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WO1999033323A1 - Procede et appareil pour une conference telephonique a plusieurs voies avec un telephone haut-parleur en mode duplex integral et une interface de communication numerique sans fil - Google Patents

Procede et appareil pour une conference telephonique a plusieurs voies avec un telephone haut-parleur en mode duplex integral et une interface de communication numerique sans fil Download PDF

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Publication number
WO1999033323A1
WO1999033323A1 PCT/US1998/026952 US9826952W WO9933323A1 WO 1999033323 A1 WO1999033323 A1 WO 1999033323A1 US 9826952 W US9826952 W US 9826952W WO 9933323 A1 WO9933323 A1 WO 9933323A1
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WO
WIPO (PCT)
Prior art keywords
signal
talker
room
echo
outside line
Prior art date
Application number
PCT/US1998/026952
Other languages
English (en)
Inventor
Jeffrey D. Keith
Kenneth E. Garey
Original Assignee
Conexant Systems, 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.)
Filing date
Publication date
Application filed by Conexant Systems, Inc. filed Critical Conexant Systems, Inc.
Priority to AU20025/99A priority Critical patent/AU2002599A/en
Publication of WO1999033323A1 publication Critical patent/WO1999033323A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00—Public address systems

Definitions

  • the present invention relates generally to speakerphones and more specifically pertains to speakerphones for multi-way conferencing which incorporate one or more digital wireless communication interfaces, such as cordless handsets.
  • a full-duplex speakerphone includes a hands-free telephone having a base with a microphone and a loudspeaker.
  • a third interface may be added to allow a third party to participate on an equal basis with the room talker and line talker for three-way conferencing.
  • Such three-way conferencing machines typically operate in the analog domain and do not utilize signal processors with echo cancellers. Moreover, the proposed devices to date are believed to work in half-duplex mode but not in full- duplex mode.
  • FIG. 1 A speakerphone 2 operates in digital domain but only has a speakerphone base 10 with a microphone 12 for room talk and a loudspeaker 14, connected to a Public Switched Telephone Network (PSTN) line, for line talk.
  • PSTN Public Switched Telephone Network
  • the speakerphone 2 allows a talker in a room to converse with another talker speaking on the PSTN input line.
  • the room signal is picked up by the microphone 12. Since the room talker may vary his distance from the microphone 12, an automatic gain control (AGC) circuit 42 is required to amplify the signal from the room talker to a listenable level and to maintain the power level of the signal, before it gets transmitted to the PSTN output line.
  • AGC automatic gain control
  • the PSTN input line signal is amplified with another AGC circuit 44 to an appropriate level before it is applied to the loudspeaker 14.
  • echo cancellers 46, 48 In some speakerphones, echo is canceled with two signal processing mechanisms called echo cancellers (ECs) 46, 48, one of which is used for acoustic echo and another for line echo.
  • Each EC 46, 48 has a signal input, a feedback input connected to the output of a corresponding summing element 50, 52 and an output connected to a negative input of the same summing element 50, 52.
  • ECs typically use linear adaptive filters, such as filters with adaptive finite impulse response (AFIR), whose coefficients are weighted in accordance with the room and line acoustics.
  • AFIR adaptive finite impulse response
  • any device that filters echo may be used for echo cancellers 46, 48.
  • Adaptive filters in the AEC 46 and LEC 48 generate a replica of the corresponding acoustic or line echo which is subtracted in the corresponding summing element 50, 52 from the signal on the positive input of the summing element 50, 52, which is either the room talk or PSTN line talk.
  • Coefficients of adaptive filters are updated every sample period and minimize error signals according to an algorithm. They must be constantly updated to account for changes in the acoustic environment.
  • this speakerphone 2 works in full-duplex mode, it can only support two-way conferencing. Therefore, there is a need for a speakerphone for multi-way conference calling, equipped with one or more digital wireless communication interfaces to allow additional parties to participate on an equal basis with the room talker and line talker. Moreover, there is a need for a speakerphone system which works in full-duplex mode.
  • Another object of the present invention is to provide a speakerphone system with a digital wireless communication interface which works in full-duplex mode.
  • a digital wireless communication interface is attached at the input of each echo canceller.
  • the acoustic echo canceller receives the room talker signal, the line input signal and a signal from the digital wireless communication interface and cancels acoustic echo from the room talker signal.
  • the line echo canceller receives the line input signal, the room talker signal and the signal from the digital wireless communication interface and cancels line echo from the line input signal.
  • the full- duplex speakerphone device is provided with a first summing device for digitally adding the signal from the digital wireless communication interface to the line input signal, a second summing device for digitally adding the signal from the digital wireless communication interface to the room talker signal, and a third summing device for digitally adding the room talker signal and the line input signal, and providing a signal to be input into the digital wireless communication interface.
  • the digital wireless communication interface is preferably a cordless handset working in spread- spectrum.
  • Each echo canceller is preferably an adaptive finite impulse response filter using a predetermined algorithm, preferably the least-means-squares algorithm, to provide an echo-canceled output signal.
  • Figure 1 is a schematic illustration of a conventional speakerphone device.
  • Figure 2 is a schematic illustration of the components of a conferencing device using a speakerphone and a digital wireless communication interface, in accordance with a preferred embodiment of the present invention.
  • Figure 3 is a schematic illustration of the components of a conferencing device using a speakerphone and two digital wireless communication interfaces, in accordance with a preferred embodiment of the present invention.
  • FIG. 4 is a schematic illustration of the components of a conferencing device using two speakerphone, in accordance with a preferred embodiment of the present invention.
  • FIG. 2 is a schematic illustration of the components of a conferencing device using a speakerphone 4 and a digital wireless communication interface 20, in accordance with the preferred embodiment of the present invention.
  • the speakerphone 4 is used to accommodate more signal sources than just one room talker and the line talker, and to allow a multi-way conversation.
  • the speakerphone base 10 is connected to one or more digital wireless communication interfaces 20.
  • the digital wireless communication interface 20 is a digital spread-spectrum cordless handset 20 with a microphone 22 and an earpiece 24.
  • the cordless handset 20 is preferably a digital 900 MHz spread-spectrum cordless telephone.
  • the speakerphone system 4 has a receive signal path and a transmit signal path. In the receive signal path, the line signal from the PSTN input line i(t) 21 is converted to digital form in an analog-to-digital converter (ADC) 58 and any line echo is estimated by the line echo canceller 48 and subtracted from the digitized input signal i(n) 23.
  • ADC analog-to-digital converter
  • the residual input signal r(n) 25 is amplified by the AGC 44 which maintains its output power at a specified level.
  • This amplified residual input signal a(n) 26 is converted into analog form in a digital-to-analog converter (DAC) 60 and output to the loudspeaker 14.
  • DAC digital-to-analog converter
  • the signal from the room talker y(t) 27 is picked up by the speakerphone microphone 12 and converted to digital form y(n) 28 in another analog-to-digital converter (ADC) 62.
  • Room echo is estimated by the acoustic echo canceller 46 and subtracted from the digitized room signal y(n) 28.
  • the amplified residual output signal x(n) 31 is converted to analog signal as signal in a digital-to-analog converter (DAC) 64 and output to the PSTN output line as signal o(t) 33.
  • DAC digital-to-analog converter
  • a signal from the handset microphone 22 m(n) 36 is added digitally into the speakerphone system 4 signals a(n) 26 and x(n) 31 with summing devices 32 and 34.
  • the digitized room signal y(n) 28, after room echo removal by AEC 46, AGC 42 and attenuator 66, and PSTN input line signal i(n) 23, after line echo removal by LEC 48, are extracted from the speakerphone 4 lines and added in a summing device 40 to form a signal e(n) 38 for transmission to the handset earpiece 24.
  • signals m(n) 36 and e(n) 38 are amplified in one of multipliers 35, 37, 39, 41. Therefore, signal m(n) 36 from the handset microphone 22 is digitally added to the signal a(n) 26, input into the acoustic echo canceller 46, and to the signal x(n) 31 , input into the line canceller 48.
  • the acoustic echo canceller 46 can be trained by a signal from either the PSTN input line i(n) 23 or the handset 20 signal m(n) 36, and the line echo canceller 48 can be trained by either the room talker signal y(n) 28 or the handset talker signal m(n) 36 (assuming absence of double-talk).
  • This connection cancels echoes heard in the handset 20 as well as by the room talker and PSTN line talker.
  • the acoustic echo canceller 46 of Figure 2 receives the digitized room talker signal y(n) 28 which has a room echo, the outside line input signal 25 after echo removal by LEC 48, and the wireless interface talker signal m(n) 36, and provides an estimate of the acoustic echo in the room talker signal y(n) 28 at its output.
  • the summing element 50 is a difference means having a first input coupled to the acoustic echo canceller 46 output and a second input coupled to the room talker signal y(n) 28.
  • the summing element 50 subtracts the acoustic echo estimate from the room talker signal y(n) 28 to produce an echo-canceled room talker signal z(n) 29.
  • the line echo canceller 48 performs the same on the outside line input signal i(n) 23 which has line echo, with the summing element 52 and produces an echo-canceled outside line input signal r(n) 25.
  • Each EC 46, 48 uses a training algorithm for the adjustment of its coefficients.
  • it is preferably the least-mean-square (LMS) algorithm, which is an implementation of the steepest descent method. It could also be a variant of the LMS algorithm, preferably the variant with partial block updating.
  • LMS least-mean-square
  • Any other filtering algorithm can also be used if it provides quick convergence and filter stability.
  • both echo cancellers 46, 48 may be trained at once using only the signal from the handset microphone 22, m(n) 36, as a training signal, if there is no room talk or the PSTN line talk. Additionally, AEC 46 can be trained when there is a significant line input signal i(n) 23 and negligible room talker signal y(n) 28. Similarly, training of the LEC 48 is not allowed when talk is detected on the PSTN input line i(n) 23.
  • the device of the present invention works in four modes: handset mode, speakerphone mode, intercom mode and conference mode.
  • speakerphone mode the handset 20 is not used and two-way conferencing is performed between a line talker and a room talker.
  • the handset 20 replaces the speakerphone base 10 as the source and destination of signals to the PSTN output line and from the PSTN input line.
  • the AEC 46 is disabled in handset mode and there is no need for cancellation of acoustic echo because there is negligible acoustic echo from the handset 20.
  • the level of the handset 20 output signal m(n) 36 does not vary much, this connection does not need an AGC circuit.
  • the intercom mode there is no PSTN input signal i(t) 21 and output signal o(t) 33 and the AGC 44 is not used.
  • the room signal y(n) 28 is directed to and received from the cordless handset 20 and instead of the line echo there is a small acoustic coupling between the earpiece 24 and the microphone 22 in the handset 20.
  • the speakerphone 4 of the illustrated invention operates as previously described.
  • the handset microphone 22 signal m(n) 36 is digitally added to the amplified residual input signal a(n) 26 in the summing device 32, positioned before the AEC 46.
  • the handset microphone 22 signal m(n) 36 is also digitally added to the amplified residual output signal x(n) 31 in the summing device 34, positioned before the LEC 48.
  • AEC 46 and LEC 48 cancel acoustic and line echo.
  • Amplified residual output signal from the room x(n) 31 and the residual input signal r(n) 25 are added in the summing device 40 and output to the handset earpiece 24 as the signal e(n) 38.
  • echo cancellers 46, 48 do not provide perfect cancellation.
  • a potential feedback loop i.e., a gain loop
  • AGC circuit 42 in the transmit path amplifies the signal z(n) 29 to be sent to the PSTN output line as o(t) 33.
  • AGC circuit 44 performs the same function for the receive path. Therefore, the gain around the gain loop may become greater than unity at some frequencies and the speakerphone 4 is prone to oscillation. This is not a problem in handset mode, since there is not enough coupling from the earpiece 24 to the microphone 22 to yield an unstable gain loop.
  • the attenuation is performed in the opposite path of the gain loop than the one where the talk is received from. For example, if there is no talk received from the PSTN input line i(t) 21 but there is room talk y(t) 27, the attenuation is applied to the receive path of the gain loop. Similarly, if there is no talk signal y(t) 27 received from the room but there is the input line talk signal i(t) 21 , the attenuation is applied to the transmit path of the gain loop. Accordingly, the present invention includes attenuation of the gain loop signal.
  • two stability attenuators 66, 68 are used to reduce gain during the time when there is no talk in that path and the amplification is not needed.
  • These stability attenuators 66, 68 are digital multipliers, each positioned after the respective AGCs 42, 44, and may be dynamically adjusted by an algorithm to stabilize the gain loop while attenuating active talkers as little as possible.
  • the handset 20 itself does not add any variation in the input signal m(n) 36 because the signal level for a given person is fairly constant and the microphone 22 of the handset is held closer to the person's mouth than the microphone 12 of the speakerphone 4. Therefore, there is no need to attenuate the amplified input signal m(n) 36 in the handset mode since there is no feedback path. Moreover, the present invention does not need additional AGC circuits for the handset-to-room and room-to-handset signals. Furthermore, the PSTN-handset two-way connection does not need any AGC circuits because the handset 20 can be easily matched to standard levels of the PSTN line input signal i(t) 21 and output signal o(t) 33 required by existing standards.
  • Another way to prevent oscillations in the gain loop and avoid the possible feedback would be to break the gain loop by using the system in half-duplex mode. Since in this mode only one party can be heard at the time and the other party cannot interrupt it, this solution produces an inferior device.
  • the system may have to be started and run for a short period of time in half-duplex mode until the training is accomplished.
  • the ECs 46, 48 are trained whenever one, but not both, room and PSTN talkers are active. AEC 46 is trained during PSTN input line talk or handset talk and LEC 48 is trained during room talk.
  • ECs 46, 48 were AFIR filters performing LMS algorithm with partial block updating.
  • the AEC 46 had 520 taps and block size 26
  • LEC 48 had 168 taps and block size 12.
  • an additional handset 70 receives at its earpiece 74 the signal m(n) 36 from the first handset 20 added in a summing element 76 to the signal e(n) 38.
  • the output from the handset microphone 72 is added to the signal e(n) 38 in the summing element 40 for transmission to the first handset earpiece 24.
  • the output from the handset microphone 72 is added to the input signal m(n) 36 in a summing element 78 for transmission to the multiplier 41 , and replaces the output of the first handset 36 as the input to the multiplier 35.
  • an additional speakerphone 80 which is preferably placed in another room, replaces the handset earpiece 24 and microphone 22.
  • Signal from the second speakerphone microphone 82 is digitized in an analog-to-digital converter 83.
  • Room echo is estimated by the acoustic echo canceller 81 and subtracted from the digitized room signal.
  • the residual output signal is amplified by an AGC 84 and sent through a stability attenuator 85 to the amplifiers 41 and 35 in the first speakerphone.
  • Signal e(n) 38 is amplified in an AGC 86 and sent through a stability attenuator 87 to a digital-to-analog converter 88 to be output to a loudspeaker 89 of the second speakerphone.
  • the present invention though applicable to any digital cordless telephone, is believed to be especially applicable to the telephones with digital spread spectrum.
  • the third and additional parties with cordless handsets may be located in the same room or be positioned at another location in the vicinity of the room with the speakerphone 4. It is understood that the principles of this invention may be applied to other digital devices which work in full duplex mode, like digital telephone answering devices.
  • Those skilled in the art will appreciate that various adaptations and modifications of the just-described preferred embodiment can be configured without departing from the scope and spirit of the invention. For instance, while the present invention has been described for three-way use, it is understood that the speakerphone 4 of the present invention could be modified to handle any number of users by incorporating additional digital wireless communication interfaces 20.
  • the digital wireless communication interface 20 when located in a room away from the speakerphone 4, may be substituted with another speakerphone.
  • This speakerphone only needs one echo canceller, AEC 46, because there is no other outside line and no other line echo signal. Therefore, it is to be understood that, within the scope of the appended claims, the invention may be practiced other than as specifically described herein.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Telephone Function (AREA)
  • Telephonic Communication Services (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)

Abstract

La présente invention concerne un téléphone haut-parleur destiné à une conférence téléphonique à plusieurs voies, comportant une ou plusieurs interfaces de communication numérique sans fil, de préférence des combinés sans fil. Le dispositif fonctionne en mode duplex total. Une pluralité de convertisseurs analogiques numériques procèdent à l'échantillonnage d'un signal de conversation interchambre et d'un signal de ligne d'entrée. Les échantillons sont d'abord additionnés numériquement aux signaux provenant de l'interface de communication numérique sans fil puis introduits dans des compensateurs d'écho pour éliminer l'écho acoustique et l'écho de ligne. Les compensateurs d'écho sont de préférence des filtres adaptifs à réponse impulsionnelle finie utilisant l'algorithme à erreurs quadratiques minimales. Ce système peut simultanément entraîner le compensateur de l'écho acoustique et le compensateur de l'écho de ligne en utilisant le signal du correspondant de l'interface sans fil.
PCT/US1998/026952 1997-12-19 1998-12-18 Procede et appareil pour une conference telephonique a plusieurs voies avec un telephone haut-parleur en mode duplex integral et une interface de communication numerique sans fil WO1999033323A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AU20025/99A AU2002599A (en) 1997-12-19 1998-12-18 Method and apparatus for multi-way conference calling with full-duplex speakerphone and a digital wireless communication terface

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US99520997A 1997-12-19 1997-12-19
US08/995,209 1997-12-19

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WO1999033323A1 true WO1999033323A1 (fr) 1999-07-01

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2003107719A1 (fr) * 2002-06-12 2003-12-24 Equtech Aps Procede d'egalisation numerique d'un son issu de haut-parleurs situes dans des pieces, et utilisation de ce procede
GB2397463A (en) * 2003-01-08 2004-07-21 Vtech Telecomm Ltd A speakerphone system having a plurality of wireless microphones
US9288572B2 (en) 2014-01-09 2016-03-15 International Business Machines Corporation Haptic microphone

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5268927A (en) * 1992-10-06 1993-12-07 Mayflower Communications Company, Inc. Digital adaptive transversal filter for spread spectrum receivers
EP0597201A1 (fr) * 1992-11-12 1994-05-18 Motorola, Inc. Appareil et procédé pour le réduction du bruit dans un téléphone à haut-parleur entièrement duplex ou similaire
WO1996014694A1 (fr) * 1994-11-04 1996-05-17 Sierra Semiconductor Corporation Procede et appareil permettant d'etablir une communication sonore en duplex integral pour un systeme a poste a haut-parleur
US5572575A (en) * 1994-03-24 1996-11-05 Matsushita Electric Industrial Co., Ltd. Cordless telephone system having speaker phone function
EP0765066A2 (fr) * 1995-09-21 1997-03-26 Rockwell International Corporation Téléphone à haut-parleur avec système efficace de suppression de réactions acoustiques

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5268927A (en) * 1992-10-06 1993-12-07 Mayflower Communications Company, Inc. Digital adaptive transversal filter for spread spectrum receivers
EP0597201A1 (fr) * 1992-11-12 1994-05-18 Motorola, Inc. Appareil et procédé pour le réduction du bruit dans un téléphone à haut-parleur entièrement duplex ou similaire
US5572575A (en) * 1994-03-24 1996-11-05 Matsushita Electric Industrial Co., Ltd. Cordless telephone system having speaker phone function
WO1996014694A1 (fr) * 1994-11-04 1996-05-17 Sierra Semiconductor Corporation Procede et appareil permettant d'etablir une communication sonore en duplex integral pour un systeme a poste a haut-parleur
EP0765066A2 (fr) * 1995-09-21 1997-03-26 Rockwell International Corporation Téléphone à haut-parleur avec système efficace de suppression de réactions acoustiques

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2003107719A1 (fr) * 2002-06-12 2003-12-24 Equtech Aps Procede d'egalisation numerique d'un son issu de haut-parleurs situes dans des pieces, et utilisation de ce procede
GB2397463A (en) * 2003-01-08 2004-07-21 Vtech Telecomm Ltd A speakerphone system having a plurality of wireless microphones
US6987992B2 (en) 2003-01-08 2006-01-17 Vtech Telecommunications, Limited Multiple wireless microphone speakerphone system and method
US9288572B2 (en) 2014-01-09 2016-03-15 International Business Machines Corporation Haptic microphone
US9666041B2 (en) 2014-01-09 2017-05-30 International Business Machines Corporation Haptic microphone

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