JP3152825B2 - 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 that removes sound.

[0002]

2. Description of the Related Art In general, acoustic echo cancellers are used for long-distance telephone lines using communication sanitation and submarine cables.
It can be broadly divided into one that removes reflection caused by impedance mismatch of the line-to-wire converter and one that removes reverberation due to acoustic coupling of speaker's voice in a loudspeaker such as a video conference system. It is composed of a variable coefficient filter for generating 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′x _k (1). Here, h = [h ₁ , h ₂ ,..., H _n ] ′ (2) x = [x _k−1 ,..., X _kn ] ′ “′”: 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 '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. Generally, a learning identification method is adopted as the adaptive operation algorithm. Successive correction of hs _k by the learning identification method is performed by _{hs k + 1 = hs k +} α (x k e k) / x k '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 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, so that high-speed acoustic reverberation can be removed. It is necessary to change the settings depending on the line status. The actual situation is that the determination of the modified loop gain α currently depends on empirical rules.

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 sequence hsk is divided into several stages because it has a long configuration of up to several thousand taps, and the amount of operation involved in updating the correction amount of the variable coefficient sequence hsk becomes enormous and cannot be realized with small-scale hardware. A method of reducing the amount of update calculation in one step is adopted. As an example, a case of the simplest two-division processing in the division update method will be described. 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 hs2 _k : (N / 2) +1 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 'x k (6) hs2 k + 1 = hs2 k + α (x k e k) / x k' 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 can be considerably improved.
It is the impulse response characteristic of the sound field that is used to weight each divided block. FIG. 2 shows an example of an impulse response characteristic of a sound field observed using a sequence code. It can be seen that the characteristic exhibits an attenuation characteristic. Using this characteristic, the convergence speed is improved by performing update processing with priority given to the first half of the impulse response where power is concentrated.

[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 attenuation characteristic of the impulse response is exponentially attenuated, and the update frequency is set accordingly. However, the actual impulse response is not uniform and varies considerably depending on the environment of the sound field. The environment is the volume and shape of the room used, the walls that make it up,
The floor and ceiling members, the distance between the microphone and the speaker, and the objects surrounding them. If the acoustic reverberation characteristics generated from this environment are treated as simple exponential decay and the update frequency is fixed, the acoustic reverberation elimination characteristics such as a reduction in the convergence speed and a reduction in the amount of erasure will be deteriorated. As a result, there has been a problem that noise on the communication line is increased, and the danger of howling is increased, thereby lowering operation stability.

SUMMARY OF THE INVENTION The present invention has been made in view of the above problems, and eliminates the above-mentioned problems, has excellent operation stability, has high followability with respect to fluctuations in an acoustic reverberation path, and has a high-speed sound. It is an object of the present invention to provide an acoustic reverberation removing device which realizes acoustic reverberation characteristics and controls acoustic reverberation of a sound field while maintaining a large acoustic reverberation amount at all times.

[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 processed by N divided coefficient blocks input from the reception signal input terminal; and a transmission signal input terminal from the reception signal output terminal via an acoustic echo path to the transmission signal input terminal. An acoustic reverberation removing apparatus in which the division coefficient is sequentially updated so as to minimize a residual signal obtained by subtracting a pseudo acoustic reverberation signal generated by the variable coefficient digital filter from an acoustic reverberation component of an input reception signal. In the above, each switch means is provided between the reception signal input terminal and the reception signal output terminal or the transmission signal input terminal and the transmission signal output terminal, A code generating circuit for generating a code for observing the acoustic reverberation characteristic of the external environment in a state where the switch means is disconnected from the transmission / reception with the destination terminal, and the code is transmitted from the reception signal output terminal via the acoustic reverberation path to the code. A calculating means for calculating the acoustic reverberation characteristic of the external environment inputted to the transmission signal input terminal, a storing means for storing the calculation result, and the switch means being connected to the destination terminal after a predetermined time has elapsed. At this time, the total power calculation means for calculating the total power for each block based on the storage data of the storage means, the comparison means for comparing the total power value between each adjacent block, and the comparison means There is provided an acoustic reverberation removing apparatus including: an updating unit configured to arbitrarily set an update frequency of each division coefficient according to a magnitude relationship between a comparison result and an interpolation threshold.

[0011]

According to the present invention, the above-mentioned means can set the division processing for updating the coefficient correction amount corresponding to the impulse response characteristics of the actual sound field, so that operation stability such as an increase in background noise on the communication line and an increase in the danger of howling occurs. High-speed and stable acoustic reverberation elimination while significantly reducing the amount of computation involved in updating, and high tracking ability even when the acoustic reverberation path characteristics fluctuate arbitrarily. Since it is ensured, the convergence speed to the steady state is excellent, so that a high level of acoustic echo cancellation can always be maintained at a high level, and high-performance acoustic control can be performed.

[0012]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 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 a first 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, reception signal output terminal 2, variable coefficient digital filter 3, transmission signal input terminal 4, subtraction circuit 5, transmission signal output terminal 6, coefficient correction Quantity operation circuit 7,
An apparatus having the same configuration as an acoustic reverberation removing apparatus employing a learning identification method as an adaptive algorithm including a reception signal input register 8, a pseudo impulse response register 9, a product-sum operation circuit 10, and a coefficient block selection circuit 11, The generation circuit 12, the first selection switch 13, the synchronous addition circuit 14, the second selection switch 15, the linear convolution integral operation circuit 16, the observation impulse response register 17, the state selection circuit 18, and the block power evaluation circuit 19 The configuration has been 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 The variable coefficient digital filter 3 to be input, the pseudo impulse response register 9 storing the coefficient series of the variable coefficient digital filter 3, the contents of the pseudo impulse response register 9, and the input signal from the reception signal input terminal 1. The product-sum operation circuit 10 for performing a convolution integral operation with
The subtraction circuit 5 for taking a difference value between the pseudo acoustic reverberation generated by the product-sum operation circuit 10 and the acoustic reverberation input from the transmission signal input terminal 4 and the variable coefficient digital filter 3 are used to calculate the acoustic reverberation. The coefficient correction amount calculation circuit 7 for adding a correction amount to the coefficient sequence of the pseudo impulse response register 9 so as to supply an approximate value, and the coefficient sequence of the pseudo impulse response register 9 are divided into N blocks, The coefficient block selection circuit 11 sends an instruction for sequentially selecting each block and performing a coefficient update operation to the coefficient correction amount calculation circuit 7 so that the entire coefficient sequence is automatically updated with the total number of steps M.
And a code generation circuit 12 for generating and generating a sequence code that is completely uncorrelated with the received signal input from the received signal input terminal 1, and an input to an input terminal a. A first selection switch 13 for outputting one of a received signal and the sequence code input to the input terminal b from the received signal output terminal 2, and an input from the transmitted signal input terminal 4 via the acoustic reverberation path The synchronous addition circuit 14 for calculating an impulse response of the obtained sequence code;
Output terminal a and output terminal b
The second selection switch 15 that switches between the output of the synchronous addition circuit 14 and the sequence code to calculate the impulse response of the acoustic reverberation path by performing convolution integration of the sequence code; An observation impulse response register 17 for storing an observation coefficient sequence of an impulse response which is an output of the convolution integration operation circuit 16;
And the state selection circuit 18 for synchronously controlling the selection switch 13 and the second selection switch 15 and issuing a discharge command of the impulse response coefficient sequence observed and stored in the observation impulse response register 17; The impulse response coefficient sequence discharged from the observation impulse response register 17 by the instruction of
, A block power evaluation circuit for calculating the total power hpn (n = 1, 2,..., N) of each block and comparing the total power hpn between adjacent blocks 19, and the total power hp performed in the block power evaluation circuit 19
The comparison of n will be described by taking the case of N = 4 as an example. hp1 / hp2 = PD1 hp2 / hp3 = PD2 (8) hp3 / hp4 = PD3 Each adjacent total power ratio PD1, calculated by the equation (8), P
D2 and PD3 compare with the interpolation thresholds S1, S2, and S3 set correspondingly. PD1 and S1 are compared and compared under the following conditions.

Condition 1: PD1 ≧ S1 Condition 2: PD1 <S1 It is “1” when Condition 1 is satisfied, and is “0” when Condition 2 is satisfied. Similarly, PD2 and S2, PD
3 and S3 are checked. "1" generated as a result
FIG. 3 shows the state of “0” in correspondence with the interpolation thresholds S1, S2, and S3. What is necessary is just to make the coefficient block selection circuit 11 interpolate the divided update block mapping in which the update frequency is changed according to the eight cases in FIG. However, not all of the impulse response characteristics corresponding to these eight types are different from each other, and there are some which are approximated. Here, as an example, the impulse response of the sound field is roughly divided into the following three ways to perform division processing. Accordingly, three patterns of divided update block mapping are interpolated in the coefficient block selection circuit 11.

Model1: hp1>hp2>hp3> hp4 model2: hp1> hp2 = hp3 = hp4 model3: hp1 = hp2 = hp3 = hp4 The relationship of each block sum power is not strict, but is a sound. From the actual measurement experience of the field impulse response observation test or the like, it is sufficient to set such that the impulse response of each state exists. Then, from this relationship, the interpolation threshold S
1, S2 and S3 are determined. According to the output state of FIG. 3, model1 is [0, 0, 0], and model2 is
[1, 0, 0] and model3 are set to three types of [1, 1, 1], and the other five types are set so as to be absorbed by any of the three types. Each mode
The number 1 is output to the coefficient block selection circuit 11.

As described above, in accordance with the result of the comparison by the block power evaluation circuit 19, the update frequency of each block in the coefficient correction amount calculation of the coefficient correction amount calculation circuit 7 is changed to divide and update the coefficient series. Acoustic echo canceller characterized by performing:

Of course, instead of using a sequence code as a means for observing the impulse response of the sound field, it is also possible to use a time stretching pulse.

FIG. 4 and FIG. 5 are diagrams showing the moving average power of an error signal observed when the system is driven using white noise as a reference signal. Compared to the result of performing the appropriate divided / updated block mapping processing in FIG. 4, the method of attenuating the error signal is slower in the conventional divided / updated processing in FIG. From this result, it can be seen that the division update processing corresponding to the impulse response characteristics of the sound field is superior.

FIG. 6 is a diagram showing acoustic echo canceling characteristics when the system is driven using a white signal as a reference signal. The rising speed in the initial erasing stage is nearly twice as fast in the method in which the divided update block mapping processing is performed as compared with the conventional method. From this result, it can be seen that the division update method using block mapping is excellent.

[0020]

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

(1) Deterioration of the convergence speed of the acoustic reverberation elimination characteristics due to the division process of updating the coefficient correction amount can be corrected, so that the acoustic reverberation can be removed at a higher speed.

(2) Without deteriorating the acoustic echo canceling performance,
Since the amount of internal operation of the adaptive algorithm can be greatly reduced, hardware can be realized with a small configuration,
Cost can be reduced.

(3) The variation factor of the echo path characteristic is a spatial movement of a person or an object near the microphone and the speaker. In other words, in the present invention in which the frequency of updating the low-order taps of the impulse response is increased and a large interpolation correction loop gain is adapted, the rising speed of the acoustic reverberation elimination characteristic is high, so that the follow-up to the reverberation path fluctuation is performed. Is very strong, and a steady state of the communication line can be quickly established.

(4) The amount of calculation related to updating the variable coefficient sequence can be reduced to half or less while ensuring high quality of the communication line.

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

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a configuration example of an acoustic reverberation apparatus according to the present description.

FIG. 2 is a diagram showing an example of an impulse response characteristic of a sound field observed by a sequence code used in the present description.

FIG. 3 is a diagram showing a state evaluation combination for determining a divided update block mapping used in the present description.

FIG. 4 is a diagram showing an example of transition of an error signal moving average power when white noise according to the present invention used in the present description is used as a reference input.

FIG. 5 is a diagram showing an example of a transition of an error signal moving average power when white noise according to the related art used in the present description is used as a reference input.

FIG. 6 is a diagram illustrating an example of acoustic echo cancellation characteristics when white noise used in the present description 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 block selection circuit 12 code generation circuit 13 first selection switch 14 synchronous addition circuit 15 second selection switch 16 linear convolution integral operation circuit 17 observation impulse response register 18 state selection circuit 19 block power evaluation circuit

Claims (1)

(57) [Claims] And 1. A received signal input terminal, and the reception signal output terminal, and a transmission signal input terminal, and the transmission signal output terminal, N pieces of divided coefficients received from said receiving <br/> talk signal input terminal a variable coefficient digital filter having a pseudo impulse response register to be processed by block, the variable coefficient digital acoustic echo component of the received signal input to the reception signal output terminal the transmission signal input terminal via the acoustic echo path from in the acoustic echo removing apparatus said division factor, such as a residual signal obtained by subtracting the pseudo acoustic echo signal generated by the filter to minimize is sequentially updated, the said reception signal input terminal receiving the signal output terminal or the provided each switch means between transmission signal input terminal and the transmission signal output terminal, the external ring in a state in which each switch means is disconnected the transmission and reception with the partner terminal Calculating a code generating circuit for generating a code for observing the acoustic echo characteristics, acoustic echo characteristics of inputted external environment to the transmission signal input terminal via the acoustic echo path the code from the reception signal output terminal calculating means for, a storage means for the operation result is accumulated, when the switch means is connected with the other party terminal after a predetermined time has elapsed, based on the accumulated data of the storage means total power for each block , A comparing means for comparing the total power value between each of its adjacent blocks, and an interpolation of the comparison result obtained by the comparing means.
An acoustic reverberation removing apparatus, comprising: updating means for arbitrarily setting an updating frequency of each division coefficient according to a magnitude relationship with a threshold value .