WO2000072566A1 - Methods and apparatus for improving adaptive filter performance by signal equalization - Google Patents

Methods and apparatus for improving adaptive filter performance by signal equalization Download PDF

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Publication number
WO2000072566A1
WO2000072566A1 PCT/EP2000/004411 EP0004411W WO0072566A1 WO 2000072566 A1 WO2000072566 A1 WO 2000072566A1 EP 0004411 W EP0004411 W EP 0004411W WO 0072566 A1 WO0072566 A1 WO 0072566A1
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Prior art keywords
far
audio signal
signal
echo
whitening
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PCT/EP2000/004411
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French (fr)
Inventor
Nils Christensson
John Philipsson
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Telefonaktiebolaget Lm Ericsson (Publ)
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Priority to US31534899A priority Critical
Priority to US09/315,348 priority
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Publication of WO2000072566A1 publication Critical patent/WO2000072566A1/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M9/00Interconnection arrangements not involving centralised switching
    • H04M9/08Two-way loud-speaking telephone systems with means for suppressing echoes or otherwise conditioning for one or other direction of traffic
    • H04M9/082Two-way loud-speaking telephone systems with means for suppressing echoes or otherwise conditioning for one or other direction of traffic using echo cancellers

Abstract

In methods and apparatus for echo cancellation, an adaptive filter reference signal (e.g., a far-end audio signal intended for input to a loudspeaker) is processed by a whitening filter before being input to an echo-cancelling adaptive filter. The whitening filter adjusts the reference signal characteristics so that all frequencies of the reference signal have similar power, and the whitening filter is dynamically adjusted to account for real-time changes in the spectra of the reference signal. Since the power spectrum of the whitened reference signal is approximately uniform, all filter coefficients of the echo-canceling adaptive filter converge more quickly and at a similar rate, and the echo-canceling filter performs as well as all frequencies of interest, irrespective of the mean power distribution of the original reference signal.

Description

METHODS AND APPARATUS FOR IMPROVING

ADAPTIVE FILTER PERFORMANCE BY

SIGNAL EQUALIZATION

Field of the Invention

The present invention relates to communications systems, and more particularly, to adaptive filtering in bi-directional communications systems.

Background of the Invention

Adaptive filtering arrangements are prevalent in communications systems of today. Such arrangements are typically used to reduce or remove unwanted signal components and/or to control or enhance signal components of interest. A common example of such a filtering arrangement relates to hands-free telephony, wherein the built-in earphone and microphone of a conventional telephone handset are replaced with an external loudspeaker and an external microphone, respectively, so that the telephone user can converse without having to physically hold the telephone unit in hand. Since sound emanating from the external loudspeaker can be picked up by the external microphone, adaptive filtering is commonly performed in order to prevent the loudspeaker output from echoing back and annoying the far-end user at the other end of the conversation. This type of adaptive filtering, or echo canceling, has become a basic feature of the full-duplex, hands-free communications devices of today. Typically, echo cancelation is achieved by passing the loudspeaker signal through an adaptive Finite Impulse Response (FIR) filter which approximates, or models, the acoustic echo path between the hands-free loudspeaker and the hands- free microphone (e.g., a passenger cabin in an automobile hands-free telephony application). The FIR filter thus provides an echo estimate which can be removed from the microphone output signal prior to transmission to the far-end user. The filtering characteristic (i.e. , the set of FIR coefficients or taps) of the adaptive FIR filter is dynamically and continuously adjusted, based on both the loudspeaker input and the echo-canceled microphone output, to provide a close approximation to the echo path and to track changes in the echo path (e.g. , when a near-end user of an automobile hands-free telephone shifts position within the passenger cabin).

Adjustment of the filtering characteristic is commonly achieved using a form of the well known Least Mean Square (LMS) adaptation algorithm developed by Widrow and Hoff in 1960. The LMS algorithm is a least square stochastic gradient step method which, as it is both efficient and robust, is often used in real- time applications. The LMS algorithm and its well known variations (e.g., the

Normalized LMS, or NLMS algorithm) do have certain drawbacks, however.

For example, the LMS and other known algorithms are dependent upon the frequency -domain power distribution of the signal being processed. In other words, certain of the FIR filter taps can converge faster than others, and the filter may perform less well at frequencies where the mean signal power is weak as compared to frequencies where the mean signal power is strong. This can be a particular problem in practical implementations where lower resolution fixed-point arithmetic is utilized to reduce computational cost and current consumption.

Although the above described problem can be avoided by resort to the well known Recursive Least Square (RLS) algorithm (see, for example, Simon Hay kin,

Adaptive Filter Theory, Third Edition, Prentice Hall Information and System Science Series, 1996, Chapter 13), the RLS algorithm is computationally complex and exceeds the limitations of the digital signal processing (DSP) components typically used today. Consequently, there is a need for improved methods and apparatus for providing echo cancelation in communications systems. Summary of the Invention

The present invention fulfills the above-described and other needs by providing echo canceling techniques wherein an adaptive filter reference signal (e.g. , a far-end audio signal intended for input to a loudspeaker) is processed by a whitening filter before being input to an echo-canceling adaptive filter. The whitening filter adjusts the reference signal characteristics so that all frequencies of the reference signal have similar power, and the whitening filter is dynamically adjusted to account for real-time changes in the spectra of the reference signal. Since the power spectrum of the whitened reference signal is approximately uniform, all filter coefficients of the echo-canceling adaptive filter converge more quickly and at a similar rate, and the echo-canceling filter performs as well at frequencies where the mean power of the original reference signal is weak as it does where the mean power of the original reference signal is strong.

An exemplary communications device according to the invention includes an adaptive echo canceler receiving a near-end audio signal and subtracting an echo estimate from the near-end audio signal to provide an echo-canceled near-end signal for transmission to a far-end user via a communications channel, the adaptive echo canceler filtering a reference signal to provide the echo estimate. The exemplary device further includes an adaptive whitening filter receiving a far- end audio signal from the far-end user via the communications channel and whitening the far-end audio signal to provide the reference signal to the adaptive echo canceler, a filtering characteristic of the adaptive whitening filter being dynamically adjusted in dependence upon the far-end audio signal.

The filtering characteristic of the adaptive whitening filter can be computed, for each frame in a succession of frames of samples of the far-end audio signal, as the inverse of the Fast Fourier Transform of the sample frame. Alternatively, the filtering characteristic of the adaptive whitening filter can be selected, for each frame of the far-end audio signal, from a number of predetermined filtering characteristics. For example, an auto-regressive estimator can be used to obtain an estimate of the spectral content of the far-end audio frame, and the predetermined filtering characteristic having the most similar spectral content can be selected for the frame. The exemplary communications device further includes a near-end whitening filter having a filtering characteristic which is updated to match the filtering characteristic of the adaptive whitening filter, the near-end whitening filter whitening the near-end audio signal before the echo estimate is subtracted. The exemplary communications device also includes an inverse whitening filter having a filtering characteristic which is updated to match an inverse of the filtering characteristic of the adaptive whitening filter, wherein the inverse whitening filter un-whitens the echo-canceled near-end signal prior to transmission to the far-end user via the communications channel.

An exemplary echo suppression method according to the invention includes the steps of filtering a reference signal to provide an echo estimate, subtracting the echo estimate from a near-end audio signal to provide an echo-canceled near-end signal for transmission to a far-end user via a communications channel, whitening a far-end audio signal received from the far-end user via the communications channel to provide the reference signal, and dynamically adjusting a filtering characteristic used in the whitening step.

A filtering characteristic used in the step of whitening the far-end audio signal can be computed, for each frame in a succession of frames of samples of the far-end audio signal, as the inverse of the Fast Fourier Transform of the sample frame. Alternatively, the filtering characteristic used in the step of whitening the far-end audio signal can be selected, for each frame in a succession of frames of samples of the far-end audio signal, from a number of predetermined filtering characteristics. For example, the spectral content of the far-end audio frame can be estimated using an auto-regressive process, and the predetermined filtering characteristic having spectral content most similar to that of the frame can be selected for the frame.

The exemplary method further includes the steps of whitening the near-end audio signal before the echo estimate is subtracted, and updating a filtering characteristic used in the step of whitening the near-end audio signal to match a filtering characteristic used in the step of whitening the far-end audio signal. The exemplary method also includes the steps of un-whitening the echo-canceled near- end signal prior to transmission to the far-end user, and updating a filtering characteristic used in the step of whitening the echo-canceled near-end signal to match an inverse of a filtering characteristic used in the step of whitening the far- end audio signal.

The above-described and other features and advantages of the invention are explained in detail hereinafter with reference to the illustrative examples shown in the accompanying drawings. Those of skill in the art will appreciate that the described embodiments are provided for purposes of illustration and understanding and that numerous equivalent embodiments are contemplated herein.

Brief Description of the Drawings

Figure 1 is a block diagram of an exemplary hands-free telephony system incorporating a conventional echo canceling arrangement.

Figure 2 is a block diagram of an exemplary hands-free telephony system incorporating an echo canceling arrangement according to the invention.

Detailed Description of the Invention Figure 1 depicts bi-directional communications device 100 including a conventional echo-canceling arrangement. As shown, the system 100 includes a microphone 110, a summing device 120, an adaptive filter 130, a filter computation processor 140, a speech encoder 150, a speech decoder 160, and a loudspeaker 170. Those of ordinary skill in the art will appreciate that the below described functionality of the components of Figure 1 can be implemented using a variety of known hardware configurations, including a general purpose digital computer, standard digital signal processing components, and/or one or more application-specific integrated circuits (ASICs). Those of ordinary skill will also appreciate that, in practice, the exemplary system 100 includes components (e.g. , an analog-to-digital converter at the output of the microphone 110 and a digital-to- analog converter at the input to the loudspeaker 170) which are omitted from Figure 1 , as they are not critical to an understanding of the present invention. In Figure 1, an audio output of the microphone 110 is coupled to an additive input of the summing device 120, and an output of the summing device 120 is coupled to an input of the speech encoder 150 and to an input of the filter computation processor 140. Additionally, an output of the speech decoder 160 is coupled to a second input of the filter computation processor 140 and to an input of each of the adaptive filter 130 and the loudspeaker 170. An output of the filter computation processor 140 is coupled to a control input of the adaptive filter 130, and an output of the adaptive filter 130 is coupled to a subtractive input of the summing device 120.

In operation, a coded far-end audio signal, including speech of a far-end user (not shown), is decoded via the decoder 160 and input to the loudspeaker 170 for presentation to a near-end user (also not shown). The loudspeaker output is then echoed back to the microphone 110 via an unknown and sometimes changing echo path, as is indicated by a variable transfer function H(z) in Figure 1. Thus, audio output from the microphone 110 includes loudspeaker echo, as well as near- end user speech and near-end background noise.

To prevent the loudspeaker echo from reaching and annoying the far-end user, the FIR filter 130 filters the loudspeaker signal to provide an estimate of the loudspeaker echo received at the microphone 110, and the resulting echo estimate is subtracted from the microphone output via the summing device 120. Echo- canceled output from the summing device 120 is then encoded via the encoder 150 and transmitted to the far-end user.

At the same time, the echo-canceled output is fed back to the filter computation processor 140 for use in adapting the filter coefficients, or taps, of the FIR filter 130 such that they converge toward and track the true echo path H(z). As is well known in the art, the filter computation processor 140 computes filter coefficient updates based on both the echo-canceled output, or error, signal and the loudspeaker input, or reference, signal (e.g. , using an LMS or NLMS algorithm). As is also well known in the art, coefficient updates can be computed either on a sample by sample basis or on a sample block by sample block basis.

Ideally, the arrangement of Figure 1 provides a quality echo estimate, and the far-end user is not distracted by delayed echoes of his or her own voice. However, as is described in the above Background of the Invention, the adaptive algorithm (i.e., the algorithm implemented in the filter computation processor

140) is dependent upon the frequency-domain power distribution of the loudspeaker signal. Thus, each of the adaptive filter taps can converge at a different rate, and the adaptive filter 130 can perform poorly at frequencies where the mean loudspeaker signal power is weak as compared to frequencies where the mean loudspeaker signal power is strong.

According to the present invention, however, the loudspeaker signal is whitened before being input to the echo-canceling adaptive filter 130. In other words, the characteristics of the loudspeaker signal are adjusted so that all frequencies have similar power. To account for real-time changes in the spectra of the loudspeaker signal, the whitening process itself (i.e. , the filtering characteristic of an adaptive whitening filter) is dynamically adjusted as described below. Since the power spectrum of the resulting whitened loudspeaker signal (i.e., the new echo canceler reference signal) is approximately uniform, all filter coefficients of the echo-canceling adaptive filter converge more quickly and at a similar rate. Consequently, the echo-canceling filter performs as well at frequencies where the mean power of the loudspeaker signal is weak as it does where the mean power of the loudspeaker signal is strong. Figure 2 depicts a bi-directional communications system 200 in which the above described aspects of the invention are incorporated. As shown, the exemplary system 200 includes a near-end whitening filter 210, an inverse whitening filter 220, a transfer function inverter 230, an adaptive whitening filter 240 and a whitening filter computation processor 250, as well as the microphone 110, the first summing device 120, the adaptive filter 130, the filter computation processor 140, the speech encoder 150, the speech decoder 160, and the loudspeaker 170 of Figure 1. As with Figure 1, those of ordinary skill in the art will appreciate that the below described functionality of the components of Figure 2 can be implemented using a variety of known hardware configurations, including a general purpose digital computer, standard digital signal processing components, and one or more ASICs. In practice, the exemplary system 200 includes components (e.g. , an analog-to-digital converter at the output of the microphone 110 and a digital-to-analog converter at the input to the loudspeaker 170) which are omitted from Figure 2, as they are not critical to an understanding of the present invention.

In Figure 2, audio output of the microphone 110 is coupled to an input of the whitening filter 210, and an output of the whitening filter 210 is coupled to the additive input of the first summing device 120. Additionally, output of the first summing device 120 is coupled to an input of the inverse whitening filter 220, and an output of the inverse whitening filter 220 is coupled to the input of the speech encoder 150 and to the first input of the filter computation processor 140.

The output of the speech decoder 160 is coupled to an input of each of the whitening filter 240 and the whitening filter computation processor 250, as well as to the second input of the filter computation processor 140 and the input of the loudspeaker 170. Also, first, second and third outputs of the adaptive whitening filter 240 are coupled to inputs of the near-end whitening filter 210, the adaptive filter 130 and the transfer function inverter 230, respectively, and an output of the whitening filter computation processor 250 is coupled to a control input of the adaptive whitening filter 240. Finally, output of the filter computation processor 140 is coupled to the control input of the adaptive filter 130, and the output of the adaptive filter 130 is coupled to the subtractive input of the summing device 120. Generally, operation of the system 200 of Figure 2 is similar to that of the system 100 of Figure 1. In other words, the FIR filter 130 and the filter computation processor 140 operate to provide an echo estimate which is subtracted from the microphone output signal to provide an echo-canceled near-end audio signal to the far-end user. Unlike the system 100 of Figure 1, however, the loudspeaker signal is not input directly to the echo-canceling adaptive filter 130. Instead, the adaptive whitening filter 240 processes the loudspeaker signal, using a filtering characteristic (i.e., a number of filter taps) provided by the whitening filter computation processor 250, to provide a whitened reference signal to the echo-canceling adaptive filter 130. The whitened reference signal is then filtered by the echo-canceling adaptive filter 130 to provide the echo estimate as before. Generally, the whitening process (i.e. , the operation of the adaptive whitening filter 240 and the whitening filter computation processor 250) results in a reference signal in which successive samples are uncorrelated and in which the power spectrum is approximately uniform across frequencies of interest. Consequently, the whitened reference signal enables the filter taps of the echo- canceling adaptive filter 130 to converge more quickly and at a uniform rate. For an introductory description of the principles of signal whitening, see for example Simon Haykin, Adaptive Filter Theory, Third Edition, Prentice Hall Information and System Science Series, 1996, page 268. According to an exemplary embodiment of the invention, the whitening filter computation processor 250 updates the filtering characteristic of the adaptive whitening filter 240 on a per-frame basis (i.e. , once for each of a succession of frames of samples of the loudspeaker signal). Specifically, a frequency -domain filtering characteristic W (i.e., a set of frequency-domain coefficients) is computed by inverting a Fast Fourier Transform (FFT) of the loudspeaker signal x as:

1 W =

FFT(x)

Thereafter, a frequency -domain representation of a frame of the whitened reference signal Xwhlle is computed by vector-multiplying the FFT of the loudspeaker frame by the resulting frequency -domain filtering characteristic Was:

X white = FFT(x) * W .

The frequency-domain version of the whitened reference signal Xwhlte is then converted to the time-domain, via an Inverse FFT (IFFT), to provide a frame of time-domain samples of the whitened reference signal xwhlte, as follows: x wh .ite = IFFT(X wh .ite ) .

The resulting time-domain frame xwhlle is then filtered by the echo-canceling adaptive filter 130 to provide the echo estimate. Since the whitening filter is continually adjusted to reflect the changing spectrum of the loudspeaker signal, the spectrum of the whitened reference signal remains substantially uniform from frame to frame.

In an alternative embodiment, computational complexity is reduced by not explicitly computing the adaptive whitening filter for each frame. Instead, a number of predetermined whitening filters are stored in memory, and an appropriate one of the fixed filters is selected at each frame. For example, the loudspeaker signal can be fed to an auto-regressive (AR) estimator to provide an estimate of the spectral content of the loudspeaker signal, and the pre-defined whitening filter having the most similar spectral characteristics can be selected. To compensate for the whitening of the echo estimate, the near-end whitening filter 210 whitens the microphone output signal to provide a near-end audio signal (i.e. , the signal coupled to the additive input of the summing device 120) which includes a whitened echo, as well as whitened near-end speech and whitened near-end background noise. To ensure that the filtering characteristic of the near-end whitening filter 210 is always identical to that of the adaptive whitening filter 240, the filter coefficients of the adaptive whitening filter 240 are copied to the near-end whitening filter once per sample frame.

The output of the summing device 120 is thus a whitened, echo-canceled signal. In other words, the whitened echo estimate is subtracted from a signal including whitened echo and whitened near-end speech and noise to provide a signal including whitened near-end speech and noise. Accordingly, the inverse whitening filter 220 un-whitens the near-end speech and noise before the signal is encoded and transmitted to the far-end user. To ensure that the filtering characteristic of the inverse whitening filter 220 is always equal to the inverse of that of the adaptive whitening filter 220, the filter coefficients of the adaptive whitening filter 220 are inverted (via the inversion processor 230) and copied to the inverse whitening filter 220 once per frame.

Generally, the invention provides methods and apparatus for improving echo canceler convergence speed and uniformity. By whitening an echo-canceling adaptive filter reference signal prior to input to the adaptive filter, all of the adaptive filter taps converge more quickly and at a similar rate. Consequently, the echo-canceling filter performs well for all frequencies of interest, irrespective of the mean power distribution of the original reference signal. Those skilled in the art will appreciate that the present invention is not limited to the specific exemplary embodiments which have been described herein for purposes of illustration and that numerous alternative embodiments are also contemplated. For example, although the exemplary embodiments have been described with respect to acoustic echo cancelation in the context of hands-free telephony, the disclosed adaptation-enhancement techniques are equally applicable to network echo cancelation (i.e. , where echoes result from impedance mismatches at a hybrid junction between a digital device and an analog network). The scope of the invention is therefore defined by the claims appended hereto, rather than the foregoing description, and all equivalents consistent with the meaning of the claims are intended to be embraced therein.

Claims

We claim:
1. A communications device for providing bi-directional audio communications between a near-end user and a far-end user via a bi-directional communications channel, comprising: an adaptive echo canceler receiving a near-end audio signal and subtracting an echo estimate from the near-end audio signal to provide an echo- canceled near-end signal for transmission to the far-end user via the communications channel, wherein said adaptive echo canceler filters a reference signal to provide the echo estimate; and an adaptive whitening filter receiving a far-end audio signal from the far-end user via the communications channel and whitening the far-end audio signal to provide the reference signal to said adaptive echo canceler, wherein a filtering characteristic of said adaptive whitening filter is dynamically adjusted in dependence upon the far-end audio signal.
2. A communications device according to claim 1, wherein the filtering characteristic of said adaptive whitening filter is computed, for each frame in a succession of frames of samples of the far-end audio signal, as the inverse of the Fast Fourier Transform of the sample frame.
3. A communications device according to claim 1 , wherein the filtering characteristic of said adaptive whitening filter is selected, for each frame in a succession of frames of samples of the far-end audio signal, from a number of predetermined filtering characteristics.
4. A communications device according to claim 3, further comprising an auto-regressive (AR) estimator providing an estimate of spectral content of the far-end audio signal, wherein a predetermined filter having spectral characteristics most similar to spectral characteristics of the far-end audio signal is selected as the adaptive whitening filter.
5. A communications device according to claim 1, further comprising a near-end whitening filter having a filtering characteristic which is updated to match the filtering characteristic of said adaptive whitening filter and whitening the near-end audio signal before the echo estimate is subtracted.
6. A communications device according to claim 1, further comprising an inverse whitening filter having a filtering characteristic which is updated to match an inverse of the filtering characteristic of said adaptive whitening filter and un-whitening the echo-canceled near-end signal prior to transmission to the far- end user via the communications channel.
7. A method for suppressing echo in a bi-directional communications device, the device being configured to provide two-way audio communications between a near-end user and a far-end user via a bi-directional communications channel, the method comprising the steps of: filtering a reference signal to provide an echo estimate; subtracting the echo estimate from a near-end audio signal to provide an echo-canceled near-end signal for transmission to the far-end user via the communications channel; whitening a far-end audio signal received from the far-end user via the communications channel to provide the reference signal; and dynamically adjusting a filtering characteristic used in said whitening step.
8. A method according to claim 7, wherein a filtering characteristic used in said step of whitening the far-end audio signal is computed, for each frame in a succession of frames of samples of the far-end audio signal, as the inverse of the Fast Fourier Transform of the sample frame.
9. A method according to claim 7, wherein a filtering characteristic of used in said step of whitening the far-end audio signal is selected, for each frame in a succession of frames of samples of the far-end audio signal, from a number of predetermined filtering characteristics.
10. A method according to claim 9, comprising the steps of auto- regressively estimating spectral content of the far-end audio signal, and selecting a predetermined filter having spectral characteristics most similar to spectral characteristics of the far-end audio signal for whitening the far-end audio signal.
11. A method according to claim 7, further comprising the step of whitening the near-end audio signal before the echo estimate is subtracted, wherein a filtering characteristic used in said step of whitening the near-end audio signal is updated to match a filtering characteristic used in said step of whitening the far-end audio signal.
12. A method according to claim 7, further comprising the step of un- whitening the echo-canceled near-end signal prior to transmission to the far-end user, wherein a filtering characteristic used in said step of whitening the echo- canceled near-end signal is updated to match an inverse of a filtering characteristic used in said step of whitening the far-end audio signal.
13. A handsfree telephone, comprising: a near-end microphone receiving sound from a near-end telephone environment and providing a near-end audio signal; a near-end whitening filter filtering the near-end audio signal to provide a whitened near-end audio signal; an adaptive echo canceler filtering a reference signal to provide an echo estimate and subtracting the echo estimate from the whitened near-end audio signal to provide a whitened echo-canceled signal; an inverse whitening filter filtering the whitened echo-canceled signal to provide an un-whitened echo-canceled signal for transmission; an adaptive whitening filter filtering a received far-end audio signal to provide the reference signal to said adaptive echo canceler; and a loudspeaker receiving the far-end audio signal and providing sound to the near-end environment, wherein a filtering characteristic of said adaptive whitening filter is dynamically adjusted in dependence upon the far-end audio signal, wherein a filtering characteristic of said near-end whitening filter is updated to match the filtering characteristic of said adaptive whitening filter, and wherein a filtering characteristic of said inverse whitening filter is updated to match an inverse of the filtering characteristic of said adaptive whitening filter.
14. A communications device according to claim 13, wherein the filtering characteristic of said adaptive whitening filter is computed, for each frame in a succession of frames of samples of the far-end audio signal, as the inverse of the Fast Fourier Transform of the sample frame.
15. A communications device according to claim 13, wherein the filtering characteristic of said adaptive whitening filter is selected, for each frame in a succession of frames of samples of the far-end audio signal, from a number of predetermined filtering characteristics.
PCT/EP2000/004411 1999-05-20 2000-05-16 Methods and apparatus for improving adaptive filter performance by signal equalization WO2000072566A1 (en)

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Cited By (1)

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Publication number Priority date Publication date Assignee Title
WO2002093774A1 (en) * 2001-05-17 2002-11-21 Stmicroelectronics Asia Pacific Pte Ltd Echo canceller and a method of cancelling echo

Citations (3)

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Publication number Priority date Publication date Assignee Title
WO1992012583A1 (en) * 1991-01-04 1992-07-23 Picturetel Corporation Adaptive acoustic echo canceller
EP0739102A2 (en) * 1995-04-20 1996-10-23 Nippon Telegraph And Telephone Corporation Subband echo cancellation method using projection algorithm
EP0758830A2 (en) * 1995-08-14 1997-02-19 Nippon Telegraph And Telephone Corporation Subband acoustic echo canceller

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1992012583A1 (en) * 1991-01-04 1992-07-23 Picturetel Corporation Adaptive acoustic echo canceller
EP0739102A2 (en) * 1995-04-20 1996-10-23 Nippon Telegraph And Telephone Corporation Subband echo cancellation method using projection algorithm
EP0758830A2 (en) * 1995-08-14 1997-02-19 Nippon Telegraph And Telephone Corporation Subband acoustic echo canceller

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* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2002093774A1 (en) * 2001-05-17 2002-11-21 Stmicroelectronics Asia Pacific Pte Ltd Echo canceller and a method of cancelling echo

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