DE19526098C1 - Active sound-dampening equipment - Google Patents

Abstract

The equipment is esp. for a fan in an air conduit, having a datum microphone (16) detecting the noise from the fan (13). A loudspeaker (18) at a minimum distance from this microphone feeds in a neutralising sound, while beyond the loudspeaker is an error microphone (17). A digital processing unit with filter (192) connected between the datum microphone output and the loudspeaker input applies an analogue control signal to the latter in antiphase to the analogue microphone output signal. Both microphones are at the side and clear of the air current in the conduit (12), to which they are acoustically coupled by a perforated flat component (22).

Description

The invention relates to a method for maximizing Damping effect of one in one at a noise source, especially blowers, connected, air flow Channel arranged device for active noise reduction regarding the noise emerging at the open end of the duct the genus specified in the preamble of claim 1.

An electronic noise reduction system with one Reference microphone, an error microphone, a speaker for feeding an interfering anti-sound and with a digital one having a digital filter Signal processing unit for generating a control signal for the loudspeaker is known from DE 39 08 881 A1. Reference and error microphone are on both sides of the Speaker and are within the airflow of the air-flow channel placed.

EP 0 612 057 A2 also describes a system for active noise reduction of a fan or blower generated noise emerging from a channel known with only one, next to the fan inside the duct arranged reference microphone works. Here too by means of a digital signal processing device Control signal for the loudspeaker generated, the one with  quasi quenching the noise generated by the fan interfering anti-sound radiates into the channel interior.

When installing the individual components of the device for active noise or noise reduction, also Active Noise canceling or noise abatement through anti-noise called, in such channels or channel-like structures, such as they z. B. are represented by extractor hoods, there are no certain characteristics and Dimensioning rules, how and where the designer for the Soundproofing essential constructive details useful executes, rather every constructive detail in their acoustic benefit only on a sample channel be verified, including the one emitted by the sample channel Noise level with and without sound absorbing measures is measured and compared. This iterative process means lengthy optimization without the certainty to have also achieved the maximum damping.

The invention is based on the object Specify optimization methods of the type mentioned at the beginning, that in a time and cost saving way to one if possible  maximum sound attenuation at the exit of the sample channel the device leads to active noise reduction.

The task is in a procedure in the preamble of Claim 1 defined genus according to the invention by the Features solved in the characterizing part of claim 1.

The method according to the invention has the advantage of three to be clearly structured procedural stages, in which each have a constructively optimized contribution in With regard to optimal sound absorption in the end result is achieved by the measures in the subsequent process steps are not affected again becomes. This is particularly the case if the mentioned process steps carried out in succession and the order from the first to the third step is followed.

In the method according to the invention, in a first The procedure was the coherence between the electrical Output signals from reference and error microphone measured, which should be greater than 0.9, preferably greater than 0.95. Without sufficient coherence of the temporal sound pressure curve on There is no reference microphone and at the loudspeaker location effective destructive superposition and therefore no sufficient cancellation of the noise by means of the antiphase loudspeaker noise. Only when the coherence exceeds the default value of 0.9, better 0.95 good sound absorption through the active Sound damping device possible. A measured one poor coherence value can be improved by clever arrangement of the reference microphone with respect to the Noise source and the air flow, so that the installation location and / or the installation position is changed empirically as long as  until a satisfactory coherence value is reached. Should if this is not possible, the entire pattern channel must be new be conceived if you want with the Sound absorption device the desired sound absorption achieve.

If the coherence is good, the second Process step the "effective sound propagation time" in the channel between the reference microphone and the error microphone. How is known must between the reference microphone and the Speakers a minimum distance should be provided. This The minimum distance is due to the speed of sound in the channel and the time given by the digital Signal processing unit needed to due to the Output signal of the reference microphone the inverse Sound pressure curve to be calculated and by means of a Radiate control signal through the speaker. Is if this minimum distance does not exist, then that comes from Speakers correctly radiated anti-phase sound in time simply too late and can no longer be present at the loudspeaker location the noise that already passes the speaker location interfering enough to cancel. Due to the known computing and delay times in the Signal processing unit and the known This minimum distance is the speed of sound in the canal known and specified for the installation of the speaker.

For example, B. that for example by means of cross correlation the electrical output signals of reference and Error microphone determined the time difference between "effective sound propagation time" corresponding to the signals multiplied by the known Sound propagation speed in the channel one "acoustic" sound path that is smaller than that  geometric installation distance between reference microphone and Loudspeaker or error microphone, this is an indication that the interference noise received by the error microphone itself from the noise source is not essentially about the Air flow in the duct, but essentially through the Channel wall spreads as (faster) structure-borne noise; because the sound propagation speed in fixed Bodies is much larger than in air. So that Structure-borne noise is not that of the digital Signal processing unit generated control signal for the Loudspeaker influences and thus one for the destructive Superposition of noise and anti-noise strong faulty sound pressure curve at the loudspeaker location causes are now constructive measures for Structure-borne noise decoupling of the reference microphone and the Error microphones. Such constructive Measures include, for example, the reference and the error microphone resilient to a large extent support and the mass in the attached to the channel Suspend the microphone module. At the same time you will do well to take constructive measures Structure-borne noise to make the Structure-borne noise transmission of the noise, which yes not with combats the device for active noise reduction can be reduced. Such constructive measures for passive noise reduction are, for example, in the coating or lining of the channel sound absorbing material. It should be noted that Structure-borne noise to ensure good coherence of the Microphone output signals leads and that therefore in the first Process step structure-borne and airborne sound transmission cannot be separated.

Now from the "effective sound propagation time"  "effective sound path" is determined, the rough with the geometrical installation distance of the reference microphone and The third microphone can match the reference microphone in the channel Process step are carried out. Here, the temporal length of the impulse response of the now constructive determined overall system determined by measurement. As is well known, the impulse response is the inverse Fourier transform of the transfer function of the Entire system to replicate the signal processing is. If the impulse response of the overall system at Signal processing not considered in full length, is that synonymous with a falsified Transfer function and therefore defective Compensation results. The third step measured impulse response length must now be below Taking into account the sampling frequency of the digital Signal processing unit in the number of Find filter coefficients of the digital filter, more specifically, the number of filter coefficients of the Digital filter equal to the product from the Impulse response length with the sampling frequency. The number the error coefficient can be larger than this Product, but because of the aimed small Signal processing times and as little as possible Storage spaces for the digital filter, however, as far as possible is avoided. Based on these measurements, know that conceived filters on too few filter coefficients, so can alternatively also reduce the sampling frequency so far until there is agreement again. Are there however, sampling theorem and filter group delays too note.

Advantageous further developments and refinements of The inventive method result from the others  Claims.

According to a preferred embodiment of the The coherence and time shift of the process Output signals from reference and error microphone using one at the outputs of the reference and error microphone connected FFT analyzer measured. The length too the impulse response is determined using the FFT analyzer determines the transfer function of the overall system missing between the two microphones. By Transformation of the transfer function in the time domain the impulse response is then received.

In an alternative embodiment of the method can the impulse response in the time domain also by triggering a acoustic Dirac shock at the noise source directly with the Error microphone can be measured, although the apparatus expenditure for the measuring device and Device for generating the Dirac shock can be larger.

The method according to the invention is in one of the Drawing shown embodiment of a air-flow channel with a device for active Noise reduction described below. Show it:

Fig. 1, partly in section a front view of a fume extractor hood, with a schematically illustrated in block diagram the electroacoustic device for the active sound attenuation,

Fig. 2 is an enlarged perspective view of a microphone module of the sound attenuation device.

The sample extractor hood 10 shown partially in section in FIG. 1 comprises an intake funnel 11 and a channel 12 continuing from it, which here is rectangular in cross-section here, but can also have any other hollow cross-sections, and which has an outgoing air opening, here not shown, is connected. A fan 13 for generating an air flow symbolized by arrows 14 is arranged in the channel 12 . The channel 12 is of sufficient length so that there remains a free flow path between the suction funnel 11 and the fan 13 , in which an electroacoustic device for active sound damping, hereinafter referred to as sound damping device 15 , is arranged.

The sound attenuation device 15 comprises a reference microphone 16 arranged near the fan 13 , an error microphone 17 and at least one loudspeaker 18 as well as a digital signal processing unit 19 and a controller 20 . The reference microphone 16 is used to record the noise emitted by the fan 13 . The loudspeaker 18 is arranged on the side of the reference microphone 16 remote from the fan 13 , viewed in the longitudinal axis of the channel, at a distance from the reference microphone 16 . The minimum distance is predetermined by the processing time of the signal processing unit 19 and the known sound propagation speed in the channel 12 and is, for example, 70 cm. The loudspeaker 18 is inserted in a wall opening 21 in the channel wall 121 of the channel 12 . The error microphone 17 is arranged on the side of the loudspeaker 18 remote from the reference microphone 16 at a distance from it, which is, for example, 25 cm. The output of the reference microphone 16 is connected to the input of the signal processing unit 19 and the output of the error microphone 17 via an analog-digital converter 201 to the input of the controller 20 . The digital signal processing unit 19 known per se, which has, inter alia, an analog-digital converter 191 , a digital filter 192 , in particular FIR filter, and a digital-analog converter 193 , outputs the analog output signal of the reference microphone 16 after amplification, filtering, delay time etc. as an analog control signal to the loudspeaker 18 , the control signal being generated in such a way that the loudspeaker 18 feeds sound into the channel 12 , which is broadband and in phase opposition to the noise generated by the fan 13 , in such a way that this so-called. Antisound interferes with quasi-extinguishing noise. The controller 20 connected with its output to the input of the digital signal processing unit 19 improves the signal processing of the output signal of the reference microphone 16 and represents a so-called backward control known per se.

The two microphones 16 , 17 are arranged laterally outside the air flow 14 guided in the channel 12 and are acoustically coupled to the latter via a perforated surface element 22 . The surface element 22 preferably consists of a honeycomb structure with a plurality of longitudinally adjacent, open-ended hollow bodies, the hollow body axes of which are aligned transversely to the direction of flow of the air flow 14 in the channel 12 . Each microphone 16 , 17 is inserted into a microphone module 23 which is attached to the channel 12 . The two microphone modules 23 for the reference microphone 16 and for the error microphone 17 are constructed identically and are shown in perspective in FIG. 2. Each microphone module 23 is placed on a wall opening 24 or 25 in the channel wall of the channel 12 , the acoustically completely transparent surface element 22 completely filling the wall opening 24 or 25 and being flush with the surface of the channel inner wall.

The microphone module 23 comprises a hood 26 with fastening flanges, not shown here, which are angled in the region of the hood opening and can be attached to the channel 12 . The surface element 22 closes off the hood opening and is held in the hood 26 . It projects so far out of the hood 26 that after the hood 26 is placed on the outside of the channel 12, the front surface of the surface element 22 is flush with the inner wall of the channel 12 . The hood 26 thus encloses a chamber 28 arranged laterally on the channel 12 , which is connected to the interior of the channel via the holes in the surface element 22 . The reference microphone 16 or the error microphone 17 is arranged in this chamber 28 , the chamber volume remaining between the surface element 22 and the hood 26 being completely filled with fiber material 29 , for example rock wool, cotton wool or the like. The microphone 16 or 17 can be completely embedded in the fiber material 29 and thus be spatially fixed within the hood 26 or, as will be explained later, as can be seen in FIG. 2, can be attached to the hood 26 in the interior of the chamber 28 his.

After the installation of the components of the active noise damping device 15 in the channel 12 of the sample hood 10 as described above, the damping effect of the noise damping device 15 with respect to the noise emerging at the open channel end, i.e. from the opening of the suction funnel 11 , is still to be optimized, for what purpose this optimization to achieve time and cost saving, the procedure described below is used. In this case, an FFT analyzer 30 is used, and the outputs from the reference microphone 16 and error microphone 17 to the digital signal processing unit 19 or the controller 20 are separated from the latter and connected to the corresponding inputs of the FFT analyzer 30 , which is shown in FIG the two switches 31 and 32 is symbolized. The optimization process is structured in three stages, whereby the individual process steps are expediently carried out in succession and in the order described:

In a first method step, the coherence between the electrical output signals of the reference microphone 16 and the error microphone 17 is measured. The two-channel FFT analyzer 30 , in which this function is implemented, serves as the coherence measuring device. The aim is to achieve a coherence value that is close to "1", but in any case must be greater than 0.9. A good coherence value is 0.93, a very good 0.95. If a sufficiently high coherence value is not measured, the installation location and / or the installation position of the reference microphone 16 must now be changed until a maximum coherence value is obtained which is greater than 0.9. For example, an improved coherence value can be achieved by arranging the microphone module 23 at a different location on the channel 12 . Thus, it may be more favorable to arrange the microphone module 23 with the reference microphone 16 behind the duct curvature or in the duct curvature itself, as is indicated in broken lines in FIG. 1, with the reference microphone 16 in the flow direction of the air flow 14 . For an improved coherence value, however, it can also be favorable that the microphone module 23 with reference microphone 16 is arranged on the wall of the channel 12 having the smaller curvature in front of or behind the channel curvature. These installation locations and installation positions of the reference microphone 16 are to be varied until a maximum coherence value is obtained. Possibly. the installation location of the error microphone 17 must also be moved or its installation position changed. If no coherence value over 0.9 can be achieved in any installation position, the design of the channel 12 in the sample hood 10 must be reconsidered, the cross-section and the length of the channel 12 and the design of the curvature possibly having to be re-interpreted.

If there is sufficient coherence, the second process step of the optimization process is now initiated. The cross-correlation function of the electrical output signals from the reference microphone 16 and error microphone 17 is formed by means of the FFT analyzer 30 and the time shift of the output signals is determined therefrom. This time shift represents the effective sound throughput time through the channel from the location of the reference microphone 16 to the location of the error microphone 17. This time shift is multiplied by the known speed of sound inside the channel 12 (c = 340 m / s). The resulting value is compared with the geometrical installation distance of the reference microphone 16 and error microphone 17 . If these two values roughly match, then no further action can be taken and the third step can be initiated.

If the sound path deviates significantly from the geometric installation distance, it is in particular much smaller than this, constructive measures for decoupling structure-borne sound from the reference microphone 16 and error microphone 17 must be carried out. The success of these measures can be assessed by the fact that the difference between the measured sound path and the geometrical installation distance becomes smaller and smaller and finally reaches an acceptable tolerance range. A successful measure for decoupling structure-borne noise from the two microphones 16 , 17 consists in increasing the inertial mass of the two microphones 16 , 17 and thereby obtaining a deeply tuned spring-mass system. This is outlined schematically in FIG. 2. The reference microphone 16 (or the error microphone 17 ) is inserted into a central bore of a somewhat 1 cm long brass ring 31 and is supported relative to the brass ring 31 by a rubber bushing 32 surrounding the microphone 16 or 17 . The brass ring 31 is held on the hood 26 by means of two tension springs 33 and 34 which act diametrically on the brass ring 31 . By using a larger diameter brass ring 31 , the structure-borne noise decoupling can be increased if necessary.

The overall system of duct 12 and sound damping device 15 is now structurally fixed, and in the third and last method step the time length of the impulse response of the overall system is determined by measurement. Since the impulse response is known to be the inverse Fourier transform of the transfer function to be simulated of the channel area between the reference microphone 16 and the error microphone 17 , the transfer function of this sound transmission system is determined by means of the FFT analyzer 30 and this transfer function is transformed into the time domain. In order to determine whether the impulse response is fully taken into account in the signal processing, the temporal impulse response length is multiplied by the sampling frequency of the digital signal processing unit 19 . The resulting number is specified as the minimum number of filter coefficients of the digital filter 192 . If the digital filter 192 already provides a number of storage locations for the filter coefficients which is equal to or greater than this determined number, then no further action is to be taken. The sound damping device 15 is optimally designed. If there are too few memory locations, then either the number of memory locations for the filter coefficients must be increased or the sampling frequency of the signal processing unit 19 must be reduced to such an extent that the number obtained by the multiplication corresponds to the intended number of filter coefficients.

Alternatively, the impulse response length can also be determined in that an acoustic Dirac shock is triggered at the location of the blower 13 , for example by a bang, and the impulse response is measured directly with the error microphone 17 . The length of the impulse response can also be easily measured here and used to check the filter coefficients provided in the digital filter.

The invention is not based on the described Embodiment limited. For example, the Time difference between the electrical output signals of reference and error microphone instead of from the Cross correlation of these output signals also from the Measurement of the transit times of the sound waves to the two Microphones can be determined. Such a measuring method for the transit times are, for example, in DE 31 16 586 C2 described.

Claims (8)

1.Method for maximizing the damping effect of a device arranged in an air-flow duct connected to a noise source, in particular a blower, for active noise damping with regard to the noise emerging at the open end of the duct, which has a reference microphone ( 16 ) arranged next to the noise source ( 13 ) for detecting the from the noise source (13) outgoing interference sound, at least one in the minimum longitudinal distance thereof disposed loudspeaker (18) for supplying a quasi destructively interfering with the interference noise anti sound, a valve disposed on the side facing away from the reference microphone (16) side of the speaker error microphone (17) and a between the output of the reference microphone ( 16 ) and the input of the loudspeaker ( 18 ) arranged digital signal processing unit ( 19 ) with digital filter ( 192 ) which to the loudspeaker input an analog phase to the analog output signal of the reference microphone ( 16 ) Ste Uersignal sets, the reference and the error microphone ( 16 , 17 ) each arranged laterally outside of the air flow forming in the channel ( 12 ) and acoustically coupled thereto via a perforated surface element ( 22 ), characterized in that in a first method step the coherence between the electrical output signals of the reference and error microphone ( 16 , 17 ) is measured and the installation location and / or the installation position of the reference microphone ( 16 ) with respect to the noise source ( 13 ) is changed until a maximum coherence value results that in a second process step determines the time shift between the output signals of the reference and error microphone ( 16 , 17 ), and the effective sound path resulting from the multiplication of the time shift by the speed of sound in the interior of the channel with the geometric installation distance of the reference and error microphone ( 16 , 17 ) is compared and at significant Deviation of the comparison values, constructive measures for decoupling structure-borne noise from the reference and error microphone ( 16 , 17 ) are carried out until the deviation is minimized, and that in a third method step the temporal impulse response length of the channel area between the reference and error microphone ( 16 , 17 ) existing sound transmission system is measured and the number resulting from the multiplication of the impulse response length and the sampling frequency of the signal processing unit ( 20 ) is specified as the minimum number of filter coefficients of the digital filter ( 192 ). 2. The method according to claim 1, characterized in that the first, second and third process steps in the sequence mentioned are carried out one after the other. 3. The method according to claim 1 or 2, characterized characterized as a maximum coherence value  A value greater than 0.9 is required. 4. The method according to claim 3, characterized in that, in order to achieve the maximum coherence value in the first method step, the installation location and / or installation position of the error microphone ( 17 ) may be changed. 5. The method according to any one of claims 1 to 4, characterized in that the coherence and the time shift of the output signals of the reference and error microphone ( 16 , 17 ) measured by means of an FFT analyzer ( 30 ) connected to the outputs of the reference and error microphone become. 6. A method according to any one of claims 1 to 5, characterized in that the impulse response by transformation with a at the outputs of the reference and error microphone (16, 17) analyzer FFT recovered connected (30) the measured transfer function of the sound transmission system in the time domain becomes. 7. The method according to any one of claims 1 to 5, characterized in that the impulse response after triggering an acoustic Dirac shock at the noise source ( 13 ) is measured directly with the error microphone ( 17 ). 8. The method according to any one of claims 5 to 7, characterized in that the time shift between the output signals of the reference and error microphone ( 16 , 17 ) is determined from the cross-correlation function of these output signals.