WO1999026390A1

WO1999026390A1 - Echo canceller employing dual-h architecture having variable adaptive gain settings

Info

Publication number: WO1999026390A1
Application number: PCT/US1998/024352
Authority: WO
Inventors: Kenneth P. Laberteaux; Richard C. Younce; Bruce E. Dunne; David S. Farrell
Original assignee: Tellabs Operations, Inc.
Priority date: 1997-11-14
Filing date: 1998-11-13
Publication date: 1999-05-27
Also published as: US20060115077A1; AU1410699A; US6614907B1; US20040086108A1; CA2307657C; JP2001523921A; EP1040633A4; AU740467C; AU740467B2; US7200222B2; CA2307657A1; EP1040633A1; US6031908A

Abstract

An echo canceller circuit for use in an echo canceller system is set forth that provides sensitive double-talk detection. The echo canceller circuit comprises a second digital filter having adaptive tap coefficients to simulate an echo response occurring during the call. The adaptive tap coefficients of the second digital filter are updated over the duration of the call using a Least Mean Squares process having an adaptive gain a. A channel condition detector is used to detect channel conditions during the call. The channel condition detector is responsive to detected channel conditions for changing the adaptive gain a during the call. For example, the channel condition detector may detect the presence of a double-talk condition and set the adaptive gain a to zero. Similarly, the channel condition detector may detect the occurrence of a high background noise condition and set the adaptive gain a to a level less than 1 that is dependent on the detected level of the background noise. Other similar channel conditions and corresponding adaptive gain settings may likewise be utilized.

Description

TITLE OF THE INVENTION

ECHO CANCELLER EMPLOYING DUAL-H ARCHITECTURE

HAVING VARIABLE ADAPTIVE GAIN SETTINGS

CROSS-REFERENCE TO RELATED APPLICATIONS

The following applications, filed on even date, herewith, are incorporated by

reference: USSN , (Attorney Docket No. 11724US01), "Echo

Canceller Employing Dual-H Architecmre Having Improved Coefficient Transfer";

USSN , (Attorney Docket No. 11998US01), "Echo Canceller Employing

Dual-H Architecmre Having Improved Double-Talk Detection" ; USSN

, (Attorney Docket No. 11999US01), "Echo Canceller Employing

Dual-H Architecmre Having Improved Non-Linear Echo Path Detection" ; USSN

, (Attorney Docket No. 12001US01), "Echo Canceller Employing Dual-H

Architecmre Having Improved Non-Linear Processor"; USSN ,

(Attorney Docket No. 12002US01), "Echo Canceller Employing Dual-H Architecmre

Having Split Adaptive Gain Settings."

STATEMENT REGARDING FEDERALLY SPONSORED RESEARCH

OR DEVELOPMENT

Not Applicable BACKGROUND OF THE INVENTION

Long distance telephone facilities usually comprise four-wire transmission

circuits between switching offices in different local exchange areas, and two-wire

circuits within each area connecting individual subscribers with the switching office. A

call between subscribers in different exchange areas is carried over a two-wire circuit in

each of the areas and a four-wire circuit between the areas, with conversion of speech

energy between the two and four-wire circuits being effected by hybrid circuits.

Ideally, the hybrid circuit input ports perfectly match the impedances of the two and

four-wire circuits, and its balanced network impedance perfectly matches the impedance

of the two-wire circuit. In this manner, the signals transmitted from one exchange area

to the other will not be reflected or returned to the one area as echo. Unfortunately,

due to impedance differences which inherently exist between different two and four-

wire circuits, and because impedances must be matched at each frequency in the voice

band, it is virtually impossible for a given hybrid circuit to perfectly match the

impedances of any particular two and four-wire transmission circuit. Echo is,

therefore, characteristically part of a long distance telephone system.

Although undesirable, echo is tolerable in a telephone system so long as the time

delay in the echo path is relatively short, for example, shorter than about 40

milliseconds. However, longer echo delays can be distracting or utterly confusing to a

far end speaker, and to reduce the same to a tolerable level an echo canceller may be

used toward each end of the path to cancel echo which otherwise would remrn to the far

end speaker. As is known, echo cancellers monitor the signals on the receive channel of a four-wire circuit and generate estimates of the actual echoes expected to remrn

over the transmit channel. The echo estimates are then applied to a subtractor circuit in

the transmit channel to remove or at least reduce the actual echo.

In simplest form, generation of an echo estimate comprises obtaining individual

samples of the signal on the receive channel, convolving the samples with the impulse

response of the system and then subtracting, at the appropriate time, the resulting

products or echo estimates from the actual echo on the transmit channel. In actual

practice generation of an echo estimate is not nearly so straightforward.

Transmission circuits, except those which are purely resistive, exhibit an

impulse response that has amplitude and phase dispersive characteristics that are

frequency dependent, since phase shift and amplitude attenuation vary with frequency.

To this end, a suitable known technique for generating an echo estimate contemplates

manipulating representations of a plurality of samples of signals which cause the echo

and samples of impulse responses of the system through a convolution process to obtain

an echo estimate which reasonably represents the actual echo expected on the echo

path. One such system is illustrated in FIG.1.

In the system illustrated in FIG. 1, a far end signal x from a remote telephone

system is received locally at line 10. As a result of the previously noted imperfections

in the local system, a portion of the signal x is echoed back to the remote site at line 15

along with the signal v from the local telephone system. The echo response is

illustrated here as a signal s corresponding to the following equation: where h is the impulse response of the echo characteristics. As such, the signal sent

from the near end to the far end, absent echo cancellation, is the signal y, which is the

sum of the telephone signal v and the echo signal s. This signal is illustrated as y at line

15 of FIG. 1.

To reduce and/or eliminate the echo signal component s from the signal y, the

system of FIG. 1 uses an echo canceller having an impulse response filter h that is the

estimate of the impulse echo response h. As such, a further signal s representing an

estimate of echo signal s is generated by the echo canceller in accordance with the

following equation:

s = h * x

The echo canceller subtracts the echo estimate signal y from the signal y to

generate a signal e at line 20 that is returned to the far end telephone system. The

signal e thus corresponds to the following equation: e = s + v - s ∞ v

As such, the signal returned to the far end station is dominated by the signal v of the

near end telephone system. As the echo impulse response h more closely correlates to

the actual echo response h, then s-bar more closely approximates s and thus the

magnitude of the echo signal component 5 on the signal e is more substantially reduced.

The echo impulse response model h may be replaced by an adaptive digital

filter having an impulse response h . Generally, the tap coefficients for such an adaptive response filter are found using a technique known as Normalized Least Mean

Squares adaptation.

Although such an adaptive echo canceller architecmre provides the echo

canceller with the ability to readily adapt to changes in the echo path response h, it is

highly susceptible to generating sub-optimal echo cancellation responses in the presence

of "double talk" (a condition that occurs when both the speaker at the far end and the

speaker at the near end are speaking concurrently as determined from the viewpoint of

the echo canceller).

To reduce this sensitivity to double-talk conditions, it has been suggested to use

both a non-adaptive response and an adaptive response filter in a single echo canceller.

One such echo canceller is described in USPN 3,787,645, issued to Ochiai et al on

January 22, 1974. Such an echo canceller is now commonly referred to as a dual-H

echo canceller.

Although the dual-H echo canceller architecmre of the '645 patent provides

substantial improvements over the use of a single filter response architecmre, the '645

patent is deficient in many respects and lacks certain teachings for optimizing the use of

a such a dual-H architecmre in a practical echo canceller system. For example, the

present inventors have recognize that the adaptation gain used to adapt the tap

coefficients of the adaptive filter may need to be altered based on certain detected

conditions. These conditions include conditions such as double-talk, non-linear echo

response paths, high background noise conditions, etc.. The present inventors have recognized the problems associated with the foregoing dual-H architecmre and have

provided solutions to such conditions.

BRIEF SUMMARY OF THE INVENTION

An echo canceller circuit for the use in an echo canceller system is set forth that

provides sensitive double-talk detection. The echo canceller circuit comprises a second

digital filter having adaptive tap coefficients to simulate an echo response occurring

during the call. The adaptive tap coefficients of the second digital filter are updated

over the duration of the call using a Least Mean Squares process having an adaptive

gain a. A channel condition detector is used to detect channel conditions during the

call. The channel condition detector is responsive to detected channel conditions for

changing the adaptive gain a during the call. For example, the channel condition

detector may detect the presence of a double-talk condition and set the adaptive gain a

to zero. Similarly, the channel condition detector may detect the occurrence of a high

background noise condition and set the adaptive gain a to a level less than 1 that is

dependent on the detected level of the background noise. Other similar channel

conditions and corresponding adaptive gain settings may likewise be utilized.

BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWINGS

Figure 1 is a block diagram of a conventional canceller.

Figure 2 is a schematic block diagram of an echo canceller that operates in

accordance with one embodiment of the present invention.

Figure 3 is a flow chart illustrating one manner of carrying out coefficient

transfers wherein the transfer conditions may be used to implement double-talk

detection in accordance with one embodiment of the present invention.

Figure 4 is a flow chart illustrating a further manner of carrying out coefficient

wherein the transfer conditions may be used to implement the double-talk detection an

accordance with one embodiment of the present invention.

Figure 5 illustrates an exemplary solution surface for the adaptive filter whereby

the desired result is achieved at the solution matching the echo response of the channel.

Figure 6 illustrates one manner of checking for various echo canceller

conditions and responding to these conditions using a change in the adaptive gain

setting of the adaptive filter of the echo canceller.

Figure 7 illustrates one manner of implementing an echo canceller system employing the present invention.

SUBSTITUTE SHEET (RULE 25) DETAILED DESCRIPTION OF THE INVENTION

Figure 2 illustrates one embodiment of a dual-h echo canceller suitable for use

in implementing the present invention. As illustrated, the echo canceller, shown

generally at 25, includes both a non-adaptive filter h and an adaptive filter h to model

the echo response h. Each of the filters h and h are preferably implemented as digital

finite impulse response (FIR) filters comprising a plurality of taps each having a

corresponding tap coefficient. The duration of each of the FIR filters should be

sufficient to cover the duration of the echo response of the channel in which the echo

canceller 25 is disposed.

The output of the non-adaptive filter h is available at the line 30 while the

output of the adaptive filter h is available at line 35. Each of the signals at lines 30

and 35 are subtracted from the signal-plus-echo signal of line 40 to generate echo

compensated signals at lines 50 and 55, respectively. A switch 45, preferably a

software switch, may be used to selectively provide either the output signal at the line

50 or the output signal at line 55 to the echo canceller output at line 60.

A transfer controller 65 is used to transfer the tap coefficients of filter h to

replace the tap coefficients of filter h . As illustrated, the transfer controller 65 is

connected to receive a number of system input signals. Of particular import with

respect to the present invention, the transfer controller 65 receives the signal-plus-echo

response y and each of the echo canceller signals e and e at lines 50 and 55 , respectively. The transfer controller 65 is preferably implemented in the software of

one or more digital signal processors used to implement the echo canceller 25.

As noted above, the art is substantially deficient of teachings with respect to the

manner in which and conditions under which a transfer of tap coefficients from h to h

is to occur. The present inventors have implemented a new process and, as such, a

new echo canceller in which tap coefficient transfers are only made by the transfer controller 65 when selected criterion are met. The resulting echo canceller 25 has

substantial advantages with respect to reduced double-talk sensitivity and increased

double-talk detection capability. Further, it ensures a monotonic improvement in the

estimates h .

The foregoing system uses a parameter known as echo-return-loss-enhancement

(ERLE) to measure and keep track of system performance. Two ERLE parameter

values are used in the determination as to whether the transfer controller 65 transfers

the tap coefficients from h to h . The first parameter, E , is defined in the following

manner:

E - -^y e

Similarly, the parameter E is defined as follows:

E = - ^y Each of the values E and E may also be averaged over a predetermined number of

samples to arrive at averaged E and E values used in the system for the transfer

determinations.

Figure 3 illustrates one manner of implementing the echo canceller 25 using the

parameters E and E to control tap coefficients transfers between filter h to h . As

illustrated, the echo canceller 25 provides a default h set of coefficients at step 80

during the initial portions of the call. After the tap coefficients values for h have been

set, a measure of E is made at step 85 to measure the performance of the tap

coefficient values of filter h .

After the initialization sequence of steps 80 and 85, or concurrent therewith, the

echo canceller 25 begins and continues to adapt the coefficients of h to more

adequately match the echo response h of the overall system. As noted in Figure 3, this

operation occurs at step 90. Preferably, the adaptation is made using a Normalized

Least Mean Squares method, although other adaptive methods may also be used (e.g. ,

LMS and RLS).

After a period of time has elapsed, preferably, a predetermined minimum period

of time, the echo canceller 25 makes a measure of E at step 95. Preferably, this

measurement is an averaged measurement. At step 100, the echo canceller 25

compares the value of E with the value of E . If the value of E is greater than the

value of E , the tap coefficients of filter h are transferred to replace the tap coefficients

of filter h at step 105. If this criterion is not met, however, the echo canceller 25 will continue to adapt the coefficients of the adaptive filter h at step 90, periodically

measure the value of E at step 95, and make the comparison of step 100 until the

condition is met.

If the echo canceller 25 finds that E is greater than E . the above-noted transfer

takes place. Additionally, the echo canceller 25 stores the value of E as a value E_max.

This operation is depicted at step 110 of the Figure 3. The value of E_max is thus the

maximum value of ΕRLΕ that occurs over the duration of the call and at which a

transfer has taken place. This further value is used thereafter, in addition to the E and

E comparison, to control whether the tap coefficients of h are transferred by the

transfer controller 65 to replace the tap coefficients of h . This further process is

illustrated that steps 115, 120, and 125 of Figure 3. In each instance, the tap

coefficient transfer only occurs when both of the following two conditions are met: 1)

E is greater than the current , and 2) E is greater than any previous value of E used

during the course of the call. (E is greater than E_max). Each time that both criteria are

met, the transfer controller 65 of echo canceller 25 executes the tap coefficient transfer

and replaces the previous E_max value with the current E_max value for future comparison.

Requiring that E be greater than any E value used over the course of the call

before the coefficient transfer takes place has two beneficial and desirable effects.

First, each transfer is likely to replace the prior tap coefficients of filter h with a better

estimate of the echo path response. Second, this transfer requirement increases the

double-talk protection of the echo canceller system. Although it is possible to have positive ERLE E during double-talk, the probability that E is greater than E_max during

double-talk decreases as the value of E_max increases. Thus an undesirable coefficient

transfer during double-talk becomes increasingly unlikely as the value of E_max increases

throughout the duration of the call.

The echo canceller 25 may impose both an upper boundary and a lower

boundary on the value of E_max. For example, E_max may have a lower bounded value of 6

dB and an upper bounded value of 24 dB. The purpose of the lower bound is to

prevent normal transfers during double-talk conditions. It has been shown in

simulations using speech inputs that during double-talk, a value of greater than 6 dB

ERLE was a very low probability event. The upper bound on E_max is used to prevent a

spuriously high measurement from setting E_max to a value at which further transfers

become impossible.

The value of E_max should be set to, for example, the lower bound value at the

beginning of each call. Failure to do so will prevent tap coefficient transfers on a new

call until the echo cancellation response of the echo canceller 25 on the new call

surpasses the quality of the response existing at the end of the prior call. However, this

criterion may never be met during the subsequent call thereby causing the echo

canceller 25 to operate using sub-optimal tap coefficients values. Resetting the E_max

value to a lower value increases the likelihood that a tap coefficient transfer will take

place and, thereby, assists in ensuring that the h filter uses tap coefficients for echo

cancellation that more closely correspond to the echo path response of the new call. One manner of implementing the E_max value change is illustrated in the echo

canceller operations flow-chart of Figure 4. When all transfer conditions are met

except E greater than E_max, and this condition persists for a predetermined duration of

time, the echo canceller 25 will reset the E_max value to, for example, the lower bound

value. In the exemplary operations shown in Figure 4, the echo canceller 25

determines whether E is greater than the lower bound of E_max at step 140 and less than

the value of E_max at step 145. If both of these conditions remain true for a

predetermined period of time as determined at step 150, and all other transfer criterion

have been met, the echo canceller 25 resets the E_max value, at step 155. This lowering

of the E_max value increases the likelihood of a subsequent tap coefficient transfer.

Choosing values for the lower and upper bound of E_max other than 6 dB and 24

dB, respectively, is also possible in the present system. Choosing a lower bound of

E_max smaller than 6 dB provides for a relatively prompt tap coefficient transfer after a

reset operation or a new call, but sacrifices some double-talk protection. A value

greater than 6 dB, however, inhibits tap coefficient transfer for a longer period of time,

but increases the double-talk immunity of the echo canceller. Similarly, varying the

value of the predetermined wait time T before which E_max is reset may also be used to

tweak echo canceller performance. A shorter predetermined wait time T produces

faster reconvergence transfers, but may sacrifice some double-talk immunity. The

opposite is true for larger predetermined wait time values.

A further modification of the foregoing echo canceller system relates to the

value stored as E_max at the instant of tap coefficient transfer. Instead of setting E_max equal to the E value at the transfer instant, E_max may be set to a value equal to the value

of E minus a constant value (e.g. , one, three, or 6 dB). At no time, however, should

the E_max value be set to a value that is below the lower bound value for E_max.

Additionally, a further condition may be imposed in that a new softened E_max is not less

than the prior value of E,^. The foregoing "softening" of the E_max value increases the

number of transfers that occur and, further, provides more decision-making weight to

the condition of E being larger than E . Further details with respect to the operation

of the echo canceller coefficient transfer process are set forth and the co-pending patent

application title "ECHO CANCELLER HAVING THE IMPROVED TAP

COEFFICIENT TRANSFER", (Attorney Docket No. ) filed on even date herewith.

Preferably, the adaptive filter h uses a Normalized Least Mean Square (NLMS)

adaptation process to update its tap coefficients. In accordance with the process,

coefficients are adapted at each time n for each tap m = 0, 1,...,N - 1 in accordance

with the following equation:

where h„(m) is the m ^th tap of the echo canceller, x„ is the far-end signal at time n , e_n

is the adaptation error of time n , and a_n is the adaptation gain at time n.

The foregoing adaptation process will converge in the mean-square sense to the

correct solution the echo path response h if 0< a_n < 2. Fastest convergence occurs when a — 1. However, for 0 < a < 1, the speed of convergence to h is traded-off

against steady-state performance.

Figure 5 is provided to conceptualize the effect of the adaptation gain on the

filter response. The graph or Figure 5 includes an error performance surface 185

defined to be the mean square error between h and h, to be a V dimensional bowl.

Each point in the bowl corresponds to the mean-square error for each corresponding h

(of length N). The bottom of the bowl is the h which produces the least mean-square

error, i.e. , h. The ΝLMS process alternatively moves the h towards h at the bottom of

the performance surface as shown by arrow 190. When a = 1, h moves to the

bottom of the bowl most quickly, but one the bottom is reached, the adaptation process

continues to bounce h around the true h bottom of the bowl, i.e. ,

but h ≠ h .

If a small a is used, then the steady-state error is smaller (h will remain closer to h),

but h requires a longer time to descend to the bottom of the bowl, as each step is

smaller.

In some cases, as the present inventors have recognized, the performance

surface will temporarily change. In such situations, it becomes desirable to suppress

the h from following these changes. This presents a challenge to choose the best a for

each scenario.

Figure 6 illustrates operation of the echo canceller 25 in response to various

detected scenarios. It will be recognized that the sequence of detecting the various

conditions that is set forth in Figure 6 is merely illustrative and may be significantly varied. Further, it will be recognized that the detection and response to each scenario

may be performed concurrently with other echo canceller processes. Still further, it

will be recognized that certain detected scenarios and their corresponding responses

may be omitted.

In the embodiment of Figure 6, the echo canceller 25 entertains whether or not a

double-talk condition exists at step 200. Double talk, as noted above, is defined as the

situation when both far-end and near-end talkers speak at the same time during a call. In such a scenario, the adaptive error signal is so severely corrupted by the near-end

speaker that it is rendered useless. As such, if a double-talk condition is detected, the

echo canceller 25 responds by freezing the adaptation process at step 205, i.e. , set a =

0, until the double talk ceases.

There are several methods that the echo canceller 25 can use for detecting a

double-talk condition. One is to compare the power of the near-end signal to the far-

end signal. If the near-end power comes close enough to the far-end power ("close

enough" can be determined by the system designer, e.g. , within 0 or 6 or lOdB), then

double talk can be declared. Another method is to compare the point-by -point

magnitudes of the near-end and far-end signals. This search can compare the current

[ C[ with the current |_y| , the current |x| with the last several \y\ , the current |>>| with the

last several |x| , etc. In each case, the max |x| and \y\ over the searched regions are

compared. If

max|_y|

> Double Talk Threshold max x where max |x| indicates the maximum |x| over the search region (\y\ is similarly

defined), then a double-talk condition is declared.

A still further manner of detecting a double-talk condition is set forth in

U.S.S.N. , entitled (Attorney Docket No. ), the teachings of which are

hereby incorporated by reference. As set forth in that patent application, a double-talk

condition is declared based on certain motored filter performance parameters.

It may be possible to further condition the double-talk declaration with other

measurements. For example, the current Echo Remrn Loss (ERL) may be used to set

the Double Talk Threshold noted above herein. The short-term power of either the far-

end, the near-end, or both, may also be monitored to ensure that they are larger than

some absolute threshold (e.g. , -50dBm or -40dBm). In this manner, a double-talk

condition is not needlessly declared when neither end is speaking.

Once a double-talk condition is declared, it may be desirable to maintain the

double-talk declaration for a set period of time after the double-talk condition is met.

Examples might be 32, 64, or 96 msec. After the double-talk condition ceases to exist,

the adaptive gain value may be returned to the value that existed prior to the detection

of the double-talk condition, or to a predetermined remrn value.

At step 210, the echo canceller 25 determines whether a high background noise

condition is present. A low level of constant background noise can enter from the near-

end, for example, if the near-end caller is in an automobile or an airport. Its effects are

in some ways similar to that of double-talk, as the near-end double-talk corrupts the

adaptive error signal. The difference is that, unlike double talk, near-end background noise is frequently constant, thus setting a = 0 until the noise ends is not particularly

advantageous. Also background noise is usually of lower power than double-talk. As

such, it corrupts the adaptation process but does not render the resulting adaptation

coefficients unusable.

As illustrated at step 215, it is desirable to choose a gain 0 < a < 1, i.e. lower

the gain from its fastest value of 1 when a high background noise condition is present.

While this will slow the adaptation time, the steady state performance increases since

the effects of noise-induced perturbations will be reduced. In other words, the tap

variance noise is reduced by lowering the adaptation gain a.

Preferably, the background noise is measured as a long-term measurement of

the power of when the far-end is silent. As this measurement increases, a decreases.

One schedule for setting the adaptive gain a as a function of background noise level is

set forth below.

Background Noise (dBm) a

> -48 .125

> -54 > -48 .25

> -60 > -54 .5

< -60

It will be readily recognized that there are other schedules that would work as well, the

foregoing schedule being illustrative.

A further condition in which the adaptive gain may be altered from an otherwise

usual gain value occurs when the adaptive filter h is confronted with a far-end signal

that is narrow band, i.e. comprised of a few sinusoids. In such a scenario, there are an

infinite number of equally optimal solutions that the LMS adaptation scheme can find.

Thus it is quite unlikely that the resulting cancellation solution h will properly identify

(i.e. mirror) the channel echo response h. Such a situation is referred to as under-

exciting the channel, in that the signal only provides information about the channel

response at a few frequencies. The echo canceller 25 attempts to determine the

existence of this condition that step 220.

Consider a situation where the far-end signal varies between periods in which a

narrow band signal is transmitted and wide band signal is transmitted. During the wide

band signal periods, the h filter should adapt to reflect the impulse response of the

channel. However, when the narrow band signal transmission period begins, the h filter will readapt to focus on canceling the echo path distortion only at the frequencies

present in the narrow band signal. Optimizing a solution at just a few frequencies is

likely to give a different solution than was found during transmission of the wide band

signal. As a result, any worthwhile adaptation channel information gained during wide

band transmission periods is lost and the h filter requires another period of adaptation

once the wide band signal returns.

When the far-end signal is narrow band, the adaptation can and should be

slowed considerably, which should discourage the tendency of the coefficients to

diverge. Specifically, when a narrow band signal is detected, a may be upper-bounded

by either 0.25 or .125. This operation is illustrated at step 225.

Narrow band signal detection may be implemented using a fourth order

predictive filter. Preferably, this filter is implemented in software executed by one or

more digital signal processors used in the echo canceller system 25. If it is able to

achieve a prediction gain of at least 3 to 6 dB (user defined) over the h filter, then it is

assumed that the received signal is a narrow band signal.

An amplimde threshold for the far-end signal is also preferably employed in

determining the existence of a narrow band signal. If the far-end power is greater than

-40 dBm, the current far-end sample is sent to the fourth order predictive filter, which

determines whether or not the far-end signal is narrow band. If the far-end power is

less than -40 dBm, the predictive filter is re-initialized to zero.

A further scenario in which it is desirable to alter the gain of the adaptive filter

h is when the echo path response is non-linear. The presence of non-linearities in the echo path encourages constant minor changes in the coefficients h in order to find

short-term optimal cancellation solutions. The detection of non-linearity of the echo

path response preferably proceeds in the manner set forth in U.S.S.N. , titled

, filed on even date herewith. The presence of a non-linear echo path

is determined that step 230.

In a non-linear echo path scenario, it is desirable to choose the adaptive gain

constant a large enough that h can track these short-term best solutions. However,

choosing a = 1 may be suboptimal in most non-linear scenarios. This is due to the

fact that the gain is too large and, thus, short-term solutions are "overshot" by the

aggressive adaptation effort. Accordingly, as shown at step 235, choosing a gain lower

than 1 is preferable. Choosing a = 0.25 was found to be the best trade off between

tracking and overshooting short term optimal solutions. The gain constant a may be further reduced if large background noise is measured, as discussed above.

A still further scenario in which the adaptive gain may be varied relates to the

convergence period of the adaptive filter h . As noted above, a large gain constant a is

desired during convergence periods while a smaller a is desired in steady state

conditions after the filter has converged. In other words, there seems little lost and

perhaps some potential gain to reduce a after an initial period of convergence is

completed. This appears to be especially valuable if the long-term performance is

found to be substandard.

In view of the foregoing, the echo canceller 25 may implement a reduced gain

mode in which an upper bound for the gain constant a is set at a lower value than 1 (e.g. , at either .25 or .125). This mode is detected at step 240 and is entered at step

245 if the ERLE remains below a predetermined threshold value (e.g. , either 6dB or

3dB) after a predetermined period of adaptation. The adaptation time is preferably

selected as a value between 100 to 300 msec. This amount of time will generally

prevent the echo canceller 25 from entering the reduced gain mode during convergence

periods.

As will be readily recognized, the echo canceller of the present invention may

be implemented in a wide range of manners. Preferably, the echo canceller system is

implemented using one or more digital signal processors to carry out the filter and

transfer operations. Digital-to-analog conversions of various signals are carried out in

accordance with known techniques for use by the digital signal processors.

Figure 7, illustrates one embodiment of an echo canceller system, shown

generally at 700, that maybe used to cancel echoes in multi-channel communication

transmissions. As illustrated, the system 700 includes an input 705 that is connected to

receive a multi-channel communications data, such as a Tl transmission. A central

controller 710 deinterleaves the various channels of the transmission and provides them

to respective convolution processors 715 over a data bus 720. It is within the

convolution processors 715 that a majority of the foregoing operations take place. Each

convolution processor 715 is designed to process at least one channel of the

transmission at line 730. After each convolution processor 715 has processed its

respective channel(s), the resulting data is placed on the data bus 720. The central

controller 710 multiplexes the data into the proper multichannel format (e.g., Tl) for retransmission at line 735. User interface 740 is provided to set various user

programmable parameters of the system.

Numerous modifications may be made to the foregoing system without departing

from the basic teachings thereof. Although the present invention has been described in

substantial detail with reference to one or more specific embodiments, those of skill in

the art will recognize that changes may be made thereto without departing from the

scope and spirit of the invention as set forth in the appended claims.

Claims

1. An echo canceller circuit comprising:

a digital filter having adaptive tap coefficients to simulate an echo response

occurring during the call, the adaptive tap coefficients being updated

during the call using a Least Mean Squares process having an adaptive

gain a; and

channel condition detection means responsive to detected channel

conditions for changing the adaptive gain a during the call.

2. An echo canceller circuit as claimed in claim 1 wherein the channel condition

detection means is responsive to a double-talk channel condition and sets the

adaptive gain a equal to zero in response thereto.

3. An echo canceller circuit as claimed in claim 1 wherein the channel condition

detection means is responsive to a high background noise channel condition and

lowers the adaptive gain a in response thereto.

4. An echo canceller circuit as claimed in claim 3 wherein the channel condition

detection means is responsive to a high background noise channel condition and

lowers the adaptive gain a to a value that is dependent on a level of the

background noise that is detected.

5. An echo canceller circuit as claimed in claim 1 wherein the channel condition

detection means is responsive to a narrow band signal condition and lowers the

adaptive gain a in response thereto.

6. An echo canceller circuit as claimed in claim 5 wherein the adaptive gain a is

set equal to about or less than 0.25.

7. An echo canceller circuit as claimed in claim 5 wherein the adaptive gain a is

set equal to about or less than 0.125.

8. An echo canceller circuit as claimed in claim 1 wherein the channel condition

detection means is responsive to a non-linear echo path condition and lowers the

adaptive gain a in response thereto.

9. An echo canceller circuit as claimed in claim 8 wherein the adaptive gain a is

set equal to about or less than 0.25.

10. An echo canceller circuit as claimed in claim 1 wherein the channel condition

detection means is responsive to convergence of the adaptive filter and lowers

the adaptive gain a in response thereto.

11. An echo canceller circuit as claimed in claim 10 wherein the adaptive gain a is

set equal to or less than 0.25.

12. An echo canceller circuit as claimed in claim 10 wherein the adaptive gain a is

set equal to about or less than 0.125.

13. An echo canceller comprising:

at least one input for receiving a far-end signal of a call;

at least one input for receiving a signal-plus-echo signal of the call, the

signal-plus-echo signal having a signal component corresponding to

an echo response of a transmission medium carrying the call;

a first digital filter receiving the far-end signal and having non-adaptive tap

coefficients to simulate the echo response;

a summer circuit for subtracting the filtered far-end output signal of the first

digital filter from the signal-plus-echo signal to generate an echo

compensated signal for transmission to a far-end;

a second digital filter receiving the far-end signal and having adaptive tap

coefficients to simulate the echo response, the adaptive tap

coefficients being updated during the call using a Least Mean

Squares process having an adaptive gain a ;

a coefficient transfer controller disposed to transfer the adaptive tap

coefficient of the second digital filter to replace the tap coefficients of the first digital filter when a set of one or more conditions is met;

and

a channel condition detector responsive to detected channel conditions for

changing the adaptive gain a during the call.

14. An echo canceller as claimed in claim 13 wherein the coefficient transfer

controller transfers the adaptive tap coefficients of the second digital filter to

replace the tap coefficients of the first digital filter when E is greater than E

and, concurrently, E is greater than E_m╬▒x, wherein E corresponds to the ratio

between a signal-plus-echo signal and a first echo compensated signal using the

first digital filter, E corresponds to the ratio between a signal-plus-echo signal

and a second echo compensated signal using second digital filter, and E_max

corresponds to the largest E occurring over a call during which a transfer has

occurred.

15. An echo canceller as claimed in claim 13 wherein the channel condition

detection means is responsive to a double-talk channel condition and sets the

adaptive gain a equal to zero in response thereto.

16. An echo canceller as claimed in claim 13 wherein the channel condition

detection means is responsive to a high background noise channel condition and

lowers the adaptive gain a in response thereto.

17. An echo canceller as claimed in claim 13 wherein the channel condition

detection means is responsive to a high background noise channel condition and

lowers the adaptive gain a to a value that is dependent on a level of the

background noise that is detected.

18. An echo canceller as claimed in claim 13 wherein the channel condition

detection means is responsive to a narrow band signal condition and lowers the

adaptive gain a in response thereto.

19. An echo canceller as claimed in claim 18 wherein the adaptive gain a is set

equal to about or less than 0.25.

20. An echo canceller as claimed in claim 18 wherein the adaptive gain a is set

equal to about or less than 0.125.

21. An echo canceller as claimed in claim 13 wherein the channel condition

adaptive gain a in response thereto.

22. An echo canceller as claimed in claim 21 wherein the adaptive gain a is set

equal to about or less than 0.25.

23. An echo canceller as claimed in claim 13 wherein the channel condition

detection means is responsive to convergence of the adaptive filter and lowers

the adaptive gain a in response thereto.

24. An echo canceller as claimed in claim 23 wherein the adaptive gain a is set

equal to or less than 0.25.

25. An echo canceller as claimed in claim 23 wherein the adaptive gain a is set

equal to about or less than 0.125.