JPH07321716A - Device for eliminating sound echo - Google Patents

Abstract

PURPOSE:To provide a sound echo eliminating device in which the deterioration in the performance of operation stability is improved by using different division update block mapping in a coefficient correction quantity division update processing. CONSTITUTION:In the division update processing where a coefficient series stored in a pseudo impulse response register 9 is divided into N and the entire coefficient series is updated in the total step numbers of M, the sound echo eliminating device uses a coefficient correction quantity arithmetic operation circuit 7 to apply arithmetic operation processing to each block according to a block mapping decided by mutual power ratio of pseudo impulse response in the divided blocks by using P-sets of position fixed blocks and Q-sets of position variable blocks used in common.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention is used in a communication line, a room sound field control device, and a high-quality voice communication conference device, and an acoustic echo component in which a signal on a receiving path appears in a transmitting path via an acoustic echo path. The present invention relates to an acoustic echo canceller that removes noise.

[0002]

2. Description of the Related Art Generally, an acoustic echo canceller is used in communication hygiene and long-distance telephone lines using a submarine cable.
It can be roughly divided into one that removes reflection caused by impedance mismatch of the line-to-four-line converter and one that removes reverberation due to acoustic coupling of speaker's voice in a loudspeaker telephone such as a video conference system. It is composed of a variable coefficient filter and a subtraction circuit that generate pseudo-acoustic echo. The basic operation of the acoustic echo canceller will be described below.

FIG. 6 is a diagram showing the basic structure of an acoustic echo canceller. 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 pseudo acoustic echo. The transmission signal from the transmission signal input terminal 4 and the pseudo-acoustic echo that is the output of the variable coefficient filter 3 are input to the subtraction circuit 5, the acoustic echo 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 sum of products 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 sum of products circuit 10. Sum of products 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 to four-wire converter in the case of a long-distance telephone line, and to a speaker and a microphone in the case of a public telephone system.

Assuming that the signal propagation characteristic of the acoustic echo path is linear and represented by an FIR type digital filter, if the impulse response h (t) and the input received signal x (t) are used, the sampling time interval is Is T, and the acoustic echo y _{k at} time kT is represented by y _k = h′x _k (1). However, h = [h ₁ , h ₂ , ..., H _n ] '(2) x = [x _k-1 , ..., x _kn ]'': transposition of the vector.

On the other hand, the estimated value of h at time kT is h
If s _k , the estimated value ys _k of y _k is given by ys _k = hs _k ′ x _k (3). In the acoustic echo canceller, when there is a voice signal in the reception signal input terminal 1 and there is no voice signal in the transmission signal input terminal 4 and only acoustic echo exists, the echo elimination operation is performed as an adaptive operation state. This adaptive motion algorithm is generally a learning identification method (Atsuhiko Noda, Jinichi Nagumo: “Learning Identification Method of System” Measurement and Control, 7, 9pp597-6).
05 (1968)) is adopted. H by learning identification method
The iterative modification of s _k is performed by hs _{k + 1} = hs _k + α (x _k e _k ) / x _{k'x k} (4). However, e _k = y _k −y s _k 0 <α ≦ 1 (5), and e _k is called residual acoustic echo. Such a calculation operation is processed in the coefficient correction amount calculation circuit 7. The contents of the pseudo impulse response register 9 include the variable coefficient series h.
s _k is stored. α is a correction loop gain for determining the sensitivity of estimation, and a larger correction amount can be given as the value is closer to 1.0, and high-speed acoustic echo removal is possible. However, in actual use, near-end noise and It is necessary to change the setting depending on the line status. At present, the actual determination of the modified loop gain α depends on an empirical rule.

When the acoustic reverberation characteristic in the loud sound field is expressed by the FIR type digital filter in this way, several hundreds
Becomes very long construction of to several 1000 taps, since the calculation amount involved in the correction amount updating of the variable coefficient series hs _k can not be realized in a small hardware becomes enormous, division processing a variable coefficient series hs _k in several stages Is performed to reduce the update calculation amount in one step. As an example, the case of the simplest two-division processing in the division updating method will be described. When the total number of variable coefficient sequences stored in the pseudo impulse response register 9 is N, the division contents of the coefficient sequence can be expressed as follows.

Hs1 _k : 0 to N / 2 hs2 _k : (N / 2) +1 to N The updating algorithm applies the above division range to formula (4).
More, expressed as _{hs1 k + 1 = hs1 k +} α (x k e k) / x k 'x k (6) hs2 k + 1 = hs2 k + α (x k e k) / x k' x k (7) things can be, M is 2, that is, an adaptive algorithm to update the entire variable coefficient series hs _k in two steps. Therefore, the calculation amount in one step can be reduced to 1/2, and of course, if the division number N is increased, the calculation amount can be reduced to 1 / N in proportion to the increase. However, although the amount of calculation can be reduced, the convergence time for eliminating a certain amount of acoustic reverberation becomes long. In order to improve the convergence time, each divided block is not updated in order, but each block is weighted to change the update frequency. As a result, the number of steps M for updating the entire coefficient series increases, but the convergence speed can be considerably improved. The impulse response characteristic of the sound field is used to weight each divided block. FIG. 5 shows an example of the impulse response characteristics of the sound field observed using the maximum period sequence code. It can be seen that the characteristic exhibits a damping characteristic. By using this characteristic, the convergence speed of the impulse response can be improved by using the position fixed block that prioritizes the part where the power of the impulse response is concentrated and the position change block that performs the update process in combination with the other part. Improvements are being made.

[0008]

When the weighted division update method is used for updating the coefficient correction amount, the basis of the weighting is that the impulse response of the sound field exhibits an attenuation characteristic. That is, it is assumed that the damping characteristic of the impulse response is exponentially decayed, and the update frequency is set in accordance with it. However, the actual impulse response is not uniform and varies considerably depending on the environment of any sound field. The environment is the volume and shape of the room used, the walls, floors and ceiling members that make up the room, the distance between the microphone and the speaker, and the installations surrounding them. That is, in an arbitrary sound field, the ratio of the direct sound and the indirect sound greatly differs. If the acoustic reverberation characteristic generated from this environment is treated as a simple exponential decay and the update frequency is fixed, the acoustic reverberation characteristic is deteriorated, such as a decrease in the convergence speed and a reduction in the amount of elimination. As a result, there has been a problem that noise on the communication line is increased or the risk of howling is increased, which lowers the operation stability.

The present invention has been made in view of the above points, eliminates the above-mentioned problems, is excellent in operation stability, and has high followability with respect to changes in acoustic echo paths, and high-speed audio. An object of the present invention is to provide an acoustic echo canceller that realizes echo canceling characteristics and performs acoustic echo control of a sound field while always maintaining a large acoustic echo canceling amount.

[0010]

DISCLOSURE OF THE INVENTION The present invention is intended to solve these problems and comprises a reception signal input terminal, a reception signal output terminal, a transmission signal input terminal, and a transmission signal output terminal. , N that receives the reception signal of the reception signal input terminal
Variable coefficient digital filter having a pseudo impulse response register divided into a number of blocks, and a coefficient sequence for selecting the number of blocks (n) to be updated at a time from the total number N of divided blocks of the pseudo impulse response register The block selection circuit and the pseudo acoustic echo generated by the variable coefficient digital filter are subtracted from the acoustic echo component of the reception signal input to the transmission signal input terminal through the acoustic echo path from the reception signal output terminal and left. In the acoustic echo canceller in which the coefficient sequence is sequentially updated by the coefficient correction amount calculation circuit for minimizing the difference signal, a first power calculation circuit for calculating the power of each coefficient stored in the pseudo impulse response register, , A power ratio for aggregating the power values for each block output by the first power computing circuit and comparing the aggregated power values Circuit, and the update frequency of each divided block adapted to an arbitrary sound field characteristic is determined based on the comparison result of the power comparison circuit, and P position fixing blocks and Q (n-P) are determined according to the update frequency. ) Provide an acoustic echo canceller that corrects coefficients by using a plurality of position variation blocks.

Further, each coefficient correction amount generated by the correction amount calculation circuit of the position fixed block having the highest update frequency among the update frequencies adapted to arbitrary sound field characteristics based on the power comparison circuit. A second power operation circuit for obtaining the power value of, and a coefficient division circuit for dividing each power value output by the second power operation circuit by each output of the first power operation circuit of the position fixed block, A coefficient variation detection circuit that compares the calculation result calculated by the coefficient division circuit with a predetermined threshold value;
The acoustic echo canceller according to claim 1, wherein when the output value of the coefficient division circuit is larger than the threshold value in the coefficient fluctuation detection circuit, the position fluctuation block performs coefficient correction calculation processing of the same divided block as the position fixed block.

[0012]

According to the present invention, since the division processing for updating the coefficient correction amount adapted to the impulse response characteristic of an arbitrary sound field can be set by the above means, the deterioration of the voice quality and the howling due to the reduction of the acoustic echo canceling amount on the communication line can be set. It is possible to prevent obstacles to operation stability such as increase in risk of occurrence, to realize high-speed and stable acoustic echo removal while significantly reducing the amount of calculation related to updating, and to have arbitrary acoustic echo path characteristics. Since high followability is ensured even when there is a change, the convergence speed to the steady state is excellent, and high-performance acoustic control is possible.

[0013]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing the configuration of a first acoustic echo canceller of the present invention. As shown in FIG. 1, according to the present invention, the conventional reception signal input terminals 1, 3 reception signal output terminals 2, variable coefficient digital filter 3, transmission signal input terminal 4, subtraction circuit 5, transmission signal output terminal 6, coefficients are used. Correction amount calculation circuit 7, reception signal input register 8, pseudo impulse response register 9,
A first power calculation circuit 12 and a power comparison circuit 13 are provided in a device having the same configuration as the acoustic echo canceling device that employs a learning identification method as an adaptive algorithm and is configured by a product-sum calculation circuit 10 and a coefficient sequence block selection circuit 11. The second power calculation circuit 14, the coefficient division circuit 15, and the coefficient variation detection circuit 16 are added. The reception signal input terminal 1, the reception signal output terminal 2, the transmission signal input terminal 4, the transmission signal output terminal 6, and the reception signal input from the reception signal input terminal 1 are input. The variable coefficient digital filter 3 and the variable coefficient digital filter 3
Of the (L-1) coefficient sequences of the pseudo impulse response register 9 and the sum of products for performing the convolution integral operation of the contents of the pseudo impulse response register 9 and the input signal from the reception signal input terminal 1. The arithmetic circuit 10, the subtraction circuit 5 for taking the difference value between the pseudo acoustic echo generated by the product-sum arithmetic circuit 10 and the acoustic echo input from the transmission signal input terminal 4, and the variable coefficient digital filter 3 Is a coefficient correction amount calculation circuit 7 for adding a correction amount to the coefficient series of the pseudo impulse response register 9 so as to supply an approximate value of the acoustic echo, and the coefficient series of the pseudo impulse response register 9 is N The block is divided into blocks, and a command for sequentially selecting each block and performing a coefficient update operation is sent to the coefficient correction amount calculation circuit 7 so that the entire coefficient sequence is automatically updated with the total number of steps M times. In the acoustic echo removing apparatus composed of a sequence block selection circuit 11, the pseudo each coefficient stored in the impulse response register 9 power pl (l =
0, 1, ..., L-1), and the sum of the outputs of the first power calculation circuit 12 and the power calculation circuit 12 is calculated for each block, and the total power hpn (n = 0, 1, , ...
., N-1) and a power comparison circuit 13 for comparing
Using the comparison result of the power comparison circuit 13, a method of determining the update frequency of each divided block adapted to an arbitrary sound field characteristic is L = 20.
00, N = 4, n = 2, P = 1, and Q = 1 will be described as an example. Hp0 = p0 + p1 + ... + p499 (8) hp1 = p500 + p501 + ... + p999 hp2 = p1000 + p1001 +. .. + p1499 hp3 = p1500 + p1501 + ... + p1999 The electric power calculation circuit 12 uses the total electric power of each block calculated by the equation (8), and the electric power comparison circuit 13 compares the total electric power. For example, if the comparison result has the relationship as shown in Expression (9) in the ideal state, hp0>hp1>hp2> hp3 (9) The maximum update frequency is given to the first block and the first block is set as the fixed position block. Update process every time. Hereinafter, the second to fourth blocks are set as position change blocks, and the update frequency is distributed. The frequency is determined based on the power difference between the blocks. That is, a threshold value may be set for each block power ratio. Then, the equation (9) is determined by performing this threshold comparison. Of course, these evaluations are performed by the power comparison circuit 13. The evaluation result of the power comparison circuit 13 is transferred to the coefficient sequence block selection circuit 11,
The internally stored divided update block mapping is selected. Equation (10) is an example where an ideal state like Equation (9) is not obtained.
Shown in.

Hp1 >> hp2 >> hp3 >> hp0 (10) This relationship represents a condition in which there is a very large initial delay. Therefore, the position-fixed block is the second block, and the others are assigned the update frequency as in the case of the equation (9).

FIG. 2 shows the division update block mapping set by the equation (9). This block mapping may be modeled based on the ratio of the direct sound and the indirect sound in an arbitrary sound field, and may be stored in the coefficient sequence block selection circuit 11 in advance. According to the block mapping adapted to this arbitrary sound field characteristic, P position-fixed blocks and Q (n
-P) An acoustic echo canceller characterized in that coefficient correction is performed by using position change blocks.

Although the initial position of each processing block at this time is arbitrary, the pseudo impulse response register 9 may be divided into the first half and the latter half for half-updating, or may be started from the ideal state of the equation (9). .

FIG. 3 shows acoustic echo cancellation characteristics according to the present invention. In the figure, a) shows that the pseudo impulse response register is divided into two, and the updating process is applied alternately in the first and second halves. In the diagram b), the update processing is applied at a ratio of the first half block divided into two three times and the second half block once. C)
Applies the method according to claim 1 of the present invention. The number L of the pseudo impulse response registers at this time was 2000. Also, the signal-to-near-end noise ratio was infinite. Acoustic Echo Cancellation (ERLE) to achieve high quality voice communication
Since about 40 dB is required, the a) method and the present invention c) method are equivalent when comparing the respective methods, but the b) method deteriorates the convergence time by about 1 sec. Overall, the method c) of the present invention is superior to the other methods in initial convergence. As the number of divided blocks increases, the difference between the method c) of the present invention and the other methods becomes more remarkable. Of course, the calculation amount of each method is the same.

Further, among the update frequencies adapted to the arbitrary sound field characteristics determined by the power comparison circuit 13, the position fixed block having the highest update frequency, here, the coefficient correction amount calculation circuit of the first block is calculated. 7. The second power calculation circuit 14 for obtaining the power pn (n = 0, 1, ..., 499) of each coefficient correction amount generated in 7, and each output Δpn (n of the second power calculation circuit 14 = 0, 1, ..., 499) is the output pn in the position fixed block of the first power operation circuit 12.
Coefficient division circuit 1 for division by (n = 0, 1, ..., 499)
5, p0 / p0 = R0 (11) p1 / p1 = R1 || Δp499 / p499 = R499 The result Rn of the coefficient division circuit 15 (n = 0, 1, ...,
499) to a predetermined threshold value Sn (n = 0, 1, ..., 49)
9) and the coefficient fluctuation detection circuit 16 to be compared with the coefficient fluctuation detection circuit 16 while the output value Rn (n = 0, 1, ..., 499) is larger than the threshold value S in the coefficient fluctuation detection circuit 16 2. The acoustic echo canceller according to claim 1, wherein the block performs the coefficient correction calculation processing of the same divided block as the position fixed block.

By performing the coefficient correction calculation processing on the divided blocks in which the position fixed block and the position variable block are the same,
It is intended to quickly adapt to rapid changes in direct sound. Actually, the sudden change in the impulse response of the sound field is due to the movement of the human body and the accompanying substances near the direct sound. Therefore, by preferentially correcting the coefficient sequence in the position fixed block that covers the part where the power concentration is large,
It is possible to improve the tracking performance when the acoustic echo path changes.

FIG. 4 shows the acoustic echo canceling characteristics when a sudden variation in the acoustic echo path occurs according to the present invention. In the figure, a) shows that the pseudo impulse response register is divided into two, and the updating process is applied alternately in the first and second halves. In the figure, b) is the system according to claim 1 of the present invention. In the figure, c) is the system according to claim 2 of the present invention. The number L of the pseudo impulse response registers at this time was 2000. The signal-to-near-end noise ratio was set to 30 dB to facilitate comparison of convergence speeds. And
The content of the path change is assumed to be a large direct sound change. It can be seen that there is a convergence speed difference of 5 times or more after the route change in the methods b) and c). From this, it can be seen that the method of the present invention is excellent in adaptive performance with respect to route fluctuation.

[0021]

As described in detail above, according to the present invention, the following excellent effects are expected.

(1) Since it is possible to perform divided block mapping adapted to arbitrary sound field characteristics, it is possible to correct deterioration of the convergence speed of the acoustic echo canceling characteristics due to the division processing for updating the coefficient correction amount, and to speed up acoustic echo removal. Can be realized. This thing is
It is very effective for a loud voice communication device in which the ratio of direct sound and indirect sound is easily changed.

(2) Without degrading the acoustic echo canceling performance,
The internal calculation amount of the adaptive algorithm can be significantly reduced. Further, since a new training mechanism for estimating an arbitrary sound field characteristic is not required, hardware can be realized with a smaller configuration than the conventional method, and the cost can be reduced.

(3) Since the position-fixed block, which is a divided block having a high update frequency in an arbitrary sound field characteristic, is always set in the portion of the impulse response where high power is concentrated, even if the number of divided blocks is increased. Acoustic echo removal processing can be performed without degrading the convergence speed.

(4) The variation factor of the acoustic echo path characteristic is the movement of the human being or the accompanying object in the space where the direct sound is predominant close to the microphone and the speaker. Therefore, in the present invention in which not only the position-fixed block that covers the direct sound portion when the acoustic echo path changes but also the position-change block covers the same block, the rising speed of the acoustic echo cancellation characteristic is high, so The followability is very strong and the steady state of the communication line can be quickly created.

(5) Since the amplitude fluctuation of the error signal due to erroneous erasure is very small, the quasi-steady state is maintained, and a relatively large level of residual acoustic echo does not exist on the communication line, so high-speed two-way communication detection is possible. It can be easily performed, and the voice deterioration such as the cut off of the head of the transmitted voice is eliminated and high sound quality is secured.

[Brief description of drawings]

FIG. 1 is a block diagram showing a configuration example of an acoustic echo canceller according to the present description.

FIG. 2 is a diagram showing state evaluation combinations for determining divided update block mapping used in this description.

FIG. 3 is a diagram showing an example of acoustic echo canceling characteristics when white noise according to the present invention used in the present description is used as a reference input.

FIG. 4 is a diagram showing an example of acoustic echo cancellation characteristics when there is a rapid acoustic echo path variation, with white noise according to the present invention used in the present description as a reference input.

FIG. 5 is a diagram showing an example of impulse response characteristics of a sound field observed by the maximum period sequence code used in this description.

FIG. 6 is a block diagram showing an example of a basic configuration of an acoustic echo canceller using a conventional general learning identification method.

[Explanation of symbols]

 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 calculation circuit 8 reception signal input register 9 pseudo impulse response register 10 sum-of-products calculation circuit 11 Coefficient sequence block selection circuit 12 First power calculation circuit 13 Power comparison circuit 14 Second power calculation circuit 15 Coefficient division circuit 16 Coefficient fluctuation detection circuit

Claims (2)

[Claims] 1. A reception signal input terminal, a reception signal output terminal, a transmission signal input terminal, a transmission signal output terminal, and N blocks into which a reception signal from the reception signal input terminal is input. A variable coefficient digital filter having a pseudo impulse response register, a coefficient sequence block selection circuit that selects the number of blocks (n) to be updated at one time from the total number N of divided blocks of the pseudo impulse response register, The residual signal is minimized by subtracting the pseudo-acoustic echo generated by the variable coefficient digital filter from the acoustic echo component of the incoming voice signal input to the outgoing voice signal input terminal from the incoming voice signal output terminal through the acoustic echo path. In the acoustic echo canceller in which the coefficient series is sequentially updated by the coefficient correction amount calculation circuit for, the power of each coefficient stored in the pseudo impulse response register Comparison of a first power calculation circuit to be obtained, a power comparison circuit that totals the power values of each block output by the first power calculation circuit, and compares the totaled power values, and the power comparison circuit Based on the result, the update frequency of each divided block adapted to an arbitrary sound field characteristic is determined, and according to the update frequency, the P fixed block and the Q (n−P) position variable block are used as coefficients. An acoustic echo canceller characterized by being modified. 2. A coefficient correction amount generated by the correction amount calculation circuit of a position fixed block having the highest update frequency among update frequencies adapted to an arbitrary sound field characteristic based on the power comparison circuit. A second power operation circuit for obtaining the power value of, and a coefficient division circuit for dividing each power value output by the second power operation circuit by each output of the first power operation circuit of the position fixed block, The position variation block is composed of a coefficient variation detection circuit that compares the calculation result calculated by the coefficient division circuit with a predetermined threshold value, and the position variation block is positioned when the output value of the coefficient division circuit is larger than the threshold value in the coefficient variation detection circuit. 2. The acoustic echo canceller according to claim 1, wherein coefficient correction calculation processing is performed on the same divided block as the fixed block.