Abstract:
A method and associated apparatus for controlling an acoustic canceler (“AEC”) are disclosed. Prior to passing audio signals to the AEC, a distortion detector is used to determine if the signals are distorted. If so, the AEC does not adapt is filter coefficients to the distorted signals. This technique improves the AEC&#39;s ability to adapt its filter coefficients to subsequent undistorted signals. For example, near-end or far-end audio signals above a predetermined threshold value are detected by a distortion detector which disables adaptive filter control logic so that distorted signals do not result in generation of erroneous filter coefficients.

Description:
BACKGROUND 
     1. Field of the Invention 
     The present invention generally relates to telecommunication systems and more particularly to acoustic echo cancellation. 
     2. Description of the Related Art 
     In telecommunication systems (e.g. telephony and video conferencing systems), audio signals such as a user&#39;s voice are transmitted to a loudspeaker in a remote location using a microphone. A microphone and a loudspeaker are provided at each location for sending and receiving audio signals. Acoustic coupling occurs in these systems whenever a microphone is placed where it can pick up the sounds from a loudspeaker in the same location. In which case, the user&#39;s voice is transmitted to a loudspeaker in a remote location, picked up by a microphone in the remote location, and re-transmitted back to the loudspeaker in the user&#39;s location, thereby resulting in the user hearing back his own voice (i.e. an echo). Acoustic coupling between microphones and loudspeakers is difficult to eliminate because microphones and loudspeakers are typically located in the same general area, such as in a conference room or a large hall. 
     Acoustic Echo Cancelers(“AEC”) have been developed to eliminate echoes caused by acoustic coupling. AECs, in general, are well known; see for example: U.S. Pat. No. 4,965,822 (Williams); H. Yasukawa et al., “Acoustic Echo Canceler with High Speech Quality,” Institute of Electrical and Electronic Engineers (“IEEE”) CH2396-0/87/0000-2125 (1987); A. Gilloire, “Experiments With Sub-Band Acoustic Echo Cancelers For Teleconferencing,” IEEE CH2396-0/87/0000-2141 (1987); J. Chen et al., “A New Structure For Sub-Band Acoustic Echo Canceler,” IEEE CH2561-9/88/0000-2574 (1988); and C. Breining et al., “Acoustic Echo Control: An Application of Very-High Order Adaptive Filters,” IEEE Signal Processing Magazine 1053-5888/99, pp. 42-69 (1999). All of the aforementioned references are incorporated herein by reference. 
     A typical AEC uses an adaptive filter to generate an echo estimate signal that is subtracted from the microphone&#39;s output signal. If the echo estimate signal matches the echo embedded in the microphone&#39;s output signal, the echo is canceled out (i.e. removed). The accuracy of the generated echo estimate signal is dependent on the AEC&#39;s capability to change the coefficients of its adaptive filter to adapt to the echo. Thus, improving the capability of AECs to adapt to echoes is highly desirable. 
     SUMMARY OF THE INVENTION 
     The present invention relates to a method and associated apparatus for controlling an AEC. In one embodiment, distortion detectors are used to determine if an audio signal is distorted. If so, the AEC is prevented from incorrectly adapting its filter coefficients to the distorted audio signal. This allows the AEC to maintain a valid set of filter coefficients that can be rapidly adapted to cancel subsequent echoes. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 shows a block diagram of an audio transmission system in the prior art. 
     FIG. 2 shows a block diagram of an audio transmission system in accordance with an embodiment of the present invention. 
     FIG. 3 shows a block diagram of an audio transmission system in accordance with another embodiment of the present invention. 
    
    
     DETAILED DESCRIPTION 
     The present invention relates to a method and associated apparatus for controlling an acoustic echo canceler (“AEC”). The invention can be used with a variety of AECs including full-band, sub-band, finite impulse response (“FIR”), and infinite impulse response (“IIR”) AECs. 
     FIG. 1 shows a block diagram of an audio transmission system  100  in the prior art. A digital audio signal from a remote location (also referred to as the “far-end”) is converted into a format that is compatible with system  100  by decoder  125 . The resulting digital audio signal x(n) (also referred to as the “far-end signal”) is converted to an analog signal by a digital-to-analog converter DAC  110 , amplified by an amplifier  111 , and then converted into sound by a loudspeaker  112 . If a microphone  113  and loudspeaker  112  are placed in the same general area, far-end signal x(n) can be picked up by microphone  113  from loudspeaker  112 , amplified by an amplifier  114 , converted to a digital signal by an analog-to-digital converter ADC  115 , converted to a format that is compatible with an audio transmission system in the far-end by coder  126 , and then heard back in the far-end as an echo. The location of microphone  113  and loudspeaker  112  is hereinafter referred to as the “near-end.” 
     In system  100 , an AEC  120  is employed to prevent echoes from being transmitted to the far-end. A model  121  includes an adaptive filter, such as a finite impulse response (“FIR”) filter, for generating an echo estimate signal s(n). The use of adaptive filters in AECs is well known. Echoes are canceled by subtracting echo estimate signal s(n) from the output signal of microphone  113 . For example, when the user in the far-end is talking while the user in the near-end is listening, AEC  120  will detect that there is audio activity in the far-end. A microphone signal y(n), the digitized output of microphone  113 , will contain some amount of the far-end audio due to acoustic coupling. Consequently, an adder  122  subtracts echo estimate signal s(n) from microphone signal y(n). If echo estimate signal s(n) accurately characterizes microphone signal y(n), a resulting error signal e(n) of adder  122  is zero. Otherwise, error signal e(n) will have some residual value that will be heard in the far-end as an echo. Error signal e(n) is fed back to an input port  124  of model  121  to provide an indication of how well the echo is canceled. Model  121  changes its adaptive filter&#39;s coefficients to adapt to the echo based on, among other criteria, samples of far-end signal x(n) taken at an input port  123  and samples of error signal e(n) taken at input port  124 . 
     Generation of an accurate echo estimate signal s(n) depends on how correctly model  121  adapts its filter coefficients to the echo. As implemented in audio transmission system  100 , model  121  adapts its filter coefficients even if far-end signal x(n) or microphone signal y(n) is distorted. For example, loud noises, such as door slams or noises created by physically moving a microphone, can saturate ADC  115  (i.e. require ADC  115  to operate beyond its allowable range of values) and result in a distorted microphone signal y(n). Because model  121  generates echo estimate signal s(n) based on the assumption that microphone signal y(n) resembles far-end signal x(n), a distorted microphone signal y(n) not only results in a large error signal e(n) that is heard in the far-end, but also causes model  121  to incorrectly adapt its filter coefficients. Models with incorrectly adapted filter coefficients take a long time to re-train for subsequent echoes. Thus, it will take several iterations of echo cancellation before model  121  generates an accurate echo estimate signal s(n), thereby degrading the capability of AEC  120  to cancel echoes for a period of time. Also, if error signal e(n) is large enough, signals (not shown) in the far-end will saturate, further distorting error signal e(n), which is propagated back to the near-end. This echo cycle will continue until the loudspeaker volume or the microphone sensitivity at either the far-end or near-end is reduced. 
     FIG. 2 shows a block diagram of an audio transmission system  200  in accordance with an embodiment of the invention. Except for the addition of a loudspeaker distortion detector  201 , a microphone distortion detector  202 , and an adapt control logic  203 , transmission system  200  is essentially identical to transmission system  100 . Loudspeaker distortion detector  201  determines whether far-end signal x(n) will be distorted upon being converted by DAC  110 , amplified by amplifier  111 , and vocalized by loudspeaker  112 . This determination can be based on known performance characteristics of DAC  110 , amplifier  111 , and loudspeaker  112 . For example, if far-end signal x(n) has a decimal value of +21,000 (on a full scale range of +32,767 to −32,768) and it is known that far-end signals x(n) having a decimal value greater than +20,000 or less than −20,000 will be distorted, loudspeaker distortion detector  201  generates a logical HIGH signal on an output port  205  to indicate to an adapt control logic  203  that far-end signal x(n) will be distorted. Microphone distortion detector  202  determines whether microphone signal y(n) is a distorted representation of the sound pressure picked up by microphone  113 . Such distortion can be due to the limited numerical range of ADC  115  or limitations in the performance characteristics of amplifier  114  and microphone  113 . For example, if ADC  115  is saturated, microphone distortion detector  202  generates a logical HIGH signal on an output port  206  to inform adapt control logic  203  that microphone signal y(n) is distorted. A sequence of large sample values of microphone signal y(n) also indicates that microphone signal y(n) is distorted and, accordingly, causes microphone distortion detector  202  to generate a logical HIGH signal. 
     Adapt control logic  203  includes an output port  204  for controlling filter coefficient adaptation at model  121 . If adapt control logic  203  detects a logical HIGH signal on output port  205  or on output port  206 , a logical HIGH signal is generated on output port  204  to indicate to model  121  that either far-end signal x(n) or microphone signal y(n) is distorted and, therefore, not suitable for adaptation. Accordingly, model  121  does not adapt and retains its existing set of filter coefficients. As a result, model  121  retains a valid set of filter coefficients that can be more rapidly adapted to cancel subsequent echoes. In the case where both output ports  205  and  206  are at a logical LOW, adapt control logic  203  generates a logical LOW signal on output port  204  to indicate that there is no distortion. Model  121  then uses well known adaptation rules, such as not adapting when there is audio activity in both the far-end and the near-end, in determining whether to adapt its filter coefficients. In other words, the signal on output port  204  of control logic  203  can be used as a gate or a condition precedent to conventional adaptation rules. 
     While the above embodiment of the invention is described using two distortion detectors, the invention is not so limited. For example, the invention can be used with a loudspeaker distortion detector  201  but without a microphone distortion detector  202 , and vice versa. This configuration simplifies implementation and may be adequate for some applications. Further, the output of loudspeaker distortion detector  201  or microphone distortion detector  202  can be directly connected to model  121  without going to a separate adapt control logic  203 . In which case, model  121  adapts its filter coefficients based on information directly received from loudspeaker distortion detector  201  or microphone distortion detector  202 . 
     In one embodiment, the invention is implemented in computer software. Far-end signal x(n) and microphone signal y(n) are sampled values in digital form stored in memory locations. Distortion detection is performed by comparing the values of far-end signal x(n) and microphone signal y(n) to memory locations containing threshold values that are indicative of saturation. Adapt control logic  203  is a logical OR function that sets a DISTORTION flag (i.e. a bit in a memory location) when a saturation threshold value is exceeded. Conventional digital signal processing techniques can also be used to detect distortion. The DISTORTION flag is taken into consideration by model  121 , which can be implemented in software or in digital signal processing (“DSP”) circuits, in determining whether to adapt its filter coefficients. Model  121  does not adapt when the DISTORTION flag is set and follows conventional adaptation rules when the DISTORTION flag is reset. Adder  122  is a summation function that generates error signal e(n) by summing microphone signal y(n) with echo estimate signal s(n) generated by model  121 . If echo estimate signal s(n) is not provided in negative form, adder  122  takes the negative of echo estimate signal s(n) before summing it with microphone signal y(n). Distortion detectors, for example, can also be logic comparators that compare microphone signal y(n) to a threshold value. Coder  126  and decoder  125  can be implemented in computer software or by using an integrated circuit (“IC”). 
     FIG. 3 shows a block diagram of an audio transmission system  300  in accordance with an embodiment of the invention. Except for the addition of a gain logic  301  and a gain control logic  302 , transmission system  300  is essentially identical to transmission system  200 . Gain logic  301  changes the magnitude of error signal e(n) by a gain value (i.e. a multiplication factor) received from gain control logic  302 . When adapt control logic  203  generates a logical HIGH signal, indicating that a distorted audio signal has been detected, gain control logic  302  lowers the gain of amplifier  301  to attenuate error signal e(n). This prevents residual echoes from propagating and also prevents further distortion of error signal e(n). Gain control logic  302  sets the gain of gain logic  301  based on the past history of audio signal levels, the maximum and minimum gain settings for a particular brand and model of microphone  113 , and the optimum rate of changing the gain (based on previous experiments, for example). 
     The description of the invention given above is provided for purposes of illustration and is not intended to be limiting. Numerous variations are possible within the scope of the invention. The invention is set forth in the following claims.