JP3121997B2 - Acoustic echo canceller - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention is used for a communication line, an indoor sound field control device, and a high-quality audio communication conference device, and a signal of a receiving path appears on a transmission path via an acoustic reflection path. The present invention relates to an acoustic reverberation removing device for removing noise.

[0002]

2. Description of the Related Art Generally, an acoustic echo canceler removes reflection caused by impedance mismatch of a two-wire / four-wire converter in a long-distance telephone line using a communication satellite and a submarine cable, and a loudspeaker such as a video conference system. , Which can be broadly divided into those that remove the reverberation due to the acoustic coupling of the speaker's voice, and are composed of a correction amount calculation circuit, a variable coefficient filter that generates a pseudo acoustic reverberation, and a subtraction circuit. The basic operation of the acoustic reverberation removing device will be described below.

FIG. 7 is a diagram showing a basic configuration of an acoustic reverberation removing apparatus. The reception signal input terminal 1 is connected to the reception signal output terminal 2, and the reception signal of the reception signal input terminal 1 is branched and supplied to the variable coefficient filter 3 to generate a pseudo acoustic echo. The transmission signal from the transmission signal input terminal 4 and the pseudo acoustic reverberation output from the variable coefficient filter 3 are input to a subtraction circuit 5, where the acoustic reverberation component in the transmission signal is removed, and the output of the subtraction circuit 5 is It is output to the transmission signal output terminal 6. The output of the transmission signal output terminal 6 and the signal of the reception signal input terminal 1 are input to the correction amount calculation circuit 7, and the filter coefficient of the variable coefficient filter 3 is corrected by the output of the coefficient correction amount calculation circuit 7. In the variable coefficient filter 3, the reception signal is input to the reception signal input register 8, and the product sum of the reception signal of the reception signal input register 8 and the pseudo impulse response of the pseudo impulse response register 9 is obtained by the product sum circuit 10. Product-sum circuit 10
Is output as a pseudo acoustic echo. The reception signal output terminal 2 and the transmission signal input terminal 4 are connected to a two-wire / four-wire converter for a long-distance telephone line, and to a speaker and a microphone for a loudspeaker system.

[0004] Assuming that the signal propagation characteristics of the acoustic reverberation path are linear and represented by an FIR type digital filter, and using the impulse response h (t) and the input received signal x (t), a sample time interval is obtained. Is ^T , the acoustic reverberation y _{k at} time kT is represented by y _k = h Tx _k (1). Here, h = [h ₁ , h ₂ ,..., H _n ] ^T (2) x = [x _k−1 ,..., X _kn ] ^{T T} : Vector transposition.

On the other hand, the estimated value of h at time kT is expressed as h
if s _k, estimated value ys _k of y _k is given by ^{_{ys k = hs k T x k}} (3). In the acoustic reverberation removing device, when there is an audio signal at the receiving signal input terminal 1 and no acoustic signal exists at the transmitting signal input terminal 4 and only acoustic reverberation exists, the acoustic reverberation operation is performed as an adaptive operation state. This adaptive operation algorithm generally includes a learning identification method (Atsuhiko Noda, Jinichi Nagumo: “System Learning Identification Method”, Measurement and Control, 7, 9, pp. 597-605 (1
968)). Successive correction of hsk by the learning identification method is performed by _{hs k + 1 = hs k +} α (x k e k) / x k T x k (4). However, the e _k = y _k -ys _k, is 0 <α ≦ 1 (5) e k is referred to as residual acoustic echo. Such a calculation operation is performed in the coefficient correction amount calculation circuit 7. The contents of the pseudo impulse response register 9 include a variable coefficient series h.
_sk is stored. α is a coefficient updating gain for determining the sensitivity of estimation, and the closer the correction gain is to 1.0, the greater the amount of correction can be given, and high-speed acoustic reverberation can be removed. It is necessary to change and set according to the line condition. The actual situation is that the determination of the coefficient update gain α currently depends on empirical rules. The coefficient update gain α may be variably controlled according to the magnitude of the residual acoustic reverberation, or may be set in accordance with the room characteristics (for example, Shoji Makino, Nobuo Koizumi: “The adaptation characteristics of the echo canceller in the room sound field. Improvement, "IEICE (A), J71-A, 12, pp. 221-2214 (1988-12)).

When the acoustic reverberation characteristics in a loudspeaker sound field are represented by an FIR type digital filter in this way, a few hundreds of
The variable coefficient series hs _K is divided into several stages because it has a long configuration of up to several thousand taps, and the amount of computation involved in updating the correction amount of the variable coefficient series hs _K becomes enormous and cannot be realized with small-scale hardware. To reduce the amount of update calculation in one step. As an example, the case of the simplest two-partition processing in the divisional update method will be described (for example, Hiroki Furukawa, Takeo Kanamori, Satoru Ibaraki, Hiroyuki Naono, Hiroyuki Tanaka: "Hands-free using division adaptive echo canceller" Telecommunication Circuit, ”Dentsu Spring Meeting, SA-7-8, pp.466-467 (1992). Assuming that the total number of variable coefficient sequences stored in the pseudo impulse response register 9 is N, the division content of the coefficient sequence can be expressed as follows.

Hs1 _k : 0 to (N / 2) -1 hs2 _k : (N / 2) to N The updating algorithm applies the above division range to obtain the equation (4).
More, expressed as _{hs1 k + 1 = hs1 k +} α (x k e k) / x k T x k (6) hs2 k + 1 = hs2 k + α (x k e k) / x k T x k (7) And M is 2, an adaptive algorithm that updates the entire variable coefficient sequence hsk in two steps. Therefore, the amount of calculation in one step can be reduced to 、. Of course, if the number of divisions N is increased, the amount of calculation can be reduced to 1 / N in proportion thereto. However, although the amount of calculation can be reduced, the convergence time for eliminating a certain amount of acoustic reverberation increases. Instead of sequentially updating the divided blocks in order to improve the convergence time, each block is weighted to change the update frequency. As a result, the number of steps M for updating the entire coefficient series
However, the convergence speed in the initial transient response process until the amount of acoustic echo cancellation exceeds 20 dB can be considerably improved. This numerical value of 20 dB is an important threshold value at which a human detects reverberation. That is, if it exceeds 20 dB, it is possible to minimize the deterioration of the sound. It is the impulse response characteristic of the sound field that is used to weight each divided block. Generally, the update process is positively assigned to the one with a smaller time delay of the impulse response.

[0008]

In a digital adaptive filter system in which coefficient updating is performed by using a division method, the convergence speed of the initial transient response process of acoustic reverberation characteristics is inevitably deteriorated as compared with the collective coefficient updating method. However, reverberation is not easily removed, and high-quality voice communication cannot be obtained. This is shown in FIG. In the figure, a represents the acoustic reverberation elimination characteristic when the pseudo impulse response register is divided into two and one of them is alternately selected and updated each time. FIG. 4B shows the same characteristics when the pseudo impulse response register divided into two is subjected to an update process with the division block update frequency set to 3: 1. In the graph b, the convergence characteristic of the initial transient response process is considerably improved, but the convergence speed is deteriorated when the acoustic echo cancellation amount exceeds 30 dB. As described above, when the initial speed is improved, there is a problem that the access speed to the steady state is deteriorated.

The present invention has been made in view of the above points, and eliminates the above-mentioned problems, and has excellent high-speed operation and operation stability.
An object of the present invention is to provide an acoustic reverberation removing apparatus which has high adaptive performance and performs acoustic control while always maintaining a large acoustic echo canceling amount.

[0010]

SUMMARY OF THE INVENTION The present invention has been made to solve these problems, and includes a reception signal input terminal, a reception signal output terminal, a transmission signal input terminal, a transmission signal output terminal, and A variable coefficient digital filter having a pseudo impulse response register divided into N blocks to which a reception signal of the reception signal input terminal is input, and n out of a total number N of divided blocks of the pseudo impulse response register (n 1 <
a coefficient sequence block selecting circuit for selecting n <N), and a variable coefficient digital filter that generates from the acoustic reverberation component of the received signal input from the received signal output terminal to the transmission signal input terminal via the acoustic reverberation path. The acoustic impulse response register divided into N pieces in an acoustic reverberation removing apparatus in which a coefficient sequence is successively updated by a coefficient correction amount operation circuit that minimizes a residual signal obtained by subtracting the obtained pseudo acoustic echo. Is calculated for each divided block, and the number of divided blocks N ′ (1 <N ′ <N) to be subjected to coefficient update processing based on the calculated coefficient accumulated average power value Variably controlled acoustic echo canceller.

[0011]

According to the present invention, since the high speed and high stability of the estimating operation are ensured by the above-mentioned means, reverberation components are hardly mixed into the communication line, the deterioration of the sound quality of the communication voice is prevented, and the communication itself is prevented. This can reduce the risk of howling occurrence, which makes it impossible to perform sound control, and enables high-quality sound control.

[0012]

Embodiments of the present invention will be described below in detail with reference to the drawings. FIG. 1 is a block diagram showing a configuration of an acoustic reverberation removing apparatus according to the present invention. As shown in FIG. 1, according to the present invention, a conventional reception signal input terminal 1, a reception signal output terminal 3, a variable coefficient digital filter 3, a transmission signal input terminal 4, a subtraction circuit 5, a transmission signal output terminal 6, a coefficient An acoustic reverberation removing apparatus employing a learning identification method as an adaptive algorithm comprising a correction amount calculating circuit 7, a received signal input register 8, a pseudo impulse response register 9, a product-sum calculating circuit 10, and a coefficient sequence block selecting circuit 11; In the device of the same configuration,
The configuration is such that a cumulative addition average power calculation circuit 12, a block average cumulative power calculation circuit 13, a block power growth rate calculation circuit 14, and a divided block number setting circuit 15 are added. The reception signal input terminal 1, the reception signal output terminal 2, the transmission signal input terminal 4, and the transmission signal output terminal 6.
And the variable coefficient digital filter 3 having the pseudo impulse response register 9 divided into N blocks to which the reception signal of the reception signal input terminal 1 is inputted, and the total number of divided blocks of the pseudo impulse response register 9 Out of N,
the coefficient sequence block selecting circuit 11 for selecting n (1 <n <N), and the acoustic echo of the received signal input from the received signal output terminal 2 to the transmission signal input terminal 4 via the acoustic echo path Each time the coefficient correction amount calculating circuit 7 selects the coefficient sequence block selecting circuit 11 by the coefficient correction amount calculating circuit 7 so as to minimize the residual signal obtained by subtracting the pseudo acoustic echo generated by the variable coefficient digital filter 3 from the component. In an acoustic reverberation removing apparatus in which n divided blocks are sequentially updated, only the variable coefficients stored in the n divided blocks of the pseudo impulse response register 9 selected by the coefficient sequence block selecting circuit 11 are cumulatively added. An accumulative average power calculation circuit for calculating an average power; And the block mean cumulative power calculating circuit 13 for calculating the table mean cumulative addition average power, the selected n
The block power fluctuation rate calculation circuit 14 for calculating the growth rate of the average cumulative addition average power of the divided blocks independently, and the coefficient update corresponding to the growth rate of each average cumulative addition power of the selected n divided blocks The number of divided blocks N ′ (1 <N ′ <N) for switching the number of divided blocks to be processed is determined by the divided block number setting circuit 15, and the total number of divided blocks to be updated is determined, and the divided update scheduling corresponding to the total number of divided blocks is performed. The acoustic reverberation removing device performs acoustic control while setting the coefficient correction amount arithmetic circuit 7. The block average cumulative power calculation circuit 13 calculates the average power PBN of each block of the cumulative addition average power of the variable coefficients stored in the pseudo impulse response register according to equation (6). The calculation for calculating the average power P _BN is as follows:
Executed only when each block is updated. That is, since the processing is performed for one block each time, the calculation amount does not need to be increased much. Further, the size of each block is uniquely determined by the scale of the constituting hardware.

P _B1 = (hs ₀ ² +... + _{Hs m1-1} ² ) / m1 P _B2 = (hs _m1 ² +... + _{Hs m2-1} ² ) / (m ² _−m 1) (6) P _BN = (Hs _mN ² + ... + hs _L-1 ² ) / (L-mN) L is the total number of variable coefficients, m1, m2,..., MN are coefficient numbers to be divided, and N is the total number of divided blocks. is there. Here, the average for each block is used as the evaluation value. However, as long as the value can represent the block, the average value may be, for example, the maximum value or the minimum value of the coefficient cumulative average power, instead of this value. The configuration of the block average accumulated power calculation circuit 13 is shown in FIG. 2 taking the 0th block as an example.

Using the average power PBN, the block power growth rate calculation circuit 14 calculates using the equation (7).

DP _BN (K + 1) = | P _BN (K + 1) −P _BN (K) | / P _BN (K) (7) The growth rate shown in the equation (7) is an example and is determined. It is not a typical calculation method. For example, the numerator may be a present value instead of a past value. Also, the present invention functions effectively even if a difference value between dP _BN (K + 1) and dP _BN (K) is used. The relationship between (K) and (K + 1) described here is before and after coefficient updating is performed.

The divided block number setting circuit 15 compares the value of the representative growth rate dPBN of one selected divided block with a threshold value inserted into the circuit, and updates the divided processing target division corresponding to a suitable threshold range. An operation for determining the total number N 'of blocks is performed. FIG. 3 shows this conceptual diagram. For example,
Assuming that the total number of divisions of the pseudo impulse response register 9 is 4, N '= 2 in the first step and N' in the next step.
= 3 and N '= 4 in the last step. And
In one update, the update process is performed only for the coefficients of two blocks, that is, in the first step, all the set coefficients are set, in the second step, the set 2/3 coefficients, and in the third step, the set coefficients are set. The coefficient of 2/4 is updated. The second and third steps are divided update, and the schedule inside can be freely set. On this occasion,
The update operation is performed in all steps in half of the total coefficient, and the amount of operation is reduced. FIG. 4 shows a result a of the adaptive processing operation according to the present invention when white noise is used as a reference input. As a comparison target, the pseudo impulse response register is divided into two, and the result is shown in the result b of the model that has been subjected to the updating process alternately. Both models have the same number of operations related to one coefficient update. The vertical axis is the amount of acoustic echo cancellation, and the horizontal axis is time. Compared to b in the figure, a is 2 in the initial response process.
It achieves more than twice the adaptation speed. After that, the method according to the present invention is superior without changing the relative relationship.

Similarly, FIG. 5 shows the result of comparison with the case where the number of updates is 3: 1. In the figure, a is the figure according to the present invention.
Is the case where is fixed to 3: 1. While maintaining the initial convergence speed, which is a very important performance as an acoustic reverberation removing device, adaptation is performed without slowing down the convergence speed even when the acoustic reverberation amount exceeds 30 dB. Compared with a in the figure, b is approximate when the sound elimination amount is 25 to 30 dB, but thereafter the characteristics gradually begin to deteriorate, and a longer time is required to reach a steady state. The detection limit at which humans have a feeling of reverberation is about 20 dB in terms of the amount of acoustic echo cancellation. Therefore, the key point is how fast the acoustic reverberation device erases up to this value. In the method b in the figure, the update frequency is changed from 3: 1 to 5:
When it is set to 1, the initial convergence speed is slightly improved, but the subsequent adaptation speed is getting worse. As described above, the conventional method of simply performing the operation by weighting the update frequency cannot improve the performance.

4 and 5, it can be said that the present invention is an excellent system in which the adaptive speed of the initial transient response process, which is important as an acoustic reverberation device, and the high adaptive convergence speed thereafter are compatible.

[0019]

As described above, according to the present invention,
The following excellent effects are expected. (1) By using the present invention, high speed and high stabilization can be realized at the same time, so that high quality voice communication can be maintained and the danger of howling can be suppressed. (2) In the adaptive operation process, the calculation processing of the correction amount related to the coefficient update is greatly reduced, so that the load on the hardware can be reduced.

(3) Since the adaptation speed of the initial transient response process of the acoustic reverberation elimination characteristic can be maintained at a high level, an audible reverberation can be removed at a high speed, and a high quality voice communication space can be realized. it can.

(4) Since a process adapted to the growth of the coefficient is performed, the variable coefficient is not updated more than necessary. Therefore, when a digital signal processor or the like is used, a calculation error and a malfunction are less likely to occur.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a configuration example of an acoustic reverberation removing apparatus according to the present invention.

FIG. 2 is a block diagram showing one configuration of a block average accumulated power calculation circuit used for describing the present invention.

FIG. 3 is a block diagram illustrating a concept of a state determination control unit.

FIG. 4 is a diagram showing an example of acoustic echo cancellation characteristics when white noise according to the present invention is used as a reference input.

FIG. 5 is a diagram showing an example of an acoustic reverberation elimination characteristic when white noise according to the present invention is used as a reference input.

FIG. 6 is a diagram showing an example of acoustic echo cancellation characteristics according to a conventional method when white noise is used as a reference input.

FIG. 7 is a block diagram showing an example of a basic configuration of a conventional acoustic reverberation removing apparatus using a general learning identification method.

[Explanation of symbols]

 REFERENCE SIGNS LIST 1 reception signal input terminal 2 reception signal output terminal 3 variable coefficient filter 4 transmission signal input terminal 5 subtraction circuit 6 transmission signal output terminal 7 correction amount operation circuit 8 reception signal input register 9 pseudo impulse response register 10 product-sum operation circuit 11 Coefficient sequence block selection circuit 12 Cumulative average power calculation circuit 13 Block average cumulative power calculation circuit 14 Block power fluctuation rate calculation circuit 15 Division block number setting circuit

Claims (1)

(57) [Claims] 1. A receiving signal input terminal, a receiving signal output terminal, a transmitting signal input terminal, a transmitting signal output terminal, and N blocks each of which receives a receiving signal of the receiving signal input terminal. A variable coefficient digital filter having a pseudo impulse response register, a coefficient sequence block selecting circuit for selecting n (1 <n <N) out of a total of N divided blocks of the pseudo impulse response register, The residual signal obtained by subtracting the pseudo acoustic echo generated by the variable coefficient digital filter from the acoustic echo component of the received signal input to the transmission signal input terminal via the acoustic echo path from the output terminal is minimized. In the acoustic reverberation removing apparatus in which the coefficient sequence is sequentially updated by the coefficient correction amount calculating circuit, each of the divided blocks of the pseudo impulse response register divided into N is used. Calculating a number cumulative addition average power value,
Based on the calculated coefficient cumulative average power value, the number of divided blocks N ′ (1 <
N '<N) is variably controlled.