EP0710947B1 - Verfahren und Apparat zur Geräuschunterdrückung in einem Sprachsignal und korrespondierendes System mit Echounterdrückung - Google Patents

Verfahren und Apparat zur Geräuschunterdrückung in einem Sprachsignal und korrespondierendes System mit Echounterdrückung Download PDF

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EP0710947B1
EP0710947B1 EP95402385A EP95402385A EP0710947B1 EP 0710947 B1 EP0710947 B1 EP 0710947B1 EP 95402385 A EP95402385 A EP 95402385A EP 95402385 A EP95402385 A EP 95402385A EP 0710947 B1 EP0710947 B1 EP 0710947B1
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signal
domain
voice signal
frequency
background noise
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French (fr)
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EP0710947A1 (de
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Ivan Bourmeyster
Frédéric Lejay
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Alcatel Lucent SAS
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Alcatel CIT SA
Alcatel SA
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    • 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
    • Y—GENERAL TAGGING OF NEW TECHNOLOGICAL DEVELOPMENTS; GENERAL TAGGING OF CROSS-SECTIONAL TECHNOLOGIES SPANNING OVER SEVERAL SECTIONS OF THE IPC; TECHNICAL SUBJECTS COVERED BY FORMER USPC CROSS-REFERENCE ART COLLECTIONS [XRACs] AND DIGESTS
    • Y10—TECHNICAL SUBJECTS COVERED BY FORMER USPC
    • Y10S—TECHNICAL SUBJECTS COVERED BY FORMER USPC CROSS-REFERENCE ART COLLECTIONS [XRACs] AND DIGESTS
    • Y10S367/00—Communications, electrical: acoustic wave systems and devices
    • Y10S367/901—Noise or unwanted signal reduction in nonseismic receiving system

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  • the present invention relates generally to method and device for suppressing a noise signal in a speech signal, typically for an application to hands-free radiotelephony. It also relates to a system implementing such a device in combination with an echo canceller.
  • the electrical signal resulting from an acousto-electric conversion of a speech is mixed with a noise signal. Since the level of the noise signal in this environment is high, for example in the cockpit of a vehicle, it is necessary to implement treatments aimed at remove the noise signal in the electrical signal from word.
  • two types of suppression treatment noise are distinguished according to the prior art: the Spectral subtraction treatment and the so-called treatment by filter bank.
  • the treatment by filter bank consists of a step of separating the input signal into a plurality of time signals representative each of a band of respective predetermined frequencies, an estimation step a signal-to-noise ratio for each of these signals temporal, a step of weighting these temporal signals by respective coefficients which are each a function of the respective one of the signal-to-noise ratios for the signal considered, and a step of adding these signals weighted temporals into a resulting signal consisting of a speech signal in which the noise signal is deleted.
  • each of the signal-to-noise ratios is estimated according to the variation of the power of the time signal considered in its frequency band respectively.
  • Spectral subtraction processing uses its share in the frequency domain, typically by use FFT Fast Fourier Transform (Fast Fourier Transform) in Anglo-Saxon terminology). It has the disadvantage principle of inducing non-linear distortion in the processed speech signal that results from the loss phase information of this signal. Indeed this treatment by spectral subtraction produces such a distortion because it applies to samples resulting from the Transform of Fast Fourier of noisy speech signal to deal with squared module functions that remove information from phase, thus making this treatment non-linear.
  • the lack of linearity of the treatment by subtraction spectral impedes its effective use in combination with echo cancellation processing, such proposed by the invention because the cancellation device echo is disturbed in its operation by this loss phase information.
  • a first objective of the present invention is to provide a method of suppressing noise in a signal of word that has the advantage of significantly reducing the computing power required in terms of Instructions per Second, compared to a treatment by filter bank.
  • a second object of the invention is to provide a not inducing nonlinear distortion in the speech signal to be treated, in contrast to the treatment by spectral subtraction.
  • Another object of the invention is to provide a system comprising a noise canceling device implementing the process steps, in combination with an echo cancellation device.
  • the calculation step is preceded, for each of the K groups of frequency components, by a step of selecting some of the components frequencies having respective predetermined ranks in said each of the groups, said selected portion symmetrical with respect to the complementary of this part among the plurality of extracted frequency components.
  • production steps and synthesis are implemented respectively by means of Fast Fourier Transform and Inverse Fourier Transform.
  • the invention also provides two variants of combined system of echo cancellation and suppression of noise.
  • the frequency processing unit 100 includes in cascade a circuit of extraction of energy components 10, an estimation circuit of signal-to-noise ratio 11, a gain calculation circuit 12 and a filter coefficient synthesis circuit 13.
  • the temporal processing circuit 14 is a time filter, of finite impulse response filter type (FIR Filter in English terminology for Finite Impulse Response Filter).
  • the sampled noisy speech signal s (nT) resulting from this sampling is applied, on the one hand, to an input of the energy component extraction circuit 10 in the frequency processing unit 100, and, on the other hand, to an input of the time filter 14.
  • Figure 2 shows schematically the processing performed in the circuit 10 which receives the noisy speech signal s (nT).
  • This sampled noisy speech signal s (nT) is in the form of successive frames of samples, four of which are T (n-2), T (n-1), T (n) and T (n). +1) being represented in a first line of FIG. 2.
  • K 3 blocks of samples B (1), B (2) and B (3) , K being an integer.
  • K 3 blocks of samples are formed according to the embodiment described from, on the one hand, this frame of rank n, T (n), and, on the other hand, 2 frames T (n-2). and T (n-1), of respective ranks (n-2) and (n-1).
  • the step noted 101 in this Figure 2 is intended to simplify the implementation of subsequent processing by selecting only a part of the frequency components in each group E (1) i to E (3) i , 0 ⁇ i ⁇ 255.
  • This step is based on the following property.
  • the Fast Fourier Transform of a real signal has a pseudo-symmetry.
  • These first 129 selected frequency components are sufficient to describe each group E (k) i , 0 ⁇ i ⁇ 255, in a complete way since the other frequency components in the group, namely the last 127 components E (k) 129 to E ( k) 255 are deduced by symmetry.
  • the frequency components E (k) O to E (k) 128 which are selected in each group have indeed a character of symmetry with respect to the complement E (k) 129 to E (k) 255 of these selected components among all the components frequencies in the group that are initially produced.
  • the frequency components E (k) O to E (k) 128 are produced for each group.
  • the 129 frequency components selected in each group are decimated by 2, in order to retain only one out of two of each selected group of components. This decimation by two 102 aims to selectively remove one component out of two, relative to a given frequency, with a view to inhibiting the interaction effect that produces on this component each of the two frequency components located at two respective frequencies of each other.
  • the device 10 extracts for a cycle relating to a noisy speech signal processing frame T (n) s, (nT) 65 energy components Em j , each representative of an energy, or power, of the noisy speech signal. (nT) for the frequency, or frequency band, considered.
  • all the steps 100, 101 and 102 described with reference to FIG. 2, although improving the implementation of the method according to the invention, can be reduced to a single step consisting of the application of a single FFT Fourier transform at M 128 samples of the T (n) frame selected for the considered treatment cycle.
  • the selection step 101 may either be operated or not be operated directly on the frequency components resulting from the FFT treatment (s).
  • the 65 energy components Em j are applied to an input of the signal-to-noise ratio estimation circuit 11.
  • the circuit It estimates a signal-to-noise ratio SNRj between the noisy speech signal s (nT) and a noise signal included in this noisy speech signal, for the energy component considered Em j .
  • SNR j not Em j not / B j not in which the exponent n identifies the processing cycle relative to the frame T (n) of rank n and B j denotes a noise energy component in the energy component Em j of rank j.
  • such an estimate of the signal-to-noise ratio is based on the calculation of the estimated noise energy component in each given energy component. For example, it uses the ratio between the energy component Em j n extracted and the noise energy component B j n-1 calculated previously during a treatment cycle that preceded the considered cycle of noise signal suppression processing in the process. frame T (n). The higher this ratio, the more it indicates the existence of a speech signal for the frequency energy component Em j n considered, in which case the noise component B j (n-1) calculated with respect to the energy component Em j ( n-1) for the rank (n-1) is maintained in the noise component B j n .
  • Circuit 11 assigns a signal to noise ratio SNR j , 0 ⁇ j ⁇ 64, to each extracted energy component Emj, 0 ⁇ j ⁇ 64, according to an estimation algorithm using such a principle.
  • SNR j As a function of these 65 signal-to-noise ratios SNR j , the circuit 12 calculates for each of them a gain G j , taking for example a value comprised substantially between 0 and 1, which is directly related to the signal-to-noise ratio SNR j for the corresponding frequency component.
  • the noise signal is weak, plus the gain G j is high.
  • the noise signal component is attenuated for each frequency energy component Em j .
  • These gains G j are such that the weighting of the energy components Em j respectively by these gains would give a discrete spectrum of weighted frequency energy components, which would be representative of the noisy speech signal s (nT) in which the noise signal is substantially suppressed.
  • This circuit 13 comprises a first circuit (not shown) for duplicating the 65 calculated gains G i , in accordance with the equation given in (1).
  • These coefficients denoted C (nT), 128 in number, are applied to a first control input of the filter 14, typically FIR filter.
  • a second input of the filter 14 receives the noisy speech signal s (nT).
  • the filter 14 convolves the coefficients C (nT) with the 128 samples of the frame T (n), in a field denoise of 128 samples forming part of the speech signal noised s * (nT).
  • the method resulting from the device described above is naturally "adaptive" in the sense that the coefficients C (nT) applied to the control input of the FIR filter 14 are modified for each frame T (n), of rank n given, depending on the processing 10, 11, 12 and 13 of the samples forming the speech signal to be processed.
  • the characteristic of the noise suppression process according to the invention is to use, on the one hand, a treatment digital frequency 100 of the noisy speech signal, for produce filtering time coefficients C (nT), and, on the other hand, a digital time processing 14 of the signal of noisy speech s (nT) according to the coefficients of filtering C (nT), to produce a speech signal s * (nT) wherein the noise signal is substantially suppressed.
  • a combined system of noise suppression and echo cancellation which is included in a terminal, typically a hands-free radiotelephone, includes a microphone 2, a loudspeaker 4, a noise suppression according to the invention 1, as described previously, a temporal processing circuit 14 'and a echo canceller 3.
  • the noise suppressor 1 is identical to the device shown in FIG. 1, and mainly includes a frequency processing unit 100 and a temporal processing circuit 14.
  • the canceller echo is formed of a subtractor 30 and a circuit 31 which produces an estimated echo signal.
  • the microphone 2 receives a speech signal to be transmitted [s (t) + e (t)] formed of a signal noisy sound word s (t) added to an echo signal and).
  • This echo signal results from the acoustic coupling between the speaker 4 and microphone 2.
  • the device noise suppression 1 treats, as described previously, the speech signal to be transmitted in a speech signal transmitted denoised [s * (nT) + e * (nT)] which is applied to a first input of the subtractor 30 including a second input receives the output of circuit 31.
  • a received speech signal r (t) from a remote terminal is applied, from a to a loudspeaker input, and, on the other hand, to a loudspeaker circuit 31 input through the processing circuit temporal 14 'preceded by a sampler 14a'.
  • the circuit of temporal processing 14 ' is at each moment strictly similar to the temporal processing circuit 14 in the noise suppression device 1 (FIG. 1).
  • This characteristic is based on the fact that the estimated echo of received signal r (t) produced by the circuit 31 is to be subtracted, in the subtractor 30, to the echo signal processed by noise suppression e * (nT) in circuit 1 and not at original echo signal e (nT).
  • This circuit 14 'is therefore a pure and simple duplication of the temporal processing circuit 14 in device 1, as indicated by the arrow in broken line at both ends in Figure 3.
  • the time processing circuit 14 'is therefore associated with each moment to the same 128 filter coefficients C (nT) as the circuit 14 in the device 1.
  • a combined system of noise suppression and echo cancellation comprises a microphone 2, a loudspeaker 4, an echo canceller 3, a frequency processing 100, a processing circuit temporal 14 and a sampler 5.
  • the units 100 and circuit 14 are identical to those described in FIG.
  • the echo canceller 3 is formed of a subtractor 30 and a circuit 31 which produces an estimated echo signal ê (nT).
  • the microphone 2 receives a transmitted speech signal [s (t) + e (t)] formed of a noisy speech signal s (t) added to an echo signal e (t). This echo signal results from the coupling between the loudspeaker 4 and the microphone 2.
  • the transmitted speech signal [s (t) + e (t)] is sampled in the sampler 5 in the signal [s (nT) + e (nT)].
  • This signal sampled is applied, on the one hand, to an input of unit 100, and, on the other hand, to an input of the circuit 14 to through the subtractor 30.
  • a speech signal received in from a remote terminal r (t) is applied, from a part of an entrance to circuit 31 and, on the other hand, to a speaker input 4.
  • Circuit 31, receiving the signal r (t) produces in response an estimated echo signal ê (nT) applied to a first input of the subtractor 30, one of which second input receives the transmitted speech signal [S (nT) + e (nT)].
  • the treatments frequencies implemented in unit 100 operate on the transmitted speech signal [s (nT) + e (nT)], and secondly the temporal processing in the circuit 14, from the coefficients C (nT) produced by unit 100, operates on the difference signal, or transmitted speech signal processed by echo cancellation, [s (nT) + e (nT) -e (nT)].
  • This variant avoids "duplicating" circuit 14 in the branch including the circuit 31, as represented by the line arrow discontinuous in Figure 3 for the previous variant.

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  • Engineering & Computer Science (AREA)
  • Human Computer Interaction (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Computational Linguistics (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
  • Interconnected Communication Systems, Intercoms, And Interphones (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Filters That Use Time-Delay Elements (AREA)

Claims (7)

  1. Verfahren zum Unterdrücken eines Rauschsignals in einem abgetasteten verrauschten Sprachsignal (s(nT)), mit den Schritten:
    digitale Verarbeitung (100) des verrauschten Sprachsignals im Frequenzbereich zum Erzeugen von zeitlichen Filterkoeffizienten (C(nT)), mit den Unterschritten:
    Extraktion einer Mehrzahl von Frequenzenergiekomponenten (Emj) in dem verrauschten Sprachsignal (s(nT)),
    Abschätzen (11), für jede der extrahierten Frequenzenergiekomponenten, eines Verhältnisses (SNRj) zwischen einem Energieniveau des verrauschten Sprachsignals (s(nT)) und einem Energieniveau des Rauschsignals,
    Bestimmen (12) einer jeweiligen Verstärkung (Gj) für jede der extrahierten Frequenzenergiekomponenten (Emj) in Abhängigkeit von dem abgeschätzen Verhältnis (SNRj) zwischen dem Energieniveau des verrauschten Sprachsignals (s(nT)) und dem Energieniveau des Rauschsignals für jede jeweils ausgewählte Frequenzkomponenten, und
    Synthese (13) der Filterkoeffizienten (C(nT)) in Abhängigkeit von den Verstärkungen (Gj) ;
    digitale Verarbeitung (14) des verrauschten Sprachsignals (s(nT)) im Zeitbereich in Abhängigkeit von den Filterkoeffizienten (C(nT)) zu einem Sprachsignal (s*(nT)), bei dem das Rauschsignal im wesentlichen unterdrückt ist.
  2. Verfahren nach dem vorhergenden Anspruch, bei dem der Schritt der Extraktion der Frequenzenergiekomponenten die folgenden Unterschritte umfasst:
    Erzeugen (100a, 100b, 100c) von K Gruppen mit jeweils einer Mehrzahl von Frequenzkomponenten (E(1)i, E(2)i, E(3)i für jeweils K verschachtelte Blöcke (B(1), B(2), B(3)) des verrauschten Sprachsignals (s(nT)), wobei K eine ganze Zahl ist, und
    Berechnen (103) eines Energiemittels von K Frequenzkomponenten von jeweils gleichem Rang (j) in den K Gruppen für die jeweils eine der extrahierten Frequenzenergiekomponenten.
  3. Verfahren nach Anspruch 2, bei dem dem Schritt des Berechnens (103) für jede der K Gruppen von Frequenzkomponenten ein Schritt des Auswählens (101) eines Teils der Frequenzkomponenten mit jeweils vorgegebenen Rängen in jeder der Gruppen (E(1)i, E(2)i, E(3)i) vorangeht, wobei der ausgewählte Teil eine Symmetrieeigenschaft in Bezug auf das Komplement dieses Teils unter der Mehrzahl der extrahierten Frequenzkomponenten aufweist.
  4. Verfahren nach einem beliebigen der Ansprüche 1 bis 3, bei dem die Schritte des Erzeugens (100a, 100b, 100c) und der Synthese (13) jeweils durch schnelle Fourier-Transformation und inverse schnelle Fourier-Transformation durchgeführt werden.
  5. Vorrichtung zur Unterdrückung eines Rauschsignals in einem abgetasteten verrauschten Sprachsignal (s(nT)), welche für jeden von aufeinanderfolgenden Verarbeitungszyklen umfasst:
    Mittel (10) zum Extrahieren einer Mehrzahl von Frequenzenergiekomponenten (Emj) in dem verrauschten Sprachsignal (s(nT)),
    Mittel (11) zum Abschätzen, für jede der extrahierten Frequenzenergiekomponenten, eines Verhältnisses (SNRj) zwischen einem Energieniveau des verrauschten Sprachsignals (s(nT)) und einem Energieniveau des Rauschsignals,
    Mittel zum Bestimmen (12) einer jeweiligen Verstärkung (Gj) für jede der extrahierten Frequenzenergiekomponenten (Emj) in Abhängigkeit von dem abgeschätzten Verhältnis (SNRj) zwischen dem Energieniveau des verrauschten Sprachsignals (s(nT)) und dem Energieniveau des Rauschsignals für jede jeweils ausgewählte Frequenzkomponente,
    Mittel (13) zum Synthetisieren der Filterkoeffizienten (C(nT)) in Abhängigkeit von den Verstärkungen (Gj), und
    Mittel zur digitalen Verarbeitung im Zeitbereich (14) des verrauschten Sprachsignals (s(nT)) in Abhängkeit von den Filterkoeffizienten (C(nT)) zu einem Sprachsignal (s(nT)), bei dem das Rauschsignal im wesentlichen unterdrückt ist.
  6. Kombiniertes System zur Echoaufhebung (3) und Rauschunterdrückung (1), mit:
    einer Rauschunterdrückungsvorrichtung (1) zum Unterdrücken eines Rauschsignals in einem zu übertragenden Sprachsignal (s(nT)+e(nT)), um ein entrauschtes Sprachsignal zu bilden,
    einem Echoaufheber (3) mit einem ersten Mittel (31) zum Erzeugen eines abgeschätzten Echosignals (ê* (t)) als Funktion eines gegebenen Sprachsignals r*(nT)) und eines Differenzsignals (s*(nT)+e*(nT)-ê*(nT)) und einem zweiten Mittel (30) zum Subtrahieren des abgeschätzten Echosignals (ê*(t)) von dem entrauschten Sprachsignal (s*(nT)+e*(nT)), um das Differenzsignal (s*(nT)+e*(nT)-ê*(nT)) zu bilden,
       dadurch gekennzeichnet, dass die Rauschunterdrückungsvorrichtung ein Mittel zur digitalen Frequenzverarbeitung (100) des zu übertragenden Sprachsignals zum Erzeugen von Filterkoeffizienten (C(nT)) im Zeitbereich umfasst, mit:
    Mitteln (10) zum Extrahieren einer Mehrzahl von Frequenzenergiekomponenten (Emj) in dem verrauschten Sprachsignal (s(nT)),
    Mitteln (11) zum Abschätzen, für jede der extrahierten Frequenzenergiekomponenten, eines Verhältnisses (SNRj) zwischen einem Energieniveau des verrauschten Sprachsignals (s(nT)) und einem Energieniveau des Rauschsignals,
    Mitteln zum Bestimmen (12) einer jeweiligen Verstärkung (Gj) für eine jede der extrahierten Frequenzenergiekomponenten (Emj) in Abhängigkeit von dem abgeschätzten Verhältnis (SNRj) zwischen dem Energieniveau des verrauschten Sprachsignals (s(nT)) und dem Energieniveau des Rauschsignals für die jede jeweils ausgewählte Frequenzkomponente,
    Mitteln (13) zum Synthetisieren der Filterkoeffizienten (C(nT)) in Abhängigkeit von den Verstärkungen (Gj),
       und dass die Rauschunterdrückungsvorrichtung ferner umfasst:
    ein erstes Mittel zur digitalen Verarbeitung im Zeitbereich (14) zur Verarbeitung des zu übertragenden Sprachsignals (s(nT)) in Abhängkeit von den Filterkoeffizienten (C(nT)) zu dem entrauschten Sprachsignal (s*(nT) + e*(nT)), bei dem das Rauschsignal im wesentlichen unterdrückt ist,
       ein zweites Mittel zur digitalen Verarbeitung im Zeitbereich (14'), des dem ersten Mittel zur Verarbeitung im Zeitbereich (14) genau entspricht, zum Verarbeiten eines von einem entfernten Endgerät empfangenen Sprachsignals (r(nT)) in Abhängigkeit von den Filterkoeffizienten (C(nT)) zu dem gegebenen Sprachsignal (r*(nT))
  7. Kombiniertes System zur Echoaufhebung (3) und Rauschunterdrückung (1) in einem zu übertragenden Sprachsignal (s(nT)+e(nT)) mit
       einem Echoaufheber (3) mit einem ersten Mittel (31) zum Erzeugen eines geschätzten Echosignals (ê*(t)) als Funktion eines von einem entfernten Endgerät empfangenen Sprachsignals (r(nT)) und eines Differenzsignals (s(nT)+e(nT)-ê(nT)) und einem zweiten Mittel (30) zum Subtrahieren des abgeschätzten Echosignals (ê*(t)) von einem zu übertragenden Sprachsignal (s(nT)+e(nT)), um das Differenzsignal (s(nT)+e(nT)-ê(nT)) zu erhalten,
       dadurch gekennzeichnet, dass es eine Rauschunterdrückungsvorrichtung (1) zum Erzeugen eines entrauschten Sprachsignals (s*(nT)+e*(nT)-ê*(nT)) durch Unterdrücken eines Rauschsignals in dem Differenzsignal s(nT)+e(nT)-ê(nT)) umfasst, wobei die Rauschunterdrückungsvorrichtung umfasst:
    einerseits ein Mittel zur digitalen Verarbeitung (100) im Frequenzbereich zum Verarbeiten des zu übertragenden Sprachsignals (s(nT)+ enT)), um zeitliche Filterkoeffizienten (C(nT)) zu erzeugen, mit:
    Mitteln (10) zum Extrahieren einer Mehrzahl von Frequenzenergiekomponenten (Emj) in dem verrauschten Sprachsignal (s(nT)),
    Mitteln (11) zum Abschätzen, für jede der extrahierten Frequenzenergiekomponenten, eines Verhältnisses (SNRj) zwischen einem Energieniveau des verrauschten Sprachsignals (s(nT)) und einem Energieniveau des Rauschsignals,
    Mitteln zum Bestimmen (12) einer jeweiligen Verstärkung (Gj) für jede jeweils extrahierte Frequenzenergiekomponente (Emj) in Abhängigkeit von dem abgeschätzen Verhältnis (SNRj) zwischen dem Energieniveau des verrauschten Sprachsignals (s(nT)) und dem Energieniveau des Rauschsignals für besagte jeweils ausgewählte Frequenzkomponente,
    Mitteln (13) zum Sythetisieren der Filterkoeffizienten (C(nT)) in Abhängigkeit von den Verstärkungen (Gj), und
    andererseits ein Mittel zur digitalen Verarbeitung im Zeitbereich (14) zur Verarbeitung des Differenzsignals s(nT)+e(nT)-ê(nT)) in Abhängigkeit von den Filterkoeffizienten (C(nT)) zu einem entrauschten Sprachsignal (s*(nT)+e*(nT)-ê*(nT)), in dem das Rauschsignal im wesentlichen unterdrückt ist.
EP95402385A 1994-10-28 1995-10-25 Verfahren und Apparat zur Geräuschunterdrückung in einem Sprachsignal und korrespondierendes System mit Echounterdrückung Expired - Lifetime EP0710947B1 (de)

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FR9412964 1994-10-28
FR9412964A FR2726392B1 (fr) 1994-10-28 1994-10-28 Procede et dispositif de suppression de bruit dans un signal de parole, et systeme avec annulation d'echo correspondant

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EP0710947B1 true EP0710947B1 (de) 2003-01-08

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US (1) US5680393A (de)
EP (1) EP0710947B1 (de)
JP (2) JPH08213936A (de)
AT (1) ATE230890T1 (de)
AU (1) AU698081B2 (de)
CA (1) CA2161575A1 (de)
DE (1) DE69529328T2 (de)
FI (1) FI955086A (de)
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FR2726392B1 (fr) 1997-01-10
DE69529328D1 (de) 2003-02-13
JP2007129736A (ja) 2007-05-24
AU3444295A (en) 1996-05-09
NZ280224A (en) 1997-02-24
EP0710947A1 (de) 1996-05-08
FI955086A0 (fi) 1995-10-25
AU698081B2 (en) 1998-10-22
JP4567655B2 (ja) 2010-10-20
FR2726392A1 (fr) 1996-05-03
JPH08213936A (ja) 1996-08-20
FI955086A (fi) 1996-04-29
ATE230890T1 (de) 2003-01-15
CA2161575A1 (fr) 1996-04-29
US5680393A (en) 1997-10-21
DE69529328T2 (de) 2003-09-04

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