EP2490218B1 - Procédé pour supprimer l'interférence - Google Patents

Procédé pour supprimer l'interférence Download PDF

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EP2490218B1
EP2490218B1 EP11155047.1A EP11155047A EP2490218B1 EP 2490218 B1 EP2490218 B1 EP 2490218B1 EP 11155047 A EP11155047 A EP 11155047A EP 2490218 B1 EP2490218 B1 EP 2490218B1
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signal
loudspeaker
band
microphone
interference
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EP2490218A1 (fr
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Arthur Wolf
Bernd Iser
Patrick Hannon
Mohamed Krini
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SVOX AG
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SVOX AG
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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

Definitions

  • an indoor or car sound player can also be active while the ICC is working, and is also coupling into the microphone.
  • Typical audio systems support stereo or even multi channel (e.g. dolby digital) audio playback. Due to the strong correlation between the single audio channels, the echo cancellation of multi channel audio output signal is also very challenging.
  • the noise components should be attenuated by noise suppression, as has already been mentioned in the above article of E. Lleida, E. Masgrau and A. Ortega.
  • the microphone of a handsfree system also picks up the background noise and the played back audio signal. These interferences should be suppressed, before transmitting the signal to the remote subscriber.
  • the microphone of an automatic speech recognition system also picks up the background noise and the played back audio signal. Noise and these interferences should be suppressed by adequate signal processing to improve the speech recognition results.
  • An indoor communication system or a handsfree telephony system or an automatic speech recognition system comprising at least one loudspeaker, at least one microphone and a signal processing system, particularly in a vehicle, wherein the microphone is recording a signal comprising communication information and interferences, the signal processing system is processing the microphone signal and providing a loudspeaker signal and the loudspeaker is emitting a sound signal corresponding to the loudspeaker signal, the at least one microphone picking up at least a portion of the loudspeaker signal and thereby causing feedback.
  • the method for interference suppression in the communication system includes the step of the signal processing system estimating an interference signal by an energy decay model with frequency dependent coupling factors, frequency dependent decay factors and frequency dependent delay factors, wherein an estimated interference signal includes at least one product of the respective coupling factor times a respective part of a loudspeaker signal delayed by the respective delay factor plus at least one product of an estimated interference signal at an earlier moment times the respective decay factor.
  • the estimated interference signal is used for generating an interference suppressed loudspeaker signal.
  • interferences for example feedback of the communication system, feedback of the audio system and background noise.
  • For at least one interference signal an estimation is made.
  • the signal quality can be enhanced by applying stepwise independently or combined at least two suppressing steps out of communication system feedback suppression, audio system feedback suppression and noise suppression, wherein for all the suppression steps corresponding estimations of interference signals are used, and the suppression modules can be arranged in any order, but are preferably arranged in the following order: communication system feedback, audio system feedback and noise suppression.
  • the estimated interference signal is a communication system feedback signal or an audio-system feedback, wherein the audio signal is a part or all of the loudspeaker signal at loudspeaker Ls. If there is more than one communication system or a system with multiple communication directions then the interference of a further communication system or a further communication direction can be treated like an additional interference and be suppressed in the above described manner.
  • a test signal can be used to determine interference model parameters.
  • a test signal is sent to the respective loudspeaker for determining system characteristics on the base of the test signal received by the respective microphone, wherein the system characteristics detected by the application of the test signal are used to determine the frequency dependent coupling factors, the frequency dependent decay factors and frequency dependent delay factors. In some applications, it may be sufficient to do it once. Considering, particularly for a car, that persons may leave the room, may open a window or the like, the model parameters may vary over time.
  • the interference model parameters can be deduced from at least one microphone signal or by automatically detecting and interpreting decaying signal slopes, wherein after deducting the parameters to be applied are updated and the parameter deduction occurs preferably when there is no local speech.
  • the interference model parameters can be deduced from the coefficients of an echo compensator preferably by placing an echo compensator parallel to the signal processing path, wherein the coefficients of the echo compensator are corresponding to the room impulse response and the parameters of the echo compensator should only be updated when there is no local speech detected, e.g. only at the decaying slopes of the microphone signal or only at the feedback of the audio signal.
  • an interference suppressed loudspeaker signal is generated by Spectral Subtraction, preferably by the application of a Wiener-filter using the estimated interference signal.
  • An overestimation ⁇ ( ⁇ ,k) of the estimated interference signal of the sub-band ⁇ can be adjusted to an estimated noise to speech signal ratio SNR( ⁇ ,k) of the sub-band ⁇ deduced by a noise suppression module.
  • a maximum attenuation ⁇ ( ⁇ ,k) of the sub-band ⁇ is adjusted to a gain correction factor V( ⁇ ,k) of the sub-band ⁇ .
  • a communication system equalizer H Eq ( ⁇ ,k) can be adjusted according to the energy decay model frequency dependent coupling factors P LsMic ( ⁇ ,k).
  • a communication system gain can be adjusted according to the estimated interference signal level, wherein this gain is dependent on the background noise level and on the level of the communication system feedback and audio system feedback, for a high communication system feedback or audio system feedback signal level the system gain should be reduced.
  • the invention relates to a software product according to claim 11 and to a system according to claim 12.
  • a room 1 such as a car cabin
  • a driver 2a In a room 1, such as a car cabin, there are a driver 2a and a passenger 2b behind.
  • a microphone 3 In front of the driver 2a is a microphone 3 so that the driver's speech is better intelligible for the passenger 2b to whom a loudspeaker 4 is assigned.
  • a (further) microphone might be near the passenger 2b to be better understood by the driver 2a to whom a loudspeaker of the type of loudspeaker 4 may also be assigned.
  • the microphone 3 will not only take the speech signal s(n) of the driver 2a, but also the noise b(n), and noise suppression in the line from the microphone 3 to the loudspeaker 4 is known per se.
  • there are interferences received by the microphone such as the audio signal f Audio,Mic (n) from a further loudspeaker 5, by means of which the driver wants, for example, to become informed about street conditions or obtains a navigation aid from an audio-source 6.
  • the output m(n) of the microphone 3 which is composed of the voice signal s(n) of the driver 2a, the noise b(n), the audio-signal f Audio,Mic (n) and the ICC-system feedback signal f ICC,Mic (n) coming from loudspeaker 4, the two latter signals forming interferences. If one feeds at least one of the interferences, such as f Audio,Ls (n), into the ICC-system 7 for suppression, speech of the driver 2a is better intelligible, because the signal to the loudspeaker 4 is enhanced (vide also Fig. 2 ).
  • the signals in time domain with the time index n are defined as lower case characters and signals in sub-band domain with the sub-band index ⁇ and the frame index k are defined as upper case characters.
  • the index Mic is used for the signal at the microphone and the index Ls is used for the signal at the loudspeaker.
  • the index Mic has a different value for each microphone, where it runs for example from 0 to the maximum number of microphones minus 1.
  • the index Ls has a different value for each loudspeaker, where it runs for example from 0 to the maximum number of loudspeakers minus 1
  • the upper case character S with an index i.e.
  • S bb ( ⁇ ,k) is used for power signals, while other capital letters, e.g. B( ⁇ ,k), are used for complex sub-band signals.
  • the sub-band microphone signal M( ⁇ ,k) of a given sub-band ⁇ at a given time k consists of the local speech signal S( ⁇ ,k), the background noise B( ⁇ ,k), the feedback of the ICC-system output F ICC,Mic ( ⁇ ,k) and the feedback of the audio system F Audio,Mic ( ⁇ ,k), in which ⁇ is the sub-band index and k is the time frame index.
  • the invention provides a method of interference suppression using a mathematical model on the base of sound energy decay inside a room. In addition to the energy decay there is also a delay effect.
  • the signal processing system is estimating an interference signal by an energy decay model with frequency dependent coupling factors, frequency dependent decay factors and frequency dependent delay factors, wherein an estimated interference signal includes at least one product of the respective coupling factor times a respective part of a loudspeaker signal delayed by the respective delay factor plus at least one product of an estimated interference signal at an earlier moment times the respective decay factor, and that the estimated interference signal is used for generating an interference suppressed loudspeaker signal.
  • Frequency dependent coupling factors, frequency dependent decay factors and frequency dependent delay factors can be calculated preferably from a room impulse response.
  • the interference model parameters are deduced form at least one microphone signal or from automatically detecting and interpreting decaying signal slopes, wherein after deducting the parameters to be applied are updated and the parameter deduction occurs preferably when there is no speech, no audio signal or no ICC-system output.
  • the coefficients of an echo compensator can be used for the adaption of the interference model parameters preferably by placing an echo compensator parallel to the signal processing path, wherein the coefficients of the echo compensator are corresponding to the room impulse response and the parameters of the echo compensator should only be updated when there is no local speech detected, e.g. only at the decaying slopes of the microphone signal or only at the feedback of the audio signal.
  • Noise components can be suppressed by a Wiener-filter in sub-band domain.
  • Signal processing can be applied in sub-band or also in a melband domain to take the psychoacoustics into the account or to reduce the algorithmic complexity.
  • the difficulty is the estimation of the ICC-system feedback and the audio signal feedback at the microphone.
  • the ICC-system feedback and the audio signal feedback are known at the loudspeaker or can be supplied as a reference channel from the output of the ICC and the audio system ( Fig. 2 ).
  • the method according to the invention uses a model for the energy decay of the room impulse response.
  • the energy decay of the room impulse response is modelled as the outcome of a non-stationary random process.
  • E h LsMic 2 n 0 for n ⁇ 0
  • E h LsMic 2 n ⁇ 2 e ⁇ gn for n ⁇ 0 .
  • G LsMic ⁇ k 0 for k ⁇ D ⁇
  • G LsMic ⁇ k P LsMic ⁇ e ⁇ ⁇ ⁇ , k ⁇ D for k ⁇ D ⁇ .
  • the parameters for this sub-band energy decay model G LsMic ( ⁇ ,k) can be estimated from the sub-band energy decay curve from the impulse response as shown in Fig. 3a .
  • Very similar energy decay models can be used e.g. for dereverberation of a microphone signal. This has been disclosed in US2009/0117948 and also by E. Habets in "Multichannel speech dereverberation based on a statistical model of late reverberation," in ICASSP, 2005 .
  • the model is used for estimation of the ICC-system output and the audio signal at the microphone, i.e. for interference estimation.
  • S ff,ICC,Mic ( ⁇ ,k) is the estimated feedback of the ICC-system output at the microphone and S ff,Audio,Mic ( ⁇ ,k) is the estimated feedback of the audio signal at the microphone.
  • S xx ( ⁇ ,k) is used in this specification for power
  • other capital letters, such as M( ⁇ ,k) or B( ⁇ ,k) are used for complex sub-band signals.
  • Spectral Subtraction e.g. Wiener-filter in the sub-band domain can be used.
  • Wiener-filter in the sub-band domain.
  • the attenuation of the filter coefficients is constrained to a maximum attenuation (spectral floor) ⁇ ( ⁇ ).
  • an overestimation factor ⁇ ( ⁇ ) For reduction of the artefacts caused by the interference suppression, also called musical noise, an overestimation factor ⁇ ( ⁇ ) will be used.
  • the overestimation factor ⁇ ( ⁇ ) is a fixed value, with e.g. 1 ⁇ ⁇ ( ⁇ ) ⁇ 3. Because these artefacts are masked by the residual noise primarily caused by the noise suppression the improved solution contains a SNR(k, ⁇ ) (signal to noise ration) dependent overestimation factor ⁇ (k, ⁇ ).
  • T( ⁇ ,k) is the feedback and noise suppressed signal and approximates the clean local speech signal S ss ( ⁇ ,k) power and S bb ( ⁇ ,k) is the estimated nose signal power.
  • This adaptation of the SNR(k, ⁇ ) dependent overestimation factor ⁇ (k, ⁇ ) depends on a characteristic which maps the SNR(k, ⁇ ) to the overestimation factor ⁇ (k, ⁇ ).
  • the overestimation factor ⁇ (K, ⁇ ) can also be defined and determined for every processing step as ⁇ F,ICC (k, ⁇ ), ⁇ F,Audio (k, ⁇ ) and ⁇ B (k, ⁇ ).
  • This filter can be applied to the disturbed microphone signal.
  • R ⁇ k M ⁇ k H F , ICC ⁇ k . to obtain the ICC-system feedback suppressed signal R( ⁇ ,k).
  • this ICC-system feedback suppressed signal R( ⁇ ,k) can advantageously be used for suppressing the audio system feedback.
  • H F , Audio ⁇ k min ⁇ F , Audio , 1 ⁇ ⁇ F , Audio k ⁇ ⁇ ⁇ ff , Audio ⁇ ,Mic ⁇ k / R ⁇ k 2 .
  • the ICC-system and audio system feedback suppressed signal L( ⁇ ,k) can be used for noise signal suppression.
  • This amplification factor can be calculated for every sub-band V( ⁇ ,k) or as scalar fullband parameter v(k).
  • One possible implementation for the update of the amplification factor is to calculate the update terms for every filter.
  • ⁇ start ( ⁇ ) is the initial value for the spectral floor.
  • ⁇ Audio ( ⁇ ,k) and ⁇ B ( ⁇ ,k) are updated the same way.
  • the described method enables to enhance the interfered signal by a very robust and efficient way with a circuit schematically shown in Fig. 3b .
  • the configuration shown in Fig. 3b depends on the actual system setup. Feedback suppression, audio suppression and noise suppression is applied stepwise, where for all these suppression steps corresponding estimations of interference signals are used.
  • the suppression modules are arranged in the following order: Feedback, Audio and Noise. Rearrangements of the used modules are possible and may in some cases be necessary. It is possible to perform every processing step independently like shown before. There the modules can also be rearranged and/or combined.
  • H ⁇ k minimum H F , ICC ⁇ k , H F , Audio ⁇ k , H B ⁇ k .
  • H( ⁇ ,k) is the combined interference suppression filter coefficients dependent on the single components ICC-system feedback suppression filter coefficients H F,ICC ( ⁇ ,k), audio system feedback suppression filter coefficients H F,Audio ( ⁇ ,k) and noise suppression filter coefficients H B ( ⁇ ,k).
  • the microphone signal M( ⁇ ,k) is transformed by a ICC-system feedback suppression step 11 to a feedback reduced signal R( ⁇ ,k).
  • a ICC-system feedback estimation step 12 preposed.
  • the output of module 12, the estimated interference signal level S ff,ICC,Mic ( ⁇ ,k), is delivered to module 11 and is used there to suppress the interference signal accordingly.
  • module 14 is now freed from feedback interference components and can, therefore, better be used for noise estimation in module 15, to feed a noise suppression stage 16. Since the enhanced signal has lost power, it is useful, to correct the signal level by a gain control module 17 which forms a power level corrected signal Y( ⁇ ,k) and is in connection with modules 11, 14 and 16. Therefore the gain control stage 17 analyzes the filter coefficients of the modules 11, 14, 16 and returns the for adjusted spectral floor factors back to the modules 11, 14, 16.
  • the delay D( ⁇ ) and the energy decay e - ⁇ ( ⁇ ) are related to the used hardware and the room characteristics e.g. the reverberation time T 60 .
  • the changes of these parameters are slow and small.
  • the coupling factor P LsMic ( ⁇ ) depends on the actual position of the passengers inside the car and is changing faster. In the majority of cases it is sufficient only to adapt this parameter during signal processing.
  • the room energy decay parameters can be estimated from the impulse response respectively the sub-band impulse response.
  • This impulse response can be measured, before calculating the signal processing and the estimated model parameters D( ⁇ ), P LsMic ( ⁇ ) and e - ⁇ ( ⁇ ) , see also Fig 3a . With these parameters, signal processing can be applied.
  • the impulse response Due to the changes of the impulse response, caused by changes of the car occupancy and environment conditions, e.g. open window or door, it is suitable to repeat the impulse response measurements for different car occupancies and environment conditions, e.g. to have different decay models for different occupancies and environment conditions.
  • the occupancy or environment conditions of the car can be detected, e.g. with seat sensors or window sensor, and the signal processing can switch to the actual predefined decay model.
  • an echo compensator 18 ( Fig. 4a ) which is placed parallel to the signal processing path 7.
  • the output of the echo compensator 18 is not used for feedback compensation, but only for updating the echo compensator.
  • the estimated coefficients Due to the correspondence of the coefficients of the echo compensator to the room impulse response (as is known from EP-A-2151983 ), the estimated coefficients can be used in a very similar way to estimate or to update the decay model parameter during the signal processing.
  • the parameters of the echo compensator should only be updated when there is no local speech detected, e.g. only at the decaying slopes of the microphone signal or only at the feedback of the audio signal.
  • a further possibility to estimate the room energy decay model parameters is to use of frequency/phase shift methods or other decorrelation methods like nonlinearities at the system output or additional noise signal.
  • the decay model parameter can be easily updated from this decorrelated loudspeaker signal. This method can be used together to the parallel echo compensator 18 to support and accelerate the adaptation of the echo compensator 18.
  • Still another possibility to estimate the room energy decay model parameters is to automatically detect and interpret the decaying signal slopes.
  • the energy decay of the slope needs to be monitored. The fastest decay appears when there is no additional excitation signal, e.g. no local speech, no audio signal or no ICC-system output.
  • the room energy decay model parameters can be updated by the estimated sub-band decay and the sub-band transfer at the beginning of the slope.
  • Another possibility is to update the decay model parameter from the calculated cross correlation between the loudspeaker signal F ICC,Ls ( ⁇ ,k) and the microphone signal M( ⁇ ,k) to estimate the room energy decay model parameters.
  • the described model for the energy decay can also be used for adjusting the coefficients of the equalizer which can also be a part of the ICC system. Therefore the sub-band coupling parameter P LsMic ( ⁇ ,k) can be used to set up the sub-band equalizer 19 (vide Fig 4b ) to improve the stability ICC-system gain in term of maximum ICC-system gain, due to the correlation between the room impulse response h LsMic (n) and the sub-band coupling parameter P LsMic ( ⁇ ,k).
  • the estimation of the interference components can also be used to set up the ICC system gain.
  • this gain is dependent on the background noise level S bb ( ⁇ ,k). But it is also dependent on the level of the feedback and audio signal. Because for a high feedback the ICC system produces many artefacts the system gain should be reduced.
  • the audio level at the microphone S ff,Audio,Mic ( ⁇ ,k) can be estimated with the described method. This signal correlates to the sound level inside the car. In relation to the ratio between the estimated audio signal and the processed signal the system gain can be reduced or the system can be deactivated in order to not disturb the passengers, while listening music.
  • the processed signal contains many artefacts caused by the signal processing.
  • the ICC-system gain should be reduced to reduce the level of the feedback signal S ff,ICC,Mic ( ⁇ ,k) or switch off the ICC-system while the ICC-system is working under inconvenient or not acceptable conditions.
  • the communication system gain can be adjusted according to the estimated interference signal level, wherein this gain is dependent on the background noise level and on the level of the communication system feedback and audio system signal, for a high communication system feedback or for a high audio signal level the system gain should be reduced.

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Claims (12)

  1. Procédé pour une suppression d'interférences pour un système de communication, comprenant au moins un haut-parleur (4, 5), au moins un microphone (3) et un système de traitement de signal (7), dans lequel le microphone enregistre un signal comprenant des informations de communication et des interférences, le système de traitement de signal traite le signal de microphone et fournit un signal de haut-parleur et le haut-parleur émet un signal sonore correspondant au signal de haut-parleur, l'au moins un microphone capturant au moins une partie du signal de haut-parleur et produisant ainsi une rétroaction, caractérisé par le fait que le système de traitement de signal estime un signal d'interférence par un modèle de décroissance d'énergie avec des facteurs de couplage dépendant de la fréquence, des facteurs de décroissance dépendant de la fréquence et des facteurs de retard dépendant de la fréquence, dans lequel un signal d'interférence estimé comprend au moins un produit du facteur de couplage respectif multiplié par une partie respective d'un signal de haut-parleur retardé par le facteur de retard respectif plus au moins un produit d'un signal d'interférence estimé à un moment antérieur multiplié par le facteur de décroissance respectif, et par le fait que le signal d'interférence estimé est utilisé pour générer un signal de haut-parleur à interférences supprimées.
  2. Procédé selon la revendication 1, caractérisé par le fait que le signal d'interférence estimé est calculé selon la formule : S ff , Mic μ k = P LsMic μ S ff , Ls μ , k − D μ + S ff , Mic μ , k − 1 e − ϕ μ ,
    Figure imgb0035
     dans laquelle Sff,Mic(µ,k) est le niveau de signal d'interférence estimé au niveau du microphone Mic, pour une sous-bande de fréquence prédéterminée µ à un instant k ; PLsMic(µ) est le facteur de couplage de la sous-bande µ entre le haut-parleur Ls et le microphone Mic ; Sff,Ls(µ,k-D(p)) est le niveau de signal d'interférence au niveau du haut-parleur Ls pour une sous-bande de fréquence prédéterminée à un instant k-D(p) ; D(µ) est le facteur de retard pour la sous-bande µ ; Sff,Mic(µ,k-1) est le niveau de signal d'interférence au niveau du microphone Mic pour une sous-bande de fréquences prédéterminée à un instant k-1 ; et e-ϕ(µ) est un facteur de décroissance exponentielle de la sous-bande µ.
  3. Procédé selon la revendication 1 ou 2, caractérisé par le fait que le signal d'interférence estimé est un signal de rétroaction de système de communication ou un signal de rétroaction de système audio, dans lequel le signal de rétroaction comprend une partie de ou tout le signal de haut-parleur au niveau du haut-parleur Ls.
  4. Procédé selon la revendication 1, dans lequel un signal d'essai est envoyé au haut-parleur respectif pour déterminer des caractéristiques de système sur la base du signal d'essai reçu par le microphone respectif, dans lequel les caractéristiques de système détectées par l'application du signal d'essai sont utilisées pour déterminer les facteurs de couplage dépendant de la fréquence, les facteurs de décroissance dépendant de la fréquence et les facteurs de retard dépendant de la fréquence.
  5. Procédé selon l'une quelconque des revendications précédentes, caractérisé par le fait que la génération d'un signal de haut-parleur à interférences supprimées est réalisée par soustraction spectrale, de préférence par l'application d'un filtre Wiener en utilisant le signal d'interférence estimé.
  6. Procédé selon l'une quelconque des revendications précédentes, caractérisé par le fait qu'une surestimation γ(µ,k) du signal d'interférence estimé de la sous-bande µ est ajustée à un rapport bruit sur signal vocal estimé SNR(µ,k) de la sous-bande µ déduit par un module de suppression de bruit (16) et par le fait qu'une atténuation maximale β(µ,k) de la sous-bande µ est ajustée à un facteur de correction de gain V(µ,k) de la sous-bande µ.
  7. Procédé selon l'une quelconque des revendications précédentes, caractérisé en outre par application pas-à-pas de façon indépendante ou combinée d'au moins deux étapes de suppression parmi une suppression de rétroaction de système de communication (11), une suppression de rétroaction de système audio (14) et une suppression de bruit (16), dans lequel pour toutes les étapes de suppression, des estimations correspondantes de signaux d'interférence sont utilisées, et les modules de suppression (11, 14, 16) peuvent être agencés selon un ordre quelconque, mais sont de préférence agencés dans l'ordre suivant : rétroaction de système de communication, rétroaction de système audio et bruit.
  8. Procédé selon l'une quelconque des revendications précédentes, caractérisé par le fait que les paramètres de modèle d'interférence sont déduits à partir d'au moins un signal de microphone ou par détection et interprétation de façon automatique de pentes de signal de décroissance, dans lequel après déduction, les paramètres à appliquer sont actualisés et la déduction de paramètres se produit de préférence lorsqu'il n'y a pas de parole locale.
  9. Procédé selon l'une quelconque des revendications précédentes, caractérisé par les autres étapes consistant à préparer au moins deux modèles d'interférence différents sur la base de différents paramètres pour des conditions d'occupations différentes ou d'environnements différents ; détecter l'occupation réelle de véhicule ou la condition d'environnement ; et sélectionner les modèles d'interférence selon la condition d'occupation ou d'environnement réelle détectée.
  10. Procédé selon l'une quelconque des revendications précédentes, caractérisé par le fait que les coefficients d'un compensateur d'écho sont utilisés pour l'adaptation des paramètres de modèle d'interférence, de préférence en plaçant un compensateur d'écho (18) parallèlement au trajet de traitement de signal, dans lequel les coefficients du compensateur d'écho sont correspondants à une réponse impulsionnelle de pièce et les paramètres du compensateur d'écho devraient être actualisés uniquement lorsqu'il n'y a pas de parole locale détectée.
  11. Produit logiciel comprenant des instructions qui, lorsque le logiciel est exécuté par un ordinateur, amènent l'ordinateur à réaliser le procédé selon l'une quelconque des revendications précédentes.
  12. Système de communication comprenant au moins un haut-parleur (4, 5) et au moins un microphone (3), ainsi qu'un système de traitement de signal (7), qui réalise un procédé selon l'une quelconque des revendications 1 à 10.
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CN104601258B (zh) * 2014-12-23 2017-03-15 中国北方车辆研究所 一种车载通信***同车多机互扰测试方法
CN106328152B (zh) * 2015-06-30 2020-01-31 芋头科技（杭州）有限公司 一种室内噪声污染自动识别监测***
US10002601B2 (en) 2015-12-30 2018-06-19 Qualcomm Incorporated In-vehicle communication signal processing
JP6567985B2 (ja) * 2016-02-17 2019-08-28 アルパイン株式会社 車載装置
CN112309364B (zh) * 2020-11-04 2024-07-02 戴欧德(广东)科技有限公司 一种dsp多通道降啸叫处理的实现方法及***和芯片
CN113611276B (zh) * 2021-07-08 2024-06-11 北京小唱科技有限公司 声反馈抑制方法、装置及存储介质

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EP1885154B1 (fr) * 2006-08-01 2013-07-03 Nuance Communications, Inc. Déreverbération des signaux d'un microphone
WO2008122930A1 (fr) * 2007-04-04 2008-10-16 Koninklijke Philips Electronics N.V. Amélioration du son dans des espaces fermés
EP2058804B1 (fr) 2007-10-31 2016-12-14 Nuance Communications, Inc. Procédé de déréverbération d'un signal acoustique et système associé
EP2151983B1 (fr) 2008-08-07 2015-11-11 Nuance Communications, Inc. Téléphonie mains libres et communication dans le véhicule

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