US3784747A - Speech suppression by predictive filtering - Google Patents

Speech suppression by predictive filtering Download PDF

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Publication number
US3784747A
US3784747A US00204509A US3784747DA US3784747A US 3784747 A US3784747 A US 3784747A US 00204509 A US00204509 A US 00204509A US 3784747D A US3784747D A US 3784747DA US 3784747 A US3784747 A US 3784747A
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Prior art keywords
signal
speech
undesired
waveform
parameter
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US00204509A
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D Berkley
O Mitchell
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AT&T Corp
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Bell Telephone Laboratories Inc
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    • 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
    • H04M9/082—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 echo cancellers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B3/00—Line transmission systems
    • H04B3/02—Details
    • H04B3/20—Reducing echo effects or singing; Opening or closing transmitting path; Conditioning for transmission in one direction or the other
    • 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
    • H04M9/087—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 different frequency bands for transmitting and receiving paths ; using phase shifting arrangements
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L2021/02082—Noise filtering the noise being echo, reverberation of the speech
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L2021/02087—Noise filtering the noise being separate speech, e.g. cocktail party

Definitions

  • ABSTRACT Speech signal energy from an undesired source is suppressed by extracting from the undesired signal a delay parameter and a gain parameter. These parameters control a delay and gain network through which both the desired and undesired signals are routed.
  • the invention is applied to several handsfree telephony situations and to the suppression of one of two speakers in a room.
  • a general object of the invention is to reduce the energy from an undesired speech source in a composite signal containing desired speech.
  • Another object of the invention is tosuppress an undesired speech signal in an electronic communications channel.
  • a specific object of the invention is to render a desired speech signal relatively more intelligible despite the presence of undesired speech energy.
  • a particular inventive object is to achieve the foregoing objects in the hands-free telephony situation.
  • Another specific object of the invention is to avoid voice switching functions and thus enable full-time duplex operation of a hands-free telephone channel.
  • Yet another inventive object is to distinguish one talker from another nearby talker and to suppress speech signals from one of them.
  • the invention is grounded in the general recognition that an unwanted speech signal can be rejected on the basis of its speech parameters.
  • the basic concept contemplated by the present invention is to extract, from the undesired signal, a gain parameter and a delay parameter. These parameters control a delay and gain network through which both desired speech signals and the undesired signal are routed.
  • the delay is approximately equal to the current duration of the pitch period of the undesired speech.
  • the gain is calculated, in accordance with one of several possible formulas, so as to bring the delayed unwanted signal to the amplitude level of the present value of the unwanted signal.
  • the gain is set equal to l.
  • the network output is then subtractively applied to the unwanted signal or to any composite signal containing the unwanted signal. The process may be carried out in analog or digital fashion.
  • the process is carried out by sampling techniques where the signal is sampled at a rate of, for example, 6 kHz that results in 30 to 60 samples per pitch period.
  • the number of samples in a pitch period will vary in accordance with the pitch frequency.
  • speech from the loudspeaker of a hands-free telephone set impinging either directly or reverberatively on the sets microphone can be largely removed from the microphone output.
  • the reverberant signal as well as the direct signal is suppressed because the unwanted speech parameters do not vary rapidly during voicing.
  • speech from, for example, two talkers in the same room is detected by a multiplicity of microphones, and the speech of one talker is suppressed using speech parameters determined by combining the outputs of the microphones.
  • the rearrangement of the Atal process constitutes in one aspect a filter; and more specifically, a comb filter with minima at the pitch frequency (and harmonics hereof) of the undesired speech.
  • a comb filter with minima at the pitch frequency (and harmonics hereof) of the undesired speech This distinguishes the predictive filter of the present invention from a conventional echo canceler which merely replicates a reverberant signal and subtractively applies the replica to the composite signal.
  • FIG. 1 is a communications network schematic block diagram containing a hands-free telephone and an inventive embodiment
  • FIG. 2 is a schematic block diagram of the inventive predictive filter
  • FIG. 3 is a schematic block diagram further delineating the inventive predictor
  • FIGS. 4-6 are graphs depicting various characteristics of the predictor
  • FIGS. 7 and 8 are two further embodiments of the invention in a communications network containing hands-free telephones.
  • FIGS. 9 and 10 are schematic diagrams of the invention as applied to suppression of speech from talkers in a room.
  • a hands-free telephone loudspeaker 1 and microphone 2 present in a reverberative enclosure 3 are shown in FIG. 1 connected to the speech processor of the present invention.
  • the desired speech signal input to microphone 2 is from source 4, the near-end talker, whose signal denoted a travels mainly the direct path 5 and also reverberative paths not shown.
  • Loudspeaker 1 which broadcasts the far-end talker signal, is a source of undesired input to microphone 2 either via the direct path denoted 6 or reverberative paths illustrated by path 7.
  • the far-end talker direct path speech signal is denoted
  • the speech processing network 8 in FIG. 1 consists of what will be called a predictive filter 9 connected in the microphone 2 output circuit.
  • filter 9 consists of two parallel legs.
  • the first leg is a predictor 11 which may be a network consisting of adelay network 12 and an amplifier 13.
  • the second leg is a direct shunt path. Both legs are connected to a subtractor 10.
  • the predictor 11 is controlled in a manner to be described, by a parameter extractor 14 connected in the loudspeaker l circuit.
  • a low pass filter l5 advantageously 3 kHz and a 6 kHz sampler 16 are serially connected in the output circuit of microphone 2.
  • a low pass filter 17 and a sampler 18 are in shunt relation to the loudspeaker ll input circuit and serially connected to parameter extractor l4.
  • a waveform representing the far-end talker signal 0 is illustrated in FIG. 4. Because a speech signal is redundant-Le, the signal changes little in shape and length of pitch period from one pitch period to the next -the present form or value of signal c can be estimated by a linear prediction based on a past value of signal c.
  • the signal 0 of FIG. 4 is shown made up of speech in consecutive pitch periods I, I 1 etc. lnherently, the speech signals in adjacent pitch periods of signal c are of unequal amplitude.
  • a gain denoted b can be calculated (in a manner to be described) that when applied to the sampled signal of the pitch period 1 will cause the latter to approximate the sampled signal in the next pitch period 1
  • the amplified signal of period I is subtractively combined with the signal value of period the result is the substantial filtering out of the signal 0.
  • a composite signal a c containingsignal c is amplified during period 1 and subtractively combined with the composite signal a 0 during period 1 the same result obtains.
  • W (the amplitude of sample n reaching subtractor via the direct path in predictive filter 9) is subtractively combined with W,, (the amplitude of the delayed sample) where k is the number of samples in a pitch period.
  • the time window over which the parameters are evaluated is of the order of the pitch period to ensure that sufficient energy is present.
  • a time window of 30 samples at a sampling rate of 6 kHz will include between one-half and all the samples in a given pitch period.
  • input speech samples from sampler 18 are stored as frames of signals.
  • the store content is then fed to an arithmetic unit which is part of parameter extractor 14, wherein for 30 samples, computational values of correlation X, are computed as follows:
  • N can advantageously be in the range 30-60 samples.
  • the computed values of X are then inspected in a peak locating network also part of extractor 14, to de termine the largest value of X, The value ofj is found such that X, is the maximum of all values of X.
  • This particular value ofj is the delay parameter, k, which is supplied to predictive filter 9 as one parameter. It is seen that k is a variable delay and that the maximum value of X, is X
  • the delay parameter k for a typical voiced segment is shown in FIG. 6.
  • the gain parameter b is calculated by computing circuitry also in parameter extractor 14, that solves:
  • the gain parameter b likewise is supplied to predictive filter 9.
  • delay parameter k and gain parameter b is but one of several systems by which, from an analysis of the speech energy content in adjacent or substantially adjacent signal segments, parameters may be calculated that when applied to a past signal segment will render the latter closely similar to the shape of the present signal segment.
  • incoming speech to loudspeaker 1 is continuously analyzed to extract therefrom an optimum delay parameter, and a gain factor. These parameters are periodically updated as for example, every 5 ms. When no incoming signal to loudspeaker l is present, the delay and gain are zero. With incoming signal, the calculated present signal value output of predictor 11 is subtracted from the undelayed, unamplified signal sample representing signals a c.
  • the filter depicted in FIG. 3 and described above has a transfer function in Z transform notation.
  • H(Z) l bZ' The magnitude of the frequency response of a typical embodiment of filter 9 is shown in FIG. 5 using predictor 11 where T is the sampling period.
  • the frequency response for gain parameter b l are shown by the solid curves and for gain parameter bzl by the broken line. Since speech is dynamic during voicing, the parameters b and k have to be optimized as stated above, and readjusted periodically as, for example, every 5 ms.
  • the parameters b and k calculated do not vary smoothly with time.
  • the optimum delay occasionally doubles during voiced segments. Also, during unvoiced segments, the optimum delayvaries rapidly over a wide range while the correlation remains relatively low. However, the gains calculated are not negligible during these unvoiced portions. Desired speech a, uncorrelated with the undesired signal 0 which is to be rejected, is degraded when passed through a filter with these rapidly varying filter parameters, while under such conditions no additional suppression of the unwanted source is accomplished.
  • the predictive filter 9 will be effective in removing part of the reverberant signal as well as the direct sound. Specifically, that part of the reverbcrant signal that has parameters not greatly different from the filter parameters will be reduced in amplitude.
  • the far-end echo picked up by the microphone 2 from loudspeaker 1 is first reduced in amplitude during voiced segments by a speech processor 8 in the manner described previously.
  • Gain and delay parameters b and k of the far-end speech are measured on the received loudspeaker signal, and the far-end echo component of the microphone signal is reduced by filtering.
  • the remaining far-end signal at the output of the speech processor 8 is then removed V by the center-clipping echo suppressor.
  • parameter delay circuit 19A which is serially connected between the output of parameter control 19 and predictive filter 9.
  • Parameter delay circuit 19A advantageously is provided with a delay duration adjustment circuit 198 with which the delay duration may be set to correspond to the transit time which characterizes each given hands-free telephone.
  • FIGS. 7 and 8 A combination of a predictive filter with a centerclipping echo suppressor of the type taught in D. A Berkley-O. M. M. Mitchell-J. R. Pierce U.S. Pat. No. 3,699,271 which is hereby incorporated by reference, is shown in FIGS. 7 and 8. This combination is a possible replacement for voice switching presently used for echo and feedback suppression.
  • FIG. 7 shows a network denoted for eliminating the echo ofthe far-end talker in a 4-wire hands-free tel-
  • the received signal is used to set the clipping levels by means of clipping control 22 so as just to remove the echo.
  • the output of D/A converter 10A is fed to filter bank 40 which comprises plural contiguous band filters in the voice frequency range.
  • filter bank 40 which comprises plural contiguous band filters in the voice frequency range.
  • center clipper 41 the signal in each subband from filter 41 is center clipped at a level determined by clipping control 22 which measures in effect the'energy level in the received signal within each of the subbands.
  • the output of clipper 41 is filtered in bank 42 which is similar to bank 40.
  • the clipping control 22 is advantageously controlled also by the parameter extractor 14. Since the echo is reduced by the predictive filter 9 during voicing, the clipping levels can be reduced by substantially the same amount during voicing. Consequently in FIG. 7, a control signal is shown (dashed line) between the parameter extractor l4 and the clipping control 22, which causes an attenuation of the input to clipping level control 22 that is equal to the suppression achieved by speech processing network 8. It will be recognized that optimum performance of clipping level control 22 will be realized by inserting a delay in its input path to compensate for the already mentioned signal transit time between loudspeaker 1 and microphone 2. With the clipping levels thus reduced during voiced segments, there will be less mutilation of the near-end speech by the center-clipping process.
  • FIG. 8 shows a circuit for eliminating both the farend echo (echo of far-end talker caused by acoustic coupling through room acoustics) and near-end echo (echo of near-end talker caused by imperfect hybrid junction) in a 2-wire hands-free telephone.
  • the far-end echo is eliminated by network 50 as described above for FIG. 7.
  • the near-end echo is eliminated by a similar circuit denoted 51 introduced on the receive side of the local 4-wire network as shown.
  • circuit 51 An alternative method of adjusting the clipping level control by the parameter extractor 14 via the parameter delay 19A is shown in circuit 51.
  • a second predictive filter designated 9a is used in circuit 51 to attenuate the clipping level control signal during voiced segments.
  • the clipping levels follow the signal at the input to the narrow band center clipper, i.e., at the output of the predictive filter 9a.
  • FIG. 9 shows the desired speech source 23 and an undesired source 24 both of whose speech signals form the input to microphones 25 and 26.
  • the undesired source 24 is positioned so that the time delays for direct sound transmission to microphones 25 and 26 are equal.
  • the output of microphone 25 is predictive filter 9 as microphone 26, enter the parameter extractor 30 wherein an arithmetic unit within the extractor calculates the computational values and A peak picking network within the extractor then selects the peaks from X, and Y, and a comparator finds the largest value peak which occurs in both sequence X, and Y for the same value ofj. This value ofj is the delay parameter k for the undesired speech supplied to the predictive filter 9.
  • FIG. 10 An alternative method of extracting the parameters is shown in FIG. 10.
  • Two additional microphones 27 and 28 are positioned so that time delays from desired speaker 24 for direct sound transmission to microphones 27 and 28 are equal to the time delays to microphones 25 and 26.
  • the outputs of all microphones 25-28 are processed by a non-linear processor 31 as described in O. M. M. Mitchell-C. A. Ross-R. L. Wallace, Jr. U.S. Pat. No. 3,644,671, which is hereby incorporated by reference.
  • the output of processor 31 contains the undesired signal and an attenuated and disturbed component of the desired signal. (The outputs may alternatively be added to merely attenuate the desired signal.)
  • the output of the non-linear processor 31 enters the speech processing network 8.
  • the output of microphone 25 is processed by speech processing network 8 which filters out the undesired talker 24 in the manner already described.
  • speech processing network 8 which filters out the undesired talker 24 in the manner already described.
  • the presence in the output of the nonlinear processor 31 of a small amount of the desired talker does not significantly affect the delay parameter k but will cause a small error in the evaluation of X and b.
  • Speech processing apparatus for suppressing voiced segments of an undesired speech signal while leaving a desired speech signal intelligible, comprising:
  • Apparatus in accordance with claim 1 further comprising means for deriving from the waveform of said undesired speech signal a gain parameter specifying the amount by which the amplitudes of corresponding values of said undesired speech signal in a past said interval must be respectively adjusted so as to produce a substantial duplicate of the undesired said speech signal of a present said interval; and which further comprises means controlled by said gain parameter for amplifying said delayed composite speech waveform prior to its being subtractively applied to said summer.
  • a communications network comprising:
  • a hands-free telephone station including a direct acoustic coupling path between the station loudspeaker and microphone, a second remote telephone station, and transmission means interconnecting said stations;
  • a communications network pursuant to claim 4 wherein said deriving means comprises:
  • X1 2 2 it) in) 71 I! ll further comprising means for rendering said gain parameter equal to zero in the absence of voiced segments of the signal in said loudspeaker path from said remote station.
  • a communications network pursuant to claim 8, whefe is the speechigflal receive? by Said fi further comprising means for adjustably delaying armlcrolphone and W" the Speech slgnal recelved rival of said delay and gain parameters at said second by Sam P path by an amount that compensates for the transit meahs for Selectmg the largest Value P from the time delay over said direct acoustic coupling path of composite P ,Vahles Ofsaid Parameters 1 and 1 speech from said remote station. for the Same Value of the term j;
  • a communications network pursuant to claim 4, means for pp y the desired and the undesired Said f th i i signals from one of said microphones to a summer filter bank means connected to the output of said irectly over a first path and alternately over a secsummer and comprising plural contiguous sub- 0nd path through a network including delay means; band s; and

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  • Computer Networks & Wireless Communication (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
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US00204509A 1971-12-03 1971-12-03 Speech suppression by predictive filtering Expired - Lifetime US3784747A (en)

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JP (1) JPS55760B2 (enrdf_load_stackoverflow)
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Cited By (24)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US3922488A (en) * 1972-12-15 1975-11-25 Ard Anstalt Feedback-cancelling electro-acoustic transducer apparatus
US4024358A (en) * 1975-10-31 1977-05-17 Communications Satellite Corporation (Comsat) Adaptive echo canceller using differential pulse code modulation encoding
US4031338A (en) * 1976-02-10 1977-06-21 Communications Satellite Corporation (Comsat) Echo suppressor using frequency-selective center clipping
US4122303A (en) * 1976-12-10 1978-10-24 Sound Attenuators Limited Improvements in and relating to active sound attenuation
US4166924A (en) * 1977-05-12 1979-09-04 Bell Telephone Laboratories, Incorporated Removing reverberative echo components in speech signals
FR2451676A1 (fr) * 1979-03-12 1980-10-10 Soumagne Joel Detecteur d'echo notamment pour systeme de communication a interpolation de parole
US4360708A (en) * 1978-03-30 1982-11-23 Nippon Electric Co., Ltd. Speech processor having speech analyzer and synthesizer
EP0106640A1 (en) * 1982-10-15 1984-04-25 British Telecommunications Noise control circuit
US4473906A (en) * 1980-12-05 1984-09-25 Lord Corporation Active acoustic attenuator
US4591670A (en) * 1982-09-30 1986-05-27 Nec Corporation Echo canceller and echo suppressor for frequency divisional attenuation of acoustic echoes
US4670903A (en) * 1981-06-30 1987-06-02 Nippon Electric Co., Ltd. Echo canceller for attenuating acoustic echo signals on a frequency divisional manner
EP0204718A4 (en) * 1984-12-14 1988-03-30 Motorola Inc FULLY DUPLEX HANDSET FOR RADIO TELEPHONES AND AERIAL CABLE.
US4819263A (en) * 1986-06-30 1989-04-04 Cellular Communications Corporation Apparatus and method for hands free telephonic communication
US4825384A (en) * 1981-08-27 1989-04-25 Canon Kabushiki Kaisha Speech recognizer
EP0472356A1 (en) * 1990-08-16 1992-02-26 Fujitsu Ten Limited Speech recognition apparatus for a vehicle, using a microphone arrangement to determine the seat from which a command is generated
US5619566A (en) * 1993-08-27 1997-04-08 Motorola, Inc. Voice activity detector for an echo suppressor and an echo suppressor
EP0881814A1 (de) * 1997-05-28 1998-12-02 Deutsche Telekom AG Verfahren zur Bestimmung des Schrifttweitenfaktors Alpha zur Einstellung der Adaptionsgeschwindigkeit des NMLS-Adaptionsalgorithmus
WO2001035118A1 (en) * 1999-11-05 2001-05-17 Wavemakers Research, Inc. Method to determine whether an acoustic source is near or far from a pair of microphones
US6249581B1 (en) * 1997-08-01 2001-06-19 Bitwave Pte. Ltd. Spectrum-based adaptive canceller of acoustic echoes arising in hands-free audio
US6442275B1 (en) 1998-09-17 2002-08-27 Lucent Technologies Inc. Echo canceler including subband echo suppressor
US20030219112A1 (en) * 2002-05-22 2003-11-27 Boland Simon Daniel Apparatus and method for echo control
US20030219087A1 (en) * 2002-05-22 2003-11-27 Boland Simon Daniel Apparatus and method for time-alignment of two signals
EP1739654A2 (de) 1999-09-08 2007-01-03 Volkswagen AG Verfahren zum Betrieb einer Mehrfachmikrofonanordnung in einem Kraftfahrzeug sowie Mehrfachmikrofonanordnung selbst
US7734034B1 (en) 2005-06-21 2010-06-08 Avaya Inc. Remote party speaker phone detection

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JP3289057B2 (ja) * 1996-02-06 2002-06-04 品川白煉瓦株式会社 浸漬ノズル交換装置
EP3806489A4 (en) 2018-06-11 2021-08-11 Sony Group Corporation SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD AND PROGRAM
CN112203188B (zh) * 2020-07-24 2021-10-01 北京工业大学 一种自动音量调节方法

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US3133990A (en) * 1962-04-27 1964-05-19 Altec Lansing Corp Automatic level-adjustment circuit
US3603744A (en) * 1965-09-29 1971-09-07 Superior Continental Corp Line tap unit for telephone system
US3631520A (en) * 1968-08-19 1971-12-28 Bell Telephone Labor Inc Predictive coding of speech signals
US3644674A (en) * 1969-06-30 1972-02-22 Bell Telephone Labor Inc Ambient noise suppressor
US3601549A (en) * 1969-11-25 1971-08-24 Bell Telephone Labor Inc Switching circuit for cancelling the direct sound transmission from the loudspeaker to the microphone in a loudspeaking telephone set

Cited By (26)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US3922488A (en) * 1972-12-15 1975-11-25 Ard Anstalt Feedback-cancelling electro-acoustic transducer apparatus
US4024358A (en) * 1975-10-31 1977-05-17 Communications Satellite Corporation (Comsat) Adaptive echo canceller using differential pulse code modulation encoding
US4031338A (en) * 1976-02-10 1977-06-21 Communications Satellite Corporation (Comsat) Echo suppressor using frequency-selective center clipping
US4122303A (en) * 1976-12-10 1978-10-24 Sound Attenuators Limited Improvements in and relating to active sound attenuation
US4166924A (en) * 1977-05-12 1979-09-04 Bell Telephone Laboratories, Incorporated Removing reverberative echo components in speech signals
US4360708A (en) * 1978-03-30 1982-11-23 Nippon Electric Co., Ltd. Speech processor having speech analyzer and synthesizer
FR2451676A1 (fr) * 1979-03-12 1980-10-10 Soumagne Joel Detecteur d'echo notamment pour systeme de communication a interpolation de parole
US4473906A (en) * 1980-12-05 1984-09-25 Lord Corporation Active acoustic attenuator
US4670903A (en) * 1981-06-30 1987-06-02 Nippon Electric Co., Ltd. Echo canceller for attenuating acoustic echo signals on a frequency divisional manner
US4825384A (en) * 1981-08-27 1989-04-25 Canon Kabushiki Kaisha Speech recognizer
US4591670A (en) * 1982-09-30 1986-05-27 Nec Corporation Echo canceller and echo suppressor for frequency divisional attenuation of acoustic echoes
EP0106640A1 (en) * 1982-10-15 1984-04-25 British Telecommunications Noise control circuit
EP0204718A4 (en) * 1984-12-14 1988-03-30 Motorola Inc FULLY DUPLEX HANDSET FOR RADIO TELEPHONES AND AERIAL CABLE.
US4819263A (en) * 1986-06-30 1989-04-04 Cellular Communications Corporation Apparatus and method for hands free telephonic communication
EP0472356A1 (en) * 1990-08-16 1992-02-26 Fujitsu Ten Limited Speech recognition apparatus for a vehicle, using a microphone arrangement to determine the seat from which a command is generated
US5619566A (en) * 1993-08-27 1997-04-08 Motorola, Inc. Voice activity detector for an echo suppressor and an echo suppressor
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JPS55760B2 (enrdf_load_stackoverflow) 1980-01-09
DE2207141B2 (enrdf_load_stackoverflow) 1980-10-09
DE2207141C3 (de) 1981-07-30
CA952439A (en) 1974-08-06
JPS4865813A (enrdf_load_stackoverflow) 1973-09-10
DE2207141A1 (de) 1973-08-02

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