1(cid:127)(cid:127)
`
`United States Patent 1191
`Leitch et al.
`
`1111111111111111111111
`
`US005689440A
`[11] Patent Number:
`[45] Date of Patent:
`
`5,689,440
`Nov. 18, 1997
`
`[54] VOICE COMPRESSION METHOD AND
`APPARATUS IN A COMMUNICATION
`SYSTEM
`
`[75]
`
`Inventors: Clifford Dana Leitch, Coral Springs;
`Robert John Schwendeman, Pompano
`Beach; Kazimierz Siwiak; William
`Joseph Kuznicki, both of Coral
`Springs; Sunil Satyamurti, Delray
`Beach, all of Fla.
`
`[73] Assignee: Motorola, Inc., Schaumburg, ID.
`
`[21] Appl. No.: 764,656
`
`[22] Filed:
`
`Dec. 11, 1996
`
`Related U.S. Application Data
`
`[63] Continuation of Ser. No. 395,747 ,Feb. 28, 1995, abandoned.
`Int. CL 6 ....................................................... H04B 7/&0
`[51]
`[52] U.S. Cl . .................... 364/514 R; 370/109; 455/54.1;
`455n0; 381/29; 381/34
`[58] Field of Search .................................. 381/29, 30, 35,
`381/36, 37, 34; 364/514 R; 395/2.14, 2.12,
`2.21, 2.24, 2.29; 340/825.44; 455/54.1,
`70, 72, 109, 47, 38.1
`
`[56]
`
`References Cited
`
`U.S. PATENT DOCUMENTS
`
`2/1974 Puckette .................................. 325/137
`3,793,589
`1/1979 Sbiki .
`4,134,069
`4/1986 Wong ..................................... 370/69.1
`4,586,174
`6/1989 Kotzin ....................................... 381/31
`4,839,923
`4,875,038 10/1989 Siwiak et al ....................... 340/825.44
`4,882,579 11/1989 Siwiak ............................... 340/825.44
`4,955,083
`9/1990 Phillips et al ..
`
`5,068,898
`5,121,391
`5,175,769
`5,216,744
`5,239,306
`5,282,205
`5,387,981
`5,533,062
`5,535,215
`
`11/1991 Dejmek et al ............................ 381/29
`6/1992 Pane th et al. .
`12/1992 Hejna, Jr. et al. ................. 340/825.44
`6/1993 Alleyne et al ............................ 381/29
`8/1993 Siwiak et al ....................... 340/825.44
`1/1994 Kuznicki ................................ 370/94.1
`2/1995 Orlen et al .............................. 358/400
`7/1996 Liberti, Jr. et al ...................... 375/334
`7/1996 Hieatt, ID .............................. 370/95.1
`
`OfHER PUBUCATIONS
`
`Verhelst and Roelands, An Overlap-Add Technique Based
`On Wavefonn Similarity (WSOIA) For High Qality Time-S(cid:173)
`cale Modification Of Speech, IEEE 1993, pp. II-554-II-557.
`Oppenheim and Schafer, Changing the Sampling Rate Using
`Discrete-Time Processing, Discrete-Time Signal Process(cid:173)
`ing, 1989, Ch. 3.6, pp. 101-112.
`
`Primary &aminer-Bllis B. Ramirez
`Assistant Examiner-Demetra R. Smith
`Attorney, Agent, or Finn-James A. Lamb
`ABSTRACT
`
`[57]
`
`The present invention comprises a method for compressing
`a plurality of voice signals within a voice communication
`resource (see HG. 6) having a given bandwidth within a
`voice communication system (100). The method comprises
`the steps of subchanneling the voice communication
`resource into a plurality of subchannels ( 441, 442, 443),
`placing a pair of the plurality of voice signals (401, 402) on
`a subchannel (441); modulating the pair of the plurality of
`voice signals ( 401, 402) about a pilot signal (581) within the
`subchannel (441) using single sideband modulation; and
`compressing the time of each of the voice signals ( 401, 402)
`within the plurality of subchannels (441,442,443), wherein
`these step provide a compressed voice signal.
`
`20 Claims, 14 Drawing Sheets
`
`(103
`
`102
`
`SELECTIVE
`CALL
`TRANSMITTER
`
`VOICE
`COMPRESSION
`CIRCUITRY
`
`101
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`:
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`
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`AMPLIFIER
`
`ALERT
`
`111
`INPUT DEVICE L----------------------------------------
`
`Apple Exhibit 1005
`Apple Inc. v. Rembrandt Wireless
`IPR2020-00033
`Page 00001
`
`

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`IPR2020-00033 Page 00002
`
`

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`IPR2020-00033 Page 00003
`
`

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`
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`
`___ __.___
`
`IPR2020-00033 Page 00004
`
`

`

`U.S. Patent
`
`Nov. 18, 1997
`
`Sheet 4 of 14
`
`5,689,440
`
`102
`
`103
`
`113
`r------------L------,
`I
`I
`150
`160
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`
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`MODULE
`
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`
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`FREQUENCY
`FA 6.25 F 8 12.5 F C 18.75
`25
`
`FIG.6
`
`IPR2020-00033 Page 00005
`
`

`

`U.S. Patent
`
`Nov. 18, 1997
`
`Sheet 5 of 14
`
`5,689,440
`
`571
`
`570
`
`SSB EXCITER SET 1---_.,..
`TO
`f A• USB
`
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`
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`
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`
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`
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`
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`
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`EXCITER
`
`IPR2020-00033 Page 00006
`
`

`

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`IPR2020-00033 Page 00007
`
`

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`IPR2020-00033 Page 00008
`
`

`

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`IPR2020-00033 Page 00009
`
`

`

`U.S. Patent
`
`Nov. 18, 1997
`
`Sheet 9 of 14
`
`5,689,440
`
`1--
`
`c310
`
`CYCLE 0 CYCLE 1 CYCLE 2
`
`CYCLE13 CYCLE14
`
`32~0-------------
`
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`
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`
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`0
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`
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`
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`
`350
`
`IPR2020-00033 Page 00010
`
`

`

`U.S. Patent
`
`Nov. 18, 1997
`
`Sheet 10 of 14
`
`5,689,440
`
`c310
`
`(cid:141)
`
`1
`
`CYCLE O CYCLE 1 CYCLE 2
`
`CYCLE13 CYCLE14
`
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`
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`
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`
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`
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`
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`
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`
`IPR2020-00033 Page 00011
`
`

`

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`
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`
`IPR2020-00033 Page 00012
`
`

`

`U.S. Patent
`
`Nov. 18, 1997
`
`Sheet 12 of 14
`
`5,689,440
`
`INPUT
`
`I
`
`...i--- NEW SAMPLES
`--so---i- So-'
`TO BE COPIED
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`
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`/
`
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`FROM ITERATION 1
`
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`COPIED
`
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`END OF ITERATION 2
`OUTPUT
`I
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`FROM ITERATION 1 OVERLAP ADD
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`
`IPR2020-00033 Page 00013
`
`

`

`U.S. Patent
`
`Nov. 18, 1997
`
`Sheet 13 of 14
`
`5,689,440
`
`INPUT
`
`1
`
`/
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`
`IPR2020-00033 Page 00014
`
`

`

`U.S. Patent
`
`Nov. 18, 1997
`
`Sheet 14 of 14
`
`5,689,440
`
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`IPR2020-00033 Page 00015
`
`

`

`5,689,440
`
`1
`VOICE COMPRESSION METHOD AND
`APPARATUS IN A COMMUNICATION
`SYSTEM
`
`2
`mitter base station and a plurality of selective call receivers.
`The transmitter base station comprises an input device for
`receiving an audio signal, a processing device for compress(cid:173)
`ing the audio signal using a time-scale compression tech-
`This is a continuation of application Ser. No. 08/395,747 5 nique and a single side band modulation technique to
`provide a processed signal and a quadrature amplitude
`filed Feb. 28, 1995, now abandoned.
`modulator for the subsequent transmission of the processed
`signal. Each of the plurality of selective call receivers
`comprises a selective call receiver module for receiving the
`transmitted processed signal, a processing device for
`demodulating the received processed signal using a single
`side band demodulation technique and a time-scale expan(cid:173)
`sion technique to provide a reconstructed signal, and an
`amplifier for amplifying the reconstructed signal into an
`reconstructed audio signal.
`In another aspect of the present invention, a selective call
`receiver for receiving compressed voice signals, comprises
`a selective call receiver module for receiving a transmitted
`processed signal, a processing device for demodulating the
`received processed signal using a single side band demodu-
`20 lation technique and a time-scale expansion technique to
`provide a reconstructed signal, and an amplifier for ampli(cid:173)
`fying the reconstructed signal into an reconstructed audio
`signal.
`In yet another aspect of the present invention, a paging
`base station for transmitting selective call signals on a
`communication resource having a predetermined
`bandwidth, comprises, an input device for receiving a plu(cid:173)
`rality of audio signals, a device for subchannelizing the
`communication resource into a predetermined number of
`30 subchannels, an amplitude compression and filtering module
`for each subchannel for compressing the amplitude of the
`respective audio signal and filtering the respective audio
`signal, a time compression module for compression of the
`time of the respective audio signal for each subchannel, and
`a quadrature amplitude modulator for the subsequent trans-
`35 mission of the processed signal.
`BRIEF DESCRIPTION OF THE DRAWINGS
`F1G. 1 is a block diagram of a voice communication
`system in accordance with the present invention.
`F1G. 2 is a block diagram of a base station transmitter in
`accordance with the present invention.
`F1G. 3 is an expanded electrical block diagram of the base
`station transmitter in accordance with the present invention.
`F1G. 4 is an expanded electrical block diagram of another
`base station transmitter in accordance with the present
`invention.
`F1G. 5 is block diagram of a speech processing, encoding,
`and modulation portion of a base station transmitter in
`accordance with the present invention.
`F1G. 6 is a spectrum analyzer output of a 6 single(cid:173)
`sideband signal transmitter in accordance with the present
`invention.
`F1G. 7 is an expanded electrical block diagram of a
`55 selective call receiver in accordance with the present inven(cid:173)
`tion.
`F1G. 8 is an expanded electrical block diagram of another
`selective call receiver in accordance with present invention.
`F1G. 9 is an expanded electrical block diagram of another
`60 selective call receiver in accordance with present invention.
`F1G. 10 is a timing diagram showing the transmission
`format of an outbound signaling protocol in accordance with
`the present invention.
`F1G. 11 is another timing diagram showing the transmis-
`65 sion format of an outbound signaling protocol including
`details of a voice frame in accordance with the present
`invention.
`
`Voice message paging is not economically feasible for
`large paging systems with current technology. The air time
`required for a voice page is much more than that required for
`a tone, numeric or alphanumeric page. With current
`technology, voice paging service would be economically
`prohibitive in comparison to tone, numeric or alphanumeric
`paging with less than ideal voice quality reproduction.
`Another constraint in limiting voice message paging is the
`bandwidth and the present methods of utilizing the band(cid:173)
`width of paging channels. In comparison, the growth of 25
`alphanumeric paging has been constrained by the limited
`access to a keyboard input device for sending alphanumeric
`messages to a paging terminal, either in the form of a
`personal keyboard or a call to an operator center. A voice
`system overcomes these entry issues since a caller can
`simply pick up a telephone, dial access numbers, and speak
`a message. Further, none of the present voice paging systems
`take advantage of Motorola's new high speed paging pro(cid:173)
`tocol structure, also known as FLEX™.
`Existing voice paging systems lack many of the FLEX™
`protocol advantages including high battery saving ratios,
`multiple channel scanning capability, mixing of modes such
`as voice with data, acknowledge-back paging (allowing for
`return receipts to the calling party), location finding
`capability, system and frequency reuse, particularly in large 40
`metropolitan areas, and range extension through selective
`re-transmission of missed message portions.
`With respect to the aspect of paging involving time(cid:173)
`scaling of voice signals and to other applications such as
`dictation and voice mail, current methods of time-scaling 45
`lack the ideal combinations of providing adequate speech
`quality and flexibility that allows a designer to optimize the
`application within the constraints given. Thus, there exists a
`need for a voice communication system that is economically
`feasible and flexible in allowing optimization within a given 50
`configuration, and more particularly with respect to paging
`applications, that further retains many of the advantages of
`Motorola's FLEX™ protocol.
`
`TECHNICAL FIELD
`
`This invention relates generally to voice compression
`techniques, and more particularly a method and apparatus of
`voice compression using efficient bandwidth utilization and
`time compression techniques.
`
`10
`
`BACKGROUND
`
`15
`
`SUMMARY OF THE INVENTION
`
`In one aspect, the present invention comprises a method
`for compressing a plurality of voice signals within a voice
`communication resource having a given bandwidth within a
`voice communication system. The method comprises the
`steps of subchanneling the voice communication resource
`while placing at least one of each of the plurality of voice
`signals on a subchannel and compressing the time of each of
`the voice signals within each of the subchannels, wherein
`these steps provide a compressed voice signal.
`In another aspect of the present invention, a communica(cid:173)
`tion system using voice compression has at least one trans-
`
`IPR2020-00033 Page 00016
`
`

`

`5,689,440
`
`3
`FIG. 12 is another timing diagram illustrating a control
`frame and two analog frames of the outbound signaling
`protocol in accordance with the present invention.
`FIGS. 13-17 illustrate timing diagrams for several itera(cid:173)
`tions of the WSOLA time-scaling (compression) method in 5
`accordance with the present invention.
`FIGS. 18-22 illustrate timing diagrams for several itera(cid:173)
`tions of the WSOLA-SD time-scaling (compression)
`method in accordance with the present invention.
`FIGS. 23-24 illustrate timing diagrams for iterations of 10
`the WSOLA-SD time-scaling (expansion) method in accor(cid:173)
`dance with the present invention.
`FlG. 25 illustrates a block diagram of the overall
`WSOLA-SD time scaling method in accordance with the
`present invention.
`DEI'AILED DESCRIPTION OF THE
`PREFERRED EMBODIMENT
`Referring to FIG. 1, a communication system illustrative
`of the voice compression and expansion techniques of the
`present invention are shown in a block diagram of the
`selective call system 100 which comprises an input device
`for receiving an audio signal such as telephone 114 ( or other
`input device such as a computer) from which voice based
`selective calls are initiated for transmission to selective call
`receivers in the system 100. Each selective call entered
`through the telephone 114 typically comprises ( a) a receiver
`address of at least one of the selective call receivers in the
`system and (b) a voice message. The initiated selective calls
`are typically provided to a transmitter base station or a
`selective call terminal 113 for formatting and queuing. Voice
`compression circuitry 101 of the terminal 113 serves to
`compress the time length of the provided voice message (the
`detailed operation of such voice compression circuitry 101
`is discussed in the following description of FIGS. 2, 3 and
`4). Preferably, the voice compression circuitry 101 includes
`a processing device for compressing the audio signal using
`a time-scaling technique and a single sideband modulation
`technique to provide a processed signal. The selective call is
`then input to the selective call transmitter 102 where it is 40
`applied as modulation to a radio frequency signal which is
`sent over the air through an antenna 103. Preferably, the
`transmitter is a quadrature amplitude modulation transmitter
`for transmitting the processed signal.
`An antenna 104 within a selective call receiver 112 45
`receives the modulated, transmitted radio frequency signal
`and inputs it to a selective call receiver module or radio
`frequency receiver module 105 for receiving the processed
`signal or radio frequency signal, where the radio frequency
`signal is demodulated and the receiver address and the 50
`compressed voice message modulation are recovered. The
`compressed voice message is then provided to an analog to
`digital converter (AID) 115. Preferably, the selective call
`receiver 112 includes a processing device for demodulating
`the received processed signal using a single sideband 55
`demodulation technique and a time-scaling expansion tech(cid:173)
`nique to provide a reconstructed signal. The compressed
`voice message is then provided to a voice expansion circuit
`106 where the time length of the voice message is preferably
`expanded to the desired value (the detailed operation of such 60
`voice expansion circuitry 106 used in the present invention
`is discussed in the following description of FIGS. 7 and 8).
`The voice message is then provided to al! amplifier such as
`audio amplifier 108 for the purpose of amplifying it to a
`reconstructed audio signal.
`The demodulated receiver address is supplied from the
`radio frequency receiver 105 to a decoder 107. If the
`
`4
`receiver address matches any of the receiver addresses
`stored in the decoder 107, an alert 111 is optionally
`activated, providing a brief sensory indication to the user of
`the selective call receiver 112 that a selective call has been
`received. The brief sensory indication may comprise an
`audible signal, a tactile signal such as a vibration, or a visual
`signal such as a light, or a combination thereof. The ampli(cid:173)
`fied voice message is then furnished from the audio ampli(cid:173)
`fier 108 to an audio loudspeaker within the alert 111 for
`message announcement and review by the user.
`The decoder 107 may comprise a memory in which the
`received voice messages can be stored and recalled repeat(cid:173)
`edly for review by actuation of one or more controls 110.
`In another aspect of the invention, portions of FIG. 1 can
`15 be equally interpreted as part of a dictation device, voice
`mail system, answering machine, or sound track editing
`device for example. By removing the wireless aspects of the
`system 100 including the removal of selective call transmit(cid:173)
`ter 102 and radio frequency receiver 105, the system can be
`20 optionally hardwired from the voice compression circuitry
`101 to the voice expansion circuitry 106 through the AID
`115 as shown with the dashed line. Thus, in a voice mail,
`answering machine, sound track editing or dictation system,
`an input device 114 would supply an acoustic input signal
`25 such as a speech signal to the terminal 113 having the voice
`compression circuitry 101. The voice expansion circuitry
`106 and controls 110 would supply the means of listening
`and manipulating to the output speech signal in a voice mail,
`answering machine, dictation, sound track editing or other
`30 applicable system. This invention clearly contemplates that
`the time-scaling techniques of the claimed invention has
`many other applications besides paging. The paging
`example disclosed herein is merely illustrative of one of
`those applications.
`Now referring to FIG. 2, there is shown a block diagram
`of a paging transmitter 102 and terminal 113 including an
`amplitude compression and filtering module 150 coupled to
`a time compression module 160 which is coupled to the
`selective call transmitter 102 and which transmits messages
`using aerial or antenna 103. Referring to FIGS. 3 and 4, a
`lower level block diagram of the block diagram of FIG. 2 is
`shown.
`Please keep in mind that this compressed voice paging
`system is highly bandwidth efficient and intended to support
`typically 6 to 30 voice messages per 25 kHz channel using
`the basic concepts of quadrature amplitude (QAM) or
`single-side band (SSB) modulation and time scaling of
`speech signals. Preferably, in a first embodiment and also
`referring to FIG. 6, the compressed voice channel or voice
`communication resource consists of 3 sub-channels that are
`separated by 6250 Hz. Each sub-channel consists of two
`side-bands and a pilot carrier. Each of these two side-bands
`may have the same message in a first method or separate
`speech messages on each sideband or a single message split
`between the upper and lower sidebands in a second method.
`The single sub-channel has a bandwidth of substantially
`6250 Hz with each side-band occupying a bandwidth of
`substantially 3125 Hz. The actual speech bandwidth is
`substantially 300-2800 Hz. Alternatively, the quadrature
`amplitude modulation may be used where the two indepen-
`dent signals are transmitted directly via I and Q components
`of the signal to form each sub-channel signal. The band(cid:173)
`width required for transmission is the same in the QAM and
`65 SSB cases.
`Note that modules 150 and 160 in FIG. 2 can be repeated
`for use by each different voice signal (up to 6 times in 25
`
`35
`
`IPR2020-00033 Page 00017
`
`

`

`5,689,440
`
`5
`KHz wide channels and up to 14 times in 50 KHz wide
`channels) to allow for the efficient and simultaneous trans(cid:173)
`mission of (up to 6 in examples shown) voice messages.
`They can all then be summed at a summing device (not
`shown, but see F1G. 5) and preferably processed as a 5
`composite signal in 102. A separate signal (not shown)
`contains the FM modulation of the FLEX™ protocol ( as will
`be described later) which may optionally be generated in
`software or as a hardware FM signal exciter.
`·
`Preferably, in the examples shown herein, an incoming 10
`speech message is received by the terminal 113. The present
`system preferably uses a time-scaling scheme or technique
`to achieve the required compression. The preferred com(cid:173)
`pression technique used in the present invention requires
`certain parameters specific to the incoming message to 15
`provide an optimum quality. Preferably, the technique of
`time-scale compression processes the speech signal into a
`signal having the same bandwidth characteristics as uncom(cid:173)
`pressed speech. (Once these parameters are computed,
`speech is compressed using the desired time-scaling com- 20
`pression technique). This time-scaled compressed speech is
`then encoded using a digital coder to reduce the number of
`bits required to be distributed to the transmitters. In the case
`of a paging system, the encoded speech distributed to the
`transmitters of multiple simulcasting sites in a simulcasting 25
`paging system would need to be decoded once again for
`further processing such as amplitude compression. Ampli(cid:173)
`tude compression of the incoming speech signals (preferably
`using a syllabic compander) is used at the transmitter to give
`protection against channel impairments.
`A time scaling technique known as Waveform Sinrilarity
`based Overlap-Add technique or WSOLA encodes speech
`into an analog signal having the same bandwidth character(cid:173)
`istics as uncompressed speech. This property of WSOLA
`allows it to be combined with SSB or QAM modulation such
`that the overall compression achieved is the product of the
`bandwidth compression ratio of multiple QAM or SSB
`subchannels (in our example, 6 voice channels) and the time
`compression ratio of WSOLA (typically between 1 and 5).
`In the present invention, a modified version of WSOLA,
`later described and referred to as "WSOLA-SD" is used.
`WSOLA-SD retains the compatibility characteristics of
`WSOLA that allows the combination with SSB or QAM
`modulation.
`Preferably, an Adaptive Differential Pulse Coded Modu(cid:173)
`lation coder (ADPCM) is used to encode the speech into data
`that is subsequently distributed to the transmitters. At the
`transmitter, the digital data is decoded to obtain WSOLA(cid:173)
`SD compressed speech which is then amplitude companded
`to provide protection against channel noise. This signal is
`Hilbert transformed to obtain a single-sideband signal.
`Alternatively, the signal is quadrature modulated to obtain a
`QAM signal. A pilot carrier is then added to the signal and
`the final signal is interpolated, preferably, to a 16 kHz
`sampling rate and converted to analog. This is then modu(cid:173)
`lated and transmitted.
`The present invention can operate as a mixed-mode (voice
`or digital) one or two way communications system for
`delivering analog voice and/or digital messages to selective
`call receiver units on a foiward channel ( outbound from the
`base transmitter) and for receiving acknowledgments from
`the same selective call receiver units which additionally
`have optional transmitters ( on an optional reverse channel
`(inbound to a base receiver). The system of the present
`invention preferably utilizes a synchronous frame structure 65
`sinrilar to FLEX™ (a high speed paging protocol by
`Motorola, Inc. and subject of U.S. Pat. No. 5,282,205, which
`
`6
`is hereby incorporated by reference) on the foiward channel
`for both addressing and voice messaging. Two types of
`frames are used: control frames and voice frames. The
`control frames are preferably used for addressing and deliv(cid:173)
`ery of digital data to selective call receivers in the form of
`portable voice units (PVU's). The voice frames are used for
`delivering analog voice messages to the PVU's. Both types
`of frames are identical in length to standard FI..EXIM frames
`and both frames begin with the standard FLEX™ synchro(cid:173)
`nization. These two types of frames are time multiplexed on
`a single forward channel. The frame structure for the present
`invention will be discuss in greater detail later on with
`regard to FIGS. 10, 11 and 12.
`With regard to modulation, two types of modulation are
`preferably used on the foiward channel of the present
`invention: Digital FM (2-level and 4-level FSK) and AM
`(SSB or QAM with pilot carrier). Digital FM modulation is
`used for the sync portions of both types of frames, and for
`the address and data fields of the control frames. AM
`modulation (each sideband maybe used independently or
`combined together in a single message) is used in the voice
`message field of the voice frames. The digital FM portions
`of the transmission support 6400 BPS (3200 Baud symbols)
`signaling. The AM portions of the transmissions support
`band limited voice (2800 Hz) and require 6.25 KHz for a
`pair of voice signals. The protocol, as will be shown later,
`takes advantage of the reduced AM bandwidth by subdivid(cid:173)
`ing a full channel into 6.25 KHz subchannels, and by using
`each subchannel and the AM sidebands for independent
`30 messages.
`Voice System of the present invention is pr