CA1262973A - Audio channel stacking with speech compression for narrow band transmission with provision for dialed calls - Google Patents

Abstract

ABSTRACT
An analog signal stacking method and apparatus permits transmission of n analog signals of given spectrum width over a narrow bans communication circuit having bandwidth approximately equal to the spectrum width of one analog signal by compressing the spectrum of each signal by a factor of l/n and frequency offsetting the compressed signals to occupy contiguous frequency bands within the bandwidth of the circuit. Received signals are processed to remove the frequency offset and expand the baseband signals to restore each analog signal to its normal spectrum. A
dial through circuit permits application of the stacking feature to place two conventional dialed calls on a single line of the telephone network using a circuitfor processing both dial and voice signals.

Description

~35~2 ~26~

PmDIO C~Y~NEL SrACKING WIIH sæ~cH C0MPP~ESSION F~R
NU~ BAND TRAN5MISSION WITH PRGVISION EO~ DI~LED CALLS
K~rle:Le~r[o.~
~ his invention relates to communication systems for sending infonmation ~y transmission between stations ove~ a ~imited band ~ircuit such as, fo~ exam~le, a telephone line or teleEhone voi oe grade circuit. lhe invention ~rcvides for sending a plurality of seE~rate messages each of which may occuEy the aYailable knc~n bandwidth under nonmal conditions but ~y ~eans of the pre æ nt i~vention a plurality of such messages can be sent over a `i comml~ cat.ion circuit whi~ ~as a bandwidth ~pproxima~ely the same as that of the individual ~essages. In particular the m vention is adapted to send a pluLalit~ of telephone voice messa~es o~er an ordinary dialed teleE~one line or circuit which h s approxImately a 3 k~z bandwidth~
There exist pre æ ntly various digitized voi oe transmission syst~ms which increase line utili2ation by data s3mple rates o~
48Q0 baud for bwo simultaneous voice channels on a 950~ baud privat2 line, but such systems involve sukstantial digital oonputer caFability with the resultant high cost and ~pai~ment in quality of the voioe signal, E~rticularly with regard ~co s~eaker reoognition.
Analog voice multiplexing is also emF~ ed for increasing cc~r~nunication line utilization through a technique kncwn as Time Assigned Speech Interpolation (TASI) in which the Yarious conversations are analyzed to insert the multip~e messages for one chanr~l in the gaps during pauses and behYeen words found in the 1.
other messages. G~3erally 'chis manipulation is d~ne in digital form with the result reoorIverted to analog for trar~missionr there~ eliminatiny the need for mod~ns c)r data s~ts,, ~hese .. ~,~
I

systems require substantial data processing ~ower and hen oe only become economically feasible when a large number of lines are prooe s~ed.
Bandwidth reduction by speech compression in real tLme has been suggested for telephone lL~e transmission as shown, for example, in the patient to French et al., 1,671,151. Such systems transform the original voice frequencies into a narrcwer sEectrun thus permitting their transmission over a narrcw ~and circuit.
Okher arrangements for reducing the required bandwidth of the tIansmission circuit for voi oe s.ignals employ a mcdulation and frequency offset demodulation technique to transfer a Eortion of the voi oe spectrum to a different F~rtion of the speech band. In theæ schemes two or more Fortions of the sEeech s~ectrum are ~requency offset to occuEy the E~rtion of ~he spectrum ordinaril~
occupied ~y a fractional part of the sFeech s~ectrum and thes!e signals are sampled at an ad~quate rate and multiplexed in time before summing so that they can be applied to a narrcwband transmission line and received by time dbmultiplexing~ An example of such systems is found in the patent to Frànoo, 3,116,374, which uses multiElexing relate~ to the number of subdivisions of the incc~ling signal s~ectrum. Ihe Eatent to Harris et al., 4,314,104, utilizes what is termed the inherent time di~ision multiplex character of the voioe si~nal to acc~mplish a sil~ilar result. The pat~nt to Morgan, 3,349,184, ~hows an arrangement for bandwidth oompression and expansion ty frequency division and multiplication and in connection with FIGS. 20 - 22 thereof discloses a five channel system with the subchannels proprotioned to occuEy a 3 k~z transmission bandwidth.

Ihe present invention provides for frequency band stacking of multiple analog si~nals utilizing speech oampression and e~pansion and includes arrangements which Eenmit teleFhone dialing to establish dialed up circuits for simult~neous teleFhone calls on a single voi oe circuit.
Tb obtain a plurality of sic~nal channels each messac~e sic~nal is ocmpressed for transfonming the sic~n31 to a fractional ~ortion of its no m al spectrum. In one ~lannel the c~mpres æ d baseband signal can be directly Fassed to a summing unit. Ihe remainir.g channels have ~he oompressed sic~l applied to introduoe frequ~ncy offset, such as ky m~dulating ~he information sic~nal on a carrier and extracting the suppresæ d carrier sideband i~formation sic~al which is then demodulated at an offset frequency ~efore being applied to the summing device. In this fashion a plurality of information signals such as a Elurality oE voi oe messages, each of which originally had a spectrum approxImately eq~al to the bandwi~th of the oommunication circuit to ke emplcyed for transmission, can ~e redu oe d in frequency by a f~ctor e~ual to the number of channels to be transmitted, and those redu oe d frequency information signals can be stacked in contiguous ~ands to occuEy the bandwidth of the transmission circuit. By providing a reciprocal relationship for signals re oeived fro~ the transnission oircuitr the contiguous frequency band signals can be separated ~y appropriate filters and the offset frequency signals transfered to occu~y ~asekand or other nonmal position in the s~ectrun before the signals are passed through speech ex?ansion means for restoring each signal to its normal ~requency spectrum on its individual output terminal.
Ascordingly it is the principal object of the present invention to provide an impr w ed simple and econo~ical sys~m which permits multiple channel analog signals to ~e txansmitted over a narrowband communication circuit without t~e re~uirement for digital processing or oomputer capab;lity while m~intaining high quality two-way communication for the multipae chann21s over a single circuit ~etween the transmitting and re oe iviny poin~s.
When ~ed for teleEhone siynals the syst~m maint~ins voi oe quality reception for normal conYersation and voice recognition while multiElying the number of voi oe channels availakle ~ver each vo; oe ¢~ 3 Page grade circuit and permits nonmal dialing to es~a~lish the desir~
circuit connection.
ESCRIPIIQN 0~ THE DR~I?~S
FIG. 1 is a block ~iagram showing a typiGal apElication of the invention to double line capacity in a dedicated or private line system, or a dial cielected circuit connection.
FIG. 2 is a block diagrc~m of a oo~pression-stAcker-expan~er as used in the system of FIG. 1 providing two-channel capacity over a single two~ay circuit for two-~ay conYersations on both ~anr~ls.
FIG. 3 shcws the analog spectrum for a two-chc~nnel system at various stages as the signal is processed for transmission through a system of the type shcwn i~ FIG. 2~
FIG. 4 is a set of spectr~m diagrams ~haw mg the pro oe ssing of t~o channel signals of the type described in FI~. 3 upon reception as they pass through a system of the type shown in EIG.

FIG. 5 is a block diagram oE an alternative form of frequency oEf æ.t circuit for use in the systen .in accord~nce with the invention.
FIGS. 6A and 6B, assembled as indicated, are a block diagram o~ a system with prcvision for dial-through connection on the dial teleEhone network with stacking of two simultaneous two-way messages on a single trunk line.
FIG. 7 ~hcws the E and M convention for single fr~quen~y dlaling for an ordinary dialed telephone conn~ction wi~h indication of when the voi oe oor.pression and expansion circuits of the present invention would be in use or byEass ~uring dialing.
FIG. 8 is a block diagram of the un~yEass logic for switching a dialed connection to s~acked transmission condition.
FIG. g is a ~lock diagram of the dbwnrconverter used in FIG. 6.
FIG~ 10 is a block diagran of the ~E-con~erter used in the ~age 5 ~ystem of PI~. 6.
- L~ l.E~_nESCFIPTION OF T~F P~F~R~D ~ DI~
Referring ncw to FIG. 1 the w erclll apElication of the invention to prcvide dual chanr.el two-way cor~munication between remote stations on a typical single narrowband line will be described. As shcwn in FIG. 1 ordinary Psx or csx sta~ions 11 ana 12 represent existing teleEhone equiEment servicing a plurality of office ~elephones and local lines. T~ interconnect the remote stations 11 and 12 a single dedicated or dialed li~e 13 is ~r w ided to intercennect the two. Interposed between the local terminals 11 and 12 and the interconnecting long distan oe line 13 are stacker units 14 constructed in accordanoe with the invention~
Each stacker 14 handles two lines from telephone sets in its respective PsX and processes both of its lines for both LnComing and outgoing messa~es on line 13, thereky doubling the capacity of line 13, as will be ~escribed.
~ e~erring ncw to FIG~ 2 a stacker 14 o~ the type used in the systen of FIG. 1 will ~e described. The systen of the invention utili es devices which are capable of c~npression and expansion o audio analog signals. In describing such elements of the invention ~he ncmenclature of the patent to Schiffm~n, 3,786,195, will be used. In that patent a system for fast or slow playback of reoorded speech is disclosed with speech o~mpression and expansion units used to convert the spectrum of the speech signal into its normal frequency band.
In the present invention sEeech oomp~ession and expansion are u~ilized in real time and the various kncwn forms of such frequency tlansformation of analog signals, such as those disclosed in the Schiffman patent and s~sequent patents of the assigneet the Variable Speech Cb~trol Campany, Gan be utilized~
In the nomen~lature used, speech o~mpression consists of reducing the frequency of an ino~ming si~nal ky a factor lfc, where c has a Yalue greater than 1. Speech expansion is defined as transforming Patje 6 the spectr~n of an inct~ning slxet~ sitJnal to increase the freguenc~ ~y a factor l/c, where c has values greater ~h~n O but less than 1. ~hus speech c~npression at a factor of c = 2 will rt~u oe the speech spectrun to occupy one half the bandh7idth and h~ve frequency values one half those of the original signal.
Similarly, speet~h expansion with a factor of c = 0.5 will increase an incYxning signal sEectrum to double its wit~ih a~ twice the inc~ming frequency.
In FIG. 2 a system for handling two channels A and B is shcwn~ Ihe num~er of cha ~ s dictates the factor c, so for two channels c = ~, and a speecn ~xnpression unit 1~ reduoe s the normal in~ning speech s~ectrum of 250-3200 HZ to an o~tEut of 125-1600 ~z. A lcwpass filter 16 having this chalacteristic pas æ s the o~mpresæ d channel A si~nal to on~ input of a summing devioe li~
Ihe second channel, channel B, is applied to a similar speech o~mpression ~nit 18, the output o~ which is at half req~ency and half spectrum width at 125-1600 ~z, which signal is applied as one input to a balan oe d modulator 19. The other input to the balan oe d mo~Nlator lg is a carrier freguency, for example 455~ k~7 obkained from a sui~ble sour oe such as oscillator 21 and divider 22.
Ihe output of balan oe d modulator 19 oon~ains ueFer and l~wer sidebands which when applied to a sideband filter 23 produoe as an output signal the s~pressed~carrier lcwer sideband at (455.0 -1.6) kHz which is applied to a ~alan oe d demodulator ~4. ~he seoond inp~ to demodLlator 24 is the offset carrier at 45~,8 kHz derived from an oscillator 25 and divi~er 26. Ihe output o~ the demodulator 24 is applied to a bandpass filter 27 having a passband of 1625-3200 HZ and this signal is applied as another input to the sumning circuit 17.
Ihus at an ou~put terminal 28 the su~ming circuit 17 supplies the channel A si~nal compressed to half spectr~m width and occuEying the band from 125 to 1600 Hz while the signal of channel Paye 7 B is o~mpLessed to one half its sFectrum width and occupies the ~and from 1625 to 3200 ~
Ihe o~eration of the Eortion of FIG. 2 just descri~ed can be further ~nderstood with reference to FIG. 3 which shcws the sFectr~m of the analog sFeech signal at various points as it is subject to frequency txar,sformation. In FIG. 3 the column entitled ~Channel A" shcws audio input as oovering the s~ectr~m from 250 to 3200 Hz which when c3npressed ~y a factor oE 2 occupies the sFectrum from 125 to 1600 Hz. Ihat same si~nal after filtering in filter 16 ap~ears as ~he channel A portion vf the o~mbined output in FIG. 3.
In the column for channel B in FIG. 3 the original and c~mpressed sFectrum signals ap~ear the same as those in channel Ab When the 1600 EZ oampressed speech signal is modu~ated cn the 455.0 kHz carrier the double sicleband sic~nal centexed at ~55~0 kHz is obtained, and the lcwer sicleband is s21ected ky filter 23 whic~
may be a sharply t~ned mechanicll filter.
he democ~ation in demodulator 24 is wi~h resEect to a ~arrier fre~uency of 451.~ kHz which has the efect of m w Lng the ælected lcwer sic~band sFectrum hicJher by 1~6 kHz~ ~here.~y creating a sFectrum ccmponent which is offset ky ~hat amcunt to occupy the sFectrum from 1.6 to 3.2 kH~. ~his is the sisnal that appears at the output of ban~pass filter 27, and, when a~plied to the summi~lg circuit 17, produ oe s the B o~mponent of the ccmbined channel A and B signals on tenminal 28.
Thus from FIG. 3 it can be seen that each o~ ~he original audio input signals which both occuFy the 3 kHz kand of the available tele~hone l me or other ccmm~nication circuit have been compres æ d to occuEy one half of that spectrum and one o~ the channels has been offset by an amount equal to one hal the band so that the two can be transmitted s~multaneously, each uccupying a contiguous half of the available bandwidth, Stacked channel A and B signals of the type just described, when transmitted from tenminal 28 o~ FIG. 2 to a ~elephone line, 3'73 `' Paye ~

are reoe ived at a remote receiving station ~t ~n input te~ninal 31, as shown in FIG. 2, where the contiguous frequency bands are ~eparated ~y a bandpass filter 32, operating at the upper h~lf o~
the band, and a lcwpass filter 33 passing signals in the lcwer half of the band. 'rhe lcwer band signals fron the fil~er 33 a~e applied to a speech expansion unit 34 which converts the 125-1600 Hz input signal to a full spectrum 250-3200 Hz voice audio signal~
~br this purpose the sEeech e~ander 34 is opera~ing with the factor c = 0.5 .
qhe output of bandpass f;lter 32 is the u~per half spectrun of the transmitted signal occuFying the band from 1625-3200 HZ an~ ~
this sicnal is applied as one inEu~ to a balanoed ~odulator 36.
Ihe other input to the balan oe d modulator 36 is -the 451.8 k~z carrier which produces at the output of ~dulator 36 the upper and lGwer sidebands centered on this carrier freq~ency which are applied to a sideband f;lter 37. Ihe upper sidekand is selected ~y the sharply tuned ~ilter 37, and this sideband is apE~ied as on~ input to a balanoed ~Rmodulator 38, the other input o~ which is the carrier frequency 455.0 kE~. This d~mo~ulation prooess prcdu oe s at its output a si~nal cc~ponent in the ~and 125-1600 ~z, and this signal band is passed by lcwpass filter 39 and ap~lied to a speech expansion unit 41~ Speech expander 41 is operating with c = 0.5 so that it produ oe s an output full spectrun 5ignal frcm 250-3200 HZ at its output terminal 42. qhus the fuLl spectrun signal of chan~el A is recovered on output ~enminal 35 as previously descriked and the full spectrum OUtp~lt signal of chanrel B is recovered at terminal 42 to prcvide ~wo separa~e teleEhone message signals.
The reception of the stacked signal bands ana ~heir conversion into separate channel ~ and channel B ~ull spectrun output si~nals can be further understood with reference to FIG. 4 which shc~s the spectruM processing throu~h the re oe i~er portion of klG. 2. ~s indicated in FIG. 4, stacked A and B spectrum signals are re oe ived and by ~eans o~ the lowpass filter channel A

t73 Paye g is separated, and after passiny throu~h it~s separate speech expander 34 apEears as the full spectr~n channel ~ output signal.
The prooe ssing steps to recover channel B fram a received signal are also sha~n in FIG. 4. Ihe band~ass f Iter 32 separates the channel B spectrum from 1625-3200 ~ and the halanced modulator 36 opera~ing with an injected carrier of 451.8 k~
pro~u oe s the upper and lcwer sidebands separated Erom ~51.8 k~
into the sFectrum 1.5 to 3.2 kHz above and -3.2 ~o -1.5 kHz bela~
the carrier frequency. The sideband filter selects the upper sideband which is above the 451.8 k~lz carrier by -tl,5 to -~3.2 kHz.
This single sideb~nd suppressed carrier signal is then ~emodulated in demodula~or 38 a~ainst a 455.0 kHz carrier to produ oe the cQmpressed channel B siynal at ba æ band from 0 t~ 1.6 k~z, Passing this signal through a low pass filter 39 and speech expander 41 produces the desired full spectrun channel B s;gnal at 3.2 kHæ baseband.
Thus in the system of FIG. 1 if a unit of the type disclosed in FIG. 2 is emplcyed for the stackers 14 shown in FIG. 1 ~he single telephone line 13 can convey two sLmultaneous two way voi oe frequency messages with excellent voi oe qualit~ accomplish this each o ~he stackers 14 receives the channel A and B inputs rom its EBX 11 or 12, respectiv~ly, and supplies its channel A
and B outputs to these æ parate telephone sets or other devices which are using the line. In eac~ case the stacked outFut tenminal 28 and stacked input terminal 31 are oonnected at the local enc~ of the voice grac}e circuit 13.
Ihe invention can be practi oe d by other means for generatiny offset sidebanc~ of o~mpressed s~eech signals. In FIG~ 5 a modification is shcwn wherein a pair of balance~ moclulators 51 ancl 52 receive the compressed channel B signal fLom sEeec~ c~mpressor 18 after it passes ~hrough an audio 90 phase sh~ft network 5 The 455.0 kHz carrier also Easses through a 90 Ehase shift n~twork 55 before the 455.0 kHz carriers at 90~ phase relation are applied respectively to the balan oe d modulators 51 and 52~ Ihe 3t7 resultiny sic'eh~nd outputs frun mcdulators 51 an~ 52 are applied to a siynal c~nhiner 56 which operates to canoel the upper sidebcmd and enhan oe the lower sideband so that only the lcwer sideband signal is applied to a balan oe d dem~dul2tor 57~ Ihe ~pper si~eband si~nal applied to c3~modulator 57 is demodulated ~ith respect to the 451.~ k~z carrier to recover a ~modulated spectrum fran 1625-3200 HZ which is selected ky ~andpass filter 27 and coupled as an input to su~ning cir~uit 17.
Ihe remaining portions of the stacking chan~l for c~mbining c~annels A and B into a stacked output are the s~e as tho~e described with referen oe to FIG. 2 and corresponding elenents therecf have the same referen oe n~merals primed.
The re~overy of ~he individual channel A an~ B signals from a stack.~d inpu~ in the system of FIG. 5 is achieve~.in an analc~ous manner to that just described for stacking and a~in with the o~mponents that correspond to those described for ~IG. 2 having the same referen oe numbers primed~ Ihe stacked channel A iIlpUt signal passes through lcwFass filter 33' and spee ~ expander 34~
to produce the normal spectrum and frequency chan~el A output at terminal 35'.
The channel B spectrum of the stacked inpu~ signal in FIG. 5 is selected by bandpass filter 32' and ap~lied t~ an audio 90 phase shift network 64 which produ oe s the signal ~t two outputs wi~h 90~ phase relation, which signals are applie~ respecti~ely to balanced modulators 61 ~d 62. Ihe balan oe d modu~ator 61 and 62 operate with respect to the 451.8 kHZ carrier which is applied thereto with 90 phase shift from a carrier phase shift ne-twork 65. The outputs of the modulator 61 and ~2 are ~plied to a signal cRmbiner 66 wh rein the lcwer sideband is suppressed and the suppressed carrier upper si~eband is enhan oe ~ and aE~ied to a balanced demodulator 67. The balan oe d ~emodulator 67 has ~he 455.0 k~z carrier applied thereto which pro~u oe s ~he campressed channel B signal at basebar.d among other o~mEone~s and the baseband si~nal is selected ky lcwp~ss filter 39r. P~ter expansion at c = 0.5 in expander ~1' the full spectrum baseband channel B signal is obtained at output terminal 42'.
Although the invention has been described as applied to voice telephone circuits where two independent voice frequency signals can be accommodated or stacked for transmission over a single voice frequency circuit, the invention is not limited to doubling the narrowband transmission circuit capacity. As is clear from the present disclosure, if the signals are compressed and expanded by a factor of 2 and stacked by frequency offset the two signals propagate through the full bandwidth of the transmission circuit. If three or more message signals are to be transmitted the compression factor c would be correspondingly increased to c = 3, 4, etc. Such high compression ratios are available with the technology such as disclosed in the referenced Schiffman patent and related patents which are assigned to the assignee of that patent. In particular, the techniques for speech compression and expansion which are disclosed in applicant's European Patent Office Application 2d No. 0,127,892 published December 24, 1984 (E.P.O. Bulletin No.
19~4/50) can be employed in the present invention, and when so employed provide extremely high quality speech compression, expansion and reproduction in real time for narrowband transmission. This form of speech compression and expansion is paxticularly well suited for real time voice circuits since this published application controls signal processing by detecting the glottal pulse signal derived from the speech signal and achieve a natural splicing between successive samples of the speech signal. In the system of the present invention since splicing of the successive components of the time waveforms occurs in both compression and expansion for each channel at both ends of the transmission line, the natural glottal pulse epoch splicing is of particular advantage.
The invention is not limited to telephone wire line circuits but can be applied to band limited transmission circuits of any type. Although the invention has been described with reference to kh/ ~

voi oe circuits it is not so limited, and can be app:Lied to transmit various al~log signal forms without the c~npLexity and expense of digital techniq~es~ lhe transmission medium ~y also be radio or other fo~n of radiant energy w~erein the ~ystem o ~he t~Fe disclosed in FIG. 2 would be interposed to pro oe ss the audio or analog input ~ignal to deliver the desired stacked messages to the mc~ulator o any con~entional cvmm~nications systen. Ihe corresponding unstacking of the demodulated signal at the receiver would produ oe at the re oe iver ~he æ parate chan~el signals which were stacked and transmitted.
R~ferring now to FIG. 6A a system termiN31 for use with a local EBX to double the channel capacity for trunk calls which are dialed in a routine m~n~er is shcwn. Thus in assigning lines or long distance calls the PBX can assign two calls to a given trunk line and these connections will be made locally to an upper channel unit 70 or a lcwer channel 71, each of which has connections for the transmission ~D~rT) and reception (RCV) of a voice message and the E and M leads for ON HOOK and OFF HOOK
signals and impulse dialLng. In this description of FIG. 6A it is assumed that there is an i~entical unit at the renote tenminal to which a call is being placed and that the two sL~ultaneous messages which can be transmitted by the sy~tem will be received by the selected telephone æ t connec~ed to u~pper channel and lcwer channel units 70' and 71' at the remote terminal of FIG. 6B ~y the PBX during the dialing hookup sequen oe. Since FIG. 6B is in this and all other resEects identical wlth FIG. 6A, only ~G. 6~ w;ll be described and the corresponding elements in FIG. 6B will be assigned corresponding referen oe numerals primedr Signalling bet~een PBX's through a trunk circuit is normally accomplished b~y the E and M convention. ~he ~ ad transmits battery or ground signals to the si~nalling system. The E lead re oe ives open or ground si~nals from the trunk. ~he near end condition of the trunk is indicated ky the M lead and the far end condition by the E lead. These two leads on each end serve as the CN-OFF HUOK communication link via the trunk circuit.
A 2600 HZ single frequency inband signal is u~ed for dial impulse transmission sin oe the trunk circuit can ke a carrier channel where DC ~annot be transmitt~d~ Ihus ~pper channel unit 70 has E and M leads which are connected to a conventional single fr~quency generator 72 ~hich produ oe s the 2600 ~ signal on ~he outgoing line 73 to the trunk. The 2600 Hz signal ~n incoming line 74 is converted ky the single frequency ~nit 72 to the appropriate E and M lLne conditions in upper channel unit 70.
The uFper channel unit 70 re oeives the caller's voi oe signal on terminals XM~T which also receive t~e dual ~one multifrequenc~
(In~F) dialing tones i the caller's set emplcys touch~o~e dialing. Ihe XMIT terminals are pass~ed directly through a switch 76X to a sta.cker 77 ~or application to the Outgoing line 73. The al~ernate position of switch 76X applies volce sign31s to a speech compressor 78 operating with C = 2 to redu oe the frequency spectrum of voi oe signals by one-half.
Ihe upper channel unit 70 has receiver (P~CV) tenminals which are connected to the re oe iver of ~he ~eleFhone æ t of the caller.
lhese RCV terminals receive the voi oe signal from SF unit 72 aFpearing on output leads 75 after the voice signal Fas æ s through an unstack unit 79 and a VSC unit ~1 operating with C - 0.5 to expand the voioe frequency signals by a factor o~ two. The 5ignal fro~ the unstacker 79 passes through a switch 76R which in the position sh~ m in FIG. 6A ~ypasses the VSC unit 81 to avoid prooessing tone dial mg signals and in the alternate position switch 76R receives the voice sisnals after frequenc~ expansion in the unit 81.
The invention provides for a second simultaneous voi oe co~versation over the same trunk line from a callîng set which is connected to lawer channel unit 71 with the voice sisna~ aFplied to the XMIT tenminals and the receiver connected to ~he RCV
tenminals of unit 71. As in unit 70~ if the teleEhone set connected to ~nit 71 employs m~ dialing, those phone signals are ~6~ ô 3 Pac~

also connecked ko the ~Irr tenninals of ~nit 71~ Either tyFe a~dio sic~nals con~ected to XMrT terrninals in unit 71 pass throucih a switch connection ~6X to SF unit 72 from whic~h they are ap~li~d to the outgoing trunk line 73. In the alternate connec~ion of switch 86X the ~MIT signals pass through VSC unit 88 operating with C = 2 to redu oe the voi oe frequency s~ectrun ky one-half.
Ihe voice sic~nal output on tenminal 75 of unit 72 is applied directly to RC~ termi~lls in unit 71 when switch 86R is in the position shown and for the alt rnate position of switch 86R voice signals from terminal 75 are expanded in VSC unit 87 oEerating at C = 0.5 to double the frequency spectrun o~ the signals~
As previously ex~olained for upper channel unit 70, ~he lower channel unit 71 has E and M termimls for im~ul æ dialir~g whi~h are connected to a single fre~uency unit 82 where the dialLng imFulses are converted to 2~00 Hz and applie~ 'LO a d~ conver~er 810 Sinoe the uEper channel unit 70 is employing the nD~maL 2600 Hz dialing impulses for E and M linR dialing it is nece~a~y to distinguish the dialing signals from the l~er ~hannel unit 71.
lhis is acoomplished in a dcwn-converter 81 which reoe iYes the 2600 Hz si~n31 on lLne 80 from single frequency unit 82 in the form of impulses as ~etermLned ~y the E and M lines of ~cwer channel unit 71. Ihe 2600 Hz inp~ si~nals on line 80 ~
dbwn-con~erter 81 ap~ear on output line 83 as 1400 Hz ~ulses or tone indicating the condition of the E and M lines in l ~ er channe~ unit 71. ~he 1400 Ez 5i9nals on line 83 are aE~ied to the ou~go.ing trunk line 73.
Ino~n~lg 1~00 Hz signals from a remote lcwer chan~l 71' u~it -which are received on line 7~ of the trunk line are applied ~y line 84 to an up-converter 85 where the 140a Ez tone i~Eulses are conNerted to corresponding 2600 Hz si~nals. These 260~ ~z signals from uE-co~verter 85 are aEplied on line 89 to the SF unit 82 for indicating the condition o~ the called remote telephone set, i.e., oN HOOK or OFF ~OOK. Ihe SF to voice logic disables the 1400 F~.
notch filter when voi oe transmission takes plaoe , ~lhe notch filter is necessary because during dialing by the caller~ the S~ from the far side is notched out so the caller cannot hear it. When the far si~e goes off hook to answer, the absen oe of the 1400 Hz on line 84 at the up cor~erter 85 e~ables the SF to voice logic to kyEass ~he notch filter for voic~ signals from the far side.
In the unmodifi~d E~M to SF CorNerter 82 where 2600 ~ tor~e dialing imEulses are used, this function is already nor~ally available.
Ihe switching for signals in upper channel unit 70 is controlled in accordan oe with the condition of an AND circuit 91 which has as inputs the E and M leads of up~er channel 70.
'Similarly the switching for signals of l~er channel unit 71 is controlled in accordance with the condition of an AND circuit 9 which is determined ~y the inputs thereto fro~ the ~ and M lead~
of l~wer channel ~nit 71.
As previously stated, all elements of the called station are p equiped wi~h corresFonding structure and function and such a called station is illustrated in FIG. 6B with the elements numbered with primed referenoe characters corresFonding to those used in FIG. 6P ~he operation of the systemL to automaticall~
dial a call~l number and transmit on only the ~pper half o~ the voice channel b~Lndwidth will ncw be descriked. Cbnsider FIGS. 6A
and 6B interconnected by a trunk circuit 73, 74, and referring to the imp~l æ dialing seq~ence of FIG. 7, the establishing of a voice circuit between a calling and a called set w;~l be described. ~ssuming that the calling party is at the station of EIG. 6A and that the PBX has connected the c31ling teleEbLon~ æ t to the tenmin31s of upper channel unit 70. Referring to FIG. 7 the caller's M lead goes from O to -48 ~-olts when the receiYer goes OFF HOOK, and the dialing impul æ s are ~enerate~ ~y the noDmal dial switch. Ihe condition o~ ~line in unit 70 is transmitted via the single frequency ~nit 72 to the called n~mber at FIG. 6B ~here the switching of the M le~d at the caller's ~ 3 station changes the E l.ead at the call~l station ~rom oEen to U, as i.ndicated~ Ihe dialed n~nber ælects the circuit to the applopriate called tele~hone set in the us~l manner, and when the called number goes OFF ~OOK its M lead changes frcn 0 to -48 volts. ~oing OFF HOOK at the called set causes the E lead for the callex's set to go from o~en to 0. With the cal-er's ~ lead at -48 ~nd the caller's E lead at 0, the connection is made to switch s~itches 76X and 76R to the alternate position frcm ~hat shown in FIGo 6A and the oonversation takes plaoe until either the caller or called station goes back ON HOOK. During the oonversa~ion the V5C units 78 and 80 in FIGo 6A and VXC units 78' and 80i in FIG~
6B are not kyEQssed but are in o~eration.
Before both the caller and called sets are ~P ~OOK ~he E and ~ line condition is transmitted on the ~.6~0 Hz ~o~e withou' mo ;fication by the VSC units kut the upper channel 70 is freq~ency shifting ~y the S~ACK and UNSTACK 77, 79 units. After both sets are OFF HOOX and the alternate position of switches 76 occurs the transnission ~rom X~rT in upFer channRl 70 is compressed in VSC unit 78 to half its spectrum wi~h and is stacked in the u~per half of the voice channel ~y SIACK unit 77 before it is transnitted on outgoing line 73. At the re oe iving end in FIG. 6B the voice frequency signal stacked in ~he upper half of the channel bandwidth is inc~ming on line 73' and emerges from the SF unit 72' to be unstacked in unit 79' and have its ~requency restored ty the VSC expander 80' which ~bubles the frequency com~onents of its input signal. Ihe full frequPn~y voice message thus reaches the RCV terminals of u~per channel 70' o FIG. 6B from the VSC unit 80'. qhe reverse process oc s for voice messases originating at the XMIT tenminals of uFper channel unit 701 m FIG. 6B. Such signals are redu oe d in bRndwidth by VSC
compression unit 78' and stacked in the ~pper hal~ of the channel bandwid~h ~y SI~CK unit 77' for transmission on trunk line 74' to line 74 in FIG. 6P~ qhe voice signals emerge fro~ SF unit 72 at terminal 75 and they are uns~acked at unit 79 and ex~anded in VSC

9'î ;:~
Pa~e 17 ~mit 80 to produce a full fr~luency voice signal at ~t~ RCV
tenminals oE upper channel unit 70.
Simultaneously with the OFF HOOK conversation occurring in the upper ~ta~ el 70 as just descriked, a second voice message circuit can be achieved on the same trunk line 73, 74 using lcwer channel unit 71 and its associated circuits~ lhus the EBX will assign the next call to lower channel 71 and the E and M dialin~
sequence will appear sn the E and M leads of unit 71. lhe SF
converter 82 converts the ~pul æ s to 2600 ~z tones and in this instance, sinoe 2600 Hz is above the lcwer ~nannel up$er limit, a down~converter 81 converts the 2600 Hz impulse si~nals into 1400 Hz signals on line 83. Ihese different frequenc~ dialing signals are a~plied to the trunk line 73 which has a notch filter to el~minate the 1400 Hz tone frnm the conver.~a~ion ~.hich is keing conducted on the upper chan~el. Ihe incGming 1400 Hz signal on line 73' of FIG. 6B pRsses through an up-converter 85' to restore the impulses to 2600 Hz on lead 89' so that the dialing impulses can pass through the SF unit 82' and ~e reco~nized ky the nonmal teleFhone single frequency dialing equiEment of the lo~er channel 71'. The reverse path ~rc~ the E and M leads oE l w er channel unit 71' pro~u oe 2600 Hz fr SF unit 82' whi~h is ~bwn oonverted in unit 81' to 1400 ~z at lLne 83' ~or tran~mission on trunk line 74'~ Ihis 1400 Hz signal received on line 74 o~ FIG. 6A is applied to an wp~converter 85 which converts the impulses to 2600 Hz at leads 89 where they are applied to SF unit 82 for nonmal operation with the E and M leads of lower channel 71. Again when both the ca~ling and called æ ts are OFF HOOK for the lcwer channel the AND circuits 91, 92 and 91', 92' alter ~he switches 76~ 86, 76', 86' to unkypQss the ~SC units~ In the lcwer chal~el 71 the ~ompression of the voi oe signal on XMrr te~min~ls of lower channel unit 71 by a factor of two in unit 88 places the signal in the lower hal~ of the chalmel bandwidth so that no stacking is required. Similarly in FIG. 6B the received signal is in the lower half of the channel bandwidth and only needs to be expan~ed ~ ~ ~ 9 ~ ~3 a in VSC ~it 87' without unstacking to be restored to a fuL~
frequency voi oe signal at P~CV terminals in l~er channel unit 71'.
~he voi oe message originating at the æ t con~ected to lower channel unit 71' in FIG. 6B tra oe s an analogo~s E~th kack to the re oe iver RCV terminals of lcwer channel unit 71 in FIG. 6A with VSC oonpression for æ nding and VSC expansion upon recepti~n, but without stacking or unstacking at either end of the line.
~ n the ~yst~m disclosed in FIGS. 6A and 6B the speech oompression and expansion and the stacking and unstacking are acc plished in the manner described herein with reEerenoe to EIGs. 1 - 40 Thus, referring to FIGS. 3 and 4, the upper ~hannel of FIGS. 6A and 6B would correspond to channel B, while the lower channel of FI~S. 6A and 6B would correspond to channel A_ In the system of FIGS. 6A and 6B where r~ touch tone ~ialing is used, such dial mg tones are appliea to the xMrr terminals of ~he calling station, such as upFer channel 70 of FIG.
6P_ The highest frequency used in ~IMF dialing is 1477 ~z, hen oe the dialing tonRs in ~he upper channel are stacked in unit 77 to transmit tones equal to t3200-FDn~F) which is the lower si~ekand when the input signal is applied for stacking in the system of FIGo 1~ ~hen these stacked dial tones are unstacked in unit 79' of FIG. 6B they are applied directly to the RCV tenminals without VSC conversion, since the OFF ~OOK condition has not yet occurred, and when ~nst~cked, they are reoonverted into their normal aual tone freq~encies of the touch tone dialing system. F~r a call originating on lower channel unit 71 using m~lF dialing the dual tones are all within the lower channel and can be transmitted direc$1y wi~hout VSC conversion and without stacking or wns~acking to r~ceiver tenminals RCV in lower channel unit 71 of FIG. 6B. ~t each end of ~le line SF units 72 and 72' process ~he OFF ~03~
condition to suEply the signlls to AND circuits 91, 92, 911, 92' for the neæssary switching to unkypass the VSC during OFF ~COK
voioe messages.
Further details of the switching to untyFass the VSC wnits in P~lqe 19 FIGSo 6A and 6B will ~e descrited with reference to FIG. 8. ~s previously d~scriked upon going OFF HOOK the calling PBX switches the ~ ad from ground to -48 volts~ As shcwn in FIG. 8, this OF~
HOOK condition of the calling set is applied on the M le~d to a level interfaoe and deboun oe circuit 101, Where the switched l~vels are converted to ~5 volts and applied through an ir~erter 102 to a talk condition detector 103. ~le E lead o~ ~he call.Lng station is n~nmally open for the ON HCOK condition and goes OFF
HOOK (winks) to signal the called end is ready to re oeive dialing information. When the E lead is open the -48 volts is applied to the rela~ wiilding in the PBX thus ~e-energizing ~t and creating the E signal for the calling end~ A level interface and debo~m oe circuit 104 converts the -48 to ground signal.on the E lead into ~5 volts which is applied directly to talk condition detector 103r When the talk condition dRtector 103 senses the OFF HOOK condition for both ~he E and M circuit its outp~t goes high and this output is fed to a detect delay circuit lOS whi~h provides approximately a one second delay to prevent the VSC fron being un~ypas~ed ky the wink back signal. Pfter this delay the output o~ d~tect delay 105 operates the unbypass swit~ling to the VSC ~nits 91, 92 and 91';
92' of FIGS. 6A and 6B ~y its output signcil on lead 1060 q1le called end of the circuit, ky going OF~ HOCK unbyF~sses VSC for both ends ky creating OFF ~OOK conditions on the E~M leads on both ends~
R~ferring to Fig. 9 the dialer lcw c~annel transmit SF up converter will be descri~ed. A 1400 HZ signal is ge~erated in a 1400 Hz oscillator 110 and fed to a tone gate circuit where the 1400 ~z signal is gated on and off and adjusted to the level of the input SF tone~ Ihe output of the tone gate 111 is then Eassed through a lcw pass filter 113 to provide a sine wave in li~e 83 of Fig. 6A~
1he transmit 2600 Hz SF signal on line 80 is detected ky a PLL tone d2tector 114 and a ~ekoun oe d logic output is used to control the tone gate 111 as follcws. Ihe SF is ~mplified in PRY~ 20 ~nplifier 115 c~d fed to an er~elope detector 116 and le~ 1 c~tector 117 to clevelop a lo~ic siyr~l represen~ing the high SF
tone levelO qhese t~o lc~ic signals fra~. to~e ~etec,~or 114 and SF
level c~tector 117 are used in the tone gate circuit 111 to gate the 1400 ~2 SF signal on and off at output lir~ 83 and adjust its level to follc~ the input SF tor,~e on li~e 80. ~e SF is a supervisory signal cburing ~N HOOK condition~ It is generated at the E&M to SF module ~y presen oe of thP ground c~ndi~ion o~ the lead. when the caller goes OFF ~COR, the SF goes from a -20dbm level (during ON HOOK) to no SF ~excess of -100d~ During dialing, the Fulsing of the ~M" lead causes the turning on of the SF to a level of -&dbm ~12d~m higher than the s~ervisory level~.
~ eferring to Fig. 10, the dialer low channel receive SF down converter will be described. Irhe low channel receive signal which contains bo~h voi oe and SF information is fed to the input of this circuit on line 84. To reduce the possibility ~E voice signals being mistakenly recognized as SF, an ANTI TALK ~FF circuit 121 is employed. ~he signals on line 84 are first sep~rated ky ~ans of a high pass ~ilter 122 for the SF tone and low p~lss filters 123, 123l, in the ~rI TALK OFF circuit. A logic si~nal is pro~uced by a tone detector 125 whenever 1400 Hz. is present at ~he pr~per level. Ihe voi oe sig~al is squared up in d~tec~or 126 and the positive edyes used to trigger a retriggerable monostable 127 px w idiny a more distinct indication o~ the pxesenoe o voice.
~he two logic signals are then fed to a SF tone ~etect logîc 128 whexe they axe ~sed to provide an SF detected signal to t~rn a tone gate 129 on and off.
The output o~ a 2600 Hz oscillator 131 is fed to the tone gate 129 wh~re it is turned off and on ~y ~he SF detect logic signal and the level adjusted ~y the level detec~ion circuits 132 and 133 sLmilar to units 116 and 117 previously ~escribed. ~he gated SF 2600 Hz tone is then passed through la~ pass filters 134r 134~ to pro~i~e a sine wave output on line 89 wh~ch ~ollows the input SF tone level.

3 ~26~ t3 :~e 21 Variations oE the invention will occu~ t~ t~ose skilled in the art fr~l the ~resent disclosure O Although v~ice messages are ~he rx:rm and are melltioned as suc:h in this disclosure, it will be ~derstood that the system oE~erates w;th a~ analog signals within the voioe fr~quenc~ band of the system. ~e insrention accordingly is to be corLsidered as not limited to 'che discloæd or suggested embodiments but only ~y the 5co~e of the aE~ended claims,

Claims (6)

THE EMBODIMENTS OF THE INVENTION IN WHICH AN EXCLUSIVE PROPERTY
OR PRIVILEGE IS CLAIMED ARE DEFINED AS FOLLOWS:

1. A terminal for two way telephone communication of two voice messages over a single voice circuit comprising:
first and second transmitting voice channel input terminals;
first and second speech compressors coupled respectively to said input terminals for reducing the spectrum of the voice signals in each channel to the lower half of the bandwidth of said voice circuit;
a first carrier frequency oscillator for supplying a first high frequency relative to the voice spectrum;
first balanced modulator means for modulating said first carrier frequency with the compressed voice signal of said second channel and producing a first suppressed carrier side band output;
means for selecting the lower side band of said output;
means for generating a second carrier frequency offset by the bandwidth of said circuit from said first carrier frequency;
first balanced demodulator means responsive to said lower sideband output and said second carrier frequency to produce the compressed voice signal of said second channel offset by half the bandwidth of said circuit;
means for selecting said lower side band offset in the upper half of said circuit bandwidth;
1. Claim 1 cont'd 2 means for summing the lower half bandwidth compressed voice signal in said first channel with the offset compressed voice signal of said second channel in the upper half bandwidth to provide an output signal of the combined upper and lower half bandwidth signals to an output terminal;
   means for coupling the combined signal to said voice circuit;
   a receiving voice circuit input terminal;
   a bandpass filter for passing signals in said upper half bandwidth;
   a lowpass filter for passing signals in said lower half bandwidth;
   means coupling said receiving voice circuit input terminal to the inputs of said bandpass filter and said lowpass filter;
   second balanced modulator means responsive to the output of said bandpass filter and said second carrier frequency for producing a second supressed carrier single sideband signal offset above said second carrier by half said voice circuit bandwidth;
   means for selecting said offset single side band signal from said second modulator;
   second balanced demodulator means responsive to said second suppressed carrier selected single sideband signal and said first carrier frequency for producing said upper half bandwidth signal as a sideband at said second carrier;
   means for detecting said sideband of said second carrier at baseband; and first and second speech expanders coupled respectively to the outputs of said lowpass filter and said detected baseband output to produce separate baseband voice spectrum signal outputs from the upper and lower half bandwidth signals coupled to said input terminal.
2. 2. An adapter for dialed telephone network calls that provides for two simultaneous calls in a single voice frequency channel comprising:
   two groups of terminals providing individual connections to first and second telephone sets, each telephone set including a transmitter, a receiver and E and M dialing lines;
   outgoing and incoming line terminals;
   means for spectrum compressing voice signals from each of said transmitters to half the voice channel bandwidth and for stacking the compressed voice signal of said first set to occupy the upper half of said bandwidth;
   first frequency converting means for converting E and M line status and dial impulses from said first set into single frequency signals in the upper half of said bandwidth;
   second frequency converting means for converting E and M line status and dial impulses from said second set into single frequency signals in the upper half of said bandwidth;
   a down-converter for converting the frequency of the output of said second single frequency converting means to a predetermined frequency within the lower half of said bandwidth;
   means for coupling the outputs of said first single frequency means and said down-converter to said outgoing line terminals to be transmitted over said channel;
   means for unstacking and spectrum expanding voice signals in the upper half of said bandwidth and passing the resulting voice signal to the receiver of said first set;
   second means for expanding voice signals in the lower half of said bandwidth and passing the resulting voice signal to the receiver of said second set;
   means for coupling signals received from said incoming line terminals to said first single frequency means and responsive to signals in the upper half of said bandwidth for determining E and M status of a called number connected to said first set and for passing voice signals for connected calls to said unstacking and expanding means, and operable to pass voice signals in the lower half of said bandwidth to said second expanding means; and an up-converter for converting incoming line signals at said predetermined frequency into output signals at said single frequency and applying said output signals to said second frequency converting means for determining E and M
   status of a called number connected to said second set and for passing voice signals for connected calls to said second expanding means.
3. 3. Apparatus according to claim 2 adapted for dual tone multifrequency (DTMF) dialing comprising in addition:
   
   means for applying respective DTMF tones as dialed to the transmitter terminals in each set;
   means responsive to local off hook condition prior to completion of a call for bypassing signals around said spectrum compressing and both expanding means; and means responsive to completion of a call for unbypassing the compressing and both expanding means for the voice signals of the sets which have connected with a called number.
4. 4. The method of dialing a plurality of independent calls through a trunk line having predetermined bandwidth prior to adapting said line for carrying a like plurality of frequency compressed and offset stacked messages comprising the steps of:
   subdividing the trunk line bandwidth into a plurality of contiguous frequency sub-channels with stacker terminals;
   assigning each originating trunk line call from a plurality of sets to an individual one of said sub-channels;
   converting as required single frequency tone dialing impulses from any set to a frequency within the sub-channel assigned for that particular call;
   responding to an answering signal from a called set which is selected by dialing impulses from a sub-channel to condition the stacker terminals at that sub-channel for two-way voice transmissions by converting local telephone set voice messages through frequency compression and offset stacking to utilize the bandwidth of the line for multiple simultaneous message transmission; and unstacking and expanding compressed voice messages received in said sub-channel to normal voice frequency range signals applied to the receiver of said set.
5. 5. In a dialed telephone network which uses the E and M
   line convention and single frequency dialing impulses an adapter which permits two telephone conversations to occur simultaneously over a single trunk line comprising:
   input couplings for an upper channel telephone set and a lower channel telephone set;
   an output coupling to said trunk line including transmit, receive and single frequency E and M status lines;
   means for frequency compression of voice frequency signals originating at said two sets and stacking the compressed signals into upper and lower channels to occupy the trunk line bandwidth;
   means for applying said compressed stacked signals to said transmit line;
   means for unstacking signals received on said receive line and to obtain separate upper and lower channel signals;
   means for frequency expansion of said separate upper and lower channel signals to normal voice frequency signals applied to said upper and lower channel sets respectively;
   means for translating the single frequency dialing impulses originating from the lower channel set into a predetermined frequency in the lower half of said line bandwidth and applying it to said transit line; and means for translating dialing impulses received on said receive line at said predetermined frequency to said single frequency and applying such dialing impulses to said lower channel;
   whereby dialing impulses originating in said upper and lower channel sets are operative to select connection to a corresponding upper or lower channel set respectively at a called end of the trunk line and said upper and lower channel voice frequency signals can be transmitted at the same time.
6. 6. Apparatus according to claim 5 adapted for dual tone multifrequency dialing which includes logic means for bypassing said means for frequency compression and said means for frequency expansion during dialing and unbypassing said frequency compression and frequency expansion means in each channel respectively upon completion of a circuit to a telephone set at the called end of the trunk line.