EP3225037B1 - Procédé et appareil de génération d'un signal sonore directionnel à partir de premier et deuxième signaux sonores - Google Patents

Procédé et appareil de génération d'un signal sonore directionnel à partir de premier et deuxième signaux sonores Download PDF

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EP3225037B1
EP3225037B1 EP14771598.1A EP14771598A EP3225037B1 EP 3225037 B1 EP3225037 B1 EP 3225037B1 EP 14771598 A EP14771598 A EP 14771598A EP 3225037 B1 EP3225037 B1 EP 3225037B1
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frequency
signal
directional
deq
sound
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EP3225037A1 (fr
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Hauke Krüger
Bernd Geiser
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Binauric Se
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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
    • 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
    • H04R2430/25—Array processing for suppression of unwanted side-lobes in directivity characteristics, e.g. a blocking matrix

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  • the present invention generally relates to the field of sound acquisition. More particularly, the present invention relates to a method and an apparatus for generating a directional sound signal from first and second sound signals, which are generated by a first and a second microphone, which are separated by a distance.
  • microphone arrays proved to be useful. They are designed to attenuate possible noise and interference components while retaining the desired source signal by exploiting different spatial (or directional) characteristics of the different signal sources (see, e.g., J. Benesty, J. Chen, and Y. Huang, "Microphone Array Signal Processing,” Heidelberg: Springer, 2008 for an overview).
  • a simple, yet efficient approach is the first-order differential microphone array described in G. Elko and A.-T. N. Pong, "A simple adaptive first-order differential microphone,” in IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), pages 169 to 172, October 1995 .
  • This microphone array which is schematically and exemplarily shown in Fig. 1 allows to place two symmetrical notches (directions of maximum attenuation) at angles of ⁇ and 360° - ⁇ .
  • three independent enhancements to the original method are proposed and a practical implementation for handsfree communication is described.
  • a possible target device is a wireless loudspeaker with two integrated miniature digital micro-electromechanical system (MEMS) microphone capsules which facilitate handsfree audio coomunication.
  • MEMS micro-electromechanical system
  • Fig. 1 shows schematically and exemplarily a differential microphone array according to G. Elko and A.-T. N. Pong.
  • Two closely spaced omnidirectional microphones M1 and M2 are used to capture the acoustic environment.
  • the corresponding digital signals x 1 ( k ) and x 2 ( k ) are sampled with a rate of f s .
  • Due to the small distance D between M1 and M2, a coherent mutual subtraction - for convenience, acausal filters are assumed herein; in practice, appropriate signal alignment is required as marked by 'o' in Figs.
  • the signals x f ( k ) and x b ( k ) can be interpreted as "forward and backward facing cardioid" signals as the respective directional responses of Eqs. (1) and (2) form cardioid shapes (see Fig.
  • a common problem with differential microphone arrays are the tolerances of the employed microphones, leading to a “microphone mismatch” and therefore noise amplification (see M. Buck and M. R michler, "First Order Differential Microphone Arrays for Automotive Applications,” in Proceedings of International Workshop on Acoustic Echo and Noise Control (IWAENC), September 2001 ).
  • the digital MEMS microphones in the possible target device usually exhibit relatively constant frequency responses; therefore, individual microphone equalization is preferably not necessary for the envisaged application.
  • their power levels may still vary to a certain extent due to mounting and assembly tolerances, which is disadvantageous since it is preferred to have fully matched input levels in order to utilize the full potential of the method.
  • a method for generating a directional sound signal from first and second sound signals, which are generated by a first and a second microphone, which are separated by a distance comprises:
  • the present invention is based on the idea that by employing these steps, a (substantially) frequency invariant notch characteristic can be obtained even for larger microphone distances.
  • a larger distance also helps to confine the noise gain of the array. Therefore, the array becomes practically usable even for higher sampling rates (e.g., 16kHz).
  • difference signal also includes the case where one or both of the first and the second sound signals is/are further temporally delayed, for example, by means of a fractional delay filter h T ( k ), as described in section 2 above.
  • the frequency-selective processing comprises weighting the difference signal with an approximated steering factor that is independent of frequency to generate a weighted difference signal and correcting for the approximation by adding a correction signal that is generated from the difference signal in dependence of frequency and the steering angle.
  • the generation of the correction signal comprises applying two separate operations, one being dependent on frequency and independent of the steering angle and one being dependent on the steering angle but independent of frequency.
  • the generation of the correction signal comprises filtering the difference signal with a filter that is dependent on frequency and independent of the steering angle to generate a filtered difference signal.
  • the generation of the correction signal further comprises weighting the filtered difference signal with a factor that is dependent on the steering angle and independent of frequency.
  • the factor is determined by using a polynomial approximation that is evaluated with the steering angle.
  • the method further comprises filtering the directional sound signal with a low-pass filter to generate a filtered directional sound signal.
  • the approximated steering factor for a time instance is adapted for the following time instance by adding an adaptation value that is scaled by a stepsize parameter, wherein the stepsize parameter is adapted in dependence of estimated energies of coherent and incoherent sound components.
  • the energy of the incoherent sound components is approximated by the estimated short-term energy of the directional sound signal and the energy of the coherent sound components is approximated by a fraction of the estimated short-term energy of the difference signal.
  • the method further comprises estimating a relative gain of the first and the second microphone and equalizing power levels of the first and the second microphone based on the relative gain.
  • the relative gain is determined based on recursively estimated variances of the first and second sound signals.
  • the first and the second microphone are omnidirectional microphones.
  • an apparatus for generating a directional sound signal from first and second sound signals, which are generated by a first and a second microphone, which are separated by a distance comprising:
  • a system comprising:
  • this transfer function should become zero for a specific angle, i.e., the so-called steering angle ⁇ .
  • ⁇ a ( ⁇ , ⁇ ) is separable with good accuracy: ⁇ ⁇ a ⁇ ⁇ ⁇ ⁇ ⁇ a ⁇ ⁇ ⁇ ⁇ a ⁇ ⁇ ⁇ ⁇ a ⁇
  • the factors ⁇ a ( ⁇ ) and ⁇ a ( ⁇ ) can be computed by marginalization of the 2-dimensional function ⁇ a ( ⁇ , ⁇ ) and appropriate normalization.
  • the factor ⁇ a ( ⁇ ) can now be regarded as the frequency response of a fixed filter. It can be transformed to the time domain via periodic extension, inverse DFT, cyclic shifting (to enforce causality) and an appropriate shortening to a desired length.
  • the resulting FIR filter coefficients h DEQ ( k ), e.g., of order 16, are independent of the steering angle ⁇ .
  • the angular dependency is then reintroduced with a polynomial approximation (e.g., order 4) of the second factor ⁇ a ( ⁇ ) after a variable transformation from ⁇ to a lin ( ⁇ ), i.e., P ⁇ a ( a lin ( ⁇ )) ⁇ ⁇ a ( ⁇ ( a lin )).
  • the effect of directional equalizing can be observed in Fig. 3 (b) , which displays an almost frequency-invari
  • the practical operation of the modified version of the microphone array does not significantly differ from the conventional version ( Fig. 1 ):
  • the desired notch angle ⁇ is still easily controlled by adapting the scalar factor a lin .
  • the polynomial P ⁇ a ( a lin ) must be evaluated.
  • the distorted notch curve of the standard differential array ( Fig. 3 (a) ) not only limits the ability to suppress interfering sound sources, but it can even compromise the accurate NLMS adaptation of the steering angle ⁇ (see section 6).
  • the goal of a notch adaptation algorithm is to automatically align the notch angle ⁇ of the differential array with the incidence angle ⁇ of the (main) interferer.
  • the standard approach to adapt the factor a (or a lin if directional equalization is used) and therefore the notch angle ⁇ is the ( normalized ) least mean square (NLMS) algorithm.
  • the goal here is to minimize the power of the output signal y ( k ), i.e.
  • This equation represents the error signal of a single-tap adaptive filter with a noisy input.
  • the noise signal n(k) is due to the incoherent (ambient) noise that cannot be suppressed.
  • the coherent contribution to y ( k ) should ideally be zero.
  • an error signal e ( k ) appears at the output.
  • E ⁇ n 2 ( k ) ⁇ is the level ⁇ ⁇ y 2 of the microphone array's output y ( k ) while for E ⁇ e 2 ( k ) ⁇ , the assumption of a fixed attenuation factor for the backward cardioid signal is made, i.e. E e 2 k ⁇ ⁇ ⁇ ⁇ ⁇ x b 2 .
  • the adaptation can be deliberately slowed down by the factor 0 ⁇ ⁇ ⁇ 1 to avoid artifacts that stem from the single-tap prediction which does not apply any smoothing.
  • the combination of the proposed NLMS notch adaptation with the directional equalizer of section 5 is straight forward.
  • the equalizer can indirectly influence and enhance the notch adaptation via the array output signal y ( k ).
  • the performance of the proposed fast notch adaptation algorithm is contrasted with the conventional NLMS using a fixed stepsize in Fig. 5 .
  • the graph illustrates the adaptation process for a synthetic sound field with a single sound source that arrives from changing angles ⁇ .
  • ⁇ 0°
  • the adaptation should not drift towards the 90° boundary but rather maintain the previously identified steering factor a .
  • the underlying assumption is that an interferer does not move while being inactive.
  • the fast version of the constant stepsize NLMS (Eq.
  • the described differential microphone array (including the proposed enhancements) has been implemented on a signal processor of a wireless loudspeaker ( Binauric Boom Boom ) which is, at the same time, a handsfree communication device.
  • a wireless loudspeaker Binauric Boom Boom
  • the microphones offer SNRs of more than 60 dB which open up the possibility of a differential microphone array with a sufficiently low noise level.
  • An example application scenario is a handsfree call in an office where another colleague is working on the opposite side of the desk. The colleague's noise (typing, voice, etc.) can then be canceled out when placing a call with Boom Boom.
  • the signal processing software has been developed with the help of the RTProc rapid real-time prototyping framework (see H. Krüger and P. Vary) - the developer interface for algorithm parameterization is shown in Fig. 6 .
  • a Matlab prototype based on framewise processing
  • several other versions have been subsequently developed: A parameterizable C version, a C version with generated parameter tables, a C version based on fixed point arithmetic with an emulated instruction set and generated parameter tables, and finally optimized assembler code for the signal processor with generated parameter tables. All versions can be verified against each other and there is the possibility to step back to Matlab and add or modify features.
  • a single unit or device may fulfill the functions of several items recited in the claims.
  • the mere fact that certain measures are recited in mutually different dependent claims does not indicate that a combination of these measures cannot be used to advantage.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)

Claims (15)

  1. Procédé de génération d'un signal sonore directionnel (y(k)) à partir de premier et de deuxième signaux sonores (x 1(k), x 2(k)) qui sont générés par des premier et deuxième microphones (M1, M2) qui sont séparés par une distance (D), dans lequel le procédé comprend:
    - générer des premier et deuxième signaux sonores différentiels (x f(k), x b,DEQ(k)) sur la base des premier et deuxième signaux sonores (x 1(k), x 2(k)), et
    - générer le signal sonore directionnel (y(k)) selon un motif de réponse directionnelle dépendant de la fréquence, sur la base des premier et deuxième signaux sonores différentiels (x f(k), x b,DEQ(k)),
    dans lequel la génération du deuxième signal sonore différentiel (x b,DEQ(k)) comprend une génération d'un signal de différence (x b(k)) des premier et deuxième signaux sonores (x 1(k), x 2(k)) et un traitement sélectif en fréquence qui dépend d'un angle de pilotage (α) qui indique une direction désirée d'atténuation maximale du motif de réponse directionnelle dépendant de la fréquence, dans lequel le traitement sélectif en fréquence adapte la direction réelle d'atténuation maximale du motif de réponse directionnelle dépendant de la fréquence de manière à ce qu'elle corresponde à l'angle de pilotage (α) pour l'essentiel indépendant de la fréquence (ω) sur la plage de fréquences du signal sonore directionnel (y(k)).
  2. Procédé selon la revendication 1, dans lequel le traitement sélectif en fréquence comprend une pondération du signal de différence (x b(k)) avec un facteur de pilotage approximé (a lin(α)) qui est indépendant de la fréquence (ω) pour générer un signal pondéré de différence (a lin(α)·x b(k)), et une correction de l'approximation en ajoutant un signal de correction (P Δa (a lin(α))·h DEQ(k) ∗ xb (k)) qui est généré à partir du signal de différence (x b(k)) en fonction de la fréquence (ω) et de l'angle de pilotage (α).
  3. Procédé selon la revendication 2, dans lequel la génération du signal de correction (P Δa (a lin(α))·h DEQ(k) ∗ xb (k)) comprend l'application de deux opérations distinctes, une étant dépendante de la fréquence (ω) et indépendante de l'angle de pilotage (α) et une étant dépendante de l'angle de pilotage (α) mais indépendante de la fréquence (ω).
  4. Procédé selon la revendication 2 ou 3, dans lequel la génération du signal de correction (P Δa(a lin(α))·h DEQ(k) ∗ xb (k)) comprend un filtrage du signal de différence (x b(k)) avec un filtre (h DEQ(k)) qui est dépendant de la fréquence (ω) et indépendant de l'angle de pilotage (α) pour générer un signal de différence filtré (h DEQ(k) * xb (k)).
  5. Procédé selon la revendication 4, dans lequel la génération du signal de correction (P Δa (a lin(α))·h DEQ(k) ∗ xb (k)) comprend en outre une pondération du signal de différence filtré (h DEQ(k) ∗ xb (k)) avec un facteur (P Δa(a lin(α)) qui est dépendant de l'angle de pilotage (α) et indépendant de la fréquence (ω).
  6. Procédé selon la revendication 5, dans lequel le facteur (P Δa (a lin(α)) est déterminé en utilisant une approximation polynomiale qui est évaluée avec l'angle de pilotage (α) ou le facteur de pilotage approximé (a lin(α)).
  7. Procédé selon l'une quelconque des revendications 1 à 6, dans lequel le procédé comprend en outre un filtrage du signal sonore directionnel (y(k)) avec un filtre passe-bas (h EQ(k)) pour générer un signal sonore directionnel filtré (y EQ(k)).
  8. Procédé selon l'une quelconque des revendications 2 à 7, dans lequel le facteur de pilotage approximé (a lin(α)) pour un instant (k - 1) est adapté pour l'instant suivant (k) en ajoutant une valeur d'adaptation qui mis à l'échelle par un paramètre de pas (µ opt), dans lequel ledit paramètre de pas (µ opt) est adapté en fonction d'énergies estimées de composantes sonores cohérentes et incohérentes.
  9. Procédé selon la revendication 8, dans lequel l'énergie des composantes sonores incohérentes est approximée par l'énergie estimée à court terme σ ^ y 2
    Figure imgb0035
     du signal sonore directionnel (y(k)), et l'énergie des composantes sonores cohérentes est approximée par une fraction k σ ^ x b 2
    Figure imgb0036
     de l'énergie estimée à court terme σ ^ x b 2
    Figure imgb0037
     du signal de différence (x b(k)).
  10. Procédé selon l'une quelconque des revendications 1 à 9, dans lequel le procédé comprend en outre une estimation d'un gain relatif (g c) des premier et deuxième microphones (M1, M2) et une égalisation de niveaux de puissance des premier et deuxième microphones (M1, M2) sur la base du gain relatif (g c).
  11. Procédé selon la revendication 10, dans lequel le gain relatif (g c) est déterminé sur la base de variances σ ^ x 1 2 , σ ^ x 2 2
    Figure imgb0038
     estimées de manière récursive des premier et deuxième signaux sonores (x 1(k), x 2(k)).
  12. Procédé selon l'une quelconque des revendications 1 à 11, dans lequel les premier et deuxième microphones (M1, M2) sont des microphones omnidirectionnels.
  13. Dispositif (2) de génération d'un signal sonore directionnel (y(k)) à partir de premier et de deuxième signaux sonores (x 1(k), x 2(k)) qui sont générés par des premier et deuxième microphones (M1, M2) qui sont séparés par une distance (D), dans lequel le dispositif (2) comprend:
    - de premiers moyens de génération destinés à générer des premier et deuxième signaux sonores différentiels (x f(k), x b,DEQ(k)) sur la base des premier et deuxième signaux sonores (x 1(k), x 2(k)), et
    - de deuxièmes moyens de génération destinés à générer le signal sonore directionnel (y(k)) selon un motif de réponse directionnelle sur la base des premier et deuxième signaux sonores différentiels (x f(k), x b,DEQ(k)),
    dans lequel la génération du deuxième signal sonore différentiel (x b,DEQ(k)) comprend une génération d'un signal de différence (x b(k)) des premier et deuxième signaux sonores (x 1(k), x 2(k)) et un traitement sélectif en fréquence qui dépend d'un angle de pilotage (α) qui indique une direction désirée d'atténuation maximale du motif de réponse directionnelle dépendant de la fréquence, dans lequel le traitement sélectif en fréquence adapte la direction réelle d'atténuation maximale du motif de réponse directionnelle dépendant de la fréquence de manière à ce qu'elle corresponde à l'angle de pilotage (α) pour l'essentiel indépendant de la fréquence (ω) sur la plage de fréquences du signal sonore directionnel (y(k)).
  14. Système (1), dans lequel le système (1) comprend:
    - un premier et un deuxième microphones (M1, M2) qui sont séparés par une distance (D) et génèrent des premier et deuxième signaux sonores (x 1(k), x 2(k)), et
    - un dispositif (2) tel que défini dans la revendication 13.
  15. Programme d'ordinateur comprenant des moyens de code de programme, qui, lorsqu'il est exécuté sur un ordinateur commandant le dispositif (2) selon la revendication 13, met en oeuvre les étapes du procédé selon l'une quelconque des revendications 1 à 12.
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US20220109511A1 (en) * 2020-10-05 2022-04-07 CUE Audio, LLC Method and system for digital communication over an acoustic channel

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JP7139628B2 (ja) * 2018-03-09 2022-09-21 ヤマハ株式会社 音処理方法および音処理装置
GB2575491A (en) * 2018-07-12 2020-01-15 Centricam Tech Limited A microphone system

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US20220109511A1 (en) * 2020-10-05 2022-04-07 CUE Audio, LLC Method and system for digital communication over an acoustic channel
US11728905B2 (en) * 2020-10-05 2023-08-15 CUE Audio, LLC Method and system for digital communication over an acoustic channel

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