EP1055317A1 - Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre - Google Patents

Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre

Info

Publication number
EP1055317A1
EP1055317A1 EP99904718A EP99904718A EP1055317A1 EP 1055317 A1 EP1055317 A1 EP 1055317A1 EP 99904718 A EP99904718 A EP 99904718A EP 99904718 A EP99904718 A EP 99904718A EP 1055317 A1 EP1055317 A1 EP 1055317A1
Authority
EP
European Patent Office
Prior art keywords
filter
echo
adaptive
attenuation
level balance
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Withdrawn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
EP99904718A
Other languages
German (de)
English (en)
Inventor
Gerhard Schmidt
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.)
Infineon Technologies AG
Original Assignee
Siemens AG
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 Siemens AG filed Critical Siemens AG
Publication of EP1055317A1 publication Critical patent/EP1055317A1/fr
Withdrawn legal-status Critical Current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

Definitions

  • the present invention relates to a method for improving the acoustic attenuation in hands-free systems with a level balance and a plurality of adaptive echo compensation filters, each of which processes a subband.
  • a further adaptive filter (shadow filter) of a lower order is connected in parallel to the adaptive echo cancellation filter in at least one subband. Room changes can then be detected by combining a performance evaluation of the two residual echo powers and a correlation analysis of the estimated and the measured microphone signal.
  • the further adaptive filter has a significantly lower order.
  • the echo cancellation is preferably implemented in frequency subbands by means of a filter bank.
  • Both performance evaluations of competing are preferably used for the adaptation or the step size control 3 adaptive filters, as well as correlation-based analyzes used.
  • the echo compensation filters provide estimates for the echo attenuation introduced by them, since these estimates can preferably be used to control the attenuation of the level balance. As a result, the attenuation to be introduced by the level balance can be further reduced and the conversation quality in the case of two-way communication can be further improved.
  • Figure 1 is a simplified model of a hands-free device connected to a digital connection.
  • FIG. 2 shows a simplified block diagram of a hands-free device
  • FIG. 4 shows an overview of the method according to the invention with shadow filter and correlation analysis
  • FIG. 5 shows the control of the power transmission factors in a clear representation
  • FIG. 1 shows a simplified model of a hands-free device 10 connected to a digital connection 12.
  • the A-law coding or decoding used in the European ISDN network is shown in the two left blocks 14, 16.
  • the speaker-room microphone system 18 (LRM system) with the local call participant 20, the user of the hands-free device, is sketched on the right-hand side.
  • the acoustic coupling between loudspeaker and microphone leads to crosstalk via the LRM system.
  • This crosstalk is perceived by the distant subscriber as a disturbing echo.
  • Acoustic waves emerge from the loudspeaker and spread out in the room. Reflection on the walls and other objects in the room creates several paths of propagation, which result in different durations of the loudspeaker signal.
  • the echo signal at the microphone thus consists of the superimposition of a large number of echo components and possibly the useful signal n (t): the local speaker.
  • the connection between the participants can also generate echoes at transitions between different transmission systems.
  • the network operators try to take special measures against such echo sources directly at the critical points, so that these echoes can be disregarded here.
  • Fork echoes which arise in telephones with an analog interface due to mismatching of the line simulation to the line impedance, can also be disregarded when using digital connections.
  • FIG. 2 An overview of a hands-free device is shown in FIG. 2.
  • the central element is a level balance 22, which is shown in the left part of FIG. 2.
  • the level balance 22 guarantees the minimum attenuation prescribed by the ITU or ETSI recommendations by adding attenuation to the transmission and / or reception path depending on the conversation situation.
  • the reception path is activated and the signal from the remote subscriber is output undamped on the loudspeaker.
  • the echoes that occur when the compensators are switched off or poorly balanced are greatly reduced by the damping inserted into the transmission path.
  • the local speaker is active, the situation is reversed.
  • the level balance 22 does not insert any attenuation into the transmission path and the signal of the local speaker is transmitted undamped. Controlling the level balance 22 in the case of two-way communication becomes more difficult.
  • both paths and thus also the subscriber signals) each receive half of the damping to be inserted or, if the control is not optimal, at least one of the two signal paths is damped. Intercom is therefore not possible or only possible to a limited extent.
  • adaptive echo cancellers 28 shown in the right part of FIG. 2. These try to digitally emulate the LRM system in order to then remove the echo component of the distant subscriber from the microphone signal. Depending on how well the compensators manage this, the total attenuation to be inserted by the level balance 22 can be reduced.
  • the echo composition was implemented in frequency subbands, the width of the individual bands being between 250 Hz and 500 Hz at 8 kHz sampling rate or between 500 Hz and 1000 Hz at 16 kHz sampling rate.
  • the use of a frequency 6 selective echo cancellation has several advantages. Firstly, by using undersampling and oversampling, the system can be operated as a multirate system, which reduces the signal processing effort. On the other hand, by dividing the sub-band, the "compensation power" can be distributed differently over the individual frequency ranges and thus an effective adaptation of the "compensation power" to speech signals can be achieved. Subband processing also has a decorrelating effect when the overall tape processing is compared with the individual subband systems. For speech signals, this means an increase in the convergence speed of the adaptive filters. In addition to these advantages, the disadvantage of subband processing must not be ignored. Breaking down a signal into individual frequency ranges always results in a runtime. However, since the method is used for video conferences or in GSM mobile phones, such runtimes are permissible
  • the runtime is mainly determined by the image processing component. Since attempts are generally made to output the image and sound of the distant subscriber lip-synchronized to the local subscriber, the running time of the acoustic echoes can increase to several hundred milliseconds. 3 shows the results of a study in which an attempt was made to find out which echo attenuation is necessary depending on the duration of this echo, so that 90, 70 and 50 percent of the respondents were satisfied with the quality of the call.
  • the echo cancellers are controlled in several stages. All power-based control units 32 work autonomously for each compensator, that is to say independently of the remaining frequency ranges. A separate adaptation and control unit 32 is therefore sketched in FIG. 2 for each compensator.
  • the control stage which is based on correlation analyzes of the estimated and the measured microphone signal, is used for intercom detection and is therefore evaluated equally in all frequency ranges. A further level takes into account the accuracy limited by the fixed point arithmetic and controls the adaptation depending on the modulation.
  • the final intercom detection is also carried out separately with its own unit, which is based on both the level balance detectors and the echo cancellers. This unit causes the level balance to reduce the total attenuation to be inserted again (in accordance with ITU recommendation G.167).
  • a second filter 36 with a significantly reduced order - hereinafter referred to as shadow filter 36 - was connected in parallel to one of the echo cancellers 34.
  • This second filter 36 is dimensioned so that it can only compensate for the direct sound. Due to its shortened length and its adapted control, it can adapt much faster than the actual echo cancellation filter 34.
  • the control of the shadow filter 36 is based only on the excitation by the distant call participant. After room changes, the residual error power (signal e ⁇ k r ),
  • the combination of these two detection methods - the shadow filter and the correlation analysis - allow fast and stable adaptation of the echo cancellers even under the difficult conditions in motor vehicles.
  • the control of the residual damping, which is to be inserted by the level balance, can be carried out permissible with the described method. Control of attenuation reduction in intercom is included.
  • the frequency band analysis and synthesis required for subband processing is implemented as a polyphase filter bank.
  • the subband echo cancellers 28 are controlled by their step sizes a Dk r ).
  • the equation for these quantities is:
  • and ⁇ e ⁇ ( (k r ) ⁇ represent smoothed estimates for the signal power of the remote subscriber and for the error power. Both estimates are determined by first-order non-linear recursive amount smoothing.
  • ⁇ ß ⁇ ) ( ⁇ ⁇ ⁇ e ⁇ (k r ) ⁇ ⁇ + ⁇ lm ⁇ e ⁇ (k r ) ⁇ ⁇ ) + (l-ß e (k r )) ⁇ e ⁇ (k r -1 )
  • with ß R if
  • Im ⁇ e ⁇ (* r ) ⁇ ⁇ > ⁇ e ⁇ k r - 1) ß ⁇ K) [2.2: ß F , otherwise 13
  • the time constants ß R and ß F are chosen so that an increase in signal power can be followed faster than a decrease in power.
  • the actual calculation of the step sizes uses a DSP-specific log
  • Subbands are estimated if the condition of the speakerphone allows.
  • the quality of these estimates also determines the quality of the entire hands-free system over the long term. Accordingly, the determination of these variables also involves a significantly higher process effort.
  • a second filter is connected in parallel with the actual adaptive filter in the first subband (frequency range 250 Hz - 750 Hz) (FIG. 6).
  • This so-called shadow filter is significantly shorter than the conventional one and is designed in such a way that it can mainly compensate for direct sound and the first reflections. Due to the reduced order, the shadow filter can adjust much faster, if not as far as the longer echo compensation filter.
  • the shadow filter c ⁇ ⁇ k r is, like the subband echo compensators ⁇ ⁇ k r ), with an NLMS algorithm rl K ) ⁇ u (r) ( )
  • the parameter ⁇ ⁇ is adjustable and should be about 1.
  • the size N sf is also adjustable and should be adjusted to the length of the shadow filter.
  • ⁇ (K) ⁇ determines the detector output, which is generated as follows:
  • the correlation coefficient can assume a value range of p ( 0 r) (k r ) e [0 ... l]. Little one
  • the correlation analysis starts from compensated compensators - the signals y ⁇ r) (k r ) and y ⁇ r ⁇ k r ) then have no running time difference. This does not apply to poorly balanced compensators. In order to enable an analysis here, too, the evaluation is also carried out for a time offset in both directions.
  • the correlation coefficients p r) (k r ) are calculated for different values of n:
  • ⁇ j o
  • the values for /? are preferably taken from an integer interval that contains the value 0.
  • P n (r k r ) is preferably calculated for five values of n. 17 To reduce the effort, the sums of the numerator or denominator can be calculated recursively.
  • the detection criterion can thus be specified as follows:
  • the threshold values should be adapted to the statistical properties of the input signal, in particular to the power density spectrum. If sufficient excitation has been detected, the spatial change detection of the shadow filter is evaluated in a second detection stage. Should the shadow filter change to "strong" room changes
  • the determination equation of the transfer factors is in the case of detected strong changes in space:
  • the first stage here is the correlation analysis already mentioned. Will the condition
  • the second level of intercom detection is evaluated with overall band signals.
  • the sizes ⁇ x (k) ⁇ and ⁇ e (k) ⁇ are according to
  • the calculation of the total band power transmission factor p EK (k) is carried out analogously to the subband transmission factors with several detectors. First, the excitation power of the distant participant is checked - if a threshold is not exceeded here, the old estimate is retained. If sufficient excitation has been detected, the error performance of the shadow filter is evaluated and the p EK (k) estimate with a correspondingly short one when the room changes are detected
  • K cs can be used to react to the variance of the variables entering condition 2.27 - it should be chosen so that two-way talk is not recognized even with slight fluctuations in the signal powers.
  • the detection should only recognize two-way communication when the measured error power exceeds the estimated power by a certain value. In such cases, the 21 Estimation of the power transfer factors (Equation 2.17; carried out very slowly, ie

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
  • Telephone Function (AREA)
  • Interconnected Communication Systems, Intercoms, And Interphones (AREA)

Abstract

L'invention concerne un procédé permettant d'améliorer l'affaiblissement acoustique du signal local dans des appareils main libre, s'utilisant notamment dans des automobiles, avec une balance de niveau (22) et plusieurs filtres de compensation d'écho (34) adaptatifs, dont chacun traite une bande partielle. Dans au moins une bande partielle, un autre filtre adaptatif (filtre d'ombre 36) d'ordre moindre est couplé en parallèle au filtre de compensation d'écho (34) adaptatif. Des modifications spatiales sont identifiées sur la base d'une évaluation combinée d'une analyse de corrélation et d'une comparaison d'erreurs résiduelles des deux filtres (34, 36) adaptatifs simultanés.
EP99904718A 1998-02-13 1999-01-18 Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre Withdrawn EP1055317A1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
DE19805942A DE19805942C1 (de) 1998-02-13 1998-02-13 Verfahren zur Verbesserung der akustischen Rückhördämpfung in Freisprecheinrichtungen
DE19805942 1998-02-13
PCT/DE1999/000094 WO1999041898A1 (fr) 1998-02-13 1999-01-18 Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre

Publications (1)

Publication Number Publication Date
EP1055317A1 true EP1055317A1 (fr) 2000-11-29

Family

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Family Applications (1)

Application Number Title Priority Date Filing Date
EP99904718A Withdrawn EP1055317A1 (fr) 1998-02-13 1999-01-18 Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre

Country Status (5)

Country Link
US (1) US6618481B1 (fr)
EP (1) EP1055317A1 (fr)
JP (1) JP2002503924A (fr)
DE (1) DE19805942C1 (fr)
WO (1) WO1999041898A1 (fr)

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DE10242700B4 (de) * 2002-09-13 2006-08-03 Siemens Audiologische Technik Gmbh Rückkopplungskompensator in einem akustischen Verstärkungssystem, Hörhilfsgerät, Verfahren zur Rückkopplungskompensation und Anwendung des Verfahrens in einem Hörhilfsgerät
DE10245667B4 (de) * 2002-09-30 2004-12-30 Siemens Audiologische Technik Gmbh Rückkopplungkompensator in einem akustischen Verstärkungssystem, Hörhilfsgerät, Verfahren zur Rückkopplungskompensation und Anwendung des Verfahrens in einem Hörhilfsgerät
US6952473B1 (en) * 2002-11-25 2005-10-04 Cisco Technology, Inc. System and method for echo assessment in a communication network
DE10317600B4 (de) * 2003-04-16 2007-10-25 Infineon Technologies Ag Integrierter Transceiverschaltkreis und Kompensationsverfahren im integrierten Transceiverschaltkreis
US8831936B2 (en) * 2008-05-29 2014-09-09 Qualcomm Incorporated Systems, methods, apparatus, and computer program products for speech signal processing using spectral contrast enhancement
US8538749B2 (en) 2008-07-18 2013-09-17 Qualcomm Incorporated Systems, methods, apparatus, and computer program products for enhanced intelligibility
US9202456B2 (en) * 2009-04-23 2015-12-01 Qualcomm Incorporated Systems, methods, apparatus, and computer-readable media for automatic control of active noise cancellation
JP5161838B2 (ja) * 2009-05-18 2013-03-13 株式会社日立製作所 エコーキャンセラおよび音響エコー消去方法
US8441515B2 (en) * 2009-09-17 2013-05-14 Sony Corporation Method and apparatus for minimizing acoustic echo in video conferencing
WO2011133075A1 (fr) * 2010-04-22 2011-10-27 Telefonaktiebolaget L M Ericsson (Publ) Annuleur d'écho et procédé correspondant
US9053697B2 (en) 2010-06-01 2015-06-09 Qualcomm Incorporated Systems, methods, devices, apparatus, and computer program products for audio equalization
JP5744236B2 (ja) * 2011-02-10 2015-07-08 ドルビー ラボラトリーズ ライセンシング コーポレイション 風の検出及び抑圧のためのシステム及び方法
US8600040B2 (en) * 2011-03-14 2013-12-03 Continental Automotive Systems, Inc Apparatus and method for convergence control
US9357080B2 (en) * 2013-06-04 2016-05-31 Broadcom Corporation Spatial quiescence protection for multi-channel acoustic echo cancellation
EP3358857B1 (fr) 2016-11-04 2020-04-15 Dolby Laboratories Licensing Corporation Gestion de système audio intrinsèquement sûr pour salles de conférence

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FR2628918B1 (fr) * 1988-03-15 1990-08-10 France Etat Dispositif annuleur d'echo a filtrage en sous-bandes de frequence
DE4227327A1 (de) * 1992-08-18 1994-02-24 Philips Patentverwaltung Teilbandechokompensator mit Teilbandcodiereinrichtung
CA2174366C (fr) * 1995-04-20 2001-03-27 Shoji Makino Methode utilisant un algorithme de projection pour supprimer des echos repartis en sous-bandes
EP0758830B1 (fr) * 1995-08-14 2004-12-15 Nippon Telegraph And Telephone Corporation Annuleur d'écho acoustique par sous bandes
JP3199155B2 (ja) * 1995-10-18 2001-08-13 日本電信電話株式会社 反響消去装置
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Publication number Publication date
JP2002503924A (ja) 2002-02-05
WO1999041898A1 (fr) 1999-08-19
US6618481B1 (en) 2003-09-09
DE19805942C1 (de) 1999-08-12

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