EP2545717A1 - Suppression de la réverbération de signaux d'un dispositif auditif biauriculaire - Google Patents

Suppression de la réverbération de signaux d'un dispositif auditif biauriculaire

Info

Publication number
EP2545717A1
EP2545717A1 EP10739560A EP10739560A EP2545717A1 EP 2545717 A1 EP2545717 A1 EP 2545717A1 EP 10739560 A EP10739560 A EP 10739560A EP 10739560 A EP10739560 A EP 10739560A EP 2545717 A1 EP2545717 A1 EP 2545717A1
Authority
EP
European Patent Office
Prior art keywords
reverberation
signal
input signals
reference signal
binaural
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Withdrawn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
EP10739560A
Other languages
German (de)
English (en)
Inventor
Marco Jeub
Heinrich LÖLLMANN
Peter Vary
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.)
Sivantos Pte Ltd
Original Assignee
Siemens Medical Instruments Pte Ltd
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 Medical Instruments Pte Ltd filed Critical Siemens Medical Instruments Pte Ltd
Priority to EP10739560A priority Critical patent/EP2545717A1/fr
Publication of EP2545717A1 publication Critical patent/EP2545717A1/fr
Withdrawn legal-status Critical Current

Links

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/43—Electronic input selection or mixing based on input signal analysis, e.g. mixing or selection between microphone and telecoil or between microphones with different directivity characteristics
    • 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
    • G10L2021/02082—Noise filtering the noise being echo, reverberation of the speech
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/55—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using an external connection, either wireless or wired
    • H04R25/552—Binaural

Definitions

  • the present invention relates to a method for providing a ⁇ enthallten binaural output signal of a binaural hearing device.
  • the present invention relates to a corresponding binaural hearing device.
  • a hearing device is understood here to mean any sound-emitting device which can be worn in or on the ear, in particular a hearing device, a headset, headphones and the like.
  • Hearing aids are portable hearing aids that are used to care for the hearing impaired.
  • different types of hearing aids such as behind-the-ear hearing aids (BTE), hearing aid with external receiver (RIC: receiver in the canal) and in-the-ear hearing aids (IDO), e.g. Concha hearing aids or canal hearing aids (ITE, CIC).
  • BTE behind-the-ear hearing aids
  • RIC hearing aid with external receiver
  • IDO in-the-ear hearing aids
  • ITE canal hearing aids
  • the hearing aids listed by way of example are worn on the outer ear or in the ear canal.
  • bone conduction hearing aids, implantable or vibrotactile hearing aids are also available on the market. The stimulation of the damaged hearing takes place either mechanically or electrically.
  • Hearing aids have in principle as essential components an input transducer, an amplifier and an output transducer.
  • the input transducer is usually a sound receiver, z. As a microphone, and / or an electromagnetic receiver, for. B. an induction coil.
  • the output transducer is usually used as an electroacoustic transducer, z. As miniature speaker, or as an electromechanical transducer, z. B. bone conduction, realized.
  • the amplifier is usually integrated in a signal processing unit. This basic structure is shown in FIG. 1 using the example of a behind-the-ear hearing device. In a hearing aid housing 1 for carrying behind the ear, one or more microphones 2 for receiving the sound from the environment are installed.
  • a signal Processing unit 3 which is also integrated in the hearing aid housing 1, processes the microphone signals and ver ⁇ strengthens them.
  • the output signal of the signal processing unit 3 is transmitted to a loudspeaker or earpiece 4, which outputs an acoustic signal.
  • the sound is optionally, for the wearer's eardrum übertra ⁇ gene via a sound tube which is fixed with a otoplasty in the auditory canal.
  • the energy supply of the hearing aid and in particular the signal processing unit 3 by a likewise integrated into the hearing aid housing 1 battery 5.
  • the object of the present invention is to more effectively reduce the reverberation in a binaural hearing system.
  • this object is achieved by a method for providing a connected, binaural Ninth-a binaural hearing by picking up a left input signal and a right input signal through the hearing, combining the two input signals to a reference signal, determining spectral weights from the reference signal or providing spectral weights, with which reverberation can be reduced later, Beauf ⁇ beat the left and right input signal with the spectral weights, determining a coherence of signal components of the weighted input signals and attenuating non-coherent signal components of both weighted input signals to reduce an early reverberation.
  • the invention provides a binaural hearing device with a recording device for recording a left input signal and a right input signal, a signal processing means for combining the two input signals to a reference signal, a Ge ⁇ weighting means for determining the spectral weights from the reference signal or providing spectral weights, with which later reverberation can be reduced and for applying the left and right input signals to the spectral weights, and a coherence means for determining a coherence of signal components of the weighted input signals and for attenuating noncoherent signal components. Both of the weighted input signals reduce early reverberation.
  • a binaura- 1er Enthallungsalgorithmus is used according to the invention, is later ⁇ After reduced in the hall with Spektraltheen consisting of a com bined ⁇ signal (right signal with left signal) in the frequency ⁇ frequency range are obtained.
  • reverberation is earlier reduced by taking into account the coherence between the left and right signals. This can ensure a high harnesswerti ⁇ ge dereverberation.
  • a ⁇ nal is used, which is obtained by combining the left and right signal of the binaural hearing device.
  • a time offset between the two input signals is preferably compensated and the two input signals are then added to the reference signal. In this way, a simple reference signal can be obtained, with which weights for the reduction of the late reverberation can be obtained for both individual input signals.
  • the spectral weights are determined from the reference signal, it is convenient to estimate the reverberation time from the Refe ⁇ rence signal to do so. To estimate the reverberation time, it is particularly advantageous to make an initial selection of segments of the Re ⁇ conference signal. In this way, the reverberation time ⁇ can be very reliably estimated and Toggle the one hand can be a result of the computational effort significantly reduced.
  • the preselection Preferably, in the preselection only those segments are selected within which a drop in the sound level is detected. This waste can be used to estimate the reverberation time.
  • one fall time of the preselected segments can be determined and the fall time that occurs with the greatest likelihood can be defined as the reverberation time. This makes it possible to achieve a more robust method for obtaining the reverberation time.
  • the length of each of the segments can be adapted to the length of its sound output . Due to the variable length of the segments significantly computational effort can be saved.
  • the energy of this late reverberation is estimated to determine the spectral weights for the reduction of the late reverberation.
  • Energy estimation does not necessarily require an estimate of the reverberation time; rather, the energy can also be determined solely from the correlation of the spectral coefficients. Only with knowledge of the energy of the noise (reverberation) can this be effectively reduced.
  • FIG. 1 shows the basic structure of a hearing aid according to the
  • FIG. 2 shows a schematic diagram of a two-stage Enthallungs- system
  • FIG. 3 shows a detailed block diagram of a two-stage
  • a binaural, two-stage algorithm which allows the common Redukti ⁇ on early and late reverberation and basically preserves the binaural sound impression.
  • Such an algorithm is in M. Jeub, M. Shepherd, T. Esch and P. Vary: "Model-based de- reverberation preserving binaural cues", Preprint 2010, IEEE Transactions on Audio, Speech and Language Processing, ⁇ be enrolled.
  • a special approach of the coherence method is described in the above-mentioned article "M. Jeub and P. Vary," Binaural dereverberation based on a dual-channel vienna filter with optimized noise field coherence, "in Proc. IEEE Int. Conference on Acoustics, Speech and Signal Processing
  • FIG 2 is a simplified block diagram of beispielhaf ⁇ th two-stage Enthallungssystems again.
  • the recharge system is implemented, for example, in a hearing aid system with two hearing aids (one for the left ear and one for the right ear).
  • the two hearing aids of the hearing aid system are in communication with each other.
  • the microphone signal of the right hearing aid transmitted to the left ⁇ Hearing device, and in the left hearing aid, the Enthallungssystem is integrated.
  • Both input signals 1 and r (left channel and right channel) according to FIG. 2 are then available to the binaural reverberation system.
  • a first processing stage I provides a corresponding algorithm for the reduction of late reverberation.
  • As the output of the first stage I results in a binaural signal having a left Zvi ⁇ rule signal 1 'and a right intermediate signal r' correspondingly to the left channel and the right channel.
  • the two intermediate signals 1 'and r' are fed to a second processing stage II.
  • a coherence ⁇ based algorithm is implemented, which improves the two signals in terms of early reverberation. Ie. in the left intermediate signal 1 ', the early reverberation is reduced, resulting in an improved left output signal 1''.
  • the right intermediate signal r ' the early reverberation is reduced, so that an improved right Struktursig ⁇ nal r''results.
  • At the end of the reverberation system there is thus an improved binaural signal with a right channel and a left channel, in which both the late reverberation and the early reverberation are reduced.
  • FIG. 3 shows a block diagram for a detailed description of the two processing stages I and II of FIG.
  • the input signals Xi ( ⁇ , ⁇ ) and X r ( ⁇ , ⁇ ) of the first processing stage I which correspond to the input signals 1 and r of FIG. 2, are present here in the frequency domain. Ie. Before processing in the illustrated reverb system, a transformation takes place in the frequency domain.
  • the index ⁇ denotes a segment or a frame of the respective input signal.
  • the input signal is segmented NaEM ⁇ Lich and converted into short-term spectra.
  • the index ⁇ denotes a frequency range.
  • the two input signals of the left and right channel of a Kombina ⁇ tion unit 10 are supplied, in which the left input signal Xi ( ⁇ , ⁇ ) and the right input signal X r ( ⁇ , ⁇ ) to a refer- ence signal X ref ( ⁇ , ⁇ ) are combined.
  • the two input ⁇ signals are combined here in such a way that the time shift of the two signals is balanced to each other and they are then added.
  • the reference signal X ref ( ⁇ , ⁇ ) is transformed back into the time domain by an inverse transformation unit 11. From the reference signal in Zeitbe ⁇ rich the reverberation time is estimated by an estimator 12.
  • the reverberation time is defined as the time Inter ⁇ vall, in which the energy of a stationary sound field by 60 dB drops below the initial level, after the sound source has been turned off.
  • the estimation of the reverberation time can take place at ⁇ play as blind, that the reverberation time is obtained from a Hall signal without knowledge of the excitation signal and the room's geometry.
  • An advanced form of reverberation time estimator 12 relies on an improved blind reverberation time estimation algorithm. This improved algorithm is preferably in the fact that a noisy and ver ⁇ halltes speech signal is first processed by a Störgehoffschunterdrü- ckungssystem to obtain a entstörtes, verhalltes speech signal.
  • a pre-selection is carried out in order to detect segments in which a sound drop (energy drop of the sound) occurs. This detection takes place in the following substeps: The dividing ⁇ te already in frames or segments input signal is divided into sub-frames and a tough ⁇ ler is initialized to zero.
  • the energy, the maximum value and the minimum value of a current subframe are compared with the values of the next subframe.
  • the counter is incremented by one. Otherwise, the counter is set to zero.
  • the counter is checked whether the counter has already reached a minimum value.
  • the minimum value is, for example, three; For at least three values, it can be assumed that this is not a random energy drop within two subframes , but rather an actually sought energy drop. If the counter has thus reached a predetermined minimum value, a sound reduction is assumed. This is also the case when the counter reaches a predetermined maximum value. A maximum value is therefore given because the number of subframes, if it reaches the maximum value, is then sufficient for an estimate.
  • the counter In both cases (the counter reaches the minimum or maximum value), the counter is set to zero and the reverberation time is calculated using a ML estimator, as described in [Ratnam et al. , 2003].
  • the estimation is for a group of the last consecutive subframes where the counter has been incremented. Therefore, the length of such a group to which the ML estimate was applied is not fixed but adjusted to the (detected) speech decay.
  • This ML estimate represents a third step of the reverberation time estimation.
  • a histogram is z staple from the ML-contemptuous loading, which have been calculated within a predetermined past time ⁇ interval ,
  • a value of the reverberation time is used to select or set the actual reverberation time.
  • the values of the estimated reverberation time are smoothed over time to reduce the Va ⁇ RIANZ the estimate.
  • the advantage of the preselection is that a significant reduction in computational complexity can be achieved.
  • the new approach uses an adaptive buffer length for the ML estimation, which increases the estimation accuracy especially for low reverberation times.
  • the actual reverberation time is determined by the maximum of the
  • a reverberation time ⁇ is thus determined in the estimation unit 12.
  • This value ⁇ is fed together with the reference signal in the frequency range of a calculation ⁇ unit 13 which determines it in a known manner ⁇ example via an energy estimate weights G 'i ate ( ⁇ , ⁇ ) for the reduction of the late reverberation.
  • This ER-mediated weights are time ge smoothes ⁇ over several segments or frames of the input signal in a smoothing unit fourteenth This finally results in the weights Gi ate ( ⁇ , ⁇ ).
  • the smoothed weights Gi ate ( ⁇ , ⁇ ) are multiplied by both the left input signal Xi ( ⁇ , ⁇ ) and the right input signal X r ( ⁇ , ⁇ ) in the multiplication units 15 and 16 , As products result for the left channel the signal S ; ( ⁇ , ⁇ ) and for the right channel the sig- n S r ( ⁇ , ⁇ ), which correspond to the intermediate signals 1 'and r' of FIG. There was thus in the first processing stage ⁇ I binaural spectral subtraction to reduce the late reverberation.
  • the filter includes an arithmetic unit 17 to recover from a coherence of the signals of the left channel and right channel corresponding weights G COh ( ⁇ , ⁇ ) for the damping noncoherent Sig ⁇ nalanmaschine.
  • the arithmetic unit 17 uses a coherence model 18 for this purpose.
  • This integrated Konos ⁇ ence model 18 considered shadowing by the head of a user in relation to the coherence of the background noise ⁇ field.
  • a coherent model used, as suggested in the article "Binaural dereverberation based on a dual-channel Wiener filter with optimized noise field cohe- rence" by M. Jeub and P. Vary.
  • the verbes ⁇ serte model refers to the coherence of the Störschall ⁇ field in contrast to an ideal, diffuse Störschallfeld without Kopfabschattung.
  • the coherence model 18 can be based on that of [Dörbecker 1998].
  • the weights G COh ( ⁇ , ⁇ ) obtained by the arithmetic unit 17 are multiplied by the signal S j ( ⁇ , ⁇ ) to obtain an equalized output signal ( ⁇ , ⁇ ) in the left channel, and the signal S r ( ⁇ , ⁇ ) of the right channel multiplied ⁇ sheet, to obtain a enthalltes signal S r ( ⁇ , ⁇ ) in the right channel.
  • the multiplication units 19 and 20 are provided.
  • the second processing stage II can take place before the first processing stage I.
  • the independent processing stages I and II can also be interwoven. Then a two-stage is not obvious.
  • Estimate unit 12 performed. Rather, a correlation of the spectral coefficients is then used for the determina ⁇ tion of the energy of the late reverberation.
  • the reverberation time is also not estimated but fixed. A compromise is found for different acoustic conditions. By specifying the value for the reverberation time can be significantly reduced computational effort at the disadvantage of less efficient reverberation reduction.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)

Abstract

L'invention concerne un procédé plus efficace pour réduire la réverbération dans des systèmes auditifs binauriculaires. Pour cela, on a développé pour un dispositif auditif binauriculaire un procédé permettant d'acquérir un signal de sortie binauriculaire exempt de réverbération (Šl (λ, μ), Šr (λ, μ). En premier lieu, on prépare un signal d'entrée gauche (X1 (λ, μ) ) et un signal d'entrée droit (Xr (λ, μ) ). Les deux signaux d'entrée sont combinés pour former un signal de référence (Xref (λ, μ) ). On détermine des pondérations spectrales (Glate (λ, μ) ) à partir de ce signal de référence ou on établit ces pondérations d'une autre façon afin de réduire avec leur aide une réverbération retardée. Pour cela, les pondérations spectrales sont appliquées aux deux signaux d'entrée. Par ailleurs, on détermine une cohérence (17) de parties des signaux d'entrée pondérés (Šl (λ, μ), Šr (λ, μ) ). Ensuite, les parties de signaux non cohérentes des deux signaux d'entrée pondérés sont atténuées afin de réduire une réverbération avancée.
EP10739560A 2010-03-10 2010-07-27 Suppression de la réverbération de signaux d'un dispositif auditif biauriculaire Withdrawn EP2545717A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP10739560A EP2545717A1 (fr) 2010-03-10 2010-07-27 Suppression de la réverbération de signaux d'un dispositif auditif biauriculaire

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
EP10156082 2010-03-10
EP10739560A EP2545717A1 (fr) 2010-03-10 2010-07-27 Suppression de la réverbération de signaux d'un dispositif auditif biauriculaire
PCT/EP2010/060849 WO2011110239A1 (fr) 2010-03-10 2010-07-27 Suppression de la réverbération de signaux d'un dispositif auditif biauriculaire

Publications (1)

Publication Number Publication Date
EP2545717A1 true EP2545717A1 (fr) 2013-01-16

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EP (1) EP2545717A1 (fr)
WO (1) WO2011110239A1 (fr)

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US9635474B2 (en) 2011-05-23 2017-04-25 Sonova Ag Method of processing a signal in a hearing instrument, and hearing instrument
CN102750956B (zh) * 2012-06-18 2014-07-16 歌尔声学股份有限公司 一种单通道语音去混响的方法和装置
US9060052B2 (en) 2013-03-13 2015-06-16 Accusonus S.A. Single channel, binaural and multi-channel dereverberation
WO2014198332A1 (fr) 2013-06-14 2014-12-18 Widex A/S Procede de traitement de signal dans un systeme d'aide auditive et systeme d'aide auditive
CN106688247A (zh) * 2014-09-26 2017-05-17 Med-El电气医疗器械有限公司 确定房间混响以用于信号增强
DE102017200597B4 (de) * 2017-01-16 2020-03-26 Sivantos Pte. Ltd. Verfahren zum Betrieb eines Hörsystems und Hörsystem
DE102018210143A1 (de) * 2018-06-21 2019-12-24 Sivantos Pte. Ltd. Verfahren zur Unterdrückung eines akustischen Nachhalls in einem Audiosignal
AU2019321519B2 (en) * 2018-08-13 2022-06-02 Med-El Elektromedizinische Geraete Gmbh Dual-microphone methods for reverberation mitigation

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US20120328112A1 (en) 2012-12-27
WO2011110239A1 (fr) 2011-09-15

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