Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
  • H04R3/00—Circuits for transducers
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
  • H04R25/00—Electric hearing aids
  • 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; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
  • H04R25/00—Electric hearing aids
  • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
  • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing

Definitions

• the present invention is generically directed on a technique for so-called "beam forming" on acoustical signals.
• the amplitude of the resulting signal A_. is proportional to the sine of the signal frequency ⁇ and to the distance p.
• f r becomes approx. 7 kHz.
• Such techniques for beam forming are well-known and have been realised using analogue signal processing, as e.g. shown in the US-A-2 237 298, US-A-4 544 927, US-A-4 703 506, US-A-5 506 908 or using digital signal processing, both in time or in frequency domain, as shown in the EP-A-0 381 498 (time domain) or in the US-A-5 581 620 (frequency domain) .
• the resulting signal is dampened at low frequencies, which results in a bad signal to noise ratio.
• the directivity index is very sensitive to matching of the individual microphone cells, especially at low frequencies.
• the distance p between the microphone cells should be large (> 12 mm) for audio range.
• the directivity largely depends upon the number of microphone cells and thus on the complexity of the overall arrangement .
• the US-A-4 653 102 proposes the use of two directional microphones aimed in target direction and of a third microphone aimed in opposite direction.
• the signal of the third microphone supposedly only containing noise is used to shape the response of the two primary microphones.
• This technique obviously has the drawback within reverberating rooms, where the desired signal is reflected on walls, floor, ceiling and furniture and is therefore considered as noise by the system.
• This technique is further unhandy as making use of at least three microphones .
• the US-5 539 859 proposes a technique wherein reception characteristic is logged in on that direction wherefrom the highest energy impinges on a pair of microphones and considered in the sound environment. Principally, all sound impinging from directions other than from highest energy direction is considered as noise and its reception is cancelled.
• an analogue to digital conversion and subsequent time to frequency domain conversion is performed on the output signals of two microphones. Exploiting the knowledge of the fixed mutual distance between the two microphones, wherefrom phase difference of the impinging signal spectra is dependent, there is determined the mutual phasing and thus impinging direction of highest energy sound signals, i.e. direction of highest en- ergy sound source within the acoustical surrounding. Signals impinging from that direction are amplified by means of in- phase shifting and adding similarly to an auto correlation technique, whereby signals from other impinging angles are cancelled as noise.
• the preferred apparatus according to the present invention is a hearing aid apparatus, and especially a one ear hearing aid apparatus . It is a further object to provide such method and apparatus with good frequency response in the audio band, i.e. between approx. 0,1 and 10 kHz.
• Still a further object of the present invention is to provide such method and apparatus which allow high signal to noise ratio realisation without unwanted side-lobes and with easily variable beam form, e.g. for acoustical zooming.
• the inventive method comprises the steps of repetitively determining from sig- nals dependent from the acoustical signals a respective mutual delay signal according to reception delay at the at least two transducers; subjecting a signal dependent from the output signal of at least one of the at least two transducers to filtering with a filtering transfer characteristic; and of control- ling the filtering transfer characteristic in dependency of the mutual delay signal; further exploiting a signal dependent from the output signal of the filtering as electrical reception signal .
• the inventive acoustical sensor apparatus comprises at least two acoustical/electrical transducers, arranged at a predetermined mutual distance in target direction, a time delay detection unit, which has at least two inputs and an output, the inputs thereof being respectively operationally connected to the outputs of the two transducers, whereby the time delay detection unit generates an output signal in dependency of the time delay of acoustical signals, impinging on the at least two spaced apart transducers, preferably a time domain to frequency domain converter unit generating the output signal of said time delay detection unit in frequency domain; a weighing unit with a predetermined weighing characteristic and with an input and with an output, whereby the input thereof is operationally connected to the output of the time delay detection unit and preferably receiv- ing the signal at said output of said time delay detection unit in frequency domain mode; with a filter unit with a controllable transfer characteristic, which has at least one input, a control input and an output and whereat the input is operationally connected to
• Fig. 1 A functional block diagram of a two-cell directional microphone arrangement according to the prior art principle of "delay and sum";
• Fig. 2 the first order cardoid amplification characteristic of prior art arrangement according to fig. 1;
• Fig. 3 departing from the prior art arrangement of fig. 1 a further arrangement following up the technique of "delay and sum" for realising second order characteristic;
• Fig. 4 the second order amplification characteristic as realised by the prior art arrangement according to fig. 3;
• Fig. 5 in dependency of frequency the amplification characteristic of the arrangement according to fig. 1 or 3 at maximum amplification impinging angle of acoustical signals;
• Fig. 6 a simplified functional block diagram of an inventive apparatus operating according to the inventive method and further showing the sequence of process signals;
• Fig. 7 in a representation according to fig. 6 a first preferred realisation form of an inventive apparatus operating according to the inventive method;
• Fig. 8 in an inventive apparatus operating according to the inventive method according to fig. 6 a further preferred form of realisation of a time delay detection unit;
• Fig. 9 a polar diagram of signals as realised by the embodi- ment of fig. 8 for explaining operation of a comparator unit as provided in the fig. 8 embodiment;
• Fig. 10 the course of comparison results in dependency of impinging angle of an acoustical signal and as realised by the embodiment according to fig. 8;
• Fig. 11 a preferred form of realising superimposing result signal dependency from impinging angle of an acoustical signal at an embodiment according to fig. 8 ;
• Fig. 12 in a representation according to fig. 10 the course of comparison results as realised with a preferred embodiment resulting in the fig. 11 dependency;
• Fig. 13 in polar diagrammatic representation the dependency of superimposing result signals from impinging angle of acoustical signals and from frequency as realised by the embodiment according to fig. 8;
• Fig. 14 a preferred realisation form of the fig. 8 embodiment, additionally counteracting frequency dependency as shown in fig. 13;
• Fig. 15 in a representation according to fig. 13 the. depend- ency of the superimposing result signal with normalisation as realised by the embodiment of fig. 14 with a first preferred normalisation frequency function;
• Fig. 16 a representation according to fig. 15, realised with a second preferred normalisation frequency function at the embodiment of fig. 14;
• Fig. 17 a first (rigid line) and second (dashed line) preferred realisation form of amplitude filter characteristic at the embodiment of fig. 6 or 7;
• Fig. 18a the effect of the amplitude filter amplitude versus amplitude transfer characteristic according to fig. 17
• Fig. 18b the representation of the output signal of a time delay detection unit passed through the amplitude filter with a transfer characteristic according to fig. 17
• Fig. 19 the spectrum of an acoustical signal converted into electrical and input to a controllable frequency fil- ter as provided by the present invention according to fig. 6;
• Fig. 20 the electrical reception result signal realised by amplitude filter characteristic according to fig. 17 (rigid line) and reception signal as exemplified in fig. 19 at the inventive embodiment according to fig.
• Fig. 21 the resulting dependency of amplification from impinging angle of an acoustical signal as realised by the fig. 17 amplitude filter characteristic (rigid and dashed lines) ;
• Fig. 22 the amplification versus impinging angle characteristic as realised by the fig. 6 or fig. 8, 14 embodiments of the invention, making use of an amplitude filter characteristic with a maximum to minimum spec- tral amplitude transfer behaviour;
• Fig. 23 in a simplified signal/functional block diagram a further preferred embodiment of the invention
• Fig. 24 in a signal flow, functional block representation, a further mode of realisation of the time delay detection unit as shown in fig. 6 and
• Fig. 25 in a signal flow, functional block representation, a further mode of realisation of the time delay detection unit following the technique as shown in fig. 8 or fig. 14.
• At least two acoustical/electrical transducers 1 and 2 are provided with a predetermined mutual distance p along axis a. Acoustical signals IN are received by the transducers 1 and 2 as they impinge from different spatial directions ⁇ . The acoustical signals IN have frequency spectra which vary in time. Output signals of transducer 1, S x (t, ⁇ ) and of transducer 2, S 2 (t, ⁇ ), are formed as electrical signals at the output of the transducers 1 and 2.
• the acoustical signals IN impinge on the transducers 1 and 2 with a time delay dt, which may be expressed by the phase difference ⁇ ⁇ at each spectral frequency ⁇ according to
• the time delay dt ⁇ becomes equal for all spectral components at the different ⁇ .
• the output signals S. and S 2 of the transduc- ers 1 and 2 are operationally connected to the respective inputs of a time delay detection unit 10, which generates an output signal A 10 according to the spectral distribution of time delays dt ⁇ , which are, as was explained, a function of the impinging angle ⁇ at which the respective frequency components impinge on the transducers 1 and 2 and thus in fact of ⁇ ⁇ .
• a possible spectrum of output signal A 10 is also shown in fig. 6.
• This spectrum varies in time according to the time variation of impinging acoustical signal IN.
• the output signal A 10 of time delay detection unit 10 is input to a weighing unit 12.
• a 12 results from respectively weighing the spectral amplitudes of A 10 according to the characteristic W.
• the weighing unit 12 determines with its characteristic W the beam shape .
• the output signal A 12 is applied to a filter unit 14 with a controllable transfer filter characteristic.
• each spec- tral line of the time varying spectrum of the output signal S- tjCo) is amplified or attenuated according to the controlling spectrum W ⁇ • A 10 ⁇ .
• unit 14 is a filter unit for input signal S 1 at which the transfer characteristic is varied, as controlled by A 12 .
• the weighing unit 12 In dependency of the kind of filter unit 14 the weighing unit 12, generally spoken, calculates adjustment of filter characteristic determining coefficients as a function of A 10 .
• the beam form can be adjusted and thus acoustical zooming is realised.
• both transducer output signals may be advantageous to subject both transducer output signals to a controlled filtering at unit 14.
• fig. 7 there is shown a first preferred form of realisation of the inventive principle according to fig. 6.
• the output signals S x and S 2 are first converted from analogue to digital form in respective analogue/digital converters 16 and 17.
• the digital output signals of the respective converters 16 and 17 are input to respective complex time domain/frequency domain converters 18 and 19.
• the output spectra S 1 (t, ⁇ ) and S 2 (t, ⁇ ) of converters 18, 19 are input to the spectral time delay detection unit 10'.
• Unit 10' computes according to formula (1) the phase difference spectrum ⁇ ⁇ divided by the respective frequency ⁇ to result in an output signal spectrum A 10 ' according to the time delay dt ⁇ as was explained in connection with fig. 6.
• the output signal of the time delay detection unit 10', A 10 ' is further treated, as was explained in connection with fig. 6, by the weighing filter unit 12 and the controllable filter unit 14.
• the unit 10' operates. Out of the spectral phase distribution ⁇ ln of signal S- and ⁇ 2n of signal S 2 the time delay dt ⁇ is calculated for each spectral line within an interesting spectral band.
• the time domain to frequency domain conversion units 18 and 19 perform complex (real and imaginary) operation.
• a second preferred realisation form of the present invention and especially as concerns realisation of the time delay detection unit 10, shall be explained with the help of fig. 8 and 9.
• the output signal of one of the transducers is fed to a time delay unit 20, wherein, in a first form of this realisation, signal S ⁇ is time delayed by a predetermined frequency independent time delay ⁇ .
• signal S x accords thus with signal A x .
• the output signal of time delay unit 20 thus accords with signal A. 1 of fig. 1.
• the time delay signal according to A x ' is superimposed to the output signal S 2 (t, ⁇ ) from transducer 2 at a superimposing unit 23 according to unit 3 of fig. 1, thus resulting in an output signal according to A_.(t, ⁇ ) of fig. 1.
• the output signal A_.(t, ⁇ ) depends from the impinging acoustical signal direction ⁇ ac- cording to the first order cardoid beam of fig. 2, which cardoid function nevertheless varies with frequency ⁇ .
• the output signal A., of superimposing unit 23 and e.g. the output signal S 2 (t, ⁇ ) from transducer 2 are input to a ratio unit 25, as a comparator unit .
• a 20 according to fig. 9 is indicative of the impinging angle ⁇ 0 .
• the division unit 25 of fig. 8 there is formed for each spectral component amplitude the ratio of A. to A., wherefrom there results a signal spectrum at the output of division unit 25 with a ratio spectrum.
• the spectrum of A 10 according to fig. 6 thus becomes the spectrum of an amplitude ratio which nevertheless is indicative of the impinging angle ⁇ , at which each frequency component of the spectrum of the acoustical signal impinges with respect to the axis a of the two transducers (see fig. 6) .
• the dotted line block indicates the delay detection unit 10 according to fig. 6. Further signal processing is performed as was explained by means of fig. 6, i.e. via weighing unit 12 and controllable filter unit 14.
• the output ratio signal of unit 25 is a measure for the time delay dt ⁇ and is input to the weighing unit 12.
• This amplitude ratio is shown for ⁇ at unit 20 of fig. 8 being selected to be
• the cardoid function as shown in the figures 2, 9 and 11 is only valid for one specific frequency considered.
• the outputs of the transducers 1 and 2 are converted into digital form by respective analogue to digital converters 16, 17 and the resulting digital signal of transducer 1 is time delayed by a time delay ⁇ ', which is larger than p/c.
• the output signal S 2 of transducer 2 is further converted into frequency domain by a linear (not complex) time to frequency domain conversion unit 18 ' , whereas the output signal A., of superimposing unit 23 is converted to frequency domain at a linear time to frequency domain conversion unit 19 ' .
• the frequency dependent polar diagram according to fig. 13 is taken into ac- count by a normaliser unit 30 which is in fact a filter.
• the transfer characteristic of the filter is selected proportional to 1/ ⁇ . This results in a frequency dependency of the pole diagram as is shown in fig. 15 for the same distance and frequency values as shown in fig. 13.
• a further, even improved normalisation function or filter characteristic at unit 30 of fig. 14 is achieved when the filter characteristic is selected as a function l/sin( ⁇ ). The result is shown in fig. 16. The characteristics match well from 0.5 to 4 kHz.
• a further advantage of this normalisation technique is improved sensitivity in backwards direction. This improved sensitivity may be exploited for adaptive beam forming, that is for selectively eliminating noise sources from the rear side.
• Fig. 24 shows in block diagram form, that the signal A 10 (dt ( ) may also be generated as the output signal of a comparator unit 60 to which on one hand the output signal of an omnidirectional transducer 61, having equal amplification of its acousti- cal/electrical reception characteristic substantially irrespective of the impinging angle ⁇ and the output signal of a directional transducer 62 with selected, beam shaped reception characteristic are led to.
• time delaying ⁇ may also be performed by one of the transducers itself.
• the output signals of the transducers 1 and 2 are first converted by respective analogue to digital converters 16 and 17 and then by respective time domain to fre- quency domain converters 18, 19 finally into frequency domain.
• One signal, as an example S 2 of the converted output signals of the transducers, which, after time to frequency domain conversion may be represented as a spectrum of S 2 ⁇ pointers, is converted to its conjugate complex pointers at a conversion unit 50. At the output of this unit 50, the conjugate, complex pointers S * 2 ⁇ are generated.
• This spectrum S 2 * and the pointer spectrum S x are multiplied to form the scalar product spectrum S 3 in a multiplication unit 52.
• the pointers S 3( of spectrum S 3 have a phase angle with respect to the real axis, which is ⁇ ⁇ .
• ⁇ ⁇ ⁇ • (p/c) • cos( ⁇ ⁇ )
• a conversion unit 53 forms the imaginary part of the pointers S 3 ⁇ and a further unit 54 forms the amplitudes I S 3 ⁇ I of these pointers .
• All the units 50, 52, 53, 54, 55 and 56 are preferably realised in one calculator unit .
• each dt M spectral line amplitude of signal A 10 (see fig. 6) is attenu- ated to zero, if such amplitude is below or above predetermined values dt min ⁇ , dt mas ⁇ and is set to be "one" if such spectral component amplitude is between these two values.
• Such selection of weighing function W results in an output signal spectrum A 12 , as shown in the figures 18a and 18b.
• Fig. 19 shows a spectrum example of signal S- L .
• All spectral lines of S 1 (Fig. b) are amplified by the value 1 according to A 12 or are nullified ac- cording to zero values of A 12 .
• the weighing function I of fig. 17 is applied to the technique according to fig. 7 there results a beam form as shown in fig. 21 in strong lines.
• If an amplitude filter characteristic is applied as shown by II in fig. 17, there results the characteristic as shown in fig. 21 in dashed line .
• Fig. 22 shows the resulting beam if in analogy to fig. 17 and with an eye on figures 8 and 9 all ratio values which exceed (A-VA- it-ax are discarded. This is realised by the amplitude filter characteristic as also indicated in Fig. 22.

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• Physics & Mathematics (AREA)
• Engineering & Computer Science (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• Health & Medical Sciences (AREA)
• General Health & Medical Sciences (AREA)
• Neurosurgery (AREA)
• Otolaryngology (AREA)
• Circuit For Audible Band Transducer (AREA)
• Measurement Of Velocity Or Position Using Acoustic Or Ultrasonic Waves (AREA)
• Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
• Investigating Or Analyzing Materials By The Use Of Ultrasonic Waves (AREA)

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• 1997
  • 1997-08-20 EP EP97114413A patent/EP0820210A3/fr not_active Withdrawn
• 1998
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