the presence of the "notch" in encoded

material. The purpose of this listening study in which you are participating

is to try to determine if the encoding process makes an audible difference in

recorded material and, if so, to what degree.

You will be asked to listen to ten musical selections that were transferred

digitally from commercially available compact discs to two tracks of a multitrack digital tape recorder. These two tracks were then copied on to six

other tracks -- two of which constitute a reference (REF) of non-encoded

(direct) material and four of which comprise two groups (A and B) of two

tracks each, where either of the groups, A or B, may be direct or encoded

material. Your task is to determine for each of the selections which group, A

or B, is encoded material, i.e., is different from the reference track and the

other group. The material in the given groups remains either direct or en- coded throughout an entire selection but is subject to change from selection

to selection in a random fashion. Further, for any given selection the ma- terial contained in the two groups A and B is always different, i.e., if for

any given selection one group is encoded material, the other group is always

direct material. You have been provided with a set of switches that enable

you to switch between any of the groups of tracks: REF, A, and B. Each of the

selections is preceded and followed by an announcement identifying the beginning and end, respectively, of the selection. The announcements are never

encoded in either group A or group B regardless of the nature of the material

preceding or following the announcements. The following is a table of example

selections giving the nature of material in the three groups of tracks and the

correct response relating to them.

H.2
Selection No.

N

N+l

Switch Position

A REF B

Correct

Response

Encoded

Direct

Direct

Direct

Direct

Encoded

A

B

Please circle the appropriate response on your answer sheet for each of the

ten musical selections.

In order to preserve the levels of the original recordings, no attempt was

made to compress the overall dynamic range contained in the recordings of the

selected material. As a result, the maximum levels from selection to selection vary over a range of approximately 30 to 35 decibels. In order to provide you with adequate volume to listen to the selections recorded at relatively low levels, it was necessary that the selections recorded at relatively

high levels be loud given the same relative volume control setting. There are

two volume controls, one on the headphone amplifier, and one in a separate box

adjacent to it. Please do not touch the volume control on the headphone

amplifier, but do adjust the level of the recorded material to your liking

using the attenuator in the separate chassis box. Be careful, however, to

return this control to mid-position between selections, particularly after

listening to a low-level recording. If possible, we ask that you record the

setting of the attenuator during the time period that you evaluated each of

the selections so that we have an idea of the nominal sound pressure level at

which you listened to the recordings. The following is a list of the ten

selections and the times, in minutes and seconds, at which they begin on the

tape. Also noted is whether the level at which the material was recorded is

relatively high or low (loud, soft, or average).

Selection

No.

Level Description Tape Time

1 Average Bizet-Sarasete : Carmen Fantasy 4:59

2 Loud Streisand: Broadway Album 6:05

3 Soft Beethoven: Symphony No. 9, Op. 125 7:55

4 Loud Copeland: Fanfare for the Common Man 8:51

5 Loud Pat Metheny: American Garage 10:11

6 Loud Prokofiev: Alexander Nevsky Op. 78 10:52

7 Soft Berstein: West Side Story 11:40

8 Average Messian: Turangalila Symphony 12:55 . 9 Loud J. S. Bach: Inventionen und Sinfonien 13:49

10 Soft Respighi: Fountains of Rome 14:39

To enable you to listen to the recordings , you have been provided with a set

of headphones. The acoustic frequency response of these headphones is dependent on the position of the phones on your head, particularly in the frequency

region of 4000 Hz . Based on measurements made at NBS , we recommend that the

headphones be positioned with the inner edge of the front of the ear cushions

next to the pinnas of your ears, i.e., that the headphones be positioned

towards the rear of your head. Also, be sure that the phones designated by

"L" and "R" cover your left and right ears, respectively.

H.3
To assist you in evaluating the recorded material, you have been provided with

a remote control unit to operate the tape recorder. This unit is capable of a number of functions, the operational details of which will be provided to you

by the test administrator. A brief description of some of the features of the

remote control unit follows. The remote control has two time displays: TAPE

TIME, which indicates the actual tape position, and LOCATE TIME, which indicates the current time data to be used for a number of different functions as

a destination address. These times may be used in either an absolute or relative mode with the mode indicated by the ABS TIME key/indicator. The tape

times given for the selections are absolute and it will probably be less

confusing to use the remote control in this time mode. The unit is capable of

storing and recalling a total of 100 cue points which can be recalled and used

as destination addresses for automatic operation of the tape transport. Cue

registers 01 thru 10 contain the start times of selections 1 thru 10, respectively, as given in the table above. Destination addresses, or cue points,

may be entered into a cue register either by using the numeric key pad and the

STO key or by using the CUE STO key which stores the TAPE TIME at the time at

which the key was pressed even if the transport is in the PLAY mode. Caution:

when the CUE STO key is pressed the current cue register is incremented by one and the cue point in this register is overwritten by the new cue point. The

cue register numbers are displayed in the CUE displays located next to the

TAPE and LOCATE TIME displays. LOCATE TIMES, or destination addresses, may be

expressed to the nearest second, or nearest millisecond, by pressing the mS/S

key. Data may be entered into the LOCATE TIME display by recalling cue register contents, by pressing the CUE STO key, or by pressing the down shift key

which transfers the data in the TAPE TIME display to the LOCATE TIME display.

Three functions may be of use to you in operating the tape transport. Pressing the LOC key starts the locate operation which moves the tape to the destination address indicated by the LOCATE TIME display. Pressing the RLB key

starts the roll back operation which moves the tape to the destination address

indicated by the LOCATE TIME display less a preset PREROLL time (the default

PREROLL time is two seconds but is easily changed using the numeric key pad

and the STO and PRE keys) . If the PLAY key is pressed at the same time as the

LOC or RLB key is pressed, playback will automatically begin when the tape is

located at the point defined by the operation. The RTN key starts the return,

or loop, operation which moves the tape to a preset start point and then sets

the transport into the PLAY mode until reaching a preset end point when it

returns the tape to the start point and stops the transport (the start and end

points are preset using the REP SET key to indirectly define destination

addresses by cue register numbers which appear in the two CUE displays next to

the time displays) . If the PLAY key is pressed at the same time as the RTN

key is pressed, the transport enters the REPEAT mode during which the transport continuously repeats playback of the tape between the start and end

points. (Press one of the transport keys to stop a REPEAT operation.) Note:

for your information the LEDs below the display should indicate the following

conditions during playback: Fs : 44.1 kHz; STATUS: VARI-SPEED - off, MASTER

SAFE -on, ALARM - off (except during selection no. 1 due to the emphasis

status of the final recording differing from that of the original compact

disc); REC MODE: INSERT. In general, you should ignore the status of these

indicators and concentrate on listening to the recorded material.

H.4
We also are interested in receiving your comments on how you thought the

encoded and direct material differed for each of the selections. A dictating

machine is available for you to record any such comments that you wish to make

as well as to record specific locations on the tape (TAPE or LOCATE TIMES)

where the action of the encoding process was most evident to you.

You may wish to use the following information as a guide to aid you in

ascertaining differences in the direct versus encoded recordings. The Encoder

used in encoding the selections contains two narrow band-reject, or band- stop,

filters each of which is simultaneously introduced into the analog signal path

of one of the two channels of a stereo recording (with the filters assigned to

separate channels) . The filters are centered nominally about the frequency of

3840 Hz and are dynamic in the sense that the Encoder may activate or deactivate the filters in such a way as to either introduce them into, or remove

them from, the signal paths of the two audio channels being encoded. The

frequency of 3840 Hz is above the fundamental frequency range of most, but not

all (e.g., the piccolo), musical instruments. Consequently, during most of

the passages of encoded material, the filters of the Encoder, when introduced

into the signal path, will affect the harmonics or overtones of lower fundamentals such as those produced by the human voice and by instruments such as the violin, trumpet, triangle, and bell. In the case of percussive instruments that have a broad spectral content such as the snare drum and cymbal

,

the apparent timbre, pitch, or loudness of an instrument may be altered by the

encoding process. Also, the material that has passed through the Encoder,

i.e., the "encoded material," may have been perceptibly altered by the Encoder

during the time periods in the encoding process when the Encoder was introducing the filters into, or removing the filters from, the signal paths of the

two audio channels. In any case, the differences in the encoded and direct

material tend to be subtle rather than blatant and are likely to require

careful, concentrated listening to detect differences between the two (for

some of the selections it may be very difficult, if not impossible, to detect

differences) . A spectral analysis of the recorded material was performed

after the recordings were completed and in all cases the analysis of the en- coded material indicated that the filters of the Encoder were present in the

signal path of the two audio channels for at least a portion of the time

during the encoding of each of the selections. Since the recordings were made

in such a way as to ensure that no perceptible differences were introduced

into the direct versus encoded material by the recording process itself, you

should concentrate on listening for differences between the direct and encoded

material due to the filtering action of the Encoder rather than for differences in the material due to such things as variations in fade in and fade out

of selections or low-frequency noise, relative time delays or lack of synchronization, etc. Also, we recommend for each of the selections that you listen

to the entire selection that you are evaluating on each of the groups of

tracks, REF, A, and B, before attempting to discern differences in the ma- terial by switching between them. (Note: a slight switching transient or "click" may be heard when switching between positions A, Ref, and B. This is an artifact of switching the digital audio output of the tape recorder and not

due to differences in the recorded material)

.

H.5
'

-t

.

.

.

'

.
APPENDIX I Schematic Diagrams of Defeat Circuits

Table 1-1

Circuits Used in Each of the Defeat Methods

Circuit Defeat Method

1 2 3 4 5 Noise Source • •

3.8 kHz Band Pass Filter • • • • •

Summer • • • • •(2)

1.9 kHz Band Pass Filter • • •

Full Wave Rectifier • • •

2.7 kHz Band Pass Filter • •

Modulator/Multiplier/VC Amplifier • • •

Full Wave Rect. and LP Filter • • •

3.5 kHz Band Pass Filter •

4.3 kHz Band Pass Filter •

1.1
NOISE SOURCE

3.8 kHz BAND PASS FILTER

1.2
SUMMER

INPUT A

INPUT B

INPUT C

22k

RESISTANCE VALUES

ARE IN OHMS

1.9 kHz BAND PASS FILTER

FULL WAVE RECTIFIER

1.3
NBS-114A (REV- 2-80

U.S. DEPT. OF COMM.

BIBLIOGRAPHIC DATA

SHEET (See instructions)

4. TITLE AND SUBTITLE

1. PUBLICATION OR

REPORT NO.

2. Performing Organ. Report No. 3. Publication Date NBSIR-88/3725 February 1988

Evaluation of a Copy Prevention Method for Digital Audio Tape Systems

5c AUTHOR(S) B.A. Bell and G.N. Stenbakken, D.R. Flynn, D.J.

V. Nedzelnitsky , and K.R. Eberhardt

6. PERFORMING ORGANIZATION (If joint or other than NBS, see in stru ction s)

Evans, E.D. Burnett

7. Contract/Grant No. national bureau of standards

DEPARTMENT OF COMMERCE

WASHINGTON, D.C. 20234

8 . Type of Report & Period Covered

9. SPONSORING ORGANIZATION NAME AND COMPLETE ADDRESS (Street. City. State. ZIP)

10.

SUPPLEMENTARY NOTES

Document describes a computer program; SF-185, FlPS Software Summary, is attached.

11.

ABSTRACT (A 200-word or less factual summary of most significant information. If document includes a significant

bi bliography or literature survey, mention it here)

The National Bureau of Standards in response to requests from the U.S. Congress tested

a system designed to prevent unauthorized copying by digital audio tape (DAT)

recorders of suitably encoded audio recordings. The system, designed by CBS Records,

filters out a narrow range of frequencies from the spectrum of the original sound in

the region of 3840 Hz, thereby encoding the material with a "notch 'in the frequency

spectrum so that a DAT recorder equipped with the system’s decoding circuitry can

sense the presence of a prescribed notch in the spectrum and inhibit recording.

The congressional questions and the NBS conclusions are:

1 . Does the copy prevention system achieve its purpose?

NBS Conclusion : The system does not achieve its stated purpose.

2 „ Does the system diminish the quality of the prerecorded material into which the

notch is inserted?

NBS Conclusion : The system's encoder alters the original electrical signal. For

some listeners for some selections, this results in a discernible difference between

prerecorded notched and unnotched material.

3 . Can the system be bypassed and, if so, how easily?

NBS Conclusion: The copy prevention system can be bypassed easily.

12.

KEY WORDS (Six to twelve entries; alphabetical order; capitalize only proper names; and separate key words by semicolon s) audio recording; copy protection; digital recording; music; sound; tape recording

13. AVAILABILITY

[X~l Unl imited

| | For Official Distribution. Do Not Release to NTIS

Order From Superintendent of Documents, U.S. Government Printing Office, Washington, D.C. 20402.

£X] Order From National Technical Information Service (NTIS), Springfield, VA. 22161

14. NO. OF

PRINTED PAGES

205

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