EP1538867A1 - Freisprechanlage für ein Kraftfahrzeug - Google Patents

Freisprechanlage für ein Kraftfahrzeug Download PDF

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Publication number
EP1538867A1
EP1538867A1 EP03022273A EP03022273A EP1538867A1 EP 1538867 A1 EP1538867 A1 EP 1538867A1 EP 03022273 A EP03022273 A EP 03022273A EP 03022273 A EP03022273 A EP 03022273A EP 1538867 A1 EP1538867 A1 EP 1538867A1
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Prior art keywords
filter
signal
adaptive
microphones
beamformer
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EP03022273A
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French (fr)
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EP1538867B1 (de
Inventor
Markus Chirstoph
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Nuance Communications Inc
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Harman Becker Automotive Systems GmbH
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Priority claimed from EP03014846.4A external-priority patent/EP1524879B1/de
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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
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/406—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers 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
    • H04R2430/23—Direction finding using a sum-delay beam-former
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10—General applications
    • H04R2499/13—Acoustic transducers and sound field adaptation in vehicles

Definitions

  • the invention is directed to a handsfree system for use in a vehicle comprising a microphone array with at least two microphones and a signal processing means.
  • handsfree systems are used more and more since they provide increased comfort and reduce the risk of an accident as the driver is distracted only marginally. Because of that, in many countries, handsfree devices are even required by law.
  • a handsfree system comprises a microphone that can be fastened to a user such as the driver.
  • the invention provides a handsfree system for use in a vehicle comprising a microphone array with at least two microphones, a signal processing means and an adaptive post-filter, the signal processing means comprising a beamformer having an input connected to the at least two microphones and an output connected to the input of the adaptive post-filter.
  • a beamformer processes signals emanating from a microphone array to obtain a combined signal.
  • a beamformer comprises a beamsteering means being responsible for time delay compensation of the different microphones and a summing means.
  • beamforming In its simplest form (Delay-and-Sum beamformer), beamforming only comprises delay compensation and summing of the compensated signals. Beamforming allows to provide a specific directivity pattern for a microphone array.
  • a beamformer can be implemented as digital system with a plurality of digital filter using, for example, digital signal processors (DSP).
  • DSP digital signal processors
  • a beamformer can be configured as an adaptive or a non-adaptive beamformer.
  • Adaptive means that relevant parameters such as filter coefficients can be re-calculated during use of the system in order to adapt the beamformer to changing conditions.
  • the system parameters are determined once by calibrating the beamformer and, then, kept unchanged.
  • the beamforming in principle, can be performed in the time domain or in the frequency domain.
  • a handsfree system in accordance with the invention shows an excellent acoustic performance in a vehicular environment. Due to the beamformer, an improved directivity is obtained and, furthermore, speech signals are enhanced and ambient noise is reduced.
  • the adaptive post-filter (responsible for filtering a signal after the beamforming) further reduces the noise in the signal.
  • the adaptive post-filter can be a filter in the time domain. If the post-filtering is performed in the time domain, the delay time is reduced and the implementation is simplified.
  • the adaptive post-filter can be a Wiener filter. It turns out that a Wiener filter is particularly suitable for filtering in a car environment.
  • the adaptive post-filter can be a linear-phase filter.
  • the adaptive post-filter can be a linear-phase Wiener filter.
  • the signal processing means can further comprise at least two adaptive filters having an input connected to the output of the beamsteering means and an output connected to the adaptive post-filter, wherein the at least two adaptive filters are configured to determine adaptive filter parameters for the adaptive post-filter.
  • background filters are provided for adaptively estimating the filter parameters for the adaptive post-filter.
  • an adaptive filter can be provided having an input connected to the output of the beamsteering means.
  • adaptive filter parameters can be determined for the adaptive post-filter.
  • the actual filter parameters of the post-filter can be given, for example, by the filter parameters determined by one of the adaptive filters or the mean of the filter parameters determined by several different adaptive filters.
  • an input of each of the at least two adaptive filters can be further connected to the output of the beamformer. This allows for an adaption of the respective filter parameters directly with respect to the beamformed signal.
  • the signal processing means can further comprise a pre-emphasis filter, in particular, comprising a pre-whitening filter, having an input connected to an output of the adaptive post-filter and/or a pre-emphasis filter, in particular, comprising a pre-whitening filter, having an input connected to the output of the beamsteering means and an output connected to the at least two adaptive filters.
  • a pre-emphasis filter in particular, comprising a pre-whitening filter, having an input connected to an output of the adaptive post-filter and/or a pre-emphasis filter, in particular, comprising a pre-whitening filter, having an input connected to the output of the beamsteering means and an output connected to the at least two adaptive filters.
  • the pre-emphasis filter can comprise a pre-whitening filter.
  • a pre-whitening filter whitens the spectral distribution of a signal.
  • the filter coefficients of such a pre-whitening filter can be determined using a linear predictive coding (LPC) analysis, for example, via an adaptive lattice predictor (ALP) algorithm.
  • LPC linear predictive coding
  • ALP adaptive lattice predictor
  • the signal processing means can further comprise an inverse filter, particularly a warped inverse filter. These filters are especially useful to adjust the microphone transfer function and to match the microphones of the array in this way.
  • the beamformer can comprise at least one inverse filter, in particular, having an output for providing an inversely filtered signal to a summing means.
  • matched microphones on the basis of silicone or paired microphones may be used.
  • the susceptibility of microphone arrays often increases with decreasing frequency. Due to this, a higher matching precision is preferred for low frequencies compared to high frequencies. A frequency depending adjustment of the microphone transfer functions with the use of warped filters reduces the required memory compared to the case of conventional FIR filters.
  • each inverse filter can be an approximate inverse of a non-minimum phase filter. This results in an inverse filter which is both stable and has no phase error.
  • an inverse filter may be combined with another filter of the handsfree system, for example, a filter of the beamformer.
  • another filter of the handsfree system for example, a filter of the beamformer.
  • the signal processing means of the above handsfree systems can comprise a non-adaptive post-filter having an input connected to an output of the adaptive post-filter.
  • the non-adaptive post-filter may directly follow the adaptive post-filter.
  • Such a filter is used to compensate for the ambient acoustics of a speaker.
  • the non-adaptive post-filter may have the form of an inverse room filter.
  • the signal processing means may further comprise an adaptive noise canceller (ANC), for electrical ANC implementations.
  • ANC adaptive noise canceller
  • the ANC can be connected to a non-acoustic sensor to determine a noise signal, for example, by using the tachometer of the vehicle.
  • the ANC advantageously, can have an output connected to the input of the beamformer and/or of the adaptive post-filter.
  • the signal processing means of the previously described handsfree systems can comprise an acoustic echo canceller AEC.
  • the AEC can comprise an echo shaping filter. In this way, a frequency selected echo attenuation may be obtained.
  • the AEC can have an output connected to the input of the beamformer and/or of the adaptive post-filter.
  • the beamformer can be a non-adaptive beamformer.
  • the computing power during operation of the system is reduced.
  • the beamformer may be a superdirective beamformer which further improves the acoustic performance.
  • the beamformer may be a regularized superdirective beamformer using a finite regularization parameter ⁇ .
  • the regularization parameter usually enters the equation for computing the filter coefficients or, alternatively, is inserted into the cross-power spectrum matrix or the coherence matrix.
  • the regularized superdirective beamformer has reduced noise and is less sensitive to an imperfect matching of the microphones.
  • the finite regularization parameter ⁇ may depend on the frequency. This achieves an improved gain of the array compared to a regularized superdirective beamformer with fixed regularization parameter ⁇ .
  • each superdirective filter may result from an iterative design based on a predetermined maximum susceptibility. This enables an optimal adjustment of the microphones, particularly with respect to the transfer function and the position of each microphone.
  • the maximum susceptibility may be determined as a function of the error in the transfer characteristic of the microphones, the error in the microphone positions and a predetermined (required) maximum deviation in the directional diagram of the microphone array.
  • the time-invariant impulse response of the filters will be determined iteratively only once; there is no adaption of the filter coefficients during operation.
  • each superdirective filter can be a filter in the time domain. Filtering in the frequency domain is a possible alternative, however, requiring to perform a Fourier transform (FFT) and an inverse Fourier transform (IFFT), thus, increasing the required memory.
  • FFT Fourier transform
  • IFFT inverse Fourier transform
  • the beamformer may have the structure of a generalized sidelobe canceller (GSC).
  • GSC generalized sidelobe canceller
  • the beamformer can be a minimum variance distortionless response (MVDR) beamformer.
  • MVDR minimum variance distortionless response
  • the microphone array can comprise at least two microphones being arranged in endfire orientation with respect to a first position.
  • An array in endfire orientation has a better directivity and is less sensitive to a mismatched propagation or delay time compensation.
  • the first position can be the location of the drivers head, for example.
  • the microphone array can comprise at least two microphones being arranged in endfire orientation with respect to a second position.
  • the handsfree system of the invention has a good directivity in two directions. Speech signals coming from two different positions, for example, from the driver and the front seat passenger, can both be recorded in good quality.
  • the signal processing means may comprise at least two beamformers.
  • a first beamformer may be used for signals from a first position and a second beamformer may be used for signals from a second position.
  • the handsfree system may further comprise a voice activity detector (VAD) and/or a switch control means.
  • VAD voice activity detector
  • the switch control and the VAD are used to determine how to combine the output of the at least two beamformers.
  • the handsfree system can comprise a residual echo suppression (RES) means and/or a dynamic volume control (DVC).
  • a RES means serves for suppression of residual echoes, in particular, being present in the signal resulting from the adaptive post filter.
  • a residual echo suppression means can comprise an input connected to the output of the adaptive post filter.
  • a RES means can comprise an input for receiving a far end signal.
  • a DVC is intended for dynamically adapting the output volume of a far end signal depending on the ambient noise level being present in the vehicle.
  • the at least two microphones in the first endfire orientation (endfire orientation with respect to a first position) and the at least two microphones in the second endfire orientation (endfire orientation with respect to a second position) can have a microphone in common.
  • a microphone array consisting of only three microphones provides an excellent directivity for use in a vehicular environment.
  • the microphone array may comprise at least two subarrays.
  • Each subarray of microphones may be optimized for a specific frequency band yielding an improved overall directivity.
  • At least two subarrays may have at least one microphone in common.
  • the above handsfree systems may comprise a frame wherein each microphone of the microphone array is arranged in a predetermined, in particular fixed, position in or on the frame. This ensures that after manufacture of the frame with the microphone, the relative positions of the microphones are known. Such an array can be easily mounted in a vehicular cabin.
  • At least one microphone may be a directional microphone.
  • the use of directional microphones improves the array gain.
  • At least one directional microphone may have a cardioid characteristic. This further improves the array gain. More preferred, the cardioid characteristic is a hyper-cardioid characteristic.
  • At least one directional microphone may be a differential microphone.
  • the differential microphone may be a first order differential microphone.
  • the invention is further directed to a vehicle, particularly a car, comprising any of the above-described handsfree systems.
  • the invention is also directed to the use of any of the previously described handsfree systems in a vehicle, in particular, a car.
  • the invention provides a method for noise reduction in a vehicular handsfree system, comprising receiving input signals resulting from a microphone array with at least two microphones, processing the input signals by a beamformer to provide a beamformed signal, and adaptively filtering a signal resulting from the beamformed signal by an adaptive post-filter.
  • This method results in an excellent acoustic performance of a handsfree system in a vehicular environment.
  • the adaptive filtering can be performed in the time domain. In this way, particularly the delay time is reduced.
  • the method can further comprise
  • adaptively filtering can further comprise receiving a signal resulting from the beamformed signal by at least one adaptive filter and wherein processing the beamsteered signal can comprise determining adaptive filter parameters using the at least one beamsteered signal and the signal resulting from the beamformed signal.
  • an adaptive filter for each beamsteered signal, can be provided for determining adaptive filter parameters using the beamsteered signal and the signal resulting from the beamformed signal.
  • receiving at least one beamsteered signal by at least one of the at least two adaptive filters can comprise processing the at least one beamsteered signal by a pre-emphasis filter, in particular, comprising a pre-whitening filter.
  • the above methods can further comprise processing a signal resulting from the microphone array by an inverse filter, in particular, a warped inverse filter.
  • the methods can further comprise non-adaptively filtering a signal resulting from the adaptively filtered signal and/or processing a signal resulting from the adaptively filtered signal by a pre-emphasis filter.
  • the above method can further comprise processing a signal resulting from the microphone array, particularly resulting from the beamformed signal, by an adaptive noise canceller (ANC) and/or an acoustic echo canceller (AEC) and/or a residual echo suppression (RES) means.
  • ANC adaptive noise canceller
  • AEC acoustic echo canceller
  • RES residual echo suppression
  • the input signals can be processed by a non-adaptive and/or superdirective and/or minimum variance distortionless response (MVDR) beamformer.
  • MVDR minimum variance distortionless response
  • the invention also provides a computer program product comprising one or more computer readable media having computer-executable instructions for performing the steps of the above described methods.
  • FIG. 1 An example of the handsfree system in accordance with the present invention is shown in Fig. 1.
  • the general structure will be shortly described, and, then, the different components will be explained in more detail.
  • the dotted lines encasing some elements simply serve for better understanding of the figures without necessarily implying any actual combination or separation of different elements.
  • the main components of the system are a microphone array, a beamformer and an adaptive post-filter in the time domain.
  • the microphone array 101 in this example, comprises four microphones 102. Each microphone 102 yields an output signal x i [ k ].
  • the microphone signals may be filtered by an optional high-pass filter 103.
  • a beamformer may be a conventional delay and sum beamformer. However, in the present example, a preferred superdirective beamformer is shown.
  • a beamformer comprises beamsteering means 104 and filters 105.
  • the output signals of the beamformer may be passed through optional inverse filters 106 and, then, are summed by summing means 107 to yield a resulting beamformed signal x [ k ].
  • This signal is passed through an adaptive post-filter 108 in the time domain which may be followed by an optional non-adaptive post-filter 109 and/or by an optional pre-emphasis filter (not shown).
  • the adaption of the post-filter 108 is performed using a set of Wiener filters 109.
  • the input signals of the Wiener filters 110 comprise, on the one hand, the individual signals resulting from the different microphones and, on the other hand, the summed signal x [ k ].
  • the microphone signals are taken after the beamsteering.
  • the beamformer comprises further (superdirective) filters 105 as in the present case, it is also possible to take the microphone signals after this additional filtering.
  • the microphone signals are passed through an optional pre-emphasis filter 111.
  • a microphone signal x [ k ] is the sum of the speech signal s [ k ] and the noise n [ k ].
  • the microphone signal will be filtered by an impulse response w ( i ) to obtain a noise reduced signal [ k ]. It is the aim to minimize the mean square error between the undisturbed speech signal s [ k ] and the output signal [ k ]:
  • the partial derivative of the mean square error with respect to the coefficients of the impulse response has to vanish yielding the Wiener-Hopf equation: wherein r xx ( l ) and r sx ( l ) are the auto-correlation function and the cross-correlation function of the microphone signal and the undisturbed speech signal.
  • r xx ( l ) and r sx ( l ) are the auto-correlation function and the cross-correlation function of the microphone signal and the undisturbed speech signal.
  • the power spectral densities in the above equation may be replaced by the corresponding short-time estimated values that may be obtained, for example, by a recursive averaging: wherein S ( ⁇ , ⁇ ) and X ( ⁇ , ⁇ ) are short-time spectra that may be determined, for example, with the help of DFT filter banks or an FFT.
  • ⁇ is the time index and ⁇ the frequency index;
  • E ⁇ . ⁇ represents the short-time average that may be obtained, for example, with the help of a first order IIR filter.
  • the short-time auto power spectral density of the speech signal in the numerator of the above equation is to be estimated in a suitable way.
  • Appropriate estimation methods include spectral subtraction (estimating the auto power spectral density of the noise), minimum mean square error short-time spectral amplitude (MMSE STSA) estimator or MMSE log-SA estimator or a speech pause detector, for example.
  • the estimated short-time auto power spectral density of the noise signal may be used to estimate the absolute value of the most probable Fourier coefficient (using, for example, a spectral subtraction algorithm or an MMSE log-SA estimator) and to reconstruct the absolute value of the spectrum of the speech signal.
  • the coherence or the cross power spectral density of the noise signals received by the microphones is vanishing.
  • the microphone signal has the form: wherein h 1 ( i ) and h 2 ( i ) are the impulse responses representing the acoustic transfer between the source of speech and the microphones. Both parts of the speech signal filtered in this way are superimposed with the uncorrelated noise signals n 1 [ k ] and n 2 [ k ].
  • the adaption of the post-filter w ( k,i ) - k being the time index and i denoting the coefficient within the impulse response - is performed in the time domain, for example, with the help of the LMS algorithm.
  • the background Wiener filters w 1 ( k , i ),..., w 4 ( k , i ) are two minimize the error signals e 1 [ k ],..., e 4 [ k ] such that, for example, the filter w 4 ( k , i ) tends towards the frequency response wherein
  • the form of the other three Wiener filters is obtained by a cyclic permutation of the indices.
  • Wiener filters 110 are not restricted to a particular number of Wiener filters 110. Furthermore, not every Wiener filter 110 is always to be used to determine the adaptive post-filter 108. For example, one may use only the Wiener filter which uses the microphone signal of the microphone proximal to the source of speech.
  • the filter coefficients of a linear-phase post-filter (with length L) can be obtained. Accordingly, the linear-phase post-filter has twice the length of one of the background filters 110 (with length L/2). Such a linear-phase filter only modifies the amplitude spectrum of the input signal of the filter without a frequency dependent distortion of the phase spectrum.
  • the performance of the filter can be further improved by smoothing its frequency response. This can be achieved by weighting the filter coefficients with a window function.
  • the inverse filters 106 serve to compensate for the acoustic transfer function of the path between the source of speech and the microphones.
  • FIG 2 the structure of a superdirective beamformer is shown.
  • the beamformer shown in this figure performs the filtering in the frequency domain, in contrast to the case of Figure 1. If a beamformer in the frequency domain were used in Figure 1, an inverse Fourier transform is to be performed on the signals before passing the signals to the Wiener filters 110 or the pre-emphasis filter 111.
  • the microphone array consists of M microphones 102, each yielding a signal x i (t).
  • the signals x i (t) are transferred to the frequency domain by fast Fourier transform (FFT) means 201, resulting in a signal X i ( ⁇ ).
  • FFT fast Fourier transform
  • the beamforming consists of a beamsteering and a filtering. The beamsteering is responsible for the propagation time compensation.
  • the signals are filtered by superdirective filters 203 that are filters in the frequency domain.
  • the filtered signals are summed yielding a signal Y ( ⁇ ).
  • IFFT inverse fast Fourier transform
  • a i ( ⁇ ) ( ⁇ ( ⁇ ) + ⁇ I ) -1 d d T ( ⁇ ( ⁇ ) + ⁇ I ) -1 d wherein I is the unity matrix.
  • the directional diagram or response pattern ( ⁇ ( ⁇ , ⁇ ) of a microphone array characterizes the sensitivity of the array as a function of the direction of incidence ⁇ for different frequencies .
  • a measure to describe the directivity of an array is the so-called gain that does not depend on the angle of incidence ⁇ .
  • the gain is defined as the sensitivity of the array in the main direction of incidence with respect to the sensitivity for omnidirectional incidence.
  • the Front-To-Back-Ratio indicates the sensitivity in front receiving direction compared to the back.
  • the white noise gain (WNG) describes the ability of the array to suppress uncorrelated noise, for example, the inherent noise of the microphones.
  • the inverse of the white noise gain is the susceptibility K ( ⁇ ),
  • the susceptibility K ( ⁇ ) describes the array's sensitivity to defective parameters. It is often preferred that the susceptibility K ( ⁇ ) of the array filters A i ( ⁇ ) does not exceed an upper bound K max ( ⁇ ) .
  • the selection of this upper bound may be dependent on the relative error ⁇ 2 ( ⁇ , ⁇ ) of the microphones and, for example, on requirements regarding the directional diagram ⁇ ( ⁇ , ⁇ ).
  • the relative error ⁇ 2 ( ⁇ , ⁇ ) in general, is the sum of the mean square error of the transfer properties of all microphones ⁇ 2 ( ⁇ , ⁇ ) and the Gaussian error with zero mean of the microphone positions ⁇ 2 ( ⁇ ).
  • the error in the microphone transfer functions ⁇ ( ⁇ ) has a higher influence on the maximum susceptibility K max ( ⁇ ) and, thus, also on the maximum possible gain G ( ⁇ ) than the error ⁇ 2 ( ⁇ ) in the microphone positions.
  • the defective transfer functions are mainly responsible for the limitation of the maximum susceptibility.
  • a higher mechanical precision to reduce the position deviations of the microphones is only sensible up to a certain point since the microphones usually are modeled as being point-like, which is not true in reality.
  • the microphones usually are modeled as being point-like, which is not true in reality.
  • ⁇ 2 ( ⁇ ) 1% which is quite realistic.
  • the error ⁇ ( ⁇ ) can be derived from the frequency depending deviations of the microphone transfer functions.
  • inverse filters may be used to adjust the individual microphone transfer functions to a reference transfer function.
  • a reference transfer function can be the transfer function of one microphone out of the array or, for example, the mean of all measured transfer functions.
  • M being the number of microphones
  • the transfer functions are not minimal phase, thus, a direct inversion would yield instable filters.
  • the approximate inversion with the help of an FXLMS (filtered X least mean square) or the FXNLMS (filtered X normalized least mean square) algorithm will be described.
  • the inverse filters may be coupled with the superdirective filters A i ( ⁇ ) such that, in the end, only one filter per viewing direction and microphone is to be implemented.
  • the FXLMS or the FXNLMS algorithm is described with reference to Figure 3.
  • the update of the filter coefficients of w[ n ] is performed iteratively, i.e. at each time step n , whereby the filter coefficient w[ n ] are computed such that the instantaneous squared error e 2 [ n ] is minimized.
  • the susceptibility increases with decreasing frequency.
  • the FIR filters for example, are to be very long in order to obtain a sufficient frequency resolution in the desired frequency range. This means that the expenditure, in particular, regarding the memory, increases rapidly.
  • the computing time does not impose a severe limitation.
  • a suitable frequency depending adaption of the transfer functions can be achieved by using short WFIR filters (warped filters).
  • a realization of the beamforming filters in the time domain - as in the system of Figure 1 - is described with reference to Figure 4.
  • Signals are recorded by microphones 102.
  • a near field beamsteering 104 is performed using gain factors ⁇ k 401 to compensate for the amplitude differences and time delays ⁇ k 402 to compensate for the propagation time differences of the microphone signals x k [ i ]
  • the realization of the superdirective beamforming is achieved using the filters (preferably, FIR filters) ⁇ k ( i ) indicated by reference sign 403.
  • the impulse responses ⁇ 1 ( i ),..., ⁇ M ( i ) can be determined as follows:
  • the microphone signals are directly processed using the beamsteering 104 in the time domain.
  • the beamsteering 104 is followed by the FIR filtering 403. After summing the filtered signals, a resulting enhanced signal y [ k ] is obtained.
  • ⁇ max d mic f a c .
  • the sampling frequency or the distance between the microphones can be chosen much higher than in the broad-side case, thus, resulting in an improved beamforming.
  • the maximum microphone distance that can be chosen depends not only on the lower limiting frequency for the optimization of the directional characteristic, but also on the number of microphones and on the distance of the microphone array to the speaker. In general, the larger the number of microphones, the smaller their maximum distance in order to optimize the Signal-To-Noise-Ratio (SNR).
  • a further improvement of the directivity, and, thus, of the gain, can be achieved by using unidirectional microphones instead of omnidirectional ones; this will be discussed in more detail below.
  • Figures 5A and 5B show preferred arrangements of microphone arrays in a vehicle.
  • the distance between the microphone array and the speaker should be as small as possible.
  • each speaker 501 may have its own microphone array comprising at least two microphones 102.
  • the microphone arrays may be provided at different locations, for example, within the headliner, dashboard, pillar, headrest, steering wheel, compartment door, visor or (driving) mirror.
  • An arrangement within the roof is also a preferred possibility that is, however, not suitable for the case of a cabriolet. Both microphone arrays for each speaker are in endfire orientation.
  • one microphone array is used for two neighboring speakers.
  • directional microphones in particular, having a cardioid characteristic, may be used.
  • the microphone array may be mounted within the mirror.
  • Such a linear microphone array may be used for both the driver and the front seat passenger. A costly mounting of the microphones in the roof can be avoided.
  • the array may be mounted in one piece, which ensures a high mechanical precision. Due to the adjustment of the mirror, the array would always be correctly oriented.
  • Figure 6A shows a top view on a (driving) mirror 601 of a car with three microphones in two alternative arrangements.
  • two microphones 602 and 603 are located in the center of the mirror in endfire orientation with respect to the driver and, preferably, have a distance d mic of about 5cm between each other.
  • the microphones 603 and 604 are in endfire orientation with respect to the front seat passenger and have a distance of about 10cm between each other. Since the microphone 603 is used for both arrays, a cheap handsfree system can be provided.
  • All three microphones may be directional microphones, preferably having a cardioid characteristic, for example, a hypercardioid characteristic.
  • microphones 602 and 604 are directional microphones, whereas microphone 603 is an omnidirectional microphone which further reduces the costs. If all three microphones are directional microphones, preferably, microphones 602 and 603 are directed towards the driver.
  • the front seat passenger beamformer Due to the larger distance between microphones 603 and 604 than between microphones 602 and 603, the front seat passenger beamformer has a better SNR at low frequencies.
  • the microphone array for the driver consists of microphones 602' and 603' located at the left side of the mirror. In this case, the distance between this microphone array and the driver would be increased, thus, decreasing the performance. On the other hand, the distance between microphone 603' and 604 would be about 20cm, which yields a better gain for the front seat passenger at low frequencies.
  • FIG. 6B A variant of two microphone arrays with improved precision is shown in Figure 6B. Also in this case, all microphones may be directional microphones, microphones 602 and 603 being directed to the driver, microphones 604 and 605 being directed to a front seat passenger.
  • the microphone array for the front seat passenger comprises the three microphones 603, 604 and 605, which increases the gain considerably.
  • subarray 701 with d mic 5 cm is used for the frequency band of 1400 - 3400 Hz
  • subarray 702 with d mic 10 cm for the frequency band of 700 - 1400 Hz
  • subarray 703 with d mic 20 cm for the band of frequencies smaller than 700 Hz.
  • a lower limit of this frequency band may be imposed, for example, by the lowest frequency of the telephone band (the frequencies used in telephone applications) which, presently, is 300 Hz in most cases.
  • the superdirective beamformer is designed as general sidelobe canceller (GSC).
  • GSC general sidelobe canceller
  • FIG. 8 Such a superdirective beamformer in GSC structure is shown in Figure 8.
  • the GSC structure is to be implemented in the frequency domain, thus, an FFT 201 is applied to the incoming signals x k ( t ) from microphones 102.
  • a time alignment using phase factors e j ⁇ k has to be performed (in this figure, a far field beamsteering is shown).
  • an inverse Fourier transform is to be performed before passing the signal to the Wiener filters 110 or the pre-emphasis filter 111.
  • X denotes a vector comprising all time aligned input signals X i ( ⁇ ).
  • a c is a vector comprising all frequency independent filter transfer functions A i that are necessary to observe the constraints in viewing direction; H is the vector of the transfer functions performing the actual superdirectivity; and B is the so-called blocking matrix projecting the input signals in X onto the "noise plane”.
  • the signal Y DS ( ⁇ ) denotes the output signal of the delay and sum beamformer, Y BM ( ⁇ ) the resulting output signal of the blocking branch, Y SD ( ⁇ ) the output signal of the superdirective beamformer x i ( t ), and X i ( ⁇ ) the input signals in the time and frequency domain that are not yet time aligned, and Y i ( ⁇ ) the output signals of the blocking matrix that ideally should block completely the desired or useful signal within the input signals.
  • the signals Y i ( ⁇ ) ideally only comprise the noise signals.
  • a GSC structure In addition to the superdirective output signal, a GSC structure also yields a delay and sum beamformer signal and a blocking output signal.
  • a blocking matrix should have the following properties:
  • a blocking matrix according to Griffiths-Jim can be used which has the general form
  • the upper branch of the GSC structure is a delay and sum beamformer with the transfer functions
  • H i ( ⁇ ) (B ⁇ nn ( ⁇ )B H ) -1 (B ⁇ nn ( ⁇ )A C ), wherein B is the blocking matrix and ⁇ nn ( ⁇ ) the matrix of the cross-correlation power spectrum of the noise.
  • ⁇ nn ( ⁇ ) may be replaced by the time aligned coherence matrix of the diffuse noise field, as discussed above.
  • a regularization and the iterative design with predetermined susceptibility may be performed in the same way as above.
  • This method may also be generalized to the three-dimensional case. Then, in addition to the parameter ⁇ being responsible for the azimuth, a further parameter ⁇ is to be introduced for the elevation angle. This yields an analog equation for the coherence of the homogeneous diffuse 3D noise field.
  • a superdirective beamformer based on an isotropic noise field is particularly useful for a handsfree system which is to be installed later in a vehicle. This is the case, for example, if the handsfree system is installed in the vehicle by the user itself.
  • an MVDR beamformer may be relevant if there are specific noise sources at fixed relative positions or directions with respect to the position of the microphone array.
  • the handsfree system can be adapted to a particular vehicular cabin by adjusting the beamformer such that its zeros point into the direction of specific noise sources.
  • a noise source may be formed by a loudspeaker or a fan.
  • a handsfree system with MVDR beamformer is already installed during manufacture of the vehicle.
  • the typical distribution of noise or noise sources in a particular vehicular cabin can be determined by performing corresponding noise measurements under appropriate conditions (e.g., driving noise with and/or without loudspeaker and/or fan noise).
  • the measured data are used for the design of the beamformer. It is to be noted that also in this case, no further adaption is performed during operation of the handsfree system.
  • the corresponding superdirective filter coefficients can also be determined theoretically.
  • FIG. 10 shows a superdirective beamformer with directional microphones 1001.
  • each directional microphone 1001 is depicted by its equivalent circuit diagram.
  • d DMA denotes the (virtual) distance of the two omnidirectional microphones composing the first order pressure gradient microphone in the circuit diagram.
  • T is the (acoustic) delay line fixing the characteristic of the directional microphone and
  • EQ TP is the equalizing low path filter yielding a frequency independent transfer behavior in viewing direction.
  • circuits and filters may be realized purely mechanically by taking an appropriate mechanical directional microphone. Again, the distance between the directional microphones is d mic .
  • the whole beamforming is performed in the time domain.
  • a near field beamsteering 104 is applied to the signals x n [ i ] coming from the microphones and being filtered by the equalizing filter EQ TP .
  • the gain factors ⁇ n compensate for the amplitude differences and the delays ⁇ n for the transit time differences of the signals.
  • the FIR filters ⁇ n [ i ] realize the superdirectivity in the time domain.
  • Mechanical pressure gradient microphones have a high quality and yield, in particular, using a hypercardioid characteristic, an excellent array gain.
  • the use of directional microphones results in an excellent Front-to-Back-Ratio as well.
  • FIG. 11 An example for another preferred embodiment of a handsfree system is shown in Fig. 11.
  • the system shown in this figure differs from the system of Fig. 1 in that an adaptive noise canceller (ANC) system 1101 is provided between the microphone array 101 and the high-pass filters 103.
  • ANC adaptive noise canceller
  • the ANC system is particularly useful to reduce motor harmonics in the signal.
  • a wanted signal source 1201 particularly corresponding to a speaker, and a noise source 1202 are shown.
  • the signal entering a microphone 102 which is part of the microphone array is the sum of a wanted signal s [ k ] and a noise signal n 0 [ k ].
  • a noise sensor 1203 is present which is to provide a pure noise signal n 1 [ k ].
  • the reference sensor 1203 is a microphone; in this case, such a microphone should be arranged at a place where no or almost no wanted signal is to be recorded.
  • the output signal y [ k ] of the adaptive filter 1204 is subtracted from the output signal of the microphone 102.
  • the input signal n 1 [ k ] for the adaptive filter 1204 and the output signal or error signal e [ k ] serve for adaption of the noise canceller.
  • the noise reduction of the adaptive noise canceller depends only on the coherence of the signals of the microphone 102 and the reference sensor 1204; this coherence function in turn is depending on the distance between microphone and reference sensor.
  • the reference sensor is not an acoustic sensor.
  • One possibility is to couple an electrical sensor with the tachometer or speed counter which is usually present in a vehicle. After determining the interrelationship between the tachometer signal and the motor noise, the latter may be subtracted from the microphone signal via the adaptive noise canceller. Such an embodiment is shown in Fig. 11 where a tachometer 1102 is coupled to the ANC 1101.
  • the ANC need not be placed directly behind the microphone array 101.
  • the ANC may be used to filter the output signal x[k] of the superdirective beamformer.
  • the ANC is to be placed between the summing circuit 107 and the adaptive post-filter 108.
  • a further noise reduction with the help of an ANC system can be achieved by using additional - acoustical or non-acoustical - noise sensors.
  • a corresponding embodiment is shown in Fig. 13.
  • the ANC system 1304 is used particularly to suppress signals coming from a loudspeaker 1301, for example, emitting a far end signal 1302.
  • the ANC system is able to create a so-called area of silence around the noise sensor or noise sensors. If the microphone array 101 is located in the vicinity of the near speaker, the whole array 101 or one of its microphones 102 may be used as noise sensor. Alternatively, one or more acoustical (1203) and/or non-acoustical (1102) noise sensors are to be installed.
  • an acoustical ANC can provide a noise reduction for both the near and far end speaker.
  • an additional acoustic echo canceller (AEC) system 1303 may also be provided.
  • AEC acoustic echo canceller
  • each microphone 102 is provided with an individual AEC filter.
  • an AEC filter may be placed between the summing circuit and the adaptive post-filter.
  • an ANC may be placed between the summing circuit and the AEC.
  • the AEC system used for this invention comprises a conventional AEC filter and an integrated frequency selected echo attenuation which acts as a residual echo suppression (RES) algorithm.
  • RES residual echo suppression
  • a preferred embodiment of such a system is shown in Fig. 14. It comprises a conventional AEC filter 1303 that filters the far end signal 1302.
  • the adaption of the conventional AEC filter 1303 is performed using the signal 1401, e.g. the output signal of an electrical ANC or of one of the microphones.
  • an echo shaping means 1402 is provided.
  • This echo shaping means has the form of an adaptive FIR filter with coefficient vector H [ k ] that filters the compensated signal e [ k ].
  • the coefficient vector H [ k ] of the adaptive filter is taken in each sampling step from another adaptive FIR filter with coefficient vector H 1[ k ].
  • the filter with coefficient vector H 1[ k ] is a linear-phase filter of low order.
  • the echo shaping means further comprises a delay element T H 1 .
  • the resulting signal z [ k ] depends on the time varying factor ⁇ [ k ].
  • the output signal y [ k ] of the ANC system is dominant ( ⁇ [ k ] close to 1) at the input of the adaptive filter with coefficient vector H 1[ k ].
  • the adaptive filter with coefficient vector H 1[ k ] can reduce the error signal E [ k ] only by suppressing the signal of the far end speaker.
  • the near speaker and local noise signals will not be attenuated by the echo shaping means.
  • the echo shaping algorithm is frequency selective.
  • the system shown in Fig. 15 is able to process speech signals from two different positions (for example, from the driver and the front seat passenger in a car).
  • the microphone array 101 has a directional diagram with two preferred directions. For example, directional microphones may be used and/or the microphones may be arranged in a suitable way.
  • One or several ANC or AEC filters can provide an estimation of the noise level present in the microphone signals that may be used in the dynamic volume control (DVC) 1501 to vary the volume of the far end speech signal 1511 in dependence of the noise level.
  • the system comprises a beamformer for both wanted signal sources each comprising a beamsteering means 1502 and 1504 and beamformer filters 1503 and 1505. Following each beamformer, adaptive post-filters 1506 and 1507 are arranged which, in turn, are directly connected to non-adaptive post-filters 1508 and 1509.
  • the output signals of the non-adaptive post-filters are fed to a unit 1510 comprising two voice activity detectors and a switch control that generates a weighting factor A for combining both signals.
  • a unit 1510 comprising two voice activity detectors and a switch control that generates a weighting factor A for combining both signals.
  • each of the signals s ⁇ / 1[ k ] and s ⁇ / 2[ k ] can be processed by a low-pass filter.
  • the contained noise signal or its level is estimated using, for example, a minimum statistics.
  • the noise signal level is subtracted from the corresponding filtered signal level.
  • the resulting signal levels are compared to a threshold value. Depending on this comparison of both signal levels, the weighting factor A is determined.
  • a possible weighting can be determined as follows. If both signal levels are larger than the threshold value, both signals are equally weighted. If one of the signal levels is larger than the threshold value and the other is smaller than the threshold value, the larger signal is weighted by a factor of 1 and the other is fully suppressed (weighting factor 0). If both signal levels are smaller than the threshold value, the signal stemming from the direction of the driver's seat is weighted by a factor of 1 and the other signal is fully suppressed.
  • the combined signal may be subject to an additional post processing 1512, for example, a residual echo suppression (RES).
  • RES residual echo suppression
  • the combined signal is weighted by a spectral short time gain in the frequency domain, wherein the gain factor depends on the spectrum of the far end speech signal.

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  • Health & Medical Sciences (AREA)
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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • General Health & Medical Sciences (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Obtaining Desirable Characteristics In Audible-Bandwidth Transducers (AREA)
EP20030022273 2003-06-30 2003-10-01 Freisprechanlage für ein Kraftfahrzeug Expired - Lifetime EP1538867B1 (de)

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CN109845287A (zh) * 2016-09-30 2019-06-04 伯斯有限公司 用于动态声音调节的噪声估计
DE102016013042A1 (de) * 2016-11-02 2018-05-03 Audi Ag Mikrofonsystem für ein Kraftfahrzeug mit dynamischer Richtcharakteristik
US10623853B2 (en) 2016-11-02 2020-04-14 Audi Ag Microphone system for a motor vehicle with dynamic directivity
JP2020504966A (ja) * 2017-01-04 2020-02-13 ハーマン ベッカー オートモーティブ システムズ ゲーエムベーハー 遠距離音の捕捉
KR20190099445A (ko) * 2017-01-04 2019-08-27 하만 베커 오토모티브 시스템즈 게엠베하 원거리 장 사운드 캡처링
US20190348056A1 (en) * 2017-01-04 2019-11-14 Harman Becker Automotive Systems Gmbh Far field sound capturing
CN110199528B (zh) * 2017-01-04 2021-03-23 哈曼贝克自动系统股份有限公司 远场声音捕获
WO2018127359A1 (en) * 2017-01-04 2018-07-12 Harman Becker Automotive Systems Gmbh Far field sound capturing
CN110199528A (zh) * 2017-01-04 2019-09-03 哈曼贝克自动系统股份有限公司 远场声音捕获
US10869126B2 (en) 2017-05-29 2020-12-15 Harman Becker Automotive Systems Gmbh Sound capturing
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CN110692257A (zh) * 2017-05-29 2020-01-14 哈曼贝克自动系统股份有限公司 声音捕获
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DE102018110759A1 (de) * 2018-05-04 2019-11-07 Sennheiser Electronic Gmbh & Co. Kg Mikrofonarray
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US11295718B2 (en) 2018-11-02 2022-04-05 Bose Corporation Ambient volume control in open audio device
US11955107B2 (en) 2018-11-02 2024-04-09 Bose Corporation Ambient volume control in open audio device

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