USBM Contract No. H0346045 

TECHNICAL SERVICES FOR MINE 

COMMlTNI CATIONS RESEARCH 

APPLICABILITY OF STATE-OF-THE-ARli7 

VOICE BANDWIDTH COMPRESSION TECHNIQUES 

FOR WIRELESS MINE COMMUNICATION 

Richard He Spencer -- Task Leader 

Warren G. Bender 

ARTHUR D. LITTLE, INC. 

Cambridge, Massachusetts 02140 

C-77229 

The views and conclusions contained in this document 

are those of the authors and should not be interpreted 

as necessarily representing the official policies or 

recommendations of the Interior ~epartment 's Bureau of 

Mines or of the U.S. Government. 

USBM CONTRACT REPORT -- TASK A (Contract No. H0346045) 

TASK ORDER NO. 1 

July 1975 

UNITED STATES 

DEPARTMliSNT OF THE INTERIOR 

BUREAU OF MINES 

Arthur D Little, fnc. 
This report was prepared by Arthur D. Little, Inc., Cambridge, Massachusetts under USBM Contract No. H0346045. The contract was initiated 

under the Coal Mine Health and Safety Program, It was administered under 

the technical direction of the Pittsburgh Mining and Safety Research 

Center with Mr. Howard E. Parkinson acting as the technical project 

officer. Mr. Michael W. College was the contract administrator for the 

Bureau of Mines. 

This report is a summary of the work recently completed as part of this 

contract during the period May 1974 to July 1975. This report was submitted by the authors in July 1975. 

No inventions or patents were developed and no applications for inventions or patents are pending. 

Arthur D Little, lnc 
TABLE OF CONTENTS 

LIST OF TABLES 

LIST OF FIGUWS 

I. EXECUTIVE SUMMARY 

A. INTRODUCTION 

B. FINDINGS 

1. Real-Time Techniques 

2. Non-Real-Time Techniques 

C. CONCLUSIONS 

D. RECOMMENDATIONS 

11. THE NATURE OF VOICE SIGNALS 

111. METHODS OF VOICE BANDWIDTH COMPRESSION 

A. MODIFICATION OF SPEECH SIGNALS 

1. Band Limiting 

2. Time Assignment Speech Interpolation 

3. Temporally Interrupted Speech 

4. Clipping 

5. Frequency Division (Stretched-out Voice) 

B. ANALYSIS-SNYTHESIS TECHNIQUES - VOCODERS 

1. Spectrum Channel Vocoder 

2. Voice-Excited Vocoders 

3. Cepstrum Vocoder 

4. Harmonic Compression 

5. Autocorrelation Vocoder 

6. Phoneme Vocoder 

7. Adaptive predictive Coding 

8. Pattern Matching Vocoder 

9. Formant Vocoder 

Page 

vi 

vii 

iii 

Arthur D Little, Inc 
TABLE OF CONTENTS (Continued) 

Page 

IV. FUNDAMENTAL LIMITS TO BANDWIDTH COMPRESSION 

V. DISCUSSION OF BANDWIDTH COMPRESSION TECHNIQUES 

A, ADAPTIVE PREDICTIVE: CODING 

B0 SPECTRUM CHANNEL VOCODER 

C, FORMANT VOCODERS 

D, BASEBAND VOICE 

E.,. BAND LIMITING 

Fo TWORALLY INTERRUPTED SPEECH 

Go FREQUENCY DIVISION (STRETCHED-OUT VOICE) 

Ho HARMONIC COMPRESSION 

I. CLIPPING 

J. PHONW CODER 

K. PATTERN CODING 

Lo COMPANDING 

VI . COMPARISON OF SYSTEMS 

A. METHODS OF COMPARISON 

B. DISCUSSION OF COMPARISON PLOT 

VII. TIME-DELAYED SPEECH 

VIII. REFERENCES 

APPENDIX I NARROW BAND VOICE USING FREQUENCY SLICING 

APPENDIX 11 ALTERNATIVES TO VOICE BANDWIDTH COMPRESSION 

SYSTEMS 

A. LIMITED SYMBOL NARROW-BAND CODE SYSTEM 

B. MORSE CODE-TYPE SYSTEMS 

iv 

Arthur D Little, lnc. 
LIST OF TABLES 

Table No. 

1 Comparison Matrix 

1-1 Twenty Frequency Bands of Equal Contribution to 

Speech Intelligibility 

Page 

30 

52 

Arthur D Little, lnc 
TABLE OF CONTENTS (Continued) 

C. AN ALPWIBET SYSTEM WITH KEYBOARD 

D. FLAG SIGNAL SYSTEMS 

E. PANTOGRAPH SYSTEM 

F. THE ELECTRONIC PAD 

APPENDIX 111 WHISPERED SPEECH 

APPENDIX IV THE NATURE OF TRANSMISSION CKANMELS 63 

A. TRANSMISSION OF ANALOG SIGNALS ON DIGITAL, 63 

CHANNELS - PCM 

Bo TRANSMISSION OF DIGITAL SIGNALS ON ANALOG 67 

C-LS - MODEMS 

C. BPS PER HERTZ IN DIGITAL TRANSNISSION ON 68 

AN ANALOG C 

D. HERTZ PER BPS IN ANALOG TRANSMISSION ON 69 

DIGITAL CHANNELS 

Eo CHANNEL CAPACITY RELATIONSHIPS 73 

APPENDIX V LINCOMPEX 77 

BIBLIOGRAPHY 81 

Arthur D Little. Inc 
LIST OF FIGURES (Continued) 

Figure No. 

IV-6 channel' Capacity As Function of Bandwidth for 

Fixed Signal Power in Presence of Flat Noise 

V-1 Simplified Block Diagram of Lincompex System 

Page 

76 

78 

Arthur D Little, Inc 
Figure No. 

LIST OF FIGURES 

Page 

Energy Density Spectrum for Male Speech 8 

Energy and Articulation for Speech Signals with 10 

Low-Pass Filter 

Energy and Articulation for Speech Signals with 11 

High-Pass Filter 

Band Limiting Speech Signal 14 

Spectrum Channel Vocoder 

Voice-Excited Vocoder 

Harmonic Compressor 21 

Autocorrelation Vocoder 

Formant Vocoder 

Articulation Index as a Function of Signal to 39 

Noise Ratio 

Comparison of Compression Techniques 41 

Formats for Reconstruction of Stretched-Out 

Messages at 6 to 1 Stretch-Out Ratio 

Relation Between A1 and Various Measures of 53 

Speech Intelligibility 

Articulation Index for Various Frequency Sliced 55 

Systems as Function of Signal Power 

Handheld Keyboard and Display 58 

PCM Quantizing Noise 66 

~its/~econd/Hertz Vs Signall~oise 70 

Differential Quantizing at 56kbps. 

l6kbps DCDM Performance 

Relative Power Required as Function of Bandwidth 75 

for Fixed Channel Capacity in Presence of Flat Noise 

vii 

Arthur D Little, Inc 
I, EXECUTIVE STJMMARY 

A. INTRODUCTION 

Since the passage of the Coal Mine Health and Safety Act of 1969, 

the Bureau of Mines has made many advances in mine communications. These 

efforts have resulted in the following developments and demonstrations: 

Surface-to-surface electromagnetic transmission of baseband 

voice signals through overburdens as deep as 600 feet. This 

transmission was achieved with equipment from the Interim 

Mine Rescue and Survival System Program, and demonstrated 

that baseband voice communications could be obtained through 

these overburdens in the absence of mine-generated noise; 

that is, when the mine power system was shut off, However, 

during mine operational conditions, the mine-generated noise 

proved to be so large that effective baseband voice communications to the desired depths could not be effected. 

r The development of beacon transmitters for locating trapped 

miners. This work was again the outgrowth of the Interim 

Mine Rescue and Survival System. Extensions of this work 

have led to reliable detection and location of trapped 

miners to depths of 1000 feet. The system, which makes use 

of a loop antenna deployed where the trapped miner is located, uses low-duty-cycle, tone-burst transmission. Portable tuned receivers and loop antennas are used on the surface to determine the location of the trapped miner, 

An extension of .this work has led to a call-alert paging . 

system concept, wherein a transmitter, similar to that used 

in the trapped miner system, is used to audibly and/or 

visually alert key mine personnel carrying pocket receivers 

that they are wanted at the nearest mine telephone. In 

addition, a roof-bolt paging system has been developed, 

wherein commercial paging and mine carrier phone equipment 

are used in conjunction with a long-wire "roof bolt" antenna 

to page individuals and relay to them short voice messages. 

1 

Arthur D Little, Inc 
Despite these advances in communication capabilities within mines, 

some major unmet needs which require further research and development 

still remain. One of these msjor needs is wtreless electromagnetic 

through-the-earth voice communication channels capable of being operated 

in the presence of both surface and subsurface noise environments. Calculations have been made by both Arthur Do Little, Inc. and Collins 

Radio C~mpany'~) relating to the power requirements for such a voice 

communication system, In both instances the power requirements under 

operational conditions for depths of 600 to 1000 feet have been quite 

substantial, in fact, prohibitive, for uplinks and downlinks between the 

surface and the subsurface. The power requirements range from a few 

kilowatts up to the 100-kilowatt range. Similar levels were also predicted for side-link communication paths within the mine. These paths 

may run from a central fixed transmitter to roving miners carrying portable units, or between two portable units. Here the problem is not so 

much reaching the roving miners from the fixed transmitter location, but 

the reverse talk-back problem of the roving miner to the central location, or from roving miner to roving miner where transmitter power is severely limited. The Collins Radio Company's calculations .for power requirements for the roving miner to roving miner indicate that portable 

battery-operated equipment appears barely adequate to meet a 600-foot 

range requirement imposed on such side-link communication systems.* 

* 

Recent in-mine EIM propagation measurements made by Collins do show an 

unexpected propagation enhancement for frequencies near 500 kilohertz. 

It is not known how consistent this enhancement will be over a range 

of coal mine environments, but should it be found to be consistent, 

then the problem of achieving adequate side-link communication range 

will be considerably alleviated. However, it appears that the mechanism that provides this enhancement of side-link propagation will 

serve to make vertical propagation more difficult and thus the problem of subsurface to surface communication may become worse under 

such conditions. 

Arthur D Little, lnc 
One of the means proposed for aiding in overcoming these severe 

power and range limitations has been the application of voice bandwidth 

compression techniques. We anticipated that such techniques would permit greater range with the same transmitter power or the same range with 

less transmitter power. 

The purpose of this study was to determine the applicability of 

state-of-the-art voice bandwidth compression techniques for mine wireless voice communication systems as one means of reducing the power 

requirements of portable units. One reason why voice bandwidth compression appeared attractive is that at VLF to MF frequencies the noise 

received by a portable receiver in a mine is largely dominated by external mine-generated noise and not by receiver-generated noise. Therefore, 

if the received noise power could be lessened by reducing the bandwidth 

required for the voice signal, without paying a corresponding penalty 

in processing power and/or required signal-to-noise ratio, the desired 

range could be either extended with the same transmitter power or obtained 

with less power. Hence, a 10:l reduction in receiver-bandwidth could be 

expected to yield a 10:l reduction in received noise and hence a 10:l 

improvement in signal-to-noise ratio for the same amount of transmitter 

power. Extending this simplistic notion even further, a 100:l bandwidth 

reduction could be expected to yield a 100:l reduction in transmitter 

power. These potential values of transmitter power reduction were large 

enough to undertake this assessment of the potentials of state-of-the-art 

voice bandwidth compression techniques for achieving the goal of significantly reduced transmitter power consumption for mine wireless communication applications. 

B. FINDINGS 

1. Real-Time Techniques 

Real-time voice bandwidth compression techniques currently do offer 

some modest savings in transmitter power requirements. The savings, 

however, are obtained at the cost of increased complexity for transmitters 

and receivers, and we therefore deem them inapplicable to most mine communication needs. There are several fundamental reasons why this is so: 

Arthur D Little Inc 
I, There has been no demonstration as yet of a system which 

even comes close to coding speech at a 50-bit-per-second 

(bps) rate, which is the information rate attributed to 

a voice transmission channel; 

2. Although substantial bandwidth reductions are achieved 

by certain of the voice bandwidth compression schemes, 

such reductions are achieved at an increase in the required signal-to-noise ratio, thus obviating some of the 

expected savings of transmitter power that would accrue 

if the systems were able to operate at the same signalto-noise ratio as for a conventional voice band; 

The emphasis of most of the work on voice bandwidth compression has been directed toward means of achieving more 

voice channels in a given bandwidth. There has been.very 

little concern in this work with the added necessity of 

increasing power per channel. It has rather been strict'ly 

based on getting more channels into a given bandwidth. For 

For this reason, the results are less applicable to mine 

communication needs than might otherwise be expected. 

Bandwidth compression of modest amounts; that is, more than 4 or 5 

to 1 in bandwidth, is generally achieved with analysis-synthesis systems. 

In this type of system, the input voice signal is first analyzed. Then 

voice parameters extracted by the analysis, parameters which will allow 

the voice signal to be reconstructed, are transmitted in a narrow band 

to the receiver. At this point these extracted parameters are used to 

reconstruct the original voice message. Research in this area in the 

193OVs, 19401s9 and 1950's was directed toward using analog circuits to 

implement these systems. Work centered on digital systems in the 1960's 

and 1970'~~ and progress has been significant. 

To digitize conventional telephone speech requires approximately 

50,000 bps. This assumes a bandwidth of about 3000 Hz and a sampling 

rate of about 10,000 Hz with quantizing to around 8 bits. Modem digital systems can encode the analyzed version of the speech into 1200 bps 

Arthur D Littlelnc 
and, in some extreme cases, into as little as 700 bps, This indicates a 

compression of 40:l is obtainable by these systems. However, the more 

advanced systems of this type require considerable computer power, placing 

them considerably beyond their application to mine communication 

requirements, 

2, Non-Real-Time Techniques 

In the area of non-real-time voice bandwidth compression techniques, 

we feel there may be some merit to the stretched-out voice system wherein 

an original message T seconds long is in effect stretched to NT seconds 

long for transmission, All the quality of the original speech may be 

preserved at the same signal-to-noise ratio as would be required for 

full bandwidth transmission, but with a bandwidth that is shrunk in 

direct proportion to the stretch-out factor. Thus, true transmitter 

power requirement savings can be made, but only at the expense of a delay 

between the time.the message is spoken and the time it is available to 

be heard at the receiver, 

C. CONCLUSIONS 

From our study we conclude that real-time state-of-the-art voice 

bandwidth compression techniques are not suitable as a practical means 

for improving mine wireless communications performance at present. However, non-real-time voice bandwidth compression techniques and real-time, 

non-voice systems do offer means for meeting the needs of through-theearth emergency subsurface to surface communications. We conclude that 

the advantage of limiting the voice signal waveform to improve received 

signal-to-noise ratio should be applied to mine wireless communication 

systems. 

D . RECOMMENDAT IONS 

Based on the findings and conclusions of our study, we recommend 

that the stretched-out voice systems be further explored as a means of 

meeting certain of the mine communication needs, particularly for links 

through the earth for emergency purposes, and most important, the link 

from underground to the surface. It is particularly true that power 

Arthur D Little Inc 
savings here can be important because of the limited amount of energy 

available to a miner trapped underground and the requirements for intrinsic safety for electrical communication equipment, We feel that in an 

emergency the delay time imposed by a stretched-out speech system becomes 

amuchlessimportantobjectionthanitis foruseintheday-to-day . 

operation of the mine; and that a suitable system could be made to operate 

at.a substantially reduced power using this means. 

We also recommend that non-voice-t,ype systems where truly narrowband operation can be achieved be studied further. One such system 

might be a very low-bandwidth teletype-quality system wherein the operator 

uses an alphabet keyboard to spell out his message and a LED display to 

present both incoming and outgoing messages. 

State-of-the-art electronic components are available today which 

make both of these alternatives to a real-time voice system.extremely 

attractive for emergency mine communication needs, and capable of being 

demonstrated in the form of prototype hardware with only a modest development effort. 

Arthur D Little. Inc 
11. THE NATURE 02' VOICE SIGNALS 

Speech is produced by the excitation of a complicated acoustic filter 

by raw acoustic energy. The filter consists of the cavities of the 

throat, nose, and mouth. The energy sources are either produced by air 

vibrating the vocal cords (voiced sounds) or by turbulent air flow between the teeth or between the tongue and the teeth or the palate (unvoiced sounds) or by some such combination. 

Typical vocal tract filter characteristics cons is t of a series of 

resonances and anti-resonances with as much as 10 to 20 dB between peaks 

and valleys. At the peaks of these resonances some of the harmonics in 

the exciting forces are reinforced relative to the others. These £requencies where reinforcement takes place are referred to as formants, 

of which there may be as many as five. 

For unvoiced sounds the spectrum of the exciting signal is like noise 

spread out over the audio range. If they go through the filter, they 

exhibit formant frequencies. Hissing sounds that do not go through the 

filter (ssss and sh) are noise-like, with a very wide frequency band. 

Voiced sounds look like a train of impulses with a fundamental £requency between 60 and 240 Hz (for men, but higher for women and children) 

with a rich harmonic structure that falls off with frequency up to perhaps 4000 Hz. These harmonics are modified by the filter and the formants 

form an envelope on the harmonics. 

Formants differ from individual to individual, making the voices of 

different individuals distinctive. However, the combined effects of 

harmonics and filter always produce a rapidly falling spectrum. Figure 

- 1 shows the envelope of spectra for male speakers and illustrates this 

feature. 

One of the most straightforward ways of decreasing the frequency 

required for voice transmission is simply to chop some of it off. Extensive tests have been conducted on the intelligibility of telephone 

speech as a function of bandwidths. 

Arthur D.Little, Inc. 
Frequency in Hertz 

Source: Ref. 3 

FIGURE 1 ENERGY DENSITY SPECTRUM FOR MALE SPEECH 

Arthur D Little, Inc. 
Figure 2 shows the effect of changing the upper end of the speech 

band with a variable cutoff low-pass filter. Xf the band is restricted 

to 1000 Hz, intelligibility is decreased to 40%, but 83% of the energy 

is still present. This curve shows that low frequencies contribute to 

energy much more than to articulation. Figure 3 shows the effect of 

eliminating low frequencies with a variable cutoff high-pass filter. 

This shows that elimination of frequencies below 900 Hz leaves intelligibility above 90%, but decreases energy to 17% of the total. While intelligibility is preserved, speaker recognizability deteriorates severely. 

Arthur D Little, Inc. 
0 1,000 2,000 3,000 4,000 5,000 

Filter Cutoff Frequency, Hz 

Source: Ref. 4 

FIGURE 2 ENERGY AND ARTICULATION 

FOR SPEECH SIGNALS WITH 

LOW-PASS Fl LTER 

Arthur D Little, Inc. 
Articulation 

\ 

I I I I I I 

Filter Cutoff Frequency, Hz 

Source: Ref. 4 , 

FIGURE 3 ENERGY AND ARTICULATION 

FOR SPEECH SIGNALS 

WITH HIGH-PASS FILTER 

Arthur D Little Inc 
(This page intentionally left blank.) 

Arthur D Little lnc. 
111. METHODS OF VOICE BANDWIDTH COMPmSSION 

Systems designed to compress the bandwidth of speech fall into two 

categories: 

1. Those which act or modify the speech signal directly, and 

2. Those which analyze the signal and transmit facts about 

this analysis to the far end where they can be used to 

reconstruct (synthesize) the original speech. 

In the following sections we treat methods of bandwidth compression 

of these two classes. 

A, MODIFICATION OF SPEECH SIGNALS 

1. Band Limiting 

An obvious way of decreasing bandwidth is to cut some of it off with 

a filter or filters. We have observed in Figures 2 and 3 that very 

little articulation is lost by cutting off a substantial portion of the 

lower frequencies or a moderate amount of the upper end of the band. 

Restricting the bandwidth decreases the noise at the receiver. Cutting 

off lower frequencies is especially attractive in a noise environment 

with large discrete frequency noise contributions at the power frequency 

and its first several harmonics. 

During World War I1 the Bell System installed "emergency banks" that 

provided two voice channels in a standard 4000 Hz slot intended for one 

voice channel. This method was accepted as a wartime measure and provided at least marginally useful channels. 

Experiments have shown that restricting the band to 500 to 2000 Hz 

results in about 60% intelligibility for isolated words, but provides 

adequate performance for connected speech with suitable repetitions. 

~r~ter'~) has found that intelligibility with a 1500 Hz bandwidth can 

be improved by dividing the band into three separate 500 Hz bands centered within the range 500- to 3000-range (see Figure 4). 

Pursuit of this simple approach may be an attractive possibility 

for mine communication, since the bands (not necessarily limited to 

Arthur D Little, Inc. 
Normal Telephone Bandwidth 

About 60% 

Intelligibility 

in This Band 

3,000 

Frequency, Hz 

I Improved by Splittin; and Spreading Out I 

0 1,000 2,000 3,000 

Frequency, Hz 

Source: Arthur D. Little, lnc. 

FIGURE 4 BAND LIMITING SPEECH SIGNAL 

Arthur D Little, Inc 
three, nor to a total of 1500 Hz) may be strategically located to avoid 

dominant noise sources. 

2. Time Assignment Speech Interpolation 

Time assignment speech interpolation (TASI) makes use of silent 

periods in speech in a given direction -- either while the source is 

listening or in silent intervals between words or sentences -- by interleaving words from different speakers. This has been effective on transatlantic cables where the medium is costly enough to warrant the expensive terminls, but it does not seem applicable to mine communications. 

3. Temporally Interrupted Speech 

Channel utilization can be improved by continuously interrupting the 

speech signal at a fairly high frequency. If a 50% duty cycle is used, 

another channel may be interleaved in the same band. The resultant intelligibility is a little better than that obtained by halving the frequency bandwidth. If the samples are run together, they can be transmitted at half speed and put back together at the receiving end, halving 

the bandwidth required in the channel. 

4. Clipping 

Where the signal-to-noise ratio is low, some improvement in intelligibility results from simple clipping. This results from increasing the 

energy in lower amplitudes at the expense of distortion on the peaks. 

Simple clipping differs from companding in that no compensating expansion takes place at the receiver. 

Clipping may be carried to the limit, giving a constant-amplitude 

"infinitely clipped" rectangular wavef om. The only in£ ormation preserved is the zero crossings of the original speech. Surprisingly, 

little intelligibility is lost, but speaker recognizability is almost 

obliterated and the quality is unpleasant. 

An improvement in effective signal power of as much as 12 dB can be 

obtained from clipping. This represents a large number of watts at the 

power levels we are considering. 

Arthur D Little, Inc. 
5. Frequency Division (Stretched-out Voice) 

One of the most effective ways of decreasing bandwidth is to divide 

all frequencies by a constant before transmission and multiply them back 

to their original values at the receiver. If all cycles of all frequencies are transmitted, this implies a slowing down of transmission by the 

constant used. An elementary way to achieve this is to record speech on 

a tape recorder and play it over the line at a slower speed with a corresponding decrease in bandwidths. At the receiver the speech is again 

recorded and played back sufficiently speeded up to reconstitute the 

original. 

In principle, there is no limit to the savings in bandwidth achievable 

in this way. From an information theory viewpoint there is absolutely no 

saving in channel capacity since a greater time is used. Time and bandwidths are exchanged. Further processing may reduce the actual channel 

capacity needed. Interrupted samples may be joined and sent as discussed 

in Section A-3. 

Real-time compressed speech may be sent by dividing in real time with, 

for example, phase-locked loops. . The divided frequency is sent for the 

amount of time -- not cycles -- that it occurs and is multiplexed back 

up for that time at the receiver. This is actually a form of "synthesisanalysis" compression where a frequency is encoded into l/n times itself 

and decoded by multiplying by n at the receiver. This type of processing 

goes beyond modification of the voice signal and will be discussed further in Section B-2, Harmonic Compression. This form of frequency division is limited to about 3 to 1. 

B. ANALYSIS-SYNTHESIS TECHNIQUES - VOCODERS 

The compression methods discussed so far try to preserve, however 

crudely, the original voice waveform. Another approach to bandwidth 

compression is to transmit only the important properties of the speech 

signal with the intention of synthesizing at the receiver a signal that 

sounds like the original. Systems that work in this way are called 

"vocoders," a term coined in 1928 by Homer Dudley of Bell Telephone 

Laboratories. 

Arthur D Little, Inc. 
Three important properties of human hearing are fundamental in the 

analysis-synthesis process in vocoaers: 

1. The ear is a spectrum analyzer with a relatively short 

integrating period ; 

2. For monaural sounds, the ear is virtually insensitive to 

phase; and 

3. Hearing is extremely sensitive to pitch variations. 

For a vocoder to be successful then, the short-term power spectrum 

has to be preserved without worrying about the phase relationships. The 

generic system for doing this is the spectrum.channe1 vocoder. 

1. Spectrum Channel Vocoder 

Figure 5 is the block diagram of a spectrum channel vocoder. A set 

of bandpass filters divides the voice signal into frequency channels. 

To determine the power in each band, the signals are rectified and passed 

through low-pass filters. The outputs of these filters are continuous 

estimates of the speech power spectrum in each channel. Working on the 

whole input signal, a pitch extractor determines the pitch and a detector determines whether the speech sound is voiced or unvoiced. 

The separate channels. are transmitted by any of a number of means, 

either analog or digital, to the receiver synthesizer which reconstructs 

the signals using the spectrum energy estimates to modulate the reconstructed' pitch and voicing signal. During voiced segments a pulse generator puts out short pulses at the pitch rate. During unvoiced sounds, 

a noise generator output is fed to the filter bank modulators. 

Vocoder research was active in the 1950's and concentrated on the 

number of channels and optimization of channel filter design. With 

increasing emphasis on digital transmission and availability of powerful computers in small spaces, recent channel vocoder research has been 

directed to digital implementation. 

Arthur D Little, lnc 
Speech In 

Analyzer Synthesizer 

Bandpass Lowpass Bandpass 

Filters Rectifiers Filters 1 I Modulators Filters 

Reconstituted Spe 

Voiced-Unvoiced 

Source: Refs. 6 and 7 

I 

FIGURE 5 SPECTRUM CHANNEL VOCODER 

Arthur D Little, Inc 
2. Voice-Excited Vocoders* 

Naturalness in channel vocoders can be increased by eliminating the 

pitch detector and voicing detectors and substituting a low-frequency 

portion of the baseband signal (hopefully containing the fundamental) . The excitation function is generated at the synthesizer by rectifying, 

filtering, and clipping the baseband signals, thus creating a spectrally 

flat signal with energy at pitch harmonics for voiced sounds, noise for 

unvoiced sounds, and silence for silence. The set of filters does not 

need to include the frequencies covered by the low-bandpass filter since 

these signals can be directly added in at the output (see Figure 6). 

3. Cepstrum Vocoder 

The channel vocoder essentially transmits the Fourier transform of 

the speech signal. Because of the nature of typical voiced speech, significant changes are more readily detectable in the Fourier transform of 

the logarithm of the power spectrum which has been named the Cepstrum. 

Cepstrum information can be more easily used by the synthesizer to separate the envelope of the spectrum (slowly changing vocal tract configuration) and excitation function. 

4. Harmonic Compression 

As described in Section A-2, frequency division without compression 

simply slows transmission. A harmonic compressor, as shown in Figure 7, 

* 

The voice-excited vocoder is an excellent example of the simplification possible for mine communications compared with telephone applications. For an acceptable telephone voice-excited vocoder, the low 

band must include the fundamental frequencies for voices of men, women, and children. The wider this band, the more it eats into possible bandwidth compression. By limiting the band to the range of fundamentals of men's voices only and optimizing the design around accompanying expected formants, much more encouraging results may be obtained than telephone research would lead one to expect. 

Arthur D Little, Inc: 
Analyzer SY nthesizer 

Bandpass Bandpass 

Filters Ci ippers Modulators Filters 

Speech In 

Low-Bandpass Filters 

Source: Ref. 8 

FIGURE 6 VOICE-EXCITED VOCODER 

Reconstituted Speech 

Arthur D Little, Inc. 
Speech In 

- 

m Synthesizer Analyzer CI: 

. - 

V) 

Bandpass Freq. Bandpass 

Filters Dividers Filters 

Bandpass Freq. Bandpass 

Filters Multipliers Filters 

- 135 270 

Hz x 2 - 

Hz 

b 

Source: Refs. 9 and 10 

FIGURE 7 HARMONIC COMPRESSOR 

Reconstituted 

Speech 

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halves the instantaneous frequency of each band, transmits the compressed 

signal, and resynthesizes the input by doubling in each of the (halved) 

bands. 

A harmonic compressor can be applied to the baseband signal of a 

voice-excited vocoder (Section B-2) to decrease the bandwidth required 

by that type of vocoder. 

5. Autocorrelation Vocoder 

A vocoder that operates on the autocorrelation function of speech is 

shown in Figure 8. In the analyzer the short time autocorrelation function is derived for a number of discrete delays. The autocorrelation 

function is band-limited to the.same band as the input signal and is 

therefore completely specified by the instantaneous values for discrete 

delays whose period is no more than the reciprocal of the Nyquist 

interval. A delay step of 0.1 msec corresponds to a 5000-Hz band-limited 

signal. Speech quality is influenced by the maximum delay (number of 

steps). For a 2.5-msec delay 26 delay channels are available. Each 

represents a time-varying sample of the short-time autocorrelation function. If each channel is limited to 20 Hz, a total gross transmission 

bandwidth without multiplexing or excitation of 520 Hz is required. 

At the synthesizer the original signal is regained by a process 

which is essentially a time-varying transverse filter multiplying the 

excitation spectrum by the power spectrum of the original speech. 

The autocorrelation vocoder is not susceptible to degradation resulting from arbitrary choice of bands of frequency that may not correspond with the harmonic structure of voiced speech sounds. However, the 

analogous problem crops up in the choice of maximum delay relative to 

the fundamental frequency being encoded. 

6. Phoneme Vocoder 

Continuous speech may be broken into perhaps 40 distinct elements, 

called "phonemes." Typical phonemes are all vowel and consonant sounds 

in a language. A system that would recognize these phonemes and send a 

simple code to the far end where the phoneme would be generated again 

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Synthesizer I I 

Tapped Delay Line 

Speech 

' Source: Ref. 7, 10 

FIGURE 8 AUTOCORRELATION VOCODER 

Arthur D Little, Inc. 
probably offers the greatest band saving possible. Vocoders of this type 

have not been very successful, although the synthesis process alone *is 

useful for speaking machines operated by a keyboard or a computer. Since 

pre-arranged phonemes are used, speaker recognizability is lost in a 

phoneme vocoder . 

7. Adaptive Predictive Coding 

A great deal of current research is based on adaptive predictive 

coding. This embodies a combination of differential quantizing and 

vocoder techniques, and depends on sophisticated computational ability ' 

in real time. In this approach both the transmitter and the receiver, 

using the same algorithms, predict what the current value of a signal 

should be based on its history. The difference between this value and 

the actual value is transmitted to the receiver where the difference is 

added to the predicted value to produce the next sample. The predictor 

at each end is adaptive -- that is, it changes its ground rules periodically to minimize the error. 

Work in this field is taking place using simulation on large digital 

computers rather than hardware implementation of systems. Workers in 

the field estimate that high quality speech may be realizable with rates 

as low as 1000 bps. 

These systems tend to be vastly confused by signals other than single talkers. Noise and double talking can make them useless. This is 

a serious drawback for use in mines. 

8. Pattern Matching Coding 

Caldwell ~mith'l') has developed a speech coding system which uses 

a set of patterns as the base for coding speech. A channel vocoder 

operates on the speech, and the time history of the output responses 

from each channel forms a pattern over a preselected period of about 

20 msec. For each pattern thus obtained, a best match is found to one 

of 1024 stored patterns, for example. The basic information transmitted 

to the receiving location is a series of pattern numbers, thus developed. 

The speech is reconstructed by a generator which produces a series of 

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sounds corresponding to the sequence of patterns. Data rates for such 

systems are from 500-1000 bps, depending primarily on the number of stored 

patterns used. Intelligibility ranged from 83 - 92% in tests of such 

systems . 

9, Formant Vocoder 

The spectral envelopes of speech sounds are characterized by several 

prominent maxima. These represent the resonances of the vocal tract 

called theformants. Below 3000 Hz three formants are typically found 

in adult speech. In a formant vocoder, the analyzer attempts to determine the frequency locations of the major formants. Signals corresponding 

to the formant locations and strengths are transmitted and utilized to 

control the resonances of a formant synthesizer consisting of three or 

more single-tuned resonant circuits. The block diagram of a formant 

vocoder is shown in Figure 9. 

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Multiplexing 

and 

Transmission 

Source: Ref. 7 

FIGURE 9 FORMANT VOCODER 

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IV . FUNDAMENTAL LIMITS TO BANDWIDTH COMPRESSION 

A number of investigators have treated the problem of the information 

rate associated with speech communication. They tend to agree on a figure of about 50 bps as being the upper limit of the information rate communicable to a person listening to speech. This rate falls far short of 

the nominally accepted 50,000-bps rate necessary to code a digital representation of speech directly. This value is arrived at by assuming a 

sampling rate of 8000 bps and a 7-bit amplitude resolution (corresponding 

to'l part in 128). It is this savings of 1000:l that is the lure for the 

development of compressed speech for use in comnunication systems such as 

might be employed in the mine environment. No one has yet developed a 

real-time voice communication system that achieves this ultimate i n b i t 

rate. However, if such an achievement were made, it should be possible 

to transmit the SO-bps rate in a communication system whose bandwidth 

in Hz is approximately this value. Indeed, by using modern digital data 

transmission techniques, it would be possible to confine the bandwidth 

to even less than this ultimate 50 Hz. However, in general, the reduction of bandwidth beyond that represented by the basic bit rate would be 

bought at the expense of added transmitter power as can be seen from an 

.examination of Appendix IV, Secti0n.E. 

There is another point of comparison that can be used in arriving at 

one of the limits to the compression of voice in real time, and that is 

to accept the number of bits per second required in a phonemic-type voice 

reproducer which responds to phonemic commands and produces an understandable voice output from these phonemic commands. Such a device is 

the Votrax manufactured by the Vocal Interface Division of Federal Screw 

~orks'~~), If pitch and inflection commands are deleted from those of 

the Votrax, then a six-bit characterization results for each phoneme. 

If one assumes a nominal speaking rate of 4 or 5 phoneme words per second, a usable output results at a 90-bps rate. These two limits, then, 

seem to be the boundaries against which voice bandwidth compression 

systems can work. It is instructive to note that a high-speed electric 

typewriter which types 1 character at a time can achieve a typing rate 

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of about 3 words per second, each word comprising, say, 6 characters. 

Thus, 5 bits per character results in a 90-bps rate of transmission. 

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V. DISCUSSION OF BANDWIDTH COMPRESSION TECHNIQUES 

Table 1 shows the classes of bandwidth compression schemes discussed 

in Section 111 and compares their respective parameters. The characteristics compared include: bandwidth or bit rate, signal-to-noise ratio 

required, band reduction ratio, estimate of sentence intelligibility, 

estimate of computational requirements for analysis and synthesis, analog signal processing requirements, acoustic background influences, the 

deterioration of performance with decreasing signal-to-noise ratio, 

pitch preservation, speaker identification, real time, delay time if it 

is not a real-time system, the state of development, cost, size, weight, 

power consumption, and an estimate of voice quality, In a number of instances'the bases for comparison were relative, not absolute, with the 

data presented meant to provide a sense of the relative behaviors of the 

various sys tems . In addition to various types of bandwidth compression 

techniques presented in the matrix, we also presented three others, which 

are not directly bandwidth compression schemes; but represent other means 

of preserving or saving signal-to-noise ratio or decreasing the required 

transmitter *power, such as method of clipping and companding. 

The basis for comparing the compressed systems is a direct baseband 

voice system. The prominent features of each of the voice bandwidth 

compression techniques tabulated in this comparison are discussed below. 

A. ADAPTIVE PREDICTIVE CODING 

Adaptive predictive coding is an area of current ongoing research 

and development. The emphasis in this area is on producing a telephonequality voice with a small bps rate. The impetus for this work is to get 

more voice channels per digital channel in communication systems. The 

general system requires computer operation to develop the codes to represent the speech from the sending station. However, the mine environment 

is a drawback to the use of such a technique. There is heavy reliance 

on very sophisticated and complex digital data processing at both ends 

of the system, and the computer power required is quite significant. 

Another problem is that the adaptive predictive coding system is quite 

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COMPARISON MATRIX 

Frequency - 

Adaptive Spectrum ' Temporally ' Division 

Predictive Channel Formant Band Interrupted (Stretched Harmonic Phoneme Pa

Coding Vocoder Vocoder AM Voice Limiting Speech Voice) Compression Clipping Vocoder C

Bandwidth Not fixed 300 - 500 Hz 300 - 500 Hz 500-3000Hz 1400 Hz 1400 Hz Any 1400 HZ 500-3000 Hz Promises Rot 

to be small 

Bit Rate 700 - 1500 N/A N/ A N/A N/A N/ A N/A N/A N/A N/A 300 

S IN 15 - 20 dB 15 dB 15 dB 6 dB 10 dB 10 dB 6 dB 10 dB 6 dB ? 1

Band N/A 101 1 1011 111 211 211 Any 211 211 Promises 

Reduct ion to be high 

Ratio 

Sentence >90X 

Intelligibility 

Corcpu ta- Large 

tional Requirements 

Analog Negligible 

Signal 

.Processing 

Acoustic Extreme 

Background 

trl Influence 

0 

Deteriora- Sudden 

tion with 

Decreasing 

S/S 

Pitch Yes 

Preservation 

>90% 

N/A 

Modest 

Quite 

Tolerant 

Yes 

Yes or 

No 

>90% 

N/ A 

Modest 

Quite 

Tolerant 

Yes 

Yes or No 

None Modest 

Unchanged Unchanged 

Direct Direct 

with with 

SIN S /N 

Yes Yes 

>90% >90% >90% >90% 90% 

Achievable 

N/A N/A N/A N/A Medium M

Modest Medium Medium Little Xedium M

Unchanged Unchanged Unchanged Troublesome Tolerant To

during 

pauses 

Direct Direct Direct Direct Sudden . S

with with with with 

SIN SIN SIN S/N 

Yes Yes Yes Yes Yes or Y

So 

Speaker Yes Yes Yes Yes Yes Yes Yes Yes Yes Not likely 

Identif ica- , 

t ion 

Real Time Essentially Essentially Essentially Yes Yes Yes No Yes Yes Essentially Esse

Delay Time 25 ms Large 

Depends on 

Ratio 

State of on-going Complete Complete Complete . Some Complete Hardware Complete Experiments Not Co

Development Experiments Design Could Be Developed 

Needed Required Usef ul 

Cost High Medium Medium Low Medium Medium Medium Medium Low Medium Med

Size Large Medium - Medium Small Medium Medium Medium Medium Small Medium H

Weight Heavy Medium Medium Low Medium Medium Medium Medium Low Medium H

Power High Low Low Low Low Low Low Low Low Low R

Consumption 

Subjective Excellent Good Good Good Good Good Good Good Fair Fair 

Voice 

O;:.ll icy 
sensitive to background acoustic noise that mixes with the original voice 

message. The system attempts to translate the total input signal into a 

representation as if all signals were of the voice message. Hence, the 

performance in a noisy environment could become quite poor. The ongoing 

research indicates that, for telephone-type use, excellent system performance will be achieved at a level of about 700 or 800 bps within the 

next few years. The two features of extreme computer power required and 

the susceptibility to background noise make this particular system quite 

unattractive for mine communication systems. 

B. SPECTRUM CHANNEL VOCODER 

Spectrum channel vocoding, which was highly developed 20 years ago, is 

an old art compared to adaptive predictive coding. Relatively unsophisticated circuitry would suffice to provide systems of this type. The 

subjective voice quality from a system of this type is classed as good. 

There will be machine-like characteristics to the reproduced voice, but 

this should offer little problem in a mine environment. An interesting 

advantage of this type of vocoder is its relative immunity to backgroundinterfering acoustic noise at the sending station. This is so because 

the device uses a bank of tuned filters and tends to emphasize the resonant formants of the speaker's voice to the exclusion of background 

noises. 

C . FORMANT VOCODERS 

The developments in formant vocoders follow the lines of the spectrum 

channel vocoders; the characteristics are generally the same. Some feel 

that the perceived subjective voice quality of the reconstructed voice is 

somewhat more machine-like than that of the spectrum channel vocoder. A 

common problem to both the spectrum channel vocoder and the formant 

vocoder is the problem of extracting accurate pitch information for reconstruction of the voice. For mine-type applications it may be possible 

to drop the requirement for, pitch entirely. This will let intelligible 

messages be transmitted, but the received voice would have a monotone 

characteristic to it, or the received voice could be made to sound like a 

whisper and it would then be entirely of an unvoiced character (see Appendix 111). 

31 

Arthur D Little, Inc. 
D. BASEBAND VOICE 

Baseband is included as a basis of comparison for the other systems 

for mine communication uses, One of the characteristics of the direct 

baseband voice -- or a band allocated to voice -- is that its deterioration with deteriorating signal-to-noise ratio is gradual and well understood. By reference to Figure ll*, it can be seen that there is no sudden deterioration of the ability to discern the message. It deteriorates 

gradually with decreasing signal-to-noise ratio, whereas most of the 

processed speech systems fall apart suddenly and completely destroy the 

ability to communicate. 

E. BAND LIMITING 

Band-limiting systems provide an opportunity to make relative simple, 

very modest ratio reductions in bandwidth required for voice communication. 

Such a system simply blocks out certain bands and transmits the remaining 

part. Performance from such band-limiting systems is presented in Appendix I. The conclusion reached is that no great benefit can be obtained 

with such systems. 

F. TEMPORALLY INTERRUPTED SPEECH 

Many experiments on temporally interrupted speech have been conducted, 

They were directed toward speeding up or slowing down recorded speech for 

either rapid comprehension of voiced messages, or during a learning process, to present the voiced message at a slow rate for greater comprehension of difficult subject matter. It is completely evident from this 

work that one can drop one-half of the voice message without disturbing 

the ability to understand the spoken messages. In this way one can, by 

various techniques, cut the bandwidth required for transmission to one-half 

of the original band. We do not expect that there will be any savings in 

transmission power for such a system, however, because it represents a 

removal of redundancy in speech, rather than a repackaging of the form of 

the speech. Again, simple systems can be made and they do work; the 

applicability to mine communication needs is not evident, however, 

-- - 

* 

See page 41, 

32 

Arthur D Little, Inc. 
G. FREQUENCY DIVISION (STRETCHED-OUT VOICE) 

It is obvious that an appreciable reduction of required bandwidth for 

transmission of voiced signals can be obtained simply by recording the 

original voiced message and then playing it back at a much reduced rate 

or stretched out in time. The bandwidth associated with it will shrink 

directly in proportion to the stretch-out factor one uses. In this 

event, one can claim a direct transmitter power savings proportional to 

I the stretched-out ratio. The major problem associated with such stretchedout voice systems is that the users of the system must adapt to the 

enforced added time delay between the sender speaking and the receiver 

hearing the spoken message. 

While an emergency voice transmission system might be able to accommodate quite large delay times imposed by this restriction, it is unlikely 

that day-to-day communications would tolerate a stretch-out factor of 

more than 10 to 1 at the most. A 10-to-1 ratio would correspond to a 

10-to-1 ratio in transmitter power savings. Applicable experience with 

systems of this type comes from the U.S. Navy's underwater telephone 

systems. In seawater the velocity of propagation of sound is approximateiy 5000 ft./sec., and it is well known that transmissions over 5 

nautical miles are obtainable with such underwater telephone systems. 

Hence, one-way time delays of 6 seconds are experienced. The telephone 

operators know that time delay is inherent in their communication system 

and thus are able to adapt to it without' severe difficult. For an 

untrained person, however, such a time delay could be annoying if he 

were not experienced with it nor expecting it. Certainly for the day-today communication needs of mines, delays of this type could be considered 

an undesirable characteristic of the system. For a true emergency voice 

communication sys.tem, we do believe that delays of this length, or even 

longer, would not pose a severe problem. The application of these system can provide a reduction of power capacity required for voice at the 

expense of the imposed time delay. Such a system would best operate with 

very short voice messages, because it is the length of message and the 

reduction ratio that determine the delay time between the transmission 

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and reception of the signal. The possibilities for using this system 

are discussed in Section VII in more detail. 

H. HARMONIC COMPRESS ION 

Harmonic compression schemes of many kinds have been developed. In 

principle, these systems take the original voice spectrum and by real-time 

signal processing reduce the bandwidth by a factor of, say, 2 to 1. Thus, 

an original voice band between, say, 200 and 3000 Hz would be reduced to 

100 to 1500 Hz, cutting the bandwidth in half. Upon reception, the signal frequencies in the received waveform would be analyzed, multiplied 

by 2, and presented to the listener. 

Band reduction ratios of 2 to 1 are readily obtainable in this way. 

Again, a somewhat higher signal-to-noise ratio than for a baseband signal would be required, and there appears to be almost no advantage in 

terms of power requirements for a voice link obtained by this means. 

I. CLIPPING 

Clipping of transmitted waveforms has long been used, particularly 

in instances where the limitation on transmitter power is the peak power 

rating of the transmitter. The ability to communicate depends on the 

received signal-to-noise ratio which is proportional to the average 

power of the transmitter. The average power can be much higher when 

clipping is practiced, thus causing the transmitter to operate at maximum power all of the time. 

However, there are some problems with clipping systems. In particular, there is a harshness to the quality of the voice as received. If 

a high degree of clipping is used, the quiet passages where the voice is 

not present tend to fill up and respond to noise. This creates an irritating property to the output speech presented. .There are mdifications 

of this technique, such as Lincompex discussed in Appendix V of this 

report. In Lincompex, in addition to hard limiting or clipping, a 

narrow-band modulating signal is transmitted which is used to control 

the level at the receiver. This operation makes the voice natural and 

also avoids the problem of the quiet periods being filled up with noise. 

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Clipping can be of considerable use in a voice communication link if 

the transmitter power is peak-limited. The degree of improvement can be 

as high as 12 dB (more than 10 to 1 in power). Thus, it would appear to 

be a desirable feature to include by itself in a mine voice communication 

system, either with or without the Lincompex type of control signal. 

J. PHONEME CODER 

There has been much talk and little resulting hardware related to a 

phonemic-type coding system. One can make up a perfectly adequate English 

language representation with as few as 32 phonemes. There is hardware 

and equipment that accepts phonemic commands. From these commands it 

can generate understandable English or an artificial speech. The Votrax 

is a device of this class. 

There does not presently exist the capability of accepting spoken 

English messages and from these messages obtaining the extraction of the 

phonemic structure to a degree that would be adequate for a voice communication system. It also appears that a device capable of extracting the 

phonemic structure of general spoken English would be a computer-based, 

complicated, non-mineworthy piece of equipment. 

It should be pointed out that present voice comunication research 

is directed toward the telephone-quality voice systems, such as those 

represented by the adaptive predicted coding already discussed, and that 

little work is being done on the phonemic coder. It may be appropriate 

to keep track of the progress being made in the area of voice-controlled 

computers (a part of continuing voice research), because many of the 

voice input systems, as part of their basic structure, need decoders 

similar to those required in a true phonemic coder. 

K. PATTERN CODING 

The pattern vocoding systems developed by Caldwell Smith are an 

interesting approach to producing voice communication systems. The 

basic Caldwell Smith approach is to look at the patterns of spectral 

energy distribution with respect to time produced by the various sections of spoken words and, by examining these patterns over periods of 

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25 to 50 milliseconds, try to find a fadly of patterns from which the 

spoken messages can be coded, then transmit by codes the sequence of 

patterns found in the spoken stream being analyzed. 

Experiments by Caldwell Smith show that bit rates of 300 to 500 bits/ 

second can provide intelligible message transmission. For mine communication needs the amount of storage of information and computer processing 

required generate the proper interpretation of the patterns quite 

large. Caldwell Smith's system used a magnetic disk storage unit, 

entirely inappropriate for use in the mine comunication systems presently 

contemplated. 

L. COMPANDING 

Although not a voice bandwidth compression device, companding is £requently used in communication systems to permit a relatively narrow dynamic range system to accommodate a substantially wider dynamic range of 

message input. It automatically compresses the loud passages of spoken 

messages and automatically increases the level of quiet passages for 

transmitting purposes. Upon reception, the opposite is done to restore 

the correct level to the passages of the spoken message. In some ways 

it suffers from the same drawback as the limiting or clipping of speech 

in that during quiet passages background noises will be amplified and 

transmitted. 

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VI. COMPARISON OF SYSTEMS 

A. METHODS OF COMPARISON 

There are a number of ways in which various voice bandwidth compression systems can be compared. These include degree of circuit complexity, 

weight, size, adaptability for mine environment, power consumption of 

processing circuit, quality of voice, processing power consumption, inherent delays in the system, and so forth. Bandwidth reduction ratio, or 

the factor by which the voice band can be compressed into a transmission 

band, is a parameter which is also central to all voice-compression techniques. However, it should be borne in mind that the application to mine 

wireless communication systems is one in which the prime measure of the 

effectiveness of candidate voice bandwidth compression techniques is not 

the bandwidth reduction ratio per se, but how much power reduction can be 

made while maintaining an adequate signal-to-noise ratio or comprehension 

level for the reconstructed voice message at the receiver. This consideration is therefore quite different from the considerations related to 

much of the past and present work on voice bandwidth compression. 

In the area of past and current progress in voice bandwidth compression, emphasis has been placed, not on saving transmitter power, but 

rather strictly on the conservation of bandwidth. Much attention has 

been directed toward adding capacity in terms of voice channels to already 

existing bandwidth-limited communication paths, which focusses the conservation efforts at bandwidth and not at power. Thus, the objectives 

sought in much of the voice bandwidth compression work are not necessarily 

compatible with the objectives of the Bureau of Mines for voice bandwidth 

compression techniques. For this reason we believe that the method of 

comparison between various candidate systems should be based strictly on 

the amount of transmitter power required compared to a standard level 

required for full-voice bandwidth transmission. Such a posture will 

clarify the real power demands placed on the transmitter of such a system. 

Then the importance of bandwidth compression ratio vanishes from the comparison criteria and the power requirements are given prime importance, 

together with the complexity required to achieve the reported performance. 

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There is a well established body of data from which the various measures of channel quality can be assessed for full-bandwidth baseband 

voice communication. In particular, we used the articulation index as 

a measure of the capability of a voice channel. Figure 10 shows how this 

quality varies as a function of signal-to-noise ratio in the channel. 

There are several conditions associated with the performance indicated. . 

The first assumption is that the background noise has a flat spectrum 

over the voice band, and the second assumption is that the signal-to-noise 

ratio is constant over the band. This assumption means that modest preemphasis of the voice signal has been used and a method of restoring the 

spectral shape of the original voice is used at the output end of the 

system. Under these conditions the plot illustrates the articulation 

index as a function of signal-to-noise ratio. The signal power for these 

plots is the long-term average signal power at the output end of the 

system. Similarly, noise power is the long-term average noise power. 

The articulation'index (A.I.) is a measure of a listener's correct 

interpretation of voice sounds. There are several features of the Figure 10 plot that are worthy of note: 

1. At a signal-to-noise -ratio of +18 dB, articulation index is 

approaching the asymptotic value, and any further increase 

in the signal-to-noise ratio would be judged solely as an 

improvement in quality of the channel, not in intelligibility of the messages received through the channel. 

2. At a -12 dB signal-to-noise ratio, essentially all articulation index is lost and communication becomes almost 

impossible. 

This form of articulation index plot can be converted to one in which the 

abscissa is signal power rather than signal-to-noise ratio merely by 

relating the total average signal power to the average power found in a 

1-Hz bandwidth due to the received noise. If this is done, various systems of bandwidth compression can be compared strictly on the basis of 

power necessary on reception to achieve various levels of articulation 

index. What this new plot will show are the relative power requirements 

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for the various bandwidth compression techniques, together with a measure of what intelligibility or articulation index can be expected for 

the various systems, Although many of the bandwidth compression techniques do not have fully identified articulation indexes as the signal-to 

noise ratio deteriorates, they do identify minimum required values of the 

signal-to-noise ratio in specific bands, and sometimes give a measure of 

articulation index, on which we can base a comparison with other systems, 

Be DISCUSSION OF COMPARISON PLOT 

The first item of concern in the modified articulation index comparison plot of Figure 11 is the way that articulation index varies as a 

function of signal power. For the full voice band (200 to 6100 Hz), it 

can be seen that once the total signal power exceeds approximately 56 dB 

relative to the noise power in a 1-Hz band, there is no further improvement in the ability to interpret voice sounds spoken through a communications system. A drop of 15 dB in the signal power leads one to a point 

where the articulation index is around 50%. As one decreases further in 

signal power, a corresponding loss of articulation index results. Generally, a communication network that works with articulation index below 

30% becomes difficult to use, although not impossible, Consequently, only 

systems that achieve something of the order of 30% articulation index 

should receive serious consideration. 

The plot of Figure 11 also shows points or regions where some of the 

voice bandwidth compression techniques fall. The ideal region for techniques for mine wireless communications is the upper left corner; they 

are also not complex. The most inappropriate.system shown on this plot is 

the two-f ormant frequency vocoder developed by Philco Ford (23) . For this 

vocoder, it is better to use the full-voice bandwidth instead of the 

160-Hz wide channel of the vocoder because the vocoder requires a 30-dB 

signal-to-noise ratio, thus leading to a higher power requirement at a 

75% articulation index than for the full-voice band. In fact, at the 

same power level an articulation index of about 90% results for the 

full-voice band as compared to a 75% level for the Philco Ford vocoder. 

Indeed, because of the nature of that vocoder, even as signal-to-noise 

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Future 700 BIS 

Voice 10-V rrors 

Signal Power: dB re Noise Power in 1 Hertz Band 

Source: Arthur D. Little, Inc. 

FIGURE 11 COMPARISON OF COMPRESSION TECHNIQUES 
ratio improves, it is unlikely that the articulation index would improve 

at all. Therefore, this system is inappropriate for mine wireless communication applications. 

The next system of interest is one identified with Flanagan (12) in 

which a 43-Hz wide bandwidth is required for vocoded speech, with a 

signal-to-noise ratio of 35 dB yielding only a 1-dB improvement over the 

full-voice bandwidth. Again, the sys tepl is inappropriate for ,application 

to mine communication needs. 

Another system of interest is the 2400-bps voice-compression system. 

Such a system, typical of the digitally processed adaptive predictive 

systems with a signal-to-noise ratio of 6 dB, would result in an improvement of some 8 dB over the full-voice band. This improvement would be 

at the cost of complicated digital processing equipment, and therefore 

we do not expect that it would be economical or mineworthy at the present 

time. 

The Caldwell Smith pat tern-recognizing vocoder seems to be occupying 

a favorable position. The use of this technique is dependent on equipment that is quite complex. Caldwell Smith used a learning process as 

well as a disk storage system to perform the analysis and synthesis required for his processing. 

A 4 to 1 non-real time, stretched-out speech system is also shown on 

the plot of Figure 11. This type of processing preserves the characteristic of the full-voice bandwidth system as a function of signal-to-noise 

ratio. Therefore, this relatively simple system, stretched-out voice, 

will have a 50% word intelligibility at the same power level that 

Caldwell Smith's pattern-recognition system would show an 80% word 

intelligibility. Thus, in terms of the communication systems, it is 

entirely reasonable to expect acceptable performance from the stretchedout voice system. 

As a final point of comparison, power requirements for a three character-per-second teletype code system operating over a 30-Hz bandwidth 

with a 6-dB signal-to-noise ratio are shown. This system appears to be . 

Arthur D Little, lnc. . 
extremely favorable from the point of view of power consumption, and does 

provide a data rate that would be useful for some purposes. This should 

be borne in mind when considering alternatives to voice bandwidth compression systems (discussed later) . 

The general overall conclusion obvious from the plot of Figure 12 is 

that available state-of-the-art voice bandwidth compression techniques 

do not offer sufficient improvements in power requirements to make their 

use universally applicable to mine wireless communication systems. 

Arthur D Little, Inc 
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Arthur D Little, Inc 
VII. TTME-DELAYED SPEECH 

As stated in Chapter VI, present real-time voice bandwidth compression systems have an extremely limited capability for decreasing the 

power requirements of voice communication systems. The non-real-time 

systems, on the other hand, do offer some improvements in this respect. 

However, this improvement is bought at the expense of time delay in the 

voice channel. It is well recognized that modest time delays in the 

voice channels offer no serious problems. For example, earth-to-moon 

voice communications have been demonstrably easy to conduct. In this 

instance, one-way delay times of about 1-112 seconds and round-trip 

'delayed times of about 3 seconds are experienced. 

Underwater acoustic voice channels suffer a delay of 1.2 seconds per 

nautical mile and are known to be useful for ranges beyond 5 nautical 

miles. At 5 nautical miles the one-way transmission delay time is 

6 seconds and the round-trip time is 12 seconds. Operators who use 

these underwater systems have no trouble using the system, but they do 

know that delay is to be expected and they so adapt. 

It therefore seems reasonable to expect that a mine wireless communication system having delays of 12 seconds could prove to be a usable 

system. Indeed, an underground-to-surface system between fixed or 

semi-portable stations could well employ such a system. It 

noted that for such a system it may be appropriate to use different 

stretch-out factors for the two directions of transmission. A large 

stretch-out factor would be used for the in-mine transmitter and a 

small stretch-out factor would be used for the surface transmitter. 

This difference in stretch-out factor is possible because the power of 

surface transmitter is not as limited as that for the in-mine transmitter, 

both from the safety aspect and the availability of raw power. 

There are a number of ways that such stretched-out voice messages 

could be handled. Figure 12 illustrates some of the possibilities. At 

the top of this figure is shown an initial voice message of 2-seconds 

duration. The second line illustrates the stretched-out version of this 

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1. Original Message 

2. Stretched-Out Version of Original Message 

3. Received Reconstructed Message 

4. Alternate Message Reconstruction at '/z Original 

Articulation Rate 

5. Repeat of Message Reconstructed at 1/2 Original 

Articulation Rate 

- - - -- 

0 123456 78 9101112 

Seconds 

FIGURE 12 FORMATS FOR RECONSTRUCTION OF STRETCHED-OUT 

MESSAGES AT 6 TO 1 STRETCH-OUT RATIO 

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message, the time stretch-out factor being 6 to 1. The generation of 

this stretched-out version can start immediately; it does not require 

that the full message be completed before transmission of the stretchedout version starts. 

There are a number of alternative ways that the receiver end can be 

handled. Perhaps the most obvious is compressing the received message 

by 6 to 1, such that the last part of this reconstruction corresponds to 

the last part of the stretched-out version. This mode is shown on line 3. 

A second interesting alternative is to compress the received version by 

only 3 to 1, but keeping the pitch to a natural one (this is a familiar 

technique for the slowed-down speech systems promoted for learning foreign languages from tape recordings) . This compression would be done by 

repeating on a subphonemic level short passages of the received and compressed waveform. The output presented at the receiver would occupy 

4 seconds and end as the stretched-out version ends. This mode is shown 

on line 4. A still further possibility is to use 3 to 1 compression and 

repeat it twice, as illustrated on line 5. It is interesting to note 

that this mode results in received messages starting to reach the operator 

only 4 seconds after the start of transmission. 

Stretched-out transmission systems seem to offer one of the few viable means of reducing transmitter power requirements for wireless 

through-the-earth voice communication links. As shown above, there also 

exists considerable flexibility in the way such systems can be 

organized. 

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Arthur D Little, 11 
VII I. REFERENCES 

1. Arthur D. Little, Inc., USBM Final Report, Contract No. H0122026, 

''Survey of Electromagnetic and Seismic Noise Related to Mine Rescue 

Communications," January 1974. 

2. Collins Radio Group, Rockwell International, Preliminary Mine Wireless Capability Investigation for Contract No. H0346067, August 1974. 

3. H. Schlinder, "Delta Modulation," IEEE Spectrum Vol., pp 69-78, 1970 

(includes bibliography with 36 entries). 

4. A. Albert, "Electrical Communication," 3rd Edition, New York, Wiley, 

1950. (Chapter 2 has 47 references on fundamentals of speech and 

acoustics.) 

5. K. D. Kryter, "speech Bandwidth Compression through Spectrum Selection," J. Acoust. Soc. Am., Vol. 32, .pp 547-556, May 1960. 

6. J. Williams, "some Bandwidth Compression Systems for Speech Transmission," Radio & Electronic Engr., Vol. 27, pp 459-472, June 1964. 

7. M. Schroeder, "Vocoder: Analysis and Synthesis of Speech," Proc. 

IEEE, Vol. 54, pp 720-734, May 1966. 

8. J. Bayless, S. Campanella and A. Goldberg, "voice Signals: Bit-bybit," IEEE Spectrum, Vol. 10, pp 28-34, October 1973. 

9. J. Savidge, "Ease Radio-Spectrum Congestion by Bandwidth Compression 

'and Expansion," EDN, pp 74-77, May 20, 1973. 

10. M. Schroeder, B. Logan and A. Prestigiacoma, "New Methods for Speech 

Analysis-Synthesis and Bandwidth Compression,".4th Int. Congress on 

Acoustics, Copenhagen, 21-28 August 1962. 

11. Caldwell Smith, "Perception of Vocoder Speech Processed by Pattern 

Matching ," J . Acoust . Soc . Am. , TJol . 40 No. 6, pp 1562-1571, December 

1969. 

12. J. L. Flanagan, "Bandwidth and Channel Capacity Necessary to Transmit 

the Formant Information of speech," J. Acoust. Soc. Am., =:4, July 1956. 

13. K. D. Kryter, "Validation of the Articulation ~ndex," J. Acoust. Soc. 

Am., 2:11, November 1962. 

Arthur D Little, Inc. 
14. K. W. Cat.termole, "Principles of Pulse Code ~odulation, " London, 

Ilif fe Books, Ltd., 1969. 

15. B. Smith, "Instantaneous Companding of Quantized !Signals," Bell System Technical Journal, Vol. 36, pp 653-709, May 1957. 

16. G. Raisbeck, "Information ~heory," MIT Press (1964), p. 37. 

17. W. Re Be~et. and J. Ro Davey, "~ata ~ransmission," New York, 

McMillian, ' 1965. 

18. J. Martin, "Telecommunications and the Computer ,'I Englewood Cliffs , 

New Jersey, Prentice-Hall, 1969. 

19. J. Salz, qt~ommunications Efficiency of Certain Digital Modulation 

Systems," IEEE Trans. On Comm-Technology, Vol. COM-16, pp 149-156, 

February 1968. 

20. A, Tomozawa and He Kaneko, "~ompanded Delta Modulation for Speech 

Transmission," IEEE Trans. Cow-Technology, Vol. COM-16, pp 149-156, 

February 1968. 

21. Jo Greefkes and Ko Riemens, "code Modulation with Digitally Controlled Companding for Speech Transmission," Philips Tech. Review, 

Vol. 31, pp 115-133, 1970. 

22. KO Do Kryter, "Methods for the Calculation and Use of the Articulation Index," Jo ACOUS~. Soc. Am., g:11, November 1972. 

23. Philco Ford, "Research and Development of an Optimized Laboratory 

Speech-Compression System for Spacecraft Application," Final Report, 

Contract NAS9-4523, Prepared for Manned Spaceflight Center, NASA, 

Houston, Texas, May 1966. 

24. Fo Lo van den Berg, "The Lincompex Radio-Telephone Terminal Equipment of the RY 741 Series ," Philips Telecommunication Review, %: 2, 

April 1972. 

25. Vocal Interface Equipment ~rou~, Federal Screw Works, 3401 Martin 

Ave., Detroit, Mich. 48210. 

26. J . Greefkes and F. DeJaeger , "Continuous Delta ~odulat ion," Philips 

Research Report NO. 22, pp 233-246, 1968. 

Arthur D Little. Inc. 
APPENDIX I 

NARROW BAND VOICE USING FREQUENCY SLICING 

It is a well known fact that the full-voice band from 200 to 6100 Hz 

is not necessary to provide completely acceptable voice channel communications. Indeed, a telephone channel whose transmission band is confined 

to 300 to 3000 Hz is generally considered a good voice channel, and, at 

adequate signal-to-noise ratio, would far exceed the requirements for 

voice communications within coal mines. One of the means of achieving 

narrow-band voice comunications is to restrict the transmission band 

even further than is done for telephone service. 

A convenient way to access the performance to be expected is by 

using the articulation index developed for voice transmission systems. 

The articulation index (AI) ranges from 0 to 100%. If a system has an 

A1 of loo%, it indicates that all isolated words spoken into the system 

would be correctly recognized by a listener. (A moderate degree of skill 

is expected for both speaker and listener.) Correspondingly, an A1 of 

50% would mean correct identification of spoken words 50% of the time. 

Kryter (13y22) and others have developed means for predicting the A1 

for voice channels. The method is based on bands of equal contribution 

to articulation index and the signal-to-noise ratio in each band. Twenty 

bands are used, each contributing a maximum of 5% to the AI. If the s/N 

ratio (long-term average signal power to average noise power) is 18 dB 

or greater, the maximum contribution of 5% per band is achieved. If this 

S/N ratio is -12 dB or less, no contribution is made to the AI. Between 

these two values, A1 contribution depends linearly on the SIN ratio in 

dB. Thus, if the S/N in a band is +3 dB, that band contributes 2.5% to 

the AI. By using this means, it is possible to estimate the performance 

of any proposed frequency-restrictive voice channel. 

The bands of equal contribution to articulation index are tabulated 

in Table I. 

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TABLE 1-1. 

TWENTY FREQUENCY BANDS OF EQUAL CONTRIBUTION 

TO SPEECH INTELLIGIBILITY 

Band Mid- Band MidNo. Limits Frequency No. Limits Frequency 

1 200 to 330 cps 270 cps 11 1660 to 1830 cps 1740 cps 

From this table we can easily determine that a telephone channel from 

330 to 3200 Hz, for example, shows an articulation index of 75% at best 

-- that is, if the S/N ratio in all bands is greater than 18 dB. If the 

S/N in all bands decreases to 8 dB (1/3 of the way to -12 dB), then the 

A1 falls to 50%. For mine communication needs, this value of A1 is 

probably more than adequate. The reason this is so can be seen from 

Figure I-1, which shows that for an A1 of 50%, a sentence intelligibility near 95% can be expected, . 

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Syllables, Words Or Sentences Understood-Percez 

-IUWP CnoOCOo 

0000000000 
It is instructive to examine how power requirements can be expected 

to vary for a frequency-sliced voice communication system, We start 

with the awiiuynption that received noise is flat across the voice band, 

We also assume that the signal-to-noise ratio of proposed systems is 

equalized across the band of interest: 200 to 6100 Hz. In an actual 

system, this would probably be done by pre-eqhasis of the voice signals, 

Because the noise is flat across the band, we can equate the total noise 

power directly to the bandwidth of a hypothetical system, and we can 

plot A1 as a function of signal power level for such hypothetical systems, 

The results of this exercise are shown in Figure 1-2 for four conditions: full-voice bad of 200 to 6100 Hz; a telephone band of 330 to 

3200 Hz; bands 1 through 9 and 11, the 10 bands of largest contribution 

to A1 per Hertz of bandwidth; and bands 1 through 5, the five bands of 

largest contribution to A1 per Hertz of bandwidth. 

It is obvious from these plots that if AI's larger than 30% are 

desired, there is no merit whatever in terms of signal power requirements to the use of frequency slicing. Indeed, the merit is found in 

the use of the widest possible bandwidth, 

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% - xapu I uo!zelna!zJV 

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Arthur D Little, Inc 
APPENDIX I1 

ALTERNATIVES TO VOICE BANDWIDTH COMPRESSION SYSTEMS 

Although we were not asked in the contract to explore alternatives 

to voice communication channels, the subject of narrow-band, coded systems of a non-voiced type surfaced at almost every interview and discussion, both amongst our own staff and those of other organizations with 

whom we discussed the voice bandwidth compression techniques. We tabulate these topics and ideas here for the sake of completeness and because of the fact that most voice bandwidth compression techniques do 

not appear to have as much merit as was orginally expected of them. 

A. LIMITED SYMBOL NARROW-BAND CODE SYSTEM 

This system would use an approach very similar to that of Westinghouse in their Interim Coal Mine Rescue and Survival System wherein a 

limited set of symbols is transmitted to provide a communication 

channel. In that instance, pushbuttons representing four or five 

messages were used to send very narrow-band, coded signals between the 

subsurface and the surface and the surface and the subsurface to provide 

a limited message-capability communication system. This approach was 

mentioned quite frequently by people with whom we discussed the general 

mine communication problem. 

B. MORSE CODE-TYPE SYSTEMS 

It is certainly possible to communicate over a very narrow bandwidth system with a Morse Code-type of communication. The problem with 

using Morse Code is that miners are likely to be unfamiliar with the 

Morse Code, and the difficulty of reading from a list of alphabetical 

characters and their Morse Code equivalents could pose a problem, particularly during emergency or dark conditions. 

C. AN ALPHABET SYSTEM WITH KEYBOARD 

Communication and data-link equipment manufacturers have brought 

about the development of compact, hand-held keyboard and display units, 

such as that illustrated in Figure 11-1. A 5-bit code is entirely 
FIGURE. 11-1 HANDHELD KEYBOARD AND DISPLAY 

58 

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adequate to represent a full alphabet of characters and messages would 

simply be spelled out in sequence by the operator and subsequently 

encoded and transmitted by the unit in a narrow-band transmission scheme. 

The keyboard on this portable unit is intended for one-finger operation; 

thus, the character rate is probably liinited to three or four per second 

and the bit rate is approximately 20 per second, a number easily handled 

by a narrow-band communication system. Received messages could be presented on the optical display part of the hand-held unit such that 

received messages are spelled out in front of the operator. Such a low 

bit-rate system would appear to be compatible with performance achieved, 

by the trapped miner detection and location equipment presently undergoing tests by the Bureau, and units suitable for use in mines should I 

not be difficult to develop. 

D. FLAG SIGNAL SYSTEMS 

On two occasions when we were talking with people about communication 

requirements in coal mines, they immediately suggested the use of flag 

codes as used at sea for developing a message structure capable of transmitting a wide variety of messages with a rather small number of symbols. 

Naturally, the flag would be replaced by coded signals, but the technique 

of communicating could be made the same. Unfortunately, in this instance 

a look-up book would be required at both transmitter and receiver to 

enable the operator to put together messages and to interpret received 

messages. 

E. PANTOGRAPH SYSTEM 

In conversations with other people concerned with the narrow-band 

communication systems, we found the consensus to be that the Pantograph 

System used in some restaurants to communicate a written order from the 

waitress to the kitchen might be feasible for mine use. In such a system, two analog signals can characterize the position on a piece of paper 

of a pen from which the original written message can be reconstructed. 

We have not given much consideration to this as a viable means, but it 

is an interesting possibility for communication. 

Arthur D Little, Inc 
FQ THE ELECTRONIC PAD 

Some of the people with whom we talked suggested that use could be 

made of an electronic pad which, again, is another way of transmitting 

a written or printed message from the sender to the receiver. In this 

instance, a stylus would be run across an electronic pad, writing characters, symbols, pictures, or words. These would be electronically 

encoded and transmitted for reconstruction at the surface. Again, it 

could be expected that the system could operate effectively in a fairly 

narrow band. Little further attention was given to this as an alternate 

possiblity . 

Arthur D Little, lnc. 
APPENDIX I11 

WHISPERED SPEECH 

The analysis/synthesis type systems which form the major bulk of the 

voice bandwidth compression systems require that pitch be measured and a 

decision be made as to whether the transmitted part is voiced or unvoiced. 

Data from Flanagan (I2) show that the channel capacity required for the 

pitch and the voiced/unvoiced decision is approximately 125 bps. Therefore, if it were possible to drop these two measurements from the analyzing 

equipment, a corresponding reduction in required channel capacity would 

occur. How much of a savings this reduction in required channel capacity 

achieves has to be referred to the bits per second required for the full 

capability of the system. These numbers range from 1000 to 30,000 bps, 

so that at best a 12% savings of channel capacity, and hence, power, 

could be achieved by the adoption of this scheme. 

One might well inquire as to the quality or ease of dealing with conversations held in whispered voice -- the result af dropping both the 

voice/unvoiced and the pitch information in an analysis/synthesis system. 

The reproductions of speech from the control signals would be produced 

as whispered speech. While one might believe it to be difficult to carry 

on a conversation in whispered speech, this perception is probably based 

on the difficulty of articulating with whispered speech, but in the systems we are contemplating the input would be in normal voice and the 

analysis/synthesis equipment would be used to create the whispered speech. 

Bernald Gold of M.I.T.'s Lincoln Lab, in a private communication, told us 

that whispered voice is perfectly understandable. We have two other 

points of comparison for this conclusion. During our visit to Caldwell 

Smith at Air Force Cambridge Research Laboratories, we witnessed demonstrations of his equipment. One of the demonstrations was to have the 

synthesis equipment operate in a whispered fashion: the whispered voice 

resulting was perfectly intelligible. The second occasion was during a 

Votrax demonstration. By operating it in a whispered mode too, the putput was found to be perfectly intelligible. 

Arthur D Little, lnc. 
Thus, it can be said with a fair degree of confidence that an 

analysis/synthesis system can be made which does not use pitch or voiced/ 

unvoiced information, and cormnunicates in'a satisfactory manner to the 

listeners in a whispered fashion, Therefore, it is a perfectly acceptable system to consider, At the present time the shortcoming of this 

system would be that its application has only a marginal effect on the 

channel capacity required for communication purposes, and thus would 

bring no very large benefits to a system, It could be noted, however, 

that by dropping the above two requirements from the system, the equipment becomes simpler to fabricate. It has generally been recognized that 

the pitch extraction requirement for analysis/synthesis systems always 

pose problems of one sort of another, The dropping of this requirement 

would therefore simplify the equipment moderately, 

Arthur D Little, lnc. 
APPENDIX IV 

THE NATURE OF TRANSMISSION CHANNELS 

A. TRANSMISSION OF ANALOG SIGNALS ON DIGITAL CHANNELS - PCM 

In the 19501s, long before the explosive growth of computer communications, Bell Telephone Laboratories developed a pulse code modulation 

(PCM) transmission system to handle voice signals over digital lines -- 

the T1 carrier system. In the original T1 carrier system (whose terminals 

are now designated as Dl channel banks) each voice channel used a 64,000- 

bps pulse stream. Despite the apparent squandering of bandwidth, T1 was 

economically competitive with contemporary analog carrier systems. This 

was a classical tradeoff of bandwidth for other advantages -- principally 

the ability to regenerate signals as they went along so that degradations 

did not accumulate with distance. 

It is interesting to observe what Dl channel banks do with their 

64,000-bps per channel: the voice bandwidth is limited to well below 

4000 Hz and then sampled at the Nyquist rate, 8000 times per second. 

(Note that the sampling rate could be lower. With expensive filters 

the telphone band could be sharply limited to 3000 Hz and the sampling 

done at a little more than 6000 per second. Since filters represent a 

per-channel cost rather than a common equipment-shared cost, the decision 

was made to use a sampling rate of 8000 per second and avoid the use of 

expensive filters .) 

In Dl each sample is encoded into a 7-bit binary word. An eighth . 

bit is added for signalling and supervision. Thus, each channel uses: 

(7 + 1) 8000 = 64,000 bps. 

The noise in a PCM system is mainly due to quantizing distortion 

which is caused by the approximation in assigning a discrete binary 

word to the value assumed by a continuous speech voltage waveform. A 

well-known theoretical analysis which assumes a uniform probability density of the quantizing error over +1/2 - the size of the least significant 

bit has the result that for an n bit binary word the ratio of the rms 

Arthirr I3 I ittlp Inc 
quantizing error to the full range of values is 1/2"m. Thus, for the 

7 bits of Dl the theoretical ratio of the ms quantizing noise is (in dB): 

20 log [(0.707) (m) (z7)/2] = 44 dB 

which is surprisingly good. If it were not for the restricted bandwidth, 

it might qualify as high fidelity. A linear PCM system, however, is 

very poorly matched to the amplitude statistics of a speech signal. Dl 

consequently was modified to provide optimum performance for speech signals by compressing the speech signal before encoding and expanding it 

after decoding. This compressing plus expanding (called companding) 

follows an experimental law such that for the compressor alone: 

where p = 100 for Dl. The effect of this logarithmic companding is to 

smear the original signal-to-noise ratio out over a wider dynamic range. 

With linear encoding, 7 bit theoretical SIN is 44 dB at full load and 

drops 1 dB per dB as the level decreases (reaching 0 dB at -44 dB signal 

level. This behavior is shown by the dotted line in Figure IV-1. With 

p = 100 the signal-to-noise ratio remins at 30 to 35 dB from 0 to -30 dB 

signal level (see Figure IV-1). Logarithmic companding is therefore a 

tradeoff, since it provides more noise distortion for large amplitudes 

and less for small aplitudes, but that is just what voice signals need. 

If Dl channels had been meant for some other signal, such as facsimile or fsk data, the companding (if any) would certainly have been 

completely different . 

A value of 100 for y was chosen for TI because of the technology of 

the late 1950's. The compressor and expander nonlinear characteristics 

were determined by diode networks, and the higher the p, the.more difficult was the problem of the two characteristics tracking to cancel each ' 

other. It should be noted that the logarithmic characteristic used in 

T1 is by no means universal and other countries use different companding 

laws 0 

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Dl channel banks were designed for exchange trunk transmission and 

did not meet the requirements for longer haul trunk carriers. The D2 

channel bank was designed about a decade after the Dl to make use of 

more modern technology to provide toll transmission without changing the 

basic T1 line format of 1,544,000 bps. It was clear that using a whole 

bit per sample (8000 per second) to transmit switchhook and dialing information was wasteful, so D2 uses this bit for voice signal transmission 

5 times out of 6. (A slight change is required to keep track of channel 

signal framing as well as channel number framing.) In D2, was changed 

to 255. No longer are coder and compander separate -- the two functions 

are combined in a single, nonlinear coder performing both functi.ons. 

These two changes result in a theoretical SIN ratio better than 35 dB 

over about a 39-dB dynamic range which meets toll requirements in the 

direct dial hierarchy (see Figure IV-1). 

It should be realized that logarithmic companding in itself is a 

powerful bandwidth compression scheme. By capitalizing on the instantaneous amplitude distribution of speech, it offers a great economy in 

number of bits needed. For example, to pick a range on the Dl curve, 

if we want to maintain a 17.5 dB SIN ratio at a signal power 40 dB below 

full load with linear PCM, 11 digits would be needed per sample instead 

of 7. Thus p = 100 companding reduces the line bit rate by 36%. 

The curves in Figure IV-1 are ideal curves. Actual PCM systems may 

be 1 to' 3 dB poorer for any of a number of reasons such as small errors 

in sampling time, finite rather than instantaneous sampling, compressorexpander characteristic mistracking, noise in decision circuits, and 

general system noise. 

The analog telephone channels derived on T systems are used interchangeably with all other analog channels in the telephone plant. Thus, 

it is not unusual for a 1200- or 2400-bps modem signal to be riding on 

a T carrier channel using 64,000 bps. This dual standard poses tariff 

inconsistencies for telephone companies. For example, to provide a 50- 

kbs data channel, 12 voice channels must be removed from an analog carrier but only one from a digital. Tariffs are not geared to the particular facility providing service. Is the fair price for a 50-kbs channel 

Arthur D Little, Inc. 
Theoretical 

D-2 Performance 

Signal Power-dB Below Full-Load Sinusoid 

Source: Ref. 14, 15 

FIGURE .IV-1 PCM QUANTIZING NOISE 

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roughly 1 or 12 times that for a voice channel? This inconsistency is 

one of the pressures that is leading to all digital systems such as Datran 

and Bell's Dataphone Digital Service (DDS). 

B. TRANSMISSION OF DIGITAL SIG~ALS ON ANALOG CHANNELS - MODEMS 

At the same time that Bell was finding it economically attractive to 

transmit analog signals on digital chanels, the inverse problem arose 

and grew rapidly. Telephone plant had evolved over the years to handle 

voice efficiently and was not well suited for raw digital data. For 

example, phone channels generally do not transmit DC or indeed any frequency below a few hundred Hertz, so they are not suitable for baseband 

DC pulses. To transform the data information into a format compatible 

with the telephone channel, it is necessary to modulate it in some way. 

To reconstitute it, it must be demodulated at the other end. This combination of a modulator-demodulator is called a modem or a data set. 

By the mid 1960's data sets for 1200 to 2400 bps using.frequency 

shift keying (FSK) or phase modulation were readily available. It is 

instructive to consider how much better transmission might be. A well 

known result from information theory is ~hannon's expression(16) for the 

information capacity of a band-limited channel in the presence of gaussian 

c = wlog2 (1 + 5) n 

where C is the channel capacity in bps, W is the bandwidth in ~e;tz, s 

is the signal power, and n the noise power in the band W. Shannon's 

equation is analogous to the Carnot cycle in thermodynamics; it sets an 

upper bound on theoretically achievable performance, but does not tell 

how to achieve or even approach this bound. 

An interesting consequence of Shannon's equation is that channel 

capacity can be increased without bound for any W or S/N ratio provided 

there is no limit on the other. Another limit to transmission rate is the 

Nyquist rate 2W -- the upper bound on which symbols can be sent through 

a band W. This is not a limit on channel capacity since there is no 

limit on the number of bits that can be encoded in each symbol; indeed 

Arthur D Little, Inc. 
it is intuitively clear that the larger S/N ratio is, the more bits per 

symbol a decoding arrangement can distinguish. Section E shows some of 

the properties of Shannon's equation. 

What does Shannon tell us about a telephone channel? For an untreated 

channel the band may be 400 to 3000 Hz for a W = 2600. If the S/N ratio 

is 30 dB, we get: 

C = 2600 log2 (1 + 1000) = 25,900 bps. 

Using complex combinations of phase and multilevel modulation, modems 

for 9600 bps are now available, considerably short of the 25,900 bits 

theory says are possible. If we take the true capacity of the channel 

as 25,900 bps, we can assess the present state of the art, in terms of 

efficiency : 

For digital transmission on analog channel: 9,600 = 337% 25,900 

For analog transmission on digital channel : 25¶900 = 46% 

56,000 

This is the money changer situation -- you lose both ways. In the next 

two sections we examine what hope there is for improving these two 

efficiencies. 

C. BPS PER HERTZ IN DIGITAL TRANSMISSION ON AN ANALOG CHANNEL 

While it is a side issue, it may be worth noting that 9600 bps on a 

telephone channel, even though it represents only 37% efficiency, may be 

close to the best that will be accomplished for a long time, maybe ever. 

Shannon's equation speaks only to the problem of gaussian noise and there 

are many other degradations in phone channels including phase distortion, 

impulse noise, crosstalk, phase jitter, frequency offsets, and nonlinear 

distortions. It has only been the development of means of coping with 

the first of these, phase distortion, with automatic transversal equalizers that speeds have risen from the 2400-bps range to 9600 bps. Cure 

of some of the remaining ills is beyond the present state-of-the-art. 

Arthur D Little, Inc. 
It is useful to examine the theoretical efficiencies of various modulation techniques, Details of the techniques themselves are beyond the 

present scope of the report; see for example, Bennet & Davey (17) or 

Martin(18). For band-limited gaussian noise, the efficiencies are 

plotted in Figure IV-2 (from Salz - Ref. 19). These curves assume that 

transmitters and receiver filters are optimized to discriminate against 

noise. Smooth curves are drawn connecting the efficiencies of 2-level, 

4-level, 8-level, etc., systems. 

Single sideband is the most efficient system although it falls far 

short of Shannon's equation. FSK is least efficient. Some of the current research in modems use novel modulation not shown on Figure IV-2, 

for example, simultaneous combination of amplitude modulation and 

phase modulation. 

I The problem of mine communication with bandwidth compression techniques is the joint problem of processing and transmission through a very 

hostile medium. Efficient interaction of the two is of utmost importance. 

An experimental program in bandwidth compression must take into account 

the overall system. For example, Figure IV-2 hints at the rapid changes 

in error rate with small changes in SIN ratio. Two equally efficient 

processing schemes mayq react quite differently to a distribution of 

digit error limits. It should also be noted that Figure IV-2 deals only 

with gaussian noise. The peculiar noise environment in a mine may give 

very different results. 

D. HERTZ PER BPS IN ANALOG TMSMISSION ON DIGITAL CHANNELS 

We have seen that companding reduces the number of bits for Dl channel performance from 11 to 7, but that the resultant efficiency still 

leaves room for improvement. A great deal of telephone-oriented research 

has been taking place in Delta modulation and related differential quantizing techniques. The prime motivation of the research has not been 

bandwidth saving per se, but rather coder-decoder simplification. 

A nonlinear PCM encoder-decoder is quite expensive. This is not 

serious in T carrier, since it is shared by 24 channels so the per channel 

Arthur D Little. lnc 
Signal-to-Noise Ratio in dB (S/N) 

Source: Ref. 19 

FIGURE IV-2 BITSISECONDIHERTZ VS SIGNALINOISE 

Arthur D Little, lnc 
cost is not excessive. There are applications in subscriber carriers for 

much smaller bundles than 24, and here the cost of the PCM encoder-decoder 

becomes a problem. A Delta modulation coder-decoder is very simple compared with its PCM counterpart. If Delta modulation can be made as good 

as PCM for a given bit rate, the economics of small digital systems become attractive. 

The basic idea of all differential quantizing schemes is to quantize 

and encode changes in the signal rather than instantaneous samples of 

the signal itself. The simplest type is Delta modulation which can be 

thought of as 1-bit differential PCM. 

A Delta coder consists of a comparator in which an analog signal 

derived from the coder output is compared with the input signal. At intervals controlled by a clock, an output pulse is sent if the input is 

larger than the output and no pulse is sent if it is smaller. Decoding 

consists of a simple integration process. The more often pulses are 

sent (higher bit rate) the better the approximation. Delta modulation 

is subject to a unique type of distortion known as slope-overload 

distortion. Here the rate of change of the input signal becomes too 

large for a series of ones to follow. As a result, curves of S/N ratio 

vs level are different for different frequencies. Fortunately, the level 

for a given distortion falls off at almost exactly the same rate as a 

normal speech energy spectrum. As a consequence, if an adequate S/N 

ratio exists at 800 Hz, speech will tend to sound natural. 

Figure IV-3 compares differential quantizing schemes with Dl theoretical performance. For normalization, a voice signal rate the same as 

Dl (i.e., 56 kbps) is assumed. It will be noted that straight Delta modulation is quite inferior. Delta modulation is appreciably improved by 

companding. ,Various companding schemes are detailed in the references . ' One of the most promising (Tomozawa and Kaneki (20)) is shown in Figure IV-3. 

More sophisticated schemes have been proposed. One of these (Greefkes 

and Riemens (21)) is for digitally controlled companded Delta modulation 

(DCDM). Here the quantization strip size is varied at the transmitter 

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0 10 20 30 40 50 60 

Signal Power-dB Below Full-Load Sinusoid 

Source: Ref. 20 

FIGURE IV-3 DIFFERENTIAL QUANTIZING AT 56KBPS 

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and receiver simultaneously under the control of the digital signal. To 

indicate the potential usefulness of this for mine communications, the 

BOO-Hz performance of a 16-kbps DCDM channel is compared with 5 kbps Dl 

PCM for reference and shown in Figure IV-4. 

E. CHANNEL CAPACITY RELATIONSHIPS 

In the investigation of the applicability of voice bandwidth compression techniques to mine communication systems, it is important to 

know the general way in which channel capacity varies as a function of 

channel bandwidth, noise, and signal power. The classical relationship 

is contained in Section B and has been recast below: 

P 

where 

C = channel capacity in bps, 

W = bandwidth in Hertz, 

P = received signal power, 0 

N = received noise spectral density (flat noise assumed). 0 

Two curves (Figures IV-5 and IV-6) derived from this relationship are 

plotted. Figure IV-5 seeks to show how transmitter power (and thus 

received power) depends on bandwidth for a fixed-channel capacity 

requirement. This plot illustrates the penalty in power that is exacted 

if one insists on a conununication system that asks for even a few bps 

per Hertz of bandwidth. Figure IV-6 shows how channel capacity varies 

for a fixed power level and variable bandwidth. 

Arthur D Little, inc. 
.- -- - - - - - .--- - - -.I 

DCDM - 16KBS 

Signal Power-dB Below Full-Load Sinusoid 

Source: Ref. 21 

FIGURE IV-4 16KBPS DCDM PERFORMANCE 

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= Bandwidth-Hz 

Source: Arthur D. Little, Inc. 

10,000 

1,000 

I00 

10 

FIGURE IV-5 RELATIVE POWER REQUIRED AS FUNCTION OF BANDWIDTH 

FOR FIXED CHANNEL CAPACITY IN PRESENCE OF FLAT NOISE 

"0 -=',[25 No -1 

- 

C= 1000 BPS 

- \ C= 100 BPS 

I I I I 

10 1 00 1,000 1 0,000 100,000 1,000,000 

Arthur D Little, Inc. 
Source: Arthur D. Little, Inc. 

FIGURE IV-6 CHANNEL CAPACITY AS FUNCTlON OF BANDWIDTH 

FOR FIXED SIGNAL POWER IN PRESENCE OF FLAT NOISE 

Arthur D Little, Inc 
APPENDIX V 

LINCOMPEX 

Perhaps the most interesting method of improving signal-to-noise ratio 

in voiced AM radio comunication developed in the past decade is the 

Lincomp ex (24) (for Linked Compressor and Expander) technique. The basic 

idea is to transmit the speech signal at the maximum level capability of 

the channel. Amplitude variations in the speech signal are almost entirely 

removed by a compressor. The amplitude information is transmitted as a 

narrow-band frequency modulated signal. In the receiver, the amplitude 

signal is used to control an expander in order to restore the normal 

amplitude variations in the speech signal. 

A simplified block diagram of the Lincompex system is shown in Figure V-l. The input speech is supplied to an amplitude measurement circuit as well as a direct speech channel. The delay in the speech channel is necessary to synchronize the amplitude measurement, which is 

delayed by a low-pass smoothing filter, with the envelope of the speech 

signal. The amplitude signal controls the compressor, a variable gain 

amplifier, so as to reduce the gain at high amplitude and increase the 

gain at low amplitude, resulting in a compressed speech signal with constant amplitude. The speech signal is then filtered to remove any components above 2700 Hz. 

The amplitude signal is also applied to a voltage-controlled oscillator to obtain a narrow-band frequency-modulated signal varying from 2840 

to 2960 HZ, which corresponds to an amplitude range of 60 dB (2 Hz per 

dB). This FM amplitude signal is now added to the filtered compressed 

speech signal to modulate the SSB transmitter. 

The speech and amplitude components of the received Lincompex signal 

are separated by filters, as shown in Figure V-1. The FM amplitude signal is detected by a discriminator, and this output is used to control 

the expander. The speech signal is also applied to the expander. The 

expander is a variable gain amplifier operated in the reverse manner 

from the compressor. Now the gain is increased by high amplitude signals 

Arthur D Little. Inc. 
Amplitude Channel /' 

Amplitude Signal 

FIGURE.V-1 SIMPLIFIED BLOCK DIAGRAM OF LINCOMPEX SYSTEM 
and reduced by low amplitude signals from the discriminator, with the 

result that the speech output of the expander has the amplitude variations 

of the original speech signal. Obviously, it is very important that the 

speech and amplitude signals be kept in synchronism if the system is to 

work properly. This necessitates several other delays not shown in the 

simplified block diagram, A high degree of frequency stability is also 

required because of the narrow-band FM used to transmit the amplitude 

information. The overall system is complex and expensive, 

While there is general agreement that Lincompex does provide significantly improved quality conversation under certain poor signal-to-noise 

conditions, there is a difference of opinion on the desirability of 

Lincompex for certain applications. The negative view contends that, in 

general, the improvement is not worth the expense, and that most of the 

time, transmission conditions are sufficiently good so that the improvement due to Lincompex is marginal, 

Arthur D Little. Inc 
œ his page intentionally left blank.) 

Arthur D Little, lnc 
PCM and Companding 

K. W. Cattermole, "Principles of Pulse Code ~odulation," London, Iliffe 

Books, Ltd., 1969. 

H. Mann, H. M. Straube and C. P. Villars, "A Companded Coder for an 

Experimental PCM Terminal," Bell System Technical Journal, Vol. 56, 

pp 173-226, January 1962. 

B. M. Oliver, J. R. Pierce, C. E. Shannon, h he Philosophy of PCM," 

Proc. I.R.E., Vol. 36, pp 1324-1331, November 1948. 

B. Smith, "Instantaneous Companding of Quantized Signals, " Bell Sys tem 

Technical Journal, Vol. 36, pp 653-709, May 1957. 

Digital Transmission on Analog Channels 

W. R. Bennet and J. Re Davey, "Data Transmission," New York, McMillian, 

1965. 

J. Martin, "Telecommunications and the Computer," Englewood Cliffs, 

New Jersey, Prentice-Hall, 1969. 

J. Salz, "Communications Efficiency of Certain Digital Modulation 

Systems," IEEE Trans. on Corn-Technology, Vol. COM-18, pp 97-102, 

April 1970. 

C. E. Shannon, "Communication in the Presence of Noise," Proc. I.R.E., 

Vol. 37, pp 10-21, January 1949. 

Delta Modulation, Differential PCM, etc. 

S. Brolin and J. Brown, "Companded Delta Modulation for ~elephony," 

IEEE Trans. Comm-Technology, Vol. COM-16, pp 157-162, February 1968. 

P . Cumminsky, N . Jayant and J . Flanagan , "Adaptive Quantizing in 

Differential PCM Coding of Speech ,I1 Bell System Technical Journal, 

Vol. 52, pp 1105-1118, September 1973. 

J . Greefkes and F . DeJaeger,, "Continuous Delta ~odulation, I' Philips 

Research Report No. 22, pp 233-246, 1968. 
J. Greefkes and K. Riemens, "Code Modulation with Digitally Controlled 

Companding for Speech Transmission," Philips Tech. Review, Vol. 31, 

pp 115-133, 1970. 

N. Jayant, "Adaptive Quantization with a One-Word ~emory , " Bell System 

Technical Joumal, Vol. 52, pp 1119-1144, September.1973. 

F. Johnson, "Calculating Delta Modulator Perf ormance, " IEEE Trans. 

Audio and Electroacoustic Vol., pp 121-129, March 1968. 

E. Protonotarios, "Slope Overload Noise in Differential Pulse Code 

Modulation Systems," Bell System Technical Journal, Vol. 46, pp 2119- 

2161, 1967. 

Ha Schlindler, "~elta ~odulation," IEEE Spectrum Vol., pp 69-78, 1970 

(includes bibliography with 36 entries). 

A. Tomozawa and H. Kaneko , "~ompanded Delta Modulation for Speech 

~ransmission," IEEE Trans. Comm-Technology, Vol. COM-16, pp 149-156, 

February 1968. 

Characteristics of Speech 

A. Albert, "~lectrical ~o~nication," 3rd Edition, New York, Wiley, 

1950 (Chapter 2 has 47 references on fundamentals of speech and 

acous tics. ) 

C. Cherry, "On Human Communication," 2nd Edition, New York, Dover, 1961. 

K. D. Kryter, "Methods for the Calculation and Use of the Articulation 

Index," J . Acoust . Soc. Am. , 24: 11 (November 1962) . 

Speech, Bandwidth Compression Systems 

B. Atal and M. Schroeder, "Adaptive Predictive Coding of Speech Signals ,I1 

Bell System Technical Journal, Vol. 49, pp 1973-1986, October 1970. 

B. H. Batson, "An Analysis of the Relative Merits of Various PCM Code 

Formats, National Aeronautics and Space Administration, Internal Note 

No. MSC-EB-R-68-5 (November 1, 1968) . 

Arthur D Little, Inc 
B. H. Batson and Re W. Moorehead, "A Digital Communications System for 

Manned Spaceflight Applications", National Aeronautics and Space 

Administration, Lyndon Be Johnson Space Center, Houston. 

. J. Bayless, S. Campanella and A. Goldberg, "voice Signals: Bit-by- it," 

IEEE Spectrum, Vol. 10, pp 28-34, October 1973. 

B. Bogert, "The Vobanc - A Two-to-One Speed Bandwidth Reduction System," 

Jour. Acoust. Soc. Am., Vol, 28, pp 399-404, May 1956. 

S, J. Campanella, D. C. Coulter, L. A. Barnes, R. E. Irons and 

J. M. Pickett, "Formant Tracking Vocoder System, Melpar Inc. a Subs. 

of Westinghouse Air Drake Co., 3000 Arlington Blvd., Falls Church, 

Virginia (September 1, 1963 to November 30, 1963), Contract No. 

DA36-039-AMC-0006(e), DA Project No. 1G640309D491; DA Task No. 

lG640309D49105. 

S. J. Campanella, D. C. Coulter and R. E. Irons, "A 1000 Bit Per second 

Speech Compression System," Melpar, Falls Church, Virginia, a report of . 

work on Contract AF33(600)-39962, 

J. Daguet, "Summary Codimex, Speech Compression System," Paper G 2 in 

Proc . .of the Speech Communication Seminar, Vol . I1 (Stockholm 1963) . 

H. Dudley and T. Tarnoczy, "The Speaking Machine of Wolfgang von 

Kempelen," Jour. Acoust. Soc. Am., Vol. 22, pp 151-166, March 1950 

(description of a successful pneumatic-mechanical phoneme vocoder 

built in 1780). 

J. L. Flanagan, "~evelopment and Rating of a Formant-Coding Speech 

Compression System," Jour. Acoust. Soc. Am., a:6, November 1956. 

J . L . Flanagan, "Speech Analysis, Synthesis and Perception, " Academic 

Press, New York, 1965. 

J. L. Flanagan, andwi width and Channel Capacity Necessary to Transmit 

the Formant Information of Speech," Jour. Acoust . Soc. Am., Vol. 28, 

pp 592-596, July 1956. 

J . E . Flood and D . I. Urquhart-Pullen , "Time-Assignment Speech Interpolation in Time-Compression-Multiplex-Transmission, Proc. IEEE - 111, 

(4) : (April 1964). 

83 
G. Font and K. N. Stevens, "Systems for Speech Compression,'' Fortschr. 

Hochfrequenztechn., 5 (1960). 

A. J. Goldberg, "~ocoded Speech in the Absence of the Laryngeal Frequency,'' 

B. Gold, Editor Group 62, Technical Note 1967-9, 3 April 1967-9 

[AD651 2531. 

A. J. Goldberg, M. J. Rose and A. Areese, "A Predictive Voice Processor 

for Narrowband Communication channels," IEEE, ICC, June 1974. 

H. Gruen and B. Olevsky , "Increasing Inf ormation Transfer , " Advanced 

Communications Laboratory, Philco Corp . 

J. Haskew, J. Kelly, R. Kelly, Jr. and T. McKinney, "Results of a Study 

of the Linear Prediction Vocoder," IEEE trans. on Corn., Vol. COM-21, 

pp 1008-1115. 

H. E. Jones, "Bandwidth Economy for Multiplexed Digital Signals," 

Westinghouse Electric Corporation Defense and Electronic Systems 

Center, Baltimore, Maryland. 

J. W. Jones and W. R. Roberts, "Performance Standards and Test- Specifications for Voice Message Amplitude and Bandwidth Compression 

~quipment," Final Report, Contract AF30(635)-12857. 

K. D. Kryter, "Speech Bandwidth Compression Through Spectrum Selection," 

Jour. Acoust. Soc. Am., Vol. 32, pp 547-556, May 1960. 

K. D. Kryter, "Validation of the Articulation Index," Jour. Acoust. Soc. 

Am., 34:11, November 1962. 

D. T. Magill, "Adaptive Speech Compression for Packet Communication 

System," Stanford Research Institute, Menlo Park, California 94025. 

C. H. Manney, "Applicability of Speech Bandwidth Compression Techniques 

in Mine Electromagnetic Communications," National Bureau of Standards 

Report 10728, Sponsored by Bureau of Mines Agreement H)111019. 

J. Markel and A. Gray, Jr., "A Linear Prediction Vocoder Simulation 

Based Upon the Autocorrelation ~ethod," IEEE Trans. on Acoustics, 

Speech and Signal Processing, Vol. ASSP-22, pp 124-134, April 1974 

(includes 15-item bibliography) . 

Arthrrr 1) i .it tic. Ini. 
W. F. Meeker and A. L. Nelson, "~ocoder Evaluation ~esearch , " National 

Technical Inf. Service, U.S. Dept. of Commerce, Springfield, Va. 22151 

(February 29, 1964), Contract No. AF19 (604) 8507, Project No. 4610, 

Task No. 4610002, AD-602955 (prepared for Air Force Cambridge ~esearch 

Laboratories Off ice of Aerospace Res. U. S. Air Force, Bedford, Mass. 

AFCRL-64-453; CR-64-558-1). 

H. L. Morgan, "Bandwidth Compression and Expansion by Frequency 

Division and Multiplication," U. S. Patent 3,349,184, October 24, 1967. 

A. Noll, "Cepstrum Pitch ~etermination," Jour. Acoust.. SOC. Am., Vol. 41, 

pp 293-309, February 1967. 

J. B. OVNeal, "Efficient Digital Encoding Methods for Improved Military 

Communications Systems," NTIS National Technical Inf. Service, U.S. Dept. 

of Commerce, Air Force Office of Scientific Res., Contract No. F44620- 

69-C-0033 (April 1, 1972). 

G. Power and C. Speaks, "~ntelligibility of Temporally Interrupted 

Speech," Jour. Acoust. Soc. Am., Vol. 54, pp 661-667, 1973. 

B. 0. Pyron and F. R. Williamson, Jr., "signal Processing By Infinite 

Clipping and Related Techniques," National Technical Inf. Service, 

U. S. Dept. of commerce, Spring, Va. 22151 (August 16 to March 31, 1964), 

Project No. A-727, Contract No. DA 49-092-ARO-21 AD600 863 (prepared for 

Army Research Office, Dept. of the Army, Washington, D.C. 

L. R. Rabiner, L. Be Jackson, R. W. Shaf er and C. He Coker, "A Hardware 

Technology, Vole COM-19, No. 6, December 1971. 

J. Savidge, "Ease Radio-Spectrum Congestion by Bandwidth Compression 

and Expansion," EDN, pp 74-77, May 20, 1973. 

M. Schroeder, "Vocoder: Analysis and Synthesis of Speech," Proc. IEEE, 

Vol. 54, pp 720-734, May 1966. 

M. Schroeder, B. Logan and A. Prestigiacoma, "New Methods for Speech 

Analysis-Synthesis and Bandwidth omp press ion," 4th Int. Congress on 

Acoustics, Copenhagen, 21-28 August 1962. 
G. S. Sebestyen and D. Van Meter, "~nvestigation of Automation of 

Speech Processing for Voice ~ommunication," Litton Systems, Inc. Data 

Systems Division (May 28, 1962), Contract No, M19(604)-8828 BD 287 887 

(reproduced by the Armed Services Technical Information Agency. Arlington 

Hall Station, Arlington, Virginia). 

H. S~O, "Speech Compression," University Microfilms Limited, High Wycomb, 

England (submitted in partial fulfillment in the Department of Elec. 

Eng., University of South Carolina) 1967. 

C. Smith, "Perception of Vocoder Speech Processed by Pattern ~atching," 

Jour. Acoust. Soc. Am., Vol. 40 No. 6, pp 1562-1571, December 1969. 

C. P. Smith, "A Phoneme ~etector," Jour. Acoust. Soc. Am., 2:4, 

July 1951. 

C. P. Smith, "Speech Compression studies ,I1 Armed Services Technical 

Information Agency, Arlington Hall Station, Arlington, Virginia 

(September 1962) AD 291 740 [presented at the 1962 Northeast Electronics Research and Engineering Meeting (NmREM), Boston, Mass,] 

J . L. Steward, "Two New Speech Corn-nication Systems ,I1 Technical 

Documentary Report No. MRL-TDR-62-10, March 1962, Biomedical Laboratory, 6570th Aerospace Medical Res. Labs, Aerospace Medical Div., 

Air Force Systems Command, Wright-Patterson Air Force Base, Ohio; 

Project No. 7233, Task No. 723305 (prepared under Contract No. AF33- 

(616)-7800 AD 278 711. 

H. A. Straight and 0. C. King, "speech Processing Studies (Voice Data 

Processing System), Melpar Inc., 3000 Arlington Blvd., Falls Church, 

Virginia, Contract No. AF19 (628) 214, Proj ect No. 4610, Task No. 

461002, Final Report, October 1963 [A~4304611) (prepared £or Cambridge 

Research Laboratories, Of £ice of Aerospace Research, U. So Air Force, 

Bedf ord, Mass. 

W. Wathen-Dunn and D. Lipke, "on the Power Gained by Clipping Speech 

in the Audio Band ," Jour. Acoust . Soc. 'Am. , Vol. 30, pp 36-45, 

January 1968. 
R. M. Wilkinson, "An Adaptive Pulse Code Modulator for Speech Signals," 

Signals Research and Development Establishment (January 1972), Report 

No. 72001, AD 750-167, Procurement Executive Ministry of Defense 

S.R.D.E., Christchurch, Hampshire, England. 

J, Williams, "Some Bandwidth Compression Systems for Speech Transmission," Radio and Electronic Engr., Vol. 27, pp 459-472, June 1964. 

R. Viswanathan and J. Makoal, "Current Issues in Linear Predictive 

Speech omp press ion," presented at EASCON, October 1974. 

Philco Ford, "Research and Development of an Optimized Laboratory 

Speech-Compression System for Spacecraft Application ," Final Report, 

Contract NAS9-4523 (prepared for Manned Spaceflight Center, NASA, 

Houston, Texas, May 1966. 

"Speech Analysis/Synthesis Survey," 1967, sponsored by Office of Aerospace Research, USAF. 

"Speech Analysis Techniques," contract No. DA-91-591-EUC-2626 (July 1, 

1962 - June 30, 1963) , Final Technical Report No. 4 AD 417356, catalogued 

by DDC Defense Documentation Center for Scientific and Technical Infor- . . 

mation, Cameron Station, Alexandria, Virginia. 

"1972 Conference on Speech Communication and Processing, Institute of 

Electrical and Electronics Engineers, Inc., Air Force Cambridge Research 

Laboratories, Cat. No. 72 CHO 596-7 AE (April 24-16, 1972) AD 742236. 

"An Evaluation of Speech Compression ~echniques ,I'Bolt, Beranek and 

Newman, Inc., Technical Report No. RADC-TDR-63-90 (February 23, 1963), 

(prepared for Rome Air Development Center), Research and Technical Division, Air Force Systems Command, U.S. Air Force, Friffiss Air Force 

.Base, N.Y.; Project No. 4519, Task No. 45350, Contract No. AF 30(602)- 

2235) AD 402 604. 

Recent Conferences 

IEEE Transactions on Audio and Acoustic, Vol. AU-21, No. 3, June 1973 - 

Special Issue on 1972 International Conference on Speech Communication 

and Processing, Newton, Mass., 24-16 April 1972. 
National Telecommunication Conference, 26-28 November 1973, Atlanta, Ga. - 

Session 29, "voice Processing for Narrowband Channels," Vol. 2 of IEEE, 

Cat. No. 73 CHO 86 2NTC. 

Other 

Arthur D. Little, Inc., USBM Final Report, Contract No. ~0122026, 

January 1974, "Survey of Electromagnetic and Seismic Noise Related to 

Mine Rescue ~ommunications." 

Collins Radio Group, Rockwell International, "Preliminary Mine Wireless 

Capability Investigation" for Contract No. H0346067, August 1967. 

J. La Flanagan, C. H. Coker, L. Re Rabiner, R. W. Schafer andN. Umeda, 

"Synthetic Voices for computers ,It IEEE Spectrum, October 1970. 

G. Raisbeck, "Information ~heory, "MIT Press (1964). 

F. L. van den Berg, "The Lincompex Radio-Telephone Terminal Equipment 

of the RY741 Series," Philips Teleconrmunication Review, - 30:2, April 1972. 

Vocal Interface Equipment Group, Federal Screw Works, 3401 Martin Ave., 

Detroit, Michigan 48210. 
I 

7. Author(s) 1 8. Performing Organization Rept. 

BIBLIOGRAPHIC DATA 

SHEET 

Little (Arthur D.), Inc. 

25 Acorn Park 

Cambridge, .Mass. 02140 

1- Report No* 

ADL-77229-Task A 

Richard H. Spencer and Warren G. Bender 

9. Performing Organization Name and Address 

1 1. Contract/Grant No. 

USBM-H0346045 1 Task Order No. 1 

I 

4. Title and Subtitl Technical Services for Mine Communications Research 

Task A (T.0.1) Applicability of State-of-the Art Voice 

Bandwidth Compression Techniques for Wireless Mine Communicatior 

No. 

, 10. Project/Task/Work Unit No. 

12. S onsorin Organization Name and Address I 8.S. #epartment of the Interior, Bureau of Mines 

5. Report Date Preparation 

July 1975 

6, 

2. 

Pittsburgh Mining and Safety Research Center 

4800 Frobes Avenue 

Pittsburgh, Pennsylvania 15213 

3. Recipient's Accession No. 

I 13. Type of Report & Period 

Covered Final (Task A) I I May 1974 - June 1975 1 

I 15. Supplementary Notes 

Initiated under the Coal Mine Health and Safety Program . 

16- Ab346045stracts The applicability of voice bandwidth compression techniques to mine wireless 

communication systems is examined. Promise had been attached to such techniques 

because the lessened bandwidth gave hope for lessened received noise and hence 

increased operating range or reduced transmitter power requirements. Analog vocoders 

and. digital adaptive predictive coders are examples of real-time techniques examined. 

For these techniques, as the bandwidth gets narrower, the signal-to-noise ratio 

required rises, so that expected benefits of reduced transmitter power are not fully 

achieved. Benefits are achieved only through the use of sophisticated and costly 

equipment deemed impractical for most mine applications. However, non-real-time 

bandwidth compression achieved by stretching out the transmission time for a voice 

signal to several times its original duration can provide a savings in transmitter 

power but with an attendant time delay. 

17. Key Words and Document Analysis. 17a. Descriptors 

Bandwidth Compression 

Mines-Communications 

Thru-the-earth Communications-Electromagnetic 

Coal Mines 

Vocoders 

Mine Wireless Communication 

Mines-Coal 

I 

17 b. Identifiers/Open-Ended Terms 

17c. COSATI Field/Group 

FORM N TIs-36 (REV. (0-73) ' ENDORSED BY ANSI AND UNESCO. THIS FORM MAY BE REPRODUCED , USCOMM- DC 826s-P74