EP1055318A2 - 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

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Publication number
EP1055318A2
EP1055318A2 EP99907267A EP99907267A EP1055318A2 EP 1055318 A2 EP1055318 A2 EP 1055318A2 EP 99907267 A EP99907267 A EP 99907267A EP 99907267 A EP99907267 A EP 99907267A EP 1055318 A2 EP1055318 A2 EP 1055318A2
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EP
European Patent Office
Prior art keywords
echo
attenuation
filter
frequency
signal
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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.)
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Application number
EP99907267A
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German (de)
English (en)
Inventor
Gerhard Schmidt
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Infineon Technologies AG
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Infineon Technologies AG
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Filing date
Publication date
Application filed by Infineon Technologies AG filed Critical Infineon Technologies AG
Publication of EP1055318A2 publication Critical patent/EP1055318A2/fr
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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 devices with a level balance, a frequency-selective, controllable echo compensation with subband processing and residual error post-filtering.
  • This object is achieved with a method having the features of claim 1.
  • Advantageous refinements of this method are specified in the subclaims.
  • control variable namely the step size vector
  • the step size vector is used both for controlling the frequency-selective echo compensation and for the Control of the additional filter used.
  • sampling rates can preferably be used. This can further reduce the computing effort.
  • the echo cancellation is preferably implemented in frequency subbands by means of a filter bank.
  • both the echo cancellers and the residual error post-filtering provide the estimates for the echo attenuation introduced by them, since these estimates can preferably be used to control the attenuation of the level balance.
  • 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.
  • FIG. 2 is a block diagram of the speakerphone according to the invention.
  • FIG. 3 curves for the attenuation requirements for the hands-free device as a function of the echo time
  • FIG. 4 shows an overview of the method according to the invention
  • FIG. 5 shows the structure of the adaptation of the subband echo compensators
  • FIG. 6 shows a model for the power transmission factors
  • FIG. 7 shows the signals of the distant and the local subscriber on the basis of which the method according to the invention is explained below;
  • FIG. 8 the resulting excitation and the disturbed error in band 1;
  • FIG. 11 the smoothing of the attenuation reduction according to the invention
  • FIG. 12 shows a detailed illustration of the post-filtering of the error signal
  • FIG. 13 the smoothing of the step sizes according to the invention (part A for the same time constants, part B for different time constants);
  • FIG. 14 shows a further example of the signals of the remote and the local subscriber, which are the basis for the processing in the further figures;
  • FIG. 15 the adjustment curve and the damping by the further filter in band 1;
  • FIG. 17 the transfer of the damping values to the level balance
  • 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 loudspeaker-room microphone system 18 (LRM system) with the local subscriber 20, the user of the hands-free device, is sketched on the right-hand side.
  • LRM system loudspeaker-room microphone system
  • the acoustic coupling between loudspeaker and microphone leads to crosstalk via the LRM system.
  • This crosstalk is echoed by the distant subscriber perceived.
  • 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.
  • 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 phones with an analog interface due to mismatching of the line simulation to the line impedance, can be disregarded by using digital connections.
  • the central element is a level balance 22, which is shown in the left part of FIG. 2.
  • two gain controls 24, 26 can be switched on in the transmit and receive path.
  • the level balance 22 guarantees the minimum attenuations prescribed by the ITU or ETSI recommendations by inserting attenuations into 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 adds to the Transmission path no attenuation and the signal of local case- ⁇ Chers is transmitted unattenuated. More difficult is the Steue ⁇ tion of the level discriminator in the duplex case.
  • 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 calculate the echo component of the distant subscriber from the microphone signal. Depending on how well the compensators manage this, the total attenuation to be introduced by the level balance can be reduced.
  • the echo composition was implemented in frequency subbands, the width of the individual bands preferably 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 frequency selective echo cancellation has several advantages. On the one hand, by using undersampling and oversampling, the system can be operated as a multirate system, which reduces the calculation 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.
  • the runtime is mainly determined by the image processing component. Since attempts are generally made to output the image and sound of the remote 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 conversation.
  • a pure audio runtime of 30 - 40 ms only requires 35 dB echo attenuation.
  • the requirement increases to 53 dB.
  • the runtime can also be more than 100 ms in GSM connections. The requirements placed on echo cancellation methods in video conferencing and GSM systems are thus higher than the requirements placed on conventional hands-free telephones.
  • a so-called post filter 30 was introduced. This evaluates the step sizes of the individual subbands together with the other detector results and filters the synthesis filter output signal again in a frequency-selective manner. Since the setting algorithm of filter 30 was designed in accordance with a Wiener approach, this post-filtering is also referred to below as Wiener filtering.
  • 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 loudspeaker and microphone signals, 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 in intercom situations to reduce the total attenuation to be inserted again (in accordance with ITU recommendation G.167).
  • the central element here is the calculation of the step size vector c (k). This is used both to control the subband echo cancellers and to calculate the coefficients of the post filter.
  • the two sub-methods each calculate the echo attenuation caused by them and communicate this information to the level balance 22.
  • the scale 22 then reduces the total attenuation set by the user and only inserts the remaining attenuation into the transmission or reception path.
  • the frequency band analysis and synthesis required for subband processing is implemented as a polyphase filter bank.
  • a step size control is described, which ensures a fast and stable adaptation of the subband echo cancellers.
  • methods are presented that estimate the echo attenuation achieved.
  • the level balance 22 can thus reduce the total attenuation based on these estimated values. For the attenuation estimate, it is irrelevant whether the attenuation of well-balanced echo cancellers is achieved by the acoustic arrangement of the loudspeaker and microphone or by an appropriate choice of the analog amplifications.
  • the adaptation of the subband echo cancellers is carried out by means of an NLMS method adapted to the signal processor used.
  • NLMS method adapted to the signal processor used.
  • the index ⁇ should show the subband number.
  • the adaptation error e (r) (k r ) is calculated by forming the difference between the estimated and the measured microphone signal:
  • This error consists of a so-called undisturbed error and the portion caused by the local speaker together:
  • the adaptation is carried out using an approximation of the NLMS algorithm
  • the coefficients of the subband echo cancellers are continuously applied to the subband impulse responses of the LRM system during the operation of the hands-free device using the adaptation methods. fit. A reduction in acoustic echoes can thus be achieved even after system changes.
  • the setting criterion for the adaptation method used is the minimization of the mean square error. According to the calculation specification of the NLMS algorithm, the coefficients undergo a strong change if the samples of the compensated signal e (r) (k r ) of the ⁇ th subband are large. Constantly large values e r) (k r ) can be attributed to two causes:
  • the adaptive filters are poorly adapted to the room impulse response. There is then no or only a slight reduction in the acoustic echoes - the uncompensated echo components cause the signals e ⁇ r) (k r ) to increase. • In such situations, the compensators should be adjusted as quickly as possible.
  • n (k) for example when the local speaker is active - also causes the signals e (r) (k r ) to increase.
  • This component is the useful signal to be transmitted for the hands-free device and for the adaptive device
  • the filter represents a malfunction that can lead to an incorrect setting of the coefficients. In such situations, the compensators should not be adjusted, or only slightly, so that the adjustment already achieved is not deteriorated again.
  • a step size control has already been presented which takes into account the two described conversation situations or states of the compensators and fulfills the demands placed on the adaptation control.
  • the step size in the -th subband should be according to
  • the disturbed error signal e (r) (k r ) in the denominator of equation 3.5 can be measured directly - the expected value of this can be determined by
  • a power transfer factor p ⁇ r) (k r ) is introduced to estimate the meter.
  • the parallel connection is switched off
  • Modeled LRM system and echo canceller including the subtraction point in a first approximation as a simple attenuator.
  • Equation 3.8 was made from for this reason the amount K ES , FT introduced. This amount is to the times in which the handsfree in to ⁇ stand Single the remote subscriber is located, beinhal ⁇ th.
  • the power transfer factor estimate should not be updated - the most recently calculated p ⁇ r) (k r ) are retained. This measure means that changes in space cannot be detected when the local speaker is active. In such cases, the power transmission factors are only adjusted after the individual subscriber status has been reached again.
  • the determination equation for the smoothed power transmission factors can thus according to
  • Nonlinear, recursive smoothing was used for the first subproblem.
  • the sum of the amount of the real part and the amount of the imaginary part of the subband signals was selected as the input signals of these filters.
  • the performance factors were calculated logarithmically - the division can therefore be replaced by a subtraction.
  • a so-called correlation measure ⁇ (k r ) was used for the second sub-problem.
  • a standardized cross-correlation analysis of the excitation signal of the distant subscriber and the microphone signal is carried out.
  • the distant subscriber speaks individually the two signals are strongly correlated and the correlation measure gives values ⁇ () «1.
  • the correlation is reduced and values ⁇ (kr) ⁇ 1 are detected.
  • the microphone signal is formed by convolution of the excitation signal with the impulse response already presented in an office room (length 2044 coefficients at 8 kHz sampling rate) and subsequent addition of the signal from the local speaker.
  • the correction factor can be dispensed with by subsequently dividing the two quantities.
  • the amount calculations were made by the more cost-effective estimates , (') (k r )
  • the power transmission factors are only determined logarithmically - the division is thus reduced to two logarithms and one subtraction.
  • the power transmission factors are thus according to
  • the time constant ß p was also chosen differently for rising and falling edges. This is intended to do justice to the non-compensable part of the system runtime (artificial delay of the microphone signal). Due to this runtime, the signal power of the excitation signal drops earlier than that of the error signal - without correcting this process, the estimate would lower the estimated value after each excitation phase. In addition, the time constants are increased when two-way communication is detected. The two-way detector used is described below.
  • the equation for the time constant ß p is: GK, GS
  • K GS is used to denote the times at which the detector described above detects intercom.
  • the set K ES , FT denotes the points in time at which the correlation measure recognizes individual speech by the distant subscriber.
  • the step size a (k r) i-n can each band from the previously calculated sizes, according to
  • the linearization is designated with LIN ⁇ ... ⁇ . If the excitation power is a limit falls below, it is assumed that the excitation consists only of background noise and the adaptation is stopped.
  • the step size in the first subband is shown logarithmically in FIG.
  • the step size is approximately 1 - in phases of individual speaking by the local subscriber ( ⁇ i and B 2 ), a difference from disturbed to undisturbed error performance of approximately 26 to 30 dB can be determined from FIG. 8 become.
  • the step size is therefore also in the expected range (approx. -27 dB) in the intercom phases.
  • the desired detector should be able to decide between single-talk and two-way talk independently of room changes and also independently of the power of the input signals.
  • a correlation measure is used - a detector that meets the above requirements. The cross correlation between the loudspeaker signal and the microphone signal is evaluated in a standardized form.
  • a release is set when the maximum of the determined correlation measures is greater than a limit value ⁇ 0 .
  • the limitfrag 0 is determined by a finite sum of non-positive powers of two
  • N tn 0 with a n e ⁇ 0, l ⁇ (3.26) approximated.
  • the threshold value comparison can then be traced back to a summation of right-shifted denominator values and a comparison:
  • N 0
  • the evaluations were only carried out in the most powerful, first subband and there only with the real parts of the complex signals. In this band, the greatest signal-to-noise ratio can be expected for voice excitation, which should improve the reliability of the detector results. As a result of this measure, the subsampling will only carry out the calculations every r sampling cycles. The time k r is then included in the set K ES , F ⁇ if one of the L 2 comparisons yields a correlation measure greater than ⁇ 0 .
  • the echo attenuation to be provided by the hands-free device can be reduced by 15 dB in intercom situations.
  • an intercom detector has been developed according to the following considerations. At the same time, this detector can be used to "more carefully” set the estimates in the step size control when two-way communication occurs.
  • the detection of intercom is carried out in two steps.
  • a first stage it is checked whether the distant speaker is active.
  • the excitation signal of the distant subscriber smoothed in magnitude, with a threshold
  • the second comparison is always necessary if the level scale brings in large attenuation values (e.g. after changes in room). In such situations, the reception path can be severely damped. Here the comparison with the smoothed input signal would not provide a reliable result.
  • the amount-smoothed excitation signal is calculated analogously to the recursive, non-linear smoothing described in the step size control. It should be noted here, however, that the higher sampling rate means that larger time constants must be used and limit cycles can occur as a result. A double-word precision calculation (32 bit) is therefore required:
  • the time constant ß xg is chosen as follows
  • the time constant ß eg is chosen as follows:
  • a (total band) power transmission factor p EK (k) is determined to estimate the undisturbed error power:
  • this variable is also smoothed recursively. Since the determination of the transmission factor only consists of smoothed quantities, it is only carried out under-sampled:
  • the attenuation requirement is reduced using a low-pass filter.
  • the time constant for the rising edge ß Gsr should be as small as possible so as not to cut off the beginning of a speech passage .
  • the time constant for the falling flank ß Gsf should be greater than the arrival be selected rose constant, thus lowering the damping Pos e r) i n short speech pauses is not completely withdrawn. This relationship is shown in FIG.
  • the smoothed damping reduction is determined as follows:
  • the time k r is included in the quantity K gs if the damping reduction is above a predetermined value.
  • An exemplary course of the damping reduction is shown in FIG. 11.
  • the total attenuation of the level balance which is prescribed by ITU-T recommendation G. 167, can be reduced by the attenuation of the overall system consisting of room and echo canceller. Even when echo compensation is switched off, the control described above estimates the transmission factor of the acoustic path from the loudspeaker to the microphone, including the analog amplifications. In this way it is possible to react to different loudspeaker or different (analog) microphone amplifications and to adjust the total attenuation (digital) according to the required values.
  • the total attenuation can also be set to a lower value in accordance with ITUT recommendation G. 167.
  • ITUT recommendation G. 167 a detector and a corresponding transfer size were presented or defined.
  • the total level balance damping D PW ⁇ k) is thus controlled (initially without taking post-filtering into account) using the following procedure:
  • D 0 is the required maximum attenuation (eg 45 dB).
  • the attenuation of the echo canceller D EK (k) is determined by the form of calculation
  • the real-time implementation of the echo cancellation method shows that the adaptive filters can never completely calculate the portion of the distant speaker from the microphone signal. This can have many different causes, three of which are listed here as examples:
  • the error signal e (k) thus contains, in addition to the portion of the local speaker n (k), also the uncompensated portion of the distant speaker, which was already referred to in the previous parts of this description as an 'undisturbed' error ⁇ (k).
  • the signal n (k) is the useful component of the signal e (k) - the signal ⁇ (k) is the disturbance from this point of view.
  • the following shows how post-filtering of the signal e (k) - to dampen the "interference" ⁇ (k) - based on a Wiener filter approach with the step size control for the Subband echo cancellers can be linked.
  • a transversal filter of order M - 1 is inserted after the synthesis filtering.
  • the parameter M is also the number of bands in the filter bank.
  • the coefficients are determined in the subband level and transformed into the time domain with an inverse DFT.
  • the coefficient determination is affected by several smoothings with an inertia and thus a running time. This runtime can be at least partially compensated for by the maximum-phase synthesis filter that lies between the determination and use of coefficients.
  • the post-filtering takes place in the time domain and frequency-selective.
  • the filter g (k) 30 is placed behind the synthesis.
  • the order of the filter is M - 1, so M coefficients must be set.
  • the filter 30 should optimally free the "disturbed” signal e (k) from the "disturbance” ⁇ (k).
  • the filter frequency response can be too
  • the filter g (k) has the order M - 1 and is to be determined from the frequency response G opt ( ⁇ ) by inverse Fourier transformation, M nodes of the frequency response must be determined. For the frequencies
  • the frequencies ⁇ ⁇ represent, in addition to the support points in the frequency range, also the band centers of the bandpasses described above when dividing the subband.
  • G opt ( ⁇ ⁇ ) can by
  • the estimated support points of the filter frequency response are smoothed over time, and are provided with a so-called overestimation factor ⁇ and a maximum attenuation G min (k).
  • the temporal smoothing is applied to the step sizes and is carried out with a first-order IIR filter with two different time constants for rising ( ⁇ r ) and falling ( ⁇ f ) edges:
  • the vector a ⁇ r) (k) used in the implementation is thus composed of the smoothed step sizes:
  • the filter frequency response is then according to
  • the overestimation factor ß accelerates the introduction of the damping and increases the damping.
  • a value between 1.0 and 3.0 is preferably chosen for ⁇ .
  • G m i n (k) of the "influence" of the Wiener filter can be controlled. In real-time tests showed that it is advisable to link the control of this parameter with the collated status of the echo cancellers.
  • the attenuation achieved by the echo cancellers is still very low.
  • the Wiener filter should intervene strongly and be able to introduce large attenuations (eg up to 45 dB according to the ITU recommendations). Is in the room in which the hands-free system is located If there is strong background noise, the echo is suppressed by the Wiener filter, but the distant participant then perceives a kind of modulation of the background noise. During the pauses in the speech, the noise is transmitted undamped while he is speaking, it experiences a (e.g. B. 45 dB) attenuation.
  • the step size control provides a suitable control variable - the estimated power transmission factor D EK (k).
  • the parameter G mln (k) is therefore set according to:
  • LIN denotes the linearization of logarithmic variables already used in the step size control.
  • the maximum insertion loss (for example 45 dB) can be set with the parameter G maXrlog .
  • This fixed value is then reduced by the attenuation D EK (k), which the echo cancellers provide on average, and the intercom reduction D GS (k) reduced.
  • the sizes D EK (k) and D GS (k) lie in the same logarithmic ⁇ mix form as the constant G max / log before. Limiting the calculated size to 0 dB serves to adapt to the linearization.
  • the attenuation D w (k) of the signal e (k) by the Wiener filter is communicated analogously to the attenuation of the echo cancellers and the attenuation reduction in the case of two-way communication via an interface of the level balance. Attenuation is approximated by the mean over all frequency ranges to be transmitted:
  • D ' ⁇ w (k) ß, ./ D (k ⁇ ) + ( ⁇ -ß rf ) D (k).
  • the use of different time constants for rising and falling edges causes the estimate to be "more careful”. If attenuation is added by the Wiener filter, the level balance reduces its attenuation more slowly. For a short time, the error signal thus exceeds the required 45 dB Conversely, if the Wiener filter reduces its attenuation, the level balance very quickly adds the remaining attenuation, and the delay due to the synthesis filtering can also result in a brief total attenuation of more than the set upper limit (eg 45 dB) .
  • the set upper limit eg 45 dB
  • the maximum attenuation G maX ⁇ og was chosen to be 60 dB.
  • the initial adjustment process of the compensators takes place in area A ⁇ .
  • the compensators have not yet been adjusted - in the end, the final adjustment status was reached in all bands. Since there is no intercom in this phase, the Wiener filter should insert the difference between 60 dB and the attenuation achieved by the echo canceller. The coefficient for this is in area Ai
  • G 'k) r ⁇ (l- ßa ⁇ r) (k)), G nl (*) ⁇ in sub-band 1 (250 -750 Hz at 8 kHz sampling rate) together with the excitation and error signal before the Wiener filter in FIG. 15.
  • the damping is not inserted immediately - this effect is partially compensated for by the transformation into the time domain and the synthesis filter in between. At least 25 dB of attenuation is thus already inserted in the overall band signal (see FIG. 18) at the beginning of the activity of the distant speaker.
  • the attenuation After about 200 ms, the attenuation has already increased to its final value of 60 dB. With increasing compensation of the compensator, the attenuation by the Wiener filter in band 1 decreases and, as expected, reaches a final value of about 30 dB (60 dB maximum limit - 30 dB echo attenuation by the compensator). Since the Wiener filter was only inserted after the synthesis, the courses of the excitation, the error, the step size and the power transmission factor in band 1 can be seen from FIGS. 9 and 10.
  • the maximum limit of the damping to be inserted G min (k) is the determining variable.
  • the total signal e (k) should be separated from its interference ⁇ (k).
  • the local participant - the useful signal in e (k) - is not active, the overall signal only consists of the disturbance.
  • the initial value of about 60 dB is determined by the set maximum attenuation G maXr ⁇ og .
  • this upper limit is reduced again by the intercom detector by 15 dB to about 15 dB.
  • the performance of the local speaker is significantly higher than that of the residual echo, this limit is not reached.
  • the determining factor in the intercom phase is the power ratio of the signal from the local speaker and the residual echo from the distant speaker. The performance of the residual echo depends on the one hand on the excitation power of the distant participant and on the other hand on the balancing state of the compensators. The better these are balanced, the less the influence of the Wiener filter will be in these passages.
  • an upper limit of the attenuation was determined in accordance with equation 4.1.
  • This upper limit was determined as a function of the attenuation already achieved, which is given by the power transmission factors in the respective band or by the intercom attenuation. Both quantities were only calculated and saved in logarithmic representation in the step size calculation. In order to be able to use the variables in the limitation function, eight linearizations are necessary. The determination of the maximum values would therefore require more computing power than the entire remaining coefficient calculation. For this reason, a uniform upper limit has been introduced for all tapes. This is also calculated according to equation 4.1, but with the total band sizes.
  • the resource requirements of the post-filtering obtained in this way are well below 1 MIPS when using 16-bit fixed-point signal processors.
  • the Wiener filter 30 When the Wiener filter 30 is switched on, the total attenuation can additionally be weakened by the attenuation of the Wiener filter 30. The maximum stroke of the level balance can thus be
  • the size D w (k) is according to

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

Abstract

L'invention concerne un procédé permettant d'améliorer l'affaiblissement acoustique du signal local, sur la base de la combinaison d'une commande d'adaptation pour processus de compensation de l'écho de bande partielle et d'un post-filtrage de bande globale afin de supprimer l'écho résiduel dans des appareils main libre, avec une balance de niveau (22) et une compensation d'écho (28) avec traitement de bande partielle, qui peut être commandée et être sélective en fréquence. Après la compensation d'écho (28) sélective en fréquence, le signal délivré est soumis à un post-filtrage dans un autre filtre (30) sélectif en fréquence, avec un algorithme de réglage selon une équation de Wiener (filtrage de Wiener). Une seule valeur d'influence (vecteur des incréments) sert à la fois à assurer la commande de la compensation d'écho sélective en fréquence et celle de l'autre filtre. Ce procédé peut par conséquent être mis en oeuvre avec un volume très limité de calculs, de manière à pouvoir également être utilisé dans des processeurs grand-public simples.
EP99907267A 1998-02-13 1999-01-21 Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre Withdrawn EP1055318A2 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
DE19806015 1998-02-13
DE19806015A DE19806015C2 (de) 1998-02-13 1998-02-13 Verfahren zur Verbesserung der akustischen Rückhördämpfung in Freisprecheinrichtungen
PCT/DE1999/000151 WO1999041897A2 (fr) 1998-02-13 1999-01-21 Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre

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EP1055318A2 true EP1055318A2 (fr) 2000-11-29

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EP99907267A Withdrawn EP1055318A2 (fr) 1998-02-13 1999-01-21 Procede pour ameliorer l'affaiblissement acoustique du signal local dans des appareils main libre

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US (1) US6834108B1 (fr)
EP (1) EP1055318A2 (fr)
JP (1) JP2002503923A (fr)
DE (1) DE19806015C2 (fr)
WO (1) WO1999041897A2 (fr)

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DE19806015A1 (de) 1999-08-26
WO1999041897A2 (fr) 1999-08-19
DE19806015C2 (de) 1999-12-23
WO1999041897A3 (fr) 1999-09-23
JP2002503923A (ja) 2002-02-05

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