JP3147207B2 - Adaptive unknown system output estimation method - Google Patents
Adaptive unknown system output estimation methodInfo
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- JP3147207B2 JP3147207B2 JP10241894A JP10241894A JP3147207B2 JP 3147207 B2 JP3147207 B2 JP 3147207B2 JP 10241894 A JP10241894 A JP 10241894A JP 10241894 A JP10241894 A JP 10241894A JP 3147207 B2 JP3147207 B2 JP 3147207B2
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- output
- filter coefficient
- unknown system
- equation
- signal
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Description
【0001】[0001]
【産業上の利用分野】この発明は、音響エコーキャンセ
ラ(反響消去装置)、適応音場制御装置などにおいて適
応的に未知系の出力を推定する、特に高速射影法により
推定する方法に関する。BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a method for adaptively estimating the output of an unknown system in an acoustic echo canceller (echo canceller), an adaptive sound field controller, and the like, and more particularly to a method for estimating the output by a high-speed projection method.
【0002】[0002]
【従来の技術】この明細書では、時間の表現は離散時間
として説明を行なう。例えば、時刻kでの信号xの値は
x(k) と表わす。この発明は、音響エコーキャンセラ、
適応音場制御などにおける適応的に未知系の出力の推定
に適用できるが、ここでは、音響エコーキャンセラを例
にとり説明を行なう。図4Aに音響エコーキャンセラの
一般的な構成を示す。入力信号x(k) は伝達関数推定部
11、畳み込み部疑似反響路12及び拡声器13へ供給
され、拡声器13よりの放音は受音器14に未知系(反
響路)15を経て収音され、受音器14の出力信号は減
算器16で畳み込み部12の出力である推定出力y^
(k) が差し引かれて推定誤差e(k) が出力される。2. Description of the Related Art In this specification, time is described as discrete time. For example, the value of signal x at time k is represented as x (k). The present invention provides an acoustic echo canceller,
Although it can be applied to the estimation of the output of an unknown system adaptively in adaptive sound field control or the like, an explanation will be given here taking an acoustic echo canceller as an example. FIG. 4A shows a general configuration of the acoustic echo canceller. The input signal x (k) is supplied to a transfer function estimator 11, a convolutional pseudo echo path 12 and a loudspeaker 13, and the sound output from the loudspeaker 13 is collected by a sound receiver 14 via an unknown system (echo path) 15. The output signal of the sound receiver 14 is output from the convolution unit 12 by the subtractor 16 at the estimated output y ^
(k) is subtracted and the estimation error e (k) is output.
【0003】ここで未知系の特性とは拡声器13および
受音器14の特性を含んだものと考える。従って、図4
Aにおける拡声器13および受音器14の特性は1であ
ること、即ち、拡声器13および、受音器14はそれぞ
れ入力信号と同一の信号を出力するものと仮定する。拡
声器13から出た音が未知系15を経て受音器14で受
音されると、音響エコーやハウリングなどの問題が発生
する。音響エコーキャンセラはこのような音の回り込み
を防止する装置である。拡声器13への入力信号(=未
知系への入力信号)x(k) と未知系の出力(=受話器1
4の出力)y(k) と、未知系の出力の推定値y^(k)
と、雑音(送話信号)n(k) と、推定誤差e(k) との関
係は次式で表わされる。Here, the characteristics of the unknown system are considered to include the characteristics of the loudspeaker 13 and the sound receiver 14. Therefore, FIG.
It is assumed that the characteristic of the loudspeaker 13 and the sound receiver 14 in A is 1, that is, the loudspeaker 13 and the sound receiver 14 each output the same signal as the input signal. When the sound output from the loudspeaker 13 is received by the sound receiver 14 via the unknown system 15, problems such as acoustic echo and howling occur. An acoustic echo canceller is a device for preventing such sound from wrapping around. Input signal to loudspeaker 13 (= input signal to unknown system) x (k) and output of unknown system (= receiver 1
4) y (k) and the estimated value y ^ (k) of the output of the unknown system
, Noise (transmission signal) n (k), and estimation error e (k) are represented by the following equation.
【0004】 e(k) =y(k) +n(k) −y^(k) (1) 未知系の伝達関数はLタップのFIRフィルタとして推
定され、そのフィルタの係数をh^1(k),h^2(k),
…,h^L (k) と表す。簡単のためにフィルタ係数を次
式で定義されるベクトルとして表わす。 h^(k) =[h^1(k),h^2(k),…,h^L (k) ]T (2) ここで、 Tは転置を表す。伝達関数推定部11では未知
系15の伝達関数を入力信号x(k) と推定誤差e(k) と
から推定する。その推定結果はフィルタ係数h^(k) と
して表し、これを畳み込み部12に転送する。畳み込み
部12では入力信号x(k) とフィルタ係数h^(k) とを
畳み込み、未知系15の応答の推定値y^(k) を生成す
る。E (k) = y (k) + n (k) −y ^ (k) (1) The transfer function of the unknown system is estimated as an L-tap FIR filter, and the coefficient of the filter is h ^ 1 (k ), H ^ 2 (k),
..., h ^ L (k). For simplicity, the filter coefficients are represented as a vector defined by the following equation. h ^ (k) = [h ^ 1 (k), h ^ 2 (k),..., h ^ L (k)] T (2) where T represents transpose. The transfer function estimator 11 estimates the transfer function of the unknown system 15 from the input signal x (k) and the estimation error e (k). The estimation result is expressed as a filter coefficient h ^ (k), which is transferred to the convolution unit 12. The convolution unit 12 convolves the input signal x (k) with the filter coefficient h ^ (k) to generate an estimated value y ^ (k) of the response of the unknown system 15.
【0005】 y^(k) =Σi=1 L x(k−i+1)h^i (k) (3) 未知系15の伝達関数は時刻とともに変化する場合が多
い。そこで、未知系15の伝達関数の推定値であるh^
(k) も未知系15の変化を検出し、適応的に修正する必
要がある。伝達関数推定部11では推定誤差e(k) と未
知系15への入力信号x(k) とからフィルタ係数修正ベ
クトルδh^(k) を計算する。フィルタ係数h^(k) は
それ自身にδh^(k) を加えることで更新される。[0005] y ^ (k) = Σ i = 1 L x (k-i + 1) h ^ i (k) (3) The transfer function of an unknown system 15 is often change with time. Therefore, h ^ which is the estimated value of the transfer function of the unknown system 15
(k) also needs to detect changes in the unknown system 15 and adaptively correct them. The transfer function estimator 11 calculates a filter coefficient correction vector δh ^ (k) from the estimation error e (k) and the input signal x (k) to the unknown system 15. The filter coefficient h ^ (k) is updated by adding δh ^ (k) to itself.
【0006】 h^(k+1) =h^(k) +αδh^(k) (4) ここで、αはステップサイズと呼ばれるスカラ量であ
る。フィルタ係数h^(k)の修正方法は適応アルゴリズ
ムとして知られている。適応アルゴリズムとしてはLM
S(Least Mean Squares)アルゴリズムや学習同定法が
よく知られているが、ここでは文献「尾関、梅田“アフ
ィン部分空間への直交射影を用いた適応フィルタアルゴ
リズムとその諸性質”、電子情報通信学会誌(A),J
67−A pp.126−132,(昭59−2)」で
提案された射影アルゴリズムを対象とする。射影アルゴ
リズムは学習同定法に比べて演算量は増加するものの、
音声入力信号に対して優れた適応特性を持つ。p次の射
影アルゴリズムでは次のp個の連立方程式が満足される
ようにフィルタ係数修正ベクトルδh^(k) を決める。H ^ (k + 1) = h ^ (k) + αδh ^ (k) (4) where α is a scalar quantity called a step size. A method of modifying the filter coefficient h ^ (k) is known as an adaptive algorithm. LM as an adaptive algorithm
The S (Least Mean Squares) algorithm and the learning identification method are well-known, but here the document "Ozeki, Umeda" Adaptive filter algorithm using orthogonal projection to affine subspace and its properties ", IEICE Magazine (A), J
67-A pp. 126-132, (Showa 59-2) ". Although the projection algorithm requires more computation than the learning identification method,
It has excellent adaptive characteristics for audio input signals. In the p-th order projection algorithm, the filter coefficient correction vector δh ^ (k) is determined so that the following p simultaneous equations are satisfied.
【0007】 y(k)=[x(k),x(k-1),…,x(k-L+1)](h^(k)+δh^(k) ) y(k-1)=[x(k-1),x(k-2),…,x(k-L)](h^(k)+δh^(k) ) (5) y(k-p+1)=[x(k-p+1),x(k-p),…,x(k-p-L+2)](h^(k)+δh^(k) ) すなわち、フィルタ係数修正ベクトルδh^(k) は式
(4)でα=1とした式により更新されたフィルタ係数
ベクトルh^(k+1)が時刻k,k−1,…,k−p+1
の入力信号に対して誤差を0にするように決められる。
これにより、入力信号が類似であれば、他の時刻でも誤
差が小さくなると期待される。Y (k) = [x (k), x (k−1),..., X (k−L + 1)] (h ^ (k) + δh ^ (k)) y (k−1 ) = [X (k−1), x (k−2),..., X (kL)] (h ^ (k) + δh ^ (k)) (5) y (k−p + 1) = [ x (k−p + 1), x (kp),..., x (kp−L + 2)] (h ^ (k) + δh) (k)) That is, the filter coefficient correction vector δh ^ (k) is .., K−p + 1 at time k, k−1,..., K−p + 1
Is determined so that the error becomes zero for the input signal of
Thus, if the input signals are similar, it is expected that the error will be reduced at other times.
【0008】図4Bに従来の射影法を伝達関数推定部1
1に適用した音響エコーキャンセラの構成を示す。つま
り伝達関数推定部11は自己相関計算部21とβ計算部
22と、フィルタ係数更新部23とからなる。従来の射
影法ではフィルタの係数の更新は次式で行なわれる。 h^(k+1)=h^(k) +α(β1(k)X(k) +β2(k)X(k-1) +…+βp (k) X(k-p+1)) (6) X(k)=[x(k) ,x(k-1) ,…,x(k-L+1)]T (7) ここで、β1(k),β2(k),…,βp (k) はプレフィルタ
係数であり、次の連立方程式の解としてβ計算部22で
求められる。FIG. 4B shows a conventional projection method using a transfer function estimating unit 1.
1 shows a configuration of an acoustic echo canceller applied to Example 1. That is, the transfer function estimating unit 11 includes the autocorrelation calculating unit 21, the β calculating unit 22, and the filter coefficient updating unit 23. In the conventional projection method, the update of the filter coefficient is performed by the following equation. h ^ (k + 1) = h ^ (k) + α (β 1 (k) X (k) + β 2 (k) X (k−1) +... + β p (k) X (k−p + 1) ) (6) X (k) = [x (k), x (k−1),..., X (k−L + 1)] T (7) where β 1 (k), β 2 (k ),..., Β p (k) are pre-filter coefficients, which are obtained by the β calculation unit 22 as solutions of the following simultaneous equations.
【0009】 β(k) T R(k)=[e(k),(1−α)e(k-1),…,(1−α)p-1 e(k-p+1) ] (8) ただし、β(k)=[β1(k),β2(k),…,βp ]T であ
る。また、R(k) は入力信号x(k) の自己相関行列であ
り、R(k) のi+1行j+1列(0≦i,j≦p−1)
の要素rij(k) は次式で定義される。Β (k) TR (k) = [e (k), (1−α) e (k−1),..., (1−α) p−1 e (k−p + 1)] (8) where β (k) = [β 1 (k), β 2 (k),..., Β p ] T. R (k) is an autocorrelation matrix of the input signal x (k), and i + 1 row j + 1 column (0 ≦ i, j ≦ p−1) of R (k)
The element r ij (k) of is defined by the following equation.
【0010】 rij(k) =rji (k)=X(k-i)T X(k-j) (9) rij(k) の更新(rij(k) からrij(k+1)を求めるこ
と)は次式のように、自己相関計算部21で行なわれ
る。 i≠0かつj≠0で、rij(k+1)=ri-1,j-1(k) その他の場合、r0j(k+1)=r0j(k) +x(k+1)x(k+1-j) −x(k-L+1)x(k-j-L+1) (10) 従来の射影法によりフィルタ係数h^(k) を1回更新す
る処理手順は図5に示すように、入力信号x(k) の自己
相関ri,j (k) が式(10)により自己相関計算部21で
行われ(S1)、次に入力信号x(k) とフィルタ係数h^
(k) との畳み込み演算が式(3)により畳み込み部12
で行われて未知系の出力の推定値y^(k) を求める(S
2)。次に減算器16で受話器14の出力から畳み込み部
12の出力推定値y^(k) を引算し、つまり式(1)を
演算して推定誤差e(k) を計算し(S3)、その推定誤
差e(k) と自己相関行列R(k)とを用いてβ計算部2
2で式(8)によりプレフィルタ係数β(k) を計算し
(S4)、そのプレフィルタ係数β(k) と入力信号x(k)
とを用いてフィルタ係数更新部23で式(6)によりフ
ィルタ係数h^(k) の更新を行う(S5)。このフィルタ
係数h^(k) の1回の更新には約(p+1)L回の積和
演算を必要とする。この積和演算量を減少するために中
間変数を用いた高速射影法を利用した音響エコーキャン
セラが知られている。これは図4Cに示すように、プレ
フィルタ係数β(k) はs更新部30で平滑化されて中間
変数更新部31へ供給され、中間変数z(k) に変換され
て畳み込み部30に供給され、自己相関計算部25より
の自己相関計数r0i(k) とs更新部30よりの平滑化プ
レフィルタ係数si (k) とが乗算器33で乗算され、そ
の乗算結果と畳み込み部32の出力とが加算器35で加
算されて出力推定値y^(k) として出力される。この高
速射影法は必要な積和演算の回数を約2L回に低減した
アルゴリズムである。高速射影法では、h^(k+1)の代
わりに次式で与えられる中間変数z(k+1) を導入する。[0010] determine the r ij (k) = r ji (k) = X (ki) T X (kj) (9) update of r ij (k) (r ij (k) from r ij (k + 1) Is performed by the autocorrelation calculation unit 21 as in the following equation. When i ≠ 0 and j ≠ 0, r ij (k + 1) = ri -1, j-1 (k) In other cases, r 0j (k + 1) = r 0j (k) + x (k + 1 ) X (k + 1-j) -x (k-L + 1) x (kj-L + 1) (10) The processing procedure for updating the filter coefficient h ^ (k) once by the conventional projection method is shown in FIG. As shown in FIG. 5, the autocorrelation r i, j (k) of the input signal x (k) is calculated by the autocorrelation calculator 21 according to the equation (10) (S 1 ). Filter coefficient h ^
The convolution operation with (k) is performed according to equation (3).
To obtain the estimated value y ^ (k) of the output of the unknown system (S
2 ). Next, the subtracter 16 subtracts the output estimation value y ^ (k) of the convolution unit 12 from the output of the receiver 14, that is, calculates Equation (1) to calculate the estimation error e (k) (S 3 ). .Beta. Calculation unit 2 using the estimation error e (k) and the autocorrelation matrix R (k).
2, the pre-filter coefficient β (k) is calculated by the equation (8) (S 4 ), and the pre-filter coefficient β (k) and the input signal x (k) are calculated.
Then, the filter coefficient updating unit 23 updates the filter coefficient hk (k) according to the equation (6) (S 5 ). One update of the filter coefficient h ^ (k) requires about (p + 1) L product-sum operations. An acoustic echo canceller using a high-speed projection method using an intermediate variable in order to reduce the product-sum operation amount is known. This is because, as shown in FIG. 4C, the pre-filter coefficient β (k) is smoothed by the s updating unit 30 and supplied to the intermediate variable updating unit 31, converted into the intermediate variable z (k) and supplied to the convolution unit 30. The multiplier 33 multiplies the autocorrelation coefficient r 0i (k) from the autocorrelation calculator 25 by the smoothing pre-filter coefficient s i (k) from the s updater 30, and multiplies the result by the convolution unit 32. Are added by an adder 35 and output as an output estimated value y ^ (k). This high-speed projection method is an algorithm in which the number of necessary product-sum operations is reduced to about 2L. In the fast projection method, an intermediate variable z (k + 1) given by the following equation is introduced instead of h ^ (k + 1).
【0011】 z(k+1)=h^(k+1)−α[s1(k)X(k)+s2(k)X(k-1) +…+sp-1(k)X(k-p+2) ] (11) ただし、 si (k)=si-1(k-1)+βi (k) ,s0(k)=0(i=1,…,p)(12) である。このsi (k)を平滑化プレフィルタ係数と呼
び、その計算はs更新部30で行なわれる。h^をzに
置換するとフィルタ係数を更新する式(6)は次のzの
回帰式となる。Z (k + 1) = h ^ (k + 1) −α [s 1 (k) X (k) + s 2 (k) X (k−1) +... + S p−1 (k) X (k−p + 2)] (11) where s i (k) = s i−1 (k−1) + β i (k), s 0 (k) = 0 (i = 1,..., p) (12) This s i (k) is called a smoothing pre-filter coefficient, and the calculation is performed by the s updating unit 30. Equation (6) for updating the filter coefficient by replacing h ^ with z is the following regression equation for z.
【0012】 z(k+1)=z(k)+αsp (k) X(k-p+1) (13) このzの時間更新は、中間変数更新部31で行なわれ
る。未知系の応答の推定値y^(k) は式(3),(11)
より、 y^(k) =h^(k) T X(k)=z(k) T X(k) +αΣsi (k-1)X(k-i) T X(k) (14) Σはi=1からp−1まで となる。自己相関値を用いれば、 y^(k) =z(k) T X(k)+αΣsi (k-1)r0i(k) (15) Σはi=1からp−1まで とかける。式(15)の右辺第1項は畳み込み部32の出
力であり、第2項以降は乗算部33の出力である。Z (k + 1) = z (k) + αs p (k) X (k−p + 1) (13) The time update of z is performed by the intermediate variable updating unit 31. The estimated value y ^ (k) of the response of the unknown system is given by equations (3) and (11).
From the above, yk (k) = h ^ (k) T X (k) = z (k) T X (k) + αΣs i (k-1) X (ki) T X (k) (14) = 1 to p-1. If the autocorrelation value is used, y ^ (k) = z (k) T X (k) + α {s i (k−1) r 0i (k) (15)} is multiplied from i = 1 to p−1. The first term on the right side of Expression (15) is the output of the convolution unit 32, and the second and subsequent terms are the outputs of the multiplication unit 33.
【0013】この高速射影法におけるフィルタ係数の更
新処理手順を図6に示す、自己相関計算部21で入力信
号x(k) の自己相関rij(k) を式(10)により計算し
(S1)、入力信号x(k) と中間変数z(k) との畳み込み
演算を畳み込み部32で、また平滑化プレフィルタ係数
si (k) と自己相関r0i(k) との内積の和を乗算器33
で行いこれらの和から未知系の出力の推定値y^(k) を
求める、つまり式(15)の計算を行う(S6)。次に減算
器16で式(1)により推定誤差e(k) を計算し
(S3)、またβ計算部22で式(8)によりプレフィル
タ係数β(k) を計算し(S4)、そのβ(k) と入力信号x
(k) とから式(12)によりs更新部30で平滑化プレフ
ィルタ係数s(k) の更新を行い(S7)、その後、そのs
i (k) を用いて中間変数更新部31で中間変数z(k) の
更新を行う(S8)。FIG. 6 shows the procedure for updating the filter coefficients in the high-speed projection method. The autocorrelation calculator 21 calculates the autocorrelation r ij (k) of the input signal x (k) according to equation (10) (S10). 1 ), the convolution operation of the input signal x (k) and the intermediate variable z (k) is performed by the convolution unit 32, and the sum of the inner products of the smoothing pre-filter coefficient s i (k) and the autocorrelation r 0i (k) To the multiplier 33
From these sums, the estimated value y ^ (k) of the output of the unknown system is obtained, that is, the calculation of equation (15) is performed (S 6 ). Next, the subtractor 16 calculates the estimation error e (k) by the equation (1) (S 3 ), and the β calculator 22 calculates the pre-filter coefficient β (k) by the equation (8) (S 4 ). , Its β (k) and the input signal x
(k), the s updating unit 30 updates the smoothing pre-filter coefficient s (k) according to the equation (12) (S 7 ).
The intermediate variable updating unit 31 updates the intermediate variable z (k) using i (k) (S 8 ).
【0014】以上説明したように、高速射影法では直接
的に伝達関数(フィルタ係数)h^(k) の更新を行なわ
ず中間変数z(k) を更新して式(15)により推定出力y
^(k) を計算する。この場合の1回の更新に必要な積和
演算量はp<<Lのときには約2Lである。これは通常の
射影法の積和演算量が(p+1)Lであるのに比べると
大幅に削減されている。As described above, in the high-speed projection method, the intermediate variable z (k) is updated without directly updating the transfer function (filter coefficient) h ^ (k), and the estimated output y is calculated by the equation (15).
Calculate ^ (k). In this case, the product-sum operation amount required for one update is about 2L when p << L. This is significantly reduced as compared with the case where the product-sum operation amount of the normal projection method is (p + 1) L.
【0015】ところで、音響エコーキャンセラーにおい
て、推定誤差e(k) が大きく増加するには2つ理由が
ある。1つは未知系15の変化であり、他の1つは雑音
(送話音声)n(k) の増加である。この2つの区別は難
しいにも関わらず、前者の場合は、フィルタ係数の更新
を続けなくてはいけないが、後者の場合にはフィルタ係
数の更新を続けると未知系の伝達関数の推定精度が劣化
するのでフィルタ係数の更新を停止する必要がある。こ
の相反する要求を満たすための方法としてFG/BG
(Fore Ground/Back Ground)方式がある。Incidentally, in the acoustic echo canceller, there are two reasons why the estimation error e (k) greatly increases. One is a change in the unknown system 15, and the other is an increase in noise (transmitted voice) n (k). Although it is difficult to distinguish between the two, in the former case the filter coefficients must be updated, but in the latter case the estimation accuracy of the transfer function of the unknown system deteriorates if the filter coefficients are updated continuously. Therefore, it is necessary to stop updating the filter coefficients. As a method for satisfying this conflicting demand, FG / BG
(Fore Ground / Back Ground) method.
【0016】図7AはFG/BG方式を説明する図であ
って、図4Aに対して入力信号x(k) に対する畳み込み
部41が更に設けられ、畳み込み部41の出力推定値と
受音器14の出力との差が減算器42でとられ、その減
算結果である推定誤差ef (k) と減算器16からの推定
残差eb (k) とから転送判断部43で転送と判断すると
44をオンとして伝達関数推定部11の推定値h^(k)
を畳み込み部41に設定する。このFG/BG方式は2
つの畳み込み部を有し、畳み込み部12がBG側、畳み
込み部41がFG側の畳み込み部と呼ばれる。FG側の
畳み込み部41のフィルタ係数は半固定である。一方、
BG側の畳み込み部12のフィルタ係数は伝達関数推定
部11の推定結果により逐次更新される。転送判断部4
3では2つの推定誤差ef (k),eb (k)のパワーの大
小を比較し、eb (k)のパワーがef (k)のパワーより
小さい、つまり、畳み込み部12のフィルタ係数がより
よく未知系15を表していると判断すると、スイッチ4
4を閉じ、BG側の畳み込み部12のフィルタ係数をF
G側の畳み込み部41に転送する。FIG. 7A is a diagram for explaining the FG / BG system. In FIG. 4A, a convolution unit 41 for the input signal x (k) is further provided, and the output estimation value of the convolution unit 41 and the sound receiver 14 are provided. Is subtracted by the subtractor 42, and the transfer determination unit 43 determines that the transfer is to be performed based on the estimated error e f (k) resulting from the subtraction and the estimated residual e b (k) from the subtractor 16. Turning on 44, the estimated value h ^ (k) of the transfer function estimating unit 11
Is set in the convolution unit 41. This FG / BG method is 2
The folding section 12 is called a BG side folding section, and the folding section 41 is called an FG side folding section. The filter coefficient of the convolution unit 41 on the FG side is semi-fixed. on the other hand,
The filter coefficients of the convolution unit 12 on the BG side are sequentially updated based on the estimation result of the transfer function estimation unit 11. Transfer judgment unit 4
In No. 3, the powers of the two estimation errors e f (k) and e b (k) are compared, and the power of e b (k) is smaller than the power of e f (k), that is, the filter of the convolution unit 12. When it is determined that the coefficient represents the unknown system 15 better, the switch 4
4 is closed, and the filter coefficient of the convolution unit 12 on the BG side is set to F
The data is transferred to the convolution unit 41 on the G side.
【0017】従って、未知系の変化により残差が増加し
た後には、適応動作によりeb (k)がef (k)より小さ
くなり、転送が起きる。一方、雑音(送話音声)が多い
場合には、BG側の畳み込み部12のフィルタ係数が乱
れ、eb (k)がef (k)より大きくなるので転送は起き
ない。[0017] Therefore, after the residual is increased by the change of the unknown system, e b (k) is smaller than e f (k) by the adaptive operation, the transfer occurs. On the other hand, if the noise (transmission voice) is large, the filter coefficient of the convolution portion 12 of the BG side is disturbed, e b (k) is the transfer does not occur becomes larger than e f (k).
【0018】[0018]
【発明が解決しようとする課題】雑音(送話音声)n
(k) が未知系の応答y(k) に比べて大きい場合にフィル
タ係数の更新を続けると、未知系の伝達関数の推定精度
が劣化する。これを避けるためそのような場合には射影
法の適応動作を停止する必要がある。そして、時刻ks
以降においてフィルタ係数の更新を停止したとすると、
その後における推定出力y^(ks +m)(m≧0)は y^(ks +m)=h^(ks )T X(ks +m) (16) となることが望ましい。SUMMARY OF THE INVENTION Noise (transmitted voice) n
If the update of the filter coefficient is continued when (k) is larger than the response y (k) of the unknown system, the estimation accuracy of the transfer function of the unknown system is degraded. To avoid this, in such a case, it is necessary to stop the adaptive operation of the projection method. And time k s
If you stop updating the filter coefficients later,
Estimated output in a subsequent y ^ (k s + m) (m ≧ 0) is y ^ (k s + m) = h ^ (k s) T X (k s + m) and made it desirable (16).
【0019】さて、高速射影法の場合、先に述べたよう
なフィルタ係数の更新の代わりとして中間変数の更新を
行なっている。そこで、フィルタ係数の更新判断機能を
有する音響エコーキャンセラとしての構成は図7Bに示
す構成が考えられる。即ち更新停止判断部51におい
て、入力信号xと推定誤差eとを入力し、入力信号xが
ゼロの場合やx<eの場合は中間変数更新部31に対し
て更新停止信号を送出する。Now, in the case of the high-speed projection method, an intermediate variable is updated instead of updating the filter coefficient as described above. Therefore, as a configuration as an acoustic echo canceller having a filter coefficient update determination function, a configuration shown in FIG. 7B can be considered. That is, the update stop judging section 51 inputs the input signal x and the estimation error e, and sends an update stop signal to the intermediate variable updating section 31 when the input signal x is zero or x <e.
【0020】時刻ks 以降において、中間変数z(k) の
更新式((13))を停止した場合、時刻ks +m(m≧
0)での未知系の出力の推定値y^(ks +m)=h^
(k s )T X(ks +m)は式(15)より、次式で求め
られる。 y^(ks +m)=z(ks )T X(ks +m) +αΣsi (ks -1)r0m+i(ks +m) (17) Σはi=1からp−1まで 上式中のr0m+i(ks +m)はi=1,…,p−2につ
いては次式により更新する。Time ksHereinafter, the intermediate variable z (k)
When the update expression ((13)) is stopped, the time ks+ M (m ≧
0), the estimated value y 出力 (k) of the output of the unknown system.s+ M) = h ^
(K s)TX (ks+ M) is obtained by the following equation from equation (15).
Can be y ^ (ks+ M) = z (ks)TX (ks+ M) + αΣsi(Ks-1) r0m + i(Ks+ M) (17) Σ is from i = 1 to p−1 r in the above equation0m + i(Ks+ M) is for i = 1,..., P-2
Then, it is updated by the following formula.
【0021】 r0m+1+i(ks +m+1)=r0m+i+1(ks +m)+x(ks -i)x(k s +m+1) −x(ks -i-L)x(ks +m-L+1) (18) この更新に要する積和演算は2(p−2)である。i=
p−1では、前時刻と比べてmの値が1増加しているの
でr0m+p(ks +m)は計算されていない。従って、式
(9)の定義により次式で求める必要がある。[0021] r 0m + 1 + i (k s + m + 1) = r 0m + i + 1 (k s + m) + x (k s -i) x (k s + m + 1) -x (k s -iL) x (k s + m-L + 1) (18) product-sum operation required for this update is 2 (p-2). i =
In p-1, r 0m + p (k s + m) is not calculated because the value of m compared with the previous time is increased by one. Therefore, it is necessary to obtain the following equation according to the definition of equation (9).
【0022】 r0m+p(ks +m+1)=X(k s +m+1) T X(k s −p+1) (19) 上式の計算にはL回の積和演算が必要となる。このよう
に、式(18)、(19)を計算するために約L+2pの新
たな演算の追加が必要になるという問題点がある。ま
た、式(18)、(19)を計算するために、x(ks −
1),…,x(ks −p−L+2)を保持する必要があ
り、L+p−2個の余分な記憶領域を必要とするという
問題点もある。図7Bに示した構成での中間変数更新の
処理手順は図8に示すようになる。即ちまず入力信号x
(k) の自己相関rij(k) を求め(S1)、次に中間変数更
新停止信号が更新停止判断部51から生じているかを調
べ(S9)、更新停止信号がなければ図6中のステップ
S6 と同様の出力の推定値y^(k) を求めるステップS
6 に移るが、更新停止信号があれば自己相関計算部21
で式(18)を計算してr0m+i(ks +m)を求め
(S10)、更に式(19)によりr0m+p(ks +m+1)
を求めてステップS6 に移る(S11)。その後は図6と
同様に推定誤差e(k) を計算し(S3)、更にプレフィル
タ係数β(k) を計算して(S4)、平滑化プレフィルタ係
数si (k) を更新する(S7)。次に再び中間変数更新停
止信号の有無を調べ(S12)、なければ中間変数z(k)
の更新を式(13)により行ってステップS1 に戻るが
(S8)、中間変数更新停止信号が有れば中間変数z(k)
の更新を行うことなくステップS1 に戻る。R 0m + p (k s + m + 1) = X (k s + m + 1) T X (k s −p + 1) (19) In the calculation of the above expression, L times of product-sum operation Is required. As described above, there is a problem that it is necessary to add about L + 2p new calculations in order to calculate Expressions (18) and (19). Further, equation (18), to calculate the (19), x (k s -
1), ..., it is necessary to hold the x (k s -p-L + 2), there is a problem that it requires L + p-2 pieces of extra storage space. FIG. 8 shows a procedure for updating the intermediate variables in the configuration shown in FIG. 7B. That is, first, the input signal x
The autocorrelation r ij (k) of (k) is obtained (S 1 ), and it is checked whether or not an intermediate variable update stop signal is generated from the update stop determination unit 51 (S 9 ). Step S in which an estimated value y ^ (k) of the output similar to step S 6 in FIG.
Then, if there is an update stop signal, the autocorrelation calculator 21
In calculates the equation (18) r 0m + i ( k s + m) the calculated (S 10), further r 0 m + p by the equation (19) (k s + m + 1)
The process proceeds to step S 6 seeking (S 11). Thereafter, the estimation error e (k) is calculated in the same manner as in FIG. 6 (S 3 ), the pre-filter coefficient β (k) is calculated (S 4 ), and the smoothed pre-filter coefficient s i (k) is updated. to (S 7). Then again checked for intermediate variable update stop signal (S 12), otherwise if intermediate variable z (k)
Returning to step S 1 the update performed by equation (13) (S 8), if there is an intermediate variable updating stop signal intermediate variable z (k)
Returns to the step S 1 without performing the update.
【0023】以上述べたように高速射影法によれば従来
の射影法よりも計算量を著しく少なくすることができる
が、雑音の影響を受けないように中間変数更新停止信号
を出力するようにする場合は、その時の計算量が多く、
かつ余分な記憶領域を必要とする、この発明はr
0m+i(ks +m)を計算することなくy^(ks +m)
(m≧0)を求め少ない計算量、記憶領域で雑音の影響
を受けない高速射影法による出力推定方法を提供するこ
とを可能とする。As described above, according to the high-speed projection method, the calculation amount can be significantly reduced as compared with the conventional projection method, but the intermediate variable update stop signal is output so as not to be affected by noise. In that case, the amount of calculation at that time is large,
In addition, the present invention requires extra storage area.
0m + i (k s + m) without calculating y ^ (k s + m)
(M ≧ 0) is obtained, and it is possible to provide an output estimation method by a high-speed projection method that is not affected by noise in a small amount of calculation and in a storage area.
【0024】[0024]
【課題を解決するための手段】この発明においては高速
射影法による未知系出力推定方法において、係数更新停
止信号を出した時刻ks 以降は、つまり適応を停止した
時刻ks 以降は適応を再開するまでプレフィルタ係数β
i (k) を、 βi (k) =0,i=1,…,p (20) とする。このようにすれば、新たな演算を追加すること
なく図4Cにおける加算器35の出力が、式(16)で示
した推定出力y^(ks +m)=h^(ks )TX(k
s +m)(m≧0)と一致する。以下にこのことを時刻
ks +1を例にとり説明する。SUMMARY OF THE INVENTION In the unknown system output estimate method with fast projection algorithm in the present invention, after the time k s that issued the coefficient update stop signal, i.e. the time k s after stopping the adaptation resume adaptation Prefilter coefficient β
i a (k), β i (k ) = 0, i = 1, ..., and p (20). In this way, the output of the adder 35 in FIG. 4C without adding new operations, estimated output shown in equation (16) y ^ (k s + m) = h ^ (k s) T X ( k
s + m) (m ≧ 0). This will be described below using the time k s +1 as an example.
【0025】式(20)とすることで、時刻ks では、式
(12)は、sp (ks )=sp-1 (ks −1)となる。
これを式(13)に代入すると、 z(ks +1)=z(ks )+αsp-1 (ks −1)x(ks −p+1)(21) となる。一方、時刻ks +1での未知系の出力の推定値
y^(ks +1)は式(15)により求まるが、これと等
しい式(14)により、 y^(ks +1)=z(ks +1) T X(ks +1) +αΣsi ( k s ) X(ks +1−i)T X(ks +1)(22) Σはi=1からp−1まで と書ける。さらに、式(12)、(20)より、s
i (ks )=si-1 (ks −1)であるから、式(22)
右辺は、 z(ks +1)T X(ks +1)+αΣsi-1 (ks −1)X(ks +1−i)T X(ks +1)=z(ks +1)T X(ks +1)+αΣsi-1 (ks −1) X(ks +1−i)T X(ks +1),(∵s0 =0) (右辺のΣはi=2からp−1まで) =z(ks +1)T X(ks +1)+αΣsi (ks −1) X(ks −i)T X(ks +1) (23) Σはi=1からp−2までとなる。また、z(ks +
1)は式(11)でk+1をk s とした式と式(21)よ
り、 z(ks +1)=h^(ks )−αΣsi (ks −1)X(ks −i)(24) Σはi=1からp−2まで となる。式(24)を式(23)に代入すると、式(22)は y^(ks +1)=h^(k) T X(ks +1) (25) となる。つまり式(20)の条件の場合は、式(15)によ
り(ks +1)について演算すると、未知系の出力の推
定値y^(ks +1)が新たな演算r 01+1+i (ks +1
+1)(式(18))、r01+p(ks +1+1)(式(1
9))を行なわずに求められる。なお式(15)中のr 0i
(k+1)は自己相関計算部21で常に逐次的に演算さ
れている。時刻ks +2以降についても式(14)によ
り、加算器35の出力はh^(ks )T X(ks +
m),m≧1となる。このように、適応を停止する時刻
ks 以降、適応を開始するまで、βi =0とすることに
より、[発明が解決しようとする課題]の項において述
べた新たな演算(式(18)、(19))と記憶領域とを必
要とせずに、未知系の出力の推定値y^(ks +m)を
求めることが可能である。By using equation (20), at time k s , equation (12) becomes s p (k s ) = s p −1 (k s −1).
When this is substituted into equation (13), z (k s +1) = z (k s) + αs p-1 (k s -1) x (k s -p + 1) becomes (21). On the other hand, the estimated value y ^ (k s +1) of the output of the unknown system at the time k s +1 is obtained by Expression (15), and by Expression (14) equivalent thereto, y ^ (k s +1) = z (k s +1) T X ( k s +1) + αΣs i (k s) X (k s + 1-i) T X (k s +1) (22) Σ is written as i = 1 to p-1. Furthermore, from equations (12) and (20), s
Since i (k s ) = s i−1 (k s −1), the equation (22)
Right side, z (k s +1) T X (k s +1) + αΣs i-1 (k s -1) X (k s + 1-i) T X (k s +1) = z (k s +1) T X (k s +1) + αΣs i -1 (k s -1) X (k s + 1-i) T X (k s +1), (∵s 0 = 0) (p-1 from the Σ of the right side i = 2 up) = z (k s +1) T X (k s +1) + αΣs i (k s -1) X (k s -i) T X (k s +1) (23) Σ is p-2 from i = 1 Up to. Also, z (k s +
1) with the formula and the formula (21) to the k +1 was k s in equation (11), z (k s +1) = h ^ (k s) -αΣs i (k s -1) X (k s - i) (24) Σ is from i = 1 to p−2. Substituting equation (24) into equation (23), equation (22) becomes y ^ (k s +1) = h ^ (k) T X (k s +1) (25). In other words, in the case of the condition of Expression (20), when (k s +1) is calculated by Expression (15), the estimated value y ^ (k s +1) of the output of the unknown system becomes a new calculation r 01 + 1 + i ( k s +1
+1) (equation (18)), r 01 + p (k s + 1 + 1) (equation (1
Required without 9)). Note that r 0i in equation (15)
(K + 1) is always calculated sequentially by the autocorrelation calculator 21. From time k s +2 onward, the output of the adder 35 is given by h ^ (k s ) T X (k s +
m), m ≧ 1. In this way, by setting β i = 0 from the time k s at which the adaptation is stopped until the adaptation is started, the new operation described in the section of [Problems to be Solved by the Invention] (Equation (18)) , (19)) and the storage area, it is possible to obtain the estimated value y + (k s + m) of the output of the unknown system.
【0026】さらに、式(12)について各ks +1,k
s +2,…についてi=1,…,pを代入してs0(k)と
βi (k) =0、i=1,…,pとの関係を考慮すると、
フィルタ係数の更新を停止してからp時刻以降では、 si (ks +m)=0,i=1,…,p、m≧p (26) となるので、図4Cの乗算器33の出力は常に0とな
る。これらのことより、 z(ks +m)T X(ks +m)=h^(ks )T X(ks +m)(27) すなわち、 z(k+m)=h^(k) ,m≧p (28) であり、中間変数zがh^に等しくなる。Further, for equation (12), each k s +1, k
Substituting i = 1,..., p for s +2,... and considering the relationship between s 0 (k) and β i (k) = 0, i = 1,.
After the time point p after updating of the filter coefficient is stopped, s i (k s + m) = 0, i = 1,..., P, m ≧ p (26), so that the output of the multiplier 33 in FIG. Is always 0. From these things, z (k s + m) T X (k s + m) = h ^ (k s) T X (k s + m) (27) i.e., z (k + m) = h ^ (k), m ≧ p (28) and the intermediate variable z is equal to h ^.
【0027】[0027]
【作用】図7Bでは更新停止信号が出ると、その間、中
間変数z(k) の更新が停止されるが、この発明では更新
停止信号が出ている間、β=0とされるが、s更新部3
0によるsi (k) 、及び中間変数更新部31によるz
(k) の更新が行われ、その結果、式(18)、式(19)の
演算が不要となる。In FIG. 7B, when the update stop signal is issued, the update of the intermediate variable z (k) is stopped during that time. According to the present invention, while the update stop signal is issued, β = 0 is set. Update unit 3
0 by s i (k) and z by the intermediate variable updating unit 31
The updating of (k) is performed, and as a result, the operations of Expressions (18) and (19) become unnecessary.
【0028】[0028]
【実施例】図1にこの発明を適用した音響エコーキャン
セラの実施例を示し、図7Bと対応する部分に同一符号
を付けてある。この実施例ではβ計算部22とs更新部
30との間にβリセット部61を直列に挿入し、更新停
止判断部51よりの更新停止信号でβリセット部61に
おいてβを0にする。図7Bとは異なり、中間変数の更
新を止めた後も、y^(k) を生成できるようにする。こ
の場合の処理手順は図2に示すように、入力信号x(k)
の自己相関ri,j (k) を式(10)により計算し(S1)、
更に入力信号x(k) と中間変数z(k) との畳み込みと、
平滑化プレフィルタ係数si (k) と自己相関r0i(k) と
の内積和とから式(15)により未知系の出力推定値y^
(k)を求め(S6)、その推定値y^(k) と受音器の出力
との差から推定誤差e(k) を式(1)により計算し(S
3)、プレフィルタ係数β(k) の計算を式(8)により計
算し(S4)、変数更新停止信号があるかをチェックし
(S9) 、なければ平滑化プレフィルタ係数si (k) の
更新を式(12)により行い(S7)、更に中間変数z(k)
の更新を式(13)により行う(S8)。つまり図8におけ
る変数更新停止信号がない場合の処理、即ち図6の処理
と同一である。FIG. 1 shows an embodiment of an acoustic echo canceller to which the present invention is applied, and portions corresponding to those in FIG. 7B are denoted by the same reference numerals. In this embodiment, a β reset unit 61 is inserted in series between the β calculation unit 22 and the s update unit 30, and β is set to 0 in the β reset unit 61 by an update stop signal from the update stop determination unit 51. Unlike FIG. 7B, y ^ (k) can be generated even after updating of the intermediate variables is stopped. The processing procedure in this case is, as shown in FIG. 2, the input signal x (k)
The autocorrelation r i, j (k) of is calculated by equation (10) (S 1 ),
Convolution of the input signal x (k) with the intermediate variable z (k);
From the sum of the inner product of the smoothing pre-filter coefficient s i (k) and the autocorrelation r 0i (k), the output estimation value y 未知 of the unknown system is obtained by equation (15).
(K) is obtained (S 6 ), and an estimation error e (k) is calculated from the difference between the estimated value y ^ (k) and the output of the sound receiver by equation (1) (S 6 ).
3 ), the pre-filter coefficient β (k) is calculated by equation (8) (S 4 ), and it is checked whether there is a variable update stop signal (S 9 ). If not, the smoothed pre-filter coefficient s i ( k) is updated according to equation (12) (S 7 ), and the intermediate variable z (k) is further updated.
Is updated by equation (13) (S 8 ). That is, the process is the same as the process in FIG. 8 when there is no variable update stop signal, that is, the process in FIG.
【0029】しかしステップS9 で変数更新停止信号が
ある場合はβリセット部61で式(20)によりβi (k)
=0としてステップS7 に移る(S13)。つまりこの実
施例では、変数更新停止信号がある間はβi (k) =0と
する以外は変数更新停止信号がない場合と同一の処理を
行うため、変数更新停止信号がない場合よりも、積和演
算の回数の増加はなく、変数更新停止信号が発生した1
時刻前のフィルタ係数h^と現時刻の入力x(k) との畳
み込みの結果が未知系の応答の推定値y^(k)となる。However, if there is a variable update stop signal in step S 9 , the β reset unit 61 calculates β i (k) according to equation (20).
= 0 as the flow proceeds to step S 7 (S 13). That is, in this embodiment, while the variable update stop signal is present, the same processing is performed as in the case where there is no variable update stop signal except that β i (k) is set to 0. There is no increase in the number of product-sum operations, and a variable update stop signal is generated 1
The result of convolution of the filter coefficient h ^ before the time and the input x (k) at the current time becomes the estimated value y ^ (k) of the response of the unknown system.
【0030】次に図4Cに示した高速射影法と図7Aに
示したFG/BG方式とを作用した音響エコーキャンセ
ラにこの発明を適用した実施例を図3に図4C、図7
A、図1と対応する部分に同一符号を付けて示す。畳み
込み部32のフィルタ係数はz(k) であるが、畳み込み
部41のフィルタ係数はh^(k) とした場合で、出力推
定値y^(k) と受音器14の出力との差であるBG側推
定誤差eb (k) が、畳み込み部41の出力推定値と受音
器14との差であるFG側推定誤差ef (k) より小さ
く、転送判断部43で転送と判断されると、その判断が
βリセット部61とスイッチ44とに与えられ、β(k)
が少くとも時刻pの間0とされ、つまり中間変数更新部
31からの中間変数z(k) がh^(k) と等しくなった
後、スイッチ44がオンにされて、その時のフィルタ係
数h^(k) が畳み込み部41に転送される。更新停止判
断部51からの更新停止信号が発生した時の動作は図1
の場合と同一である。Next, FIG. 3 shows an embodiment in which the present invention is applied to an acoustic echo canceller which operates the high-speed projection method shown in FIG. 4C and the FG / BG method shown in FIG. 7A.
A, parts corresponding to those in FIG. 1 are denoted by the same reference numerals. The filter coefficient of the convolution unit 32 is z (k), but the filter coefficient of the convolution unit 41 is h ^ (k), and the difference between the output estimated value y ^ (k) and the output of the sound receiver 14 is obtained. in a BG side estimation error e b (k) is smaller than a difference between the output estimated value and the sound receiving unit 14 of the convolution portion 41 FG side estimation error e f (k), determines the transfer in the transfer determination unit 43 Then, the judgment is given to the β reset unit 61 and the switch 44, and β (k)
Is set to 0 for at least the time p, that is, after the intermediate variable z (k) from the intermediate variable updating unit 31 becomes equal to h ^ (k), the switch 44 is turned on, and the filter coefficient h at that time is set. ^ (k) is transferred to the convolution unit 41. The operation when the update stop signal is generated from the update stop determination unit 51 is shown in FIG.
Is the same as
【0031】[0031]
【発明の効果】以上説明したように、この発明により雑
音(送話音声)が大きい場合に、高速射影法の適応動作
を止めて、フィルタ係数の推定精度の劣化を防ぐことが
でき、しかも、演算量と記憶領域の増加が無く中間変数
からフィルタ係数に変換することができる。As described above, according to the present invention, when the noise (transmitted voice) is large, the adaptive operation of the high-speed projection method can be stopped, and the deterioration of the estimation accuracy of the filter coefficient can be prevented. It is possible to convert an intermediate variable into a filter coefficient without increasing the amount of calculation and the storage area.
【0032】また高速射影法及びFG/BG方式を用い
た場合に、転送時に、βをp時刻以上0とすることによ
り、そのFG側の畳み込み部にフィルタ係数h^(k) を
設定することができ、比較的簡単な構成とすることがで
きる。When the high-speed projection method and the FG / BG method are used, β is set to be equal to or longer than the p time at the time of transfer, so that the filter coefficient h ^ (k) is set in the convolution section on the FG side. And a relatively simple configuration can be achieved.
【図1】請求項1の発明の実施例を適用した音響エコー
キャンセラの機能構成例を示すブロック図。FIG. 1 is a block diagram showing a functional configuration example of an acoustic echo canceller to which an embodiment of the invention of claim 1 is applied.
【図2】図1における処理手順の例を示す流れ図。FIG. 2 is a flowchart showing an example of a processing procedure in FIG. 1;
【図3】請求項2の発明の実施例を適用した音響エコー
キャンセラの機能構成例を示すブロック図。FIG. 3 is a block diagram showing a functional configuration example of an acoustic echo canceller to which an embodiment of the invention of claim 2 is applied.
【図4】Aは音響エコーキャンセラの一般的機能構成を
示すブロック図、Bは従来の射影法を用いた音響エコー
キャンセラの機能構成を示すブロック図、Cは従来の高
速射影法を用いた音響エコーキャンセラの機能構成を示
すブロック図である。4A is a block diagram illustrating a general functional configuration of an acoustic echo canceller, FIG. 4B is a block diagram illustrating a functional configuration of an acoustic echo canceller using a conventional projection method, and FIG. FIG. 3 is a block diagram illustrating a functional configuration of an echo canceller.
【図5】従来の射影法を用いた音響エコーキャンセラに
おけるフィルタ係数更新手順を示す流れ図。FIG. 5 is a flowchart showing a filter coefficient updating procedure in an acoustic echo canceller using a conventional projection method.
【図6】従来の高速射影法を用いた音響エコーキャンセ
ラにおける中間変数更新手順を示す流れ図。FIG. 6 is a flowchart showing an intermediate variable updating procedure in a conventional acoustic echo canceller using a high-speed projection method.
【図7】Aは従来のFG/BG方式を用いた音響エコー
キャンセラの機能構成を示すブロック図、Bは従来の高
速射影法を用いた音響エコーキャンセラに更新停止判断
機能を付けた構成を示すブロック図である。FIG. 7A is a block diagram illustrating a functional configuration of a conventional acoustic echo canceller using the FG / BG system, and FIG. 7B illustrates a configuration in which an update stop determination function is added to a conventional acoustic echo canceller using the high-speed projection method; It is a block diagram.
【図8】図7Bのエコーキャンセラの中間変数更新手順
を示す流れ図。FIG. 8 is a flowchart showing an intermediate variable updating procedure of the echo canceller shown in FIG. 7B.
───────────────────────────────────────────────────── フロントページの続き (56)参考文献 特開 平3−226026(JP,A) 特開 平6−13940(JP,A) 1990年電子情報通信学会春季全国大会 講演論文集[分冊3](1990−3−18〜 21)p.3−322 IEIEC Transaction s on Fundamentals of Electronics,Com munications and Co mputer Sciences E78 −A[10](1995−10)p.1355−1361 (58)調査した分野(Int.Cl.7,DB名) H03H 21/00 H04B 3/23 JICSTファイル(JOIS) 実用ファイル(PATOLIS) 特許ファイル(PATOLIS)────────────────────────────────────────────────── ─── Continuation of the front page (56) References JP-A-3-226026 (JP, A) JP-A-6-13940 (JP, A) 1990 National Institute of Electronics, Information and Communication Engineers Spring Annual Conference Proceedings [Section 3] (1990-3-18-21) p. 3-322 IEIEC Transactions on Fundamentals of Electronics, Communications and Computer Sciences E78-A [10] (1995-10) p. 1355-1361 (58) Fields investigated (Int. Cl. 7 , DB name) H03H 21/00 H04B 3/23 JICST file (JOIS) Practical file (PATOLIS) Patent file (PATOLIS)
Claims (1)
(k) とから上記未知系の出力の推定を行なう方法であっ
て、 上記未知系への入力信号x(k) の自己相関rij(k)(0≦
i,j≦p−1)を求め、中間変数z(k) と上記入力信
号x(k) とを畳み込んだ結果と、上記自己相関値r
ij(k) と平滑化プレフィルタ係数の内積との和から、上
記未知系の出力の推定信号y^(k) を求め、 上記未知系の出力信号y(k) と上記推定出力y^(k) と
の差信号e(k) を算出し、 その差信号e(k) と上記自己相関値rij(k) とを用いて
プレフィルタ係数β(k) を求め、 そのプレフィルタ係数β(k) を用いて、上記平滑化プレ
フィルタ係数si (k)を求め、 その平滑化プレフィルタ係数si (k) を用いて上記中間
変数z(k) を更新する高速射影法による出力の推定方法
において、 上記入力信号x(k) と上記推定誤差e(k) とに基づいて
係数更新停止信号を出し、 その係数更新停止信号の間は上記プレフィルタ係数β
(k) を0にし、 上記入力信号x(k) とフィルタ係数h^(k) とを第2畳
み込み手段で畳み込み、 その第2畳み込み手段の出力と上記未知系の出力信号と
の差信号ef (k) を算出し、 その差信号ef (k) と上記差信号e(k) とを比較し、e
(k) が小さいと上記プレフィルタ係数βを、p時刻以上
0とし、 その後、上記中間変数z(k) を上記第2畳み込み手段に
上記フィルタ係数h^(k) として設定することを特徴と
する適応的未知系出力推定方法。1. An unknown input signal x (k) and an output signal y
A method from the (k) to estimate the output of the unknown system, the autocorrelation r ij of the input signal x to the unknown system (k) (k) (0 ≦
i, j ≦ p−1 ), convolving the intermediate variable z (k) with the input signal x (k) and the autocorrelation value r
From the sum of ij (k) and the inner product of the smoothing prefilter coefficients, an estimated signal y ^ (k) of the output of the unknown system is obtained, and the output signal y (k) of the unknown system and the estimated output y ^ ( k), a pre-filter coefficient β (k) is calculated using the difference signal e (k) and the autocorrelation value r ij (k), and the pre-filter coefficient β with (k), I determined the smoothing pre-filter coefficients s i (k), output by the high speed projection method for updating the intermediate variable z (k) using the smoothing pre-filter coefficients s i (k) In the estimation method, a coefficient update stop signal is generated based on the input signal x (k) and the estimation error e (k), and the pre-filter coefficient β
(k) is set to 0, the input signal x (k) and the filter coefficient h ^ (k) are convolved by the second convolution means, and the difference signal between the output of the second convolution means and the output signal of the unknown system e f (k) is calculated, and the difference signal e f (k) is compared with the difference signal e (k).
If (k) is small, the pre-filter coefficient β is set to 0 at time p or more, and then the intermediate variable z (k) is set in the second convolution means as the filter coefficient h ^ (k). <br/> suitable応的unknown system output estimate how to.
Priority Applications (5)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP10241894A JP3147207B2 (en) | 1994-05-17 | 1994-05-17 | Adaptive unknown system output estimation method |
| DE69532394T DE69532394T2 (en) | 1994-02-10 | 1995-02-08 | Method and device for echo cancellation using the "fast projection scheme" |
| EP95101682A EP0667700B1 (en) | 1994-02-10 | 1995-02-08 | Echo cancelling method and apparatus using fast projection scheme |
| US08/385,989 US5539731A (en) | 1994-02-10 | 1995-02-09 | Echo cancelling method and apparatus using fast projection scheme |
| CA002142147A CA2142147C (en) | 1994-02-10 | 1995-02-09 | Echo cancelling method and apparatus using fast projection scheme |
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| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP10241894A JP3147207B2 (en) | 1994-05-17 | 1994-05-17 | Adaptive unknown system output estimation method |
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| Publication Number | Publication Date |
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| JP3147207B2 true JP3147207B2 (en) | 2001-03-19 |
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Non-Patent Citations (2)
| Title |
|---|
| 1990年電子情報通信学会春季全国大会講演論文集[分冊3](1990−3−18〜21)p.3−322 |
| IEIEC Transactions on Fundamentals of Electronics,Communications and Computer Sciences E78−A[10](1995−10)p.1355−1361 |
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