EP2378513B1 - Method and system for active noise reduction - Google Patents

Description

The present invention relates to a method for active noise reduction of the sound field generated by a vibrating radiating surface of a primary sound source and to an active noise reduction system for carrying out such a method.

Active noise reduction systems and methods, which are also referred to as antinoise systems or methods, typically use one or more secondary sound sources in the form of loudspeakers, one or more sensors and a control device for reducing the primary sound radiated from a source of noise also referred to as a primary source connected to the speakers and the sensors. The control device controls the loudspeakers on the basis of the signals supplied by the sensors in such a way that the entire sound field generated by the combination of the primary sound source and the loudspeakers is favorably influenced in the sense of the aim of sound reduction. In this case, one or more sensors can be used to generate reference signals, on the basis of which control signals for the secondary sound sources are determined, and one or more further sensors can serve as error sensors, with the aid of which the quality of the control signals is checked and their determination is adjusted as needed.

The desired influence of the sound field can be based on various physical mechanisms. In addition to the most well-known case of destructive interference, it is also possible that sound is reflected at the locations of the secondary sound sources, that sound is absorbed by the secondary sound sources and the sound energy is dissipated via the respective actuators or that the primary sound source and the secondary sound sources influence each other such that that of the combination of primary and secondary sound sources radiated total sound power is minimized. For the latter case, the interference affects a reduction in the ability of sound sources to emit sound. This may for example be based on the fact that the secondary sound sources reduce the effective resistance of the primary sound source by acting on the acoustic modes of the sound field or act on the air molecules located in front of the primary sound source, that they less resistance to the movement of the radiation surface of the primary sound source , There is always the difficulty that the contribution of the secondary sound sources to the sound field must not overcompensate the benefits achieved in a negative way.

For example, one known type of active noise reduction employs one or more error microphones as sensors, each of which locally measures the sound pressure produced by all existing sound sources, including the primary sound source and one or more secondary sound sources. The measurement results are processed by the control device, which then controls the secondary sound sources in such a way that the sound pressure at the microphones is minimized as far as possible by destructive interference and / or sound reflection at the locations of the secondary sound sources. This can be a local noise reduction can be achieved at the microphone positions. This principle, which is an example of a sound pressure-based control, has the disadvantage that the local noise reduction at the microphone positions is generally accompanied by a noise amplification in other areas. Furthermore, only the local sound effect is influenced in the form of the sound pressure, without combating the cause in the form of the sound power output by the primary sound source.

Further exemplary sound-pressure-based controls, which include, among others Elliot, SJ et al., In Flight Experiments on the Active Control of Propeller-induced Cabin Noise, Journal of Sound and Vibration (1990), No. 140 (2), pages 219-238 are known and have the goal of global noise reduction based on the mechanism of acting on the primary sound source via acoustic modes and have the disadvantage that the number and distribution of microphones must be chosen so that detection of the excited modes is possible. Furthermore, the scope is limited by the fact that for each application separately knowledge about the physical interaction of the speakers and sensors used and the primary sound field must be present.

Overall, for a global sound pressure minimization in cases of higher modal density, the microphones must also be distributed globally and the secondary sound sources must be arranged so that they can excite the same modes as the primary sound source. It is also problematic to take into account changing environmental influences in the implementation of the control. Further, because the microphones measure the overall sound pressure, these methods may fail in the presence of additional noise sources because the controller can not account for the contribution of the various sound sources. Despite these drawbacks, controls based on sound pressure measurements are most commonly used because the necessary measurements are technically easy to implement.

In contrast, measurement of energy quantities of the sound field are in principle better suited to be used in the context of controllers which are intended to achieve a global reduction of noise by minimizing the radiated active power of all the sound sources in the room. There is the advantage that the corresponding error sensors can be arranged in the vicinity of the secondary sound sources, whereby the installation and optimization effort can be reduced. However, the corresponding prior art proposals have presented significant problems that have led them to abandon the academic trial stage. The problems arise in part because energy quantity sensors, such as sound intensity sensors, are more expensive than simple sound pressure sensors and the complexity of the controls due to a larger number of input variables (the sound intensity is determined for example by the sound pressure and the sound velocity) and associated multi-channel configurations ,

One approach to energy-based control, for example, is from the documents Elliott, SJ et al., Power output minimization and power absorption in the active control of sound, Journal of the Acoustic Society of America (1991), No. 90 (5), pages 2501 to 2512 and Bullmore, AJ et al, The active minimization of harmonic enclosed sound fields, Part I-III, Journal of Sound and Vibration (1987), 117, pages 1-58 known. There, it is described on the basis of theoretical derivations for the example of two point sound sources that the radiated total active power of a pair of sound sources from a primary source and a secondary source is minimal if and only if the secondary source is driven in phase or phase opposition, or equal to or in relation to the primary source oscillates in phase opposition, and the secondary source emits no active sound power. The active sound power is the real part of the overall sound power usually represented by a complex size and corresponds to the actual net energy transport per second perpendicular to a surface, such as the emission surface of a sound source. In contrast, the dummy sound power represented by the imaginary part of the total sound power is due to the energy transport by medium mass, the only moved, but not compressed. In these documents, however, no feasible proposals for the selection, design and arrangement of sensors and for the design of the control are made.

Experimental studies on this approach are in the two documents Tohyama, M., Suzuki, A, Sugiyama, K., Active Power Minimization of a Sound Source in a Reverberant Closed Space, IEEE Transactions on Signal Processing (1991), No. 39 (1), pages 246 to 248 and Kang, SW, Kim, YH, Active Global Noise Control by Sound Power, ACTIVE 95: Proceedings of the 1995 International Symposium on Active Control of Sound and Vibration, Newport Beach (USA), New York, Noise Control Foundation, 1995 described.

In both cases identical loudspeakers are used as primary and secondary source. In this case, the secondary source is driven either with a control signal for the primary source in the same or opposite phase (in the case of the former document) or an opposite in relation to the drive signal for the primary source drive signal (in the case of the latter document), so that The design uniformity can not be waived, and the amplitude of the drive signal for the secondary source is set manually. Furthermore, the sensors used are either a large number of microphones randomly distributed in space or a sound intensity sensor comprising two microphones spaced apart from each other. This means a relatively high amount of hardware. Finally, no feasible approaches for a suitable control are given overall.

It is an object of the invention to provide a method and a system for active noise reduction of the sound field generated by a vibrating radiating surface of a primary sound source, the energy-based control to minimize by realize a primary sound source and a secondary sound source radiated active sound performance and expand the possibilities for using this approach over the prior art.

This object is achieved by an active noise reduction method according to claim 1 and an active noise reduction system according to claim 9. Advantageous embodiments of the method and the system are the subject of the respective dependent claims.

The method according to the invention is designed for active noise reduction of the sound field generated by a vibrating emission surface of a primary sound source. For this purpose, a physical quantity q is measured for the primary sound source with a reference sensor arranged, for example, at a given time, which characterizes the sound-generating movement of the emission surface of the primary sound source, at least in a predetermined frequency range provided for noise reduction. This quantity may advantageously be, for example, an acceleration or a velocity of or at the sound-radiating surface of a sound source. In particular accelerometers are inexpensive, reliable and simple. The time-dependent measurement yields a corresponding time-dependent reference parameter q _PQ (t), which simply represents the temporally varying measured values, such as the time-dependent acceleration of the sound-emitting surface, and contains the phase information of the primary sound source.

To minimize the radiated active sound power, a secondary sound source is provided and suitably arranged, which is preferably formed by a loudspeaker. In general, the influence of the secondary sound source on the sound emission through the primary sound source is increasing Remove distance between the two sound sources. It is therefore advantageous to arrange the sound sources in spatial proximity to each other, preferably in a mutual distance, which does not exceed half the wavelength - or at several wavelengths half the smallest wavelength - of the sound to be canceled. With a sound pressure sensor arranged in the immediate vicinity of a radiating surface of the secondary sound source and preferably on or approximately on the radiator center axis of the radiating surface, the sound pressure p (t) is measured as a function of time, at least in a predetermined frequency range provided for noise reduction. It is preferred that the distance between the sound pressure sensor and the emission surface does not exceed the value of one-tenth of the wavelength - or at several wavelengths one-tenth of the smallest wavelength - of the sound to be canceled out. In a preferred embodiment, the sound pressure sensor is a microphone.

The time-dependent measured variables q _PQ (t) and p (t) are transformed into the frequency domain by a suitable method, so that a number of complex frequency-dependent amplitudes q _PQ , _tm (each) are provided for each of a plurality of discrete and preferably immediately successive time intervals t _m. f _j ) or p _tm (f _j ) for a number and - although it is in principle also conceivable to consider only a single frequency, ie only to reduce the sound of a single frequency - preferably a plurality of frequencies f _j are obtained. Here, just as in the further course of the application underlined symbols denote complex-valued sizes, while not underlined symbols denote real sizes. The frequencies f _j. Will generally be discrete frequencies, each representing a frequency band. However, in principle it is also conceivable to use continuous or quasi-continuous frequency values. The complex amplitudes q _PQ , tm (f _j ) and p _tm (f _j ) for a specific time interval t _m are then in customary manner for the time course of q _PQ (t) or p (t) in this time interval t _m characteristic (at least in a predetermined, provided for noise reduction frequency range) and can be converted by reciprocal back transformation in the time domain back into the corresponding time signals , The totality of the frequencies f _j determines the frequency range provided for noise reduction.

wobei k_Tn (f_j) reelle Verstärkungsfaktoren sind, die in jedem Zeitabschnitt T_n zeitlich konstant sind. Sie können für jede Frequenz f_j identisch sein, sind aber bevorzugt für die Frequenzen f_j unterschiedlich, um die erzielbaren Ergebnisse zu verbessern. Durch die Bedingung, dass die k_Tn (f_j) reell sind, wird die gleich- oder gegenphasige Schwingung der Sekundärschallquelle gewährleistet, d.h. dass die Schwingungen der Abstrahlflächen von Primär- und Sekundärschallquelle bei jeder Frequenz f_j um 0° bzw. um 180° phasenverschoben sind. Dabei entsprechen Verstärkungsfaktoren k_Tn(f_j) > 0 einer gegenphasigen Schwingung und k_Tn (f_j) < 0 einer gegenphasigen Schwingung, und es ist bevorzugt, den Aufbau so zu wählen, dass die letztere Bedingung zutrifft, die insbesondere durch die oben bereits angesprochene bevorzugte relative räumliche Anordnung von Primär- und Sekundärschallquelle in einem gegenseitigen Abstand, der die Hälfte der Wellenlänge - bzw. bei mehreren Wellenlängen die Hälfte der kleinsten Wellenlänge - des auszulöschenden Schalls nicht überschreitet, erfüllt wird. Es ist darauf hinzuweisen, dass es grundsätzlich auch denkbar ist, dass die Verstärkungsfaktoren einiger Frequenzen f_j negativ und anderer Frequenzen f_j positiv sind.The secondary sound source is driven with a time-dependent drive signal in such a way that the emission surface of the secondary sound source at least at the frequencies f _j equal or out of phase oscillates to the emission surface of the primary sound source and their corresponding sound-generating movement in each of immediately consecutive discrete time periods T _n , each one or more of the discrete time intervals t _m , characterized by a specific time profile q _SQ (t) of the physical quantity q. It should be noted that the drive signal does not have to have the curve q _SQ (t) itself and generally does not have, but is designed and generated so that the emitting surface of the secondary _sound source according to q _SQ (t) oscillates. Here, between the complex frequency-dependent amplitudes q _SQ , tm (f _j ), which would be obtained by a transformation of q _SQ (t) into the frequency domain using the above transformation method, and the complex amplitudes q _PQ , _tm (f _j ) for the frequencies f _j in the discrete periods T _n each have the relationship $${\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{.}$$

where k _Tn (f _j ) are real gain factors which are temporally constant in each time period T _n . They can be identical for each frequency f _j , but are preferably different for the frequencies f _{j in} order to improve the achievable results. By the condition that the k _Tn (f _j ) are real, the same- or opposite-phase oscillation of the secondary sound source is guaranteed, ie that the vibrations of the radiating surfaces of primary and secondary sound source at each frequency f _j are 0 ° or 180 ° out of phase. In this case, amplification _factors k _Tn (f _j )> 0 correspond to an antiphase oscillation and k _Tn (f _j ) <0 to an antiphase oscillation, and it is preferable to select the construction such that the latter condition applies, in particular as already described above addressed preferred relative spatial arrangement of primary and secondary sound source in a mutual distance, the half of the wavelength - or at several wavelengths half the smallest wavelength - does not exceed the sound to be canceled, is met. It should be pointed out that in principle it is also conceivable that the amplification factors of some frequencies f _{j are} negative and other frequencies f _{j are} positive.

verwendet, wobei µ ein reeller Konvergenzfaktor ist und wie im weiteren Verlauf der Anmeldung Im[x] den Imaginärteil einer komplexen Zahl x und x* die zu einer komplexen Zahl x komplex konjugierte Zahl bezeichnet.Furthermore, the amplification _factors k _Tn (f _j ) are recalculated and set for the different time intervals T _n , wherein, if certain conditions exist, one or more of the frequencies f _j may possibly be dispensed with for recalculation. The recalculation takes place in each case for a time segment T _{n + 1 on} the basis of the amplification _factors k _Tn (f _j ) in the immediately preceding period T _n and the complex frequency-dependent amplitudes q _{PQ, tm} (f _j ) and p _tm (f _j ) a time interval t _m in the period T _n (it is also possible, but not preferred, to perform the recalculation for the period T _{n + 1} from a more recent period T _nx , in which case the corresponding references would be to T _n replaced by T _nx ). The recalculation is the update equation $${\mathrm{k}}_{\mathrm{Tn}\mathrm{+}\mathrm{1}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\mathrm{-}\mathrm{\mu \; Im}\left[{\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{p}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{*}\right]\mathrm{+}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

where μ is a real convergence factor and, as in the further course of the application in [ x ], the imaginary part of a complex number x and x * designates the number complexed to a complex number x complex.

It can be shown that this update _equation converges at the stationary primary sound source to the optimal global solution for the amplification _factors k _Tn (f _j ) in order to achieve the desired equal or opposite phase control of the secondary sound source without active power radiation of the secondary sound source. In this case, by varying the convergence factor μ, the convergence behavior can be adjusted, ie to achieve convergence in fewer steps but larger jumps or in more steps but smaller jumps. For example, a suitable value for the convergence factor μ can be determined by changing it stepwise from the value 0, which corresponds to an infinitely slow convergence, until a convergence speed desired for the particular application is reached. Here μ is generally negative. As a limit, in cases where the quantity q is an acceleration or a speed, for example, the inverse of the imaginary part of the transmission number between secondary acceleration and secondary pressure or the inverse of the real part of the transmission impedance between secondary speed and secondary pressure can be assumed.

The update equation and the corresponding adaptive control have the advantage that, in addition to a reference signal, they only require the measurement of a single sound pressure. Therefore, the hardware cost can be kept low and in particular a single-channel system can be used which, in addition to a suitable control device or control device, for example, only a reference sensor, such as an accelerometer, a microphone and a speaker. Both the secondary sound source and the sensors can advantageously be arranged in the vicinity of the sound transmission path of the primary sound source. The secondary sound source can be constructed differently than the primary sound source, so that various different suitable speakers can be used flexibly. Further, the controller simple and insensitive to changing environmental conditions, ie no re-calibration system is necessary. The method is universally applicable both indoors and outdoors, and it has been found that a significant global reduction of the sound pressure level can be achieved, in particular in the case of noise generated by concave-vibrating surfaces. For example, using a loudspeaker with a membrane area of 0.01 m ² as secondary sound source for tonal sound radiation in a room with a volume of 400 m ³ , a 4.8 m ² structure measured an average global noise reduction of 8 dB. The method is particularly suitable for stationary noise, and it is preferred that the primary sound source is a sound source that generates stationary noise.

In this connection it should also be pointed out that it is of course possible to provide more than one secondary sound source in order to take account of different areas of a structure or construction representing the primary sound source in the manner described.

In a preferred embodiment of the method, the time intervals t _m , which are used to recalculate the gain _factors k _{Tn + 1} (f _j ), are temporally spaced from each other. In this way disturbances in the method are avoided, which can result from the fact that the calculations and signal processing necessary for the recalculation and readjustment of the amplification factors can lead to a time delay and consequently the readjustment does not have an immediate effect.

mit vorbestimmten Schwellenwerten ε_j gilt, und für die übrigen Frequenzen f_j die Verstärkungsfaktoren k_Tn+1(f_j) auf Null gesetzt werden. Mit anderen Worten muss der Referenzparameter einen bestimmten Schwellenwert überschreiten, damit für die entsprechende Frequenz überhaupt ein neuer Verstärkungsfaktor berechnet wird. Dadurch kann die Ansteuerung der Sekundärschallquelle für wenig Lärm oder ohne vorhandenes Referenzsignal verhindert werden. Es wird nur die Lärmquelle aktiv in der Abstrahlung unterdrückt, bei der das System installiert ist, während Schall anderer Quellen keinen Einfluss auf das Verfahren und sein Ergebnis hat. Außerdem kann durch Wahl von ε_j der Bereich eingestellt werden, ab dem eine aktive Lärmunterdrückung ökonomisch sinnvoll ist. ε_j kann für die unterschiedlichen Frequenzen f_j unterschiedlich gewählt werden oder als Konstante über alle Frequenzen vorgegeben werden.It is preferred that the recalculation of the gain _factors k _{Tn + 1} (f _j ) is performed only for frequencies f _j for which the condition $$\left|{\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\right|\ge {\mathrm{\epsilon }}_{\mathrm{j}}$$

with predetermined threshold values ε _j , and for the remaining frequencies f _j the amplification _factors k _{Tn + 1} (f _j ) are set to zero. In other words, the reference parameter must exceed a certain threshold in order to calculate a new gain factor for the corresponding frequency at all. As a result, the control of the secondary sound source for little noise or without existing reference signal can be prevented. Only the noise source is actively suppressed in the radiation at which the system is installed, while sound from other sources has no influence on the process and its result. In addition, the range from which active noise suppression makes economic sense can be set by selecting ε _j . ε _j can be chosen differently for the different frequencies f _j or as a constant over all frequencies.

vorgegeben werden, wobei A_max,_fj die maximal zulässige Amplitude der Sekundäransteuerung bei der Frequenz f_j ist. k_max (f_j) wird in diesem Fall automatisch an die Amplitude des zur Verfügung stehenden Referenzsignals angepasst. Durch Vorgabe von Schwellenwerten kann eine Überlastung der Sekundärschallquelle verhindert werden.It is further preferred that after each recalculation of a gain _factor k _{Tn + 1} (f _j ) its amount is limited by checking whether it exceeds a predetermined maximum value k _max (f _j ) for the magnitude, and kT _{n + 1} ( f _j ) if it is exceeded, depending on whether k _{Tn + 1} (f _j ) is negative or positive, set to -kmax (f _j ) or + k _max (f _j ). This limit can, for example, by $${\mathrm{k}}_{\mathrm{Max}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{A}}_{\mathrm{Max}\mathrm{.}\mathrm{fj}}\mathrm{/}\left|{\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\right|$$

where A _max , _{fj is} the maximum allowable amplitude of the secondary control at the frequency f _j . k _max (f _j ) is automatically adjusted in this case to the amplitude of the available reference signal. By setting threshold values, overloading of the secondary sound source can be prevented.

In a preferred method embodiment, the transformation of q _PQ (t) and p (t) into the frequency domain for determination of the complex frequency-dependent amplitudes q _PQ , _tm (f _j ) and p _tm (f _j ) by means of discrete Fourier transformation. For this purpose, q _PQ (t) and p (t) must be time-discretized appropriately. This transformation process is fast, reliable and cost-effective to implement.

zu berechnen und auf Basis der komplexen frequenzabhängigen Amplituden q _{SQ, tm}(f_j) mit Hilfe einer entsprechenden komplementären Rücktransformation in den Zeitbereich den zeitabhängigen Ansteuerparameter q_SQ(t) zu erhalten, der die zu erzielende schallerzeugende Bewegung der Abstrahlfläche der Sekundärschallquelle kennzeichnet. Wird zur Transformation in den Frequenzbereich in bevorzugter Weise eine diskrete Fouriertransformation verwendet, wird die Rücktransformation in dieser Ausführungsform auf Basis der komplexen frequenzabhängigen Amplituden q _sQ,_tm (f_j) in den Zeitbereich zur Bestimmung des zeitabhängigen Ansteuerparameters q_SQ (t) mittels inverser diskreter Fouriertransformation durchgeführt.In an advantageous embodiment, the generation of the time-dependent drive signal for the secondary _{sound source comprises} the steps for each of the time intervals t _m from the complex frequency-dependent amplitudes q _PQ , _tm (f _j ) and the corresponding gain _factors k _Tn (f _j ) for the time period T _n for the frequencies f _j complex frequency-dependent amplitudes q sQ, tm (f _j ) according to $${\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

to calculate and on the basis of the complex frequency-dependent amplitudes q _{SQ, tm} (f _j ) with the aid of a corresponding complementary inverse transformation in the time domain, the time-dependent control parameter q _SQ (t), which characterizes the sound-generating movement of the emitting surface of the secondary sound source to be achieved. If a discrete Fourier transformation is preferably used for the transformation into the frequency range, the _inverse transformation in this embodiment is based on the complex frequency-dependent amplitudes q _sQ , _tm (f _j ) in the time domain for determining the time-dependent control parameter q _SQ (t) by means of inverse discrete Fourier transformation performed.

angepasst und die angepassten komplexen frequenzabhängigen Amplituden q'_{SQ, tm}(f_j) für die Rücktransformation in den Zeitbereich zur Bestimmung des zeitabhängigen Ansteuerparameters q_SQ(t) verwendet. Dabei ist G ^-1(f_j) ein vorbestimmter komplexer Zeitverzögerungskorrekturparameter, mit dessen Hilfe Phasenverschiebungen berücksichtigt werden können, die durch Signalverarbeitungsschritte und elektroakustische Umwandlungen verursacht werden können. Derartige Zeitverzögerungen und die damit unter Umständen verbundenen zusätzlichen Phasenverschiebungen können ansonsten dazu führen, dass die Abstrahlflächen von Primär- und Sekundärschallquelle bei der genannten Vorgehensweise nicht wie gewünscht gleich- oder gegenphasig schwingen. Der Zeitverzögerungskorrekturparameter kann im Verlaufe einer anfänglichen Systemkalibrierung bestimmt werden und ist ansonsten unabhängig von sich ändernden Umgebungsbedingungen.In this embodiment, furthermore, after each calculation of the complex frequency-dependent amplitudes q _SQ, tm (f _j ) according to q _{SQ, tm} (f _j ) = k _Tn (f _j ) qP _Q , tm (f _j ) for a time interval t _m the complex frequency-dependent amplitudes q _sQ , _tm (f _j ) according to $${\underset{\mathrm{}}{\mathrm{q\; \text{'}}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\underset{\mathrm{}}{{\mathrm{G}}^{\mathrm{-}\mathrm{1}}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

adapted and the adjusted complex frequency-dependent amplitudes q ' _{SQ, tm} (f _j ) for the inverse transformation into the time domain used to determine the time-dependent control parameter q _SQ (t). Here, G ^-1 (f _j ) is a predetermined complex time delay correction parameter that can be used to account for phase shifts that may be caused by signal processing steps and electroacoustic transformations. Such time delays and the possibly associated additional phase shifts can otherwise lead to the radiating surfaces of the primary and secondary sound source not swinging in the above-mentioned procedure, as desired, in the same or opposite phase. The time delay correction parameter may be determined during an initial system calibration and is otherwise independent of changing environmental conditions.

anzupassen. Der Sensorphasenkorrekturparameter ϕ(f_j) berücksichtigt, dass sich durch eine unterschiedliche Bauart des Referenzsensors und das Schalldrucksensors unterschiedliche Phasengänge und demzufolge eine Phasenverschiebung zwischen q_PQ(t) und p(t) bzw. den entsprechenden frequenzabhängigen komplexen Amplituden ergeben kann, die - wie sich aus der Aktualisierungsgleichung unmittelbar ergibt - in nachteiliger Weise die Berechnung der optimalen Verstärkungsfaktoren unmittelbar beeinflusst. Der Sensorphasenkorrekturparameter ϕ(f_j) kann im Verlaufe einer anfänglichen Systemkalibrierung durch Bestimmung der tatsächlichen Phasenlage der durch die Sensoren gemessenen physikalischen Größen bestimmt werden und ist ansonsten unabhängig von sich ändernden Umgebungsbedingungen.It is advantageous if the determination of the complex frequency-dependent amplitudes p _tm (f _j ) has the step for each time interval t _{m of} the complex frequency-dependent amplitudes p ' _tm (f _j ) obtained from the transformation into the frequency range $${\underset{\mathrm{}}{\mathrm{p}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{e}}^{\mathrm{-}i\mathrm{\phi }\left(\mathrm{fj}\right)}{\underset{\mathrm{}}{\mathrm{p\text{'}}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

adapt. The sensor phase correction parameter φ (f _j ) takes into account that a different design of the reference sensor and the sound pressure sensor can result in different phase responses and consequently a phase shift between q _PQ (t) and p (t) or the corresponding frequency-dependent complex amplitudes resulting directly from the update equation - adversely affects the calculation of the optimal gain factors directly. The sensor phase correction parameter φ (f _j ) may be determined in the course of an initial system calibration by determining the actual phase position of the physical quantities measured by the sensors and is otherwise independent of changing environmental conditions.

The described method and its advantageous embodiments can preferably be carried out with the aid of an active noise reduction system, which is described below. To avoid repetition, details of the method relating to the operation of the system will not be discussed in detail.

The system has a reference sensor, preferably an acceleration sensor or a fast sensor for the sound-radiating surface of a sound source, for the time-dependent measurement of a physical quantity q, which characterizes the sound-generating movement of the emission surface of the primary sound source, at least in a predetermined frequency range provided for noise reduction. In this case, the reference sensor is adapted to provide in operation a corresponding time-dependent reference parameter q _PQ (t), which includes the phase information of the primary sound source, in the form of a corresponding reference signal. The reference sensor may for example be arranged at the primary sound source or be provided for such an arrangement.

Furthermore, the system has a secondary sound source, eg in the form of a loudspeaker, with a radiation surface suitable for sound radiation and a sound pressure sensor, which is preferably a microphone, for time-dependent measurement of the sound pressure p (t) - at least in a predetermined frequency range provided for noise reduction which is arranged in the immediate vicinity of the emission surface of the secondary sound source or adapted for such an arrangement. In this case, the sound pressure sensor is adapted to provide a corresponding sound pressure signal during operation. As has already been explained above, it is preferred that the distance between the sound pressure sensor and the radiating surface in operation is one tenth of the wavelength - or at several Wavelengths one tenth of the smallest wavelength - of the sound to be canceled. It has also been stated above that it is advantageous if the sound sources are arranged in spatial proximity to one another during operation, preferably at a mutual distance which is half the wavelength - or at half wavelengths half the smallest wavelength - of the sound to be canceled does not exceed

• q_PQ (t) und p(t) in den Frequenzbereich zu transformieren, so dass für jedes von diskreten Zeitintervallen t_m jeweils eine Anzahl von komplexen frequenzabhängigen Amplituden q _{PQ, tm}(f_j) bzw. p _tm (f_j) für eine Anzahl und bevorzugt eine Vielzahl von Frequenzen f_j erhalten werden,
• das zeitabhängige Ansteuersignal in der Weise zu erzeugen, dass die Abstrahlfläche der mit dem Ansteuersignal angesteuerten Sekundärschallquelle zumindest bei den Frequenzen f_j gleich- oder gegenphasig zur Abstrahlfläche der Primärschallquelle schwingt und ihre entsprechende schallerzeugende Bewegung in jedem von aufeinanderfolgenden diskreten Zeitabschnitten T_n, die jeweils eines oder mehrere der diskreten Zeitintervalle t_m umfassen, durch einen zeitlichen Verlauf q_SQ (t) der physikalischen Größe q gekennzeichnet ist, bei dem zwischen den einer Transformation von q_SQ (t) in den Frequenzbereich entsprechenden komplexen frequenzabhängigen Amplituden q _{SQ, tm}(f_j) und den komplexen Amplituden q _PQ, _tm (f_j) für die Frequenzen f_j in den diskreten Zeitabschnitten T_n jeweils die Beziehung $${\underline{\mathrm{q}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$
  
  
  mit den in jedem Zeitabschnitt T_n zeitlich konstanten reellen Verstärkungsfaktoren k_Tn(f_j) besteht, auf die oben bereits ausführlich eingegangen worden ist, und
• jeweils ausgehend von den Verstärkungsfaktoren k_Tn (f_j) in einem Zeitabschnitt T_n und den komplexen frequenzabhängigen Amplituden q _PQ,_tm(f_j) bzw. p _tm(f_j) für ein Zeitintervall t_m in dem Zeitabschnitt T_n die Verstärkungsfaktoren k_Tn+1 (f_j) für den nächsten Zeitabschnitt T_n+1 für zumindest eine der Frequenzen f_j durch die Aktualisierungsgleichung $${\mathrm{k}}_{\mathrm{Tn}\mathrm{+}\mathrm{1}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\mathrm{-}\mathrm{\mu \; Im}\left[{\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{p}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{*}\right]\mathrm{+}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$
  
  
  neu zu berechnen und einzustellen, wobei µ ein reeller Konvergenzfaktor ist und Im[x] den Imaginärteil einer komplexen Zahl x und x* die zu einer komplexen Zahl x komplex konjugierte Zahl bezeichnet.

Finally, the system comprises a control device or control device and preferably a digital control device or control device which has, for example, a suitably programmed processor with inputs for receiving the reference signal and the sound pressure signal and an output for outputting a time-dependent control signal for the secondary sound source on. The control device is adapted to operate in the manner already explained

• q _PQ (t) and p (t) to transform in the frequency domain, so that for each of discrete time intervals t _m each have a number of complex frequency-dependent amplitudes q _{PQ, tm} (f _j ) and p _tm (f _j ) for a Number and preferably a plurality of frequencies f _j are obtained
• to generate the time-dependent drive signal in such a way that the emission surface of the driven with the drive signal secondary sound source at least at the frequencies f _j equal or opposite phase to the emission surface of the primary sound source and their corresponding sound-generating movement in each of successive discrete periods T _n , each one or several of the discrete time intervals t _m , is characterized by a time course q _SQ (t) of the physical quantity q, in which between the one complex of a transformation of q _SQ (t) in the frequency domain Frequency-dependent amplitudes q _{SQ, tm} (f _j ) and the complex amplitudes q _PQ , _tm (f _j ) for the frequencies f _j in the discrete periods T _n respectively the relationship $${\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$
  
  
  with the real gain _factors k _Tn (f _j ) that are constant over time in each time segment T _n , which have already been discussed in detail above, and
• in each case based on the amplification _factors k _Tn (f _j ) in a time interval T _n and the complex frequency-dependent amplitudes q _PQ , _tm (f _j ) and p _tm (f _j ) for a time interval t _m in the time interval T _n, the gain factors k _{Tn + 1} (f _j ) for the next period T _{n + 1} for at least one of the frequencies f _j through the update equation $${\mathrm{k}}_{\mathrm{Tn}\mathrm{+}\mathrm{1}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\mathrm{-}\mathrm{\mu \; Im}\left[{\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{p}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{*}\right]\mathrm{+}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$
  
  
  where μ is a real convergence factor and where [ x ] is the imaginary part of a complex number x and x * is the number conjugated to a complex number x complex.

For this purpose, the control device preferably has in each case one or more devices set up for the said steps.

In this context, it should again be pointed out that it is of course possible to provide more than one secondary sound source in order to be able to take account of different regions of a structure or construction representing the primary sound source in the manner described.

The system is particularly suitable for stationary noise, and it is preferred that the primary sound source be a sound source that generates stationary noise.

In a preferred embodiment, the control device is set up such that the time intervals t _m , which are used to recalculate the gain factors k _{T + 1} (f _j ), are temporally spaced from one another.

mit vorbestimmten Schwellenwerten ε_j gilt, und für die übrigen Frequenzen f_j die Verstärkungsfaktoren k_Tn+1 (f_j) auf Null gesetzt werden.In a preferred embodiment, the control means is arranged to recalculate the gain _factors k _{Tn + 1} (f _j ) only for frequencies f _j for which $$\left|{\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\right|\ge {\mathrm{\epsilon }}_{\mathrm{j}}$$

with predetermined threshold values ε _j , and for the remaining frequencies f _j the amplification _factors k _{Tn + 1} (f _j ) are set to zero.

In a preferred embodiment, the controller is adapted to limit its magnitude after each recalculation of a gain k k _{Tn + 1} (f _j ) by checking whether that magnitude exceeds a predetermined maximum k _max (f _j ) for the magnitude, and k _{Tn + 1} (f _j ) when exceeded depending on whether k _{Tn + 1} (f _j ) is negative or positive, set to - k _max (f _j ) or + k _max (f _j ). One way of setting these limits has been explained above.

zu berechnen und auf Basis der komplexen frequenzabhängigen Amplituden q _SQ,tm(f_j) mit Hilfe einer Rücktransformation in den Zeitbereich den zeitabhängigen Ansteuerparameter q_SQ (t) zu erhalten, der die zu erzielende schallerzeugende Bewegung der Abstrahlfläche der Sekundärschallquelle kennzeichnet. In diesem Fall ist es bevorzugt, wenn die Steuereinrichtung ferner angepasst ist, um die Transformation von q_PQ (t) und p(t) in den Frequenzbereich zur Bestimmung der komplexen frequenzabhängigen Amplituden q _PQ,tm(f_j) und p _tm(f_j) mit Hilfe diskreter Fouriertransformation durchzuführen. Insbesondere ist es dann auch zusätzlich bevorzugt, wenn die Steuereinrichtung angepasst ist, um die Rücktransformation auf Basis der komplexen frequenzabhängigen Amplituden q _SQ,_tm(f_j) in den Zeitbereich zur Bestimmung des zeitabhängigen Ansteuerparameters q_SQ(t) mit Hilfe inverser diskreter Fouriertransformation durchzuführen.In a preferred embodiment, the controller is adapted to generate the time-dependent drive signal for the secondary _{sound source} for each of the time intervals t _m from the complex frequency-dependent amplitudes q _{PQ, tm} (f _j ) and the corresponding gain _factors k _Tn (f _j ) for the period T _n for the frequencies f _j complex frequency-dependent amplitudes q _{SQ, tm} (f _j ) according to $${\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{PQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

and to obtain on the basis of the complex frequency-dependent amplitudes q _{SQ, tm} (f _j ) by means of a back transformation into the time domain the time-dependent control parameter q _SQ (t), which determines the sound-generating movement to be achieved Radiating surface of the secondary sound source features. In this case, it is preferable that the control means is further adapted to perform the transformation of q _PQ (t) and p (t) into the frequency domain for determining the complex frequency-dependent amplitudes q _{PQ, tm} (f _j ) and p _tm (f _j ) using discrete Fourier transform. In particular, it is then additionally preferred if the control device is adapted to carry out the inverse transformation on the basis of the complex frequency-dependent amplitudes q _SQ , _tm (f _j ) into the time domain for determining the time-dependent control parameter q _SQ (t) by means of inverse discrete Fourier transformation ,

anzupassen und die angepassten komplexen frequenzabhängigen Amplituden q'_SQ,_tm (fj) für die Rücktransformation in den Zeitbereich zur Bestimmung des zeitabhängigen Ansteuerparameters q_SQ(t) zu verwenden, wobei G ^-1(f_j) vorbestimmte komplexe Zeitverzögerungskorrekturparameter zur Berücksichtigung von durch Signalverarbeitung und elektroakustische Umwandlungen verursachte Phasenverschiebungen sind.In this embodiment, it is also advantageous if the control device is adapted, after each calculation of the complex frequency-dependent amplitudes q _{SQ, tm} (f _j ) according to q _{SQ, tm} (f _j ) = k _Tn (f _j ) q _{PQ, tm} (f _j ) for a time interval t _m the complex frequency-dependent amplitudes q _{sQ, tm} (f _j ) according to $${\underset{\mathrm{}}{\mathrm{q\; \text{'}}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\underset{\mathrm{}}{{\mathrm{G}}^{\mathrm{-}\mathrm{1}}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underset{\mathrm{}}{\mathrm{q}}}_{\mathrm{SQ}\mathrm{.}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

and to use the adjusted complex frequency dependent amplitudes q ' _SQ , _tm (fj) for the inverse transformation in the time domain to determine the time dependent drive parameter q _SQ (t), where G ^-1 (f _j ) predetermined complex time delay correction parameters for taking into account by signal processing and electroacoustic transformations caused phase shifts.

mit einem Sensorphasenkorrekturparameter ϕ (f_j) anzupassen.In a preferred embodiment, the control device is adapted to use for the determination of the complex frequency-dependent amplitudes p _tm (f _j ) for each time interval t _m from the transformation in the frequency domain obtained complex frequency-dependent amplitudes p ' _tm (f _j ) according to $${\underset{\mathrm{}}{\mathrm{p}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{e}}^{\mathrm{-}i\mathrm{\phi }\left(\mathrm{fj}\right)}{\underset{\mathrm{}}{\mathrm{p\text{'}}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

with a sensor phase correction parameter φ (f _j ).

The control device can be, for example, a hardware-permanently implemented device with various elements for carrying out the various mentioned steps, a device adapted and set up for this purpose programmatically having a programmable processor and possibly a memory device, or a hybrid having a programmable processor but in which individual steps are performed by hardware-implemented elements.

Figur 1

zeigt schematisch den prinzipiellen Aufbau eines er- findungsgemäßen aktiven Lärmreduktionssystems.

Figur 2

zeigt ein Flussdiagramm einer bevorzugten Vorgehens- weise zur Neuberechnung eines Verstärkungsfaktors k_Tn+1 (f_j) für eine Frequenz f_j mit Hilfe der Aktualisie- rungsgleichung durch die Steuereinrichtung.

Figur 3

zeigt schematisch eine bevorzugte Anordnung zur Be- stimmung eines Zeitverzögerungskorrekturparameters G^-1 (fj)_.

Figur 4

zeigt schematisch eine bevorzugte Anordnung zur Be- stimmung eines Sensorphasenkorrekturparameters ϕ (f_j).

Figur 5

zeigt schematisch ein Blockschaltbild eines bevorzug- ten Gesamtsystems.

In the following the invention will be further explained by means of exemplary embodiments on the basis of the drawings.

FIG. 1

schematically shows the basic structure of an inventive active noise reduction system.

FIG. 2

FIG. 12 shows a flowchart of a preferred procedure for recalculating an amplification _factor k _{Tn + 1} (f _j ) for a frequency f _j with the aid of the updating equation by the control device.

FIG. 3

schematically shows a preferred arrangement for determining a time delay correction parameter G ^-1 (fj) _.

FIG. 4

schematically shows a preferred arrangement for the determination of a sensor phase correction parameter φ (f _j ).

FIG. 5

schematically shows a block diagram of a preferred overall system.

In FIG. 1 Reference numeral 1 denotes a primary sound source in the form of a vibrating surface, which forms the emitting surface of the sound source. On the surface of the primary sound source 1 is serving as a reference sensor acceleration sensor 2, with which the acceleration q of the radiating surface is measured as a time-dependent reference parameter q _PQ (t) and fed in the form of a corresponding reference signal via a reference signal line 3 to a reference signal input 4 of a digital control or regulating device 5. Further, a loudspeaker 6 is provided, in front of the radiating surface, ie its loudspeaker diaphragm, a sound pressure sensor in the form of a microphone 7 is arranged in the immediate vicinity of the radiator central axis, with which the time-dependent sound pressure p (t) measured on the speaker 6 and in the form of a corresponding Sound pressure signal via a sound pressure signal line 8 a sound pressure signal input 9 of the digital control device 5 is fed.

The control device 5 generates based on the reference signal and the sound pressure signal, a time-dependent control signal, which is output at a Ansteuersignalausgang 10 and is controlled by the loudspeaker 6 via a Ansteuersignalleitung 11. In this case, the control device 5 generates the drive signal in such a way that the total active power radiated by the source pair formed by the primary sound source 1 and the loudspeaker 6 is minimized in the manner described above. It is clear that the system, with the exception of the design of the control device 5 in its basic structural design is similar to a simple system with sound pressure-based control using a single microphone and is therefore advantageously designed and installed in an easy way.

In FIG. 2 _{FIG. 4} is a flow chart of a procedure to be advantageously implemented in the controller 5 for recalculating a gain k _{kn + 1} (f _j ) for a frequency f _j using the update equation. In step 20, various parameters are initialized once before the first recalculation step, the meaning of which has been indicated above or will be explained in more detail below.

In step 21, the two of the control means 5 by means of discrete Fourier transformation (see. Figures 3 and 5 ) of q _PQ (t) and p (t) for the frequency f _j specific complex amplitudes q _PQ , _tm , _Δtz (f _j ) and p _tm , _Δtz (f _j ) in the process _flow entered. q _PQ , _tm , _Δtz (f _j ) and p _{tm, Δtz} (f _j ) denote complex amplitudes, which consist of the signals q _PQ (t) and p (t) in a designated by the parameter z subinterval .DELTA.t _z of the time interval t _m designated by the parameter m, wherein each time interval t _{m is formed} by z _max immediately following subintervals Δt _z . _PQ in step 22 q _{PQ, tm, Δtz} (f _j) and p _{tm, Δtz} be (f _j) then q to the complex _amplitudes add _tm (f _j) and p _tm (f _j), which in step 20 were initially set to zero. It is then checked whether z is equal to the maximum value z _max (step 23) and, if this is not the case, z is increased by 1 (step 24) and jumped back to step 21. There, the discrete Fourier transform for the next, designated by the new value of z sub-interval .DELTA.t _z certain complex amplitudes q _{PQ, tm, .DELTA.tz} (f _j ) and p _{tm, .DELTA.tz} (f _j ) are entered into the process _flow and then in step 22 to the complex amplitudes q _{PQ, tm} (f _j ) and p _tm (f _j ) added. Since this loop is not left until the check in step 23 shows that the value z is equal to z _max , it can be seen that in step 22 the total of the corresponding sums for all z _max subintervals _{Δtz are} formed.

Thereafter, these sums q _PQ , _tm (f _j ) and p _tm (f _j ) are divided by z _max in step 26, so that q _{PQ, tm} (f _j ) and p _tm (f _j ) are then over the time interval t _m average values for the complex amplitudes. By this averaging is avoided the complex amplitudes q _{PQ, tm} (t _j ) and p _tm (f _j ) used for the recalculation of the amplification _factors are falsified by measurement errors or artefacts of the discrete Fourier transformation.

In step 27, it is checked whether the magnitude of the complex amplitude q _{PQ, tm} (f _j ) is at least equal to a threshold value ε _j , and only if so, in step 30 is the gain _parameter k _{Tn + 1} (f _j ) for the subsequent period T _{n + 1} compared to the value k _Tn (f _j ) for the period T _n recalculated. Otherwise, the gain _parameter k _{Tn + 1} (f _j ) is set to the value zero for the subsequent period T _{n + 1} (step 29). If a recomputation is to be performed, then in step 30, using the update equation discussed above, k _{Tn + 1} (f _j ) = -μ In [ q _{PQ, t m} (f _j ) p _tm (f _j ) *] + k _Tn ( f _j ).

Next, in step 31, it is checked whether the magnitude of gain k _{Tn + 1} (f _j ) for the subsequent period T _{n + 1} exceeds a predetermined positive maximum value k _max (f _j ) and, if necessary, kT _{n + 1} (f _j ) is limited in magnitude to the sign (step 32).

In step 33, the amplification _factor k _{Tn + 1} (f _j ) for the subsequent period T _{n + 1 is} finally output for use by the controller 5 in generating the drive signal for the speaker 6.

Step 34 ensures that no further update is performed for a certain period of time t _{pause to} ensure that the new value of the gain factor has had an effect on the sound emission by the loudspeaker 6 before the next update calculation. Finally, in step 35, the parameters n, z, q _{pQ, tm} (f _j ) and p _tm (f _j ) are reinitialized for the next pass.

FIG. 3 ¹ schematically shows a preferred arrangement for determining a time delay correction parameter G ^-1 (f _j ) for one or more of the frequencies f _j and preferably for each frequency f _j and at the same time also shows the essential elements of the control device 5 for generating the drive signal for the loudspeaker 6 the reference signal. These elements comprise an input element 40, a multiplication element 41 for applying the gain k k _Tn (f _j ) and an output element 42. The input element 40 has the reference signal input 4, and the output element 42 has the drive signal output 10. In addition, the input element 40 has an output 43 for outputting the complex amplitude q _PQ, tm (f _j ), which is output via the multiplication element 41 in which it generates the complex amplitude q _{SQ, tm} (f _j ) with the gain factor k _Tn (fj) is fed to an input 44 of the output element 42.

By this arrangement, the drive signal in principle (neglecting later discussed phase correction measures) generated by the reference signal first in the input element 40 successively a preamplifier 45, a high-pass filter 46, an analog-to-digital converter 47 and an element 48 for discrete Fourier transform (DFT ), whereby the complex amplitude q _{PQ, tm} (f _j ) is obtained at the output 43. The complex amplitude q _{SQ, tm} (f _j ) fed to the input 44 then successively passes through an element 49 for inverse discrete Fourier transformation (IDFT), a digital-to-analog converter 50, a low-pass filter 51 and an amplifier 52 in the output element 42, whereby Output 10, the drive signal is obtained. It should be pointed out once again that the drive signal does not have to have the curve q _SQ (t) itself and generally does not have, but by suitable configuration of the digital-to-analog converter 50, the low-pass filter 51, the amplifier 52 and, if necessary, further elements configured and generated so that the emitting surface of the speaker 6 according to q _SQ (t) oscillates.

In order now to determine the time delay correction parameter G ^-1 (f _j ) for the frequency f _j , which takes into account the signal propagation times within the elements shown, a loudspeaker 1 'is used as the primary sound source, which is identical to the secondary loudspeaker 6 and with the aid of a signal generator 53 sinusoidally excited at the frequency f _j . An acceleration pickup 54 corresponding to the sensor 2 is arranged on the emission surface of the secondary loudspeaker 6, the signal of which via a further input 55 of the input element 40 in the same manner as the reference signal successively its preamplifier 45, high-pass filter 46, analog-to-digital converter 47 and the element 48th to the discrete Fourier transform (DFT), whereby at a further output 56 a complex comparison _amplitude q _{SQ, tm, vg1} . (f _j ) is obtained.

It can be seen that when the multiplication element 41 is temporarily set to a multiplication by 1, the time-dependent accelerations q _PQ (t) and q _SQ (t) measured by the acceleration transducers 2 and 54 should be identical. This also means that the complex amplitude q _{PQ, tm} (f _j ) and the complex comparison amplitude q _{SQ, tm, cf.} (f _j ) should be identical at the outputs 43 and 56, respectively. However, this will not be the case due to the signal propagation times and the transmission behavior of the loudspeaker 6. Therefore, by means of a division element 57, the time delay correction parameter G ^-1 (f _j ) for one, several or each of the frequencies f _j according to q _PQ , _tm (f _j ) = G ^- 1 (f _j ) q _{SQ, tm, cf.} (f _j ) and stored in a memory location 58 for further use.

FIG. 4 shows a preferred arrangement for determining a sensor phase correction parameter φ (f _j ) for the frequency f _j , with the phase error due to the design-related different phase response of the accelerometer 2 and the microphone 7 are taken into account. The arrangement corresponds largely to in FIG. 3 shown arrangement. In addition, it contains a further multiplication element 59 connected downstream of the multiplication element 41 for multiplication by the time delay correction parameter G ^-1 (f _j ), ie for carrying out the time delay correction, which can also be referred to as predistortion for the reference signal. In this case, since the multiplication element 41 is temporarily set to a multiplication by -1, due to this correction, the emitting surfaces of the speakers 1 'and 6 oscillate in anti-phase and they both radiate no effective sound power. Is their mutual distance low, the acceleration at the primary speaker 1 'and the sound pressure at the location of the microphone 7 are approximately in phase.

Therefore, by means of a comparison made by a division element 60 and a phase angle element 61, the phase shift between the values q _PQ (t) and p (t) actually measured by the accelerometer 2 and the microphone 7 can be determined as a sensor phase correction parameter φ (f _j ) a memory location 62 are stored for further use.

After the in the Figures 3 and 4 shown system calibration measures, the system is ready for regular operation. It is completely finished again FIG. 5 shown, in contrast to FIG. 4 in addition to the multiplication elements 41 and 59, another multiplication element 63 for multiplying the complex pressure amplitude p ' _tm ( _fj ) by e ^-i φ ( _fj ) applied to a pressure amplitude output 64 of the input element 40 to obtain the complex pressure amplitude p tm ( _fj ) is provided. Furthermore, the updating element 65 is shown, by which the steps according to FIG FIG. 2 are performed and the multiplier 41 is optionally set via line 66 to the current value for the gain factor.

Claims (15)

1. A method for active noise reduction of the sound field generated by a vibrating emission surface of a primary sound source (1), wherein
   a physical quantity q is measured for the primary sound source (1) in a time-dependent manner with a reference sensor (2), which physical quantity q characterizes the sound generating movement of the emission surface of the primary sound source (1), in order to obtain a corresponding time-dependent reference parameter q_PQ (t) which includes the phase information of the primary sound source (1),
   the sound pressure p(t) is measured for a secondary sound source (6) in a time-dependent manner with a sound pressure sensor (7) disposed directly adjacent the emission surface of the secondary sound source (6),
   q_PQ (t) und p(t) are transformed into the frequency domain such that for each of discrete time intervals t_m a respective plurality of complex frequency dependent amplitudes q _{PQ, tm} (f_j) and p _tm (f_j), respectively, are obtained for a plurality of frequencies f_j,
   the secondary sound source (6) is driven with a time-dependent drive signal in such a manner that the emission surface of the secondary sound source (6) vibrates in phase or in antiphase with respect to the emission surface of the primary sound source (1), and that its sound generating movement in each of successes discrete time periods T_n, each comprising one or several of the discrete time intervals t_m, is characterized by a time-dependence q_SQ (t) of the physical quantity q for which between the complex frequency dependent amplitudes q _{SQ, tm} (f_j) corresponding to a transformation of q_SQ (t) into the frequency domain and the complex amplitudes q _{PQ, tm} (f_j) the relationship $${\underline{\mathrm{q}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

   

   exists for the frequencies f_j in the discrete time periods T_n with the real gain factors k_Tn (f_j) which are constant over time in each time period T_n,
   wherein starting respectively from the gain factors k_Tn (f_j) in a time period T_n and the complex frequency dependent amplitudes q _{PQ, tm} (f_j) and p _tm (f_j), respectively, for a time interval t_m in the time period T_n the gain factors k_Tn+1 (f_j) for the next time period T_n+1 are newly calculated and set for at least one of the frequencies f_j by means of the updating equation $${\mathrm{k}}_{\mathrm{Tn}\mathrm{+}\mathrm{1}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\mathrm{-}\mathrm{\mu \; Im}\left[{\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{p}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{*}\right]\mathrm{+}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right),$$

   

   wherein µ is a real convergence factor and Im[x] designates the imaginary part of a complex number x and x* designates the complex conjugate of a complex number x.
2. The method according to claim 1, wherein the physical quantity q is an acceleration or a particle velocity of the or at the sound emitting surface of a sound source.
3. The method according to any of the preceding claims, wherein the new calculation of the gain factors k_Tn+1 (f_j) is only carried out for frequencies f_j for which $$\left|{\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\right|\ge {\mathrm{\epsilon }}_{\mathrm{j}}$$

   

   applies with predetermined threshold values ε_j, and wherein the gain factors k_Tn+1 (f_j) are set to zero for the remaining frequencies f_j.
4. The method according to any of the preceding claims, wherein after each new calculation of a gain factor k_Tn+1 (f_j) its amount is limited by checking whether it exceeds a predetermined maximum value k_max (f_j) for the amount, and wherein in case the maximum value is exceeded k_Tn+1 (f_j) is set to -k_max(f_j) or +k_max(f_j), depending on whether k_Tn+1 (f_j) is negative or positive.
5. The method according to any of the preceding claims, wherein the generation of the time-dependent drive signal for the secondary sound source (6) comprises the steps of calculating for each of the time intervals t_m from the complex frequency dependent amplitudes q _{PQ, tm} (f_j) and the corresponding gain factors k_Tn (f_j) for the time period T_n for the frequencies f_j complex frequency dependent amplitudes q _SQ, _tm (f_j) according to $${\underline{\mathrm{q}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right),$$

   

   
   and
   obtaining on the basis of the complex frequency dependent amplitudes q _{SQ, tm} (f_j) by means of a back transformation into the time domain the time-dependent drive parameter q_SQ (t), which characterizes the sound generating movement of the emission surface of the secondary sound source (6) to be achieved.
6. The method according to any of the preceding claims, wherein the transformation of q_PQ (t) and p(t) into the frequency domain for the determination of the complex frequency dependent amplitudes qp_{Q, tm} (f_j) and p _tm (f_j) is carried out by means of discrete Fourier transform.
7. The method according to claim 5 and claim 6, wherein the back transformation on the basis of the complex frequency dependent amplitudes q _SQ,tm (f_j) into the time domain for the determination of the time-dependent drive parameter q_SQ (t) is carried out by means of inverse discrete Fourier transform.
8. The method according to any of claims 5 to 7, wherein after each calculation of the complex frequency dependent amplitudes q _{SQ, tm} (f_j) pursuant to q _SQ, _tm (f_j) = k_Tn (f_j) q _{PQ, tm}(f_j) for a time interval t_m the complex frequency dependent amplitudes q _{SQ, tm}(f_j) are adapted according to $${\underline{\mathrm{qʹ}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\underline{{\mathrm{G}}^{\mathrm{-}\mathrm{1}}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{q}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right),$$

   

   and the adapted complex frequency dependent amplitudes q'_{SQ, tm} (f_j) are utilized for the back transformation into the time domain for the determination of the time-dependent drive parameters q_SQ (t), wherein G ^-1(f_j) are predetermined complex time delay correction parameters for taking into account phase shifts caused by signal processing and electroacoustic conversions.
9. Active noise reduction system for carrying out the method according to any of claims 1 to 8, wherein the system comprises:

   a reference sensor (2) for the time-dependent measurement of a physical quantity q, which characterizes the sound generating movement of the emission surface of the primary sound source (1), wherein the reference sensor (2) is adapted for providing, during operation, a corresponding time-dependent reference parameter q_PQ (t), which includes the phase information of the primary sound source (1), in the form of a corresponding reference signal,

   a secondary sound source (6) having an emission surface suitable for sound emission,

   a sound pressure sensor (7) disposed directly adjacent the emission surface of the secondary sound source (6) for the time-dependent measurement of the sound pressure p(t),

   wherein the sound pressure sensor (7) is adapted for providing, during operation, a corresponding sound pressure signal,

   a control means (5) having inputs for receiving the reference signal and the sound pressure signal and an output (10) for outputting a time-dependent drive signal for the secondary sound source (6), wherein the control means is adapted to, during operation,

   - transform q_PQ (t) und p(t) into the frequency domain such that for each of discrete time intervals t_m a respective plurality of complex frequency dependent amplitudes q _PQ,tm (f_j) and p _tm (f_j), respectively, are obtained for a plurality of frequencies f_j,

   - generate the time-dependent drive signal in such a manner that the emission surface of the secondary sound source (6) driven with the drive signal vibrates in phase or in antiphase with respect to the emission surface of the primary sound source (1), and that its sound generating movement in each of successes discrete time periods T_n, each comprising one or several of the discrete time intervals t_m, is characterized by a time-dependence q_SQ (t) of the physical quantity q for which between the complex frequency dependent amplitudes q_{SQ, tm} (f_j) corresponding to a transformation of q_SQ (t) into the frequency domain and the complex amplitudes q_{PQ, tm} (f_j) the relationship $${\underline{\mathrm{q}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)$$

   

   exists for the frequencies f_j in the discrete time periods T_n with the real gain factors k_Tn (f_j) which are constant over time in each time period T_n, and

   - newly calculate and set, starting respectively from the gain factors k_Tn (f_j) in a time period T_n and the complex frequency dependent amplitudes q_{PQ, tm} (f_j) and p _tm (f_j), respectively, for a time interval t_m in the time period T_n, the gain factors k_Tn+1 (f_j) for the next time period T_n+1 for at least one of the frequencies f_j by means of the updating equation $${\mathrm{k}}_{\mathrm{Tn}\mathrm{+}\mathrm{1}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\mathrm{-}\mathrm{\mu \; Im}\left[{\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{p}}}_{\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{*}\right]\mathrm{+}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{,}$$

   

   wherein µ is a real convergence factor and Im[x] designates the imaginary part of a complex number x and x* designates the complex conjugate of a complex number x.
10. The system according to claim 9, wherein the new calculation of the gain factors k_Tn+1 (f_j) is only carried out for frequencies f_j for which $$\left|{\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\right|\ge {\mathrm{\epsilon }}_{\mathrm{j}}$$

    

    applies with predetermined threshold values ε_j, and wherein the gain factors k_Tn+1 (f_j) are set to zero for the remaining frequencies f_j.
11. The system according to claim 9 or claim 10, wherein the control means (5) is adapted to limit after each new calculation of a gain factor k_Tn+1 (f_j) the amount thereof by checking whether it exceeds a predetermined maximum value k_max (f_j) for the amount and setting k_Tn+1 (f_j) to -k_max (f_j) or +k_max (f_j), depending on whether k_Tn+1 (f_j) is negative or positive, in case the maximum value is exceeded.
12. The system according to any of claims 9 to 11, wherein the control means (5) is adapted to
    calculate for each of the time intervals t_m from the complex frequency dependent amplitudes q _{PQ, tm} (f_j) and the corresponding gain factors k_Tn (f_j) for the time period T_n for the frequencies f_j complex frequency dependent amplitudes q _{SQ, tm}(f_j) according to $${\underline{\mathrm{q}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}{\mathrm{k}}_{\mathrm{Tn}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{q}}}_{\mathrm{PQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right),$$

    

    
    and
    obtain on the basis of the complex frequency dependent amplitudes q _SQ,tm (f_j) by means of a back transformation into the time domain the time-dependent drive parameter q_SQ (t), which characterizes the sound generating movement of the emission surface of the secondary sound source (6) to be achieved, for generating the time-dependent drive signal for the secondary sound source (6).
13. The system according to any of claims 9 to 12, wherein the control means (5) is adapted carry out the transformation of q_PQ (t) and p(t) into the frequency domain for the determination of the complex frequency dependent amplitudes q _{PQ, tm} (f_j) and p _tm (f_j) by means of discrete Fourier transform.
14. The system according to claim 12 and claim 13, wherein the control means (5) is adapted to carry out the back transformation on the basis of the complex frequency dependent amplitudes q _{SQ, tm} (f_j) into the time domain for the determination of the time-dependent drive parameter q_SQ (t) by means of inverse discrete Fourier transform.
15. The system according to any of claims 12 to 14, wherein the control means (5) is adapted to adapt after each calculation of the complex frequency dependent amplitudes q _SQ,tm (f_j) pursuant to q _{SQ, tm} (f_j) = k_Tn (f_j) q _{PQ, tm}(f_j) for a time interval t_m the complex frequency dependent amplitudes q _SQ,tm(f_j) according to $${\underline{\mathrm{qʹ}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right)\mathrm{=}\underline{{\mathrm{G}}^{\mathrm{-}\mathrm{1}}}\left({\mathrm{f}}_{\mathrm{j}}\right){\underline{\mathrm{q}}}_{\mathrm{SQ}\mathrm{,}\mathrm{tm}}\left({\mathrm{f}}_{\mathrm{j}}\right),$$

    

    and to utilize the adapted complex frequency dependent amplitudes q'_SQ,tm (f_j) for the back transformation into the time domain for the determination of the time-dependent drive parameters q_SQ (t), wherein G ^-1 (f_j) are predetermined complex time delay correction parameters for taking into account phase shifts caused by signal processing and electroacoustic conversions.

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