AU615820B2

AU615820B2 – Computer controlled adaptive speakerphone
– Google Patents

AU615820B2 – Computer controlled adaptive speakerphone
– Google Patents
Computer controlled adaptive speakerphone

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Publication number
AU615820B2

AU615820B2
AU47216/89A
AU4721689A
AU615820B2
AU 615820 B2
AU615820 B2
AU 615820B2
AU 47216/89 A
AU47216/89 A
AU 47216/89A
AU 4721689 A
AU4721689 A
AU 4721689A
AU 615820 B2
AU615820 B2
AU 615820B2
Authority
AU
Australia
Prior art keywords
signal
communication line
transmit
speech
receive
Prior art date
1988-12-28
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)

Ceased

Application number
AU47216/89A
Other versions

AU4721689A
(en

Inventor
Richard Henry Erving
William Albert Ford
Robert Raymond Miller Ii
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)

AT&T Corp

Original Assignee
American Telephone and Telegraph Co Inc
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
1988-12-28
Filing date
1989-12-22
Publication date
1991-10-10

1989-12-22
Application filed by American Telephone and Telegraph Co Inc
filed
Critical
American Telephone and Telegraph Co Inc

1990-07-05
Publication of AU4721689A
publication
Critical
patent/AU4721689A/en

1991-10-10
Application granted
granted
Critical

1991-10-10
Publication of AU615820B2
publication
Critical
patent/AU615820B2/en

2009-12-22
Anticipated expiration
legal-status
Critical

Status
Ceased
legal-status
Critical
Current

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Classifications

H—ELECTRICITY

H04—ELECTRIC COMMUNICATION TECHNIQUE

H04M—TELEPHONIC COMMUNICATION

H04M9/00—Arrangements for interconnection not involving centralised switching

H04M9/08—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic

H04M9/085—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic using digital techniques

Description

I ii- r S F Ref: 114705 FORM COMMONWEALTH OF AUSTRALIA PATENTS ACT 1952 1 5 8 COMPLETE SPECIFICATION 6
(ORIGINAL)
FOR OFFICE USE: Class Int Class i Complete Specification Lodged: Accepted: Published: Priority: Related Art: Name and Address of Applicant: American Telephone and Telegraph Company 550 Madison Avenue New York New York 10022 UNITED STATES OF AMERICA Address for Service: Spruson Ferguson, Patent Attorneys Level 33 St Martins Tower, 31 Market Street Sydney, New South Males, 2000, Australia Complete Specification for the invention entitled: Computer Controlled Adaptive Speakerphone The following statement is a full description of this invention, including the best method of performing it known to me/us 0 5845/4 i i ;»ii -1 Computer Controlled Adaptive Speakerphone Background of the Invention 1. Technical Field This invention relates to audio systems and, more particularly, to voice switching circuits which connect to an audio line for providing two-way voice switched communications.
2. Description of the Prior Art The use of analog speakerphones have been the primary hands free means of communicating during a telephone conversation for a great number of years. This convenient service has been obtained at the price of some limitations, however.
These speakerphone usually require careful and expensive calibration in order to *operate in an acceptable manner. They are also designed to operate in a worst-case electrical and acoustic environment thereby sacrificing the improved performance that is possible in a better environment.
The operation of conventional analog speakerphones is well known and is described in an article by A. Busala, «Fundamental Considerations in the Design of a Voice-Switched Speakerphone,» Bell System Technical Journal, Vol. 39, No. 2, March 1960, pp 265-294. Analog speakerphones generally use a switched-loss technique through which the energy of the voice signals in both a transmit and a receive direction are sensed and a switching decision made based upon that information. The voice signal having the highest energy level in a first direction will I be given a clear talking path and the voice signal in the opposite direction will be attenuated by having loss switched into its talking path. If voice signals are not S° present in either the transmit direction or the receive direction, the speakerphone 25 goes to an «at rest» mode which provides the clear talking path to voice signals in a receive direction favoring speech from a distance speaker. In some modern analog *00 o i i -2speakerphones, if voice signals are not present in either the transmit direction or the ii receive direction, the speakerphone goes to an idle mode where the loss in each direction is set to a mid-range level to allow the direction wherein voice signals first Sappear to quickly obtain the clear talking path.
5 Most high-end analog speakerphones also have a noise-guard circuit to adjust the switching levels according to the level of background noise present.
Switching speed is limited by a worst-case time constant that assures that any speech energy in the room has time to dissipate. This limitation is necessary to prevent «self switching», a condition where room echoes are falsely detected as near-end speech. No allowance is made for a room that has good acoustics, i.e. low echo energy return and short duration echoes.
A disadvantage associated with analog speakerphones is that they are .difficult to calibrate, or require precision voltage references to assure consistent 15operation. In some designs, the newly manufactured analog speakerphone performs well, but over the coursc of a few years, its performance degrades to the point where it becomes unusable. In one known example, a critical calibration value relied on the stability of two different power supplies in the speakerphone. Over a period of time, one or the other of the supplies tended to drift enough to significantly change the speakerphone’s performance.
In order to provide appropriate switching in an analog speakerphone, transmit and receive signal strengths are measured to provide a logic switching unit *in the speakerphone with information as to what the current state of the speakerphone should be. This logic unit usually consists of circuitry that compares .5 the current audio levels against calibrated thresholds provided by the voltage references. The result of this comparison determines the state of the speakerphone.
Thus, these thresholds must be precisely controlled in order to keep speakerphone performance optimal.
of ‘d i i3 i~E i ia i:I ii I 0e 0 *0 so *00 0 0 0 *as.
a 0 The analog speakerphone is also unable to adapt to the hybrid it faces when attached to a telephone line. Even a digital telephone within a private branch exchange (PBX), which does not employ a hybrid, faces an unpredictable hybrid on calls outside of the PBX. As with other parameters, a worst case trans-hybrid loss must be assumed. This assumption also requires the insertion of more switched loss than might be necessary in order to assure that the system will remain stable. A high «break in» threshold is similarly required in order to prevent a bad hybrid from reflecting enough transmit speech to falsely switch the speakerphone into the receive state.
While the above arrangements may have been acceptable in the past in providing reasonable hands free communications for a user, it is now desirable to have an efficient and cost effective speakerphone without the disadvantages and limitations associated with the operation of these systems.
Summary of the Invention An adaptive speakerphone under the control of, for example, a computer, measures the energy of incoming transmit and receive signals and also develops information about the signal and noise levels for self calibration and efficient operation.
For accurately determining when the speakerphone should be in each of three operating states, i. transmit, receive or idle, in accordance with the invention, the computer recalibrates its operating parameters before operating by updating thresholds used to determine its state. These updated thresholds counteract parts variation and aging and are obtained by passing a computergenerated test tone signal through the speakerphone circuitry at two different levels 25 and measuring the resulting response.
In accordance with the calibration process and the inveation, the speakerphone measures the acoustics of the room in which it operates. This it 0 *0 0 0 0 -4achieves by emitting a tone burst through its loudspeaker and measuring the returned time-domain acoustic response with its microphone. Obtained from this response and processed by the computer are the maximum amplitude of the returned signal, and the duration of the echoes. The amplitude of the returned signal determines what level of transmit speech will be required to break in on receive speech. The greater the acoustic return, the higher that threshold must be to protect against self-switching. The duration of the echoes determine how quickly speech energy injected into the room will dissipate, which, in turn, controls how fast the speakerphone can switch from a receive to a transmit state.
In order to compensate for the inherent gain between the loudspeaker and the microphone, a certain amount of loss is inserted at some point in the speakerphone circuitry to maintain stability. The amount of this loss depends upon the amount of hybrid return, the amount of acoustic return and the volume level 9 setting. The speakerphone determines these conditions and inserts the amount of switched loss necessary to maintain stability.
9 Brief Description of the Drawing FIG. 1 is a block representation of the major functional components of a computer controlled adaptive speakerphone operative in accordance with the principles of the invention; FIG. 2 is a partial schematic of the speakerphone including a calibration «circuit, an amplifier for remotely provided speech signals, a microphone and an associated amplifier and multiplexers employed in this invention; 0 FIG. 3 is a partial schematic of the speakerphone including mute controls and high pass filters employed in this invention; 25 FIG. 4 is a schematic of a programmable attenuator and a low pass filter 9. 9 employed in a transmit section of this invention; 99 9 9 9
I
1 St I
I
0* *0 a 0 FIG. 5 is a schematic of a programmable attenuator and a low pass filter employed iri a receive section of this invention; FIG. 6 depicts a general speakerphone circuit and two types of coupling that most affect its operation; FIG. 7 is a state diagram depicting the three possible states of the speakerphone of FIG. 1; FIG. 8 depicts a flow chart illustrating the operation of the speakerphone of FIG. 1 in determining whether to remain in an idle state or move from the idle state to a transmit or a receive state; FIG. 9 depicts a flow chart illustrating the operation of the speakerphone of FIG. 1 in determining whether to remain in the transmit state or move from the transmit state to the receive state or idle state; FIG. 10 depicts a flow chart illustrating the operation of the speakerphone of FIG. 1 in determining whether to remain in the receive state or move from the receive state to the transmit state or idle state; FIG. 11 are illustrative waveforms which depict impulse and composite characterizations of an acoustic environment performed by the speakerphone of FIG. 1; FIG. 12 is a block representation of the functional components of a 20 speakerphone operable in providing echo suppression loss insertion; FIG. 13 depicts a flow chart illustrating the operation of the speakerphone of FIG. 12 in the application of echo suppression loss insertion; and FIG. 14 are waveforms illustrating the application of echo suppression loss insertion.
o a -6- Detailed Description FIG. 1 is a functional block representation of a computer controlled adaptive speakerphone 100 operative in accordance with the principles of the invention. As shown, the speakerphone generally comprises a transmit section 200, a receive section 300, and a computer 110. A microcomputer commercially available from Intel Corporation as Part No. 8051 may be used for computer 110 with the proper programming. A microphone 111 couples audio signals to the speakerphone and a speaker 112 receives output audio signals from the speakerphone.
By way of operation through illustration, an audio signal provided by a person speaking into the microphone 111 is coupled into the transmit section 200 to a multiplexer 210. In addition to being able to select the microphone speech signal as an input, the multiplexer 210 may also select calibration tones as its input. These calibration tones are provided by a calibration circuit 113 and are used, in this instance, for calibration of the hardware circuitry in the transmit section 200.
15 Connected to the multiplexer 210 is a mute control 211 which mutes the transmit path in response to a control signal from the computer 110. A high pass filter 212 connects to the mute control 211 to remove the room and low frequency background noise in the speech signal. The output of the high pass filter 212 is coupled both to a programmable attenuator 213 and to an envelope detector 214. In *0 «20 response to a control signal from the computer 110, the programmable 0 •00* attenuator 213 inserts loss in the speech signal i, three and one half dB steps up to a Stotal of sixteen steps, providing 56 dB of total loss. This signal from the programmable attenuator 213 is coupled to a low pass filter 215 which removes any spikes that might have been generated by the switching occurring in the 25 attenuator 213. This filter also provides additional signal shaping to the signal before the signal is transmitted by the speakerphone over audio line 101 to a hybrid (not shown). After passing through the envelope detector 214, the speech signal from the filter 212 is coupled to a logarithmic amplifier 216, which expands the -7i dynamic range of the speakerphone to approximately 60 dB for following the envelope of the speech signal.
The receive section 300 contains speech processing circuitry that is functionally the same as that found in the transmit section 200. A speech signal received over an input audio line 102 from the hybrid is coupled into the receive section 300 to the multiplexer 310. Like the multiplexer 210, the multiplexer 310 may also select calibration tones for its input, which are provided by the calibration circuit 113. Connected to the multiplexer 310 is a mute control 311 which mutes the receive path in response to a control signal from the computer 110. A high pass filter 312 is connected to the mute control 311 to remove the low frequency background noise from the speech signal.
The output of the high pass filter 312 is coupled both to an envelope detector 314 and to a programmable attenuator 313. The envelope detector 314 obtains the signal envelope for the speech signal which is then coupled to a 15 logarithmic amplifier 316. This amplifier expands the dynamic range of the speakerphone to approximately 60 dB for following the envelope of the receive speech signal. The programmable attenuator 313, responsive to a control signal from the computer 110, inserts loss in the speech signal in three and one half dB steps in sixteen steps, for 56 dB of loss. This signal from the programmable 20 attenuator 313 is coupled to a low pass filter 315 which removes any spikes that might have been generated by the switching occurring in the attenuator 313. This filter also provides additional signal shaping to the signal before the signal is coupled to the loudspeaker 112 via an amplifier 114.
The signals from both the logarithmic amplifier 216 and the logarithmic *25 amplifier 316 are multiplexed into an eight-bit analog-to-digital converter 115 by a S. multiplexer 117. The converter 115 presents the computer 110 with digital S» information about the signal levels every 750 microseconds.
_111 -8- The computer 110 measures the energy of the incoming signals and develops information about the signal and noise levels. Both a transmit signal average and a receive signal average are developed by averaging samples of each signal according to the following equation: SI t Yt-1 Yt-1 if st t Is it Yt-1 Yt= Yt-1 3 if Ist -1 where *oo0 Sampling rate 1333 per second I I t new sample t-1 old average 10 Yt new average This averaging technique tends to pick out peaks in the signal applied. Since speech tends to have many peaks rather than a constant level, this average favors So* detecting speech.
Both a transmit noise average and a receive noise average are also developed.
The transmit noise average determines the noise level of the operating environment of the speakerphone. The receive noise average measures the noise level on the line from the far-end party. The transmit noise average and the receive noise average are both developed by measuring the lowest level seen by the converter 115. Since background noise is generally constant, the lowest samples provide a reasonable 20 estimate of the noise level. The transmit and receive noise averages are developed using the following equation: -9- Yt-1 if ISIt Yt-1 ;Ye= 1II f Yt-1 if Is receive state for receiving speech signals from the communication line and a transirit state for transmitting speech signals over the communication line, the gl -31 method comprising the steps of: testing the operational readiness of speech processing circuitry in the controller; determining the type of acoustic environment in which the voice signal controller is employed; inserting loss alternately in a receive path for attenuating speech signals received from the communication line and in a transmit path for attenuating speech signals for transmission over the communication line; adjusting threshold switching levels at which the controller switches between a receive state for receiving speech signals and a transmit state for transmitting speech signals responsive to the operational readiness testing step; and adjusting the level of attenuation inserted by the loss insertion step in response to the acoustic environment determining step.

The method of processing speech signals in a voice signal controller as in claim 9 wherein the operational readiness testing step further includes the step of determining the type of communication line to which the voice signal controller is .connected, the loss insertion step and the threshold switching levels adjusting step

11. The method of processing speech signals in a voice signal controller as in claim 10 wherein the line determining type step further includes the step of receiving a signal from the communication line, the signal receiving step providing a signal indicative of the return level of a transmit speech signal provided by the voice signal controller to the communication line for transmission over the S communication line, the threshold switching levels adjusting step being operably 25 adjusted by the line determining type step.

12. The method of processing speech signals in a voice signal controller as in claim 10 wherein the line determining type step further includes the step of receiving a signal from the communication line, the signal receiving step providing a SNOR -32- signal indicative of the return level of a transmit speech signal provided by the voice signal controller to the communication line for transmission over the communication line, the loss insertion step being operably adjusted by the line determining type step.

13. The method of processing speech signals in a voice signal controller as in claim 9 wherein the operational readiness testing step further includes the steps of generating a tone signal, coupling this tone signal in a loop configuration through the speech processing circuitry and detecting the returned tone signal, responsive to the detecting step, the threshold switching levels adjusting step adjusting the switching levels to compensate for any change in the form of the returned tone signal.

14. The method of processing speech signals in a voice signal controller as in claim 13 further including the step of inserting echo suppression loss in the transmit path for attenuating speech signals for transmission over the communication line.

15. The method of processing speech signals in a voice signal controller as in 0000.0 S» claim 14 wherein the echo suppression inserting step includes the steps of measuring •a predetermined threshold coupling level and comparing the speech signal received •from the communication line with the threshold coupling level, the echo suppression P e: inserting step being operable for providing additional loss in the transmit path Si 20 when the level of the received speech signal exceeds that of the threshold coupling level. j

16. The method of processing speech signals in a voice signal controller as in S claim 15 wherein the predetermined coupling threshold level is operably adjusted by the threshold switching levels adjusting step. s.ee DATED this FIRST day of DECEMBER 1989 «American Telephone and Telegraph Company 9 003 0000 )0 Patent Attorneys for the Applicant SPRUSON FERGUSON 0 0 0 SO a. O•

AU47216/89A
1988-12-28
1989-12-22
Computer controlled adaptive speakerphone

Ceased

AU615820B2
(en)

Applications Claiming Priority (2)

Application Number
Priority Date
Filing Date
Title

US298531

1988-12-28

US07/298,531

US5007046A
(en)

1988-12-28
1988-12-28
Computer controlled adaptive speakerphone

Publications (2)

Publication Number
Publication Date

AU4721689A

AU4721689A
(en)

1990-07-05

AU615820B2
true

AU615820B2
(en)

1991-10-10

Family
ID=23150921
Family Applications (1)

Application Number
Title
Priority Date
Filing Date

AU47216/89A
Ceased

AU615820B2
(en)

1988-12-28
1989-12-22
Computer controlled adaptive speakerphone

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US
(1)

US5007046A
(en)

EP
(1)

EP0376582B1
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JP
(1)

JPH02260856A
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AU
(1)

AU615820B2
(en)

CA
(1)

CA2004171C
(en)

DE
(1)

DE68916218T2
(en)

ES
(1)

ES2055796T3
(en)

HK
(1)

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(en)

*

1985-08-06
1987-09-10
Outel Oy

MELLANFOERSTAERKARE SOM ANVAENDS I TVAOLEDNINGSFOERBINDELSER I ETT VALBART TELEFONNAET SAMT FOERFARANDE FOER REGLERING AV DESS FOERSTAERKNING.

JPS6260343A
(en)

*

1985-09-10
1987-03-17
Seiko Epson Corp
Telephone set

US4887288A
(en)

*

1988-12-28
1989-12-12
American Telephone And Telegraph Company
Self calibration arrangement for a voice switched speakerphone

1988

1988-12-28
US
US07/298,531
patent/US5007046A/en
not_active
Expired – Lifetime

1989

1989-11-29
CA
CA002004171A
patent/CA2004171C/en
not_active
Expired – Fee Related

1989-12-19
ES
ES89313269T
patent/ES2055796T3/en
not_active
Expired – Lifetime

1989-12-19
DE
DE68916218T
patent/DE68916218T2/en
not_active
Expired – Lifetime

1989-12-19
EP
EP89313269A
patent/EP0376582B1/en
not_active
Expired – Lifetime

1989-12-22
AU
AU47216/89A
patent/AU615820B2/en
not_active
Ceased

1989-12-26
JP
JP1335229A
patent/JPH02260856A/en
active
Granted

1995

1995-03-23
HK
HK43595A
patent/HK43595A/en
not_active
IP Right Cessation

Patent Citations (2)

* Cited by examiner, † Cited by third party

Publication number
Priority date
Publication date
Assignee
Title

US4368361A
(en)

*

1980-07-28
1983-01-11
Bell Telephone Laboratories, Incorporated
Automatically adjustable bidirectional-to-unidirectional transmission network

GB2161047A
(en)

*

1984-06-28
1986-01-02
Stc Plc
Improvements in telephone instruments

Also Published As

Publication number
Publication date

US5007046A
(en)

1991-04-09

CA2004171C
(en)

1994-08-02

EP0376582B1
(en)

1994-06-15

CA2004171A1
(en)

1990-06-28

DE68916218D1
(en)

1994-07-21

EP0376582A3
(en)

1990-12-05

EP0376582A2
(en)

1990-07-04

DE68916218T2
(en)

1995-02-02

ES2055796T3
(en)

1994-09-01

JPH02260856A
(en)

1990-10-23

JPH0544220B2
(en)

1993-07-05

AU4721689A
(en)

1990-07-05

HK43595A
(en)

1995-03-31

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