EP3225037A1 - 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

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Publication number
EP3225037A1
EP3225037A1 EP14771598.1A EP14771598A EP3225037A1 EP 3225037 A1 EP3225037 A1 EP 3225037A1 EP 14771598 A EP14771598 A EP 14771598A EP 3225037 A1 EP3225037 A1 EP 3225037A1
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EP
European Patent Office
Prior art keywords
frequency
directional
signal
generating
sound
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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.)
Granted
Application number
EP14771598.1A
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German (de)
English (en)
Other versions
EP3225037B1 (fr
Inventor
Hauke Krüger
Bernd Geiser
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.)
Binauric Se
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Binauric Se
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Classifications

    • 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

Definitions

  • 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.
  • a possible target device is a wireless loudspeaker with two integrated miniature digital micro-electromechanical system (MEMS) microphone cap- sules which facilitate handsfree audio communication.
  • 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 M 1 and M2 are used to capture the acoustic environment.
  • the corresponding digital signals x t (k) and x 2 (fe) are sampled with a rate of f s .
  • the signals x f (fc) and x (fc) 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 . 3 in G. Elko and A.-T. N. Pong).
  • 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 generating of the second differential sound signal comprises generating a difference signal of the first and the second sound signals and a frequency-selective processing that depends on a steering angle, which indicates a desired direction of maximum attenuation of the frequency-dependent directional response pattern, wherein the frequency-selective processing adjusts the actual direction of maximum attenuation of the frequency-dependent directional response pattern to correspond to the steering angle substantially independent of frequency over the frequency range of the directional sound signal.
  • 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).
  • the term "difference signal" as used herein 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. It is further preferred that 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 param- eter, 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:
  • first generating means for generating first and second differential sound signals based on the first and second sound signals
  • generating of the second differential sound signal comprises generating a difference signal of the first and the second sound signals and a frequency-selective processing that depends on a steering angle, which indicates a desired direction of max- imum attenuation of the frequency-dependent directional response pattern, wherein the frequency-selective processing adjusts the actual direction of maximum attenuation of the frequency-dependent directional response pattern to correspond to the steering angle substantially independent of frequency over the frequency range of the directional sound signal.
  • a first and a second microphone which are separated by a distance and generate first and second sound signals
  • Fig . 1 shows schematically and exemplarily a differential microphone array according to G. Elko and A.-T. N. Pong,
  • Fig . 2 illustrates the optimal steering factor vs. a linear approximation
  • H(a), (l>) 2j - e- J ⁇ (c ⁇ +1 sin I—— (1 + cos0) 1— a ⁇ sin I—— (1— cos0)
  • the steering angle a should, ideally, be adapted to match the interference or noise incidence angle ⁇ . This is discussed in section 6.
  • the factors ⁇ ( ⁇ ) and ⁇ ( ⁇ ) can be computed by marginalization of the 2-dimensional function ⁇ ( ⁇ , ⁇ ) and appropriate normalization.
  • the factor ⁇ ( ⁇ ) 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 3 ⁇ 4 DEQ (fe), 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 ⁇ ( ⁇ ) after a variable transformation from to lin (a), i.e., ⁇ ⁇ ( ⁇ ⁇ ( ⁇ )) ⁇ ⁇ ( ⁇ ( ⁇ ⁇ ⁇ ))-
  • 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).
  • a smaller microphone distance D could be used so that the product ⁇ in Eq. (6) remains sufficiently small.
  • the downside of this approach is a stronger highpass effect of the array which, in turn, requires heavier output equalization with a more pronounced lowpass filter h (k). In a real system, this leads to a higher amplification of the microphone noise, particularly at low frequencies.
  • D 1.8 cm
  • less than half of the original distance is required to obtain a comparably straight directional response. This, however, comes at the cost of a significantly increased noise gain (+10dB) over a wide frequency range.
  • 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 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(fe), i.e. where, usually, 0 ⁇ a ⁇ 1, i.e., 180° > a ⁇ 90° is enforced.
  • the stepwise update this method is (e.g., H. Puder) ⁇
  • 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(fe) should ideally be zero.
  • an error signal e(k) appears at the output.
  • the optimal (adaptive) stepsize parameter is
  • the best approximation of £ ⁇ n 2 (fc) ⁇ is the level ⁇ 2 of the microphone array's output y(fe) while for £ ⁇ e 2 (fc) ⁇ , the assumption of a fixed attenuation factor for the backward cardioid signal is made, i.e. £ ⁇ e 2 (fc) ⁇ ⁇ ⁇ ⁇ i 2 b .
  • ⁇ 0.01 (assumed attenuation of 20 dB) and the adaptive stepsize parameter is hence with the recursively estimated short term powers ⁇ , 2 and 8 , which leads to the new NLMS update rule
  • a(k) a(k - 1) + ⁇ — — - x b (fe) - y(fe)
  • 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(fe).
  • 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 ⁇ .
  • 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. (12)) (gray curve) for example drifts towards 90° easily in case of activity of the desired sound source, but even the slower version (blue curve) is not able to maintain a once identified steering factor in all situations.
  • 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.
  • 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. Kriiger 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 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).
  • the noise (or interferer) suppression works more reliable in a broader range of acoustic scenarios. Also, the desired source does not compromise the notch adaptation anymore. Moreover, the combination with the directional equalizer leads to a more stable direction of arrival tracking.
  • 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)

Abstract

La présente invention concerne un procédé de génération d'un signal sonore directionnel (y(k)) à partir de premier et deuxième signaux sonores (x 1 (k), x 2 (k)), qui sont générés par un premier et un deuxième microphone (M1, M2), séparés par une distance (D). Le procédé comporte les étapes consistant à générer des premier et deuxième signaux sonores différentiels (x f (k), X b,DEQ (k)) en fonction des premier et deuxième signaux sonores (x 1 (k), x 2 (k)), et à générer le signal sonore directionnel (y(k)) selon un diagramme de réponse directionnel dépendant de la fréquence en fonction des premier et deuxième signaux sonores différentiels (x f (k), X b,DEQ (k)). La génération du deuxième signal sonore différentiel (X b,DEQ (k)) comporte l'étape consistant à générer un signal de différence (xb(k)) des premier et deuxième signaux sonores (x 1 (k), x 2 (k)) et un traitement sélectif en fréquence dépendant d'un angle d'orientation (α) qui indique une direction souhaitée d'atténuation maximale du diagramme de réponse directionnel dépendant de la fréquence, le traitement sélectif en fréquence réglant la direction réelle d'atténuation maximale du diagramme de réponse directionnel dépendant de la fréquence de façon à correspondre à l'angle d'orientation (α) sensiblement indépendant de la fréquence (ω) sur la gamme de fréquence du signal sonore directionnel (y(k)).
EP14771598.1A 2014-09-23 2014-09-23 Procédé et appareil de génération d'un signal sonore directionnel à partir de premier et deuxième signaux sonores Active EP3225037B1 (fr)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
PCT/EP2014/070243 WO2016045706A1 (fr) 2014-09-23 2014-09-23 Procédé et appareil de génération d'un signal sonore directionnel à partir de premier et deuxième signaux sonores

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EP3225037A1 true EP3225037A1 (fr) 2017-10-04
EP3225037B1 EP3225037B1 (fr) 2019-05-08

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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
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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US9202475B2 (en) * 2008-09-02 2015-12-01 Mh Acoustics Llc Noise-reducing directional microphone ARRAYOCO

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EP3225037B1 (fr) 2019-05-08
WO2016045706A1 (fr) 2016-03-31

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