Patent Application
20170178617
This application claims priority to EP application Serial No. 15200631.8 filed Dec. 17, 2015, the disclosure of which is hereby incorporated in its entirety by reference herein.
The present invention relates to the art of reduction of noise in a listener environment. In particular, the present invention relates to the reduction of noise by adaptive filtering, for example, the reduction of noise in the passenger compartment of a vehicle.
Two-way speech communication of two parties mutually transmitting and receiving audio signals, in particular, speech signals, often suffers from deterioration of the quality of the audio signals by background noise. Background noise in noisy environments can severely affect the quality and intelligibility of voice conversation and can, in the worst case, lead to a complete breakdown of the communication.
A prominent example is hands-free voice communication in vehicles. Hands-free telephones provide comfortable and safe communication systems of particular use in motor vehicles. In the case of hands-free telephones, it is mandatory to suppress noise in order to guarantee the communication. The amplitudes and frequencies of the noise signals are temporally variable due to, for example, the speed of the vehicle and road noises. Moreover, noise heavily affects enjoying consumption of multimedia by a passenger in a vehicle, for example, an automobile, wherein a multimedia content is presented to a front/rear passenger by some front/rear seat entertainment system providing high-fidelity audio presentation using a plurality of loudspeakers arranged within the vehicle passenger compartment.
Herein, noise (or “disturbing sound”), in contrast to a useful sound signal, is considered a sound that is not intended to be perceived by a receiver (for example, a listener positioned in a vehicle compartment). With respect to motor vehicles noise can include sound signals generated by mechanical vibrations of an engine, fans or vehicle components mechanically coupled to the engine or fans and the wind as well as road noise as sound generated by the tires.
Noise within a listening environment can be suppressed using a variety of techniques. For example, noise may be reduced or suppressed by damping the noise signal at the noise source. The noise may also be suppressed by inhibiting or damping transmission and/or radiation of the noise. In many applications, however, these noise suppression techniques do not reduce noise levels in the listening environment below an acceptable limit. This is especially true for noise signals in the bass frequency range. Therefore, it has been suggested to suppress noise by means of destructive interference, i.e., by superposing the noise signal with a compensation signal. Typically, such noise suppression systems are referred to as “active noise cancelling” or “active noise control” (ANC) systems. The compensation signal has amplitude and frequency components that are equal to those of the noise signal; however, it is phase shifted by 180°. As a result, the compensation sound signal destructively interferes with the noise signal, thereby eliminating or damping the noise signal at least at certain positions within the listening environment.
Typically, active noise control systems use digital signal processing and digital filtering techniques. For example, a noise sensor such as, for example, a microphone or a non-acoustic sensor may be used to obtain an electrical reference signal representing the disturbing noise signal generated by a noise source. This reference signal is fed to an adaptive filter that outputs an actuator driving signal. The actuator driving signal is then supplied to an acoustic actuator (for example, a loudspeaker) that generates a compensation sound field, which has an opposite phase to the noise signal, within a portion of the listening environment. This compensation field thus damps or eliminates the noise signal within this portion of the listening environment. A residual noise signal may be measured using a microphone. The microphone provides an “error signal” to the adaptive filter, where filter coefficients of the adaptive filter are modified such that a norm (for example, power) of the error signal is reduced.
The adaptive filter may use known digital signal processing methods, such as an enhanced least mean squares (LMS) method, to reduce the error signal, or more specifically, the power of the error signal. Examples of such enhanced LMS method include a filtered-x-LMS (FXLMS, x denotes the input reference signal) algorithm or modified versions thereof, or a filtered-error-LMS (FELMS) algorithm.
A model that represents an acoustic transmission path from the acoustic actuator (i.e., the loudspeaker) to the error signal sensor (i.e., the microphone) is used when applying the FXLMS (or any related) algorithm. This acoustic transmission path from the loudspeaker to the microphone is usually referred to as a “secondary path” of the ANC system. In contrast, the acoustic transmission path from the noise source to the microphone is usually referred to as a “primary path” of the ANC system. The estimation of the transmission function (i.e., the frequency response) of the secondary path of the ANC system has a considerable impact on the convergence behavior and stability of an adaptive filter that uses the FXLMS algorithm. Particularly, a varying secondary path transmission function heavily affects the overall performance of the active noise control system. In order to improve the stability normalization of the reference signal has been employed thereby arriving at a normalized filtered-x-LMS (NFXLMS).
However, despite the engineering progress of the recent years there are still problems with respect to stability and overall processor load and speed involved in ANC. Therefore, it is an object of the present application to enhance stability and speed of adaptive filtering comprised in ANC.
The following presents a simplified summary of the disclosure in order to provide a basic understanding of some aspects of the disclosure. This summary is not an exhaustive overview of the disclosure. It is not intended to identify key or critical elements of the disclosure or to delineate the scope of the disclosure. Its sole purpose is to present some concepts in a simplified form as a prelude to the more detailed description that is discussed later.
In view of the above-mentioned problems, in the present invention it is provided a method of noise reduction, comprising the steps of:
filtering reference signals xk[n], k=1, . . . , K, K being an integer denoting the number of reference signals (channels) in the time domain, representing noise by an adaptive filter comprising adaptive filter coefficients to obtain actuator (loudspeaker) driving signals ym[n], m=1, . . . , M, M being an integer;
outputting the actuator driving signals ym[n] by M loudspeakers to obtain loudspeaker (output) signals (M denoting the number or loudspeakers (loudspeaker output channels in the time domain));
detecting the loudspeaker signals by L microphones, L being an integer (denoting the number of microphones and error channels; see below);
filtering the reference signals by estimated transfer functions representing the transfer of the loudspeaker signals output by the M loudspeakers to the L microphones to obtain filtered reference signals;
updating the filter coefficients of the adaptive filter based on
the filtered reference signals and
previously updated filter coefficients of the adaptive filter multiplied by leakage factors.
The method may comprise transforming the reference signals xk[n] into the frequency domain to obtain reference signals in the frequency domain Xk[k] and the filtering of the reference signals by estimated transfer functions may be performed in the frequency domain.
For example, a noise sensor such as, for example, a microphone or a non-acoustic sensor may be used to obtain the reference signals. Whereas in the art, updating is performed based on previously updated filter coefficients (at a time n) and transfer functions representing the transfer of the loudspeaker signals output by the M loudspeakers to the L microphones to obtain updated filter coefficients (at a time n+1) a leakage matrix consisting of leakage factors is employed according to an embodiment of the invention. By a leakage matrix, pre-determined ones of the previously updated filter coefficients can be modified, for example, set to zero by multiplication with zero-valued leakage factors either in the time or frequency domain (in terms of processor load processing in the frequency domain may be preferred). For example, pre-determined ones of the previously updated filter coefficients can be multiplied by leakage factors in the range of 0.5 to 0.01 or 0.0001 or 0. Thereby, the stability of the adaptation algorithm for updating the filter coefficients of the adaptive filter can be significantly improved (see also detailed description below). The method according to this embodiment as well as the methods according to the embodiments described below can be applied in the context active noise cancellation, particular, road noise cancellation, in vehicle compartments. In-vehicle communication/entertainment in automobiles, for example, can be improved by implementation of the methods in in-vehicle communication/entertainment systems.
It has to be noted that the introduction of leakage factors may slow down the convergence speed of the adaptation procedure for updating the filter coefficients. Depending on the actual application this may be considered acceptable given the advantage of the increased stability. On the other hand, the convergence speed may be increased by the introduction of non-constant adaptation sizes. For example, according to the Filtered X Least Mean Square (FXLMS) algorithm of the art updating of coefficients w of a matrix is basically achieved according to w(n+1)=w(n)+μ e(n) z(n), with e(n) denoting a residual error and z(n) denoting a reference signal filtered through a secondary path model and μ being the constant adaptation size governing speed and stability of the convergence process. Contrary, according to an embodiment an adaptation step size of the updating of the filter coefficients of the adaptive filter is not constant, in particular, frequency dependent. In fact, the adaptation step sizes may be individually fine-tuned according to the actual application thereby increasing the overall convergence of the filter coefficient adaptation.
It is noted that the approaches of the introduction of the leakage factors and the introduction of non-constant adaptation step sizes may be combined or may be alternatively implemented independently from each other. Thus, it is also provided herein a method of noise reduction, comprising the steps of:
filtering reference signals xk[n], k=1, . . . , K, K being an integer denoting the number of reference signals (channels) in the time domain, representing noise by an adaptive filter comprising adaptive filter coefficients to obtain actuator (loudspeaker) driving signals ym[n], m=1, . . . , M, M being an integer;
outputting the actuator driving signals ym[n] by M loudspeakers to obtain loudspeaker (output) signals (M denoting the number or loudspeakers (loudspeaker output channels in the time domain));
detecting the loudspeaker signals by L microphones, L being an integer (denoting the number of microphones and error channels; see below);
filtering the reference signals by estimated transfer functions representing the transfer of the loudspeaker signals output by the M loudspeakers to the L microphones to obtain filtered reference signals; and
updating the filter coefficients of the adaptive filter based on
the filtered reference signals; and
previously updated filter coefficients of the adaptive filter;
and wherein the updating is performed using non-constant, in particular, frequency-dependent, adaptation step sizes.
The method may comprise transforming the reference signals xk[n] into the frequency domain to obtain reference signals in the frequency domain Xk[k] and the filtering of the reference signals by estimated transfer functions may be performed in the frequency domain.
In any case, the above-described embodiments may be supplemented by determining at least one control parameter of a vehicle, for example, selected from a group consisting of the speed of the vehicle, a pressure of a tire of the vehicle, information indicating that the vehicle is off-road, information on a driving mode of the vehicle, information on a closed/open state of doors and/or the trunk and/or windows and/or the roof of the vehicle and an audio level adjusted for an audio device of the vehicle and controlling the adaptation step sizes depending on the determined at least one parameter of the vehicle. In particular, the adaptation step sizes may depend on time-dependent control parameters. Depending on different applications and/or driving situations different sets of adaptation step sizes may be used in the process of updating the filter coefficients of the adaptive filter. Thus, the updating process can be dynamically adjusted to the current circumstances, for example, the current driving situation in the context of automotive applications.
In all of the above-described examples the updating of the filter coefficients of the adaptive filter may at least partly be performed in the frequency domain in order to save processing time. In this case, a matrix of the Fourier transformed previously updated filter coefficients can be multiplied by a matrix of leakage coefficients (given in the frequency domain). As known in the art, signal representations in the time domain may be transformed into the frequency domain by (Fast) Fourier transforms and signal representations in the frequency domain may be transformed into the time domain by Inverse (Fast) Fourier transforms.
According to a particular embodiment, the updating of the filter coefficients of the adaptive filter is performed according to:
w
k,m
[n+1]=IFFT(Woldk,m[k]Vk,m[k]−Ck,m[k]),
wherein wk,m[n+1] are the filter coefficients of the adaptive filter updated at time step n+1, IFFT is an Inverse Fast Fourier Transform, and Woldk,m[k] denotes the filter coefficients wk,m[n] of the previous time step n transformed into the frequency domain, Vk,m[k] a leakage matrix comprising the frequency dependent leakage factors and wherein Ck,m[k] is the product of the adaptation step sizes (μ, μk,m[k] or μSPk,m[k]; see below) used for the updating of the filter coefficients and a summed cross spectrum
where conj denotes the conjugate operation (matrix), Xk[k] are the reference signals transformed into the frequency domain, Ŝm,l[k] is a matrix of the estimated transfer functions (of the secondary path, i.e., representing the transfer of the loudspeaker signals output by the M loudspeakers to the L microphones) in the frequency domain and El[k], with l=1, . . . , L, are error signals in the frequency domain obtained by the L microphones. As usual the error signals measure the success of noise cancellation and have to be minimized by adaptation of the adaptive filter.
In principle, when using the concrete algorithm wk,m[n+1]=IFFT(Woldk,m[k] Vk,m[k]−Ck,m[k]), the adaptation step sizes can be given by a global constant adaptation step size μ used for all k, m or a frequency-dependent matrix μk,m[k] comprising values of the adaptation step sizes or a time-dependent and frequency-dependent matrix μSPk,m[k] comprising values of the adaptation step sizes. Dynamic control parameters may be determined and the adaptation step sizes may be given by a time-dependent and frequency-dependent matrix μSPk,m[k] comprising values of the adaptation step sizes that depend on the determined dynamic control parameters. The dynamic control parameters may be selected from a group consisting of a current vehicle speed, tire pressure, vehicle on- or off-road status, dynamic driving modes, door/rooftop/trunk open/close states, windows/sunroof open/close states or an infotainment/entertainment operation/audio level.
Furthermore, it is provided herein a computer program product comprising one or more computer readable media having computer-executable instructions for performing the steps of the method according to one of the above-described embodiments of the method of noise reduction when run on a computer.
In order to address the above-mentioned object it is also provided a noise reduction apparatus, comprising:
a first adaptive filter comprising filter coefficients configured for adaptively filtering reference signals xk[n], k=1, . . . , K, K being an integer, representing noise to obtain a actuator (loudspeaker) driving signals ym[n];
M loudspeakers configured for outputting the actuator driving signals ym[n], m=1, . . . , M, M being an integer, to obtain loudspeaker signals;
microphones configured for detecting the loudspeaker signals;
a second filter configured for filtering the reference signals by estimated transfer functions representing the transfer of the loudspeaker signals output by the M loudspeakers to the microphones to obtain filtered reference signals;
an adaptation unit configured for updating the filter coefficients of the adaptive filter based on the filtered reference signals and previously updated filter coefficients of the adaptive filter including multiplying at least some of the values of the previously updated filter coefficients by leakage factors.
The noise reduction apparatus may be configured to perform the steps of any of the above-described embodiments of the method of noise reduction. Particularly, it is provided a noise reduction apparatus, comprising:
a first adaptive filter comprising filter coefficients configured for adaptively filtering reference signals xk[n], k=1, . . . , K, K being an integer, representing noise to obtain a actuator driving signals ym[n];
M loudspeakers configured for outputting the actuator driving signals ym[n], m=1, . . . , M, M being an integer, to obtain loudspeaker signals;
microphones configured for detecting the loudspeaker signals;
a second filter configured for filtering the reference signals by estimated transfer functions representing the transfer of the loudspeaker signals output by the M loudspeakers to the microphones to obtain filtered reference signals; and
an adaptation unit configured for updating the filter coefficients of the adaptive filter based on the filtered reference signals and previously updated filter coefficients of the adaptive filter and non-constant (for example, frequency-dependent) adaptation step sizes.
Examples of the herein provide a signal processor that can be advantageously used in a variety of electronic communication devices. In particular, it is provided an Active Noise Control system, in particular, an Active Noise Control system, comprising the noise reduction apparatus as described above.
Additional features and advantages of the present invention will be described with reference to the drawings. In the description, reference is made to the accompanying figures that are meant to illustrate preferred embodiments of the invention. It is understood that such embodiments do not represent the full scope of the invention.
While the present disclosure is described with reference to the examples as illustrated in the following detailed description as well as in the drawings, it should be understood that the following detailed description as well as the drawings are not intended to limit the subject matter to the particular illustrative embodiments disclosed, but rather the described illustrative embodiments merely exemplify the various aspects, the scope of which is defined by the appended claims.
The present invention relates to active noise cancellation, in particular, in automotive applications. For example, methods and an apparatus are provided that are suitable for the reduction of noise in vehicle compartments wherein the noise can be road noise.
A vehicle communication system may be installed in a vehicle passenger compartment 111 having a front end 112 and a rear end 113. A front seat 114 provides seating for a driver, and a rear seat 115 provides seating for the rear passengers. For example, four microphones 120-126 are located adjacent to four loudspeakers 130-136 in the vehicle passenger compartment 111. The first microphone 120 and the second microphone 122 are located at the front end 112 of the vehicle. A third microphone 124 and a fourth microphone 126 are located at the rear end 113 of the vehicle. First and second loudspeakers 130 and 132 are located adjacent to the first and second microphones 120 and 122 and third and fourth loudspeakers 134 and 136 are located adjacent to the third and fourth microphones 124 and 126. The loudspeakers 130-136 may be used by an audio entertainment system. Input signals from the microphones 120-126 are provided to a signal processing circuit 140 which interprets the signals and provides output signals to the loudspeakers 130-136. The signal processing circuit 140 can be located behind a vehicle dashboard 116, for example.
In the following, the ANC system 10 of
The multichannel reference signals xk[n] are fed to an adaptive filter 11, for example, a finite impulse response (FIR) filter. The loudspeaker driving signals (compensation signals) ym[n] are supplied to loudspeakers 12 that output compensation sound fields with opposite phase as compared to the reference signals xk[n] within at least a portion of a listener environment, for example, a vehicle compartment. The index in denotes the loudspeaker output channels (m=1, . . . , M, M being the number of the loudspeakers 12). Residual noise signals are measured by microphones 13. The microphones 13 provide error signals el[n] where l=1, . . . , L, L being the number of the microphones 13). In principle, the adaptive filter coefficients wk,m[n] of the adaptive filter 11 are to be adjusted (updated) such that a norm (for example, the power) of the error signals is reduced (minimized). The signals detected by the microphones 13 results from the combination of the multichannel reference signals xk[n] after being modified according to the transfer functions pk,l[n] of the acoustic transmission path of the listener environment from the noise source to the microphones 13 (primary path of the ANC system 10) and the loudspeaker output signals modified according to the transfer functions sm,l[n] of the acoustic transmission path of the listener environment from the loudspeakers 12 to the microphones 13 (secondary path of the ANC system 10). The loudspeaker signals as detected by the microphones 13, i.e., after having traveled the acoustic transmission path from the loudspeakers 12 to the microphones 13 are denoted by y′m[n]. The multichannel reference signals modified according to the transfer functions pk,l[n] of the acoustic transmission path of the listener environment from the noise source to the microphones 13 are denoted by x′k[n]. The microphones 13 are installed in the listener environment and the error signals el[n] output by the microphones 13 measure the difference between y′m[n] and x′k[n]. A model that represents the secondary path has to be used when applying an appropriate algorithm for adjusting (updating) the adaptive filter coefficients wk,m[n] of the adaptive filter 11 in order to minimize the error signals el[n]. The signal power of the error signals el[n] may be regarded as a quality measure for the noise cancellation obtained by ANC system 10.
According to the example illustrated in
where conj denotes the conjugate operation (matrix). Moreover, the processor 17 calculates the updated filter coefficients of the adaptive filter 11. The processor 17 reads data from a memory 20 used for the updating process.
According to an embodiment, the processor 17 reads a leakage matrix Vk,m[k] comprising frequency dependent leakage factors from the memory 20. Alternatively or additionally the processor 17 reads a matrix of frequency dependent adaptation step sizes μk,m[k] from the memory 20. In the following, examples of the updating algorithm according to the invention will be described in detail. After adaptation of the filter coefficients in the frequency domain by the processor 17 the adapted filter coefficients are input in an Inverse Fast Fourier Transform 18 to provide the adaptive filter 11 with the adapted filter coefficients in the time domain.
In principle, the summed cross spectrum SCSk,m[k] could be used for updating the filter coefficients wk,m[n] of the adaptive filter 11 simply according to:
w
k,m
[n+1]=wk,m[n]−μIFFT(SCSk,m[k]), (Equation 1)
where μ is the constant adaptation step size and IFFT denotes an Inverse Fast Fourier Transform operation. This procedure is known to be applied in the Filtered X Least Means Square (FXLMS) algorithm of the art.
However, stability of the FXLMS algorithm is heavily affected by the accuracy of the estimation of the secondary path of the ANC system 10 and the level of disturbances in the multichannel reference signals xk[n]. Particularly, time dependent variations of the secondary path and the disturbances in the multichannel reference signals xk[n] cause instabilities of the FXLMS algorithms of the art. According to an embodiment of the present invention stability of the updating procedure is significantly improved by a leakage matrix used in an updating time step n+1 to modify values of filter coefficients obtained for a previous time step n.
An example of the employment of a leakage matrix is illustrated in
As shown in
As shown in
w
k,m
[n+1]=IFFT(Woldk,m[k]Vk,m[k]−Ck,m[k]), (Equation 2)
where again IFFT denotes an Inverse Fast Fourier Transform operation.
Whereas employment of the leakage matrix Vk,m[k] increase stability, it may reduce convergence speed. According to another embodiment, that might be combined with the embodiment related to the leakage matrix Vk,m[k], convergence (adaptation, updating) speed can be enhanced by the employment of frequency dependent adaptation step sizes μk,m[k] instead of a global constant adaptation step size μ. In this an algorithm according to:
w
k,m
[n+1]=wk,m[n]−IFFT(μk,m[k]SCSk,m[k]) (Equation 3)
or
w
k,m
[n+1]=wk,m[n]−IFFT(μk,m[k]
might be employed.
The adaptation step sizes μk,m[k] are shaped over all frequency bins for each filter matrix index ‘m’ and ‘k’ according to one particular pre-determined step size tuning set. In principle, it is possible to provide for a plurality of different step size tuning sets. In the automotive context, this might prove helpful in order to adapt to different vehicle variants and dynamic conditions as, for example, the vehicle body and suspension variant, tire pressure, type of tire, information about dynamic chassis/suspension control (e.g. sport/comfort mode), weather conditions, road conditions or other RNC resonance related control information. A particular one of tuning sets that might be stored in the memory 20, for example, in form of a look-up table, of the ANC system 10 can be selected (for example, by user input or automatically based on reception of accordingly designed control signals, based on the vehicle variants and/or dynamical conditions.
As compared to updating of the filter coefficients of the adaptive filter 11 based on a global constant adaptation step sizes μ employment of frequency dependent adaptation step sizes μk,m[k] is more expensive in terms of the processor load and memory demands. However, employment of frequency dependent adaptation step sizes μk,m[k] allows for improving the updating process significantly.
Instead of being restricted to one single global adaptation step size, the adaptation step size can be individually adjusted for a particular configuration of an in-vehicle communication system, for example, particular loudspeakers, accelerometers, external boundary conditions, etc. Moreover, the individually adjusted adaptation step sizes offer the flexibility to fine-tune each seat position in a vehicle, for example, by an individual weighting with respect to rumble and torus in order to increase the adaptation performance or with respect to individual frequency roll-off definitions in order to increase the adaptation stability. Beside resonances such a technique can also handle individual seat location constraints, because front and rear suspension, if mechanically decoupled, show decoupled noise impact on different seat positions within the vehicle compartment. Thereby, the system performance can be improved because the algorithm is more focused to cancel around the resonance frequencies and as such, the robustness of the adaptation algorithm will be increased since a disturbing noise that is not coherent to road noise will be ignored within tuned notches if the adaptation step size design is properly selected.
Additionally, the maximum frequency of operation can be defined individually by applying a roll-off in order to further enhance stability of the adaptation procedure. For example, the roll-off frequency can be set to 500 Hz. In particular, simulation studies have proven that when the roll-off frequency is beneficially set the system robustness against temporal changes in the secondary path can be significantly improved. Since road noise is showing dedicated resonances in rumble and torus inside the vehicle compartment the employment of frequency dependent adaptation step sizes μk,m[k] is particularly advantageous in the context of RNC.
According to different embodiments, the frequency dependent adaptation step sizes μk,m[k] may be static or may be adjusted in a time dependent manner (“on the-fly”), in the following time-dependent and frequency dependent adaptation step sizes depending on dynamic control parameters are denoted by μSPk,m[k]. In this case, the μk,m[k] may be functions of time-dependent control parameters. The time-dependent control parameters can be parameters that potentially have an impact to level and pitch of the RNC related chassis and body resonances. The time-dependent control parameters may be chosen from the group comprising the current vehicle speed, tire pressure, vehicle on- or off-road status, dynamic driving modes as, for example, sport and comfort modes, door/rooftop/trunk open/close states, windows/sunroof open/close states, an infotainment/entertainment operation/audio level, etc.
Although this approach based on time-dependent and frequency dependent adaptation step sizes μSPk,m[k] is relatively expensive in terms of processor loads and requires a detailed understanding, for example, of the correlation between the speed and the corresponding resonances, it may nevertheless be implemented due to the enhancements that may be achieved. For example, it allows for dynamic scaling and pitching of the adaptation step sizes based on speed dependent resonances which increase performance of the adaptation algorithm. The approach allows for the reduction or limitation of the spectral bandwidth of the adaptation step size for vehicle events having an impact on secondary path modifications such as opening/closing of doors or other openings such as a sunroof. Thereby, the stability of the adaptation algorithm can be increased. Moreover, this approach allows for a temporary freeze of the filter adaptation due to special vehicle/user conditions. Such conditions may comprise a set high music volume beyond 70 dBSPL(A), for example, a vehicle in off-road status wherein many impulsive disturbances are to be expected, and a vehicle speed above some pre-defined limit wherein wind noise is the most dominant factor μSPk,m[k] may prove useful.
If time-dependent adaptation step sizes μSPk,m[k] are used it might be useful to set upper μmax[k] and lower μmin[k] boundary limits in order to guarantee stability of the adaptation algorithm, i.e., μSPk,m[k]ε[μmax[k], μmin[k]].
An example for implementation of time-dependent adaptation step sizes μk,m[k] being functions of time-dependent control parameters is illustrated in
As already mentioned above the embodiments related to the leakage matrix and the frequency-dependent adaptation sizes μk,m[k] (as well as time-dependent and frequency dependent adaptation step sizes μSPk,m[k]) can be combined with each other. In particular, Ck,m[k]=μ SCSk,m[k] in Equation 2 may be replaced by Ck,m[k]=μk,m[k] SCSk,m[k] or Ck,m[k]=μSPk,m[k]SCSk,m[k], respectively.
All previously discussed embodiments are not intended as limitations but serve as examples illustrating features and advantages of the invention. It is to be understood that some or all of the above described features can also be combined in different ways.
| Number | Date | Country | Kind |
|---|---|---|---|
| 15200631.8 | Dec 2015 | EP | regional |
