JP2002261660A - Multi-channel echo canceling method, its apparatus, its program and its recording medium - Google Patents
Multi-channel echo canceling method, its apparatus, its program and its recording mediumInfo
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Abstract
(57)【要約】
【課題】 ステレオ信号間にパワーの差があり、相互相
関が強い場合でも収束速度を速くする。
【解決手段】 受話信号x1(k),x2(k)の各所定
時間の2乗和‖x′1(k)‖2,‖x′2(k)‖2
を求め(321 ,322)、その小さい方に、例えば‖
x′1(k)‖に対し、μ1=√‖x′1(k)‖/
√(‖x′1(k)‖2 +‖x′2(k)‖2)を、
他方にはμ2 を更新ステップサイズとして反響路推定部
191 ,192で各受話信号x1(k),x2(k)と残
留反響信号e(k)により疑似反響路181 ,182 に
対する修正ベクトルを計算する。
(57) [Problem] To increase the convergence speed even when there is a difference in power between stereo signals and cross-correlation is strong. A received signal x 1 (k), x 2 2 sum ‖X each predetermined time (k) '1 (k) ‖ 2, ‖x' 2 (k) ‖ 2
(32 1 , 32 2 ), and, for example, ‖
For x ′ 1 (k)}, μ 1 = {x ′ 1 (k)} /
√ (‖x ′ 1 (k) ‖ 2 + ‖x ′ 2 (k) ‖ 2 )
On the other hand, the reverberation path estimators 19 1 and 19 2 set the μ 2 as the update step size and use the received signals x 1 (k) and x 2 (k) and the residual reverberation signal e (k) to generate pseudo echo paths 18 1 and 18. Compute the correction vector for 2 .
Description
【0001】[0001]
【発明の属する技術分野】この発明は、多チャネル再生
系を有する例えば通信会議システムにおいて、ハウリン
グの原因および聴覚上の障害となる室内反響信号を消去
する多チャネル反響消去方法、その装置、そのプログラ
ム及びその記録媒体に関するものである。BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a multi-channel echo canceling method for erasing an indoor echo signal which causes howling and impairs hearing in, for example, a communication conference system having a multi-channel reproducing system, a multi-channel echo canceling method, an apparatus therefor, and a program therefor. And its recording medium.
【0002】[0002]
【従来の技術】同時通話性能に優れ反響感の少ない拡声
通話システムを提供するために、反響消去装置がある。
まず、1チャネル用の反響消去装置について、その反響
消去方法および装置構成を図4を参照して説明する。拡
声通話において、受話端子11に相手の発話等で得られ
る受話信号は、そのままスピーカ12から再生される場
合と、スピーカ12へ送る前に、受話信号の振幅やパワ
ー等の大きさに応じて自動的に利得を調節するなど、受
話信号に何らかの加工が施された後に、スピーカ12か
ら再生される場合とがある。このため、この明細書で受
話信号x1(k)とは、相手からの受話信号そのものとは
限らず、受話信号に対し加工が施された場合は、その加
工された後の受話信号を指すものとする。kは離散時間
を表す。反響消去装置14は、受話信号x1(k)がスピ
ーカ12から反響路15を経て、マイクロホン16に集
音されて得られる反響信号y1(k)を消去する。ここ
で、反響信号y1(k)は、時刻kにおける反響路15の
インパルス応答をh11(k,n)として、 y1(k)=Σh11(k,n)x1(k−n) (1)Σ はn=0からL−1までのような畳み込み演算で得
られるものとモデル化できる。Lはタップ数で、反響路
15の残響時間に対応させて、あらかじめ設定しておく
定数である。まず、受話信号蓄積・ベクトル生成部17
において、受話信号x1(k)をL−1時刻過去のものま
で蓄積しておく。蓄積された信号は、受話信号ベクトル
x1(k)、すなわち x1(k)=[x1(k),x1(k−1),…, x1(k−L+1)]T (2) として出力される。但し、*T はベクトルの転置を表
す。疑似反響信号生成部18では、式(2)の受話信号
ベクトルx1(k)と、反響路推定部19から得られる
疑似反響ベクトルh^11(k)との内積演算 y^1(k)=h^11 T (k)x1(k) (3) を行ない、その結果として、疑似反響信号y^1(k)を
生成する。この内積演算は、式(1)のような畳み込み
演算と等価である。反響路推定部19では、疑似反響信
号生成部18で用いる疑似反響路ベクトルh^
11(k)を生成する。この反響路推定に用いる最も一般
的なアルゴリズムは、NLMSアルゴリズム(学習同定
法)である。NLMSアルゴリズムでは、時刻kにおけ
る受話信号ベクトルx1(k)と、残留反響信号e
1(k)、すなわち差回路21でマイクロホン16の出力
y1(k)から疑似反響信号y^1(k)を差し引いた信号 e1(k)=y1(k)−y^1(k) (4) とから、時刻k+1において用いる疑似反響路ベクトル
h^11(k+1)を次式のように求める。h ^11(k+1) =h^11(k) +μe1(k)x1(k)/(x1 T (k) x1(k)) (5) 但し、μはステップサイズパラメータと呼ばれ0<μ<
2の範囲で適応動作の調整に用いる。以上のような処理
を繰り返すことにより、反響路推定部19では、次第に
疑似反響路ベクトルh^11(k)を、真の反響路15
のインパルス応答h11(k,n)の時系列を各要素とし
て持つ反響路ベクトルh11(k)、すなわち h11(k) =[h11(k,0),h11(k,1),…, h11(k,L-1)]T (6) と一致させることが可能となり、その結果式(4)の残
留反響信号e1 (k)を小さくすることができる。2. Description of the Related Art In order to provide a loudspeaker communication system which is excellent in simultaneous call performance and has little reverberation, there is an echo canceller.
First, an echo canceling method and device configuration of the echo canceling device for one channel will be described with reference to FIG. In a loudspeaker call, a reception signal obtained by the other party's utterance or the like at the reception terminal 11 is reproduced from the speaker 12 as it is, or before being transmitted to the speaker 12, it is automatically transmitted depending on the amplitude, power, etc. of the reception signal. In some cases, the received signal is reproduced from the speaker 12 after some processing has been performed on the received signal, such as by adjusting the gain. Therefore, in this specification, the reception signal x 1 (k) is not limited to the reception signal itself from the other party, and refers to the reception signal after the reception signal is processed when the reception signal is processed. Shall be. k represents a discrete time. The echo canceller 14 cancels the echo signal y 1 (k) obtained by collecting the received signal x 1 (k) from the speaker 12 via the echo path 15 and the microphone 16. Here, the reverberation signal y 1 (k) is represented by y 1 (k) = Σh 11 (k, n) × 1 (k−n) where h 11 (k, n) is the impulse response of the reverberation path 15 at time k. (1) Σ can be modeled as one obtained by a convolution operation such as n = 0 to L−1. L is the number of taps and is a constant that is set in advance so as to correspond to the reverberation time of the reverberation path 15. First, the reception signal accumulation / vector generation unit 17
, The received signal x 1 (k) is stored up to the past L-1 times. The stored signal is a received signal vector x 1 (k), that is, x 1 (k) = [x 1 (k), x 1 (k−1),..., X 1 (k−L + 1)] T (2 ) Is output as. Here, * T represents transposition of a vector. The pseudo echo signal generation unit 18 calculates the inner product y 積1 (k) of the reception signal vector x 1 (k) of Expression (2) and the pseudo echo vector h ^ 11 (k) obtained from the echo path estimation unit 19. = h ^ 11 performs T (k) x 1 (k ) (3), as a result, it generates an estimated echo signal y ^ 1 (k). This inner product operation is equivalent to a convolution operation such as Expression (1). The echo path estimating section 19 generates the pseudo echo path vector h ^ used in the pseudo echo signal generation section 18.
11 (k) is generated. The most common algorithm used for this echo path estimation is the NLMS algorithm (learning identification method). In the NLMS algorithm, the received signal vector x 1 (k) at time k and the residual echo signal e
1 (k), that is, a signal e 1 (k) = y 1 (k) −y ^ 1 (k) obtained by subtracting the pseudo echo signal y ^ 1 (k) from the output y 1 (k) of the microphone 16 in the difference circuit 21. From (4), the pseudo echo path vector h ^ 11 (k + 1) to be used at time k + 1 is obtained as in the following equation. h ^ 11 (k + 1) = h ^ 11 (k) + μe 1 (k) x 1 (k) / (x 1 T (k) x 1 (k)) (5) where, mu is a step size parameter Called 0 <μ <
It is used for adjusting the adaptive operation in the range of 2. By repeating the above processing, the echo path estimating unit 19 gradually converts the pseudo echo path vector ^ 11 (k) into the true echo path 15.
The echo path vector h 11 (k) having a time series of the impulse response h 11 (k, n) of each as an element, that is, h 11 (k) = [h 11 (k, 0), h 11 (k, 1) ,..., H 11 (k, L-1)] T (6), and as a result, the residual echo signal e 1 (k) of the equation (4) can be reduced.
【0003】一般にN(>2)チャネルの再生系とM
(>1)チャネルの集音系とで構成される通信会議シス
テムの場合の反響の消去は、図5に示すような構成によ
り行われる。すなわち再生側の全Nチャネルと集音側の
各1チャネルとの間にN入力1出力時系列信号を処理す
るNチャネル反響消去装置221,222,…,22M をそ
れぞれ接続した反響消去システム23として実現され
る。この場合システム全体でN×M個の反響路15
nm(1<n<N,1<m<M)が存在する。このシステ
ムの構成単位である再生側の全Nチャネルと集音側の各
1チャネルとの間に接続されるNチャネル反響消去装置
221,222,…,22M については、図4に示した反響
消去装置14の構成を拡張して、図6に示すように構成
される。これは例えば電子情報通信学会論文誌 '86/
10 Vol.J69−A No.10「多チャネル適
応ディジタルフィルタ」に詳しく述べられている。ここ
で、集音側が第m集音チャネル(1<m<M)に接続さ
れているNチャネル反響消去装置22m を考える。第m
チャネルのマイクロホン16m で集音される反響信号
は、各再生チャネルの受話信号がそれぞれの反響路15
1m〜15Nmを経て集音側で全て加算されることにより得
られるために、反響路推定をどの再生チャネルについて
も、統一的に同じ1つの残留反響信号em (k)のみを
評価して行なうための工夫が必要となる。まず、各再生
チャネルの受話信号について、受話信号蓄積・ベクトル
生成部(171,172,…,17N )により、受話信号ベ
クトル x1(k)=[x1(k),x1(k−1),…, x1(k−L1 +1)]T (7) x2(k)=[x2(k),x2(k−1),…, x2(k−L2 +1)]T (8) : xN (k)=[xN (k),xN (k−1), …, xN (k−LN +1)]T (9) を生成する。但し、L1,L2,…,LN はタップ数で、各
反響路151m,152m,…,15Nmの残響時間に対応さ
せて、あらかじめ設定する定数である。これらのベクト
ルをベクトル結合部24によって、 x(k) =[x1 T (k),x2 T (k),…,xN T (k)]T (10) と結合する。また、反響路推定部19m においても、各
再生チャネルと第m集音チャネルとの間のN個の反響路
を模疑するための、各疑似反響路ベクトルh^ 1m(k),
h^2m(k),…,h^Nm(k) を結合して h^m (k) =[h^1m T (k),h^2m T (k),…,h^Nm T (k)]T (11) として扱う。疑似反響路結合ベクトルh^m (k) の更
新は、NLMSアルゴリズムを用いた場合、 h^m (k+1)=h^m (k)+μem (k) x(k) /(xT (k) x(k)) (12) のように行なわれる。疑似反響信号生成部18m では、
内積演算 y^m (k)=h^m T (k) x(k) (13) により、第m集音チャネルで集音された反響信号y
m (k)に対する疑似反響信号y^m (k)を生成す
る。このように、各チャネル毎のベクトルを結合して1
つのベクトルとして扱うことにより、基本的な処理の流
れは、図4に示した1チャネル反響消去装置と同様とな
る。In general, N (>2) Channel playback system and M
(>1) Communication conference system consisting of a channel sound collection system
Elimination of the echo in the case of the system
Is performed. That is, all N channels on the playback side and the
Process N-input 1-output time-series signal between each channel
N-channel echo canceller 221, 22Two,…, 22MTo
Realized as a connected echo cancellation system 23
You. In this case, N × M reverberation paths 15 in the entire system are used.
nm(1<n<N, 1<m<M) exists. This system
Each of the N channels on the playback side and the
N-channel echo canceller connected between one channel
221, 22Two,…, 22MAbout the echo shown in Figure 4
The configuration of the erasing device 14 is expanded as shown in FIG.
Is done. This is, for example, IEICE Transactions '86 /
10 Vol. J69-A No. 10 "Multi-channel suitable
Digital filters ". here
And the sound collecting side is the m-th sound collecting channel (1<m<M) connected
N-channel echo canceller 22mthink of. M-th
Channel microphone 16mEcho signal collected by the
Indicates that the reception signal of each reproduction channel is
1m~ 15NmThrough the summation on the sound collection side
To estimate the echo path for
Also, the same one residual echo signal em(K) only
It is necessary to devise a way to evaluate it. First, each playback
Received signal accumulation / vector for received signal of channel
Generation unit (171, 17Two,…, 17N), The received signal
Kutor x1(k) = [x1(k), x1(k-1),…, x1(k-L1 +1)]T (7) xTwo(k) = [xTwo(k), xTwo(k-1),…, xTwo(k-LTwo +1)]T (8): xN(k) = [xN(k), xN(k−1),…, xN(k−LN+1)]T (9) is generated. Where L1, LTwo,…, LNIs the number of taps, each
Echoway 151m, 152m, ..., 15NmCorresponding to the reverberation time
Is a constant to be set in advance. These vectors
X (k) = [x1 T(k), xTwo T(k),…, xN T(k)]T Combine with (10). In addition, the echo path estimation unit 19mIn each
N echo paths between the playback channel and the m-th sound pickup channel
To simulate 1m(k),
h ^2m(k),…, h ^Nm(k) and h ^m(k) = [h ^1m T(k), h ^2m T(k),…, h ^Nm T(k)]T (11) Pseudo echo path coupling vector h ^m(k)
Newly, when the NLMS algorithm is used, h ^m(k + 1) = h ^m(k) + μem(k) x (k) / (xT(k) x (k)) (12). Simulated echo signal generator 18mThen
Dot product operation y ^m(K) = h ^m T(k) x (k) (13), the reverberation signal y collected in the m-th sound collection channel
mPseudo echo signal y ^ for (k)mGenerate (k)
You. In this way, the vectors for each channel are combined to form 1
By treating them as two vectors, the basic processing flow
This is similar to the one-channel echo canceller shown in FIG.
You.
【0004】この多チャネル反響消去方法において、各
チャネルの受話信号の平均電力が異なると、収束速度が
大幅に劣化する。この点から前記論文法には、各チャネ
ルごとに受話信号を電力により正規化することにより改
善することが提案されている。次のベクトルを定義す
る。 Z(k)=[x1 T(k)/P1(k),…,xN T(k)/PN(k)] T (14) Pn (k)=xn T (k)xn(k)/Ln (15) Pn (k)は各チャネルごとに電力を正規化するための
正規化係数である。疑似反響路結合ベクトルh
m (k)の更新式(12)はこのベクトルZ1 (k)
を用いて次式で表わせる。In this multi-channel echo cancellation method,
If the average power of the received signals of the channels is different, the convergence speed
Degrades significantly. From this point, each of the channel methods
By normalizing the received signal for each
It is proposed to be good. Define the following vector
You. Z (k) = [x1 T(K) / P1(K), ..., xN T(K) / PN(K)] T (14) Pn(K) = xn T(K) xn(K) / Ln (15) Pn(K) is for normalizing power for each channel.
It is a normalization coefficient. Pseudo echo path coupling vector h
mThe updating equation (12) of (k) is obtained by calculating the vector Z1(K)
Can be expressed by the following equation.
【0005】 h^m(k+1)=h^m(k)+μem(k)Z(k)/(ZT(k )x(k)) (16) ZT (k)x(k)では次の性質が成り立つ。 ZT (k)x(k)=Σn=1 N Ln (17) このように各チャネルの受話信号を電力により正規化す
ることにより収束速度を改善できる。この場合、各疑似
反響路h^1m,…,h^Nmごとに独立にその更新を
考える。説明を簡単にするため、図7に示すようにステ
レオの2チャネルの受話信号と1チャネルの送話信号と
を考える。つまり受話信号x1(k)が疑似反響路181
へ供給されて、疑似反響信号h^1 T(k)x
1(k)が生成され、受話信号x2(k)が疑似反響路1
82 へ供給されて疑似反響信号h^2 T(k)x
2(k)が生成され、これら疑似反響信号が加算回路2
5で加算され、その加算信号が疑似反響信号y^(k)
として、マイクロホン16の集音信号y(k)から差し
引かれ、残留反響信号e(k)が出力される。この残留
反響信号e(k)と受話信号x1 (k)とにより反響路
推定部191 でスピーカ121 とマイクロホン16間の
反響路h1 が推定され、疑似反響路181 の疑似反響
路181 の特性h^1 が更新される。同様に残留反響
信号e(k)と受話信号x2 (k)とにより反響路推定
部192 でスピーカ122 とマイクロホン16間の反響
路h 2 が推定され、疑似反響路182 の特性h^2
が更新される。H ^m(K + 1) = h ^m(K) + μem(K) Z (k) / (ZT(K) x (k)) (16) ZTIn (k) x (k), the following properties hold. ZT(K) x (k) = Σn = 1 NLn (17) In this way, the received signal of each channel is normalized by power.
By doing so, the convergence speed can be improved. In this case, each pseudo
Echo path h ^1m, ..., h ^NmUpdate each time independently
Think. For simplicity of explanation, the steps shown in FIG.
Leo's two-channel reception signal and one-channel transmission signal
think of. That is, the reception signal x1(K) is a pseudo echo path 181
Supplied to the pseudo echo signal h 信号1 T(K) x
1(K) is generated and the received signal xTwo(K) is pseudo echo path 1
8TwoSupplied to the pseudo echo signal h ^Two T(K) x
Two(K) is generated, and these pseudo echo signals are added to the addition circuit 2
5 and the added signal is a pseudo echo signal y ^ (k)
From the sound pickup signal y (k) of the microphone 16
Then, a residual echo signal e (k) is output. This residue
Echo signal e (k) and received signal x1(K) and the reverberation path
Estimator 191Speaker 121Between the microphone 16
Echo path h1Is estimated, and the pseudo echo path 181Pseudo-resonance of
Road 181Characteristic h ^1Is updated. As well as residual echo
Signal e (k) and received signal xTwo(K) Echo path estimation
Part 19TwoSpeaker 12TwoBetween microphone and microphone 16
Road TwoIs estimated, and the pseudo echo path 18TwoCharacteristic h ^Two
Is updated.
【0006】反響路推定部191 ,192 での反響路推
定、つまり各疑似反響路181 ,182 の特性の更新を
式(16)より求める。この場合のベクトルZ(k)
は Z(k)=[x1 T(k)/P1(k),x2 T(k)/P2(k)]T (18) となる。また式(16)中のh^m(k),x
(k),Z(k)はそれぞれ以下の通りである。h ^m(k)=h^(k)=[h^1 T(k),
h^2 T(k)]Tx (k)=[x1 T(k),x2 T(k)]TZ (k)=[x1 T (k)/(x1 T (k)x1
(k)/L1),x 2 T (k)/(x2 T (k)x
2(k)/L2)]T=[L1x1 T(k)/(x
1 T(k)x1(k)),L2x2 T(k)/(x2 T
(k)x2(k))]T 式(16)の分母ZT (k)x(k)はZT (k)x(k)=[L1x1 T (k)/(x1
T (k)x1(k)),L2 x2 T (k)/(x
2 T (k)x2 (k))][x1 (k),x
2 (k)]T =L1 +L2 これを式(16)に代入すると、h^(k+1)=
h^(k)+μe(k)Z(k)/(L1 +L2 )
はThe reverberation path estimator 191, 19TwoResonance path in
Fixed, that is, each pseudo echo path 181, 18TwoUpdate the characteristics of
It is obtained from equation (16). The vector Z (k) in this case
Is Z (k) = [x1 T(K) / P1(K), xTwo T(K) / PTwo(K)]T (18) H ^ in equation (16)m(K), x
(K) and Z (k) are as follows. h ^m(K) = h ^ (k) = [h ^1 T(K),
h ^Two T(K)]Tx (k) = [x1 T(K), xTwo T(K)]TZ (k) = [x1 T(K) / (x1 T(K) x1
(K) / L1), X Two T(K) / (xTwo T(K) x
Two(K) / LTwo)]T= [L1x1 T(K) / (x
1 T(K) x1(K)), LTwoxTwo T(K) / (xTwo T
(K) xTwo(K))]T Denominator Z in equation (16)T(K) x (k) is ZT (K) x (k) = [L1x1 T(K) / (x1
T(K) x1(K)), LTwoxTwo T(K) / (x
Two T(K) xTwo(K))] [x1(K), x
Two(K)]T= L1+ LTwo Substituting this into equation (16) gives h ^ (k + 1) =
h ^ (k) + μe (k) Z (k) / (L1+ LTwo)
Is
【0007】[0007]
【数1】 (Equation 1)
【0008】となり、h^1 とh^2 はそれぞれ次
式となる。 h^1 (k+1)=h^1 (k)+μ(e(k)/(L1 +L2 ))( L1 x1 (k)/(x1 T (k)x1 (k)) (19) h^2(k+1)=h^2(k)+μ(e(k)/(L1+L2))(L2 x2 (k)/x2 T(k)x2(k)) (20)xn T (k)xn(k)=‖xn(k)‖2、‖xn
(k)‖2=xn 2(k)+xn 2(k−1)+…+
xn 2(k−Ln+1)またタップ数が等しくL1=L2
であれば h^1(k+1)=h^1(k)+0.5μe(k)x1(k)/‖x1 (k)‖2 (21) h^2(k+1)=h^2(k)+0.5μe(k)x2(k)/‖x2 (k)‖2 (22) となる。反響路推定部191 ,192 において式(2
1)、(22)によりそれぞれ反響路h1(k),
h2(k)を推定して、疑似反響路181 ,182 を
それぞれ更新することになる。式(21)、(22)の
各右辺第2項は受話信号x1(k),x2(k)のベ
クトルがパワー(電力)でそれぞれ正規化されたものと
なっている。Where h ^ 1 and h ^ 2 are respectively given by the following equations. h ^ 1 (k + 1) = h ^ 1 (k) + μ (e (k) / (L 1 + L 2)) (L 1 x 1 (k) / (x 1 T (k) x 1 (k)) ( 19) h ^ 2 (k + 1) = h ^ 2 (k) + μ (e (k) / (L 1 + L 2)) (L 2 x 2 (k) / x 2 T (k) x 2 (k)) (20) x n T (k) x n (k) = {x n (k)} 2 , ‖x n
(K) || 2 = x n 2 (k) + x n 2 (k-1) + ... +
x n 2 (k−L n +1) and the number of taps is equal and L 1 = L 2
If h ^ 1 (k + 1) = h ^ 1 (k) + 0.5μe (k) x 1 (k) / ‖x 1 (k) || 2 (21) h ^ 2 ( k + 1) = h ^ 2 ( k) +0.5 μe (k) × 2 (k) / {x 2 (k)} 2 (22) In the echo path estimating units 19 1 and 19 2 , the expression (2)
The reverberation paths h 1 (k),
By estimating h 2 (k), the pseudo echo paths 18 1 and 18 2 are updated respectively. The second term on each right side of Expressions (21) and (22) is a vector of the received signals x 1 (k) and x 2 (k) normalized by power.
【0009】[0009]
【発明が解決しようとする課題】前述したように、各チ
ャネルの受話信号をその電力により正規化することによ
り、各チャネルの受話信号の平均電力が異なっても収束
速度を劣化させないことができる。しかし、2チャネル
の受話信号間に平均電力の差があってもステレオ信号の
ように受話信号の相互相関が強い場合は、疑似反響路が
真の反響路の特性に収束しないため、遠隔で話者が交代
し、例えばその交代前はx1 (k)の平均電力がx
2 (k)の平均電力より大であったのが、話者交代によ
りx1 (k)の平均電力がx2 (k)の平均電力より小
さくなったりすると、反響が生じ疑似反響路の収束に時
間がかかり、例えば話者が2秒程度で交代することを繰
り返すと、疑似反響路は収束しない。このことは受話2
チャネル、収音1チャネルの多チャネル反響消去方法に
限らず、一般の受話Nチャネル、収音Mチャネルの多チ
ャネル反響消去方法においても同様のことが言える。As described above, by normalizing the received signal of each channel by its power, it is possible to prevent the convergence speed from being deteriorated even if the average power of the received signal of each channel is different. However, if there is a strong cross-correlation between the received signals, such as a stereo signal, even if there is a difference in the average power between the received signals of the two channels, the pseudo echo path does not converge to the characteristics of the true echo path. Are changed, for example, before the change, the average power of x 1 (k) is x
Although it was larger than the average power of 2 (k), if the average power of x 1 (k) became smaller than the average power of x 2 (k) due to speaker change, reverberation would occur and convergence of the pseudo-echo path would occur. Takes a long time, for example, if the speaker alternates in about 2 seconds, the pseudo echo path does not converge. This is received 2
The same can be said not only for the multi-channel echo cancellation method of one channel and sound collection, but also for the general multi-channel echo cancellation method of N-channel reception and M sound collection channels.
【0010】この発明の目的は、受話信号間にその平均
電力に差があっても、また相互相関が比較的強く、しか
も受話信号の平均電力比が変化しても比較的速く疑似反
響路を収束させることができる多チャネル反響消去方
法、その装置及びそのプログラム及び記録媒体を提供す
ることにある。An object of the present invention is to form a pseudo echo path relatively quickly even if there is a difference in the average power between received signals, the cross-correlation is relatively strong, and the average power ratio of the received signal changes. It is an object of the present invention to provide a multi-channel echo canceling method, an apparatus thereof, a program thereof, and a recording medium capable of converging.
【0011】[0011]
【課題を解決するための手段】この発明の方法によれ
ば、Nチャネル(Nは2以上の整数)の受話信号を、N
個の反響路を模疑した疑似反響路を通してN個の疑似反
響信号を生成し、これらN個の疑似反響信号を加算し
て、総合疑似反響信号を求め、その総合疑似反響信号を
収音信号から差し引いて残留反響信号を求め、特に各チ
ャネルの受話信号の大きさ、つまり振幅又は電力をそれ
ぞれ求め、これら大きさの中からその大きさが最大のも
のと対応する受話信号チャネルの更新ステップサイズよ
り、上記受話信号チャネル以外の受話信号のチャネルの
更新ステップサイズを小とし、一般には前者の更新ステ
ップサイズを最も収束速度が速いように1と、後者のそ
れを1より小とし、各チャネルごとにその受話信号を上
記そのチャネルの更新ステップサイズと上記残留反響信
号とによりそのチャネルの疑似反響路の修正ベクトルを
求め、その修正ベクトルによりそのチャネルの疑似反響
路の特性を修正する。According to the method of the present invention, a reception signal of N channels (N is an integer of 2 or more) is converted to N
N pseudo echo signals are generated through a pseudo echo path that simulates a number of echo paths, and these N pseudo echo signals are added to obtain a total pseudo echo signal. To obtain the residual echo signal, and in particular, to determine the size of the received signal of each channel, that is, the amplitude or power, and to determine the update step size of the received signal channel corresponding to the largest one of these sizes. Therefore, the update step size of a channel of a reception signal other than the reception signal channel is set to be small, and generally, the update step size of the former is set to 1 so that the convergence speed is the fastest, and that of the latter is set to be smaller than 1, and The received speech signal is obtained by using the updated step size of the channel and the residual echo signal to obtain a correction vector of a pseudo echo path of the channel, and the correction vector Le By modifying the properties of the estimated echo path in the channel.
【0012】[0012]
【発明の実施の形態】以下この発明の実施形態を図面を
参照して説明する。理解がし易いように、受話チャネル
が、例えばステレオの2チャネルで、送話(収音)チャ
ネルが1チャネルの場合を例として図1を参照して説明
する。図1Aにおいて図7と対応する部分には同一参照
番号を付けてある。受話信号x1 (k)とx2 (k)は
それぞれ疑似反響路181 と182 に供給されて、各反
響路h1 とh2 と通った反響信号に相当する疑似反響信
号を生成する。つまり疑似反響路181 ,182 の各タ
ップ数をLとすると、疑似反響路181 のインパルス応
答h^1 T(k)と受話信号ベクトルx1(k)の内
積h^1 T(k)x1(k)及び疑似反響路182 の
インパルス応答h^2 T(k)と受話信号ベクトルx
2(k)の内積h^2 T(k)x2(k)が生成され
る。Embodiments of the present invention will be described below with reference to the drawings. For ease of understanding, a case will be described with reference to FIG. 1 as an example in which two reception channels are stereo channels and one transmission (sound collection) channel is one channel. 1A, parts corresponding to those in FIG. 7 are denoted by the same reference numerals. The received signals x 1 (k) and x 2 (k) are supplied to the pseudo echo paths 18 1 and 18 2 , respectively, to generate pseudo echo signals corresponding to the echo signals passing through the echo paths h 1 and h 2 , respectively. . That is, assuming that the number of taps of each of the pseudo echo paths 18 1 and 18 2 is L, the inner product h ^ 1 T (k) of the impulse response h ^ 1 T (k) of the pseudo echo path 18 1 and the reception signal vector x 1 (k). ) X 1 (k) and the impulse response h ^ 2 T (k) of the pseudo echo path 18 2 and the received signal vector x
2 inner product of (k) h ^ 2 T ( k) x 2 (k) is generated.
【0013】ここで h^1(k)=[h^1(k,0),h^1(k,1),…,h^1(k,L −1)]T h^2(k)=[h^2(k,0),h^2(k,2),…,h^2(k,L −1)]T x1(k)=[x1(k),x1(k−1),…,x1(k−L+1)]T x2(k)=[x2(k),x2(k−1),…,x2(k−L+1)]T (23) である。これらh^1 T(k)x1(k)とh^2 T
(k)x2(k)が合成され、総合疑似反響信号y^
(k)とされ、収音信号y(k)との差e(k)=y
(k)−y^(k)が残留反響信号として求められる。Here, h ^ 1 (k) = [h ^ 1 (k, 0), h ^ 1 (k, 1),..., H ^ 1 (k, L−1)] T h ^ 2 (k ) = [h ^ 2 (k , 0), h ^ 2 (k, 2), ..., h ^ 2 (k, L -1)] T x 1 (k) = [x 1 (k), x 1 .., X 1 (k−L + 1)] T x 2 (k) = [x 2 (k), x 2 (k−1),..., X 2 (k−L + 1)] T ( 23) These h ^ 1 T (k) x 1 (k) and h ^ 2 T
(K) x 2 (k) is synthesized, and the total pseudo echo signal y ^
(K), and the difference e (k) = y from the picked-up signal y (k)
(K) −y ^ (k) is obtained as a residual echo signal.
【0014】この実施形態では各チャネルの受話信号x
1(k)とx2(k)がステップサイズ決定部31に入力
されて、各チャネルに対する更新ステップサイズμ1 ,
μ2が計算される。この場合受話信号の所定時間の大き
さが大きいチャネルには収束が速いような更新ステップ
サイズを、受話信号の所定時間の大きさが小さいチャネ
ルには前記更新ステップはより小さい更新ステップとす
る。例えば図1Bに示すように、受話信号の大きさとし
て電力を用い、各受話信号のx1(k),x2(k)の所
定時間内の2乗和‖x′1(k)‖2,‖x′
2(k)‖2 、つまり所定時間内の電力が2乗和計算部
321 ,322 が計算される。In this embodiment, the reception signal x of each channel
1 (k) and x 2 (k) are input to the step size determination unit 31, and the update step sizes μ 1 ,
μ 2 is calculated. In this case, the update step size is such that the convergence is fast for a channel in which the size of the received signal is large, and the update step is a small update step for a channel in which the size of the received signal is small. For example, as shown in FIG. 1B, power is used as the size of a received signal, and the sum of squares of each received signal x 1 (k) and x 2 (k) within a predetermined time {x ′ 1 (k)} 2. , ‖X '
2 (k) ‖ 2, i.e. power square sum calculation unit 32 1 in the predetermined time, 32 2 are calculated.
【0015】‖x′1(k)‖2=x1 2(k)+x
1 2(k−1)+…+x1 2(k−L′+1) ‖x′2(k)‖2=x2 2(k)+x2 2(k−1)+…
+x2 2(k−L′+1) この2乗和の計算区間L′は例えば16サンプル程度の
短い区間が正しく動作する点で好ましい。これら2乗和
は最小値検出部33に入力され、2乗和‖x′
1(k)‖2,‖x′2(k)‖2の最小値が検出され、
この例では何れの方が小さいかが検出される。その2乗
和の最小値(‖x′min(k)‖2とする)が、開平除
算部34へ供給される。一方、各チャネルの2乗和‖
x′1(k)‖2 ,‖x′2(k)‖2が加算部35で
加算され、この加算値が開平除算部34に入力される。
開平除算部34では下記の計算が行われる。‖X '1(K) ‖Two= X1 Two(K) + x
1 Two(K-1) + ... + x1 Two(K−L ′ + 1) ‖x ′Two(K) ‖Two= XTwo Two(K) + xTwo Two(K-1) + ...
+ XTwo Two(K−L ′ + 1) The calculation section L ′ of the sum of squares is, for example, about 16 samples.
Short sections are preferred in that they operate correctly. These sum of squares
Is input to the minimum value detection unit 33 and the sum of squares ‖x ′
1(K) ‖Two, ‖X 'Two(K) ‖TwoIs found,
In this example, which is smaller is detected. Its square
The minimum value of the sum (‖x ′min(K) ‖Two) But Kaikai removal
The data is supplied to the calculation unit 34. On the other hand, the sum of squares of each channel ‖
x '1(K) ‖Two , ‖X 'Two(K) ‖TwoIs the addition unit 35
The sum is added to the square root division unit 34.
The square root division unit 34 performs the following calculation.
【0016】 μmin=√‖x′min(k)‖2/√(‖x′1(k)‖2+‖x′2(k )‖2) (24) 出力部36では2乗和の最小値‖x′min(k)‖2と
対応するチャネルに対する更新ステップサイズをμmin
として出力し、その他のチャネルに対する更新ステップ
サイズをμ=1として出力する。この例では2チャネル
の場合であるから‖x′1(k)‖2<‖x′
2(k)‖2であれば、μ1=μmin ,μ2=1とし、‖
x′1(k)‖2>‖x′2(k)‖2であれば、μ1
=1,μ2=μminを出力する。Μ min = {x ′ min (k)} 2 / {({x ′ 1 (k)} 2 + {x ′ 2 (k)} 2 ) (24) The output unit 36 calculates the sum of squares. The update step size for the channel corresponding to the minimum value {x ′ min (k)} 2 is μ min
And the update step size for the other channels is output as μ = 1. Since this example is for two channels, {x ′ 1 (k)} 2 <{x ′
If 2 (k) ‖ 2, μ 1 = μ min, and mu 2 = 1, ‖
If x '1 (k) || 2>‖x' a 2 (k) || 2, μ 1
= 1, μ 2 = μ min are output.
【0017】これら更新ステップサイズμ1 ,μ2 はそ
れぞれ反響路推定部191 ,192へ供給され、反響路
推定部191 ,192 にはそれぞれ受話信号x
1 (k),x 2 (k)と残留反響信号e(k)が入力さ
れ、反響路推定部191 ,192 ではそれぞれ式(2
1)、(22)と対応して次式が計算される。 h^1(k+1)=h^1(k)+0.5μ1e(k)x1(k)/‖x1 (k)‖2 (25) h^2(k+1)=h^2(k)+0.5μ2e(k)x2(k)/‖x2 (k)‖2 (26) これらh^1(k+1),h^2(k+1)が疑似反
響路181 ,182 にそれぞれインパルス応答として設
定される。つまり、前回のインパルス応答h^
1(k),h^2(k)がそれぞれ式(25)、(2
6)の右辺第2項目の修正ベクトルにより修正されるこ
とになる。この修正ベクトルの演算は例えば式(25)
についてみれば、図2に示すように受話信号x1(k)
が入力されてベクトル生成部41で式(23)に示した
受話信号ベクトルx1(k)が生成され、また2乗和
回路42でベクトルx1(k)の各要素の2乗和‖
x1(k)‖ 2が計算され、これらx1(k),‖
x1(k)‖2と、更新ステップサイズμ 1 と残留反響
信号e(k)とが修正ベクトル演算部43に入力されて
式(25)の右辺第2項の演算が行われる。The update step size μ1, ΜTwoHaso
Each echo path estimation unit 191, 19TwoSupplied to the reflection path
Estimator 191, 19TwoHas the received signal x
1(K), x Two(K) and the residual echo signal e (k) are input.
Reverberation path estimation unit 191, 19TwoThen, the formula (2)
The following equation is calculated corresponding to (1) and (22). h ^1(K + 1) = h ^1(K) + 0.5μ1e (k) x1(K) / ‖x1 (K) ‖Two (25) h ^Two(K + 1) = h ^Two(K) + 0.5μTwoe (k) xTwo(K) / ‖xTwo (K) ‖Two (26) These h ^1(K + 1), h ^Two(K + 1) is pseudo anti
Hibiki 181, 18TwoEach as an impulse response.
Is determined. That is, the previous impulse response h ^
1(K), h ^Two(K) can be expressed by equations (25) and (2), respectively.
6) Correction by the correction vector of the second item on the right side
And The calculation of the correction vector is performed by, for example, Expression (25)
, The received signal x as shown in FIG.1(K)
Is input to the vector generation unit 41 as shown in Expression (23).
Received signal vector x1(K) is generated and the sum of squares
The vector x in the circuit 421Sum of squares of each element of (k) ‖
x1(K) ‖ TwoIs calculated and these x1(K), ‖
x1(K) ‖TwoAnd the update step size μ 1And residual echo
The signal e (k) is input to the correction vector operation unit 43 and
The calculation of the second term on the right side of Expression (25) is performed.
【0018】このように更新ステップサイズμ1 ,μ2
として、パワーの小さい方を式(24)により計算した
値を用い、他方を1とすることにより、受話信号x
1(k),x2 (k)の平均パワーに差があり、話者の
交代などでその平均パワー比が変化して、疑似反響路が
真の反響路に速い速度で近ずく、その計算機シミュレー
ションの結果を図3に示す。受話信号x1(k)とx
2(k)とのパワー比が6dB、疑似反響路181 ,1
82 のタップ数が1024、2乗和回路321 ,322
における加算サンプル数が16の場合である。As described above, the update step sizes μ 1 and μ 2
By using the value calculated by equation (24) for the smaller power and setting the other to 1, the received signal x
The average power ratio of 1 (k) and x 2 (k) is different, and the average power ratio changes due to the change of speakers, etc., and the pseudo echo path approaches the true echo path at a high speed. FIG. 3 shows the result of the simulation. Received signals x 1 (k) and x
2 The power ratio with (k) is 6 dB, and the pseudo echo path 18 1 , 1
8 number 2 taps 1024,2 square sum circuit 32 1, 32 2
Is the case where the number of added samples is 16.
【0019】実線がこの実施形態の場合であり、破線が
学習同定法を用いた従来技術の場合である。この図より
この発明によれば収束速度が速く、従って、話者が比較
的頻繁に交代してもこれに追従して、反響消去が行われ
る。上述において受話チャネルが3つ以上の場合にもこ
の発明を適用でき、この場合は各チャネルの受話信号x
1(k),…,xN(k)の所定時間の2乗和‖x′1
(k)‖2,…,‖x′N(k)‖2の最小のもの(‖
x′min(k)‖2とする)を求め、その最小値のチャ
ネルについては μmin=√‖x′min(k)‖2/√(‖x′1(k)
‖2+…+‖x′N (k)‖2 ) を計算して、これを更新ステップサイズとし、その他の
チャネルのステップサイズは全て1とすればよい。上述
においてはステップサイズ決定部31で2乗和が最小の
ものを求め、そのチャネルの更新ステップサイズを決定
し、他のチャネルのステップサイズを1とした。このよ
うにしたのはもともと、受話信号の大きさが大きけれ
ば、更新ステップサイズを1とすることにより、比較的
高速に収束するが、受話信号の大きさが小さいチャネル
が存在すると、それと対応する反響路の推定を正しく行
うことが困難になり、これに影響されて他の受話信号の
大きさが大きいチャネルと対応する反響路の推定が正し
く行われなくなるため、受話信号の大きさが小さいチャ
ネルの反響路推定の影響を受け難いように、その更新ス
テップサイズを小さくするためである。従って例えば受
話チャネルが3つの場合に、1つのチャネルの受話信号
の大きさは比較的大きく、他の2つのチャネルの受話信
号の大きさが小さければ、大きさが大きいチャネルの更
新ステップサイズは1とし、他の2つの大きさが小さい
受話信号の更新ステップサイズを1より小さい値、例え
ば式(24)と対応した式により求めてもよい。要は各
チャネルの受話信号の所定時間の大きさ(例えば2乗
和)を求め、これら大きさからその大きいもの、つまり
雑音に影響されず反響路推定を行うことができるチャネ
ルには収束の速いように更新ステップサイズを例えば1
とし、大きさが小さいもの、つまり雑音に影響されて、
反響路推定がうまくゆかないようなチャネルには大きさ
が大きいチャネルに対する更新ステップサイズより小さ
い更新ステップサイズとすればよい。大きさとしては電
力のみならず振幅を用いてもよい。The solid line is the case of this embodiment, and the broken line is
This is the case of the prior art using the learning identification method. From this figure
According to the present invention, the convergence speed is high, so that the speaker can compare
Even after a frequent change, echo cancellation is performed following this
You. This also applies to the case where there are three or more receiving channels in the above.
In this case, the received signal x of each channel can be applied.
1(K), ..., xN(K) Sum of squares of predetermined time ‖x '1
(K) ‖Two, ..., ‖x 'N(K) ‖TwoThe smallest of (‖
x 'min(K) ‖TwoAnd find the minimum value
Μ formin= √‖x 'min(K) ‖Two/ √ (‖x '1(K)
‖Two+ ... + ‖x 'N (K) ‖Two), And use this as the update step size.
All channel step sizes may be one. Above
In the step size determination unit 31,
And determine the update step size for that channel
Then, the step sizes of the other channels were set to 1. This
Originally, the size of the received signal was large.
For example, by setting the update step size to 1,
A channel that converges quickly but has a small received signal size
If there is, the estimation of the corresponding echo path is performed correctly.
This makes it difficult to
The estimation of the echo path corresponding to the large channel is correct
Will not be performed properly, and the
Update so that it is less susceptible to the
This is to reduce the step size. So, for example,
When there are three talk channels, the received signal of one channel
Is relatively large, and the other two channels receive
If the size of the signal is small, updating the larger channel
The new step size is 1 and the other two are smaller
Set the update step size of the received signal to a value smaller than 1, for example
For example, it may be obtained by an equation corresponding to equation (24). In short, each
The size of a predetermined time (for example, the square of the reception signal of the channel)
Sum), and from these sizes, the larger one, that is,
Channel that can perform echo path estimation without being affected by noise
The update step size is set to 1
And it is small, that is, influenced by noise,
Large for channels where echo path estimation is not successful
Is smaller than the update step size for large channels
Update step size. The size is electric
Not only force but also amplitude may be used.
【0020】また収音(送話)チャネルが複数の場合、
図5に示した各Nチャネル反響消去装置221 ,…,2
2M についても各受話信号ごとに疑似反響路を通し、そ
の各反響路推定における更新ステップサイズを前述した
ように選定すればよい。また修正ベクトルを求める際
に、各チャネルごとにその受話信号をその電力で正規化
したものを用いる場合に限らず受話信号をその振幅で正
規化したものを用いてもよい。更にこの発明は学習同定
法に限らず射影法など他のアルゴリズムを用いる場合に
も適用できる。When there are a plurality of sound pickup (transmission) channels,
Each of the N-channel echo cancellers 22 1 ,..., 2 shown in FIG.
As for 2 M , a pseudo echo path is passed for each received signal, and the update step size in each echo path estimation may be selected as described above. When the correction vector is obtained, the received signal is not limited to the one obtained by normalizing the received signal for each channel with the power, but the received signal normalized by the amplitude may be used. Further, the present invention can be applied not only to the learning identification method but also to a case where another algorithm such as a projection method is used.
【0021】上述したこの発明による多チャネル反響消
去装置をコンピュータによりプログラムを実行させて機
能させることもできる。その場合この多チャネル反響消
去プログラムをCD−ROM、フロッピー(登録商標)
ディスク、磁気ディスクなどからコンピュータのプログ
ラムメモリにインストールし、あるいはそのプログラム
メモリに通信回線を介してダウンロードして、そのプロ
グラムメモリに格納された多チャネル反響消去プログラ
ムを実行することになる。The above-described multi-channel echo canceller according to the present invention can be operated by executing a program by a computer. In this case, the multi-channel echo canceling program is stored in a CD-ROM, a floppy (registered trademark).
The multi-channel echo canceling program stored in the program memory is installed by being installed from a disk, a magnetic disk, or the like into the program memory of the computer or downloaded to the program memory via a communication line.
【0022】[0022]
【発明の効果】以上述べたようにこの発明によれば多チ
ャネル受話信号のパワー比がある場合、相互相関が強く
ても収束速度を改善でき演算量の増加がない。As described above, according to the present invention, when there is a power ratio of a multi-channel received signal, even if the cross-correlation is strong, the convergence speed can be improved and the amount of calculation does not increase.
【図1】Aはこの発明の実施形態の機能構成例を示す
図、Bはそのステップサイズ決定部31の具体的機能構
成例を示す図である。FIG. 1A is a diagram illustrating a functional configuration example of an embodiment of the present invention, and FIG. 1B is a diagram illustrating a specific functional configuration example of a step size determination unit 31;
【図2】図1A中の反響路推定部191 中の一部の具体
的機能構成例を示す図。FIG. 2 shows a portion of a specific example functional configuration of the echo path estimation unit 19 in 1 in Figure 1A.
【図3】この発明の実施形態と従来技術における疑似反
響路の収束性の計算機シミュレーションの例を示す図。FIG. 3 is a diagram showing an example of a computer simulation of the convergence of a pseudo echo path in the embodiment of the present invention and the prior art.
【図4】従来の1チャネル反響消去装置の機能構成を示
す図。FIG. 4 is a diagram showing a functional configuration of a conventional one-channel echo canceller.
【図5】従来の多チャネル反響消去装置の機能構成を示
す図。FIG. 5 is a diagram showing a functional configuration of a conventional multi-channel echo canceller.
【図6】図5中のNチャネル反響消去装置の具体的機能
構成を示す図。FIG. 6 is a diagram showing a specific functional configuration of the N-channel echo canceller in FIG. 5;
【図7】従来改善された2チャネル反響消去装置の機能
構成を示す図。FIG. 7 is a diagram showing a functional configuration of a conventionally improved two-channel echo canceller.
Claims (6)
信号を、N個の反響路を模疑した疑似反響路を通してN
個の疑似反響信号を生成し、 これらN個の疑似反響信号を加算して、総合疑似反響信
号を求め、 その総合疑似反響信号を収音信号から差し引いて残留反
響信号を求め、 これらN個の受話信号のうち強度が最大の受話信号チャ
ネルの更新ステップサイズより、上記受話信号チャネル
以外の受話信号チャネルの更新ステップサイズを小と
し、 各チャネルごとにその受話信号をその受話信号で正規化
したものと、上記そのチャネルの更新ステップサイズと
上記残留反響信号と、によりそのチャネルの疑似反響路
の修正ベクトルを求め、 その修正ベクトルによりそのチャネルの疑似反響路の特
性を修正することを特徴とする多チャネル反響消去方
法。1. A reception signal of N channels (N is an integer of 2 or more) is passed through a pseudo echo path simulating N echo paths.
The pseudo echo signals are generated, and the N pseudo echo signals are added to obtain a total pseudo echo signal. The total pseudo echo signal is subtracted from the collected signal to obtain a residual echo signal. In the received signal, the update step size of the received signal channels other than the received signal channel is smaller than the updated step size of the received signal channel having the highest strength, and the received signal is normalized by the received signal for each channel. And a correction vector of the pseudo echo path of the channel based on the update step size of the channel and the residual echo signal, and correcting the characteristics of the pseudo echo path of the channel by the correction vector. Channel echo cancellation method.
イズは、各チャネルの受話信号の所定時間の2乗和を求
め、これら2乗和中の最小のものを求め、そのチャネル
の更新ステップサイズとして、その2乗和を全体チャネ
ルの2乗和の和で規格化した値と対応した値を用い、そ
の他のチャネルの更新ステップサイズを1とすることを
特徴とする請求項1記載の多チャネル反響消去方法。2. The update step size for each channel is obtained by calculating the sum of squares of the reception signal of each channel for a predetermined time, obtaining the minimum of these square sums, and defining the update step size of the channel as 2. The multi-channel echo cancellation method according to claim 1, wherein a value corresponding to a value obtained by normalizing the sum of squares by the sum of the sum of squares of all channels is used, and the update step size of other channels is set to 1. .
信号がそれぞれ入力され、疑似反響信号を生成して出力
するN個の疑似反響路と、 上記N個の疑似反響信号を加算して総合疑似反響信号を
生成する加算部と、 上記総合疑似反響信号を収音信号から差し引いて残留反
響信号を求める減算部と、 N個のチャネルの受話信号のうち強度が最大の受話信号
チャネルの更新ステップサイズより、上記受話信号チャ
ネル以外の受話信号チャネルの更新ステップサイズを小
さい値に決定して出力するステップサイズ決定部と、 各チャネルの受話信号とそのチャネルと上記残留反響信
号とが入力され、上記受話信号をその受話信号により正
規化して修正ベクトルを演算し、その修正ベクトルによ
り対応するチャネルの上記疑似反響路の特性を更新する
N個の反響路推定部と、 を具備する多チャネル反響消去装置。3. An N-channel (N is an integer of 2 or more) receiving signal is input, and N pseudo echo paths for generating and outputting pseudo echo signals are added to the N pseudo echo signals. An adder for generating a total pseudo echo signal by subtracting the total pseudo echo signal from the picked-up signal to obtain a residual echo signal; A step size determining unit that determines and updates the update step size of the reception signal channel other than the reception signal channel to a smaller value than the update step size, and receives the reception signal of each channel, the channel, and the residual echo signal. , Normalizing the received signal with the received signal to calculate a correction vector, and using the corrected vector to update the characteristics of the pseudo echo path of the corresponding channel. Multi-channel echo canceller comprising a number of echo path estimation unit.
ルの受話信号の所定時間内の2乗和‖x′1‖,…,
‖x′N‖を演算し、これら2乗和の最小値‖x′
min‖2 について√‖x′min ‖2 /√(‖x′1
‖+…+‖x′N ‖)を演算し、その結果μmin を、
‖x′min ‖2 が得られたチャネルの更新ステップサ
イズとし、その他のチャネルの更新ステップサイズμを
1として出力する機能を備え、 上記反響路推定部は当該チャネルの受話信号の所定時間
の各値を要素とするベクトルxn (k)を生成し、そ
のベクトルxn (k)の各要素の2乗和‖x n (k)
‖2 を演算し、 μn e(k)xn (k)/‖xn (k)‖2 を修正ベクトルとし演算する機能を備えることを特徴と
する請求項3記載の多チャネル反響消去装置。4. The method according to claim 1, wherein the step size determination unit is configured to:
Sum 乗 x 'of the received signal of the1‖ 、… 、
‖X 'N、, and the minimum value of these sums of squares ‖x ′
min‖TwoAbout √‖x 'min‖Two/ √ (‖x '1
‖ + ... + ‖x 'N‖), And as a result μminTo
‖X 'min‖TwoChannel update step
And update step size μ for other channels.
The echo path estimating unit has a function of outputting the received signal of the channel for a predetermined time.
Vector x with each value ofn(K), and
Vector xnSum of squares of each element of (k) ‖x n(K)
‖Two, And μne (k) xn(K) / ‖xn(K) ‖Two With the function of calculating as a correction vector
The multi-channel echo canceller according to claim 3.
タに実行させるための多チャネル反響消去プログラム。5. A multi-channel echo canceling program for causing a computer to execute the method according to claim 1.
ログラムが記録されたコンピュータ読み出し可能な記録
媒体。6. A computer-readable recording medium on which the multi-channel echo canceling program according to claim 5 is recorded.
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Cited By (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2007067855A (en) * | 2005-08-31 | 2007-03-15 | Nippon Telegr & Teleph Corp <Ntt> | Echo canceling method, echo canceling apparatus, program, recording medium |
| JP2007067854A (en) * | 2005-08-31 | 2007-03-15 | Nippon Telegr & Teleph Corp <Ntt> | Echo canceling method, echo canceling apparatus, program, recording medium |
| JP2011114696A (en) * | 2009-11-27 | 2011-06-09 | Nippon Telegr & Teleph Corp <Ntt> | Method, device and program for echo cancellation |
| JP2012529211A (en) * | 2009-06-02 | 2012-11-15 | コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ | Acoustic multichannel cancellation |
-
2001
- 2001-02-27 JP JP2001051388A patent/JP3628267B2/en not_active Expired - Fee Related
Cited By (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2007067855A (en) * | 2005-08-31 | 2007-03-15 | Nippon Telegr & Teleph Corp <Ntt> | Echo canceling method, echo canceling apparatus, program, recording medium |
| JP2007067854A (en) * | 2005-08-31 | 2007-03-15 | Nippon Telegr & Teleph Corp <Ntt> | Echo canceling method, echo canceling apparatus, program, recording medium |
| JP2012529211A (en) * | 2009-06-02 | 2012-11-15 | コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ | Acoustic multichannel cancellation |
| JP2011114696A (en) * | 2009-11-27 | 2011-06-09 | Nippon Telegr & Teleph Corp <Ntt> | Method, device and program for echo cancellation |
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| Publication number | Publication date |
|---|---|
| JP3628267B2 (en) | 2005-03-09 |
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