JPH07226697A - Echo canceler, echo canceling method and transmitter/ receiver having echo canceler - Google Patents

Abstract

PURPOSE:To rightly estimate a necessary acoustic coupling amount and to delete an echo signal by judging whether an echo signal exists in a voice signal or not according to the convergence state of an echo canceling part. CONSTITUTION:The canceling means of an echo canceling part 200 forms a pseudo echo path 210 from the transmission signal x (n) to an echo path and an echo signal y (n) and inputs the signal x (n) and the estimated value of the impulse response of the echo path from an estimating means 220 to the echo path 210. A pseudo echo signal y' (n) is subtracted from the signal y (n), the signal y (n) is canceled, and the error signal e (n) and an input signal z (n) are outputted. The loss amount determination means 310 of a loss control part 300 judges whether a signal except a transmitting signal s (n) is mixed in a signal z (n) or not by using the signals e (n) and z (n) and a receiving signal x (n), and has a means deciding as a double talk state when the signal is mixed. A means 310 selects a method estimating the acoustic coupling amount by an acoustic transmission characteristic based on the result of the deciding means.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an echo canceller, a echo canceller, and a handset having an echo canceller, and more particularly, in a two-wire to four-wire conversion system and a loudspeaker communication system which is reproduced by a loudspeaker. The present invention relates to an echo canceller that cancels or suppresses echo signals that cause howling and interfere with hearing, an echo canceling method, and a handset having the echo canceller.

[0002]

2. Description of the Related Art FIG. 10 shows a system configuration for a voice call. The system shown in FIG.
3, the receiving speaker 2, 4, the transmitting signal amplifiers 5, 7, the receiving signal amplifiers 6, 8 and the transmission line 9, and the speaker 10
And the receiver 11 transmits and receives a voice signal.

The transmitted voice, which is generated by the transmitter 10 from the transmitting microphone 1, is amplified by a transmitting signal amplifier 5,
It is transmitted to the receiving side via the transmission line 9. Received signal amplifier 8
Amplifies the received signal and inputs it to the receiving speaker 4,
Is played.

Unlike the conventional telephone call system, the above-mentioned loudspeaker call system does not require a handset to be held in the hand, so that it is possible to make a call while working, or to realize a natural face-to-face call. Widely used in teleconferences, videophones, and public telephones.

On the other hand, a problem with this voice communication system is that the voice transmitted from the transmitting side to the receiving side is re-input from the receiving side microphone 3 and reverberates to the transmitting side. For the speaker 10, this reverberation phenomenon is called an acoustic echo or the like because the voice uttered by the speaker 10 is reproduced from the speaker 2, and causes adverse effects such as a call disturbance and discomfort. Furthermore, the sound reproduced from the speaker 2 is the microphone 1
Is received by and forms a closed loop of the signal. When the loop gain is larger than 1, the howling phenomenon occurs and the call becomes impossible.

A loss control device and an echo canceller are used in order to solve the above-mentioned problems of the voice communication system.

[First Conventional Example] FIG. 11 shows the configuration of a conventional loss control device. In the figure, the same components as those in FIG. 10 are designated by the same reference numerals. In addition, in this example, the loss control device is provided on the transmitting side and the receiving side.
Only the receiving side is shown, and the receiving side will be described. Further, the amplifier is omitted in FIG. 11 is shown in FIG.
The case where the loss control device 12 is applied to the receiver side of the voice call system shown in FIG. The loss control device 12 is provided between the transmission line 9 and the speaker 4 and the microphone 3. Loss control device 1
2 includes a loss control circuit 13 and variable loss devices 14 and 15. The loss control circuit 13 receives the reception signal x (n),
Output signals z (n) from the microphone 3 are input, and the transmission / reception state is determined based on the magnitudes of these signals. The output signal z (n) from the microphone 3 is a signal (z (n)) obtained by combining the echo signal y (n) from the speaker 4 and the voice signal s (n) input from the microphone 3 on the receiving side.
= Y (n) + s (n)).

The loss control circuit 13 uses, for example, the reception signal x (n) and the output signal z as a method of determining the transmission / reception state.
The short-time powers Px (n) and Pz (n) of (n) are calculated and their magnitudes are compared. As a result of comparison, echo signal y
Assuming that (n) is smaller than the reception signal x (n), if there is no transmission signal s (n) (z (n) = y (n)), Px (n)> Pz (n) It is established, and at this time, it is determined to be the receiving state. On the other hand, the transmission signal s above a certain level
If (n) exists, Px (n) <Pz (n),
At this time, it is determined to be in the transmitting state. Further, if it is determined that it is also in the receiving state, the loss is inserted into the loss device 15 on the transmitting side. As a result, the echo signal y (n) received by the microphone 3 from the speaker 4 is transmitted to the loss unit 1
5 reduces the effect of reverberation.

On the other hand, if it is judged by the loss control circuit 13 that the transmission state is present, the loss of the transmission side loss device 15 is 0 dB.
As a result, the loss is inserted into the loss device 14 on the receiving side. As a result, the transmission signal s (n) received by the microphone 3 is transmitted without being attenuated. Further, by inserting the loss into the loss device 14 on the receiving side, the closed loop gain can be maintained at 1 or less, and the howling phenomenon can be prevented.

As described above, by using the loss control device 13, it is possible to reduce the phenomenon in which the voice uttered by the transmitting side echoes from the speaker 4 on the transmitting side, but the structure is simplified. From the meaning, the insertion loss amount is fixed,
The constantized value is inserted into the loss unit 14 or 15. Generally, a large value (for example, 20 dB) is given as the value of the insertion loss amount assuming all situations.

However, if the amount of loss inserted in the loss unit 14 or 15 exceeds 10 dB, the transmission / reception judgment is delayed and the beginning or end of the voice is cut off, resulting in the problem that the communication quality is deteriorated. To do. Therefore, there has been proposed an adaptive loss control device that adaptively controls the loss amount according to the acoustic coupling amount in the situation of actual use. Here, the acoustic coupling means that, in a device in which the speaker 4 and the microphone 3 are connected to the same device, the audio signal output from the speaker 4 is received by the microphone 3. Therefore, from the viewpoint of the device, the speaker is a speaker. The sound output from the microphone 4 is spatially combined and received by the microphone 3. That is, the signals are not coupled by electrical connection, but are spatially coupled as sound. Therefore, the acoustic coupling amount indicates how the input signal x (n) to the speaker 4 and the sound reception signal y (n) to the microphone 3 are coupled in power.

Although the amplifier is omitted in FIG. 11, the sound reception signal y (n) is y (n) = (amplifier 8) * (acoustic transfer characteristic) * (amplifier 7) * x ( n).

[Second Conventional Example] FIG. 12 shows the configuration of a conventional adaptive loss control apparatus. In the figure, the same components as those in FIG. 11 are designated by the same reference numerals. The adaptive loss control device 12 shown in the figure has a configuration in which an insertion loss amount determination circuit 41 is added to the configuration of FIG. Insertion loss determination circuit 41
Is the received signal L _R x after passing through the variable loss unit 14.
(N) and the output signal z (n) from the microphone 3, and the speaker 4 based on the levels of these signals.
To estimate the acoustic coupling amount G from the microphone to the microphone 3,
Determine the amount of insertion loss.

For example, the predicted value G ^ of the acoustic coupling amount is calculated by the short-term power PL _R x (n), Pz of the output signal L _R x (n) of the variable loss device 14 and the output signal z (n) of the microphone 3.
It can be calculated as follows using (n).

G ^ = Pz (n) / PL _R x (n) At this time, if the predicted value G ^ of the acoustic coupling amount is larger than 1 (0 dB), the received signal x (n) and the variable loss device 15 are connected to each other. The insertion loss amount determination circuit 41 calculates a loss amount such that the gain between the transmitted signal L _T x (n) and the transmitted signal is 1 or less.

For example, when the predicted value G ^ is 4 (6 dB), the input signal (received signal) x (n) and the transmission signal L _T
In order to keep the open loop gain between x (n) and 1 (0 dB) or less, the amount of insertion loss is 0.5 (6 dB in power) or less. Also, the acoustic coupling amount changes, and the predicted value G ^ is 2 (3d
In case of B), the insertion loss amount is the loss control device 1
3 is transferred. The loss control circuit 13 gives the insertion loss amounts a and b to the loss units 14 and 15 based on the above prediction.

However, the adaptive loss control device 12 described above is used.
Is the microphone 3 when estimating the acoustic coupling amount.
It is desired that the output signal z (n) from the transmission signal s (n) does not include the transmission signal s (n). However, since it is difficult to determine whether or not the output signal z (n) includes the transmission signal s (n), the estimation of the predicted value G ^ is often initially learned.

As described above, if the adaptive loss control device 12 is used, the echo phenomenon at the transmitting side 10 can be reduced, but the echo signal y (n) and the transmission signal are added to the output signal z (n). Under the condition that s (n) etc. are included and the amount of acoustic coupling is large,
Still, there remains a problem that the call quality is deteriorated.

Next, an echo canceller has been proposed to solve such a problem.

[Third Conventional Example] FIG. 13 shows the structure of a conventional echo canceller. The echo canceller 21 shown in FIG.
It includes a pseudo echo path 22, an echo path estimation circuit 23, and a subtractor 24. The speaker 4 receives the received signal x (n), and the microphone 3 receives the echo signal y from the speaker 4.
(N) is input, and there is h (n) as the echo path transfer characteristic (impulse response) between the speaker 4 and the microphone 3.

The echo canceller 21 includes an echo path estimation circuit 23.
Estimates the impulse response of the echo path (between the speaker 4 and the microphone 3) and transfers the estimated value h ^ (n) to the pseudo echo path 22. Next, in the pseudo echo path 22, the estimated value h ^ (n) output from the echo path estimation circuit 23 and the received signal x
The convolution operation with (n) is executed and the pseudo echo signal y ^
(N) is output to the subtractor 24. The subtractor 24 subtracts y ^ (n) from the output signal z (n) of the microphone 3. In the echo path estimation circuit 23, if the echo path impulse response is well estimated, the echo signal y (n) and the pseudo echo signal y ^ (n) are substantially equal.
If the error signal e (n) is 0 as a result of this subtraction, the echo signal y (n) included in the output z (n) of the microphone 3 is generated.
Is erased. Here, the pseudo echo path 22 needs to follow the temporal change of the echo path characteristic h (n). Therefore, the echo path estimation circuit 23 uses an adaptive algorithm to estimate the echo path impulse response. The adaptive algorithm is an algorithm that uses the received signal x (n) and the error signal e (n) to determine the estimated value h ^ (n) of the impulse response so that the power of the error signal e (n) is minimized. There are known LMS method, learning identification method, ES method, and the like.

However, the above-mentioned algorithm for estimating the echo path estimates the pseudo echo path in the wrong direction by regarding the transmission signal of the near-end speaker as an error when the signal is present. . Therefore, it is necessary to detect whether or not the received signal x (n) and the transmitted signal s (n) of the near-end speaker are mixed (double talk), and if there is double talk, it is necessary to stop the estimation.

As a method of solving the situation where adaptive estimation is stopped at the time of table talk, the FG / BG which satisfactorily estimates the echo path without explicitly detecting double talk.
A (foreground / background) method has been proposed.

[Fourth Conventional Example] FIG. 14 shows the structure of an echo canceller using the FG / BG method. The same components as those in FIG. 13 are designated by the same reference numerals. The echo canceller 50 shown in FIG. 14 includes an FG-side pseudo echo path 51, an estimation circuit 54, and a BG.
Side pseudo anti-common path 55, subtractors 60 and 61, transfer determination circuit 6
It consists of 2.

In the FG / BG echo canceller 50, the echo path estimation circuit 54 estimates the echo response of the echo path from the error signal eb (n) from the subtractor 60 and the received signal x (n). The estimated value h ^ b (n) is transferred to the pseudo echo path 55 on the BG side. Next, the pseudo echo path 55 on the BG side executes a convolution operation of the estimated value h ^ b (n) and the received signal x (n) to synthesize the pseudo echo signal y ^ b (n) and subtract it. Input to the container 60. The subtractor 60 calculates the pseudo echo signal y ^ b from the output signal z (n) from the microphone 3.
Subtract (n) and output the error eb (n). If the echo path impulse response is estimated well, the echo signal y (n) and the pseudo echo signal y ^ (n) are substantially equal.

Next, the transfer judgment circuit 62 causes the macrophone 3
Of the pseudo-echo signal y ^ b (n) output from the subtractor 60 and the magnitude of the error eb (n) of the pseudo echo signal y ^ b (n), for example, the short-time powers Pz (n) and Peb (n), Use and compare. Here, the received signal x (n) is larger than a preset threshold Th: The short-time power Peb (n) of the error eb (n) is the short-time power Pz (n) of the output signal z (n) of the microphone 3.
Smaller than: smaller than power Pef (n) of power Peb (n) of power eb (n) on the BG side: when three conditions are satisfied, the pseudo echo coefficient h on the BG side estimated by the estimation circuit 54 ^ B (n) is FG
It is considered that the actual echo path is better simulated as compared with the side pseudo echo path coefficient h ^ f (n). Therefore, BG
The side pseudo echo path 55 transfers the BG pseudo echo path coefficient h ^ b (n) to the FG pseudo echo path 51. The coefficient h ^ f (n) of the FG side pseudo echo path 51 is updated only when the above conditions are satisfied. Therefore, when the estimation circuit 54 makes an erroneous estimation during the table talk, the BG-side coefficient h ^ b (n) is not transferred to the FG-side pseudo echo path 51 because the above condition is not satisfied. As a result, the FG that holds the coefficient only when the echo cancellation is clearly obtained on the BG side
On the side, there is no disturbance in the coefficient due to double talk.

However, the FG / BG echo canceller is
Until the coefficient on the BG side converges at the initial stage and is transferred to the FG side, the coefficient on the FG side is zero and the echo signal is not canceled at all. Furthermore, even if the echo path changes during use, the echo path is correctly estimated on the BG side, and F
The echo canceling amount decreases during the period until the transfer to the pseudo echo path 51 on the G side.

The problem that the echo canceling amount decreases with respect to the change in the echo path during the initial stage and during use is the same as in the above-described normal echo canceling device. As a method of solving this problem, it is possible to perform initial learning or, if the echo canceller is not converged, insert a loss to the extent that howling does not occur by the loss controller. .

The case where the echo canceller and the adaptive loss controller are combined will be described below.

[Fifth Conventional Example] FIG. 15 shows a structure of an echo canceller which combines a conventional echo canceller and an adaptive loss controller. FIG. 16 shows a configuration of an echo canceller that combines an echo canceller using a conventional FG / BG method and an adaptive loss controller. 15 and 16, the same components as those in FIGS. 12, 13 and 14 are designated by the same reference numerals.

In FIG. 15, connection points 71 and 72 are provided between the echo canceller 21 and the adaptive loss controller 12.
Similarly, in FIG. 16, connection points 71 and 72 are also provided between the echo canceller 50 and the adaptive loss controller 12.

The echo canceller 21 and the adaptive loss controller 12,
In the device in which the echo canceller 50 and the adaptive loss controller 12 are combined, the adaptive loss controller 12 includes the echo canceller 2
The acoustic coupling amount between the connection points 71 and 72 including 1 and 50 is estimated. As a result, when the echo cancellation amount of the echo cancellation units 21 and 50 is small at the initial stage, the insertion loss amount of the adaptive loss control device 12 is increased to suppress the echo, and the echo cancellation units 21 and 50 are controlled. When the echo signal is removed to some extent, the insertion loss amount of the adaptive loss control device 12 can be reduced.

[0033]

In the above-mentioned fifth conventional example, howling suppression, which is a problem in the echo cancellers of the first to fourth conventional examples, when the amount of echo cancellation is small is echoed. This can be solved by combining the erasing unit and the adaptive loss control unit, but in the insertion loss amount determination circuit 41 of the loss control device 12 that adaptively determines the insertion loss amount with respect to the change in the acoustic coupling amount, the correct acoustic coupling amount is obtained. In order to make prediction, it is necessary to judge whether or not the output signal z (n) from the microphone 3 contains the transmission signal s (n). However, there is a problem that it is difficult to determine an appropriate insertion loss amount.

The present invention has been made in view of the above points, solves the above-mentioned conventional problems, and determines the amount of loss to be inserted when the adaptive loss control section is used in combination with the echo canceling section. At that time, correctly estimate the required acoustic coupling amount,
It is an object to provide an echo canceller capable of canceling an echo signal, an echo canceling method, and a handset having the echo canceller.

A further object of the present invention is to determine the double talk state from the converged state of the echo canceller, and change the estimation method of the acoustic coupling amount based on the result of the determination, thereby minimizing the amount of loss to be inserted. An echo canceller having a loss control unit that suppresses

Another object of the present invention is to provide an echo canceling method capable of removing a signal echoing from a speaker to a microphone and receiving only clear voice from the other party.

Another object of the present invention is to provide a handset having an echo canceller that cancels echo signals and minimizes the amount of loss input to a communication path.

[0038]

FIG. 1 is a block diagram showing the principle of the present invention.

According to the present invention, when the received signal reverberates through the reverberation path 100, the echo canceling section 200 having a canceling means for outputting information as to whether or not the echo signal has been cancelled, and information input from the echo canceling section 200. Based on this information (echo cancellation amount), it is determined whether or not the echo signal is mixed in the transmission signal, and the loss amount determining means 320 for determining the loss amount to be inserted in the communication path according to the determination result, and the loss amount. There is provided loss control means 330 for inserting the amount of loss determined by the amount determining means 320 into at least one of the reception signal side and the transmission signal side, and one round loop gain of the communication path exceeds 1. Is combined with an adaptive loss control circuit 300 that suppresses

The canceling means of the echo canceller 200 of the present invention uses the pseudo echo path 210 from the echo signal y (n) sent to the echo path and the echo signal y (n) after the echo signal has passed through the echo path.
And the output signal x (n) and the estimated value of the impulse response of the echo path are input to the pseudo echo path 210 from the estimation means 220, and the pseudo echo signal y ^ obtained from the pseudo echo path 210.
Eliminating the echo signal y (n) by subtracting (n) from the echo signal y (n), subtracting the error signal e (n),
The input signal z (n) is output, and the loss amount determining means 320 of the loss control unit 300 uses the signals e (n), z (n) and the reception signal x (n) input from the echo canceling unit 200. ,
It is determined whether a signal other than the transmission signal s (n) is mixed in the input signal z (n), and if it is mixed, a double talk state judging means for judging a double talk state is provided, and a loss occurs. The determining unit 320 selects a method for estimating the acoustic coupling amount by acoustic coupling based on the acoustic transfer characteristics based on the determination result of the double talk state determining unit.

Further, the double talk state judging means of the present invention has the pseudo echo signal y ^ (n), the input signal z (n) and the pseudo echo signal y ^ (n) obtained by the estimating means 220 of the echo canceller 200. ), The error e (n) between the input signal z (n) and the pseudo echo signal y ^ (n), and the short-term power Px (n), Pz (n), using the received signal x (n).
Pe (n) is calculated, and the short-time power Px (n) of the transmission signal x (n) exceeds a predetermined threshold value xth, and Pe (n) <Th * Pz between the threshold value Th set to 1 or less. If the relationship of (n) is established, it is determined that the double talk state is not established, and Px (n)> xth and Pe (n)> Th * Pz.
When the relationship of (n) is established, it is determined that the double talk state or the echo path is changed.

The loss amount determining means 320 of the present invention is
When the double-talk state determining means determines that the state is not the double-talk state, Px (n) / Pz (n) is set as the loss amount, and when the double-talk state or the echo path changes, the double-talk state The loss amount is calculated using a time longer than the time interval expected to continue and is input to the loss control means 330.

Further, in the loss amount determining means 320 of the present invention, when the double talk state or the echo path is changed, the power PPx (n) of the transmission signal x (n) and the input signal whose integration time is increased. Power PPz (n) of z (n)
The loss amount is calculated by PPx (n) / PPz (n) using, and is input to the loss control means 330.

Also. In the loss amount determining means 320, when the double talk state or the echo path changes,
A value obtained by averaging a value obtained by Px (n) / Pz (n) estimated in a short time over a long time from a predetermined time interval is set as a loss amount and input to the loss control means.

FIG. 2 is a flow chart for explaining the principle of the present invention.

According to the present invention, in the echo path, the echo signal y
(N) (however, y (n) = h (n) * x (n)) is deleted from the input signal z (n) (step 1), and the deleted echo signal amount is transferred to the loss controller (step). 2), the loss control unit changes the echo signal y to the transmission signal based on the echo signal amount.
It is determined whether (n) is mixed (step 3), a method for determining the loss amount is selected based on the result of the determination (step 4), the loss amount is determined based on the selected determination method, and transmission is performed. It is inserted into at least one of the signal communication path and the received signal communication path to suppress the one-loop gain of the communication path from exceeding 1 (step 5).

In the step 1 of eliminating the echo signal, the echo signal z (n) after passing through the echo path and the signal x (n) sent to the echo path constitute a pseudo echo path, which is obtained from the pseudo echo path. The pseudo echo signal y ^ (n)
(However, the echo signal is eliminated by subtracting from z (n) = y (n) + s (n)).

Further, in the step 4 for selecting the method for determining the loss amount, when the echo signal is mixed in the transmission signal, a time longer than the time in which the echo signal is mixed is used. The method of determining the amount of loss according to the echo signal mixed in is selected, and when the echo signal is not mixed in the transmission signal, the short-time power Px (n) of the transmission signal x (n) is reflected. A method of determining the loss amount is a value obtained by dividing the sum z (n) of the signal y (n) and the transmission signal s (n) by the short-time power Pz (n).

Further, in the handset having the echo canceller of the present invention, the speaker for outputting the reception signal, the microphone for inputting the voice, and the reception signal received through the communication path reverberate through the echo path. In this case, an echo canceller having a canceller that outputs the echo canceling amount that cancels the echo signal,
The echo cancellation amount is input from the echo cancellation unit, and it is determined whether the echo signal is mixed in the transmission signal based on the echo cancellation amount,
Depending on the determination result, the loss amount determining means for determining the loss amount to be inserted into the communication path, and the loss amount determined by the loss amount determining means to the reception signal side or the transmission signal side in at least one of the communication paths. The echo canceller has a loss control means to be inserted, and is combined with an adaptive loss control section for suppressing the one-loop loop gain of the communication path from exceeding 1.

[0050]

According to the present invention, in the adaptive loss controller, it is determined whether the echo signal generated from the echo path between the speaker and the microphone exists in the voice signal s (n) from the microphone (double talk state) or not. , Judgment is made according to the state of convergence (echo signal elimination) of the echo canceller. That is, the insertion loss amount is distinguished according to the method of estimating the acoustic coupling amount according to the state in which the echo cancellation is performed to some extent in the echo cancellation unit, the state in which the echo cancellation is not performed, or the double talk state. Can be correctly estimated.

[0051]

Embodiments of the present invention will now be described in detail with reference to the drawings.

[First Embodiment] FIG. 3 shows the system configuration of the first embodiment of the present invention. In the figure, the same components as those in FIG. 10 are designated by the same reference numerals. The amplifier in FIG. 10 is omitted for simplicity. The system shown in FIG. 3 is provided with a loss control unit 120 and an echo canceller 21 in addition to the configuration of FIG. In FIG. 3, the echo canceller 21
Outputs a pseudo echo signal from the pseudo echo path, and the echo path 10
0 (between the speaker 4 and the microphone 3) subtracts the pseudo echo signal (y ^ (n)) from the output signal z (n) of the microphone 3 including the echo signal, and the adaptive loss control unit 120
To enter.

The echo canceling unit 120 receives the error signal between the output signal from the microphone 3 including the echo signal from the echo canceling unit 21 and the output signal from the pseudo echo path, and determines whether echo cancellation is performed. Determine the amount of loss to be inserted depending on the situation.

FIG. 4 shows a configuration in which the loss control section and the echo canceling section of the first embodiment of the present invention are combined. In the figure, the same components as those in FIG. 11 are designated by the same reference numerals. The adaptive loss control unit 120 shown in FIG. 4 is the adaptive loss control unit 1 of FIG.
2 has a configuration in which an echo canceling convergence state determination circuit 80 as double talk state determination means is added.

Received signal x (n) (L _R x from the receiving side)
(N)) is input to the speaker 4, an echo path 100 is formed between the speaker 4 and the microphone 3, and the echo signal y (n) from the speaker 4 is input to the microphone 3. However, as shown in FIG.
When only (n) is present, the sound reception signal of the microphone 3 receives only the reception signal x (n) affected by the acoustic transfer characteristic h (n).

On the other hand, as shown in FIG. 6, the received signal x
When (n) and the transmission signal s (n) are mixed, the transmission signal s (n) is a signal to be transmitted and cannot be erased. If there is a signal that cannot be completely removed by the echo canceller 21, the current transmission signal s
(N) means that the signals are mixed, or
To distinguish whether the echo signal cannot be canceled is
Since it is difficult, the sound elimination unit 21 to the loss control unit 120
, The difference e (n) between the output signal z (n) from the microphone 3 and the pseudo echo signal y ^ (n) via the pseudo echo path 22 and the echo signal y (n) is transmitted (however, y (n n) = h
(N) * x (n)).

The microphone 3 synthesizes, as the output signal z (n), the echo signal y (n) and the audio signal s (n) input to the microphone 3 (z (n) = y (n) +).
s (n)) is output to the subtractor 24. This is because the voice signal s (n) is the voice uttered by the listener, voice from the near end, etc., and the echo signal y (n) is removed, and only the original voice signal s (n) is transmitted to the partner type. It is what

The estimation circuit 23 as the estimation means of the echo canceller 21 estimates the impulse response h (n) of the echo path and transfers the estimated value h ^ (n) to the pseudo echo path 22.
At this time, when the estimation circuit 23 obtains the estimated value h ^ (n), it is assumed that the adaptive algorithm is used as in the third conventional example. The pseudo echo path 22 receives the received signal L _R x.
(N) and the estimated value h ^ (n) are convolved and the pseudo echo signal y ^ (n) is output to the subtractor 24. The subtractor 24 subtracts the output signal z (n) from the microphone 3 and the pseudo echo signal y ^ (n) to obtain an error signal e (n).

E (n) = z (n) -y ^ (n) The error signal e (n) from the subtractor 24 is input to the estimation circuit 23, and the estimation circuit 23 uses the adaptive algorithm to obtain the echo path impulse response. h ^ (n) is estimated and h ^ (n) is minimized so that the short-time power of the error signal e (n) is minimized.
To decide. The error signal e (n) from the subtractor 24 thus minimized is also output to the loss controller 21.

Echo cancellation convergence determination circuit 8 of the loss control section 21
0 is the received signal L _R x (n), the signal z (n) from the microphone 3 and the error signal e (n) via the loss device 15.
Is entered. The echo cancellation convergence determination circuit 80, for example,
The acoustic coupling amount G is estimated as follows and output to the insertion loss amount determining circuit 41 as the loss amount determining means.

The short-term powers Px (n), Pz (n) and Pe (n) of the input signal L _R x (n), the microphone input signal z (n) and the error signal e (n) are calculated.
Here, Px (n) exceeds the set threshold value xth, and 1
If Pe (n) <Th * Pz (n) with the threshold value Th set below, it is determined that echo cancellation has been achieved. In this case, the input signal z (n) of the microphone 3 is s
Since it can be considered that (n) is not included, the insertion loss amount determination circuit 41 estimates the acoustic coupling amount G as G ^ = Px (n) / Pz (n). Further, Px (n)> xth and Pe (n)> Th * Pz
When (n), s (n) is included in z (n), or the echo path is changed and the pseudo echo path 22 is not correctly estimated. In the double-talk state, the correct acoustic coupling amount can be estimated, but since the echo path may have changed, it is necessary to estimate the acoustic coupling amount.

In such a case, the amount of acoustic coupling is estimated in the insertion loss amount determining circuit 41 using a time longer than the time interval (about 3 seconds) expected to continue the double talk state. In the double-talk state, the necessary echo cancellation amount cannot be calculated in a short time, so it must be estimated in a long time.

The insertion loss amount determining circuit 41 is, specifically,
The acoustic coupling amount is estimated by the following method. (1) Powers PPx (n) and PPz with a long integration time
Method of setting G ^ = PPx (n) / PPz (n) using (n): (2) Method of averaging G ^ Px (n) / Pz (n) estimated in a short time over a long time : Etc.

The acoustic coupling amount G estimated by the insertion loss amount determining circuit 41 is input to the loss control circuit 13 as loss control means. The loss control circuit 13 determines the transmission / reception state based on the estimated acoustic coupling amount G ^, the reception signal x (n), and the error signal e (n) from the attenuator 24. When it is determined that the state is the receiving state, the loss is inserted into the loss device 15 on the transmitting side. As a result, the reverberation phenomenon is reduced by being attenuated by the loss device 15. On the other hand, by the loss control circuit 13,
When it is determined that the state is the transmitting state, the loss is inserted into the loss device 14 on the receiving side. As a result, the transmission signal s (n) input from the microphone 3 is transmitted without being attenuated.

[Second Embodiment] FIG. 7 shows the system configuration of the second embodiment of the present invention. FIG. 7 shows the system configuration of the second embodiment of the present invention. In the figure, the same components as those in FIG. 3 are designated by the same reference numerals. The configuration shown in FIG.
In the embodiment, the estimation circuit of the echo canceller 21 estimates the echo response of the echo path by using the adaptive algorithm. However, in this embodiment, the echo canceller 50 of the FG / BG method cancels the echo from the microphone 3. The difference between the output signal and the pseudo echo signal is obtained, and in the adaptive loss control unit 120, the magnitudes thereof are compared using the short-time power of the input signal, and the estimated value estimated by the estimation circuit is the actual echo. Double talk state is judged by the degree of simulating the road. The adaptive loss control unit 120 determines the loss amount depending on whether or not it is in the double talk state, and controls the loss amount.

FIG. 8 shows a configuration in which the loss control section of the second embodiment of the present invention and the FG / BG type echo canceling section are combined. In the figure, the same components as those in FIG. 16 are designated by the same reference numerals. The configuration of the present embodiment is shown in FIG. 16 in that the echo canceling convergence state determination circuit 80 is added in the loss control unit 120.
The configuration is different.

The echo cancellation convergence determination circuit 80 of this embodiment is
In the first embodiment, instead of the error signal e (n) input from the subtractor 24, the error signal eb from the subtractor 60
(N) is input. The echo cancellation convergence determination circuit 80, for example, the short-time power Px (n) and Pz (n) of the input signal L _R x (n), the input z (n) from the microphone 3, and the error signal eb (n), respectively. , Peb (n) is calculated.

The short-time power is obtained as follows. Here, an example of obtaining x (n) is shown, but other signals can be similarly obtained by substituting.

If all signals are discretized, the short-time power from the discretized time n−N to the present time n is

[0070]

[Equation 1]

If α is set to α = N-1 / N, Px (n) = x ² (n) + αPx (n-1) may be obtained. It can also be obtained by adding the squared value to the power one hour before and subtracting the squared value data past N times. Px (n) = Px (n -1) + x 2 (n) -x 2 (n-
N) Here, Px (n) exceeds the set threshold value xth, and 1
If Pe (n) <Th * Pzb (n) (1) with the threshold value Th set below, it is determined that echo cancellation has been achieved. In this case, it can be considered that the output z (n) from the microphone 3 does not include the voice signal s (n), and therefore the insertion loss amount determination circuit 41 determines the acoustic coupling amount as G ^ = Px. It is estimated as (n) / Pz (n) (2). When Px (n)> xth and Pe (n)> Th * Pzb (n) (3), the output z (n) from the microphone 3 includes the audio signal s (n). It is a double talk state in which the echo path is changed or the pseudo echo path 55 is not correctly estimated. In the double talk state,
Although it is not possible to correctly estimate the acoustic coupling amount, it is necessary to estimate the acoustic coupling amount because the echo path may have changed. In such a case, the amount of acoustic coupling is estimated in the insertion loss amount determination circuit 41 by using a time longer than the time interval in which the double talk state is predicted to continue. The specific estimation method is the same as in the first embodiment described above.

From the above, the operation of the system shown in FIG.
It looks like this:

When the reception signal L _R x (n) is input to the speaker 4, the echo signal y (n) is input to the macrophone 3 via the echo path formed between the speaker 3 and the microphone 3. The estimation circuit 54 estimates the impulse response of the echo path between the speaker 4 and the microphone 3, and estimates its estimated value h ^.
(N) is transferred to the pseudo echo path 55 on the BG side. Next, the pseudo echo path 55 is BG by h ^ (n) and L _R x (n).
The side pseudo echo signal y ^ b (n) is output to the subtractor 60. The subtractor 60 outputs the output signal z (n) of the microphone 3.
The error signal eb (n) is calculated by subtracting y ^ b (n) from the error signal eb (n) and sent to the transfer judgment circuit 62.

Next, the transfer judgment circuit 62 outputs the error signal eb between the output signal z (n) of the microphone 3 and y ^ (n).
The sizes of (n) are compared. The method of comparison is the same as that of the fifth conventional example, and therefore the description is omitted.

If the comparison result condition is satisfied, BG
Assuming that the coefficient of the pseudo echo path 55 on the side has converged,
The BG side pseudo echo path 55 transfers the coefficient to the FG side pseudo echo path 51.

The subtracter 61 obtains the error ef (n) of the output signal z (n) of the microphone 3 and the coefficient h ^ f from the FG side pseudo echo path 51, transfers it to the transfer judgment circuit 62, and cancels the echo. Input to the section 120. The echo canceling convergence determining circuit 80 of the echo canceling unit 120 outputs the error signal eb (n) from the subtractor 60 and the output signal z (n) from the microphone 3.
And the received signal L _R x (n) are input, and the short-term powers of these signals are calculated. Here, the received signal L _R x (n)
If the power PL _R x (n) of 1 does not exceed 1, it is determined that the double talk state is not entered, and the determination result is input to the insertion loss amount determination circuit 41. The insertion loss amount determination circuit 41
The insertion loss amount input to the loss control circuit 13 is determined based on the determination result from the echo cancellation convergence determination circuit 80.

When the double talk state is not set (1),
The acoustic coupling amount G ^ is estimated using the above (2). Also,
When it is assumed that the double talk state or the echo path is changing as in (3), a time longer than the time interval in which the double talk state is expected to continue is set in the same manner as in the first embodiment. The acoustic coupling amount G ^ is estimated by.

The insertion loss amount determination circuit 41 inputs the estimated acoustic coupling amount G ^ to the loss control circuit 13. The loss control circuit 13 determines the transmission / reception state as in the first conventional example, and inserts the loss into the loss unit 14 or 15.

In the above embodiment, the estimation circuit 2
3, an adaptive algorithm is used to obtain the estimated value h ^ (n), but the present invention is not limited to this example.
A method for obtaining the estimated value of the impulse response may be applied. In the second embodiment, the condition of the FG / BG method is set to estimate the impulse response of the echo path, but the present invention is not limited to this example, and the estimated value of the impulse response of the echo path is actually There is no error with the impulse response value of the echo path, and the pseudo echo signal y ^ (n) and the echo signal y (n) may be approximated to the same value without limit, or any method may be used.

In the first embodiment, the estimation circuit 23, the pseudo echo path 22 and the subtractor 24 are used in the echo canceller 21, and in the second embodiment the echo canceller 50 is used.
Although the echo canceling circuit of the FG / BG system is used for the above, the echo canceling unit is not limited to the above two embodiments, and as shown in FIG. 7
Information indicating whether or not the echo signal is erased at 0 is input to the loss control unit 120. Thereby, the loss control unit 1
The double talk state is determined at 20, and the loss control device determines the amount of loss at which the gain of one cycle does not exceed 1 based on the determination, and the loss amount is inserted into the loss devices 14 and 15.

In the above embodiment, the case of applying to a telephone having a loudspeaker function is described, but the present invention can also be applied to a case where the echo adversely affects a transceiver or the like capable of bidirectional communication.

[0082]

As described above, according to the present invention, it is possible to judge in a double-talk state from the state of convergence of the echo canceller, and in a state where the echo canceller performs the echo canceller to some extent, it takes a short time. The acoustic coupling amount is estimated, and when the echo cancellation is not performed (double talk state), the acoustic coupling is estimated for a long time. As a result, the accuracy of estimation of the acoustic coupling amount in the adaptive loss control unit is improved, the insertion loss amount can be suppressed to the minimum, and the communication quality is improved.

[Brief description of drawings]

FIG. 1 is a principle configuration diagram of the present invention.

FIG. 2 is a flow chart for explaining the principle of the present invention.

FIG. 3 is a system configuration diagram of a first embodiment of the present invention.

FIG. 4 is a configuration diagram in which a loss control unit and an echo canceller according to the first embodiment of the present invention are combined.

FIG. 5 is a diagram (No. 1) for explaining the first embodiment of the present invention.

FIG. 6 is a diagram (No. 2) for explaining the first embodiment of the present invention.

FIG. 7 is a system configuration diagram of a second embodiment of the present invention.

FIG. 8 is a loss control unit and FG / B according to a second embodiment of the present invention.
It is a block diagram which combined the echo cancellation part of G system.

FIG. 9 is a configuration diagram of another embodiment of the present invention.

FIG. 10 is a block diagram of a conventional voice communication system.

FIG. 11 is a configuration diagram of a conventional loss control device.

FIG. 12 is a configuration diagram of a conventional adaptive loss control device.

FIG. 13 is a block diagram of a conventional echo canceller.

FIG. 14 is a configuration diagram of a conventional FG / BG type echo canceller.

FIG. 15 is a configuration diagram in which a conventional echo canceller and an adaptive loss controller are combined.

FIG. 16 is a configuration diagram in which an echo canceller and an adaptive loss controller using the conventional FG / BG method are combined.

[Explanation of symbols]

 1, 3 Microphones 2, 4 Speakers 5, 6, 7, 8 Amplifier 9 Transmission line 10 Speaker 11 Listener 12 Loss control unit 13 Loss control circuit 14, 15 Variable loss unit 21 Echo canceling unit 22 Pseudo echo path 23 Estimating circuit 24 attenuator 41 insertion loss amount determination circuit 50 FG / BG system echo canceller 51 FG side pseudo echo path 54 estimation circuit 55 BG side pseudo echo path 60 subtractor 62 transfer judgment circuit 70 echo cancellation unit 80 echo cancellation convergence judgment circuit 100 Echo Path 120 Loss Control Circuit 200 Echo Canceller 210 Pseudo Echo Path 220 Estimator 230 Subtractor 300 Adaptive Loss Control Unit 310 Loss Amount Determiner 330 Loss Control Means 410 Talking Means 500 Echo Canceller

 ─────────────────────────────────────────────────── ─── Continued Front Page (72) Inventor Masafumi Tanaka 1-1-6 Uchisaiwaicho, Chiyoda-ku, Tokyo Nihon Telegraph and Telephone Corporation

Claims (10)

[Claims] 1. An echo canceller having canceling means for outputting information indicating whether or not the echo signal has been canceled when a received signal received through the communication channel echoes through the echo channel, and the echo canceller. A loss amount determining means for inputting the information from the erasing unit, determining whether the echo signal is mixed in the transmission signal based on the information, and determining the loss amount to be inserted into the communication path according to the determination result. And loss control means for inserting the loss amount determined by the loss amount determining means into at least one of the reception signal side and the transmission signal side, and one loop loop gain of the communication path An echo canceller characterized by being combined with an adaptive loss control unit for suppressing exceeding 1. 2. The echo canceller of the echo canceller forms a pseudo echo path from a sent signal x (n) to the echo path and an echo signal y (n) after the sent signal has passed through the echo path. , The transmitted signal x (n) and the estimated value of the impulse response of the echo path are input to the Guy pseudo echo path by the estimating means, and the pseudo echo signal y ^ (n) obtained from the pseudo echo path is converted to the echo signal y.
The echo signal y (n) is eliminated by subtracting it from (n), and the subtracted error signal e (n) and input signal z (n)
The loss amount determining means of the loss control section outputs the signals e (n), z (n) and the reception signal x input from the echo canceling section.
Using (n), the transmission signal s (n) is added to the input signal z (n).
Other than the signal is mixed, if mixed, has a double talk state determination means to determine the double talk state, the loss determination means, the determination result of the double talk state determination means The echo canceller according to claim 1, wherein a method for estimating the acoustic coupling amount due to acoustic coupling based on the acoustic transfer characteristic is selected based on the above. 3. The double talk state determination means, the pseudo echo signal y ^ (n), the input signal z (n) and the pseudo echo signal y obtained by the estimation means of the echo canceller.
^ (N), the input signal z (n) and the pseudo echo signal y ^
Using the error e (n) with respect to (n) and the received signal x (n), the respective short-time powers Px (n), Pz (n), Pe
(N) is calculated, and the short-time power Px of the transmission signal x (n) is calculated.
If (n) exceeds a predetermined threshold value xth and a threshold value Th that is set to 1 or less is satisfied, Pe (n) <Th * Pz (n) is established, it is determined that the double talk state is not established, and P
x (n)> xth and Pe (n)> Th * Pz (n)
3. The echo canceller according to claim 2, wherein it is determined that the double talk state or the echo path is changed when the relationship is satisfied. 4. The loss amount determining unit determines Px (n) / Pz (n) when the double-talk state determining unit determines that the state is not a double-talk state.
Is the loss amount, and when the double talk state or the echo path is changing, the loss amount is obtained using a time longer than the time interval expected to continue the double talk state, and is input to the loss control means. The echo canceller according to claim 2. 5. The power PPx of the transmission signal x (n) having a longer integration time in the loss amount determining means when the double talk state or the echo path is changing.
5. The echo canceller according to claim 4, wherein the loss amount is calculated by PPx (n) / PPz (n) using (n) and the power PPz (n) of the input signal z (n), and is input to the loss control means. . 6. The loss amount determining means sets a value obtained by Px (n) / Pz (n) estimated in a short time to a predetermined time when a double-talk state or the echo path is changed. The echo canceller according to claim 4, wherein a value averaged over a longer period than a section is set as a loss amount and is input to the loss control means. 7. In the echo path, a step of canceling an echo signal from an audio signal, a step of transferring the deleted echo signal amount to a loss control section, the loss control section transmitting speech based on the echo signal quantity. The step of determining whether the echo signal is mixed in the signal, the step of selecting the method of determining the loss amount based on the result of the determination, and the loss amount is determined based on the selected determination method, and the transmission signal communication path is determined. A method of canceling an echo signal, comprising the step of inserting the received signal into at least one of the communication paths and suppressing the one-loop gain of the communication path from exceeding 1. 8. The step of eliminating the echo signal comprises forming a pseudo echo path by the echo signal z (n) that has passed through the echo path and a signal x (n) sent to the echo path. 8. The echo signal elimination method according to claim 7, wherein the echo signal is eliminated by subtracting the echo signal z (n) from the pseudo echo signal y ^ (n) obtained from the pseudo echo path. 9. The step of selecting the method of determining the loss amount, when the echo signal is mixed in the transmission signal, a time longer than a time period in which the echo signal is mixed continues. Select a method of determining the amount of loss according to the echo signal mixed using, when the echo signal is not mixed in the transmission signal,
A value obtained by dividing the short-time power Px (n) of the transmission signal x (n) by the short-time power Pz (n) of the sum z (n) of the echo signal y (n) and the transmission signal s (n) is lost. The echo signal canceling method according to claim 7, wherein a method of determining the quantity is selected. 10. A speaker for outputting a reception signal, a microphone for inputting a voice, and a reception signal received through a communication path, when the reception signal reverberates through the echo path, the echo cancellation amount by which the echo signal is eliminated. And an echo canceller having an canceller for outputting the echo canceller, and the echo canceller receives the echo canceller from the echo canceller and determines whether the echo signal is mixed in the transmission signal based on the echo canceller. Accordingly, the loss amount determining means for determining the loss amount to be inserted into the communication channel, and the loss amount determined by the loss amount determining means for at least one of the reception signal side and the transmission signal side. An echo canceller having a loss control means to be inserted, and an echo canceller combined with an adaptive loss controller for suppressing that the one-loop loop gain of the communication path exceeds 1. Sending and receiving Machine.