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JP2004320204A - Method and apparatus for echo cancellation and echo cancellation program - Google Patents

Method and apparatus for echo cancellation and echo cancellation program Download PDF

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Publication number
JP2004320204A
JP2004320204A JP2003108875A JP2003108875A JP2004320204A JP 2004320204 A JP2004320204 A JP 2004320204A JP 2003108875 A JP2003108875 A JP 2003108875A JP 2003108875 A JP2003108875 A JP 2003108875A JP 2004320204 A JP2004320204 A JP 2004320204A
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filter
error
signal
coefficient
reproduction
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JP4156428B2 (en
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Suehiro Shimauchi
末廣 島内
Yoichi Haneda
陽一 羽田
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Nippon Telegraph and Telephone Corp
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Nippon Telegraph and Telephone Corp
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Abstract

<P>PROBLEM TO BE SOLVED: To accelerate a speed for approaching a filter coefficient used in a filter processing means to true characteristics for echo cancellation. <P>SOLUTION: An apparatus for the echo cancellation includes a disturbance ratio estimating means for sequentially estimating a disturbance ratio representing the ratio of the amplitude of the disturbance signal component, except an echo signal included in a sound pickup signal to the amplitude of simulated error outputted from a simulated error output means at each frequency range; and a second updating quantity regulating means for multiplying the characteristics error of the filter coefficient or the characteristic error of the filter coefficient multiplied by a first regulating coefficient by a second regulating coefficient taking a predetermined value from "0" at each frequency range, in such a manner the larger the disturbance ratio is, the larger the second regulating coefficient in response to the amplitude of the disturbance ratio in the front or rear stage of a first updating quantity regulating means. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

【0001】
【発明の属する技術分野】
本発明は、スピーカのような音響再生手段からマイクロホンのような音響収音手段へと回り込む反響を消去するための反響消去装置に関する。
【0002】
【従来の技術】
音響再生手段から音響収音手段へ回り込む反響を消去する反響消去装置は、図6のように接続される。図6に示す10は従来の反響消去装置を示す。従来の反響消去装置10内では、音響再生手段1と音響収音手段2間の反響路のインパルス応答hを推定し、推定したインパルス応答h´と再生信号入力端子3に入力された再生信号xの畳込み演算h´xを生成し、実際の反響信号yから減算することで、反響消去信号出力端子4に反響消去信号eを得る。
しかし、推定したインパルス応答と再生信号畳込み演算には、多くの演算量を必要とし、実装上の問題となっている。
【0003】
近年、この問題を解決するため、再生信号や反響信号を一旦周波数領域変換し、反響路のインパルス応答の周波数領域変換に対応したパラメータを推定し、畳込みの代わりに乗算処理を用いたり(非特許文献1)、あるいはより小さい畳込み演算に分割したりする(特許文献1)などして、演算量を削減する方法が提案されている。周波数領域変換の例としては、(高速)離散フーリエ変換、(高速)離散コサイン変換、(高速)ハートレー変換などがある。
図6で音響再生手段1としてスピーカを挙げているが、音響再生手段1としては、再生前段の増幅器やバッファも含む。また、同様に音響収音手段2は、マイクロホンの後段の増幅器やバッファも含む。
【0004】
また、(非特許文献1)などに記載されている構成を模式的に図7に示す。図7の構成は以下の手段を含んでいる。すなわち、再生信号入力端子3から入力されて音響再生手段1へと出力される再生信号を入力し一定時間蓄積し再生信号列を得る再生信号入力手段101と、音響再生手段1と同一空間に存在する音響収音手段2から収音信号を入力し一定時間蓄積し収音信号列を得る収音信号入力手段102と、再生信号列を周波数領域変換し再生信号変換列を得る再生信号変換手段103と、再生信号変換列を周波数領域毎に入力しフィルタ処理することにより、音響再生手段1から音響収音手段2へと回り込む反響信号の周波数領域変換列を模擬する模擬反響信号変換列を生成するフィルタ処理手段104と、模擬反響信号変換列と収音信号列とを入力し模擬反響信号変換列の模擬誤差を出力する模擬誤差出力手段105と、再生信号変換列と模擬誤差を入力し、フィルタ処理手段104のフィルタ係数の周波数領域毎の特性誤差を計算するフィルタ誤差計算手段106と、フィルタ誤差計算手段106が計算した特性誤差に周波数領域毎に第一調整係数を乗じる第一更新量調整手段107と、第一調整係数を乗じられた特性誤差を加えることによりフィルタ処理手段104のフィルタ係数を更新するフィルタ更新手段108とを含む。
【0005】
【特許文献1】
特開平9−116472号(図6)
【非特許文献1】
Simon Haykin 著 「適応フィルタ理論」 科学技術出版、2001年1月10日、pp.500−541
【0006】
【発明が解決しようとする課題】
図7の構成において、フィルタ誤差計算手段106が、反響消去のための真の特性とフィルタ処理手段104で用いるフィルタ係数との特性誤差を正しく計算し出力できれば、フィルタ誤差計算手段106から出力される特性誤差に対して、第一更新量調整手段107が乗じる第一調整係数は1でよい。すなわち最も速く特性誤差を修正できる。
然し乍ら、実際にフィルタ誤差計算手段106が特性誤差を正しく計算し出力することは困難である。その原因の一つは、音響収音手段2が収音した収音信号に含まれる反響信号以外の周囲騒音、回路雑音などの外乱成分の影響である。この外乱成分が、フィルタ誤差計算手段106が出力する特性誤差の計算に悪影響を及ぼす。従って、この影響を避けるため、第一更新量調整手段107を導入し、第一調整係数を小さく設定したのが従来の技術である。
【0007】
従来は第一係数を小さく設定するため、フィルタ処理手段104で用いるフィルタ係数が反響消去のための真の特性に近づく速度は低くなり、完全な反響消去を達成するまでに長い時間を要していた。
本発明の目的は、上記のように、音響収音手段2が収音した収音信号に含まれる外乱成分の影響を最小限とし、第一更新調整手段107が極力大きな第一調整係数をとることができ、フィルタ処理手段104で用いるフィルタ係数が反響消去のための真の特性に近づく速度を高めることにある。
【0008】
【課題を解決するための手段】
この発明の請求項1では音響再生手段へと出力される再生信号を入力し一定時間蓄積し再生信号列を得る再生信号入力処理と、音響再生手段と同一空間に存在する音響収音手段から収音信号を入力し一定時間蓄積し収音信号列を得る収音信号入力処理と、再生信号列を周波数領域変換し再生信号変換列を得る再生信号変換処理と、再生信号変換列を周波数領域毎に入力しフィルタ処理することにより音響再生手段から音響収音手段へと回り込む反響信号の周波数領域変換列を模擬する模擬反響信号変換列を生成するフィルタ処理と、模擬反響信号変換列と収音信号列とを入力し模擬反響信号変換列の模擬誤差を出力する模擬誤差出力処理と、再生信号変換列と模擬誤差を入力し前記フィルタ処理のフィルタ係数の周波数領域毎の特性誤差を計算するフィルタ誤差計算処理と、フィルタ誤差計算処理が計算した特性誤差に周波数領域毎に第一調整係数を乗じる第一更新量調整処理と、第一調整係数を乗じられた特性誤差を加えることによりフィルタ処理のフィルタ係数を更新するフィルタ更新処理とを実行する反響消去方法において、模擬誤差出力処理で出力される模擬誤差の大きさに対する収音信号に含まれる反響信号以外の外乱信号成分の大きさの比率を表わす外乱比率を周波数領域毎に逐次推定する外乱比率推定処理と、第一更新量調整処理の前段又は後段において、フィルタ係数の特性誤差又は前記第一調整係数を乗じたフィルタ係数の特性誤差に、0から所定値の間で外乱比率の大きさに応じ、外乱比率が大きい程小さい値をとる第二調整係数を周波数領域毎に乗じる第二更新量調整処理とを実行する反響消去方法を提案する。
【0009】
この発明の請求項2では音響再生手段へと出力される再生信号を入力し一定時間蓄積し再生信号列を得る再生信号入力手段と、音響再生手段と同一空間に存在する音響収音手段から収音信号を入力し一定時間蓄積し収音信号列を得る収音信号入力手段と、再生信号列を周波数領域変換し再生信号変換列を得る再生信号変換手段と、再生信号変換列を周波数領域毎に入力しフィルタ処理することにより音響再生手段から音響収音手段へと回り込む反響信号の周波数領域変換列を模擬する模擬反響信号変換列を生成するフィルタ処理手段と、模擬反響信号変換列と収音信号列とを入力し模擬反響信号変換列の模擬誤差を出力する模擬誤差出力手段と、再生信号変換列と模擬誤差を入力しフィルタ処理手段のフィルタ係数の周波数領域毎の特性誤差を計算するフィルタ誤差計算手段と、フィルタ誤差計算手段が計算した特性誤差に周波数領域毎に第一調整係数を乗じる第一更新量調整手段と、第一調整係数を乗じられた特性誤差を加えることによりフィルタ処理手段のフィルタ係数を更新するフィルタ更新手段とを有する反響消去装置において、模擬誤差出力手段が出力する模擬誤差の大きさに対する収音信号に含まれる反響信号以外の外乱信号成分の大きさの比率を表わす外乱比率を周波数領域毎に逐次推定する外乱比率推定手段と、第一更新量調整手段の前段又は後段において、フィルタ係数の特性誤差又は第一調整係数を乗じたフィルタ係数の特性誤差に、0から所定値の間で外乱比率の大きさに応じ、外乱比率が大きい程小さい値をとる第二調整係数を周波数領域毎に乗じる第二更新量調整手段とを有する反響消去装置を提案する。
【0010】
この発明の請求項3ではコンピュータが解読可能な符号列によって記述され、コンピュータに請求項1記載の反響消去方法を実行させる反響消去プログラムを提案する。
作用
この発明の構成によれば音響収音手段2が収音した収音信号に含まれる外乱成分が、フィルタ誤差計算手段が出力するフィルタ係数の特性誤差に及ぼした比率に応じて、計算されたフィルタ係数の特性誤差の不確かな部分を小さくすることができる。このため、第一更新量調整手段では極力大きな第一調整係数をとることができ、フィルタ処理手段で用いるフィルタ係数が反響消去のための真の近づく速度を高めることができる。
【0011】
外乱比率推定手段における、模擬誤差出力手段が出力する模擬誤差の大きさに対する音響収音手段からの収音信号に含まれる反響信号以外の外乱信号成分の大きさの比率を表わす外乱比率の推定は、次のように実現できる。ここで、N分割した周波数領域ω(n=1…N)における模擬誤差出力手段が出力する模擬誤差の大きさを|E(ω)|とし、周波数領域ωにおける外乱の大きさの推定値を|N(ω)|とする。外乱推定値|N(ω)|は、無音声区間等に事前測定した値などを用いるものとする。このとき外乱比率DR(ω)の計算は、
DR(ω)=|N(ω)|/|E(ω)| …(1)
として計算できる。ここで、外乱比率DR(ω)の値は、0から所定値Tで例えばT=1の値をとるものとし、実使用環境で、この範囲を逸脱した場合は、0から所定値の範囲になるよう数値の打ち切りを行なう。DR(ω)の値は、|E(ω)|に対する|N(ω)|の割合が大きくなるほど、大きな値を与えるようにすれば、(1)式以外に基づいてもよい。
【0012】
また、第二更新量調整手段の第二調整係数β(ω)は0から所定値例えば1までの値をとり、上記の外乱比率DR(ω)を用いて、
β(ω)=(T−DR(ω1/p …(2)
のように決定できる。ここで、pは正の実数とする。また、(2)式の他、図4のAに示すように、DR(ω)の値に対して滑らかに対応させたり、図4のBにしめすように、不連続に対応するようなものであっても、DR(ω)の値が大きくなるほど、β(ω)の値が小さくなるようなものであれば、本発明の効果を得られる実施例に含まれる。DR(ω)が(1)式以外の形で求められた場合でも、ここに示したDR(ω)とβ(ω)の対応の中から適切なものを選択可能である。
【0013】
【発明の実施の形態】
図1にこの発明による反響消去装置の一実施例を示す。図1に示す反響消去装置を反響消去装置方法と共に説明する。
図1に示す100はこの発明による反響消去装置を示す。この発明による反響消去装置100は図6に示した従来の反響消去装置10に以下の手段が付加される。すなわち、模擬誤差出力手段105が出力する模擬誤差の大きさに対する音響収音手段2からの収音信号に含まれる反響信号以外の外乱信号成分の大きさの比率を表わす外乱比率DR(ω)を周波数領域毎に逐次推定する外乱比率推定手段111と、実際にフィルタ誤差計算手段106が出力するフィルタ係数の特性誤差に、外乱比率推定手段111が推定した外乱比率の大きさに応じ、外乱比率が0パーセントの時は所定値、例えば1、外乱比率が100パーセントの時は0、その間の外乱比率の時は0から所定値の間の値をとる第二調整係数β(ω)を周波数領域毎に乗じる第二更新量調整手段110とを含む。
【0014】
この構成により、音響収音手段2が収音した収音信号に含まれる外乱成分が、フィルタ誤差計算手段106が出力するフィルタ係数の特性誤差に及ぼした比率に応じて、計算されたフィルタ係数の特性誤差の不確かな部分の値を小さくすることができる。このため、第一更新量調整手段107では極力大きな第一調整係数をとることができ、フィルタ処理手段104で用いるフィルタ係数が反響消去のための真の特性に近づく速度を高めることができる。
【0015】
発明の変形実施例
図1に示した実施例では第二更新量調整手段110を第一更新量調整手段107の前段に設けた例を説明したが、図2に示すように第二更新量調整手段110を第一更新量調整手段107の後段に配置しても図1に示した実施例と同様の作用効果を得ることができる。
【0016】
また、図3に示すように、フィルタ誤差計算手段106とフィルタ更新手段108の間に第三更新量調整手段112を配置し、この第三更新量調整手段112で第一調整係数と第二調整係数との例えば積を求め、この積の値を第三調整係数とし、この第三調整係数を周波数領域毎のフィルタ係数の特性誤差に乗じる構成としても図1に示した実施例と同様の作用効果を得ることができる。
以上説明したこの発明による反響消去装置および反響消去方法はコンピュータが解読可能な符号列によって記述された音響消去プログラムをコンピュータに実行させて実現される。図5にこの発明による音響消去プログラムをコンピュータ20にインストールした状況を示す。
【0017】
コンピュータ20はよく知られているように、中央演算処理装置(CPU)21と、コンピュータの立上げおよび立下げ等を制御する基本プログラム等を記録した読み出専用メモリ(ROM)22と、データを一時記録する他に、この発明の反響消去方法を実行するためのプログラムを格納する読み出し、読み出し・書き込み可能なメモリ(RAM)23と、入力ポート24、出力ポート25等により構成することができる。
入力ポート24には再生信号入力端子3と、音響収音手段2が接続され、再生信号Xと、収音信号とが入力される。また、出力ポート25には音響再生手段1と反響消去信号出力端子4とが接続され、再生信号に対応する音響と、反響消去信号e=y−y´が出力される。
【0018】
読み出し、書き込み可能なメモリ23にはデータ格納領域23Aの他に、再生信号入力手段101として動作する再生信号入力処理プログラム23Bと、収音信号入力手段102として動作する収音信号入力処理プログラム23Cと、再生信号変換手段103として動作する再生信号変換処理プログラム23Dと、フィルタ処理手段104として動作するフィルタ処理プログラム23Eと、模擬誤差出力手段105として動作する模擬誤差出力処理プログラム23Fと、フィルタ誤差計算手段106として動作するフィルタ誤差計算処理プログラム23Gと、第一更新量調整手段107として動作する第一更新量調整処理プログラム23Hと、フィルタ更新手段108として動作するフィルタ更新処理プログラム23Iと、第二更新量調整手段110として動作する第二更新量調整処理プログラム23Jと、外乱比率推定手段111として動作する外乱比率推定処理プログラム23Kとがインストールされ、これらの各プログラムが中央演算処理装置21によって解読されて反響消去動作を実行する。読み出し、書き込み可能なメモリ23にインストールされた各プログラム23B〜23Kは予めコンピュータが読み出し可能な磁気ディスク或はCD−ROM等に記録され、これらの記録媒体からインストールされるか又は通信回路を通じてインストールされる。
【0019】
【発明の効果】
本発明による反響消去装置は、フィルタ処理手段で用いるフィルタ係数が反響消去のための真の特性に近づく速度の低下を抑えながら、音響収音手段が収音した収音信号に含まれる外乱成分の影響を低減でき、反響消去性能を高めることができる。
【図面の簡単な説明】
【図1】この発明による反響消去装置の一実施例を説明するためのブロック図。
【図2】この発明による反響消去装置の他の実施例を説明するためのブロック図。
【図3】この発明による反響消去装置の更に他の実施例を説明するためのブロック図。
【図4】この発明の要部の動作を説明するための特性曲線図。
【図5】この発明による反響消去装置をコンピュータで実現した場合の構成を説明するための構成概念図。
【図6】反響消去装置の概要を説明するためのブロック図。
【図7】従来の反響消去装置を説明するためのブロック図。
【符号の説明】
1 音響再生手段 105 模擬誤差出力手段
2 音響収音手段 106 フィルタ誤差計算手段
3 再生信号入力端子 107 第一更新量調整手段
4 反響消去信号出力端子 108 フィルタ更新手段
100 反響消去装置 110 第二更新量調整手段
101 再生信号入力手段 111 外乱比率推定手段
102 収音信号入力手段
103 再生信号変換手段
104 フィルタ処理手段
[0001]
TECHNICAL FIELD OF THE INVENTION
The present invention relates to a reverberation canceling device for canceling reverberation circulating from a sound reproducing unit such as a speaker to a sound pickup unit such as a microphone.
[0002]
[Prior art]
A reverberation canceling device for canceling reverberation circulating from the sound reproducing means to the sound collecting means is connected as shown in FIG. Reference numeral 10 shown in FIG. 6 indicates a conventional echo canceller. In the conventional echo canceller 10, the impulse response h of the echo path between the sound reproduction means 1 and the sound pickup means 2 is estimated, and the estimated impulse response h 'and the reproduction signal x input to the reproduction signal input terminal 3 are obtained. generating a convolution h'* x, is subtracted from the actual echo signal y, to obtain a echo cancellation signal e to the echo cancellation signal output terminal 4.
However, the estimated impulse response and the reproduction signal convolution operation require a large amount of operation, which is a mounting problem.
[0003]
In recent years, in order to solve this problem, the reproduction signal and the reverberation signal are once frequency-domain transformed, parameters corresponding to the frequency domain transformation of the impulse response of the reverberation path are estimated, and multiplication processing is used instead of convolution (non-convolution). There has been proposed a method of reducing the amount of calculation by, for example, dividing the image into smaller convolution operations (Patent Document 1) or a smaller convolution operation (Patent Document 1). Examples of frequency domain transforms include (fast) discrete Fourier transform, (fast) discrete cosine transform, and (fast) Hartley transform.
Although a speaker is shown as the sound reproducing means 1 in FIG. 6, the sound reproducing means 1 also includes an amplifier and a buffer in a stage before reproduction. Similarly, the sound pickup means 2 also includes an amplifier and a buffer at the subsequent stage of the microphone.
[0004]
FIG. 7 schematically shows a configuration described in (Non-Patent Document 1) and the like. The configuration of FIG. 7 includes the following means. That is, a reproduction signal input means 101 for inputting a reproduction signal input from the reproduction signal input terminal 3 and output to the sound reproduction means 1 and accumulating the reproduction signal for a certain period of time to obtain a reproduction signal sequence; A sound pickup signal input means 102 for inputting a sound pickup signal from the sound pickup means 2 and accumulating it for a certain period of time to obtain a sound pickup signal sequence, and a reproduction signal conversion means 103 for frequency domain transforming the reproduction signal sequence to obtain a reproduction signal conversion sequence. Then, a simulated reverberation signal conversion sequence that simulates a frequency domain conversion sequence of a reverberation signal circulating from the sound reproducing unit 1 to the sound pickup unit 2 is generated by inputting the reproduction signal conversion sequence for each frequency domain and performing a filtering process. Filter processing means 104, simulation error output means 105 for inputting the simulated reverberation signal conversion sequence and the picked-up signal sequence, and outputting a simulated error of the simulated reverberation signal conversion sequence; A filter error calculating means for inputting and calculating a characteristic error of the filter coefficient of the filter processing means for each frequency domain, and a first adjusting coefficient for multiplying the characteristic error calculated by the filter error calculating means for each frequency domain. It includes an update amount adjustment unit 107 and a filter update unit 108 that updates the filter coefficient of the filter processing unit 104 by adding a characteristic error multiplied by the first adjustment coefficient.
[0005]
[Patent Document 1]
JP-A-9-116472 (FIG. 6)
[Non-patent document 1]
Simon Haykin, "Adaptive Filter Theory," Science and Technology Publishing, January 10, 2001, pp. 139-214. 500-541
[0006]
[Problems to be solved by the invention]
In the configuration of FIG. 7, if the filter error calculation means 106 can correctly calculate and output the characteristic error between the true characteristic for echo cancellation and the filter coefficient used in the filter processing means 104, the filter error calculation means 106 outputs The first adjustment coefficient by which the first update amount adjustment unit 107 multiplies the characteristic error may be one. That is, the characteristic error can be corrected most quickly.
However, it is difficult for the filter error calculation means 106 to correctly calculate and output the characteristic error. One of the causes is the influence of disturbance components such as ambient noise and circuit noise other than the reverberation signal included in the sound pickup signal picked up by the sound pickup means 2. This disturbance component has an adverse effect on the calculation of the characteristic error output from the filter error calculation means 106. Therefore, in order to avoid this influence, the conventional technique introduces the first update amount adjusting means 107 and sets the first adjustment coefficient small.
[0007]
Conventionally, since the first coefficient is set to be small, the speed at which the filter coefficient used in the filter processing means 104 approaches the true characteristic for echo cancellation is low, and it takes a long time to achieve complete echo cancellation. Was.
As described above, the object of the present invention is to minimize the influence of the disturbance component included in the sound pickup signal picked up by the sound pickup means 2 and to make the first update adjustment means 107 take the largest possible first adjustment coefficient. The purpose is to increase the speed at which the filter coefficient used in the filter processing means 104 approaches the true characteristic for echo cancellation.
[0008]
[Means for Solving the Problems]
According to the first aspect of the present invention, a reproduction signal input process for inputting a reproduction signal output to the sound reproduction means and accumulating the reproduction signal for a certain period of time to obtain a reproduction signal sequence, and collecting the reproduction signal from the sound pickup means existing in the same space as the sound reproduction means. A sound collection signal input process for inputting a sound signal and accumulating it for a certain period of time to obtain a sound collection signal sequence, a reproduction signal conversion process for frequency domain transforming a reproduction signal sequence to obtain a reproduction signal conversion sequence, and a reproduction signal conversion sequence for each frequency domain Filter processing to generate a simulated reverberation signal conversion sequence that simulates a frequency domain conversion sequence of a reverberation signal that circulates from the sound reproduction means to the sound pickup means by inputting and filtering the simulated reverberation signal conversion sequence and the picked-up signal And a simulation error output process for outputting a simulation error of a simulated reverberation signal conversion sequence and a simulation error output process for inputting a reproduction signal conversion sequence and a simulation error, and measuring a characteristic error of a filter coefficient of the filter process for each frequency domain. A filter error calculation process, a first update amount adjustment process of multiplying the characteristic error calculated by the filter error calculation process by a first adjustment coefficient for each frequency domain, and a filter by adding the characteristic error multiplied by the first adjustment coefficient. In the reverberation elimination method of performing the filter update process of updating the filter coefficient of the process, the magnitude of the disturbance signal component other than the reverberation signal included in the sound pickup signal with respect to the magnitude of the simulation error output in the simulation error output process. A disturbance ratio estimation process for sequentially estimating a disturbance ratio representing a ratio for each frequency domain, and a characteristic error of a filter coefficient or a characteristic error of a filter coefficient multiplied by the first adjustment coefficient in a stage before or after the first update amount adjustment process. The second adjustment coefficient is multiplied for each frequency domain by a second adjustment coefficient that takes a smaller value as the disturbance ratio increases according to the magnitude of the disturbance ratio between 0 and a predetermined value. Suggest echo cancellation how to run the new amount adjustment process.
[0009]
According to a second aspect of the present invention, a reproduction signal input means for inputting a reproduction signal output to the sound reproduction means and accumulating the reproduction signal for a certain period of time to obtain a reproduction signal sequence, and a sound pickup means existing in the same space as the sound reproduction means. A sound signal input means for inputting a sound signal and accumulating the sound signal for a certain period of time to obtain a sound signal sequence; a reproduction signal conversion means for frequency domain converting the reproduction signal sequence to obtain a reproduction signal conversion sequence; and a reproduction signal conversion sequence for each frequency region. Filter processing means for generating a simulated reverberation signal conversion sequence that simulates a frequency domain conversion sequence of a reverberation signal circulating from the sound reproducing means to the sound pickup means by inputting and filtering the simulated reverberation signal conversion sequence and sound pickup A simulation error output means for inputting a signal sequence and outputting a simulation error of a simulation echo signal conversion sequence, and a characteristic error for each frequency domain of a filter coefficient of a filter processing means for inputting a reproduction signal conversion sequence and the simulation error A filter error calculating means for calculating, a first update amount adjusting means for multiplying the characteristic error calculated by the filter error calculating means by a first adjusting coefficient for each frequency domain, and adding a characteristic error multiplied by the first adjusting coefficient. An echo canceller having a filter updating means for updating a filter coefficient of the filter processing means, wherein the magnitude of a disturbance signal component other than the reverberation signal included in the picked-up signal with respect to the magnitude of the simulation error output by the simulation error output means is determined. A disturbance ratio estimating means for sequentially estimating a disturbance ratio representing a ratio for each frequency domain, and a preceding or subsequent stage of the first updating amount adjusting means, the characteristic error of the filter coefficient or the characteristic error of the filter coefficient multiplied by the first adjusting coefficient. , A second adjustment coefficient multiplied for each frequency domain by a second adjustment coefficient having a smaller value as the disturbance ratio is larger according to the magnitude of the disturbance ratio between 0 and a predetermined value. Suggest echo canceller and a new quantity adjusting means.
[0010]
According to a third aspect of the present invention, there is provided an echo canceling program which is described by a code string which can be decoded by a computer, and which causes the computer to execute the echo canceling method according to the first aspect.
Action <br/> disturbance component according to the configuration sound pickup unit 2 included in the collected sound signal picked up in this invention, in accordance with the ratio of the filter error calculation means had the characteristic error of the filter coefficients to be output In addition, it is possible to reduce the uncertainty of the calculated characteristic error of the filter coefficient. For this reason, the first updating amount adjusting means can take the largest first adjusting coefficient as much as possible, and can increase the speed at which the filter coefficient used in the filter processing means approaches the true value for echo cancellation.
[0011]
The disturbance ratio estimating means estimates the disturbance ratio representing the ratio of the magnitude of the disturbance signal component other than the reverberant signal included in the sound pickup signal from the sound pickup means to the magnitude of the simulation error output by the simulation error output means. Can be realized as follows. Here, the magnitude of the simulation error output by the simulation error output means in the frequency domain ω (n = 1... N) divided into N is | E (ω n ) |, and the magnitude of the disturbance in the frequency domain ω n is estimated. The value is | N (ω n ) |. As the estimated disturbance value | N (ω n ) |, a value measured in advance in a non-voice section or the like is used. At this time, the calculation of the disturbance ratio DR (ω n )
DR (ω n ) = | N (ω n ) | / | E (ω n ) | (1)
Can be calculated as Here, the value of the disturbance ratio DR (ω n ) is assumed to be, for example, T = 1 from 0 to a predetermined value T. If the value deviates from this range in an actual use environment, a range of 0 to a predetermined value is set. Censor the numerical value so that The value of DR (ω n ) may be based on a formula other than equation (1) as long as the ratio of | N (ω n ) | to | E (ω n ) |
[0012]
The second adjustment coefficient β (ω n ) of the second update amount adjustment means takes a value from 0 to a predetermined value, for example, 1, and uses the above-described disturbance ratio DR (ω n )
β (ω n ) = (T-DR (ω n ) p ) 1 / p (2)
Can be determined as follows. Here, p is a positive real number. In addition, in addition to the equation (2), as shown in FIG. 4A, it is possible to smoothly correspond to the value of DR (ω n ), or to correspond to discontinuity as shown in FIG. Even if the value is such that the value of β (ω n ) decreases as the value of DR (ω n ) increases, it is included in the embodiment in which the effects of the present invention can be obtained. Even when DR (ω n ) is obtained in a form other than equation (1), an appropriate one can be selected from the correspondence between DR (ω n ) and β (ω n ) shown here.
[0013]
BEST MODE FOR CARRYING OUT THE INVENTION
FIG. 1 shows an embodiment of the echo canceling apparatus according to the present invention. The echo canceller shown in FIG. 1 will be described together with the echo canceller method.
Reference numeral 100 shown in FIG. 1 indicates an echo canceller according to the present invention. The echo canceling apparatus 100 according to the present invention is obtained by adding the following means to the conventional echo canceling apparatus 10 shown in FIG. That is, a disturbance ratio DR (ω n ) representing a ratio of the magnitude of the disturbance signal component other than the reverberation signal included in the sound pickup signal from the sound pickup means 2 to the magnitude of the simulation error output from the simulation error output means 105. Ratio estimation means 111 for successively estimating the frequency ratio for each frequency domain, and the characteristic error of the filter coefficient actually output by the filter error calculation means 106, and the disturbance ratio according to the magnitude of the disturbance ratio estimated by the disturbance ratio estimation means 111. Is a predetermined value when 0% is 0%, for example, 1 when the disturbance ratio is 100%, and a second adjustment coefficient β (ω n ) which takes a value between 0 and a predetermined value when the disturbance ratio is in between. And a second update amount adjusting unit 110 for multiplying each region.
[0014]
According to this configuration, the disturbance coefficient included in the sound pickup signal picked up by the sound pickup means 2 affects the characteristic error of the filter coefficient output from the filter error calculation means 106 in accordance with the ratio of the calculated filter coefficient. The value of the part where the characteristic error is uncertain can be reduced. For this reason, the first update amount adjusting means 107 can take the largest possible first adjustment coefficient, and can increase the speed at which the filter coefficient used in the filter processing means 104 approaches the true characteristic for echo cancellation.
[0015]
Modified embodiment of the invention In the embodiment shown in FIG. 1, an example in which the second update amount adjusting unit 110 is provided before the first update amount adjusting unit 107 has been described. However, as shown in FIG. Even if the second update amount adjusting unit 110 is arranged at the subsequent stage of the first update amount adjusting unit 107, the same operation and effect as the embodiment shown in FIG. 1 can be obtained.
[0016]
As shown in FIG. 3, a third update amount adjusting unit 112 is arranged between the filter error calculating unit 106 and the filter updating unit 108, and the third update amount adjusting unit 112 For example, the same operation as in the embodiment shown in FIG. 1 may be performed by obtaining a product with the coefficient, using the value of the product as a third adjustment coefficient, and multiplying the third adjustment coefficient by the characteristic error of the filter coefficient for each frequency domain. The effect can be obtained.
The above-described reverberation canceling apparatus and reverberation canceling method according to the present invention are realized by causing a computer to execute a sound canceling program described by a computer-readable code string. FIG. 5 shows a state in which the sound erasing program according to the present invention is installed in the computer 20.
[0017]
As is well known, the computer 20 includes a central processing unit (CPU) 21, a read-only memory (ROM) 22 storing a basic program for controlling start-up and shut-down of the computer, and the like; In addition to the temporary recording, the read / write / read / write memory (RAM) 23 for storing a program for executing the echo canceling method of the present invention, an input port 24, an output port 25, and the like can be used.
The reproduction signal input terminal 3 and the sound pickup means 2 are connected to the input port 24, and the reproduction signal X and the pickup signal are input. The output port 25 is connected to the sound reproducing means 1 and the echo canceling signal output terminal 4, and outputs the sound corresponding to the reproduced signal and the echo canceling signal e = yy- '.
[0018]
In the readable / writable memory 23, in addition to the data storage area 23A, a reproduction signal input processing program 23B operating as the reproduction signal input means 101, and a sound collection signal input processing program 23C operating as the sound collection signal input means 102 A reproduction signal conversion processing program 23D operating as the reproduction signal conversion means 103, a filter processing program 23E operating as the filter processing means 104, a simulation error output processing program 23F operating as the simulation error output means 105, and a filter error calculation means A filter error calculation processing program 23G operating as 106, a first update amount adjustment processing program 23H operating as the first update amount adjustment means 107, a filter update processing program 23I operating as the filter update means 108, and a second update amount Coordinator A second update amount adjustment processing program 23J operating as 110 and a disturbance ratio estimation processing program 23K operating as the disturbance ratio estimating means 111 are installed, and these programs are decoded by the central processing unit 21 to perform the echo cancellation operation. Execute Each of the programs 23B to 23K installed in the readable and writable memory 23 is recorded in advance on a computer-readable magnetic disk or CD-ROM, and is installed from these recording media or installed through a communication circuit. You.
[0019]
【The invention's effect】
The echo canceller according to the present invention suppresses a decrease in the speed at which the filter coefficient used in the filter processing unit approaches the true characteristic for echo cancellation, and suppresses a disturbance component included in the sound pickup signal picked up by the sound pickup unit. The influence can be reduced, and the echo canceling performance can be improved.
[Brief description of the drawings]
FIG. 1 is a block diagram illustrating an embodiment of an echo canceller according to the present invention.
FIG. 2 is a block diagram for explaining another embodiment of the echo canceller according to the present invention.
FIG. 3 is a block diagram for explaining still another embodiment of the echo canceller according to the present invention.
FIG. 4 is a characteristic curve diagram for explaining the operation of the main part of the present invention.
FIG. 5 is a conceptual diagram illustrating a configuration when the echo canceller according to the present invention is implemented by a computer.
FIG. 6 is a block diagram for explaining the outline of the echo canceller.
FIG. 7 is a block diagram for explaining a conventional echo canceller.
[Explanation of symbols]
REFERENCE SIGNS LIST 1 sound reproduction means 105 simulated error output means 2 sound collection means 106 filter error calculation means 3 reproduction signal input terminal 107 first update amount adjustment means 4 echo cancellation signal output terminal 108 filter update means 100 echo cancellation device 110 second update amount Adjusting means 101 Reproduction signal input means 111 Disturbance ratio estimation means 102 Sound collection signal input means 103 Reproduction signal conversion means 104 Filter processing means

Claims (3)

音響再生手段へと出力される再生信号を入力し一定時間蓄積し再生信号列を得る再生信号入力処理と、前記音響再生手段と同一空間に存在する音響収音手段から収音信号を入力し一定時間蓄積し収音信号列を得る収音信号入力処理と、前記再生信号列を周波数領域変換し再生信号変換列を得る再生信号変換処理と、前記再生信号変換列を周波数領域毎に入力しフィルタ処理することにより前記音響再生手段から前記音響収音手段へと回り込む反響信号の周波数領域変換列を模擬する模擬反響信号変換列を生成するフィルタ処理と、前記模擬反響信号変換列と前記収音信号列とを入力し模擬反響信号変換列の模擬誤差を出力する模擬誤差出力処理と、前記再生信号変換列と前記模擬誤差を入力し前記フィルタ処理のフィルタ係数の周波数領域毎の特性誤差を計算するフィルタ誤差計算処理と、前記フィルタ誤差計算処理が計算した特性誤差に周波数領域毎に第一調整係数を乗じる第一更新量調整処理と、前記第一調整係数を乗じられた特性誤差を加えることにより前記フィルタ処理のフィルタ係数を更新するフィルタ更新処理とを実行する反響消去方法において、
前記模擬誤差出力処理で出力される前記模擬誤差の大きさに対する前記収音信号に含まれる前記反響信号以外の外乱信号成分の大きさの比率を表わす外乱比率を周波数領域毎に逐次推定する外乱比率推定処理と、前記第一更新量調整処理の前段又は後段において、前記フィルタ係数の特性誤差又は前記第一調整係数を乗じた前記フィルタ係数の特性誤差に、0から所定値の間で、前記外乱比率の大きさに応じ、外乱比率が大きい程小さい値をとる第二調整係数を周波数領域毎に乗じる第二更新量調整処理とを実行することを特徴とする反響消去方法。
A reproduction signal input process for inputting a reproduction signal to be output to the sound reproduction unit and accumulating the reproduction signal for a certain period of time to obtain a reproduction signal sequence; and inputting a predetermined sound collection signal from the sound collection unit existing in the same space as the sound reproduction unit. A sound collection signal input process for accumulating time to obtain a sound collection signal sequence, a reproduction signal conversion process for frequency domain transforming the reproduction signal sequence to obtain a reproduction signal conversion sequence, and a filter for inputting the reproduction signal conversion sequence for each frequency domain A filtering process for generating a simulated reverberation signal conversion sequence that simulates a frequency domain conversion sequence of a reverberation signal that wraps around from the sound reproduction unit to the sound collection unit by processing; And a simulation error output process for inputting the sequence and outputting a simulation error of the simulation echo signal conversion sequence, and for each frequency domain of the filter coefficient of the filter process after inputting the reproduction signal conversion sequence and the simulation error. A filter error calculation process for calculating a characteristic error, a first update amount adjustment process for multiplying the characteristic error calculated by the filter error calculation process by a first adjustment coefficient for each frequency domain, and a characteristic multiplied by the first adjustment coefficient. And a filter updating process for updating a filter coefficient of the filtering process by adding an error.
A disturbance ratio that sequentially estimates a disturbance ratio representing a ratio of a magnitude of a disturbance signal component other than the reverberation signal included in the sound pickup signal to a magnitude of the simulation error output in the simulation error output process for each frequency domain. Estimation processing and before or after the first update amount adjustment processing, the characteristic error of the filter coefficient or the characteristic error of the filter coefficient multiplied by the first adjustment coefficient, between 0 and a predetermined value, the disturbance A second update amount adjustment process of multiplying a second adjustment coefficient having a smaller value as the disturbance ratio increases in accordance with the ratio, for each frequency domain.
音響再生手段へと出力される再生信号を入力し一定時間蓄積し再生信号列を得る再生信号入力手段と、前記音響再生手段と同一空間に存在する音響収音手段から収音信号を入力し一定時間蓄積し収音信号列を得る収音信号入力手段と、前記再生信号列を周波数領域変換し再生信号変換列を得る再生信号変換手段と、前記再生信号変換列を周波数領域毎に入力しフィルタ処理することにより前記音響再生手段から前記音響収音手段へと回り込む反響信号の周波数領域変換列を模擬する模擬反響信号変換列を生成するフィルタ処理手段と、前記模擬反響信号変換列と前記収音信号列とを入力し模擬反響信号変換列の模擬誤差を出力する模擬誤差出力手段と、前記再生信号変換列と前記模擬誤差を入力し前記フィルタ処理手段のフィルタ係数の周波数領域毎の特性誤差を計算するフィルタ誤差計算手段と、前記フィルタ誤差計算手段が計算した特性誤差に周波数領域毎に第一調整係数を乗じる第一更新調整手段と、前記第一調整係数を乗じられた特性誤差を加えることにより前記フィルタ処理手段のフィルタ係数を更新するフィルタ更新手段とを有する反響消去装置において、
前記模擬誤差出力手段が出力する前記模擬誤差の大きさに対する前記収音信号に含まれる前記反響信号以外の外乱信号成分の大きさの比率を表わす外乱比率を周波数領域毎に逐次推定する外乱比率推定手段と、前記第一更新量調整手段の前段又は後段において、前記フィルタ係数の特性誤差又は前記第一調整係数を乗じた前記フィルタ係数の特性誤差に、0から所定値の間で前記外乱比率の大きさに応じ、外乱比率が大きい程小さい値をとる第二調整係数を周波数領域毎に乗じる第二更新量調整手段とを有することを特徴とする反響消去装置。
A reproduction signal input means for inputting a reproduction signal to be output to the sound reproduction means and accumulating the reproduction signal for a certain period of time to obtain a reproduction signal sequence; and inputting a fixed sound signal from a sound collection means existing in the same space as the sound reproduction means. Sound collection signal input means for accumulating time to obtain a sound collection signal sequence, reproduction signal conversion means for frequency domain transforming the reproduction signal sequence to obtain a reproduction signal conversion sequence, and a filter for inputting the reproduction signal conversion sequence for each frequency domain A filter processing unit for generating a simulated reverberation signal conversion sequence that simulates a frequency domain conversion sequence of a reverberation signal circulating from the sound reproduction unit to the sound collection unit by processing; the simulated reverberation signal conversion sequence and the sound collection A simulation error output means for inputting a signal sequence and outputting a simulation error of a simulation echo signal conversion sequence, and a frequency of a filter coefficient of the filter processing means for inputting the reproduction signal conversion sequence and the simulation error Filter error calculating means for calculating a characteristic error for each area, first updating adjusting means for multiplying the characteristic error calculated by the filter error calculating means by a first adjusting coefficient for each frequency area, and multiplying by the first adjusting coefficient. And a filter updating means for updating the filter coefficient of the filtering means by adding the characteristic error.
Disturbance ratio estimation for sequentially estimating a disturbance ratio representing a ratio of a magnitude of a disturbance signal component other than the reverberation signal included in the picked-up signal to a magnitude of the simulation error output by the simulation error output means for each frequency domain. Means, at a stage before or after the first update amount adjusting unit, the characteristic error of the filter coefficient or the characteristic error of the filter coefficient multiplied by the first adjustment coefficient, the disturbance ratio between 0 and a predetermined value A reverberation canceling apparatus, comprising: a second update amount adjusting unit that multiplies, for each frequency region, a second adjustment coefficient that takes a smaller value as the disturbance ratio increases according to the magnitude.
コンピュータが解読可能な符号列によって記述され、コンピュータに請求項1記載の反響消去方法を実行させる反響消去プログラム。An echo cancellation program which is described by a computer-readable code string and causes the computer to execute the echo cancellation method according to claim 1.
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