Controlled delay line signal processor for sound reproduction

Controlled delay lias signal Processor for sound reproduction

Cambridge Research and Development Group, Sanford David Greenberg _p DS Liquidating Partnership and Murray Morton Schiffman

C?37920

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1»׳ ־ BACKGROUND OF THE INVENTION

The field of this invention is the processing of human speech signals or similar signals for ultimate comprehension by ^; the human listener at substantially the natural or normal frequency component distribution but at time interval durations which are usually different from the original time duration of the speech utterance־ ; Sound compression and expansion systems which utilize rela’ tive motion between a magnetic tape record medium and the air gap of the pick-up head which senses the recorded signal on ; the magnetic medium are well known as exemplified by the patent to Schuller 2,352,023־ Devices of this type suffer from the usual operational, cost and weight limitations involved in equipment which utilizes substantial mechanical motion components־ A delay line version of time compression and

-----------expansion for real time signals is also well known as shown, .. for example,. in the patent to French et al 1,671,151, where a voice signal is propagated along a delay line and a ; movable pick-up repeatedly scans the delay line to sense the . .

signal propagating therethrough with the relative velocity

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between the pick-up head and the propagation velocity of the wave in the medium giving bandwidth compression or expansion for t; the purpose of transmitting the signal over a narrow band

Ms • telephone line־ Later workers^in the field eliminated the

2י' mechanical motion portions of systems such as French et al . by substituting electronic switching sequentially along taps on the electrical delay line thereby providing frequency compression j or expansion of the signal for transmission over a narrow band line. The application of the sequentially scanned tapped delay .i . .

i line for modifying the time duration of recorded speech signals ' without altering the frequency components thereof is disclosed by Greenberg et al 3,480,737. By relating the speed of scan of ) the delay line to the velocity of propagation in the delay medium and the relative speed of the reproduction of a recorded • message compared to the speech utterance from which it originated

Greenberg et al achieve time expansion or compression of a recorded speech signal without altering .the frequency components thereof.

Another form of frequency-time transformation is known in the art using a signal, controlled variable time delay line for error correction. Systems of this type detect an unwanted frequency effect due to time irregularities in the pulse train ; of a repetitive signal or such variations in an audio system where the speed of the record medium past the pick-up head is subject to periodic variations which result in the production

H of the audible irregularity known as wow. In reproducing the

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=3׳ original signal these systems eliminate speed errors by servocontrol of the time delay of a delay line which is interposed in the signal channel. Audio systems which employ a reference or timing signal track with variable delay line compensation for

5׳ ;; playback speed are shown for example in Fig. 9 of Coleman, Jr.

^!i .3,202,769 who shows an open loop servo. Woodruff3,347,997״ shows 1;

ן a closed loop servo which adjusts playback speed to compensate

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I;-.'' .! for the relatively low frequency wow component whereas lj imperfections of a high frequency nature (i. e., flutter) are t .j compensated by a variable delay line. Such compensation j;

P systems depend for their operation on the repetitive character i;' of the error signal and thus by providing a delay line of maximum delay time adequate to compensate the maximum expected error do * not encounter the problem which would be presented for a system that required indefinite increase in the delay quantity to continuously modify the time-frequency character of a signal.

The systems described in the patents to Schuller, French :1 ’

I: et al and Greenberg et al, when used to reduce the frequency of ;i OfU1 'a speech signal, while compressing the time in-which a given segment of speech is ropE&daeed, inevitably involve discarding

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!1 a portion of the original speech wave. The ratio of the

I : speech signal discarded to that which is used is directly ;; . related to the compression ratio and the discard loss is inherently and fundamentally related to this process of reducing / the frequency and compressing the time for the processing of a L ' . ־ _Ί given passage of speech« Since the portion of the speech which

. . ־ is reproduced alternates with portions which are discarded the problem of merging to reproduce sections in continuous time i! slots presents some problem and various solutions have been . offered, i;

Thus Schliller suggests a skewed air gap in his rotating magnetic pick-up head or skewed tape approach to the point of contact with the rotating air gap.so that the arrival and departure of the magnetic tape record relative to the air gap will occur gradually as the skew provides a transition from zero to full air gap contact with the recording medium.

The patent to French et al suggests a number of alternatives

׳.! including the use of two spaced transducers rotating in unison ; with respect to the delay medium such that the message kA reproduced in compressed ^time^in one transducer has superposed ל thereon from the other transducer the message which would :! ordinarily have been discarded by the first transducer.

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;'. These efforts to compensate for the discontinuity caused by j: ' i ! periodically discarding portions of the continuous speech wave

j. have, in general, been possible only because the output signal . . I ! is derived from a sampling along the propagating speech signal ' ' . . I ; path since with such sampling the signal stored in the line

I can be readily accessed at any point along the line־ For the ' above-mentioned variable delay line correction systems, discard ¹

J ' !has not been considered or compensated־

1® j j- BRIEF SUMMARY OF THE INVENTION

The present invention provides compression-expansion systems ! ' ' . I ' for speech or other coded signals in which the active frequency0&׳י^רי&|ί t time conversion element is a^signal responsive delay line which , ’i is directly interposed in the path between the signal source and i ' ^:!

^: the ultimate reproducer or utilization device which receives the • converted speech wave־ Such a system does not avoid theί !

inherent problem in time compression occasioned by the!

! discontinuity resulting from discarding alternate portions of the

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2Θ . speech wave־ The discard portion of the speech wave may be f stored cancelled in the delay line or diverted from entering

I the input terminal of the delay line which is directly in the

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(: signal channel. In any case this signal and transientsI

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I' produced by line switching must be discarded as the variable h '’ ., delay line is repetitively controlled by the delay signalן !: between its minimum and maximum delay values־ Line switching

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:: and discard occur coincidentally with the requirement for : making contiguouytwo originally spaced, portions of the speech i* . .

wave and hence both of these functions are accomplished by a »; number of alternative embodiments herein disclosed which function i .

j in a manner consistent with requirements imposed by the parameters

Of the speech signal itself.

i^; ; Accordingly, it is the principal object of the present i invention to provide a speech compression-expansion system which

p. utilizes a signal controlled variable delay line located directly ;. in the signal channel between the signal source and sound reproducer which delay line is repeatedly sequenced between ! maximum and minimum delay values to modify the frequency-time characteristic of the sound reproduced from the original signal.

It is a feature of the invention to control the input and outt put bandwidth of the system for the speech frequencies which &lt;-_t are required to be reproduced for intelligibility in relation to the compression ratio thereby to exclude those frequencies which would produce distortion and intermodulation due to insufficient sampling rate or excessive phase shift per delay : stage for the high frequencies and inadequate output chunk length to attain frequency conversion for the lowest frequencies.

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A further feature of the invention provides maximum discard ii . . 1

j. intervals as determined by the maximum delay line length which j in relation to the compression ratio limits the actual ¹I

I! original speech message discard to a value which minimizesI j! ;

;; the loss of significant cues or transitions in the speech code ;

!j .j ji thereby minimizing the loss of information content transferred ti «..

ji to the listener.

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j: A still further feature of the present invention is to provide, ij. I ί: in a system utilizing variable delay line speech compression or .

KI !ן expansion, for signal processing at the point of juncture ofj / two reproduced speech portions to suppress distracting noise!

I; components and also to avoid the introduction of false cuesj j! 'i

;. which could modify the information conveyed in the subsequent • speech segment״ To this end, if required to suppress suchj 'I noise, the transition between successive reproduced speech ₍. samples can be modified by simple transfer function selection or control, or the transition can be eased by the introduction !

• of synthetic or speech-derived signal portions to approximate :

fi !: a smooth transition within a time interval which does not lose ^! !, .

: actual cues and under such conditions that do not introduce : j:. 'j ji false cues. .| יי ׳ ^f ^object of the present invention to provide improved sound signal processing for frequency conversion and time expansion or compression including unity or real time operation to provide substantially continuous m. jin ף ».

signal output and~wit-h alternate and^ftproved־^zers->®M--©i delay Jin 0 vnV״ tAVr .r Λ *'־iV signal'storage_zto accomplish the frequency conversion of the sound signal.

I!', These and other features and advantages of the invention

I ' will be apparent from the following detailed description !

li . . . I li taken in conjunction with the accompanying drawings״ !1 .

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,'I BRIEF DESCRIPTION OF THE DRAWINGS ' 1 .' ί jj Figs« 1(a) to 1(h) show diagrams representing the!

'1* ׳i reproduction of a message recorded on magnetic tape and the !* operation of the system of the present invention at different ' ΐίί ¹⁰ ,. compression ratios״I ־'׳' Figs2 ״(a) and 2(b) are diagrams for compression and , !1i ! expansion, respectively, showing input-output signal time:

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: relations״i ’ '' ־י ' י ׳' Fig« 3(a) shows a set Of curves relating various parameters involved in the processing of speech at various compression !

ratios greater than one and Fig3 ־(b) indicates similar relations for ratios less than one (i. e», expansion)״

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‘י: Figs» 4 (a)Yandj4 (f) show waveforms useful in describing forms of processing of a transition between adjacent!

reproduced speech samples« !! Figs5 ״(a) to 51(d) show a set of curves representing '

1: '* active processing of the transition between adjacent samples״ ₍ } Figs6 ״(a) to 6(e) show waveforms useful in describing l| the. use of two delay lines to effect transition betweeni

I,²⁵ adjacent speech samples־ i

! Fig» 7 shows a block diagram of a speech compressorexpander system in accordance with the invention.

Figs» 8(a) to 8(d) show waveforms useful in describing ;

the operation of the system of Fig» 7»

' ן ji Fig» 9 shows a block diagram of a dual delay line' system

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I in accordance with the invention.

j Figs» 10(a) to 10(d) show waveforms useful in describing the operation of the system of Fig» 9 for compression»

Figs» 11(a) to 11(d) show waveforms useful in J .

ji describing the operation of the system of Fia. 9 for expansion« !!I . Fig12 ־ is a partial block diagram of a modification. ' iI

Figs» 13(a) to 13(c) show waveforms useful in describing

1־ the operation of circuit of Fig» 12 for compression.

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t: Figs. 14(a) to 14(c) show waveforms useful in describing *J| ‘ the operation of the modification of Fig. 12 for expansion.

i; 'i i;. Fig» 15 is a partial block diagram of a dual delay line ׳ binaural system. .i ii. . i jj Figs. 16(a) to 16(f) show waveforms useful in describing j: the operation of the system of Fig. 15» li ׳; ' Fig. 17 is a partial block diagram of a speech processor t׳ '!

i in accordance with the invention using an analog shift

י. י ן ! register as the variable delay element, a

Fig18 ־ is a block diagram showing gap filling with signal continuity in a system similar to Fig. 17.

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Fig. 15 is a partial showing of an embodiment of the invention with variable delay provided by an r-bit parallel

° ' . ' ',־ .{; digital shift register» I ־ . .

ij Fig» 20 shows an embodiment of the invention with variable delay provided by a serial digital shift register»

I* Fig» 21 shows an embodiment using analog storage ’ t! .

׳׳ memory matrix for variable delay» i!י

!. Fig» 22 shows an embodiment using an r-bit digital random ;! access memory־ .t ‘ j; Fig» 23 shows a logic diagram of a directional zero signal !

!ΐ . level gating control־ |! Fig24 ־ shows waveforms useful in describing the operation ( Of the circuit of Fig23 ־.

Fig25 ־ shows graphically the clock frequency and maximum

2-⁵ signal frequency for the system of Fig. 17־

Fig» X is a block diagram of a dual delay storage line system employing analog shif registers with separate read and write clock pulse signals.

Fig. \(a) and (b) show a waveform diagram of the gating control signals for the system of Fig־3 ־ 1:

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i BRIEF DESCRIPTION OF THE PREFERRED EMBODIMENTS !

i ' ’ _י I ! . The description of the preferred embodiments will be preceded by a discussion of the parameters of the speech signal particularly as they relate to speech compression for reproducing ΐ a given speech message in a shorter period of time־ Because . ; of the fundamental and unavoidable limitations involved in ' speech compression, the following discussion will proceed with primary attention directed to the method and apparatus used i

' in compression mode־ Compression mode operation in the time , domain results in the discard of a fraction of the original .

information directly proportional to the compression factor

J I which is also the factor by which the time to present a given speech sequence is decreased. The method and apparatus are usable however in expansion mode and the considerations involved for use of reproduced signals which occupy a greater '־ length of time than the original speech utterance will be ________ described separately hereinafter־ The system is also capable of frequency transformation without a. corresponding time change to achieve a desired frequency signal, such as may be involved £_n generating speech in a medium having a propagation ! velocity different than air־

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:. Referring to Fig1 ־, a particular system using a delay i!

.j line־ which provides maximum time delay of 6ms for the final

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portion of the sample is shown. Assuming that the speech signal

Ίi !־ to be processed is limited to frequency components between 333 , :

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I! and 5000 Hz״ certain parameters of the playback system for 1 !1I .! compression can be defined־ A magnetic tape 21 has the speech ; ' ־־״. ' ' ׳.I ;1 signal recorded thereon of which the lowest frequency component iI :! at 333 Hz is depicted by sine wave 22 with the tape being drawn .! past a pick-up transducer 23 as it is wound on a take-up reel

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24 at speed S. The electrical signal produced by the

;. ־ transducer 23 passes through a compression processor 25 and is ' reproduced as an audible signal from speaker 26־.

'j The system 23-26 shown on line (a) of Fig. 1 just described

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!: reproduces the recorded signal on the tape 21 without frequency ^! ־i or time change if the speed of take-up reel 24 draws the tape ־ . . ־i past the transducer 23 at the speed of recording S and for this condition processor 25 would introduce a fixed constant delay >1 time of any value. Thus in line, (b) of Fig. 1 where c=l, the i reproduction of the 333 Hz sinusoidal signal without change

J. ¹’ ; other than fixed phase delay (which has been ignored) is shown.

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1. For speech compression the tape speed is increased by a I

ί.ן jj factor c and the processor 25 is operated to vary the delayJ ‘ linearly from its minimuni to maximum value. As shown in curves i: i ף (c), (d) and (e) of Fig. 1, a compression ratio of c = 2 restores, 25 J ן . for a 6ms final signal delay requiring 8ms delay line, a !!י

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. i ii 12ms portion of the original recorded wave 22 which now retains half and discards half the amount of signal originally :: vela/c׳ J .; occupying 24ms of recorded time־ This^portion is designated j *׳a chunk״ and is shown in speeded form prior to processing on:

• I ' line (c) of Fig. 1 to include cycles number 1, 2, 3 and 4־ By .

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1' virtue of the compression process and since the max-i^'es final ./ ⁴I / signal delay is maintained at 6ms corresponding to 8ms in the׳ delay line at the end of the sample, a 6ms portion of the;

h1 !; original information representing 12ms at the recorded speed:

is discarded and on line (c) is labelled discard. Thisj

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J; discard includes cycles 5, 6, 7 and 8 of the original wave 22

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If and represents the gap in the information content between;

successive chunks which are reproduced as audible signals ־ ׳! This audible output is represented on line (d) where the chunk is 'i depicted as a piece of tape 31 played at speed 2S and containing : ' .....- . - .....-.... . - . I • cycles 1-4 which after processing is effectively stretched into

1' . ׳ ?’ j! a piece of tape 32 occupying the original 12ms of recorded time and containing the cycles 1-4 at their original recorded frequency.

j: In line (d) it will be noted that the next cycle reproduced is i; - ⁰ I . cycle number 9 of the original wave after cycles 5-8 inclusive have been discarded־ The representation in line (d) of a i *I : smooth transition between the end of cycle 4 and the start of

I .1 ! cycle 9 should not be taken as representative of real signal ,

׳ ] ׳ : conditions as would be obvious from a consideration of an actual ! ־I !! signal as opposed to the idealized signals presented in Fig־1 ־

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Lines (f) , (g) and (h) of Fig1 ־ illustrate the situation

Which prevails when the. compression ratio is equa.l to five with the tape speed drawn past transducer 23 at five times thei recorded speed S. With a final signal delay of 6ms maximum' i this compression ratio results in a chunk length of 15־!

containing 2 1/2 cycles of the 333 Hz wave 25 of line (a) and again a discard interval of 6 ms equal to the final signal delay and corresponding to a delay line length of 10ms at the end of.

I the sample־ The information gap, however, has increased to the point where the last half of cycle number 3 and first half of j cycle number 13 and all the intervening information in the original recorded wave have been lost in the discard and this gap in the message represents 30ms of the originally recorded speech utterance־| t The relations among the parameters of a speech compression

I . system and those pertaining to the information content of the speech coding interrelate in a manner. to specify optimumj conditions and place outside limits on the mode of operation of . i the systems of the invention for a given intelligibility factor־ i These parameters can be examined with respect to a particular ! system for various compresssion ratios and for this purpose !

I the system parameters for a system using a delay line with a [

I maximum final signal delay AT_max of Cms^set forth in the ^! following table־ j . TABLE I

I T5j׳pical Parameters for Speech Compressor

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i. , Chunk/Discard Ratio

<td colspan="2"> Comp. Ratio i</td><td rowspan="2"> Line Length d־ T_out (ms)</td><td rowspan="2"> (Playback cTime) ^in/^Tmax</td><td rowspan="2"> (Recording Time' ^out/^cATmax (ms)</td><td rowspan="2"> Sample Period T (ms)</td><td rowspan="2"> Rep Rate 1/T 1/P</td><td rowspan="2"> Cycles Sample (fmin= 333 Ηχ)</td>

<td> c</td><td> \</td>

<td> It !I. !&gt;.</td><td> י 2/9 '25 1 1</td><td> 6 2/3</td><td> 24/6</td><td> 30/7.5</td><td> 30</td><td> 33.3</td><td> 10</td>

<td></td><td rowspan="3"> 5 .2/5 1 1 I ^2/31 ^II 1</td><td> 7 1/5</td><td> 12/6</td><td> 18/9</td><td> 18</td><td> 55.6</td><td> 6</td>

<td> li</td><td> 8</td><td> 6/6</td><td> 12/12</td><td> 12</td><td> 83.3</td><td> 4</td>

<td> p</td><td> 9</td><td> 3/6</td><td> 9/18</td><td> 9</td><td> 111</td><td> 3</td>

<td rowspan="2"> 1 I⁴ 5׳</td><td> I 6/5 1</td><td> 9 3/5</td><td> 2/6</td><td> 8/24 o</td><td> 8</td><td> 125</td><td> 2 2/3</td>

<td> ) 4/3 1 1. 1</td><td> 10 .</td><td> 1.5/6</td><td> ־w/a/ao-</td><td> 7 1/2</td><td> 133</td><td> 2 1/2</td>

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The basis for the frequency-time transformation^ employed in the present invention can be derived as follows» Consider a sine wave V=E sin wt recorded with a tape recorder־ If the tape is played back at c times the original recording rate, the result is :1 i> !;

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V ־ E sin c <!>t where c is called the compression ratio.

If c>l the time is 9 ; compressed for any given speech passage and^if c<l_?time is

1' : expanded by the factor e where e = 1/c.

If the signal is then applied to a delay line in which the ‘1

I; delay of the line is caused to increase linearly with time at ;! the rate d, so as to cause the average delay of the signal to t;

] be c* which represents the delay any point on the waveform will ,: experience in passing through the line, then the signal (1) becomes

V = E sin (c - c*J wt (2)

(.The original signal is restored if the delay is c’t = c’t^to restore) ⁼ (^c³^ ־־^ such that ¹ d _c' = ד (c +1), the average delay rate of the line (4)

' which is derived as one half the sum of the final and li j! initial delay values of the delay line, which when multiplied

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j. by time, t, thus yields the total delay, c’t.

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Fig, 2(a) shows a plot for a given signal sample of signal . ׳ I output time, t_QUt, vs. the corresponding input time, t_in&lt;,Thus a

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line_ftwith a slope of 4.represents signal of four times the original frequency or speed of presentation, and 1/4 the -ד!MnWy periodicity while a line^with a slope of^ represents theן 'J resultant restored, or unchanged signal״ In order to convert j such a signal (as represented by the line I of slope c = 4) to ׳ one corresponding to line II with a slope of 1, and with.a I corresponding frequency decrease, it is necessary to increasingly delay the input signal, ct£_n! by an amount c't [or (c-l)t) as ' i 1 shown at line III־ Thus a signal chunk, T£_n! has an ordinate which intersects III at the ordinate value c'T£_n and this ₍ value when added to the time abscissa value^at the point cTj._{n cn} i

I, delays the signal to T_out on line II־ The delay dt [ introduced by the delay line is shown by line IV. Such a delay line has the effect of delaying the instantaneous signal, tj_, ' through a linearly increasing amount d°t for the interval from tin to t_out as shown by line IV־ Thus^as in the case of the end signal at time t=T one-half the sum 3 of the initial delay, dTin, and the final delay, dT_out yields an average delay value on line IV of c’T^, the amount required for restoration־ ; Hence, c'T£_n = ^aTin + ^dTout

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!d j (c-l)T_in = 2 (T_in + cT_in)

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I .c-1 ^d “ ² c+1

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In more general terms the restoration may be achieved by

<td> cumulatively</td><td> delaying an input signal, t^_n, by an amount</td>

<td> pout.</td><td> f(t)... dt = c't. ־־ (c-Dt. (5)</td>

<td> . ^η</td><td> tout.“ t^n</td>

׳ For a linearly varied delay line with a rate of change in delay,

1: d, f(t) = d״t and !! C τ x.

!; c’T_in ־־ (c-l)Tin = d t dt (6) • . A** / m ' ^{J in} *out - *in

-------}^T°^Ut ־ d (c² - l)T²_n^Tout ־ T_{in T}._n 2 _(c_ _1)τ^ from which we obtain (4) ־ c'T. ־= d (_C + 1) h

Hence, d 2 ־־־ c + 1 ? Thus for c>l, 0<d<2 ‘C<1, -2<d<0

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”191 ! In Fig. 2(b) the corresponding relations for signal expansion are shown« Line I_e with a slope of 1/4 represents i!j ||- a signal of 1/4 the original frequency or speed of presentation« p!

!j In order to convert such a signal to one corresponding to line II i . . I :i with a slope of 1, and with a corresponding frequency increase, i’ , . '

[| it is necessary to decreasingly delay the input signal, ct,-_n I !j .

i! /_ 1 >. _ 1 t . ׳ י ץ ־' e ⁱⁿ¹ 4 ־ⁿ⁾ ' :י l| by an amount c't (= — t = -|t) from an initial delay of . e 4 .j

I c'T_out׳.This amount of delay, c't* , at any point shifts the signal

J ' . :.I ;1 to the corresponding ordinate value on line II. The delay dt' introduced by the delay line is shown by line ΐνθ. Such a delay ' line has the effect of delaying the instantaneous signal by I |! a linearly decreasing amount d’t' for the interval from tj,_n to

!. | i t_out ^as shown by line ΐνθ. Thus as in the case of the initial i' signal at time, t=0, one half the sum of the initial delay, ן

Ί י ' i j! -d^־Ti_nand its final delay, “άτθ^ yields an average delay value ' I;

;j on line IV_e of -c’T_IN p .Hence, c'T_w= ^dTput ke_ ^T.H= f (Τ.Λ.+ h ©

1-e ώ = 2 1_+e

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The process of linearly increasing time delay cannot continue indefinitely, and from time to time the delay line must be ί returned to its original length־ If this process is repeated at periodic intervals, provided the interval.is longer than the

I period of the lowest frequency component of the signal, chunks . . i of the original signal will be played back at the angular frequency (c - c')w and the rest discarded. When (3) is ' i satisfied, the system operates as though sections were cut out ' of the original tape, pasted together, and played back at normal.

ן ־ . . ' ’ ' speed. The sections of signal are heard at the correct frequency but the. information is transmitted in a shorter time (if c>l.)־ The speech has been compressed to 1/c of its | original length.

• * ן י ί| The values set forth in Table I have been plotted in Fig3 ־(a), ! For any given compression ratio the sample time is given by the .

i ־ I j curve T ^an&lt;־* the chunk length is shown by the curve T^^־ The ;

difference between these two curves is the discard which is equal׳ ‘ ,i j to the final delay.to the signal at the end of the samplej ; period(6ms in the example shown in Fig.(3a). Entering the curve ! at any compression ratio, such as c= 5 in Fig.(3a)/one obtains.

the chunk and discard times for the tape running at c times;

i the recorded speed and these values projected to the time axis show the actual original recorded time for the respectiveן chunk and discard portions־ As indicated for c = 5 the chunk is j 1.5 ms long and the discard is 6 ms long representing respectively &lt; 7.5 ms of recorded and reproduced information and 30 ms of'

I ⁰ ־ '' I discarded information. This latter value is represented by : the quantity c AT_max which is also plotted in Fig. 3(a)!

For a speech signal in which the lowest frequency 333Hz has j a period of 3 ms, a chunk length of 15־ ms at c - 5 corresponding i . ־ j to 7.5 ms of recorded time will contain 25־ cycles of the 333H_Z1 ׳.

],signal. For any higher frequency components in the speech i.

signal more cycles will be contained in the 1.5 ms chunk. The i

' ! length of the chunk should exceed the period of the lowest frequency to be passed (i.e.,should include at least a full cycle) otherwise satisfactory compression will not be obtained. As indicated in Fig. 3(a) below the time axis at 3ms־, the 333Hz .

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J signal if processed at sample periods approaching 3 ms would₃;

i| with its samples reassembled, accordingly produce a compressed' >1I

j. output of poor quality since the sampling would then bej causing a disruptive, discontinuity for nearly every cycle of \ the 333 Hz signal processed־ Sample periods less than 3ms _t ;י would not permit completion of any one cycle so that the׳ i -' ^; . ׳ . ־-I 'j resultant reassembled output would not only contain the said ;; disruption but would also begin to exhibit a basic change in'

Iΐ its frequency characteristic in the form of waveform compression :!;

by truncatbn to produce false frequencies. While thisi *i condition does not represent a real condition for a speech'

ΊI

-ו wave due to the complexity of the waveforms, this principle׳

A is controlling and sample periods less than the period of the , ’! lowest frequency wave in the speech signal will not provide .:י proper compression.

i ! Sample periods greater than the period of the lowest * frequency wave will produce compression and an interval of ί f ’ i ־! disruption exists fromthe region where the sample period is only

Ϊ slightly greater than the period of the lowest frequency wave as i 'i indicated on the time axis between 3 ms and 6 ms in Fig. 3(a). יI ; The result obtained within this period of disruption is a:

!!

i distorted expanded wave in which the effect of disjunctions between samples becomes extremely severe as the single cycle

׳1ר • point is approached and diminishes as the number of cycles in , ! the sample increases. As a practical matter two and one-half ’ cycles per sample is indicated as the desired limit in Fig. 3(a) but in general the more cycles in the sample the less the disturbance factor.

In order to avoid.the extreme distortion produced by!

waves which have a longer wavelength than the sample period, ! . these lower frequencies should be filtered out before the speech ! 'ייj ί signal enters the delay line otherwise these disjointed and highly distorted waves will be propagated down the line and '' i ,intermodulate with the desired signal and may severly degrade j

the system performance.j ¹ י . . .I j For lower values of the compression ratio than c = 5, and, !

' ־ - ־I

I keeping ΔΤ = 6ms, the chunk length, increases with the result !1QQX

,. that the actual time sample increases to greater than 7.5 ms and ; 'I ; therefore more than the minimum number of cycles for the lowest i frequency wave component will be present in the chunk. Thus it ף would be at the user's option to operate the line over less .i .i j than the Sms indicated delay for AT_max to reduce the amount of

! ׳. 1 f;

discard.

' r.

; Considering the discard portion of the sample as a constant ' 6ms long at the compressed rate of playback, the actual* infomation loss is the compression ratio times 6ms so that

I ; with c = 5 the actual information discarded for each sample ׳ is 30ms of recorded time. As shown on the time axis of Fig. 2 this is the interval from 75־ ms to 37.5 ms and the relation ' of this loss of information to the intelligibility of the ! , ׳ i j reproduced speech signal must be examined״ ’ In general, human speech represents an extremely complex *I coding of a relatively limited set of sounds called phonemes!

,!

j which taken in context with the various attributes of the speech .[ code such as the voiced-unvoiced'components, pit.ch, formant !

j frequencies and the continuum of sound pattern represented by ‘ sound energy (and the absence thereof) connected by the all , important transititions between the temporal components thereof constitutes an acoustic stream of infinite variety and versatility־ The ability of the human ear to receive this ;

ί . 1 ’ acoustic message and the ear-brain system to decode the message

I I 'I is not altogether understood since it appears that the readily comprehended information rate far exceeds the mere acoustic j : response characteristics of the ear as a receiver־

Fortunately, the ability of the ear-brain system ta comprehend i the message which is conveyed by human speech signals is ΐ sufficiently good to permit large portions of the actual acoustic i ,.׳׳.

stream to be lost or discarded without significant loss in the

T perception and comprehension of the message information content

I Of the acoustic signal־ Since the comprehehsion of message * content decreases more rapidly than the recognition of individual

; ׳ י • words as the message is presented to the listener at increasing ί

rate, the problem associated with the discard of a portion of the

I signal stream can be resolved in favor of comprehension and short of the point where intelligibility of individual words deteriorates־ This latter point is reached where the loss or alteration of transitions or other cues representing the

; connection between a consonant and vowel sound results effectively ti . ^: !j from the discard of much or all of a given cue or cues so as .

&lt;1 n ;

!! to alter the apparent information content of contiguous I !

J concatenated chunks. Even before the point of absolute loss

11ί

H i ί! of intelligibility is reached the limit of tolerance due to ,1 . .i !1 discomfort for sustained listening occurs as a result of the

J ־ . i :! unnatural sounds and the fatigue which develops in the intense 11'» i .I concentration required in attempting to extract the information .i content in the presence of excessive time. clipping.j

For the purpose of speech compression the loss of intellig-

ז ’ .

I t ibility can be associated with discarding portions of the;

l. message containing significant'cues or phonemes wHch components '׳. vary in length with the shortest being approximately 10ms to J !

20ms long־. These short cues do not dominate speech but occur ' ן י with sufficient regularity to make their systematic loss ;

undesirable and hence a desirable upper limit for the discard period would be considered to be 30 ms and preferably closer : to 15ms־ With this limit set for intelligibility of the reproduced syllables and words the rate of presenting a given message can be increased to the comprehension limit for any i given listener and degree of difficulty of the subject matter ;j. with minimum concern for the limitation which would be imposed ,! by permitting loss or distortion of the word content or the ' I ' generation of false cues from the concatenated message chunks.

ן »

¹ . Fig. 3(a) indicates the recording time discard relation to ׳

I : compression ratio as the linear function cAT_max with the range »8 | ! from r5,ms to 30 ms designated the discard uncertainty range« ; Thus the Sms discard at c = 5 projects to include the real!

time recording interval from time t = 75־ to t = 375־ which;

1. ' . !־ .! approaches the upper limit permitted for discard without undue ¹ . .. i loss, of intelligibility as required not to contribute significantly to the loss of comprehension in the messagej ^:ί perceived־ Smaller values of c result in smaller actual discard 'i time and hence the intelligibility is improved especially for those cues which are at the lower end of the time scale, i.e־,:

: ' ¹, in the neighborhood of 10ms־ ii ! While Table I and Fig. 3(a) represent, parameters for aJ '! typical speech compression system having a final signal delay of 6ms and define the limits of operation within faHy narrow limits, . . . .i ' it will be appreciated that thejprinciples involved can be :

adapted for use over, a wider range of operation. Thus the

!. variation of the actual frequency band of the speech signal ' r ^,,־I ף and the maximum length of the delay line are both important / design factors which influence the selection of the chunk-to. discard ratio and sample period for a given range of thej ' compression ratio c. On the other hand, the actual frequency i ! ־ ׳ range of the signal has an important bearing on'the design of i i .!

the delay line which must accommodate the frequency spectrum

27present in the signal as to such quantitative and qualitative ! factors as the voice pitch,the presence of all or only some of the formant frequencies for an individual voice and the

J width of the signal spectrum over which linear phase-frequency ( properties must be preserved. The ultimate system used however ; will embody design choipes of the factors involved within the ' . . 1.

; broad limits herein defined. ! * . Fig. 3(b) is a plot of corresponding relations for signal i

! expansion showing the initial gap, output chunk and maximum . delay line length variation with expansion ratio e for a given j input sample interval τ&lt;_η. The output gap occurs at the . 1 i ; Start of each sample period and thereafter for the balance of ; the sample period the reduced frequency time-expanded output chunk appears. The maximum delay d T_in required is also shown 15 . as a function of the expansion ratio e.

I /

J

- וי One aspect of the'speech compression system described in .ין __________.^{with Fi}9. 1 has not been .treated.,namely, the audible _ ^of .^the tranducer 26 when'the variable delay processing ;i 25 is switched from maximum to minimum delay at the end of the sample period. Just prior to switching the delay line is loaded with the speech signal which is to be discarded and if the line is instantaneously switched to zero delay all of this information unless cancelled or predeleted, will be presented in highly condensed form in the '!output signal. As a practical matter with conventional delay lines utilizing R and L or C components there will be a time interval required for switching the line from maximum to minimum delay and it has been found i that even if the line does not contain signal information this ! .switching of a line has a significant minimum time constant

U) h 1 Ji . associated with it^produces a' disturbing transient audible in the output signal where the repetition rate of this transient is the reciprocal of the sample period־ Because of the limitations imposed by the parameters of the system as previously set forth herein, this switching frequency and spectral components of the transient itself will always be within the audio range and thus present as.a highly disagreeable !1 A '1 intermoduation component in the audio output of the device.

The’present invention provides a number of implementations for transient suppression and message gap bridging arrangements for the purpose of minimizing the disagreeable noise effects involved־ In more elaborate systems the substitution of pseudo or real message components further improves the transition from one sample to the next and can be adapted to fill in a portion of what is discarded in the compression process

Referring now to Fig4.־ a portion of the 333 Hz wave at the transition point illustrated in Fig1.־(d) has been reproduced in which cycle 4 and cycle 9 of the original recorded 333 Hz wave are shown as a smooth uninterrupted sine wave. The junction

29between the end of cycle 4 and the beginning of cycle 9 at point 41, although shown as a continuous portion of the sine wave, is in actuality, as previously stated, almost never so related in the ήοη-selective periodic sampling of independent complex waveforms and thus instead of a smooth transition point 41 a disjunction between the end of one chunk and the beginning of the next chunk in successive samples is to be expected. This disjunction could undoubtedly be accommodated with no loss of intelligibility if the transient from switching the line (either loaded or unloaded) did not have to be dealt with at exactly this point in time. Since this transient is responsible for a highly annoying audible output from the system it must be eliminated and for this purpose a gating signal e j as indicated in' Fig. 4(b) may be applied symmetrically with |i .

respect to the transition point 41 to produce the output signal [1 shown in Fig. 4(c). By making the gate long enought to :: encompass the transient resulting from switching the line, the ״ i^!

I!. audible noise so generated is eliminated. The improvement jj’ obtained by this expedient,while significant, is not ideal since 20 j; the introduction of the gate signal within the audio range is !1 !! itself audible as a repetitive disjunctive gap which inter!

modulates with the audio signal. This effect can be reduced by using an output filter designed for the particular repetition rate and gate width to smooth the abrupt transition shown in ! Fig. 4(c) and this output response is indicated in Fig4 ־(d).

-־

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:i

. ן

A further improvement is possible by using the gating signal ,as a gain control signal and tapering the off and perhaps the Ί

Jon transitions of th^e gate so that a gradual transition of the i 1

Jaudio output from off to on is accomplished and a .

relatively smooth transition as indicated in Fig. 4(f) results.

;The object is to minimize the gap effect which in itself has an !audio, characteristic and can act like a cue. Tapering the i trailing edge of the gate helps this considerably whereas an :anticipating start (or relative delay of the speech signal) would be preferable for gradual onset for the leading edge. ! ' . ׳ ' i’ ׳ .

:^-With these relatively simple expedients the smooth transition

i., .between adjacent chunks which arb disjunctively joined by the

J ' !

operation of the compression-disbard process are achieved in a

!. . .i manner which is satisfactory forj many applications.

. 1 '.:׳.

J.⁵ » Referring now to Fig. 5, the more elaborate arrangements , ;.for bridging the gap between adjacent samples will be described.

;As shown in Fig. 5(a) a disjunctive transition^which is the ΐnorm to be expected^represents a; sharp discontinuity in the : message signal and has superposed thereon the noise transient ! from switching the line as previously described. By introducing . ' ' j a gate signal. Fig. 5(b) of sufficient width to encompass the

I i . line switching transient and conditioning the gate to coincide ]with a zero level and same direction of change for the adjacent '! signals being processed a .zero level gating transition as shown

-חג Fig. 5 (c) can be achieved. This transition which is free . xו

I ' .!

: of line switching noise and essentially continues an existing 'i ,!i :!zero amplitude signal level during the interval of the gate has ;1 been found to provide little or no disturbance to the average 1 listener.

י'ו .

'.

:[ Because of the nature of the human hearing phenomenon, pari .’ .יticularly the ability of the ear| to synthesize the message it is, ' concentrating upon even in the presence of noise, it may be useful in certain circumstances to introduce a pseudo or real!

.i jmessage component in the zero level interval indicated in:

’ י ΐ 'i !Fig. 5(c). For this purpose suitably selected noise orI ' יi ׳signal components of approximately the same amplitude and

'. ' 1' ' I

,. frequency can be inserted in what is otherwise a quiet gap;

’ interval in the message stream and this arrangement of the invention • is indicated in Fig. 5(d). Where the gap is to be filled with i ׳ 'noise components, a suitable source and symmetrical switching ! to introduce noise from the source into the signal channel can . , 'i

II be readily applied during the gating interval.!

Fig6 ־ represents a preferred form of gap filling where two ׳ / . . i /signal controlled delay lines are used־ The speech signal is j ί i • applied to both delay lines designated channel A and channel B ; in Figs. 6 (a) and 6 (b) respectively and these two lines are

. . ' י . ף

I . / ; signal controlled to have symmetrical complementary gain :: characteristics and overlapping variable delay characteristics . ii as shown in Figs. 6(c) and 6(d)־ Here the delay control signals ^! as shown in Fig. 6 (d) are phased to overlap at least an amount • ׳ ' t :

• corresponding to the transition portion of the gain control i i ' characteristics of Fig. 6(c)־ The outputs of both delay channels

Ί .j A and B are combined to produce the combined output shown m :׳ i ’ _:: Fig6 ־ (e) ־

I Generally the length of thej delay lines used for channels i ׳ ‘'. I

A and B in Fig6 ־ will employ one full length delay line and one relatively shorter length delay line for storing the, signal used ,: for gap filling purposes־ This arrangement will ׳reduce the cost of the equipment' represented by the multiple section delay lines necessary to obtain the ' required maximum delay length fcjr system performance requirements־ . On the other hand, for systems where cost is not a primary factor, 'two equal full length variable delay lines can be employed and 4 their control signals can be alternately applied so that the I :;signal channel is through first one and then the other delay line : thereby giving a full signal period for switching the inactive ’ delay lino back to minimum delay condition prior to its use for .I ׳ .

' -33' signal transmission again־ For such symmetrical delay lines it

-י'.’׳'' וי i ΐ .י ¹: may still be useful to provide some overlap during the transition .ל.'. i I ״ as indicated in Fig. 6(d) with appropriate gain control signals t applied as indicated in Fig6 ־(c)־ . . h- . .sj ״ Referring now to Fig. 7, a'basic speech compression-expansion p ‘.

j; system in accordance with the invention will be described־ This ‘ '׳II ' !, system comprises a variable speed playback device 51 which is . i ״ indicated to be a tape transport with a manual select speed ; h ^! * i j; control input ,52. The signal derived from transporting the I

L tape past a magnetic transducer is applied to an AGC amplifier 53 which also passes the signal through a band pass filter having .an adjustable low and high frequency cutoff. The selection of the : cutoff frequencies for the filter may be operated from manual ! 'וי׳ י _י . . ׳ . i control 52 in conjunction with the selection of playback speed' . for the playback device 51־ The manual control 52 also supplies an amplitude control signal to a fine voice pitch adjust control * 54 which supplies on line 55 a iignal to control the end amplitude i!

of the linearly increasing waveform which controls the variable delay, line as hereinafter described.

The signal .after passing through the amplifier and filter ' | 53 enters a variable delay line 56 which can be signal controlled Ί between minimum and maximum delay limits. This control signal applied on line 57 is derived from a ramp level and amplitude . charger 58 which receives as its input either a compression triangular waveform on line 59 o? an expansion inverse of the ^: waveform on line 59 which appears on line 61 after passing through an inverter 62. One or the other of the lines 59 and 61 ;־is energized with a ramp waveform depending upon the setting of a switch 63 which supplies the basic ramp waveform from ramp pulse train generator 64. The repetition period of the ramp waveform is selectable by a manual control 65־ A pulse coincident with the reset of the linear portion of the ramp waveform appears on line 66 and is applied to a blanking pulse the width of which generator 67 to produce a blanking pulse output can be controlled by manual adjustment 68 and which is synchronized with the input pulse on line 66.

The output of the variable blanking circuit and amplifier signal depending upon the blanking pulse(B)appli from generator 67 and when the i----- , ־ th־ delay sl.j»al Is ־rpllod to'a speech bandpass delay line 56 is applied to a which transmits or blocks the ed on line 72 blanking pulse is not present(?) filter 73 the ' output ot which is applied «> «״ «prod74 ״־״

In addition to the amplitude excursion established for the linear voltage ramp signal from generator 64 which is controlled ;j by manual control 52 the absolute level of the voltage applied can be controlled by level adjust means 60

The variable delay line 56 will generally be of any known type may be 360 RC filter stages where the shunt and in particular resistor is provided : by a FET or other semiconductor device which varies resistance )

‘ in response to a controlled voltage or current« Such delay lines : generally perform best with resjbect to distortion of the signal . passing therethrough if the phase delay per stage is kept well below the maximum possible valu!s of 90°. Accordingly, the line can be designed to operate with 45° to 60° phase delay,per stage maximum and the number of stages is then determined as greater than the quantity: N>(6or 8) c(f_max)AT_max. In the .: above inequality the digits 6 and 8 represent the number of I ' ¹ .{* stages per electrical cycle of the highest frequency to be .L ־ . ' , • passed corresponding to a phase delay of 60°'or 45’, respectively

I

I

I i

i וas the maximum phase shift per stage which is to be utilized ;

the quantity c is the compression ratio; the quantity f_max is the highest frequency being passed by the line; and AT_max is the piaximum. signal delay desired as dictated by. the maximum .

permissible discard interval previously specified« Many other forms of delay line constructions which are capable of being

G^i! signal controlled are known in the art and the present

J - ן י .’! invention is not to be considered as limited to any particular ,! form of delay line. .

; Referring now to Figs. 8(a) and 8-(b) the operation of the r system of Fig. 7 will be described. The sample period waveform | 81 has an adjustable period set by control 65 for producing an . t , .

;1 asymmetrical sawtooth waveform 82 which produces a relatively

J ; long negative going linear voltage followed by a shorter ί I j! positive going linear voltage. This waveform is used directly i;

;׳ on line 59 for speech compression while, after inversion in

J ' inverter 62, its inverse is used on line 61 for expansion.

i!

The expansion waveform is indicated in dotted lines at 83 in Fig. '!׳.־ .I .

.;. 8(a). For a variable delay line 56 which increases delay as the control voltage becomes more negative, the waveforms 82 and 83 have the proper sense for controlling the delay interval and ׳the magnitude of the delay is determined by amplitude control relative to a voltage level set by the level adjust 60. Thus the operating point in the excursion of the waveform 82 is i selected for a given compression ratio in conjunction with the 20 : sample period which will be a predetermined combination for ί . . ⁰ any given compression ratio assuming the maximum delay 'i .

^{; &amp;}*max» .*ⁿ line 56 is a fixed value as obtained by selecting lino length according to the value d«T_out as given in Fig. 3(a) ' and Table I. for the desired compression ratio.

: If the maximum delay to the signal is not maintained constant

! ־ ן j¹ the discard period will change correspondingly as is evident

1׳ from the description of Fig. 1 and corresponding adjustments in i! the amplitude of the wave will be required to give the slope d . required for a compression ratio c, Similar considerations ! apply for the slope of curve 83 which must be set atits correspondir.c

׳ '1 .

i; . ' I J! value d for an expansion ratio e.j t ל .I :! The operation of blanking pulse generator 67 is shown to

I.I f produce a pulse 84 in Fig. 8(b) of predetermined width in r ׳ ׳ !i response to the start pulse of the sample period signal 81 t !; received on line 66. This pulse may be applied in gain control

J, fashion to the circuit 71 with modified trailing edge as

-I |: previously described to reduce the transient signal and provide a gradual onset of voice sound signals which are passed to the transducer 74. The width B of the blanking pulse is selected, with control 68 and is normally made of sufficient / duration to permit the short steep linear portion of the ramp waveform to return the delay line 56 to its zero or minimum ־יdelay condition and dissipate the signal energy therein (or the ^:.

transient caused by switching the line itself) prior to enabling . the signal channel which energizes the transducer 74 with the

J; subsequent speech signal segments־[

1: .

i:

J

I i -38!! .

The blanking period B and enabled period B for expansion , !׳ . ^!_t: mode are shown in Fig. 8(c)־ The expanded chunks with an initial. :1 output gap are shown in Fig8 ־(d).| ii . ,,. . ί ;j The system of Fig7 ־ can also be used to substitute noise, or '1 I pseudo signal gap filling signals corresponding to the system ί' . ¹ (! described in Fig. 5־ For this purpose a source 75 of such signals !! .' ־ !

I! is arranged to supply the input signal to filter 73 during the !!I blanking interval־. By means of a switch 76 this gap filling !!'!

'! during the blanking interval can be made optional. The gap;

! . . '׳.

filling signal 75 can. also be derived from the message signal .

« . . ־ ן! output of amplifier 53.

I .׳! Referring now to Fig. 9, a modified form of the invention (tI ii ’ particularly’suitable for accomplishing the various gap filling .!

. procedures for the speech compression systems previously!

:: described will be disclosed. Portions of Fig. 9 which are ״ essentially the same as those described in Fig. 7 have corres• ponding reference numerals and accordingly only the additions ? and.changes will be further described־ In addition to the variable delay line 56 a second variable delay line 91 receives the signal wave from aplifier 53־ The output of the delay lines ' 56 and 91 are applied respectively to complementary blanking !! . ; [ circuits 92 and 93. Signals passed by these blanking circuits ״ i j; 92 and 93 are amplified and filtered in element 73 and passed

1־ i to the acoustic reproducer 74 as heretofore described. ׳

I ' ־39־ i 1

I

I .

j* ^A pulse train generator 94 produces a pulse wave train as : shown in Fig. 10(a) having a selectable pulse repetition rate !

. determined by the setting of manual control 65 thereby £ establishing the basic sample period. The output pulse from!

generator 94 is delayed in delay unit 95 and applied to a first:

i /ramp generator 96 and in undelayed form is applied to a second ij־ i

;.' ramp generator 97. The ramp generators 96 and 97 are subject to ' . . .i

I waveform level control from manual adjust element 60 and ramp ;

.jI £ linear wave amplitude control from the manual adjust element 52.

. As previously stated, the fine pitch adjustment 54 may be pro.: vided for slightly modifying the ramp slope as h voice pitch' i

j. adjustment by effectively altering the frequency conversion ovex a

! ׳ ׳ . . ,,' :1 j: small range. In addition the blanking width interval of each ^: j,I j; generator is adjustable with controls 68 and 70 respectively. ׳ .־ .: The outputs of the ramp generators 96 and 97 are applied respective ;: ly to delay lines 56 and 91 to control the time delay of signals

’ ין.

ί passing through the respective lines in accordance withthe * control signals applied. By means of c or e select controls the , sense of the slope of the ramp waveforms can be selected for 20 _ compression or expansion.

, j: The level and amplitude controls for setting the respective ramp generators 96 and 97 are preferably relatively adjustable i;

to permit selection of the relationship between the two ramp !: waveforms. By making the delay and phasing of the unit 95 1!

²⁵ j adjustable any desired delay line overlap can also be achieved.

i It is also possible to rearrange the components to ;have the complementary gating at the inputs of the two delay i

. I i -40-

,י ‘’.

switched to be j lines 5S and 91 with the outputs^combined in a common channel to ;

amplifier 73. This alternative discards the portion of the ' speech signal that is not utilized by each line before it enters , ^: . . . . I ; the line and thus, eliminates the necessity for dissipating these portions when the lines are switched between active periods.

! Referring now to Fig. 10, the operation of the speech i ; compression system of Fig. 9 will be described. The pulse train . .I ; generator 94 produces the timing waveform of Fig. 10(a). This , i _| ί .pulse triggers the transition of waveform C2 in pulse rampן generator 97 which produces the blanking pulse indicated in ; Fig. 10(c) with the predetermined width of B and &amp; being ' determined by the blanking pulse width control 68. After the i delay indicated in Fig. 10(b) the pulse from generator 94 triggers 'the ramp generator 96 to produce the waveform Cl shown in . !

Fig. 10(b). With this arrangement the control wave Cl for the

I 'delay line 56 is overlapped in time by waveform C2 having slope . I in the same sense and bridging the steep return slope waveform of ramp wave Cl. With the asymmetrical time intervals shown in t ! Fig. 10, the arrangements for gap filling modes of operations

. י shown in Figs. 5 and 6 can be practiced. By making the waveforms j Cl and C2 have symmetrical rising and falling portions the

J arrangement is suitable for alternate switching of the lines ί 56 and 91 to provide alternate compressed (or expanded) chunks t ί

i ׳/./.

, -Λ ו _ ’ of the speech sample. The choice of the relative lengths of sample through line 56 and 91 will generally be dictated by

I '!

! manufacturing costs for the delay line. Thus for a main delay ji ן-line 56 of adequate length for the compression ratio desired,' a relatively shorter line 91 used only for gap filling purposes ‘ .1

I . will generally be more economical. On the other hand, two full length lines which are alternately active to pass speech sample ί 'י!

. chunks thereby providing adequate time for the non-active i '' ! line to be returned to its minimum delay condition will provide i

I : for smooth transitions, any desired overlap and the maximum ( time interval for discharging the line to minimum delay condition • prior to its processing the next speech sample. The action of the i system of Fig. 9 in the gap filling mode is indicated in

Fig. 10(d) and generally corresponds to that previously described with respect to Fig. 5. :

The operation of the system of Fig. 9 for speech expansion,

i. e., increasing the time duration for a given speech utterance ; and increasing the frequency components thereof from a : reproducer running at a slower than recorded rate is shown in ίX

Fig. 11־ Here the ramp generators 96 and 97 have inverted

I.

ן outputs to produce the expansion waveforms El and E2 shown in

I Figs. 11(a) and 11(c) respectively and the blanking waveform .I ‘i׳ j has been made symmetrical such that the delay lines 56 and 91 are

׳ !

! used,alternately for approximately equal periods. By the nature i .

i of speech expansion,a gap in the signal output will always ן occur since the lines are controlled to change from maximum delay

- וi ;1!

at the start of the sample to minimum or zero delay at the end ׳! I ¹י of the sample־ Thus when the line is switched to maximum delay י . !

) there will inevitably be a time gap before delayed signal emerges ¹t ' ׳!

. from the output end of the line־ Applying the control sequence ; indicated at Fig. 11 the speech samples processed by lines 56 and are overlapped so as to fill the gap as indicated in Fig11 ־(d) !i ׳ ;;ΐ : by the solid and dotted outlined signal chunks Ej-!and Ej?־ The .

presence of a slight overlap in the reproduced signal does not i

׳' significantly interfere with intelligibility since it generally _; is not noticeable and at worst may result in a slight echo I .׳ 'i effect of the type commonly encountered in a telephone con. versation־ The time expanded speech waveform obtained using the mode of operation indicated in Fig11 ־ is useful for the!

:/recognition and comprehension of difficult passages and for ; analysis and study of foreign languages and the like־|

[ The system shown in Fig12 ־ represents a simplification of the • '1 system of Fig9 ־ where a fixed delay line 101 is used in place of . the second variable delay line 91 of Fig־9 ־ The control of ¹ blanking circuits 92’, 93' is simplified in that the variable ' width blanking gate B as derived from pulse train generator 94 U .

Correspondingly produces gaps in the output signals which have been ' . . ΐ i delayed by passage through variable delay lino 56־ The fixed I i

I ]^,:delay of־ line 101 is selected to further delay some portion of the signal emerging from the delay line 56 by an amount sufficient !to fill the gap caused by blanking pulse B thereby essentially i, ' repeating some portion of each message chunk while the variable :delay line 56 is switched back to its minimum delay condition־ h :

];Again, this repetition is not objectionable and may merely . . i; . _־ ji introduce a slight echo effect which is much less objectionable j;than the presence of the gap in the message signal. This sequence ’¹of operation is shown in Fig. 13 where the variable chunk C_vI;

.and the fixed chunk Cp alternate in supplying the output.

The expansion mode for operation of the circuit of Fig» 12 is shown in Fig. 14 where the ramp signals are inverted for the • >!

I, expansion waveform which controls the delay line 56 to vary from i;

:maximum delay to minimum delay over the linear ramp portion.E shown in Fig» 14(a). The blanking waveform B is selected to pass some portion of the signal chunk through the appropriate amount of delay to fill the gap between chunks in the output as • indicated in Fig. 14(c), Thus the output is composed of chunks Ep and E_y in alternation for continuous signal.

' il ji

. . . ו--

י. .1 ![ The system of Fig. 12 could be further simplified by jeliminating delay line 101 and conditioning gate 93' to ; introduce in the gap interval any pseudo or noise signal from . a suitable source which would simulate the frequency content ;י of the actual speech signal. While this version would be less jdesirable than using the actual speech signal for gap filling Jit would, nevertheless, be better than reproducing the .I speech signal with the message gaps present since the audible ' l&lt;

;; effect of gaps becomes detrimental to recognition of the message 'i ,content, especially at high compression ratios. This modification t ; gives a mode of operation similar to the optional, noise gap ; ' jf filling described for Fig. 7״ 'ί ׳ ' .......... . .. :----------------

ן ו

: 1' ί

I

I ΐ 1

Fig15 ־ shows a modification of the invention for binaural ; processing־ The speech signal from the band pass filter . 53 is applied to symmetrical variable delay lines VDL1 and . VDL2 controlled by waveform generator 102־ The output of : VDL1 is applied as an input to gates 103 and 105־ The output of VDI«2 is applied as an input to gates 104 and 106־ The delay line VDLj. is cohtrolled for linear variation of delay according to the waveform of Fig16 ־(c)־ The delay line VDL₂ is controlled for linear variation of delay according to

I® * the waveform of Fig. 16(d). Each of these waveforms has its ' rapid return transition at the mid-point of thelinear delay portion of the other waveform.

The gates 103 and 106 are cohtrolled by gating waveforms and B| shown in Fig16 ־(e). Gate 103 passes signal during and is blocked during Bp Gate 106 is blocked during and passes signal during Bp Amplifier 107 combines the outputs of gates 103 and 106 and applies the combined signal to an audio reproducer 108.

The gates 104 and 105 are controlled by the gating waveforms

B2 and B2 shown in Fig. 16(f). Gate 104 passes signal during *B2 and is blocked during B₂־ Gate 105 is blocked during B₂and passes signal during B₂. Amplifier 109 combines the outputs

j. of gates 104 and 105 and applies the combined signal to an audio ' reproducer 110.

The system of Fig15 ־ operates to reproduce the entire

I“׳

i.

J '.

i ''

I.

original signal (for compression ratio equal to two) since each delay line processes the portion which is the discard for the other line as is evident from Figs. 16(a) and 16(b).

; For compression ratios greater than two some message • discard occurs and for ratios less than two the overlap or ¹ ' : message duplication increases in the output. By.listening : binaurally, however, the intelligibility is enhanced since the ¹ overall discard is eliminated (or greatly reduced for the i higher compression ratios) and the overlap or repeat of message portions is not detrimental to word detection by the »

. listener.

A binaural system without supplemental gap filling (as just described) would be achieved by removing gates 105 and

106 in Fig. 15. The linesVDI^ and VDL₂ would supply the processed signal in alternation to the respective output ' ׳ transducers 108 and 110 for binaural output.

!ן Fig» 17 shows the invention using a form of delay line i! capable of processing speech signals in a manner which greatly י reduces the problems associated with discarding stored !

ן; information in the line. The system shown in Fig. 17 comprises * an analog shift register having a plurality of stages ASR_lf ASR₂, 11 ־' ASRj^, which has a speech signal input applied on line 111 and a !!

j: compressed or expanded speech signal output on line 112» ;j Alternate stages of the delay line are clocked by two-phase ij clocking signals applied at lines 113 and 114 as derived from a ^! , shift frequency generator 115. The frequency variation of the ;! generator 115 is such that the inverse of the clockingj .ί

I! frequency., namely, the pulse to pulse perio^varies as a linear ¹$ function of time with the frequency varying from high frequency ^! זזן, to low frequency for compression and from low frequency to high l frequency for expansion mode operation.ΐ . ׳ί i The analog shift register shown in Fig. 17 is'of the general :. type disclosed in the article by p. l. J. Sangster published in ; the 1970 IEEE International Solid States Circuits Conference *

; .... .. ׳ . !;.

: Proceedings, pages 74-75 and 185. such shift registers sample an analog signal and pass the sampled value along the line at the clock rate by a charge or charge deficit storage mode which permits the signal sample to be recovered at the output of the

I delay line after a time delay which is proportional to the ! 1' .j

1: clocking rate. In the present invention, wi^h the clocking rate'

I •

!;I

- | : varied so that the inverse thereof is linear function of time, !i !: the delay line operates to expand or compress the speech signal |i and by tailoring the length of the line and the repetition rate !! of the linear control function in accordance with the principles 'j _ י י ij herein set forth, continuous processing of random speech signals is achieved. In the compression mode at the end of each linear !! segment of the control function generated by generator 115 all

I , i^: stages of the delay line may be reset if a reset input is

I available or the line may be simply emptied during the blanking ¹⁰ period as the line is reloaded with the beginning of the next

J ^II segment of the speech signal at the high frequency clock rate.

t This rate may be made rapid enough to make the required blanking

י.־'.^,' .

so short as to be aurally imperceptible. Thus the problems associated with gap filling or smoothing or blanking may be .

; minimized in this version of the invention.

The design parameters for the analog shift register (ASR) can be established by applying the criteria previously _rset forth, as will now be described.

The instantaneous delay τ(t) of an analog shift register ' at time t is ז {t) = dt + τ_ο where d is the rate of change of delay and τ_ο is the initial delay. The delay dt becomes in the ii ASR for N stages N ° - where f_t is the frequency of the

J ^{P ft} ף clock shift signal at time t. Defining N _ as N , the delay I! . ־ . P ί becomes τ (t)= N1 + -) ־*־ ) where f_o is the initial clock i ^fo shift frequency.

*! The slopes of the time delay function for restoring the i;

} original speech frequencies are the same quantities as before.

.:, For compression:

.i c= total delay to signal entering at time t I! ^v = (c-l)t'for compression ratio. . c, and ׳ ' c-1 _ Ν' /1 _ 1 \

L ¹³ = י c+1 ־־ t f_ol 1׳

For expansion:

: C׳t= AT_t = 'd = 2 j; . Hence the of time multiplied by Ν', produces the delay needed to obtain

,. . . .I

1! compression or expansion of the speech wave with original!

I:

speech frequencies restored.

I! ''

׳ ' !!

I * i 0 \ / I i

1-e . . j t for expansion ratio e, and

1-e N* /1 _ 1 \

1+e “ t ί It r_oj inverse of the shift frequency as a linear function

I

Referring to Fig18 ־, the reset time t_M to clear N stages is which represents the time needed for the first

N pulses to refill the line־ By suppressing the sample-switching transients by filtering or blanking or any of the other methods herein described, and holding t^ to less than 02־ms the effect of gap intermodulation previously discussed becomes

׳1 |: virtually imperceptible.

.!

!: There is a limit to be observed for the sampling clock (' I ' ij frequency to accommodate the highest signal frequency f_max to be !

׳ י יI (passed by the line־ Referring to Fig25 ־ the signal frequency '! ״ | ( for compression decreases linearly as it passes through the delay i . .j line .as indicated by line 201. The clock frequency varies as a

I! .i hyperbolic function indicated at 202 and must throughout the samnle ;:period be equal or greater than the values on line 201.for which the relation is given by f. ; 2p _f’ I . ^J t p-1 ¹max ^,(which for a two phase ASR is 4f_maxJ to provide for at least two ;; samples per cycle of f_max throughout. . i

I .

i! ' ί ’ . יי

I 1 ΐί

I;

1! Fig18 ־ shows a modification of the analog shift '!

' register embodiment shown in Fig17 ־, but with provision i01 ׳' the exact substitution of processed signal at the initiation

I! of reset for the delay line־ The analog shift register delay , ; line 121 processes input signals from line 122 in accordance j: with the variable pulse shift frequency derived from square wave generator 123 as previously described in connection with i'

Fig. 17־ The pulse frequency is such that the pulse spacing j! varies linearly as indicated from pulse signal source 124 where י, the reciprocal of frequency is shown to be linear with!

i respect to time. At a point 125 on the analog shift' register the line branches to have duplicate shift register stage paths 126 and 127. The number of stages required in each of the boxes represented by 126 and 127 is sufficient to continue processing the signal while the line 121 is being reset־ The branch outputs from stages 126 and 127 are subject to complementary gating controls 128 and 129 ,i ' and when gated to pass signal both supply input signals to a combining amplifier 130.

I .

! .

.-521

If In addition to the control of the main line 121-126

I . from generator 123 and 124 the branch line 127 is controlled . through gate B 131 from pulse generator 124 which triggers |!

jj a second square wave generator 132 and when gated during the j| _י B interval by gating unit 133 the trigger rate for generate” 132 |!

is derived from a fixed pulse generator 134. The pulse .1 Xו l| generator 134 may also operate at the pulse rate -, ;1 X. f’

1. .

j! The operation of .the system of Fig. 18 can be describ°d !1 with reference to the waveform associated with output line 135,

II jj For a given sample period the variation in frequency of generator .j; 124 starts and controls the ASR line 121 as previously described, ן; For this condition the gates B are permitting signals to pass I! and the output of stage 126 is transferred to the input of amplifier 130 thus producing the frequency converted output signal indicated during the sample period of the waveform.

_!

; At the same time the B control for gate 131 permits the same ' P control pulse signal from generator 124 to trigger generator !j 132 thereby keeping the branch'ASR stages 127 operatingj :! in synchronism with the corresponding stages 126. The blanking !I control B in gate 129, however, prevents the output from stage ij' !1 127 from reaching the input of amplifier 130. At the blanking |- or reset period for the main ASR line 121 and generator 124 the

I.Vί

B and B gates switch thereby interrupting signal flow from !

. stage 125 to amplifier 130 and permitting signal flow from

I' stage 127 to the input of amplifier 130״ Since stages 126 and, | 127 were in synchronism this switching will be of identicalj h signals and thus imperceptible at the output line 135 of the| j: amplifier 130. At the same time the B and B switching in gates . ΐ '! 133 and. '131 interrupts trigger pulses from generator 124 and ;1 i !1 '! passes trigger pulses from generator 134 to the square wave1 ;! generator 132, This switching of control assures that the ji 'generator 132 will continue to process signal m stages 127ן

1® j: while the generator 124 can be reset for the start of the next !! sample״ At the end of the blanking period there will be some ji 1 - . . , ־ _-,I discontinuity as the gates B and B switch back to theirj

L original state thereby returning control to pulse generator 124 ;־ inasmuch as the start of the next sample period will not

1' . .

. ^produce signals which exactly coincide with_the signals terminated ; . at the end of the blanking pulse which were under control of the !/'pulse generator 134.

;; Fig» 19 shows a modification of the present invention in ' . !

' which the variable delay line under the control of the variable 20' 1 . .i j; frequency generator 136 operates with a J repetition rate!

ן control function exactly analagous to that described with j' reference to Fig. 17. In Fig. 19 instead of carrying the

¹י י . ' . ^: :

j analog signal through the successive stages of the shift »

I-54I :׳ register, the input signal on line 137 is first converted into a digital word in an A/D converter 138, the parallel output i! ’ ־ : of which applies a parallel word to the input registers of the Ji first stage 139 with this digital value transferred sequentially i through the serial stages until it reaches an output D/A ij j converter 140 where it is converted into an analog signal on ^!i .

; output line 141. This operation is entirely analagous to the ;1 ί system described with reference to Fig. 17 except for the coding .1 of the information as it passes through the serial stages which are driven with the variable clock frequency to provide the :i frequency conversion required. One advantage of the system of Fig. 19 is in the provision of a reset signal from generator 136 on line 142 which can be applied to all the registers of all the stages simultaneously thereby effecting an instantaneous clearing and resetting of the line at the end of the sample ־ period.

Fig. 20 shows a modification of the invention which is analogous to that shown in Fig. 19 except that the digital signal is processed serially through a serial shift register 150 after the digital output of A/D 138 has been serialized in a serializer i_; 151. The shift register 150 is under the control of the shift frequency generator 136 including a reset input line 142. The 1' .

&lt; output of the serial digital shift register 150 is applied to a f .

!; parallclizcr 152 which converts the serial bit string into l. _

1'

־551.

i

I

I

׳1

l.' I

I.I hI ? . .I

!. .ן .׳ ף a parallel digital word for conversion by D/A 140 into the !:I

I, required analog, output signal on line 141.!

!j ‘I r1 £ Fig21 ־ shows an embodiment of the invention in which an

I*׳ ין. analog storage matrix with provision for addressable read-in and ,I

-ו. read-out signal storage is provided. A charge storage matrix &lt;ו

:. 161 is shown haying a plurality of X write lines 162 and a

1’ second plurality of Y write lines 163 the intersections of which !' ׳ i define the matrix address at which are located analog storage ״ elements. Typically, an analog storage matrix will haveί ' I ;:. a capacitor charge storage device at each intersection of!

. i ; the X and Y lines defining the matrix for storing an analog ¹ ’1 ; ׳i ' value represented by the charge on the capacitor.ן • ί

Bach such memory location is also accessed by a plurality of

X read lines 164 and corresponding plurality of Y read lines 165 where the intersection of the lines 164 and 165 correspond with • ,the location of the charge storage elements located at the intersections of the write lines 162 and 163.i

To store an analog signal in a charge storage memory 161 ^:_: an analog signal input on line 166 is applied and the;

I: 'i instantaneous value thereof is stored in the charge storage :* element associated with the coincidentally energized X and Y write ♦

1:I ! line intersection. from an X counter-write enable 167 and

I’ .

׳ a Y counter-write enable 168. Typically the X and Y counters ׳

167 and 168 will be operating at a predetermined pulse rate ־56V.

I

II. .

I li

It .

' ί derived from a pulse generator 169 with the number of .ן pulses in the X line sequencing the X write lines 162 after which j: the Y counter 168 is stepped and the next row of X intersections with the then active Y line being energized by the next sequence ,: of pulses from the pulse generator 169־ Accordingly, the memory ' i; ' . ׳i

161 has a storage capacity of X x Y number of storage elements ϋ1 . י :!I i corresponding to the number of intersections of the X and Y lines.

' With pulse generator 169 operating at a constant frequency this j write-in of the analog signal on line 166 occurs at a pre- J .^: . . .I determined rate and storage capacity is selected to store!

a signal sampled in accordance with the general requirements ⁰ !

• hereinbefore set forth.I

A frequency converted output signal is derived from output line 171 which receives serially from the charge storage elements , of the memory 161 the analog values stored therein as the matrix ' intersections are sequenced in regular order by the operation of ; X counter-read enable control 172 and Y counter-read enable :

:I . control 173. The pulse rate.for counters 172 and 173 is| '!

ו; determined in accordance with a ramp .voltage generator 174;

״ controlling a voltage control oscillator 175 with the variable' !; rate selected to produce the desired signal compression orI ii .!

expansion in accordance with the principals of this invention. ^:

I .

I .I ‘1

Γ f

ז

:⁴ ׳ i

I:

^!; . 1

For this purpose a rate control potentiometer 176 is provided for i! ' :

; selection of the ramp voltage slope in generator 174 and this '׳ slope will generally be a gang control with the tape playback I ! . , _ ! j; speed control 52 described with reference to the embodiment shown ;; in Fig. 7. This dual control function is indicated by line 177.

i: A further control from the rate control potentiometer 176י

'. is applied on line 178 to the pulse generator 169 to control its ; pulse repetition rate in relation to the maximum read-out rate ,i

j. established by control of the ramp voltage generator 174 and oscillator 175. In particular the writing pulse rate must j

I .. ! be maintained higher than the maximum read-out pulse rateI i

I.

in order to avoid having the read-out sequencing overtake the 1 write-in entry. As soon as any storage element has been read it is available storing the value on the next signal sequence and can be either reset upon read-out or upon entry of the next write signal. The ramp generator 174 provides a reset on

- '....... i ־line 179 for resetting the counters at the end of each ramp voltage period for the start of the next signal sample storage sequence.

Fig. 22 shows a system employing a random access memory

I 181 with write and read controls 182 and 183 which operates in 1 ! a manner analogous to that described for the'system of .Fig. 21. Since the memory 181 stores binary information however the

I. input signal on line 184 is required to be converted in A/D ’ converter 185 and the corresponding output must be converted in ; D/Λ converter 186. The sequencing of write-in and rcad-c11׳t for the memory matrix generally corresponds to that previously | I

I described for Fig. 21. ..

Where gap filling is employed a particular means for minimizing the disturbance caused by the discontinuity at the

I beginnings and/or the ends of the signal samples is illustrated j in Fig. 23 with gating signal control indicated in Fig. 24. The j logic control and sequencing elements therein are so configured ¹ as to cause the primary signal sample 191 to end at a zero i ! crossing and the supplementary gap filling signal 192 to start

J at its next zero crossing in the same direction and then at

I .

i the end of the blanking period for the primary signal for said

Ί. supplementary signal to end in a zero crossing to be followed j by the new primary signal sample 191 at its next zero crossing

I .

! m the same direction. Thus, the source signals 193 and 194 : after filtering through, low pass filters 195 and 196 to remove .j processing and spurious high frequenc;^ components are fed into

I '

j. their respective gates 197 and 198 and voltage comparators 199 and 200, the latter units so connected to ground by directional ; circuitry 201 and 202 as to trigger pulse generator(PG) 203 i . .

or 204 whenever a positive going zero crossing by the respective ' signal 193 or 194 occurs. Gates 197 and 198 are actuated so as i.

! to pass signals 193 and 194 by the reset output lines 205 and

206 from flip flops 207 and 208. Flip flop 207 is set by a pulse on line 209 from gate 211 as conditioned by the reset i - output line 213 from flip flop 208 and by the inverted outout t*

216 of the sample period pulse train generator 219. Flip flop

207 is reset by the pulse output of gate 217 as conditioned by

59; the direct output 215 of the PTG 219» Similarly Flip Flop 208 I ί is set by pulse 210 from gate 212 as conditioned by the reset . •1 ij output of line 214 from flip flop 208 and by the direct output . . i

215 : '—־ of PTG 219־ Flip flop 20 8 is reset by the pulse output

I .

.j of gate 218 as conditioned by the inverted output on line 216

Ϊ of PTG 219־ Gate pairs 211 and 217, or 212 and 218 are pulsed

J by the output of pulse generator 203 or 204, respectively, ף whenever a positive going signal zero crossing occurs as Ί .! described above. Thus with flip flop 207 in the set condition

״ . 1'

1G allowing the primary signal 191 to pass and with flip flop 208

J in the reset condition blocking the filler signal 192, when PTG

'. output line 215 goes positive as at 220, gate 217 allows the | next pulse from PG 203 to reset flip flop 207, thus blocking '1 primary signal to line 191־ At the same time, gate 212 is

IS conditioned to pass the next pulse from PG₂ 204 to set flip flop :. 208 and allow the supplementary signal to pass until the end . of the gap period. At this time the inverted PTG output 216 goes positive as at 221 allowing gate 218 to pass the next ' pulse from PG₂ 204 and reset flip flop 208, cutting off the 20 supplementary signal 192 and conditioning gate 211 to pass the next pulse from PG 203. This sets flip flop 207 allowing primary : signal 191 to pass to amplifier 222 and be outputted at 223־ J ’ .; The process then repeats in the sequence described above־ י

וi *

- ־ feifiaurallyT-h^w^^^ r....^6-44^^144^349-14¾ ty in enhan-eed—aa^ccHh^ overall discard is eliminated (or greatly reduced for tix'nigher , compression ratios) and the overlap or repeat ofpdssage portions is not detrimental to word detect^en by the listener־

A binaural system without supplemental gap filling ( as distinguished from the system idst described) would be achieved / . 1 by removing gates 5 andjrin Fig־1 ־ The lines VDL^ and VDL₂ ; would supply the nancessed signal in alternation to the respective output transducers 108 and 110 for binaural output in a system otherwise generally similar to that shown in Fig9 ־ of the »Derain rcforonood oo-ponding applicxartien, *i

Referring now to Fig־ \ a dual delay line system employing analog shift registers having separate read and write clock rates will be described־ As shown in Fig. X an input line Ή. receives a sound input signal from any source such as a tape recorder driven at a speed different than the recording speed or some other signal source supplying a sound signal message which it is desired to convert as to frequency components and ! also change its time duration from zero to some longer or shorter time than the normal period during which the sound message was originated. The signal on line ^,¾4,, is controlled :

' !

for application to an analog shift register ASR^ by passing . *2.33 '!

through a gate G24&lt;and is also controlled for application to an analog shift register ASR₂ by passing through a gate G

The outputs of the analog shift registers ASR^ and ASR₂ are j ־ΖΊ.Ζ ‘I combined on an output line X by passing from the output of. '

I .ί &lt;Q. 'I

5.

2.35 . ;

ASR, through a gate and from the output of ASR₂ through ׳_o a gate

The analog shift registers ASR^ and ASR₂ are multi-stage registers adapted for passing the signal input thereto stage-by־ ; stage to the output with the transfers occurring at clock rate determined by the clock pulse rate applied at clock terminals ! 3333 2־© J

2^and'7־a, respectively. The number of stages in the arrangement, of the analog shift registers to transfer analog signal samples therethrough are all in accordance with the description given in applicant's co-pending application. In particular, a write pulse rate generator supplies an adjustable write pulse repetition rate through G-gateclock input 2^ and through _ 3+7 ί

G-gate '*ST. to clock input terminal 2-$. A read pulse rate generator S₂ supplies a relatively fixed read clock pulse rate through 3+3 2.37 7‘H-3*38

G-gate '3^ to terminal 2^ and through G-gate X to terminal The gates G and G are supplied by a gating pulse generator Sj!

which may have an adjustable period and produces substantially.

symmetrical squarewave output for both the G and G gating functions .

The write pulse rate generated by the generator is variabk as indicated and will generally be set in relation to a variable ί

.... ... ־5^.2 . speed control device 5^which controls the rate at which the tape recording or other sound signal source reproduces the sound !

I message signal at a time rate different than that of the original .

2Jf5 j speech utterance. Thus, if the speech control is set to ! play a tape recorder at twice the normal speed, the write clock i pulse of generator can be set at twice the clock rate of I generator S₂ ! thereby providing write-in at a clock rate which is twice the rate at which the information will be read-out as the generators and S₂ are alternately applied to control the shift registers. If desired, a feedback control can be

S iⁿ accordance with an error signal to compensate for WOW and flutter characteristics of a turntable or other periodic .variation in the signal source which is desired to eliminate.

applied at to modify the clock rate of generator .2.0

The frequency of squarewave generator S can be adjusted

2.4-7 ' ³ by control Ή and generally its period T will be given by the relation T equal ¹ equal to or less than ^P-1 χ ^N where P is the phase of the analog shift register (e.g« two phases per stage,for example) and N is the total number of stages. For expansion in order to avoid gaps, the period T should be equal

P-1 ν' ‘Z.q-'?

^to__^x ___ « For this purpose the frequency control of

T f!

^s3 may be tied to the manual control for S during expansion

X 0 operation.

In addition, trimming controls 38 and 39 may be provided and interconnected for the generators S₉ and S_ if desired.

Operation of the system of Fig. will now be described with reference to the waveforms of Figs, X^a) and.(b). The general principals of operation set forth in applicant’s parent appliestion. are applicable here and the compression ratio C that is obtained will be equal to the ratio ' ¹ where f and f are f 12 the frequencies of the squarewaves generated by generators and S_2< respectively, obviously, for expansion,C is the fractional quantity and corresponds to the expansion factor e mentioned 23¢ in the parent application. Input signals arriving on line are gated through gate >•3. to load ASRj^ during the G-gate portion and the stages of ASRj are filled at a rate

2.37׳ clock squarewave on line *M. which is derived 24./ through G-gate During this period there shown in Fig־\(a) determined by the from generator is no output from ASR, but to assure the absence of spurious ¹ _ Z3S or noise signals on the output, G-gate blocks, signals from ?32.

the output of ASR^ to output line generator S, supplies clock pulses on 23C

ASR-n and the G-gate passes signals

232.

to output line

When the squarewave generator S-j

During the G-gate the line M. to shift register from the output of ASRj changes state the G-gates are conditioned to pass signal and.the

G־־gates are conditioned the G intervals indicated to block signal passage. Thus, during

231 in Fig. \(b) , signals on input line*^

34־2 >4 into ASRo at the clock rate of generator S. applied through

ΖΨ2 238

G-gate M on line >8, and the signals stored in register ASRj־ are • _ ^Λ Z.3S....... 232........ י י ־ — through G-gate to output line ^,2¾ at the rate of

...._.....2H . _ ..... .

S2 applied through ־ G-gate 3A. to line 7¾. Thus, on half cycles of G and G depicted in Fig. \ the input are loaded through gate outputted generator alternate alternately stored in ASR^ and ASR₂ and while storage place in one of the registers, the storage signal

.32־2 the other register is outputted to the output line The signal is is taking in rates at which these take place, are determined by the repetition rate of generators S! and S_2׳ respectively,and as previously described, for different frequencies of these generators either compression or expansion of the signal wave on line 2^can be obtained at the output line X,. .

«F-Hie

Thus, in accordance with the present_ftinvention_ja further form of signal storage delay for frequency transformation is provided using analog shift registers operated at different input and output clock rates. This arrangement permits processing 231 of analog signals on line*M. including complex speech waves and the like without the necessity for digitizing the input signal or cftherwise conditioning it for the delay and frequency transformation processing. A further advantage of operating the analog shift registers at different input and output clock rates as opposed to operating them as variable delay lines, is the elimination of the need for an inverse frequency control function generator such as the Unit 115 disclosed in Fig. 17_o«־&־ the r-cforonecd oo-panda-n-g— &pp׳i»eation o-f-appl-ieanfe׳> In the e^/cd.O< ^^*2.(5,

Peasant application of thshift aregaotier h the'clock rates are fixed but different for the input and output control to the analog shift registers and the ratio of the clock rates directly determines the compression ratio (or expansion ratio) which is applied to the signal which passes therethrough.

While the invention has been described with reference to frequency-time transformations of the original signal, the disclosed embodiments are also useful for.frequency transformation, where required, due to other factors such as a change in the velocity of propagation of the sound'waves. For example, a human breathing in an artificial atmosphere such as θη© having a high helium content speaks with a higher than normal voice pitch but with other parameters substantially unchanged. By using the speech compression modes provided by the present invention, such speech can be restored to its normal frequency range without change in time scale.

Obviously, the methods and apparatus herein disclosed can be used for coded audible signals other than speech such as music, for example, by taking due consideration of the corresponding parameters significant to comprehension as herein disclosed. Many other modifications can also be made without departing from the scope of the invention as defined in the appended claims.