The present invention relates to audio signal processing, and, in particular, to an apparatus and method for reducing the complexity of multichannel interference cancellation and for low complexity multichannel interference cancellation.

Modern hands-free communication devices employ multiple microphone signals, for, e.g., speech enhancement, room geometry inference or automatic speech recognition. These devices range from voice-activated assistants, smart-home devices and smart speakers, to smart phones, tablets or personal computers. Many smart devices, such as voice-activated assistants, smart-phones, tablets or personal computers, are equipped with loudspeakers. Considering such devices, for example, a device which also integrates at least one loudspeaker, an acoustic interference canceler is applied to each microphone's output to reduce the electroacoustic coupling.

Acoustic echo cancellation (AEC) (see, e.g., [1]) is the most widely-used technique to reduce electro-acoustic coupling between ioudspeaker(s) and microphone(s) in hands-free communication set-ups. Given such a set-up, microphones acquire, in addition to the desired near-end speech, acoustic echoes and background noise. AEC uses adaptive filtering techniques (see, e.g., [2]) to estimate the acoustic impulse responses (AIRs) between loudspeaker(s) and microphone(s). Subsequently, acoustic echo estimates are computed by filtering the available loudspeaker signal with the estimated AIRs. Finally, the estimated acoustic echoes are subtracted from the microphone signals, resulting in the cancellation of acoustic echoes.

In the particular case of acoustic echo cancellation (AEC), electro-acoustic coupling is caused by the far-end speaker signal that is reproduced by the loudspeaker. Yet, in the aforementioned hands-free communication devices, it can also be caused by the device's own feedback, music, or the voice assistant. The most straightforward solution to reduce the electro-acoustic coupling between loudspeaker and microphones is to place an acoustic interference canceler at the output of each microphone (see, e.g., [3]).

Relative transfer functions model the relation between frequency-domain AIRs, commonly denoted as acoustic transfer functions (ATFs). RTFs (RTF means relative transfer function) are commonly used in the context of multi-microphone speech enhancement (see, e.g., [5], [8], [12]). Considering more related applications, the estimation of residual echo relative transfer functions was employed in [13], [14] to estimate the power spectral density of the residual echo, e.g., the acoustic echo components that remain after cancellation, of a primary channel. To enhance the estimation process, a second microphone signal is used. The proposed method in [13], [14] estimates the relation between the primary signal after cancellation and a secondary microphone signal, providing a relation between the error in the estimation of the primary AIR and a secondary AIR. Finally, the residual echo relative transfer function is used to compute the power spectral density of the primary residual acoustic echo.

Considering the specific application of microphone array processing, several methods have been presented which aim at the complexity reduction of the entire speech enhancement algorithms, e.g. spatial filtering combined with AEC. For example, the use of a single AEC placed at the output of the spatial filter was first studied in [3], [15]. Some alternative methods which aim at integrating acoustic echo cancellation and microphone array processing have been proposed in [8], [16], [18].

As the complexity of a multi-microphone acoustic interference canceler is proportional to the number of microphones, for many modern devices, such an increment in complexity is not attainable.

It would therefore be highly appreciated, if low complexity concepts for multichannel interference cancellation would be provided.

The object of the present invention is to provide low complexity concepts for multichannel interference cancellation. The object of the present invention is solved by an apparatus according to claim 1 , by a method according to claim 14 and by a computer program according to claim 15.

An apparatus for multichannel interference cancellation in a received audio signal comprising two or more received audio channels to obtain a modified audio signal comprising two or more modified audio channels according to an embodiment is provided.

The apparatus comprises a first filter unit being configured to generate a first estimation of a first interference signal depending on a reference signal.

Moreover, the apparatus comprises a first interference canceller being configured to generate a first modified audio channel of the two or more modified audio channels from a first received audio channel of the two or more received audio channels depending on the first estimation of the first interference signal.

Furthermore, the apparatus comprises a second filter unit being configured to generate a second estimation of a second interference signal depending on the first estimation of the first interference signal.

Moreover, the apparatus comprises a second interference canceller being configured to generate a second modified audio channel of the two or more modified audio channels from a second received audio channel of the two or more received audio channels depending on the second estimation of the second interference signal.

Embodiments provide concepts, e.g., an apparatus and a method, for multichannel interference cancellation using relative transfer functions.

For example for AEC, concepts according to embodiments use an estimate of a primary acoustic echo signal to compute estimates of the remaining, or secondary, acoustic echo signals. In order to do so, the relation between primary acoustic impulse responses (AIRs), e.g., the AIRs between the loudspeaker and the primary microphones, and secondary AIRs, e.g., the AIRs between the loudspeaker and secondary microphones, are identified. Subsequently, the secondary acoustic echo signals are computed by filtering a primary acoustic echo signal with the estimated relation between AIRs. Finally, cancellation is applied to each and every microphone signal. If the inter-microphone distance is small, these relations can be modeled using relatively short filters. Thus, the computational complexity can be reduced.

Moreover, a method for multichannel interference cancellation in a received audio signal comprising two or more received audio channels to obtain a modified audio signal comprising two or more modified audio channels according to an embodiment is provided.

The method comprises:

Generating a first estimation of a first interference signal depending on a reference signal.

- Generating a first modified audio channel of the two or more modified audio channels from a first received audio channel of the two or more received audio channels depending on the first estimation of the first interference signal.

Generating a second estimation of a second interference signal depending on the first estimation of the first interference signal. And:

Generating a second modified audio channel of the two or more modified audio channels from a second received audio channel of the two or more received audio channels depending on the second estimation of the second interference signal.

Furthermore, a computer program is provided, wherein the computer program is configured to implement the above-described method when being executed on a computer or signal processor.

In the following, embodiments of the present invention are described in more detail with reference to the figures, in which:

Fig. 1 a illustrates an apparatus for for multichannel interference cancellation according to an embodiment,

illustrates an apparatus for for multichannel interference cancellation according to another embodiment,

illustrates an apparatus for for multichannel interference cancellation according to a further embodiment,

illustrates Multi-microphone AEC,

Fig. 3 illustrates multi-microphone AEC according to an embodiment,

Fig. 4 illustrates multi-microphone AEC in the STFT domain,

Fig. 5 illustrates multi-microphone AEC in the STFT domain according to an embodiment,

Fig. 6 depicts the results corresponding to the simulations with truncated AIRs,

Fig. 7 depicts a comparison between AETF and RETF-based AEC with

T60 = 0.15s and L = 256 taps, and

Fig. 8 illustrates a comparison between AETF and RETF-based AEC with

Γ60 = 0.35s and L = 1024 taps.

Fig. 1 a illustrates an apparatus for multichannel interference cancellation according to an embodiment.

The apparatus comprises a first filter unit 1 12 being configured to generate a first estimation of a first interference signal depending on a reference signal x(t) .

Moreover, the apparatus comprises a first interference canceller 1 14 being configured to generate a first modified audio channel e1 (t) of the two or more modified audio channels from a first received audio channel yx (t) of the two or more received audio channels depending on the first estimation ) of the first interference signal.

Furthermore, the apparatus comprises a second filter unit 122 being configured to generate a second estimation of a second interference signal depending on the first

estimation of the first interference signal.

Moreover, the apparatus comprises a second interference canceller 124 being configured to generate a second modified audio channel 0 of the two or more modified audio

channels from a second received audio channel ) of the two or more received audio channels depending on the second estimation of the second interference signal.

Embodiments are based on the finding that the first estimation of the first interference signal may be used to generate the second estimation of a second interference signal. Reusing the first estimation of the first interference signal for determining the second estimation of the second interference signal reduces computational complexity compared to solutions that generate the second estimation of the second interference signal by using the reference signal instead of using the first estimation of the first interference signal.

Some of the embodiments relate to acoustic echo cancellation (AEC).

In an embodiment, the first estimation of the first interference signal may, e.g., be a first estimation of a first acoustic echo signal, the second estimation of the second interference signal is a second estimation of a second acoustic echo signal.

The first interference canceller 114 may, e.g., be configured to conduct acoustic echo cancellation on the first received audio channel (e.g., by subtracting the first estimation of the first acoustic echo signal from the first received audio channel) to obtain the first modified audio channel.

The second interference canceller 124 may, e.g., be configured to conduct acoustic echo cancellation on the second received audio channel (e.g., by subtracting the second estimation of the second acoustic echo signal from the second received audio channel) to obtain the second modified audio channel.

Fig. 1 b illustrates an apparatus for for multichannel interference cancellation according to another embodiment.

Compared to the apparatus of Fig. 1 a, the apparatus of Fig. 1 b further comprises a third filter unit 132 and a third interference canceller 134.

In the embodiment of Fig. 1 b, the received audio signal comprises three or more received audio channels, and the modified audio signal comprises three or more modified audio channels.

The third filter unit 132 is configured to generate a third estimation of a third

interference signal depending on the first estimation of the first interference signal.

The third interference canceller 134 is configured to generate a third modified audio channel e3(t) of the three or more modified audio channels from a third received audio channel y3 (t) of the three or more received audio channels depending on the third

estimation of the third interference signal.

Fig. 1c illustrates an apparatus for for multichannel interference cancellation according to a further embodiment.

Compared to the apparatus of Fig. 1a, the apparatus of Fig. 1c further comprises a third filter unit 132 and a third interference canceller 134.

In the embodiment of Fig. 1c, the received audio signal comprises three or more received audio channels, and the modified audio signal comprises three or more modified audio channels.

The third filter unit 132 is configured to generate a third estimation of a third

interference signal depending on the second estimation of the second interference

signal. Thus, the embodiment of Fig. 1c differs from the embodiment of Fig. 1b in that generating the third estimation
of the third interference signal is conducted

depending on the second estimation
of the second interference signal instead of depending on the first estimation of the first interference signal.

The third interference canceller 134 is configured to generate a third modified audio channel e3 (t) of the two or more modified audio channels from a third received audio channel y3 (t) of the two or more received audio channels depending on the third estimation ) of the third interference signal.

In other embodiments (which implement the optional dashed line 199 in Fig. 1 c), the third filter unit 132 is configured to generate a third estimation of a third interference

signal depending on the second estimation
of the second interference signal and depending on the first estimation of the first interference signal.

Fig. 2 illustrates multi-microphone AEC according to the prior art. In that prior art approach, a first filter unit 282 is used to generate a first estimation of a first

interference signal from a reference signal .

A first interference canceller 284 then generates a first modified audio channel e, (t) from a first received audio channel yx (t) of the two or more received audio channels depending on the first estimation of the first interference signal.

In the prior art approach of Fig. 2, a second filter unit 292 generates a second estimation of a second interference signal from the reference signal x(t) that was also used

by the first filter unit 282.

A second interference canceller 294 then generates a second modified audio channel eN (t) from a second received audio channel yN (t) of the two or more received audio

channels depending on the second estimation of the second interference signal.

Some embodiments reduce the complexity of multi-microphone Acoustic Echo Cancellation (AEC) that is depicted in Fig. 2, by using a relative transfer function (RTF) based approach, as depicted in Fig. 3. Relative transfer functions are described in [4], [7].

Fig. 3 illustrates multi-microphone Acoustic Echo Cancellation (AEC) according to embodiments. In Fig. 3, a first filter unit 312 is used to generate a first estimation of
a first interference signal from a reference signal x{t) .

A first interference canceller 314 then generates a first modified audio channel ex (t) from a first received audio channel yx (t) of the two or more received audio channels depending on the first estimation of the first interference signal.

The apparatus of Fig. 3 now differs from Fig. 2 in that, a second filter unit 322 generates a second estimation
of a second interference signal depending on the first estimation

of the first interference signal that was generated by the first filter unit 312.

A second interference canceller 324 then generates a second modified audio channel eN (t) from a second received audio channel yN if) of the two or more received audio

channels depending on the second estimation of the second interference signal.

Some embodiments reduce the complexity of multi-microphone Acoustic Echo Cancellation (AEC) that is depicted in Fig. 2, by using a relative transfer function (RTF) based approach, as depicted in Fig. 3. Relative transfer functions are described in [4], [7].

Embodiments use an estimate of a primary interference signal, to compute estimates of the remaining, or secondary, interference signals. To estimate a primary interference signal, a primary filter, which characterizes the relation between a reference signal and a primary received signal, is identified. An estimate of a primary interference signal is then obtained by filtering a reference signal with an estimate of a primary filter. Afterwards, the secondary filters, e.g. the filters that characterize the relation between an estimated primary interference signal and the secondary received signals, are identified. Subsequently, estimates of the secondary interference signals are computed by filtering an estimate of a primary interference signal with the estimated secondary filters. Finally, cancellation is applied to reduce the electro-acoustic coupling. If the distance between microphones is small, secondary filters are shorter than primary filters (see, e.g., [10], [19]), which leads to the reduction of the computational complexity.

Some embodiments are used for acoustic echo cancellation. To this end, Fig. 3 depicts a hands-free communication scenario with one loudspeaker (one transmitter) and N microphones (receivers). In this particular case, the reference signal is the loudspeaker signal x(t), the primary microphone signal is y^t), without loss of generality, and / denotes the discrete time index. Further, an estimate of the primary filter is denoted as

being an estimate of the primary acoustic echo (interference) signal and the

signal after cancellation As it can be observed, a secondary acoustic

echo signal is computed by filtering an estimate of the primary acoustic echo signal

with an estimate of a secondary filter It should be noted that a delay of

D≥ 0 samples is introduced to the secondary microphone signal. This is done to ensure that D non-causal coefficients of the secondary filters are estimated. In case the microphones need to be synchronized, the primary signal after cancellation also needs to be delayed by D samples. In contrast, classical interference cancellation schemes (as depicted in Fig. 2), compute estimates of the N received signals by filtering the reference x(t) signal with N estimated primary filters.

In the following, a step-by-step approach according to some of the embodiments is provided:

1.) A primary interference signal is estimated using a reference signal. In the specific application of acoustic echo cancellation, the former is the acoustic echo signal, and the latter is the loudspeaker signal. To do so:

1.1. ) a primary filter, which characterizes the relation between a reference signal and a primary receiver signal, this being either

(a) a single receiver signal,

(b) a linear combination of receiver signals,

is identified by using, e.g., adaptive filtering techniques.

1.2. ) a reference signal is filtered with an estimate of a primary filter to compute an estimate of a primary interference signal.

1.3. ) interference cancellation is applied by subtracting an estimate of a primary interference signal from a primary received signal, this being either

(a) a single receiver signal.

(b) a linear combination of receiver signals.

2. ) A secondary interference signal is estimated based on an estimate of a primary interference signal. To do so:

2.1. ) a secondary filter, which characterizes the relation between an estimate of a primary interference signal and a secondary received signal, is identified by, e.g.,

i. ) optimization of a cost-function or error criterion (e.g. mean- squared error, (weighted) least-squared error, etc..)

ii. ) an adaptive filtering technique in time, frequency or sub-band domain.

using a secondary receiver signal or a secondary signal after cancellation, and an estimate of a primary interference signal. (The secondary filter may, e.g., be considered as a filter configuration.)

2.2. ) an estimate of a primary interference signal is filtered with an estimate of a secondary filter, to compute an estimate of a secondary interference signal.

2.3. ) interference cancellation is applied by subtracting an estimate of a secondary interference signal from a secondary receiver signal.

3. ) Repeat 2. for each secondary interference signal.

4. ) Repeat 1., 2. and 3. for each reference signal.

5. ) Where a transmitter is a loudspeaker and a receiver is a microphone.

6. ) Where an estimate of a secondary interference signal can be used as an estimate of a primary interference signal leading to a cascaded configuration.

7. ) Where for more than two receivers, subsets of receivers can be defined, each of them having a primary receiver.

Further embodiments apply only some of the steps above and/or apply the steps in a different order.

In the following, embodiments which use STFT-domain adaptive filters are described (STFT means short-time Fourier transform).

Given a hands-free communication set-up with one loudspeaker and N microphones, the n -th microphone signal can be expressed in the STFT domain as

where I and k are, respectively, the time frame and frequency indexes. Further,

is the near-end signal, which comprises near-end speech and background noise, and
is the n -th acoustic echo. The latter is the result of the loudspeaker signal being propagated through the room, and acquired by the » -th microphone. Its

exact formulation in the STFT domain (see, e.g., [20]) is

where superscripts denote transpose and

conjugate transpose, respectively, and K is the transform length. Further, the 6 -th partition of the which is a vector

containing all frequency dependencies (AETF means

acoustic echo transfer function).

It should be noted that AETFs in the STFT domain, which are extensively analyzed in [20], are non-causal. Moreover, the number of partitions, or input frames, that are necessary to estimate L AIR coefficients is where R denotes the

frameshift between subsequent input frames. Due to the non-causality of the AETFs, look-ahead frames of are needed to compute the echo signals.

Let us assume that the frequency selectivity of the STFT analysis and synthesis windows is sufficient such that the frequency dependencies can be neglected. In addition, for notational brevity, according to embodiments, it is assumed that a delay of Bnc frames is introduced to the reproduction path as depicted in Fig. 4. In practice, the capturing path is commonly delayed instead, see, e.g., [7], [20].

The signals in Fig. 4 are signals in a transform domain. In particular, the signals in Fig. 4 are signals in the short-time Fourier transform domain (STFT domain).

In Fig. 4, a first filter unit 482 is used to generate a first estimation of a first

interference signal from a reference signal X(l,k).

A first interference canceller 484 then generates a first modified audio channel

from a first received audio channel of the two or more received audio channels

depending on the first estimation of the first interference signal.

In the approach of Fig. 4, a second filter unit 492 generates a second estimation

of a second interference signal from the reference signal X(l,k) that was also used by the first filter unit 482.

A second interference canceller 494 then generates a second modified audio channel from a second received audio channel of the two or more received

audio channels depending on the second estimation of the second interference

signal.

Fig. 4 illustrates multi-microphone AEC in the STFT domain. In practice, the capturing path is commonly delayed instead, see, e.g., [7], [20]. Now, by using the convolutive transfer function (CTF) approximation (see, e.g., [7]) it is possible to write

where ·* denotes complex conjugation, and, for brevity,
. Adaptive algorithms in AEC are driven by the error signal after cancellation, e.g.,

where is used to denote estimates,

and Superscript H indicates Hermitian. Most

adaptive filters used in AEC are of gradient descent type (see, e.g., [2]), thus a generic update equation is given by

where is the step-size matrix of the adaptive filter, whose formulation depends

on the specific adaptive algorithm used.

In the following, the usage of relative echo transfer functions according to embodiments is described.

Due to computational complexity restrictions, the implementation of multiple-microphone AEC as depicted in Fig. 4 is not always feasible.

According to embodiments, it is proposed to reduce the complexity by using a RETF-based approach, as depicted in Fig. 5 (RETF means relative echo transfer function). Fig. 5 illustrates multi-microphone AEC in the STFT domain according to an embodiment.

Again, the signals in Fig. 5 are signals in a transform domain. In particular, the signals in Fig. 5 are signals in the short-time Fourier transform domain (STFT domain).

Claims

1. An apparatus for multichannel interference cancellation in a received audio signal comprising two or more received audio channels to obtain a modified audio signal comprising two or more modified audio channels, wherein the apparatus comprises:

a first filter unit (1 12; 312; 512) being configured to generate a first estimation of a first interference signal depending on a reference signal,

a first interference canceller (114; 314; 514) being configured to generate a first modified audio channel of the two or more modified audio channels from a first received audio channel of the two or more received audio channels depending on the first estimation of the first interference signal,

a second filter unit (122; 322; 522) being configured to generate a second estimation of a second interference signal depending on the first estimation of the first interference signal, and

a second interference canceller (124; 324; 524) being configured to generate a second modified audio channel of the two or more modified audio channels from a second received audio channel of the two or more received audio channels depending on the second estimation of the second interference signal.

2. An apparatus according to claim 1 ,

wherein the first estimation of the first interference signal is a first estimation of a first acoustic echo signal,

wherein the second estimation of the second interference signal is a second estimation of a second acoustic echo signal,

wherein the first interference canceller (114; 314; 514) is configured to conduct acoustic echo cancellation on the first received audio channel to obtain the first modified audio channel, and

wherein the second interference canceller (124; 324; 524) is configured to conduct acoustic echo cancellation on the second received audio channel to obtain the second modified audio channel.

3. An apparatus according to claim 1 or 2, wherein the two or more received audio channels and the two or more modified audio channels are channels of a transform domain, and wherein the reference signal and the first and second interference signals are signals of the transform domain.

4. An apparatus according to one of the preceding claims, wherein the two or more received audio channels and the two or more modified audio channels are channels of a short-time Fourier transform domain, and wherein the reference signal and the first and second interference signals are signals of the short-time Fourier transform domain.

5. An apparatus according to one of the preceding claims,

wherein the second filter unit (122; 322; 522) is configured to determine a filter configuration depending on the first estimation of the first interference signal and depending on the second received audio channel, and

wherein the second filter unit (122; 322; 522) is configured to determine the second estimation of the second interference signal depending on the first estimation of the first interference signal and depending on the filter configuration.

6. An apparatus according to claim 5,

wherein the second filter unit (122; 322; 522) is configured to determine the filter configuration by minimizing a cost function or by minimizing an error criterion.

7. An apparatus according to claim 5 or 6,

wherein the second filter unit (122; 322; 522) is configured to determine the filter configuration according to

where in is a covariance matrix of

wherein is the cross-correlation vector between

wherein indicates the first estimation of the first interference signal,

wherein indicates the second received audio channel,

wherein ί denotes a time index, and wherein k indicates a frequency index.

8. An apparatus according to one of claims 1 to 4,

wherein the second filter unit (122; 322; 522) is configured to determine the filter configuration for a second time index depending on the filter configuration for a first time index that precedes the second time index in time, depending on the first estimation of the first interference signal for the first time index, and depending on a sample of the second modified audio channel for the first time index.

9. An apparatus according to claim 8,

wherein the second filter unit (122; 322; 522) is configured to determine the filter configuration for the second time index according to

wherein
indicates the second time index, and wherein indicates the first time index, and wherein k indicates a frequency index,

wherein
is the filter configuration for the second time index, and

wherein is the filter configuration for the first time index,

wherein is the first estimation of the first interference signal for the first

time index,

wherein is the second modified audio channel for the first time index,

wherein is a step-size matrix.

10. An apparatus according to one of claims 1 to 3,

wherein the two or more received audio channels and the two or more modified audio channels are channels of a partitioned-block frequency domain, wherein each of the two or more received audio channels and the two or more modified audio channels comprises a plurality of partitions, and

wherein the reference signal and the first and the second interference signals are signals of the partitioned-block frequency domain, wherein each of the reference signal and the first and the second interference signals comprises a plurality of partitions.

1 1. An apparatus according to claim 10,

wherein the second filter unit (122; 322; 522) is configured to determine a filter configuration depending on the first estimation of the first interference signal and depending on the second received audio channel,

wherein the second filter unit (122; 322; 522) is configured to determine the second estimation of the second interference signal depending on the first estimation of the first interference signal and depending on the filter configuration,

wherein the second filter unit (122; 322; 522) is configured to determine the filter configuration for a second time index depending on the filter configuration for a first time index that precedes the second time index in time, depending on the first estimation of the first interference signal for the first time index, and depending on a sample of the second modified audio channel for the first time index.

12. An apparatus according to claim 1 1 ,

wherein the second filter unit (122; 322; 522) is configured to determine the filter configuration in the partitioned-block frequency domain according to

wherein indicates the second time index, and wherein £ indicates the first

time index, and wherein k indicates a frequency index,

wherein is the filter configuration for the second time index, and wherein

is the filter configuration for the first time index,

wherein is the first estimation of the first interference signal for the first time

index,

wherein C„ is a step-size matrix,

wherein is the second modified audio channel for the first time index, and

wherein is a circular convolution constraining matrix.

13. An apparatus according to one of the preceding claims,

wherein the received audio signal comprises three or more received audio channels, and wherein the modified audio signal comprises three or more modified audio channels,

wherein the apparatus further comprises a third filter unit (132) and a third interference canceller (134),

wherein the third filter unit (132) is configured to generate a third estimation of a third interference signal depending on at least one of the first estimation of the first interference signal and the second estimation of the second interference signal,

wherein third interference canceller (134) is configured to generate a third modified audio channel e3 (t) of the three or more modified audio channels from a third received audio channel y3 (t) of the three or more received audio channels

depending on the third estimation of the third interference signal.

14. A method for multichannel interference cancellation in a received audio signal comprising two or more received audio channels to obtain a modified audio signal comprising two or more modified audio channels, wherein the method comprises:

generating a first estimation of a first interference signal depending on a reference signal,

generating a first modified audio channel of the two or more modified audio channels from a first received audio channel of the two or more received audio channels depending on the first estimation of the first interference signal,

generating a second estimation of a second interference signal depending on the first estimation of the first interference signal, and

generating a second modified audio channel of the two or more modified audio channels from a second received audio channel of the two or more received audio channels depending on the second estimation of the second interference signal.

15. A computer program for implementing the method of claim 14 when being executed on a computer or signal processor.