EP4002875A1 - Method for adapting anc headphones - Google Patents

Abstract

The invention relates to a method for adapting or calibrating ANC headphones, which has at least one signal processor on which at least one ANC filter, in particular an IIR filter, and its parameters are stored, comprising the following steps: a) putting on of the headphones to a measuring device, for example a coupler, an artificial head with measuring microphones, or a real head with probe microphones, containing EDP with data transmission,b) measuring relevant transmission paths x(n), m(n) and p(n) for feedforward and h(n) for feedback, stimulating the ANC circuit of a headphone, for example by chirping or noise,c) definition of at least one target function, ―p(n) for feedforward or t(n) for feedback, for the calculation of the complementary function, f(n) for feedforward or i(n) for feedback, at least one branch of the ANC circuit (feedforward or feedback) of the headphones,d) calculating at least one impulse response of the complementary function on/en, f(n) for feedforward and/or i(n) for feedback, from the measurements of the relevant transmission links,e) approximation of the parameters of the ANC filter(s) using the Prony method, which is used to achieve the complementary function/ en are necessary, f) entering or activating the calculated parameters in the signal processor.

Description

The invention relates to a method for adjusting or calibrating ANC headphones, according to the preamble of claim 1 and WO 2010/049241 A1 .

This publication, according to the U.S. 9,779,714 proposes that, in the course of the manufacture of ANC headphones, these should be placed on an artificial head with microphones with a filter (equivalent to an equalizer) derived from experiments, that the signal emitted by the loudspeakers be detected in the presence of (known) external noise, and adjust the gain level of the ANC microphone to achieve the best possible cancellation of ambient noise. This setting of the microphone is then used for the lifetime of the headphones.

Despite the simplicity of this method, it has a number of shortcomings, in particular no deviations in the characteristics of the test phone can be taken into account. The final setting of the level can also change the changes that occur over the course of the life of a headphone, be it the quality of the shielding by the headphone shells or their cushions, be it electronic drift in the amplifiers or loudspeakers or in the microphone due to aging processes necessary moving membranes etc. are taken into account.

the CN 111 800 694 A shows a method for adjusting ANC headphones, in which the headphones to be adjusted are placed on a measuring device and the ANC circuit is stimulated. A filter function calculation is then carried out in the frequency range. Also the US2011/222696 shows a similar method for adjusting ANC headphones in the frequency domain. The disadvantage of this method is that this type of signal processing is not suitable for all applications and then leads to an increased computing effort.

the U.S. 2019/080682 A1 represents a more distant state of the art. It is not dedicated to the filter calculation or the determination of the parameters for Configuration of the ANC headphones after the measurement, but the focus is on the measurement itself.

In general, it can be said of the prior art that ANC headphones are tuned by measuring the impulse responses of prototypes and determining "mean" (optimized) filter characteristics, which are programmed accordingly on the signal processors (usually ANC ICs). These filter characteristics are used in all headphones in this series. In the course of manufacturing the actual products, the real impulse response is measured piece by piece and the gain of the ANC microphone is adjusted as best as possible to the difference between the measurement results and the desired end result.

In the application and the claims, ANC headphones are also understood to mean earphones, so-called in-ear, earbud, on-ear and circumaural-ear headphones and also hearing aids of all kinds.

It is the aim and object of the invention to specify a method which is able to take into account such deviations as well as differences from the desired filter characteristics which are ideal for the respective individual product.

According to the invention, these objects are achieved by the characteristics indicated in the characterizing part of claim 1, in other words; a desired transmission path is determined in the course of the development of the headphones, the transmission paths of the produced ANC headphones are measured, complementary filter functions are calculated from the measurements and using the Prony method (recursive) filters for the signal processor are determined in such a way that the headphones achieve the desired (Ideal) transmission path, the filter coefficients and/or amplification factors determined in this way are stored or activated on the signal processor.

1. a) Aufsetzen des Kopfhörers auf ein geeignetes Messmittel, (beispielsweise einen Kuppler, oder einen Kunstkopf mit Messmikrofonen, oder einen realen Kopf mit Sondenmikrofonen) enthaltend eine analoge oder digitale EDV mit einer Datenübertragung,
2. b) Vermessung der relevanten Übertragungsstrecken unter Anregen des ANC-Systems (ANC-Kreises) des Kopfhörers, beispielsweise durch Chirpen oder Rauschen,
3. c) Definition zumindest einer Zielfunktion zur Kalkulation der Komplementärfunktion zumindest eines Zweigs der ANC Anwendung (des ANC-Kreises),
4. d) Berechnen zumindest einer Impulsantwort der Komplementärfunktion/en aus den Messungen der relevanten Übertragungsstrecken,
5. e) Approximation der Parameter des/der ANC-Filter/s mittels Prony Methode, die zum Erreichen der Komplementärfunktion/en notwendig sind,
6. f) Eingeben bzw. Aktivieren der berechneten Parameter im Signalprozessor

Executed in more detail, the method according to the invention for calibrating or adjusting an ANC headphone, which has at least one signal processor has, on which at least one ANC filter, in particular an IIR filter, and its parameters are stored, the following steps:

1. a) Putting the headphones on a suitable measuring device (e.g. a coupler, or an artificial head with measuring microphones, or a real head with probe microphones) containing an analog or digital computer with data transmission,
2. b) Measurement of the relevant transmission paths by stimulating the ANC system (ANC circuit) of the headphones, for example through chirping or noise,
3. c) Definition of at least one target function for calculating the complementary function of at least one branch of the ANC application (the ANC circle),
4. d) calculating at least one impulse response of the complementary function(s) from the measurements of the relevant transmission links,
5. e) Approximation of the parameters of the ANC filter(s) using the Prony method, which are necessary to achieve the complementary function(s),
6. f) Entering or activating the calculated parameters in the signal processor

Generally speaking, the method according to the invention includes that in the course of the measurement of the finished product, the filter characteristics are individually changed and adapted to the respective individual product, which is made possible by the structure of the signal processor because they have Bluetooth or other wireless connection options, or via one galvanic interface such as USB or similar are accessible. Since the measurement results are available in digital form and are usually processed digitally (without this being absolutely necessary), it is easy to adapt the digital filter coefficients of the signal processor as desired using these data transmission options.

Such an adjustment can, of course, also be carried out on the used product years later, if an appropriate measuring station is available in order to compensate for any changes in the acoustic behavior due to the aging processes mentioned at the beginning.

It should only be briefly pointed out that the desired filter properties are not only tailored to the most complete possible reduction in external noise, but also always take into account the stability of the ANC circuit(s) in order to avoid all types of clipping and distortion, Da Feedback here is more sensitive than feedforward. Feedback systems work as a closed control loop and must meet stability criteria (amplitude <0dB at phase 360 degrees (range: +/-180) in order to avoid "howling" or overloading in the event of instability.

In one embodiment of the invention, the transfer functions determined in this way are broken down into second-order polynomials, which makes it possible to use the biquadratic cascades that are frequently used in signal processors.

The EDP with data transmission is an integral part of the measuring equipment. Accordingly, the measuring arrangement consists not only of the coupler, but also of an EDP that can further process the signal.

• die Fig. 1 den Graphen einer Approximation einer erfindungsgemäß errechneten Impulsantwort (Komplementärfunktion), aus gemessenen Übertragungsstrecken und Zielfunktion, mittels Prony Methode,
• die Fig. 2 ein Beispiel eines IIR Filters von 48kHz umgesetzt für 384kHz. Die linke Spalte zeigt das Original, die Rechte die Version bei höherer Abtastrate. Oben sind die Pole und Nullstellen in der komplexen Ebene dargestellt, unten Magnituden-(strichliert) und Phasengang (durchgezogen),
• die Fig. 3 das Ergebnis der Abtastratenanpassung der Impulsantwort am Beispiel der mit biquadratischen Filtern (IIR) approximierten Komplementärfilterfunktion und
• die Fig. 4 die Einzelfunktionen eines Feedback-Systems mit h(n) (links oben), t(n) (links unten) und durch Dekonvolution erhaltene und approximierte Komplementärfilter - Impulsantwort (rechts oben), rechts unten: das Ergebnis der Faltung von Sekundärstrecke und approximiertem Komplementärfilter - Impulsantwort zur Kontrolle der erfolgreichen Synthetisierung der Zielfunktion.

The invention is explained in more detail below with reference to the drawing. while showing

• the 1 the graph of an approximation of an impulse response (complementary function) calculated according to the invention, from measured transmission paths and target function, using the Prony method,
• the 2 an example of an IIR filter of 48kHz converted to 384kHz. The left column shows the original, the right the higher sampling rate version. The poles and zeros in the complex plane are shown above, magnitude (dashed) and phase response (solid) below
• the 3 the result of the sampling rate adaptation of the impulse response using the example of the complementary filter function approximated with biquadratic filters (IIR) and
• the 4 the individual functions of a feedback system with h ( n ) (top left), t ( n ) (bottom left) and complementary filters obtained and approximated by deconvolution - impulse response (top right), bottom right: the result of the convolution of the secondary path and the approximated complementary filter - Impulse response to control the successful synthesis of the target function.

An example of the method according to the invention or the optimization method (method) used is described below:
The method according to the invention uses IIR filters since these require significantly less computing power and memory than FIR filters with the same result. In addition, some ICs only allow IIR filters, so this method can be used universally. This can be a feedforward, feedback or hybrid system. A measuring system determines the impulse responses of the transmission paths (feedforward and feedback). This can be done using all the usual methods, such as excitation using chirps or noise, but is not limited to these. The known characteristics of the microphones and the driver, which are already available, must be added (by means of convolution) to the determined impulse responses of the passive sections. Thus the impulse response of the feedforward path is given by: $$x\left(n\right)\ast m\left(n\right)\ast f\left(n\right)=-p\left(n\right)$$

Where x(n) is the transmission path from the loudspeaker of the ANC headphones to the extinction point of the feedforward ANC system (coupler microphone, artificial head microphone, real head with probe microphone, or similar) and m(n) is the transmission path between the external loudspeaker (noise source) and the feedforward microphone correspond to. In a feedforward system, the objective function p ( n ) corresponds to the passive transmission path to the extinction point. From the present three For distances x ( n ), m ( n ) and p ( n ), the desired complementary function f ( n ) can be calculated analogously to the feedback system by deconvolution and then approximated using the method according to the invention. As is customary in the prior art, the cancellation point describes the point at which the counter wave generated by the ANC system cancels out the sound wave penetrating the headphones from the outside.

The feedback path is determined by achieving a target impulse response t(n) through deconvolution. $$H\left(n\right)\ast i\left(n\right)=t\left(n\right)$$

The measured real impulse response h(n), also known as the secondary path (corresponding to the transmission path between ANC headphones, loudspeaker and feedback microphone), convolved with the calculated impulse response i ( n ) results in the target function t ( n ). Thus, the impulse responses for feedforward and feedback are given. The individual functions and the resulting already approximated complementary filter impulse response are in Fig.4 shown.

A given impulse response can be viewed as an FIR filter of the length of the impulse response, with the values of each sample acting as the filter coefficients. The transfer function thus has the form $$H\left(\mathrm{e.g}\right)={\displaystyle \sum _{i=0}^M{b}_i{\mathrm{e.g}}^{-i}}$$

An IIR filter, on the other hand, has the form $$H\left(\mathrm{e.g}\right)=\frac{{\displaystyle {\sum }_{i=0}^M{b}_i{\mathrm{e.g}}^{-i}}}{{\displaystyle {\sum }_{i=0}^N{a}_i{\mathrm{e.g}}^{-i}}}$$

A given integrated circuit does not necessarily have the ability to use an FIR filter for ANC due to hardware limitations (such filters require too many taps). With an IIR filter, however, it is already possible, since the feedback structure requires fewer taps. It is therefore advantageous to approximate the given impulse response using an IIR polynomial.

There are several ways to do this: In one embodiment, the Prony method, in 1 shown, used, which approximates the given impulse response by exponentially damped cosine oscillations. The advantage of the Prony method is that, in contrast to filter approximation in the frequency domain, where the result is FIR filters which then require further methods to generate IIR polynomials, the result is an IIR polynomial. The Prony method approximates impulse responses; that means it has to be worked in the time domain. Impulse responses are already given as transmission routes via the measurement, which also makes it obvious not to leave the time range. The approximation of the ANC filter in point e) of the procedure corresponds to the conversion of the ideal FIR impulse response of the filter into an IIR filter function using the Prony method. $$\widehat{f}\left(t\right)={\displaystyle \sum _{i=1}^M{A}_i{e}^{{\sigma }_{i}t}}\cos \left({\omega }_{i}t+{\phi }_i\right)$$

Due to the approximation, the determined transfer function has more coefficients than required: ANC filters are usually defined up to 2kHz, since good passive damping can be expected above this. Optionally, the order of the transfer function can be reduced.

Impulse responses are typically recorded at a lower sample rate than used in the ANC system. 44.1 or 48kHz are common, while an ANC system is more likely to be clocked at 192 or 384kHz. The determined IIR filter must therefore be scaled from e.g. 48kHz to 384kHz, whereby the frequency response in absolute terms (in Hz) should remain the same (in a relevant range).

The scaling of an IIR filter is not obvious because the transfer function has to change. The method proposed here uses poles and zeros of the

transfer function H(z). The unit circle on the z-plane maps the frequency response (which can also be determined via the Fourier transformation), because there $$\mathrm{e.g}=e^{\mathit{j\omega }}$$

The DC component (0Hz) can be found at the (Cartesian) coordinate 1 + 0j, while half the sampling frequency can be found at -1 + 0j (Nyquist frequency). For 48kHz, the Nyquist frequency is 24kHz. In radians it is π, which is half the unit circle. For the higher sampling rate (384kHz), π equals 192kHz. That means there is more bandwidth (in Hz) in the same range in radians.

If a straight line is drawn from the point of origin (0 + 0j) through a pole (or a zero) to the unit circle, its angle (relative to the abscissa) for the low sampling rate is initially evident. The point on the unit circle corresponds to a frequency f in Hz. $$f=\frac{\varphi }{\pi }\frac{f_s}{2}\mathit{in}\mathit{Hz}$$

The point on the unit circle in Hz can be converted into corresponding radians for the increased sampling rate. $${\varphi }_{\mathrm{and}}=\frac{f}{0.5f_{\mathit{su}}}\pi$$

Thus, the angle through which the poles and zeros must be rotated to scale the transfer function for the higher sampling rate is known. In addition, the damping factors must be adjusted. The lines of constant damping of the s-plane become spiral-shaped root locus curves on the z-plane by appropriate mapping (bilinear transformation, impulse invariance, or similar). From the original point of a pole or a zero, the position root function to the DC point must be found. The new position of a pole/zero is at the intersection of the square root function and the new angle for the higher sampling rate. The result of this adjustment is in Fig.2 (right above and below).

Special regulations of the procedure are necessary for poles/zeros which are at the Nyquist frequency for the low sampling rate: Since this moves for the higher sampling rate (from π to < π), these poles/zeros must also move and be mirrored along the abscissa by a real value get filters. This process can result in more zeros than poles, resulting in an ill-defined transfer function. Add poles close to the origin point so that their influence is small but the transfer function becomes well defined. It is known that a polynomial with more coefficients in the numerator than in the denominator is not well defined since it would be anti-causal.

After scaling the transfer function, according to the 3 , this is to be broken down into second-order functions (biquadratic cascade), since these can usually be implemented in Ics and are also more stable. Optionally, the scaling can be done after the decomposition into biquadratic filters. The decomposition can be done using partial fractions.

The coefficients thus obtained for biquadratic, thus recursive, filters can be converted into a difference equation and used in a suitable integrated circuit. Since most ANC Ics in your programming interface/development environment have the option of entering filter coefficients usually IIR sometimes also FIR directly, this can be carried out without any problems.

In this case, for example, the feedback filter is programmed in the signal processor of a feedback ANC system with the coefficients of the calculated complementary function approximated according to the above description. This programming is usually accomplished via the development environment of the respective signal processors (ANC Ics) or by importing firmware provided with the coefficients using the methods already explained.

The invention can be modified and changed in various ways, so the measuring device can have or consist of any other arrangement of microphones in addition to the possibilities mentioned of an artificial head, etc., as long as only the required data are recorded that are familiar to the person skilled in the art with knowledge of the invention.

Whether any filter is configured in the signal processor at the beginning of the method or a filter that has already been roughly adapted based on experience is ultimately irrelevant, since running through the method again from step b), possibly with further repetitions, will quickly achieve the optimum achievable state.

Since ANC headphones change during their life cycle and their relevant transmission paths can change as a result, recalibration can make sense. The omission of a) from this repetition of the procedure is due to the fact that the repetition can also take place immediately after the first fitting, and the headphones are therefore still on the measuring head.

This is noticeable in particular in the final fitting of a whole series of headphones, in which an initial configuration can be assumed, which is achieved, for example, by the first headphones configured according to the invention.

Adjustments to headphones that have been in use for a long time are also possible without any problems.

Embodiments of the invention provide, for example, that in step b) the transmission links are measured digitally at the sampling rate of the measuring system, that the ANC headphones have a clock rate given by the digital signal processor, that the clock rate is higher than the sampling rate and that the /the approximated ANC filter(s) are scaled in the ratio of the sampling rate to the clock rate, whereby the frequency response of the approximated complementary filter(s) remains the same in absolute terms, in Hertz. In other words, the scaling of already approximated ANC filters according to the ratio between the clock rate of the working signal processor in the ANC listener and the sampling rate of the measuring system means that the coefficients of the IIR filter function are changed numerically in order to be identical with the higher clock rate of the signal processor generate frequency response.

A further embodiment provides that in step e) the order of the approximated complementary filter(s) is/are higher than the signal processor(s) can process and that the order of the approximated complementary filter(s) is reduced appropriately for the performance of the signal processor(s).

The need to reduce the order results from the fact that the signal processes in the headphones must be processed in real time during operation, since ANC only makes sense for real-time applications. Digital signal processors (DSP) have hardware-related properties known to those skilled in the art that represent restrictions that must be taken into account when they are used (e.g. arithmetic operations/cycle, clock rates inherent in the processor, energy requirements, etc.). This results in countless reasons or limitations why signal processing processes (e.g. filters) could not be carried out in real time. The list is therefore not to be regarded as conclusive. The utilization of the signal processors can of course depend not only on the pure generation of an ANC signal but also on other factors, such as the processing of the audio signal to be emitted (e.g. music) or Bluetooth streaming. However, this is not a compelling circumstance. For example, it would be possible to install several signal processors in one headphone, with each one taking on its own tasks. The disadvantage of this is obvious and can be found in the higher costs and greater consumption of installation space and/or energy. Since headphones traditionally have to make do with limited space in their housing, it is advantageous, but not absolutely necessary, to assign several tasks to a signal processor. Due to this possibility of adapting the system resources on the basis of the allocation of several tasks to a signal processor, it can therefore become necessary to adapt the order of the approximated complementary filter(s) to the signal processor(s).

The IIR polynomial or the 2nd order IIR cascade is already a filter implementation. The order is reduced if the signal processor does not have enough power to execute a cascade of 16 second-order filters, for example. Here, for example, 8 filters of the 2nd order are to be reduced. This reduction must be done in such a way that ANC performance is not significantly affected. This is neither trivial nor obvious given that there are already the slightest deviations from the ideal leads to strong negative effects on the sound suppression. This in turn is familiar to anyone skilled in the field and is assumed.

Since digital signal processors for ANC applications have to work in real time in order to make sound suppression possible at all, they work at much higher clock rates (e.g. max. 768kHz, typically 384kHz) than typical acoustic measuring systems (typically 48kHz). Routine conversion of filters for a higher sampling rate is possible with known design methods and is known to a person skilled in the art. Here, however, there is a filter cascade that does not follow a known design method (as explained in the text by approximating the impulse response using the Prony method), so the scaling of the filters is no longer obviously feasible. The pole/zero positions of the IIR filter polynomials, which are calculated based on the transmission paths sampled at 48 kHz, for example, must not shift in their position with respect to the natural frequency of the filter during upsampling, otherwise the filter characteristics no longer match the required complementary function. However, the poles and zeros on the z-plane must be modified to match the higher sampling rate of the ANC system; so that the characteristic of the filter cascade with regard to its natural frequency is retained. Different problems arise: Treatment of special cases such as poles/zeros on the real axis (0 or Pi) which also have to be shifted at 0 rad/s in order to retain the characteristic. Furthermore at Pi rad/s: Here the poles/zeros are moved away from the axis, have to be mirrored in order to maintain a real-valued filter. All of this affects the magnitude and phase response of the filter cascade and must be compensated accordingly. For this reason, this clock rate adjustment is decisive for the quality of the ANC.

The determination described as to whether a reduction in complexity is required is made by the operating specialist or a corresponding algorithm after the measurement according to the method. This decision does not have to be made in real time since the storage of the polynomial is not time sensitive since the adjustment is done before the headphone is sold. In this way it can be ensured that the ANC system can later act in real time.

1. a) Aufsetzen des Kopfhörers auf ein Messmittel, beispielsweise einen Kuppler, einen Kunstkopf mit Messmikrofonen, oder einen realen Kopf mit Sondenmikrofonen, enthaltend eine EDV mit einer Datenübertragung,
2. b) Vermessen relevanter Übertragungsstrecken x(n), m(n) sowie p(n) für Feedforward und h(n) für Feedback, unter Anregen des ANC-Kreises eines Kopfhörers, beispielsweise durch Chirpen oder Rauschen,
   dadurch gekennzeichnet, dass es weiters die Schritte
3. c) Definition zumindest einer Zielfunktion, -p(n) für Feedforward oder t(n) für Feedback, zur Kalkulation der Komplementärfunktion, f(n) für Feedforward oder i(n) für Feedback, zumindest eines Zweigs des ANC-Kreises (Feedforward oder Feedback) des Kopfhörers,
4. d) Berechnen zumindest einer Impulsantwort der Komplementärfunktion/en, f(n) für Feedforward und/oder i(n) für Feedback, aus den Messungen der relevanten Übertragungsstrecken,
5. e) Approximation der Parameter des/der ANC-Filter/s mittels Prony Methode, die zum Erreichen der Komplementärfunktion/en notwendig sind,
6. f) Eingeben bzw. Aktivieren der berechneten Parameter im Signalprozessor umfasst.

In order to clarify the invention once again with a view to the figures, it is hereby stated that the invention relates to a method for calibrating or adjusting ANC headphones which have at least one signal processor on which at least one ANC filter, in particular an IIR filter , and whose parameters are stored, comprising the following steps:

1. a) Putting the headphones on a measuring device, for example a coupler, an artificial head with measuring microphones, or a real head with probe microphones, containing an EDP with data transmission,
2. b) Measurement of relevant transmission paths x (n), m ( n ) and p ( n ) for feedforward and h ( n ) for feedback, while stimulating the ANC circuit of a headphone, for example by chirping or noise,
   characterized in that it further comprises steps
3. c) Definition of at least one objective function, - p ( n ) for feedforward or t ( n ) for feedback, for the calculation of the complementary function, f ( n ) for feedforward or i ( n ) for feedback, of at least one branch of the ANC circle (feedforward or feedback) of the headphones,
4. d) calculating at least one impulse response of the complementary function/s, f ( n ) for feedforward and/or i ( n ) for feedback, from the measurements of the relevant transmission links,
5. e) Approximation of the parameters of the ANC filter(s) using the Prony method, which are necessary to achieve the complementary function(s),
6. f) entering or activating the calculated parameters in the signal processor.

Claims (5)

Method for calibrating or adjusting ANC headphones which has at least one signal processor on which at least one ANC filter, in particular an IIR filter, and its parameters are stored, comprising the following steps: a) Putting the headphones on a measuring device, for example a coupler, an artificial head with measuring microphones, or a real head with probe microphones, containing an EDP with data transmission, b) Measurement of relevant transmission paths x (n), m ( n ) and p ( n ) for feedforward and h ( n ) for feedback, while stimulating the ANC circuit of a headphone, for example by chirping or noise,
characterized in that it further comprises steps c) Definition of at least one objective function, - p ( n ) for feedforward or t ( n ) for feedback, for the calculation of the complementary function, f ( n ) for feedforward or i ( n ) for feedback, of at least one branch of the ANC circle (feedforward or feedback) of the headphones, d) calculating at least one impulse response of the complementary function/s, f ( n ) for feedforward and/or i ( n ) for feedback, from the measurements of the relevant transmission links, e) Approximation of the parameters of the ANC filter(s) using the Prony method, which are necessary to achieve the complementary function(s), f) entering or activating the calculated parameters in the signal processor. Method according to Claim 1, characterized in that in step b) the transmission links are measured digitally with a sampling rate of the measuring system, that the ANC headphones have a digital signal processor which has a given clock rate, that the clock rate is higher than the sampling rate and that the /the already approximated ANC filter(s) are scaled in the ratio of the sampling rate to the clock rate, whereby the frequency response of (an) approximated complementary filter(s) remains the same in absolute terms, in Hertz. Method according to claim 1 or 2, characterized in that in step e) the order of a polynomial of the / the approximated complementary filter / s is / are higher than the / the signal processor (s) can process in real time (can) and that the order of the /of the approximated complementary filter(s) is reduced appropriately for the performance of the signal processor(s). Method according to one of the preceding claims, characterized in that it is repeated from step b) for control and/or more precise calibration. Method according to any one of the preceding claims, characterized in that the headphone is one of a production series in which the initial parameters of the ANC filter are set or activated based on previous tests, which are then repeated by repeating the method over the lifetime of the headphone be adjusted.

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