JP3403473B2 - Stereo echo canceller - Google Patents

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a stereo echo canceller for canceling echo of stereo channels in a teleconference system or the like.

[0002]

2. Description of the Related Art In recent years, a teleconference system has been put into practical use, and attempts have been made to further improve the sense of presence by converting a voice into a stereo signal for transmission and diffusion. When performing stereo loudspeaking, a monaural acoustic echo canceller cannot sufficiently cancel echoes, so that a stereo echo canceller is desired to be put into practical use. To meet such a demand, a stereo echo canceller in which adaptive filters are linearly combined has been proposed. For example, B.Widro
w, et al .: Adaptive noise cancelling: Principle and
applications, Proc. IEEE, 63, 12, pp1692-1716, (1
975).

The above-mentioned stereo echo canceller will be described with reference to the drawings. FIG. 5 is a block diagram showing the overall configuration of a conventional stereo echo canceller.
In the figure, the input terminals 1 and 2 are input portions for audio signals output from first (left) and second (right) microphones installed in an acoustic space at a remote location. When the subscript j is a sample number or a sampling point, the first reception signal x1 _j transmitted to the input end 1 is power-amplified and given to the output end 4, and the second reception signal transmitted to the input end 2. x2 _j is power-amplified and given to the output terminal 3. First and second speakers are connected to the output terminals 3 and 4, respectively. The first and second speakers are the left and right speakers installed in the acoustic space at this point. In addition, microphones (not shown) are installed on the left and right of the acoustic space at this point. The respective audio signals input to these microphones are the first transmission input y.
1 _j and the second transmission input y2 _j are given to the input terminals 5 and 6. The output terminals 7 and 8 are output units for transmitting the local audio signals to a remote location as a first transmission output e1 _j and a second transmission output e2 _j , respectively.

The input terminal 1 has a first adaptive filter 71,
The third adaptive filter 73 is connected, and the input terminal 2 is connected to the second adaptive filter 72 and the fourth adaptive filter 74. The adaptive filter 71 is a filter for adaptively estimating the acoustic impulse response from the speaker at the output end 3 to the microphone at the input end 5, and the adaptive filter 72 is an acoustic filter from the speaker at the output end 4 to the microphone at the input end 5. It is a filter that adaptively estimates the impulse response. Similarly, the adaptive filter 73 is a filter that adaptively estimates the acoustic impulse response from the speaker at the output end 3 to the microphone at the input end 6, and the adaptive filter 74
Is a filter for adaptively estimating the acoustic impulse response from the speaker at the output end 4 to the microphone at the input end 6.

The first subtraction means 75 includes an adaptive filter 71.
Echo replica yh11 _j generated by and the adaptive filter 7
Send echo replica yh12 _j generated in 2 and input y
In the circuit for subtracting from 1 _j , the subtraction result is output from the output end 7 as a transmission output e1 _j . The second subtraction means 76
Echo replica yh21 _j generated by the adaptive filter 73
And the echo replica yh22 generated by the adaptive filter 74
_This is a circuit for subtracting _j and _j from the transmission input y2 _j , and the subtraction result is output from the output end 8 as a transmission output e2 _j .

FIG. 6 is a block diagram showing a concrete configuration of the adaptive filters 71 and 72 and the subtracting means 75. 6, the input portion of the first X register 81 is the input terminal 1 of FIG.
It is connected to the. The X register 81 samples the received signal x1 _j at 8 kHz, for example (all the registers below shall sample the input signal at 8 kHz).
Received signal sequence X of past N samples at sample point j
It is a register for storing 1 _j = {x1 _j , x1 _j-1 , ..., X1 _{j-N + 1} }.

The input part of the second X register 82 is connected to the input end 2 of FIG. The X register 82 receives the received signal x
2 _j is sampled at 8 kHz, and received signal sequence X2 _j = {x2 _j , x2 of past N samples at sample point j
_j-1 , ..., X2 _{j-N + 1} }. The first H register 83 estimates the impulse response H11 _j = {h11 _0j , h11 _1j , ..., H11 of the echo path from the speaker at the output end 3 to the microphone at the input end 5.
_N-1j } is a register for storing _N-1j }. Second H register 8
4 is an estimated value H12 _j = of the impulse response of the echo path from the speaker at the output end 4 to the microphone at the input end 5
This is a register for storing {h12 _0j , h12 _1j , ..., _{H12 N-1j} }.

The output of the X register 81 and the output of the H register 83 are given to the first product-sum means 85. Accumulating means 85
Is N of the X register 81 and the H register 83 for each sample.
Echo replica yh11
_This is a circuit that synthesizes _j . The output of the X register 82 and H
The output of the register 84 is given to the second sum-of-products means 86. The sum-of-products means 86 receives the X register 82 and the H
This is a circuit for performing a convolution operation on N pieces of data in the register 84 to synthesize an echo replica yh12 _j .

[0009] subtraction means 87 corresponds to a subtracting means 75 of FIG. 5, the circuit subtracting each synthesized echo replica yh11 _j, yh12 _j from the transmission input y1 _j in the product-sum means 85 and 86, the transmission input y1 _j The transmission output e1 _{j in} which the included echo is canceled is output. First received power calculation means 8
8 is the received signal sequence X1 stored in the X register 81
_j of square sum P1 _j (hereinafter, referred to as a first received power) is a circuit for calculating the. The second received power calculating means 89 is a circuit for calculating the sum of squares P2 _j (hereinafter, referred to as second received power) of the received signal series X2 _j stored in the X register 82.

The outputs of the reception power calculation means 88 and 89 are given to the update gain calculation means 810. Update gain calculation means 8
Reference numeral 10 is a circuit for outputting the update gain σ1 _j to the first and second updating means 811 and 812 by dividing the constant α (0 <α ≦ 1) by the sum of the received power P1 _j and the received power P2 _j. is there. The first updating means 811 multiplies the output signal e1 _j of the subtracting means 87 by the updating gain δ1 _j , and the multiplication result e1 _j ·
σ1 _{j is set} to N pieces of data {x1 _j , x1 in the X register 81.
_j-1 , ..., X1 _{j-N + 1} }. The calculation result here is added to the H register 83, and the estimated value H11 _{j of} the impulse response previously held in the H register 83.
Will be updated.

The second updating means 812 multiplies the output signal e1 _j of the subtracting means 87 by the updating gain σ1 _j , and the multiplication result e
1 _j · σ1 _{j is set} to N data of the X register 82 {x
2 _j , x2 _j-1 , ..., X2 _{j-N + 1} }. The calculation result here is added to the H register 84,
The impulse response estimate H12 _j previously held in the H register 84 is updated.

The configurations of the adaptive filters 73 and 74 and the subtracting means 76 shown in FIG. 5 are the same as those in FIG. Hereinafter, the estimated value of the impulse response stored in the H register of the adaptive filter 73 is H21 _j, and the estimated value of the impulse response stored in the H register of the adaptive filter 74 is H22 _j . Further, the echo replicas synthesized by the adaptive filters 73 and 74 are yh21 _j and yh22 _j, and the second transmission output is e.
2 _j .

The operation of the conventional stereo echo canceller configured as described above will be described using mathematical expressions. The updating means 811 of FIG. 6 inputs the received signal sequence X1 _j from the X register 81 and estimates a new impulse response by the normalized stereo echo canceller algorithm shown by the following equation (1).

[Equation 1] Estimated value H11 _j = of the impulse response obtained by the equation (1)
{ _{H11 0j} , h11 _1j , ..., h11 _N-1j } is set in the H register 8
Store in 3.

Next, the sum of products means 85 calculates the echo replica yh11 _j using the following equation (2).

[Equation 2]

On the other hand, the reception power calculation means 88 and 89 calculate the first reception power P1 _j and the second reception power P2 _j , respectively, using the following equations (3) and (4).

[Equation 3]

[Equation 4] These values are input to the update gain calculation means 810, and the update gain σ1 _j is calculated using the following equation (5).

[Equation 5] The echo replica yh11 _j thus generated by the adaptive filter 71 of FIG. 5 is output to the subtracting means 75.

Similarly, the adaptive filter 72 calculates each coefficient of the impulse response H12 _j of the echo path from the speaker at the output end 4 to the microphone at the input end 5 using the following equation (6).

[Equation 6] Then, the echo replica yh12 _j is generated using the following equation (7).

[Equation 7]

Next, the subtracting means 75 is adapted to the adaptive filters 71 and 7.
The echo replica generated in 2 is converted into the transmission output e1 _j using the following equation (8).

[Equation 8]

When the first reception input is given and a sufficient time elapses, the sum of the echo replicas synthesized by the adaptive filters 71 and 72 becomes substantially equal to the transmission input y1 _j . Therefore, the echo component of the transmission output e1 _j becomes almost zero, and only the voice signal of the speaker at this point is transmitted to the listener at the remote point via the output end 7.

Similarly for the second transmission output e2 _j ,
It is generated by the adaptive filters 73 and 75 and the subtraction means 76. That is, signal processing is performed using the following equations (9) to (14).

[Equation 9]

[Equation 10]

[Equation 11]

[Equation 12]

[Equation 13]

[Equation 14]

[0020]

[SUMMARY OF THE INVENTION However, the adaptive filter 71, 72 as mean square of the transmission output e1 _j is minimized, adaptive filter 73, 74 is a transmission output e2 _j
Since the filter coefficients of the adaptive filters are updated so that the root mean square of each is minimized, when there is a correlation between the received signal x1 _j and the received signal x2 _j , each adaptive filter converges on the impulse response to be estimated. There are times when you don't. For example, the impulse response sequence of the echo path from the speaker at the output end 3 to the microphone of the input end 5 is G11, and the impulse response sequence of the echo path from the speaker at the output end 4 to the microphone at the input end 5 is G12. Further, the impulse response sequence of the echo path from the speaker at the output end 3 to the microphone at the input end 6 is set to G21, and the output end 4
Let G22 be the impulse response sequence of the echo path from the speaker to the microphone of the input terminal 6.

Next, the adaptive filters 71, 72, 73, 74
The impulse response sequences estimated by H11, H12, and H2, respectively.
If H1 and H22 are set, H11 = G11, H12 = G12, H21 = G21, H22 = G22 if the adaptation is sufficiently advanced.
The impulse response of the adaptive filter should converge so that However, if the received signal x1 _j and the received signal x2 _j are exactly the same, that is, if the correlation is very strong, then H11
= H12 = (G11 + G12) / 2, H21 = H22 = (G21 + G
Even if it converges to 22) / 2, the echo components of the transmission outputs e1 _j and e2 _j can both be made zero.

As described above, when there is a correlation between the first received signal and the second received signal, each adaptive filter may erroneously converge on an impulse response different from the original one.
When the degree of correlation between received signals changes midway due to a change or movement of the speaker in a state where the impulse response is converged to such an incorrect impulse response, the impulse response of the adaptive filter is further updated from that point. Therefore, an unnecessary echo is generated for a while, and there is a problem that the residual echo included in the transmission output increases.

The present invention has been made in view of the above-mentioned conventional problems, and realizes a stereo echo canceller which suppresses an increase in residual echo when a speaker is changed.

[0024]

According to the invention of claim 1 of the present application, one acoustic space in which a microphone and a speaker are respectively provided at a first position and a second position, and one acoustic space are separated from each other. Stereo audio information is transmitted and received using a transmission signal and a reception signal to and from the other acoustic space which is located only at the third position and has a microphone and a speaker respectively at the third position and the fourth position. A stereo echo canceller used in a teleconference system, comprising:
The first reception signal output from the microphone at the position is adaptively has a tap coefficient equivalent to the acoustic impulse response when the first reception signal is input to the microphone at the first position via the speaker at the first position. The adaptive filter of No. 1 and the second received signal output from the microphone of the fourth position are equivalent to the acoustic impulse response when the second received signal is input to the microphone of the first position via the speaker of the second position. A second adaptive filter adaptively having tap coefficients,
A tap coefficient equivalent to the acoustic impulse response when the first received signal output from the microphone at the third position is input to the microphone at the second position via the speaker at the first position is adaptively set. A third adaptive filter that is provided, and an acoustic impulse response when the second received signal output from the microphone at the fourth position is input to the microphone at the second position via the speaker at the second position. Echo replicas generated by the first and second adaptive filters are subtracted from the fourth adaptive filter having an equivalent tap coefficient adaptively and the voice signal of the microphone at the first position to cancel the echo component. The first subtraction means for generating the first transmitted output and the echo signals generated by the third and fourth adaptive filters from the voice signal of the microphone at the second position. The second subtraction means for subtracting the respective Rica to generate the second transmission output in which the echo component is canceled and the first and second received signals are input, and the change of the speaker in the other acoustic space is detected. And a loss amount control means for generating control signals for increasing the loss of the first and second transmission outputs, respectively, when the speaker change is detected by the speaker change detecting means. A first loss means and a second loss means for applying a loss to the outputs of the first and second subtraction means for a certain period of time based on the control signal of the control means and outputting an audio signal to the other acoustic space. It is a feature.

In the invention of claim 2 of the present application, the speaker change detecting means includes first and second average power calculating means for calculating the average power of the first and second received signals, respectively. Two
Using the output of the average power calculation means, the division means for calculating the division value β, and the division value β of the division means and two predetermined constants β1 and β2 (β1>1> β2) are compared in magnitude relation. The present invention is characterized by including a magnitude relationship comparing means for comparing and a status change detecting means for detecting an output change of the magnitude relationship comparing means and determining a speaker change if there is an output change.

The invention of claim 3 of the present application is characterized in that logarithmic operation means for logarithmically converting the value of the division value β is provided between the division means and the magnitude relationship comparison means.

According to a fourth aspect of the present invention, the first transmission input and the first transmission input from the microphone at the first position are provided.
From the first input / output ratio calculation means for calculating the ratio of the first transmission output to the average signal power output from the subtraction means and outputting the result to the loss amount control means, and the microphone at the second position. A second input / output for calculating a ratio between the second transmission input that is input and the average signal power of the second transmission output that is output from the second subtraction means, and outputting the result to the loss amount control means. Ratio calculating means is provided, and the loss amount controlling means outputs a control signal for increasing the loss amount of the first loss means when the power ratio calculated by the first input / output ratio calculating means during the speaker change is small. However, when the power ratio calculated by the second input / output ratio calculation means is small, a control signal for increasing the loss amount of the second loss means is output.

[0028]

According to the inventions of claims 1 to 3 having the above characteristics, the first adaptive filter estimates the impulse response of the echo path from the first reception output to the first transmission input, Combine the echo replicas. The second adaptive filter estimates the impulse response of the echo path from the second reception output to the first transmission input, and synthesizes the echo replica. Then, the echo component contained in the first transmission input is canceled by subtracting the echo replica synthesized by the first and second adaptive filters from the first transmission input by the first subtraction means. Similarly, the third adaptive filter estimates the impulse response of the echo path from the second reception output to the first transmission input, and combines the echo replicas. The fourth adaptive filter estimates the impulse response of the echo path from the second reception output to the second transmission input and synthesizes the echo replica thereof. Then, the second subtraction means subtracts the echo replicas synthesized by the third and fourth adaptive filters from the second transmission input, thereby canceling the echo component included in the second transmission output. Further, the speaker change detecting means detects the speaker change by observing the first and second received signals, and the loss amount control means detects the loss of the first and second loss means for a certain period during the speaker change. Control to increase. Then, by adding a loss to the transmission output, it is possible to prevent an increase in residual echo that occurs when the speaker changes.

Further, according to the invention of claim 4 of the present application, the echo removal amount in the transmission output can be known by calculating the ratio of the average signal power of the transmission input and the transmission output by the input / output comparison means. When the average signal power ratio of the input calculation comparison means is input, the loss amount control means reduces the loss amount of the loss means when the echo removal amount is large. When the echo removal amount is small, the echo component loss amount is increased. In this way, in addition to the above-mentioned effects, even if the transmission input contains a signal that should not be lost as in the case of double talk, the loss of that signal is minimized and the residual echo at the time of speaker change is suppressed. You can

[0030]

DESCRIPTION OF THE PREFERRED EMBODIMENTS A stereo echo canceller according to a first embodiment of the present invention will be described with reference to FIG. Figure 1
FIG. 3 is a block diagram showing the overall configuration of the stereo echo canceller of the first embodiment. The same parts as those in the conventional example are designated by the same reference numerals and detailed description thereof will be omitted. In the figure, input terminals 5 and 6 are input terminals for inputting audio signals of microphones respectively provided at a first position (left) and a second position (right) in one (medium) acoustic space. The output terminals 3 and 4 are the output terminals of the audio signal given to the speakers provided in the first and second positions of the one acoustic space, respectively. Similarly, the input terminals 1 and 2 are input terminals for inputting the audio signals of the microphones respectively provided at the third position (left) and the fourth position (right) in the acoustic space of the other (remote point). The output terminals 7 and 8 are the output terminals of the audio signal given to the respective speakers provided at the third and fourth positions in the one acoustic space.
And similarly to the conventional example, the first to fourth adaptive filters 11 to 11
14, first and second subtraction means 15 and 16 are provided respectively.

Unlike the prior art, in this embodiment, the input terminals 1 and 2 are
A speaker change detecting means 17 is connected between the subtracting means 15,
A first loss means 18 and a second loss means 1 are provided at the output parts of the sixteen.
9 are connected respectively. The speaker change detecting means 17 inputs the received signals x1 _j and x2 _j, and from the change of the signal level,
This is a circuit that detects the change of speakers at a remote location. The output of the speaker change detecting means 17 is given to the loss amount controlling means 110. The loss amount control means 110 is a circuit that generates a loss amount control signal based on the output of the speaker change detection means 17.
The loss means 18 and 19 are circuits that attenuate the transmission outputs e1 _j and e2 _j , respectively, based on the control signal of the loss amount control means 110.

The operation of the stereo echo canceller of this embodiment having the above configuration will be described. When the speaker at the remote location is located at the center of the left and right microphones in the acoustic space of the destination, the correlation between the received signals x1 _j and x2 _j is large. In this case, the filter coefficient of each of the adaptive filters 11 to 14 may be updated by mistake. After this state, if the correlation between the received signals x1 _j and x2 _j changes due to the change of the speaker, the echo replica is not correctly synthesized by the adaptive filter, and the echo increases for a while in the transmission output. Therefore, by observing the received signals x1 _j and x2 _j ,
The speaker change detecting means 17 detects the change of the speaker, and when the speaker change is detected, the loss amount controlling means 110 causes the loss means 18,
Increase the loss amount of 19. Then, the transmission outputs e1 _j and e2 _j are suppressed when the speaker is changed, and an increase in echo immediately after the change can be suppressed. As described above, according to the present embodiment, it is possible to prevent an increase in echoes when the speaker is changed by adding a loss to the transmission output when the speaker is changed.

Next, a stereo echo canceller according to the second embodiment of the present invention will be described with reference to FIG. The present embodiment embodies the speaker change detecting means of the first embodiment, and other configurations are the same as those in FIG. FIG. 2 is a block diagram showing the configuration of the speaker change detecting means of this embodiment.

In FIG. 2, first and second input terminals 1 and 2 are provided.
The average power calculation means 21 and 22 are connected respectively. The average power calculating means 21 and 22 are circuits for calculating the average power of the received signals x1 _j and x2 _j , respectively, and the result is the dividing means 23.
Given to. The division means 23 divides the output of the average power calculation means 21 by the output of the average power calculation means 22 and outputs the divided value β to the magnitude relationship comparison means 24. The magnitude comparison means 24 uses the divided value β and two predetermined constants β 1 and β.
This is a circuit for comparing with 2 (β1>1> β2) to determine in which range the divided value β falls. This determination result is given to the state change detecting means 25. The state change detecting means 25 checks whether or not the range of the division value β has changed, and outputs a speaker change detection signal, assuming that the speaker has changed, when the range β has changed.

The operation of the stereo echo canceller having the speaker change detecting means as described above will be described. For example,
The first reception input side is defined as the left, the second reception input side is defined as the right,
It is assumed that each speaker is located on either the left side, the center, or the right side. Then, β1 and β2 are set in advance so that β>β1> 1 when the left speaker speaks and 1>β2> β when the right speaker speaks. Then, first, when the left speaker speaks, the first average power P1 _j
Becomes larger than the second average power P2 _j , the output β of the dividing means 23 becomes larger than β1.

When the speaker in the center utters, β =
1 and β1>β> β2 holds. Similarly, when the speaker on the right speaks, β <β2. In this way, the relationship between β and β 1 and β 2 is determined by the position of the speaker, and the change of speaker can be detected by detecting the boundary of the change in the state of β by the state change detecting means 25.

As described above, according to this embodiment, the change of the speaker is detected by observing the change of the division value β of the average powers of the received signals x1 _j and x2 _j . Then, in this case, by increasing the respective loss amounts of the loss means 18 and 19 of FIG. 1, it is possible to prevent an increase in echoes at the time of speaker change.

Next, a stereo echo canceller according to the third embodiment of the present invention will be described with reference to FIG. FIG. 3 is a block diagram showing the configuration of the speaker change detecting means of the third embodiment. In this embodiment, a logarithmic calculation means 31 is added to the speaker change detection means of the second embodiment. Other configurations are the same as those in FIGS. 1 and 2, and the same parts as those in the second embodiment are designated by the same reference numerals and the description thereof will be omitted.

In FIG. 3, the division value β of the division means 23 is output to the logarithmic calculation means 31. The division value β is the received signal x1
_Since it is represented by the ratio of the average power of _{j to} the average power of the received signal x2 _j , β is concentrated near 1 when there are multiple speakers near the center. In this case, a difference in the value of β is less likely to appear than when the speaker is located at the end, and it is difficult to determine the speaker change. For this reason, if the logarithmic calculation means 31 is provided after the division means 23, even if the value of β changes slightly around 1, the reference value γ of the logarithmic value γ (= log β).
Changes to ₁ (= log β1) and γ ₂ (= log β2) can be detected sensitively. Therefore, even when a plurality of speakers are located near the center, it is easy to detect the speaker change. As described above, according to the present embodiment, even when the speaker is located near the center, it is possible to detect the speaker change, and it is possible to prevent an increase in echoes during the speaker change.

Next, a stereo echo canceller according to the fourth embodiment of the present invention will be described with reference to FIG. FIG. 4 is a block diagram showing the overall configuration of the stereo echo canceller of this embodiment. In this embodiment, a first input / output ratio calculation means 41 and a second input / output ratio calculation means 42 are added to the circuit of the stereo echo canceller of the first embodiment, and the output thereof is input to the loss amount control means 111. That's what I did. The other structure is the same as that of FIG. 1, and the same parts as those of the first embodiment are designated by the same reference numerals and their description is omitted.

In the circuit as shown in FIG. 1 or 7, by calculating the ratio of the average signal power between the first transmission input y1 _j and the first transmission output e1 _j , the amount of echo cancellation at the transmission output can be known. . Therefore, as shown in FIG. 4, the input / output ratio calculating means 41 is provided, and the input signal and the output signal of the subtracting means 15 are respectively input to calculate the average signal power ratio. When the average signal power ratio of the input / output ratio calculation means 41 is input to the loss amount control means 111, when the echo removal amount is large, the loss means 1
The loss gain of 8 is reduced, and the loss gain of the echo component is increased when the amount of echo removal is small. In this way, the loss amount control means 1 is set so that the loss amount of the transmission output e1 _j is optimized.
At 11, the loss amount is controlled. By this operation, even when a signal that should not be lost is included in the transmission input as in the case of double talk, the loss of the signal can be minimized and the residual echo at the time of speaker change can be suppressed.

The input / output ratio calculation means 42 also operates in the same manner as the input / output ratio calculation means 41. As described above, according to the present embodiment, by detecting the echo removal amount by the input / output ratio calculation means 41, 42, even if the normal input signals are mixed in the transmission inputs y1 _j , y2 _j , the loss of the signals is eliminated. It can be minimized, and the increase in residual echo when the speaker changes can be suppressed.

[0043]

As described above, according to the first to fourth aspects of the invention, by providing the speaker change detecting means,
And the change of the speaker in the other acoustic space can be detected from the second received signal. In this case, by giving a loss to the transmission output of one acoustic space, it is possible to suppress the residual echo generated when the speaker changes.

[Brief description of drawings]

FIG. 1 is a block diagram showing an overall configuration of a stereo echo canceller according to a first embodiment of the present invention.

FIG. 2 is a block diagram showing a configuration of speaker change detecting means used in a stereo echo canceller in a second embodiment of the present invention.

FIG. 3 is a block diagram showing a configuration of speaker change detecting means used in a stereo echo canceller in a third embodiment of the present invention.

FIG. 4 is a block diagram showing an overall configuration of a stereo echo canceller according to a fourth embodiment of the present invention.

FIG. 5 is a block diagram showing a configuration example of a conventional stereo echo canceller.

FIG. 6 is a block diagram showing a specific configuration example of an adaptive filter used in a conventional stereo echo canceller.

[Explanation of symbols]

1, 2, 5, 6 input end 3, 4, 7, 8 output end 11-14, 71-74 Adaptive filter 15, 16, 75, 76, 87 Subtracting means 17 Speaker change detection means 18,19 Loss means 21,22 Average power calculation means 23 Division means 24 Large / small relationship comparison means 25 State change detection means 31 Logarithmic calculation means 41,42 Input / output ratio calculation means 81,82 X register 83, 84 H register 85,86 Sum of products means 87 subtraction means 88,89 Received power calculation means 110,111 Loss amount control means 810 Update gain calculation means 811 and 812 update means

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Claims (4)

(57) [Claims] 1. An acoustic space in which a microphone and a speaker are provided at a first position and a second position, respectively,
Stereo using a transmission signal and a reception signal between the acoustic space on the other side, which is located apart from the acoustic space on the one side and in which the microphone and the speaker are respectively provided at the third position and the fourth position. A stereo echo canceller used in a teleconference system for transmitting and receiving audio information of, wherein a first reception signal output from a microphone at a third position is transmitted through a speaker at the first position A first adaptive filter that adaptively has a tap coefficient equivalent to an acoustic impulse response when input to a microphone, and a second received signal output from a microphone at a fourth position is a speaker at a second position. A second adaptive filter adaptively having a tap coefficient equivalent to the acoustic impulse response when input to the microphone at the first position via A third adaptively having a tap coefficient equivalent to the acoustic impulse response when the first received signal output from the microphone at the position is input to the microphone at the second position via the speaker at the first position. Adaptive filter and the second received signal output from the microphone at the fourth position is tapped equivalent to the acoustic impulse response when the second received signal is input to the microphone at the second position via the speaker at the second position. From the voice signal of the microphone at the first position and the fourth adaptive filter that adaptively has a coefficient,
And the first replicating means for respectively subtracting the echo replicas generated by the second adaptive filter to generate the first transmission output in which the echo component is canceled, and the voice signal of the microphone at the second position, Third
And second subtraction means for respectively subtracting the echo replicas generated by the fourth adaptive filter to generate the second transmission output in which the echo component is canceled, and the first and second reception signals are input, Speaker change detection means for detecting the change of the speaker in the other acoustic space, and a control signal for increasing the loss of the first and second transmission outputs when the change of the speaker is detected by the speaker change detecting means. Based on a control signal of the loss amount control means, the first and second loss amount control means for generating
A first and second loss means for applying a loss to the output of the subtraction means for a certain period of time and outputting a voice signal to the other acoustic space. 2. The speaker change detecting means, first and second average power calculating means for calculating average power of the first and second received signals, respectively, and the first and second average power. Division means for calculating the division value β using the output of the calculation means, the division value β of the division means, and two constants β1 determined in advance,
a magnitude relation comparing means for comparing magnitude relations with β2 (β1>1>β2); a state change detecting means for detecting an output change of the magnitude relation comparing means, and determining a speaker change if there is an output change; The stereo echo canceller according to claim 1, further comprising: 3. The stereo echo canceller according to claim 2, wherein a logarithmic calculation means for logarithmically converting the value of the division value β is provided between the division means and the magnitude relationship comparison means. 4. A ratio between an average signal power of a first transmission input input from the microphone at the first position and a first transmission output output from the first subtraction means is calculated,
A first input / output ratio calculation means that outputs the result to the loss amount control means, a second transmission input that is input from the microphone at the second position, and a second output that is output from the second subtraction means. Second input / output ratio calculation means for calculating a ratio of the transmission output of the device to the average signal power and outputting the result to the loss amount control means,
The loss amount control means outputs a control signal for increasing the loss amount of the first loss means when the power ratio calculated by the first input / output ratio calculation means at the time of speaker change is small,
2. The stereo echo canceller according to claim 1, wherein when the power ratio calculated by the second input / output ratio calculation means is small, a control signal for increasing the loss amount of the second loss means is output.