EP4109929A1 - Procédé de traitement directionnel du signal des signaux d'un dispositif microphone - Google Patents

Procédé de traitement directionnel du signal des signaux d'un dispositif microphone Download PDF

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Publication number
EP4109929A1
EP4109929A1 EP22178700.5A EP22178700A EP4109929A1 EP 4109929 A1 EP4109929 A1 EP 4109929A1 EP 22178700 A EP22178700 A EP 22178700A EP 4109929 A1 EP4109929 A1 EP 4109929A1
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EP
European Patent Office
Prior art keywords
signal
frequency
input signal
directional
space
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Application number
EP22178700.5A
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German (de)
English (en)
Inventor
Henning Puder
Eghart Fischer
Tobias Daniel Rosenkranz
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Sivantos Pte Ltd
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Sivantos Pte Ltd
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Publication of EP4109929A1 publication Critical patent/EP4109929A1/fr
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • 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
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/405—Arrangements for obtaining a desired directivity characteristic by combining a plurality of transducers
    • 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
    • H04R25/35—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using translation techniques
    • H04R25/353—Frequency, e.g. frequency shift or compression
    • 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
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • 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
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/407—Circuits for combining signals of a plurality of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic

Definitions

  • the invention relates to a method for the directional signal processing of signals from a microphone arrangement, which comprises at least a first microphone for generating a first input signal from ambient sound and a second microphone for generating a second input signal from ambient sound, the first input signal and the second input signal each being in the frequency domain are transformed, and a first frequency domain directional signal is formed on the basis of the first input signal thus transformed and the second input signal thus transformed in the frequency domain, frequency-dependent first amplification factors being generated in the frequency domain derived signal, the first amplification factors and the frequency domain being used on the basis of the first amplification factors - Directional signal an output signal is generated.
  • Frequency-resolved signal processing for a microphone arrangement with one or more microphones is usually carried out in the frequency domain by dividing one or more microphone signals of the microphone arrangement into individual frequency bands, with signal components of the individual frequency bands being processed separately from one another, and in particular being amplified differently and /or compressed and possibly combined to form directional signals.
  • the individual signal components in the frequency bands are then "synthesized" into a single output signal in the time domain.
  • the decomposition into individual frequency bands leads to a latency of the signal processing, which depends on the desired frequency resolution in the frequency domain, and is usually around 5-8 ms.
  • Comb filter effects are often perceived as unpleasant, which is why they should preferably be avoided.
  • One way to do this is to noticeably reduce the latency in signal processing or, if possible, avoid it entirely.
  • this significantly limits the possibilities for frequency-resolved signal processing, since an auxiliary signal path ("analysis path") often has to be used for this purpose, in which the signal processing is to be determined as it is to be applied to the signal processing of the signal components of the main signal path.
  • direction-dependent signal processing of the individual input signals which are generated by a number of microphones in the microphone arrangement, can only be carried out with difficulty in said scenario.
  • the invention is therefore based on the object of specifying a method for directional signal processing of signals from a microphone arrangement, which method should have the lowest possible latency of a generated output signal in relation to the input signals of the microphone arrangement to be processed.
  • a method for directional signal processing of signals from a microphone arrangement which has at least a first microphone for generating a first input signal from an ambient sound and a second microphone for generating a second Includes input signals from the ambient sound, with a reference signal being formed on the basis of the first input signal, with the reference signal being transformed into the frequency domain, and as a result a frequency-space reference signal being generated, with the first input signal and the second input signal each being transformed into the frequency domain, and a first frequency domain directional signal is formed on the basis of the first input signal transformed in this way and the second input signal transformed in this way in the frequency domain.
  • a microphone is intended to include any electroacoustic input transducer that is set up to generate a corresponding input signal from the ambient sound, with sound pressure fluctuations of the ambient sound being translated into corresponding voltage or current fluctuations by the input transducer.
  • a microphone arrangement is to be understood accordingly as any spatial arrangement of at least two such input transducers, with the spatial distance between the input transducers enabling direction-dependent signal processing.
  • the formation of the reference signal based on the first input signal includes in particular the two cases that on the one hand the reference signal is formed only based on the signal components of the first input signal, but other signal components, in particular those of the second input signal, not in the reference signal received, or that on the other hand the reference signal can be formed on the basis of the signal components of both input signals, in particular by means of a time-delayed superimposition of the two input signals.
  • the reference signal is a signal in the time domain.
  • the aforementioned transformations of a signal into the frequency domain can in this case be carried out in particular by means of a correspondingly set up filter bank.
  • the frequency domain also includes in particular the time-discrete time-frequency domain in which the spectral components of the individual transformed signals are updated according to a time variable.
  • the first frequency space directional signal can be generated using the transformed first input signal and the transformed second input signal in such a way that the resulting directional characteristic for the first frequency space directional signal varies across the individual frequency bands (and that, for example, a linear combination of the two transformed input signals or a linear combination of intermediate signals derived from the two input signals, such as cardioid/anticardioid signals, has different linear factors across the frequency bands).
  • the frequency space reference signal is frequency-resolved, ie preferably compared by frequency band with the first frequency space directional signal or a signal derived therefrom in the frequency domain, so that the frequency-dependent first amplification factors are generated as a quantitative result of this comparison.
  • This comparison can be done, for example, by spectral division (the first amplification factors can then be determined directly for each frequency band as the respective quotients that are formed when the frequency space reference signal is divided by the first frequency space directional signal).
  • Other comparisons for example in the form of a relative deviation and/or in the form of a non-linear, preferably monotonic function of the frequency-band-wise deviation of the two signals mentioned from one another just as conceivable.
  • a second frequency space directional signal is preferably used, the signal components of which can be derived directly from the first frequency space directional signal and preferably without the addition of further signal components, i.e. for example via amplifying the first frequency domain directional signal frequency band by frequency.
  • the first amplification factors which are a measure of the deviation of the first frequency-space directional signal from the frequency-space reference signal in the respective frequency band (possibly implicit in the case of a derived signal), are now used to generate a time filter in the time domain.
  • the time filter is preferably a filter with a finite impulse response (Finite Impulse Response, FIR).
  • FIR Finite Impulse Response
  • the time filter can be generated here by a transformation of a transfer function from the frequency domain into the time domain, which represents the application of the first amplification factors in the frequency domain for each frequency band.
  • the time filter is particularly preferably generated as a minimum-phase filter, which therefore has a minimum possible latency for a given magnitude frequency response.
  • the reference signal is filtered—in the time domain—and an output signal is generated.
  • the output signal can now in particular be transmitted to a receiver, recorded or used further.
  • the output signal is preferably converted into an output sound signal by a loudspeaker (in the broadest sense, an electroacoustic output converter of the hearing instrument). Before said conversion, the output signal can be subjected to further signal processing, for example to suppress acoustic feedback which can occur between the loudspeaker and the microphone arrangement.
  • the use of a time filter, which is applied to the reference signal in the time domain, results in a significant reduction in the latency of the signal processing compared to the signal processing in the frequency domain. Due to the mapping of the first amplification factors, which are obtained using the first frequency space directional signal, to the time filter, the information regarding the direction-dependent sound image in the individual frequency bands can be implemented in the time filter, as is the case with the generation of the first Frequency space directional signal results. In this case, however, the differences are taken into account which the directional signal processing on the first frequency domain signal compared to the reference signal--transformed into the frequency domain--has.
  • the frequency space reference signal Based on the comparison of the frequency space reference signal with the first frequency space directional signal or with a signal preferably derived directly from it, it is determined which frequency band-wise amplification factors are to be applied to the frequency space reference signal (which actually only represents the reference signal transformed into the frequency domain), in order to approximate as well as possible the frequency band-wise sound properties, in particular with regard to signal volumes and levels etc., which are present for the first frequency space directional signal.
  • an application of the time filter, in particular an FIR filter, in the time domain corresponds to an application of frequency- and time-dependent attenuation or amplification factors in each frequency band with a subsequent signal synthesis.
  • the frequency-band-wise directional dependency coded in the first frequency-space directional signal is “coded” into the first amplification factors for the frequency-space reference signal.
  • the frequency-resolved comparison of the frequency-space reference signal with the first frequency-space directional signal or the signal derived from the first frequency-space directional signal in the frequency domain is preferably carried out on the basis a spectral division, which is used to generate the frequency-dependent first amplification factors.
  • the comparison by means of a spectral division can be implemented particularly efficiently and, on the other hand, it takes into account the fact that the first amplification factors thus represent a transfer function between the two signals mentioned in the frequency domain.
  • the frequency-dependent first amplification factors can be generated in particular by dividing the amounts of the frequency-space reference signal and the first frequency-space directional signal or the signal derived from the first frequency-space directional signal in the frequency domain.
  • a second frequency space directional signal is advantageously generated as a signal derived from the first frequency space directional signal in the frequency domain for the frequency-resolved comparison with the frequency space reference signal, which second frequency space directional signal is generated on the basis of the application of frequency-dependent second amplification factors to the first frequency space directional signal.
  • the first frequency space directional signal can also be subjected to noise suppression, for example by means of a Wiener filter, and dynamic compression if necessary, so that the frequency space reference signal for the first gain factors is compared with the resulting second frequency space directional signal.
  • the time filter is expediently formed using a mapping of the frequency-dependent first amplification factors into the time domain.
  • the term mapping can mean in particular that a transfer function corresponding to the first amplification factors or a transfer function which corresponds to a product of the first amplification factors with further amplification factors per frequency band is transformed from the frequency domain into the time domain.
  • the time filter is generated in particular as an FIR filter.
  • frequency-dependent second amplification factors are determined for the first frequency-space directional signal, with the time filter being formed using a joint mapping of the first amplification factors and the second amplification factors in the time domain.
  • the frequency-dependent second amplification factors for the first frequency space directional signal are determined using noise suppression and/or dynamic compression and/or a hearing impairment to be corrected in a receiver of the output signal.
  • the recipient of the output signal is in particular a user of a hearing instrument which includes the microphone arrangement.
  • Noise suppression and/or dynamic compression which is/is to be applied to the first frequency space directional signal depending on the frequency band, is determined according to the respective specifications for a signal-to-noise ratio ("signal-to-noise -ratio", SNR) or the specifications for maximum signal levels in the frequency bands (which can also be individually tailored to a hearing impairment of the user of the hearing instrument that includes the microphone arrangement) instantaneous second amplification factors for the first frequency space directional signal.
  • signal-to-noise -ratio SNR
  • maximum signal levels in the frequency bands which can also be individually tailored to a hearing impairment of the user of the hearing instrument that includes the microphone arrangement
  • instantaneous second amplification factors for the first frequency space directional signal.
  • the properties of the noise suppression or dynamic compression are therefore not included in the time filter via the comparison of the transformed reference signal with a second frequency space directional signal, which is generated by the application of noise suppression or dynamic compression on the first frequency space directional signal results, but directly via a mapping of the second amplification factors determined for the noise reduction or the dynamic compression in the time domain.
  • the reference signal is only formed from signal components of the first input signal. This means in particular that about the signal components of the first input signal do not include any signal components of other signals in the reference signal, and that the first input signal is preferably used as a reference signal either directly or after a single-channel signal processing.
  • the effect of the directional microphony thus results in the formation of the first frequency-space directional signal (which indeed contains the frequency-space reference signal, ie the transformed first input signal in the frequency domain) in the frequency domain, which is to be compared with the transformed reference signal.
  • the resulting first amplification factors then carry the full spectral information on how the directional microphony affects the transformed reference signal.
  • a time filter created by mapping these first gain factors into the time domain, then transfers this spectral information to the reference signal.
  • the reference signal is formed as a time directional signal based on the first input signal and the second input signal using directional microphones in the time domain.
  • the directional microphony in the time domain can be implemented here in particular as a time-delayed and possibly differently weighted superimposition of the two input signals (the time delay being implemented in the time domain and in particular being the same for all spectral components of the two input signals).
  • the directional effect can be increased overall, with broadband and/or dominant interference noise, for example, which are strongly localized, being able to be removed in the time-directional signal.
  • a frequency band fine-tuning of the directional microphone then takes place based on a comparison of the transformed time directional signal (ie the frequency space reference signal) with the first frequency space directional signal via the resulting time filter.
  • the microphone arrangement for carrying out the method preferably further comprises a third microphone for generating a third input signal from the ambient sound, the third input signal being transformed into the frequency domain, and the first frequency space directional signal also being formed using the third input signal transformed in this way in the frequency domain.
  • the microphone arrangement can correspondingly include a fourth or additional microphones. The method described can easily be transferred to such a microphone arrangement with three or more microphones.
  • the invention also specifies a method for directional signal processing in a hearing instrument, the hearing instrument comprising a microphone arrangement with at least a first microphone for generating a first input signal from ambient sound and a second microphone for generating a second input signal from ambient sound and also a control unit, and wherein an output signal of the hearing instrument intended for playback is generated on the basis of the first input signal and the second input signal according to the method described above.
  • the invention also specifies a hearing instrument with a microphone arrangement which comprises at least a first microphone for generating a first input signal from an ambient sound and a second microphone for generating a second input signal from the ambient sound, and further with a control unit, the control unit being set up to to carry out the above-mentioned method on the basis of the first and the second input signal.
  • the hearing instrument shares the advantages of the method for directional signal processing of signals from a microphone array. The advantages that are specified for said method and for its developments can be transferred to the hearing instrument.
  • the hearing instrument can be designed as a hearing device which is provided and set up to treat a hearing impairment.
  • a hearing instrument 1 is shown schematically in a block diagram, which includes a microphone arrangement 2 with a first microphone 4 and a second microphone 6 .
  • the signal processing of the in figure 1 The hearing instrument 1 shown is designed according to the prior art.
  • the first microphone 4 is set up to generate a first input signal 8 from an ambient sound 7 which impinges on the microphone arrangement 2 .
  • the second microphone 6 is set up to generate a second input signal 10 from the ambient sound 7 .
  • a possible pre-amplification and/or digitization of the first or second input signal 8, 10 is already included in the corresponding first or second microphone 4, 6.
  • the first and the second input signal 8, 10 are each supplied to a first filter bank 12 and are each transformed there into the frequency domain. so that a transformed first input signal 14 and a transformed second input signal 16 are generated.
  • a first frequency space directional signal 20 is formed from the transformed first input signal 14 and the transformed second input signal 16 in the frequency domain.
  • Any algorithm suitable for forming a frequency-band-wise directional signal can be used in the directional microphone module 18 to form the first frequency-space directional signal 20, ie in particular delay-and-sum-beamforming, delay-and-subtract-beamforming, adaptive differential directional microphony or the like
  • the first frequency space directional signal 20 is subjected to noise suppression 22, in which a useful signal component and an interference signal component are estimated for the individual frequency bands in particular, and an amplification factor is determined for each frequency band as a function of the said useful signal or interference signal components, so that frequency bands with increased relatively with a high useful signal component and lowered relatively frequency bands with a high interference signal component.
  • an amplification module 24 which can include, in particular, an AGC for dynamic compression of the frequency band-by-frequency signal components, and can also compensate for an individual hearing impairment of a user of the hearing instrument 1 via a corresponding adjustment of frequency-band-by-frequency band gain factors.
  • a processed second frequency-space directional signal 26 results from the amplification module 24.
  • This second frequency-space directional signal 26 is fed to a synthesis filter bank 28, which combines the frequency-band-wise signal components of the second frequency-space directional signal 26 and transforms them into the time domain.
  • the output signal 30 resulting from said transformation is converted into an output sound signal 34 by a loudspeaker 32 of the hearing instrument 1 .
  • the signal processing described here from the first filter bank 12 to the synthesis filter bank 28 preferably takes place on a corresponding equipped signal processor or a processor unit comprising such a signal processor.
  • Such a processor unit is in figure 1 shown schematically as a control unit 35.
  • the output signal 30 Due to the first filter bank 12 and the synthesis filter bank 28, the output signal 30 inevitably experiences a latency in relation to the two input signals 8, 10 and thus in relation to the ambient sound 7, which is all the greater the higher the frequency resolution of the first filter bank 12 is.
  • This latency is usually around 4 to 7 ms.
  • a significant reduction in the latency can be achieved by reducing the frequency resolution, but this is also at the expense of the possibilities of beamforming and interference signal components in the two input signals 8, 10 by means of the noise suppression 22, and possibly the amplification in the amplification module 24 individually to suit the user of the hearing instrument 1.
  • FIG 2 1 is a schematic block diagram of a modification of the hearing instrument 1 figure 1 1, which attempts to solve the described problems of latency that arise as a result of the first filter bank 12 and the synthesis filter bank 28. Also the in figure 2 The signal processing shown is designed according to the prior art.
  • a time directional signal 38 is formed from the first input signal 8 and the second input signal 10 by means of a time domain directional microphone module 36 .
  • the time direction signal 38 is transformed by the first filter bank 12 into the frequency domain, and a transformed time direction signal 40 resulting therefrom is fed to the noise suppression 22 and then to the amplification module 24 .
  • amplification factors gj are determined for each frequency band, which are mapped onto a time filter 44 in the time domain by means of a mapping 42, which can in particular include a Fourier transformation.
  • the control unit 35 after figure 1 is not shown in FIG.
  • the time filter 44 thus implicitly contains the properties of the frequency band-wise amplification factors gj and their effects on the transformed time directional signal 40, but now in the time domain. Accordingly, the time filter 44 is applied to the (original) time direction signal 38 in the time domain in order to generate the output signal 30 therefrom.
  • the temporal filter 44 is determined to be a minimum phase filter.
  • a hearing instrument 1 is shown schematically in a block diagram, in which the latency is to be kept as low as possible for as frequency-selective, direction-dependent signal processing as possible.
  • the first input signal 8 is used as a reference signal 46.
  • the effects of frequency-selective directional microphones and noise suppression on the individual frequency bands are determined using the reference signal 46 in a manner to be described below, and the time filter 44 to be applied to the reference signal 46 is thus determined.
  • the first input signal 8 as the reference signal 46 and the second input signal 10 are similar to that based on FIG figure 1 illustrated embodiment, transformed by the first filter bank 12 into the frequency domain, and thereby the transformed first input signal 14 as a frequency space reference signal 48 or the transformed second input signal 16 is generated.
  • the first frequency-space directional signal 20 is then generated by the directional microphone module 18 from the transformed first input signal 14--that is, the frequency-space reference signal 48--and from the transformed second input signal.
  • first amplification factors are used for each frequency band g1j obtained, which are determined by a spectral division 45 of the frequency space reference signal 48 and a second frequency space directional signal 50 derived from the first frequency space directional signal 20 .
  • the magnitudes of the frequency space reference signal 48 and of the second frequency space directional signal 50 derived therefrom, or variables derived from the magnitudes can also be divided by frequency band in order to generate the first amplification factors g1j.
  • Said second frequency-space directional signal 50 is generated by feeding the first frequency-space directional signal 20 to the noise suppressor 22 and the amplification module 24, where the first frequency-space directional signal 20 is amplified or reduced by frequency band by frequency-band-wise second g2j to the first frequency-domain directional signal 20 be applied.
  • the second amplification factors g2j can, for example, be formed in each frequency band from the successive application of the individual factors which were determined in the noise suppression 22 and in the amplification module 24 for the respective frequency band.
  • the spectral division 45 is de facto determined in which way the signal processing applied to the first frequency space directional signal 20, which takes place in the noise suppression 22 and in the amplification module 24, is to be modified or compensated if the input variable is not the said first frequency space Directional signal 20, but instead the frequency space reference signal 48. If the first amplification factors g1j resulting from the spectral division 45 were applied to the frequency space reference signal 48, the resulting signal would correspond in magnitude to the second frequency space directional signal 50, which is derived from the Application of the noise suppression 22 and the amplification module 24 (or from the second amplification factors g2j determined there) to the first frequency space directional signal 20 results.
  • the first amplification factors g1j resulting from the spectral division 45 are now transferred from the frequency domain to the time filter by means of the mapping 42 44 imaged in the time domain, which is preferably provided by an FIR filter.
  • the time filter 44 is thus the equivalent in the time domain to the "modification” or “compensation” of the signal processing of the first frequency-space directional signal 20 just described, which must be applied to the frequency-space reference signal 46 .
  • the influence of the transformed input signal 16 on the second frequency space directional signal 50 also enters the time filter 44 via the spectral division 45 . Accordingly, the output signal is generated from an application of the temporal filter 44 to the frequency domain directional signal 48 .
  • the latency can be kept very low by the time filter 46 in the time domain, since latencies, which arise, for example, as a result of the first filter bank 12, do not affect the propagation of the reference signal 46 through the signal flow, but only lead to the time -Filter 44, which is applied to the reference signal 46, is no longer "up-to-date" by the amount of latency, but this is a trade-off compared to the significantly reduced latency of the output signal 30 compared to the exemplary embodiment figure 1 can be accepted.
  • figure 4 is schematically in a block diagram an alternative embodiment of the basis of figure 3 described signal processing shown, which also elements of the embodiment figure 2 by applying the time filter 44 to a directional signal in the time domain in a manner to be illustrated.
  • a time directional signal 38 is first generated from the first input signal 8 and the second input signal 10 by means of the time domain directional microphone module 36 . This can be done, for example, by delaying one of the two input signals 8, 10 in relation to the other, which can vary over time but always has the same effect on all signal components (and therefore acts in particular independently of frequency).
  • the time directional signal 38 can also be generated in that in the time-domain directional microphone module 36 on one of the two input signals 8, 10 an all-pass filter with a frequency-dependent Apply delay, so that the time-directional signal 38 itself can already have a certain frequency dependence in terms of directivity.
  • the first and the second input signal 8, 10 are also transformed into the frequency domain by means of the first filter bank 12, and the first frequency-space directional signal 20 is generated from the transformed first and second input signal 14, 16 generated in this way by means of the directional microphone module 18 in the frequency domain .
  • the time direction signal 36 generated as described above is used in the present exemplary embodiment as the reference signal 46, which is transformed into the frequency domain by means of a second filter bank 52, as a result of which the transformed time direction signal 40 is generated as a frequency space reference signal 48.
  • This and the first frequency space directional signal 20 are subjected to spectral division 45 for comparison with one another, as a result of which the first amplification factors g1j are determined for the respective frequency bands.
  • the noise suppressor 22 and the amplifier module 24 determine second amplification factors g2j, which would have to be applied accordingly to the first frequency space directional signal 20 in order to increase the noise suppression effect of the noise suppressor 22 or the amplifying effect of the amplifying module 24 for the first frequency space directional signal 20 to achieve.
  • this noise suppression effect or amplification effect is not achieved directly with the first frequency space directional signal 20 .
  • the second amplification factors g2j corresponding to the said effects which were determined in the noise suppressor 22 and in the amplifier module 24, are now obtained together with the first amplification factors g1j, which are obtained from the spectral division 45 of the frequency space reference signal 48 and the first frequency space directional signal 20 were, through the Figure 42 into the time domain on the time filter 44 pictured.
  • the time filter 44 determined in this way which is also preferably designed as an FIR filter here, is then applied to the reference signal 46 in the time domain—ie to the time direction signal 38—and the output signal 30 is thereby generated. Finally, the output signal 30 is converted into the output sound signal 34 by the loudspeaker 32 .
  • the spectral division 45 is in the embodiment figure 4 determines the extent to which the frequency-selective beamforming of the directional microphone module 18 (frequency domain) differs from the broadband beamforming of the time-domain directional microphone module 36, so that the first gain factors g1j generated in this way de facto represent the amount in the instantaneous gain by which the transformed time - Directional signal 40 is to be compensated for in frequency bands in order to obtain the intrinsic, direction-sensitive sound behavior that is inherent in the first frequency space directional signal 20 .
  • This directionally sensitive sound behavior characterized by the first amplification factors g1j is therefore mapped onto the time filter 44 in the time domain, together with the noise suppression and amplification effect of the noise suppression 22 and the amplification module 24 characterized by the second amplification factors g2j, so that said sound behavior and said effects by applying the time filter 44 to the time-domain equivalent of the transformed time-direction signal 40, i.e. precisely to the time-direction signal 38.
  • the latency of the output signal 30 relative to the two input signals 8, 10 and thus relative to the ambient sound 7 can also be reduced in this exemplary embodiment compared to the exemplary embodiment figure 1 be kept very low.

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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Neurosurgery (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP22178700.5A 2021-06-25 2022-06-13 Procédé de traitement directionnel du signal des signaux d'un dispositif microphone Pending EP4109929A1 (fr)

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DE102021206590.2A DE102021206590A1 (de) 2021-06-25 2021-06-25 Verfahren zur direktionalen Signalverarbeitung von Signalen einer Mikrofonanordnung

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EP4109929A1 true EP4109929A1 (fr) 2022-12-28

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Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE102014204557A1 (de) * 2014-03-12 2015-09-17 Siemens Medical Instruments Pte. Ltd. Übertragung eines windreduzierten Signals mit verminderter Latenzzeit
WO2021110924A1 (fr) * 2019-12-04 2021-06-10 Widex A/S Prothèse auditive et procédé de fonctionnement de prothèse auditive

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN103907152B (zh) * 2011-09-02 2016-05-11 Gn奈康有限公司 用于音频信号噪声抑制的方法和系统

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE102014204557A1 (de) * 2014-03-12 2015-09-17 Siemens Medical Instruments Pte. Ltd. Übertragung eines windreduzierten Signals mit verminderter Latenzzeit
WO2021110924A1 (fr) * 2019-12-04 2021-06-10 Widex A/S Prothèse auditive et procédé de fonctionnement de prothèse auditive

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US12047747B2 (en) 2024-07-23
CN115529532A (zh) 2022-12-27
US20230007408A1 (en) 2023-01-05

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