EP0765562A1 - Agencement de microphone adaptatif et procede d'adaptation a un signal de bruit cible d'arrivee - Google Patents

Agencement de microphone adaptatif et procede d'adaptation a un signal de bruit cible d'arrivee

Info

Publication number
EP0765562A1
EP0765562A1 EP95922851A EP95922851A EP0765562A1 EP 0765562 A1 EP0765562 A1 EP 0765562A1 EP 95922851 A EP95922851 A EP 95922851A EP 95922851 A EP95922851 A EP 95922851A EP 0765562 A1 EP0765562 A1 EP 0765562A1
Authority
EP
European Patent Office
Prior art keywords
signal
calibration
noise
arrangement according
beamformer
Prior art date
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.)
Withdrawn
Application number
EP95922851A
Other languages
German (de)
English (en)
Inventor
Sven Nordebo
Sven Nordholm
Ingvar Claesson
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.)
Nokia Oyj
Original Assignee
Nokia Mobile Phones Ltd
Volvo AB
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 Nokia Mobile Phones Ltd, Volvo AB filed Critical Nokia Mobile Phones Ltd
Publication of EP0765562A1 publication Critical patent/EP0765562A1/fr
Withdrawn legal-status Critical Current

Links

Classifications

    • 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/18—Methods or devices for transmitting, conducting or directing sound
    • G10K11/26—Sound-focusing or directing, e.g. scanning
    • G10K11/34—Sound-focusing or directing, e.g. scanning using electrical steering of transducer arrays, e.g. beam steering
    • G10K11/341—Circuits therefor
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H21/00—Adaptive networks
    • H03H21/0012—Digital adaptive filters
    • 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
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • 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
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02166—Microphone arrays; Beamforming
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
    • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
    • H04R2201/403—Linear arrays of transducers

Definitions

  • the present invention relates to an adaptive microphone ar ⁇ rangement as referred to in the first part of claim 1.
  • the invention furthermore relates to a method for adapting to an incoming target signal.
  • the conditions under which a microphone arrangement is to be used vary to a great extent. Sometimes the environment is very noisy, as for example in a car or any moving vehicle or simi ⁇ lar, moreover also in workshops, storehouses etc. When so called hands-free operation is applied, the requirements on the microphone arrangement is even more demanding among others due to the distance from the source of the speech or whatever it may, be to the microphones. E.g. the noisy environment in a car severely degrades the performance of so called hands free mobile telephones and speech recognition devices.
  • a further object of the invention is to provide a method for adapting to an incoming target signal.
  • the signal forming arrangement comprises an adaptive beamformer and a filtering beamformer.
  • the calibration signal is a speech signal or even more particularly a typical speech signal or a signal with a speech influenced spectrum.
  • the calibration signal is recorded on site, i.e. it is recorded using the same equipment and in an ad ⁇ vantageous embodiment at the same location as when the target target-noise signal is produced.
  • the storage com ⁇ prises a digital storage, or even more particularly one digital storage for each input calibration signal, each for a separate microphone.
  • the calibration signal may comprise a number of (secondary) calibration signals, i.e. calibration signals from each microphone which are combined into a so called desired signal.
  • the adapting means uses an adaptive al ⁇ gorithm which e.g. may be the so called LMS (Least Mean Square) algorithm or some other algorithm, for example the RLS (Recu ⁇ rsive Least Square) or any other appropriate algorithm. Par- ticularly either one of the calibration signals or a combina ⁇ tion of two or more thereof often is used as a so called desired signal in the algorithm means with which the sum of the calibration signal and the noise signal is compared in a manner known per se.
  • LMS Least Mean Square
  • RLS Recu ⁇ rsive Least Square
  • the adaptive beamformer During adaptation, during which no target signal or no speech is provided, a number of filtering coefficients are obtained in the adaptive beamformer in a manner known per se. The filtering coefficients are copied to and used in the second beamformer or the filtering beamformer.
  • a target (target-noise) signal When a target (target-noise) signal is input, or a speaker or similar is active, the adaptation of the adaptive beamformer is switched off and no adaptation takes place. Then the target signal or e.g. the speech signal is filtered through the filtering beamformer.
  • the first and second beamformers comprise filters such as e.g. FIR-filters (Finite Impulse Response), the adaptation coefficients thereof being optimized adaptively to the actual noise level or noise situation and to the equipment "on site".
  • Fig. 1 illustrates a calibration phase
  • Fig. 2 illustrates an adaptive filtering phase.
  • an array of microphones is arranged for example in a car.
  • an array comprising n microphones (MP X , MP 2 ,..., MP n ) is illustrated wherein n can be any number from one upwards and is chosen depending on the actual circumstances and the relev ⁇ ant environment. Thus there may be either one or more micro ⁇ phones.
  • 8 microphone are used but this of course merely constitutes an example.
  • the microphones may be of any appropriate quality or of any kind. If however they are of a standard quality, they generally have a con- siderable spread in performance which in turn poses high demands on the beamformer as to easily incorporate a calibra ⁇ tion step.
  • training sequences are recorded from different positions in the environment of e.g. a true speaker position in a real situation with the actual system and with no noise present.
  • the training sequences or the calibration signals are then gathered into a storage and later used as so called training signals in the adaptive phase. Therethrough an inherent calibration signal is obtained and it is generally possible to wheigh interesting frequency bands and spatial points.
  • the arrangement according to the invention is accurate for the actual situation and it does not depend on the geometry of the array of microphones or similarities between elements or on calibration or matching of amplifiers or other electronic equipment etc.
  • the microphone arrangement generally uses two sets of input data, namely the target-noise signals in a filtering beamformer and the recorded calibrations signals plus the noise signals in the adaptive beamformer.
  • the signals are filtered with so called FIR-filters or Finite Impulse Response filters or a so called tapped delay line, which carries out a linear combination of input data.
  • the microphone arrangement may particularly be used for so called hands free operation.
  • the microphone arrangement according to Fig. 1 comprises a number of microphones MP j , MP 2 , ..., MP n wherein the micropho ⁇ nes are arranged and placed in any desired manner.
  • the input calibration signals M l r ..., M n undergo, an anti - aliasing operation and an A/D conversion in a conversion block 1 where ⁇ after the signals, now designated m l r m 2 , ..., m n are recorded in a calibration signal storage 2.
  • the calibration signals m l f ... , m n are also used in the adaptive means as will be further described later on.
  • the calibration signal is to be provided as a pure calibration signal, i.e.
  • a typical speech signal or signal with a speech influenced spectrum, from the typical speaker position is recorded in the calibration signal storage 2.
  • This is preferab ⁇ ly a digital storage or more particularly a number of digital storages, each for one microphone channel.
  • These recorded signals form calibration signals l r ..., ⁇ L,.
  • the adaptive means or the adaptive beamformer 4 can advantageously be calibrated on-site in a car or similar e.g. by using either a loudspeaker or letting the speaker read a representative sequence. The sequences received in each microphone channel are gathered into the calibration signal storage 2.
  • the channels from the speaker or the loudspeaker or similar to A/D converters are included.
  • the environmental noise level should be as low as possible in order to obtain a good signal-to-noise ratio in a desired signal which may be either one of the input calibration signals m l r ... , m nl , or a combination of two or more of the calibration signals m x , m 2 , .... , ⁇ ,.
  • the situa ⁇ tion as well as the equipment is generally time-invariant wherethrough the microphone arrangement has been provided with calibration signals which can be combined to form the desired signals as referred to above.
  • the separate microphones MP X , MP 2 , ..., MP n and their placement can be chosen in any appropriate manner.
  • the speaker position or the loudspeaker position is changed in such way that it is moved around and in the vicinity of the speakers normal position during the recording of the calibration signal into the storage.
  • the recorded calibration signals from different positions are according to a preferred embodiment superimposed to provide weighted average training signals or calibration signals or reference signals. As already referred to above, these signals are gathered into the storage- 2. As can be seen from Fig.
  • those signals, m,, m 2 , m n and m r which forms so called calibration signals, or reference sig ⁇ nals, are then used as well as training signals as, e.g. in a combined form, as a desired signal or reference signal for use during adaptation.
  • an adaptive phase follows. During this phase there is no calibration input signal.
  • the situation is very generally a noisy situation, in the case of the car it may relate to a situation wherein the speaker is silent and wherein the car is moving, i.e. the motor is running etc.
  • the input signals to the adapting beamformer 4 are formed by the sum of the in the storage 2 stored calibra ⁇ tion signals m l r m 2 , ...,11 ⁇ and the noise signals H l r N 2 , ..., N n respectively.
  • the storage also comprises an arrangement (not shown) wherein e.g. a combined desired signal m r is formed.
  • a known reference signal or a desired signal m r which has passed through the same electronic equipment when no noise was present is also obtained.
  • the adaptive filters of the adapting beamformer 4 therethrough are provided with all the information that is needed to adapt to the correct filter coefficients e.g. in the least square sense or applying the LMS-algorithm (or any other appropriate algorithm).
  • the adaptive microphone arrangement will be calibrated "on site” to the prevailing acoustic enviro ⁇ nment and to the placement of the microphones etc. as well as to the individual properties of the microphones, amplifiers, A/D - converters and so on.
  • the coefficients of the digital filters of the adaptive beamformer 4 has been optimized adaptively to the current noise situation and to the actual equipment, these are copied to the second beamformer or the filtering beamformer 5.
  • the filtering beamformer 5 operates when the speaker or similar is active.
  • the adaptation is switched off, either automatically or manually e.g. by a "push- to-talk"-function. This relates to a preferred embodiment; it is however not necessary. If the adaptation is switched off, however, this is done to avoid echo-effects and/or to provide a more robust system so that the adaptive filters cannot operate on the real speech signal.
  • the target signal or the speech signal comprising speech plus noise, sn 1( sn 2 , ..., sn 3 is merely filtered through the filtering beamformer 5.
  • the filtering coefficients are fixed and the output signal is obtained from the filtering beamformer 5.
  • the filtering beamformer preferably works continuously and without any calibration signal.
  • the different components of the microphone arrangement can be of any desired kind. A number of different known microphone types can be used. Different filters can also be used of which so called FIR-fliters merely constitute one example. Also the storage can be chosen in any appropriate way. The sampling frequency may likewise take a number of different values. The invention may also in a number of other aspect be varied in a number of different ways merely being limited by the scope of the claims.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Health & Medical Sciences (AREA)
  • Signal Processing (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Quality & Reliability (AREA)
  • Computational Linguistics (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Filters That Use Time-Delay Elements (AREA)

Abstract

Agencement de microphone adaptatif muni d'un ou plusieurs microphones (MP1, ..., MPn) comportant un dispositif de détection de signaux destiné à détecter des signaux d'entrée cibles, un dispositif de formation de signaux et une unité de stockage de signaux. Les signaux d'entrée comportent un signal d'étalonnage (m1, ..., mn) et un second signal de bruit (N1, ..., Nn), le signal d'étalonnage d'entrée étant enregistré et stocké dans une unité de stockage (2). Le dispositif de formation de signaux comporte une première unité de formation de signaux (4) et une seconde unité de formation de signaux (5), la première unité de formation de signaux (4) comportant une unité d'adaptation servant à traiter la somme du signal d'étalonnage et d'un signal de bruit afin d'obtenir des coefficients de filtrage qui sont ensuite recopiés dans la seconde unité de formation de signaux (5) et utilisés dans celle-ci pour traiter le signal de bruit cible d'entrée. En outre, les signaux d'adaptation et les signaux de bruit cibles sont introduits essentiellement dans les mêmes conditions.
EP95922851A 1994-06-14 1995-06-13 Agencement de microphone adaptatif et procede d'adaptation a un signal de bruit cible d'arrivee Withdrawn EP0765562A1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
SE9402088 1994-06-14
SE9402088A SE502888C2 (sv) 1994-06-14 1994-06-14 Adaptiv mikrofonanordning och förfarande för adaptering till en inkommande målbrussignal
PCT/SE1995/000718 WO1995034983A1 (fr) 1994-06-14 1995-06-13 Agencement de microphone adaptatif et procede d'adaptation a un signal de bruit cible d'arrivee

Publications (1)

Publication Number Publication Date
EP0765562A1 true EP0765562A1 (fr) 1997-04-02

Family

ID=20394386

Family Applications (1)

Application Number Title Priority Date Filing Date
EP95922851A Withdrawn EP0765562A1 (fr) 1994-06-14 1995-06-13 Agencement de microphone adaptatif et procede d'adaptation a un signal de bruit cible d'arrivee

Country Status (5)

Country Link
EP (1) EP0765562A1 (fr)
JP (1) JPH10501668A (fr)
AU (1) AU2759495A (fr)
SE (1) SE502888C2 (fr)
WO (1) WO1995034983A1 (fr)

Families Citing this family (17)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5796819A (en) * 1996-07-24 1998-08-18 Ericsson Inc. Echo canceller for non-linear circuits
AUPO714197A0 (en) * 1997-06-02 1997-06-26 University Of Melbourne, The Multi-strategy array processor
US6430295B1 (en) 1997-07-11 2002-08-06 Telefonaktiebolaget Lm Ericsson (Publ) Methods and apparatus for measuring signal level and delay at multiple sensors
US6023514A (en) * 1997-12-22 2000-02-08 Strandberg; Malcolm W. P. System and method for factoring a merged wave field into independent components
DE19801389A1 (de) 1998-01-16 1999-07-22 Cit Alcatel Verfahren zur Echounterdrückung mit adaptiven FIR-Filtern
US6549627B1 (en) * 1998-01-30 2003-04-15 Telefonaktiebolaget Lm Ericsson Generating calibration signals for an adaptive beamformer
DE19812697A1 (de) 1998-03-23 1999-09-30 Volkswagen Ag Verfahren und Einrichtung zum Betrieb einer Mikrofonanordnung, insbesondere in einem Kraftfahrzeug
EP0974329A3 (fr) * 1998-07-02 2001-09-12 Altura Leiden Holding B.V. Appareil de commande pour installations sanitaires
US6594367B1 (en) * 1999-10-25 2003-07-15 Andrea Electronics Corporation Super directional beamforming design and implementation
DE60010457T2 (de) 2000-09-02 2006-03-02 Nokia Corp. Vorrichtung und Verfahren zur Verarbeitung eines Signales emittiert von einer Zielsignalquelle in einer geräuschvollen Umgebung
US7274794B1 (en) * 2001-08-10 2007-09-25 Sonic Innovations, Inc. Sound processing system including forward filter that exhibits arbitrary directivity and gradient response in single wave sound environment
US20030161485A1 (en) * 2002-02-27 2003-08-28 Shure Incorporated Multiple beam automatic mixing microphone array processing via speech detection
WO2004025989A1 (fr) * 2002-09-13 2004-03-25 Koninklijke Philips Electronics N.V. Calibrage d'un premier et d'un second microphone
US8031881B2 (en) 2007-09-18 2011-10-04 Starkey Laboratories, Inc. Method and apparatus for microphone matching for wearable directional hearing device using wearer's own voice
US8223988B2 (en) * 2008-01-29 2012-07-17 Qualcomm Incorporated Enhanced blind source separation algorithm for highly correlated mixtures
US10244333B2 (en) 2016-06-06 2019-03-26 Starkey Laboratories, Inc. Method and apparatus for improving speech intelligibility in hearing devices using remote microphone
CN106710603B (zh) * 2016-12-23 2019-08-06 云知声（上海）智能科技有限公司 利用线性麦克风阵列的语音识别方法及系统

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4956867A (en) * 1989-04-20 1990-09-11 Massachusetts Institute Of Technology Adaptive beamforming for noise reduction

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See references of WO9534983A1 *

Also Published As

Publication number Publication date
JPH10501668A (ja) 1998-02-10
WO1995034983A1 (fr) 1995-12-21
SE502888C2 (sv) 1996-02-12
AU2759495A (en) 1996-01-05
SE9402088L (sv) 1995-12-15
SE9402088D0 (sv) 1994-06-14

Similar Documents

Publication Publication Date Title
CN100446530C (zh) 校准波束形成器的方法和消除回声的方法
WO1995034983A1 (fr) Agencement de microphone adaptatif et procede d'adaptation a un signal de bruit cible d'arrivee
US8000482B2 (en) Microphone array processing system for noisy multipath environments
EP2277323B1 (fr) Amélioration de l intelligibilité de la parole en utilisant de multiples microphones sur de multiples dispositifs
FI111310B (fi) Laite ja menetelmä epälineaarisia vääristymiä sisältävien akustisten kaikujen poistamiseksi kaiutinpuhelimissa
US5353376A (en) System and method for improved speech acquisition for hands-free voice telecommunication in a noisy environment
CN1169312C (zh) 非线性电路的回波抵消器
KR101210313B1 (ko) 음성 향상을 위해 마이크로폰 사이의 레벨 차이를 활용하는시스템 및 방법
CN1798217B (zh) 限制接收音频的系统
EP2048659B1 (fr) Gain et réglage de forme spectrale dans un traitement de signal audio
CN105825864B (zh) 基于过零率指标的双端说话检测与回声消除方法
EP1879180A1 (fr) Réduction de bruit de fond dans systèmes mains libres
US7206418B2 (en) Noise suppression for a wireless communication device
US7035415B2 (en) Method and device for acoustic echo cancellation combined with adaptive beamforming
US6157909A (en) Process and device for blind equalization of the effects of a transmission channel on a digital speech signal
EP1885154A1 (fr) Déreverbération des signaux d'un microphone
CN107636758A (zh) 声学回声消除系统和方法
WO2002018969A1 (fr) Systeme et procede pour le traitement d'un signal emis a partir d'une source de signaux cible dans un environnement bruyant
CN1331552A (zh) 用于移动终端和其他设备的声音接近检测
US20020138263A1 (en) Methods and apparatus for ambient noise removal in speech recognition
EP1692685A2 (fr) Formeur de faisceaux adaptatif avec robustesse dirigee contre le bruit non correle
Ryan et al. Application of near-field optimum microphone arrays to hands-free mobile telephony
US5905969A (en) Process and system of adaptive filtering by blind equalization of a digital telephone signal and their applications
Neo et al. Robust microphone arrays using subband adaptive filters
JP4409642B2 (ja) 音響獲得の間における外乱信号の最適化された処理のための方法および装置

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

17P Request for examination filed

Effective date: 19970114

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): DE ES FR GB NL SE

RAP1 Party data changed (applicant data changed or rights of an application transferred)

Owner name: NOKIA MOBILE PHONES LTD.

17Q First examination report despatched

Effective date: 20020122

RAP1 Party data changed (applicant data changed or rights of an application transferred)

Owner name: NOKIA CORPORATION

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: THE APPLICATION IS DEEMED TO BE WITHDRAWN

18D Application deemed to be withdrawn

Effective date: 20020802