JP2004509543A - Hearing aid with adaptive filter to suppress acoustic feedback - Google Patents

Abstract

The present invention relates to a hearing aid having an adaptive filter for suppressing acoustic feedback in the hearing aid. The hearing aid further determines a first parameter of an acoustic feedback loop of the hearing aid, compensates for acoustic feedback by adjusting a second parameter of the hearing aid in response to the first parameter, and produces undesirable sound. And a control device configured to substantially prevent This significantly reduces the gain safety margin requirement.

Description

ここで、γは、ワーピング・パラメータである。これにより、ＦＩＲフィルタにおける固定遅延は、周波数に依存した遅延に置き換えられて、大きな遅延が低周波数で得られ、高周波数では遅延が小さくなる。また、全域通過要素（ａｌｌｐａｓｓ ｅｌｅｍｅｎｔｓ）は、内部巡回形（ｉｎｔｅｒｎａｌｌｙ ｒｅｃｕｒｓｉｖｅ）であり、したがってワープ形ＦＩＲフィルタは無限インパルス応答を有することにも注意されたい。このように、ワープ形ＦＩＲという用語は、どちらかと言えば矛盾するが、トランスバーサル形（ｔｒａｎｓｖｅｒｓａｌ）ＦＩＲフィルタとの構造的アナロジーをよく表現している。
【００５０】
この発明の実施例において、ワープ形ＦＩＲフィルタの次数は、同等の仕様のＦＩＲフィルタの次数より大幅に低くしうる。このため、所与の回路構成において、ワープ形ＦＩＲフィルタは、ＦＩＲフィルタより優れたフィルタ特性を提供しうる。さらに、ワーピング・パラメータγは、伝達関数、すなわち周波数スペクトルにおける共振および遮断周波数の位置決めを制御する制御パラメータとして用いることができ、これによってフィルタ出力信号と所望の信号との差であるエラー（誤差）信号ｅ（ｎ）のスペクトルを所望の周波数範囲内において最小にすることができる。
【００５１】
ＦＩＲフィルタまたはワープ形ＦＩＲフィルタにおいて、次のサンプルＹ（ｔ＋Ｔ）は、次式にしたがって計算される。
【数７】

である。
【００５２】
ｕは、信号ｕの最新のＮ個のサンプルを含むＮ次元ベクトルであり、ｃは、Ｎ次フィルタのＮ個の係数を含むベクトルであることに注意されたい。Ｔは、サンプリング周期である。
【００５３】
上式において、ｕ（ｔ）は、実時間（現在時間）ｔにおける実際値であり、ｕ（ｔ−ｉＴ）は、実時間ｔよりもｉ回サンプリング周期前における信号値である。離散時間系においては、簡略表記が用いられることがよくある。ここでは、記号ｕ（ｉ）は、時間ｔ−ｉＴにおける信号値を表す、すなわち上式におけるｕ（ｔ−ｉＴ）である。
【００５４】
たとえばクリュワー・アカデミック出版社から１９９７年に発行されたパウロＳ．Ｒ．ディニッツの適応フィルタリング（Ａｄａｐｔｉｖｅ Ｆｉｌｔｅｒｉｎｇ ｂｙ Ｐａｕｌｏ Ｓ． Ｒ． Ｄｉｎｉｚ， Ｋｌｕｗｅｒ Ａｃａｄｅｍｉｃ Ｐｕｂｌｉｓｈｅｒｓ， １９９７）を参照すると、適応フィルタにおけるフィルタ係数の更新に下式の最小二乗平均（ｌｅａｓｔ ｍｅａｎ ｓｑｕａｒｅ） アルゴリズムが用いられることはよく知られている。
【数８】

【００５５】
前記の簡略表記を用いると（ｎは実際のサンプルの参照番号である）、上式は次のように書き直される。
【数９】

【００５６】
または、さらに簡略化して、下記のように書き直される。
【数１０】

ここで、ｉは、個別のベクトル要素を示す。
【００５７】
フィルタ係数の更新には、次式の漏洩最小二乗平均（ｌｅａｋｙ ｌｅａｓｔ ｍｅａｎ ｓｑｕａｒｅ）アルゴリズムを用いることが好ましい。
【数１１】

ここで、ｕ_ｉは、ｎ 番目のサンプリング周期とｉ−１回の先行するサンプリング周期とにおいてディジタル処理装置の出力信号から導出される１組の信号値であり、ｃ_ｉは１組のフィルタ係数であり、ｅはエラー信号の現在値であり、λ およびμは、スケーリング・ファクタである。μの値は、一般に、１０^−６の大きさであり、λの値は、一般に約０．９９である。λは漏洩（ｌｅａｋａｇｅ）（リーケージ）と呼ばれ、λ＜１のとき、フィルタ係数はそのそれぞれの初期値ｃ_ｉ（０）の方にドリフトする。μは収束速度であり、適応フィルタが変化に適応する速度を決める。この適応速度は、μの値が増加するに伴って増加する。
【００５８】
さらに、アルゴリズムを正規化して、適応フィルタが入力信号の瞬間的な動的変化に実質的に応答しないようにすることが有利である。音響帰還信号を推定するためには、所望の入力信号は不適切であり、適応フィルタの収束性能を低下させる雑音になることに注意されたい。次式の正規化されたアルゴリズムは、正規化最小二乗平均（ｎｏｒｍａｌｉｓｅｄ Ｌｅａｓｔ Ｍｅａｎ Ｓｑｕａｒｅ）（ｎＬＭＳ）アルゴリズムと呼ばれる。
【数１２】

【００５９】
しかし、上式において、パワーの計算には、多大な処理パワーが必要になり、このため、次式にしたがったパワー推定を用いることが好ましい。
【数１３】

ここで、αは、Ｐ_ｕ推定が変化する速度を決定する所定の定数である。このアルゴリズムは、パワー正規化最小二乗平均アルゴリズムと呼ばれる。また、このパワー推定は、入力信号のパワーにおける突然の変化が適応アルゴリズムに及ぼす影響を最小にするように、入力トランスデューサからの出力信号に基づくことができる。
【００６０】
さらに、漏洩符号最小二乗平均（ｌｅａｋｙ ｓｉｇｎ ｌｅａｓｔ ｍｅａｎ ｓｑｕａｒｅ）アルゴリズムと呼ばれる次式のような第３の更新アルゴリズムを用いて、適応フィルタ係数を更新してもよい。
【数１４】

ここで、μ_ｓは、ｅ（ｎ）信号の符号（ｓｉｇｎ）とμとの積である。
【００６１】
さらにまた、漏洩符号−符号最小二乗平均（ｌｅａｋｙ ｓｉｇｎ−ｓｉｇｎ ｌｅａｓｔ ｍｅａｎ ｓｑｕａｒｅ）アルゴリズムと呼ばれる次式のような第４の更新アルゴリズムを適応フィルタ係数に用いてもよい。
【数１５】

ここで、ｓｇｎ（ｕ_ｉ（ｎ））は、ｕ_ｉ（ｎ）の符号（ｓｉｇｎ）である。
【００６２】
フィルタ係数は、たとえば、他の信号と組み合わされるか、平均化されるか、そうでなければフィルタリングされる等の処理を受ける差分信号に基づいて更新してもよい。フィルタリングは、従来技術において知られている集束的態様（ｆｏｃｕｓｓｅｄ ｍａｎｎｅｒ）で行なわれうる。
【００６３】
また、この発明にしたがった多重チャネル（マルチチャネル）形補聴器においては、チャネルの適応フィルタは、同一個数のタップを有する必要はないことに注意されたい。たとえば、低周波チャネルにおいて動作する適応フィルタに、より多くのタップを、含ませることが望ましい。
【００６４】
上述のように、制御装置は、補聴器の音響帰還ループの第１のパラメータの決定に応じてλとμとを調節することができる。
【００６５】
補聴器のパラメータのさまざまな組（ｓｅｔ）を、使用者が聴取することを望む、たとえばスピーチや音楽等の各種の音の種類のそれぞれに対して、および使用者が置かれる、たとえば静寂、騒音、反響、群集、屋外、室内、ヘッドセット等の各種の音響環境のそれぞれに対して設けることができる。たとえば、周波数の関数としての各種のゲイン設定を設けてもよく、入力信号レベルの関数としての各種のゲイン設定を設けてもよく、処理装置の動作ゲインの関数としての各種の収束速度を設けてもよいといった具合である。パラメータの各組は、補聴器の特定の動作モードを規定し、補聴器が特定のパラメータの組を用いて動作するときは、対応するモードで動作すると称せられる。したがって、特定の動作モードにおいては、補聴器の特定のパラメータ値が、特定の音響環境において対応する特定の音を適切に処理するために設定される。同様に、パラメータの自動調節が、現在の動作モードにしたがって行なわれうる。
【００６６】
音の種類（タイプ）は、使用者によって選択されるか、または、たとえば周波数分析、さまざまな周波数における信号対雑音比の分析、音響力学分析、言語認識、ニューラルネットワークによる認識等により、補聴器によって自動的に検出されうる。
【００６７】
同様に、音響環境の種類（タイプ）は、使用者によって選択されるか、または、たとえば周波数分析、さまざまな周波数における信号対雑音比の分析、音響力学分析、ニューラル・ネットワークによる認識等により、補聴器によって自動的に検出されうる。
【００６８】
たとえば、使用者が、音楽を聴取することを望む場合がある。第１の適応フィルタの第１の収束速度は、その場合には、音楽の自動相関に適合する値に設定されうる。さらに、ゲイン調節または第１の収束速度の調節もまた音楽の自動相関に適合して行なわれうる。第１の収束速度、たとえば１個以上のスケーリング・ファクタが、処理装置のゲインの関数として制御される場合は、その関数は、使用者が聴取すると決めた音楽やスピーチ等の特定の音を伴う特定の音響環境において用いるのに適合した関数の組の中から選択されうる。
【００６９】
さらに、調節（調整）もまた、たとえば音響帰還経路の帰還ゲイン等の、測定されるパラメータの変化速度にしたがって行なわれうる。
【００７０】
【実施例】
以下に、添付図面を参照してこの発明をさらに詳細に説明する。
【００７１】
図１は、この発明の実施例のブロック略図である。図１に示されている回路がディジタルもしくはアナログ回路またはこれらの組合せを用いて実現されうることは当業者に自明であろう。この実施例においては、ディジタル信号処理が採用され、したがって処理装置（プロセッサ）７および適応フィルタ１０はディジタル信号処理回路である。この実施例において、補聴器の全てのディジタル回路は、単一のディジタル信号処理チップ上に設けられるか、または複数個の集積回路チップ上に適切な方法で分散して設けられる。
【００７２】
この補聴器において、マイクロホン等の入力トランスデューサ１が用いられ、これによって、音声信号（音信号）が受信され、この音声信号が受信音声信号（受信音信号）を表す対応する電気信号に変換される。補聴器は複数個の入力トランスデューサ１を備えてもよく、これによってたとえばある一定方向の感知特性を提供できる。入力トランスデューサ１は伝達関数Ｈ_ｍを有する。入力トランスデューサ１は、音声信号（音信号）をアナログ信号に変換する。このアナログ信号は、Ａ／Ｄ変換器（図示せず）によってサンプリングされるとともにディジタル化され、補聴器内においてディジタル信号処理されるディジタル信号４となる。ディジタル信号４は結合ノード９に送られ、この結合ノードにおいて、後述する帰還補償信号８５と組み合わせられる。結合ノード９は出力信号８６を出力し、この出力信号はディジタル信号処理装置（ディジタル信号プロセッサ）７に与えられ、所望の周波数特性と圧縮関数とにしたがって増幅されて、使用者の聴覚障害を補償するのに適した出力信号８０が得られる。
【００７３】
出力信号８０は、出力トランスデューサ５と任意の（オプションとしての）遅延Δとに送られ、遅延された信号８３が適応フィルタ１０に供給される。出力トランスデューサ５は、出力信号８０を音響出力信号６に変換する。この音響信号の一部分は、伝達関数Ｈ_ｆｂをもつ帰還経路に沿って入力トランスデューサ１に伝搬する。好ましくは、遅延線Δの時間遅延は、信号６の出力トランスデューサ５から入力トランスデューサ１までの通過時間と実質的に等しい。他の時間遅延を選択してもよい。しかし、より短い時間遅延または零の時間遅延は、フィルタリングを複雑にする。たとえばフィルタが有限インパルス応答フィルタである場合には、より長いフィルタ、すなわちより多くのタップを有するフィルタが必要になる。このため、さらに他の遅延を処理装置７の出力側において回路内に挿入して、遅延信号を出力トランスデューサ５とオプションの遅延Δとに供給し、これによって入力信号４とフィルタリングされた信号８５との間の相関を減少させてもよい。
【００７４】
適応フィルタ１０において、遅延信号８３がフィルタリングされて、音響帰還の推定であるフィルタリングされた信号８５が得られる。すなわちフィルタリングされた信号８５は、出力トランスデューサ５に由来する音を受信することによって生成されるトランスデューサ生成信号４の一部分の推定である。フィルタリングされた信号８５は、結合ノード９においてディジタル入力信号４から減じられ、これによって帰還補償信号８６が得られ、これがディジタル処理装置７に入力される。音響帰還経路における変化を補償するために、適応フィルタ１０のフィルタ係数は、フィルタリングされた信号８５が帰還信号６と実質的に同一に維持されるように、連続的に更新される。
【００７５】
フィルタ１０は、上記したような漏洩符号−符号最小二乗平均（ｌｅａｋｙ ｓｉｇｎ−ｓｉｇｎ ｌｅａｓｔ ｍｅａｎ ｓｑｕａｒｅ）アルゴリズムを有する有限インパルス応答（ＦＩＲ）フィルタまたはワープ形（ｗａｒｐｅｄ）ＦＩＲフィルタである。
【００７６】
制御装置は、処理装置７における実際のゲインに応じてλとμとを調節（調整）する。ゲインの関数としてのスケーリング・ファクタ（ｓｃａｌｉｎｇ ｆａｃｔｏｒｓ） λおよびμのグラフが図１０に示されている。これらの関数は、補聴器の動作モードに依存することに注意されたい。補聴器の現行（現在の）動作モードにしたがって制御装置１３により選択されうる、図１０に示されるような関数の選択可能なサブセットの組（セット）が提供される。さらにこれらの関数は、たとえば音響帰還経路内における減衰等の、測定されるパラメータの変化速度（ｒａｔｅ）にしたがって選択される。
【００７７】
図１の実施例において、制御装置１３は、ディジタル処理装置７からライン１５を介して情報を受取る。ライン１５を介して受取られた、ディジタル処理装置７における現行動作ゲインに関する情報にしたがって、制御装置は、適応フィルタ１０のフィルタ係数の適応速度（ａｄａｐｔａｔｉｏｎ ｒａｔｅ）を調節（調整）する。この図において、破線および矢印は、処理信号の信号経路の一部分を構成しない制御ラインを示すことに注意されたい。
【００７８】
フィルタ１０の実施例であるＦＩＲフィルタが、図６にさらに詳細に示されている。簡単にするために、第１の４個のタップのみが図示されているが、フィルタは、任意の適切な個数のタップを含むことができる。演算子Ηを１に設定し、かつ演算子Β をμ（ｅ（ｎ））に設定すると、漏洩最小二乗平均（ｌｅａｋｙ ｌｅａｓｔ ｍｅａｎ ｓｑｕａｒｅ）アルゴリズムが達成（実現）される。λを１に設定すると、単純最小二乗平均（ｓｉｍｐｌｅ ｌｅａｓｔ ｍｅａｎ ｓｑｕａｒｅ）アルゴリズムが達成される。Ηを１に設定し、かつΒをμｓｇｎ（ｅ（ｎ））に設定すると、漏洩符号最小二乗平均アルゴリズム（ｌｅａｋｙ ｓｉｇｎ ｌｅａｓｔ ｓｑｕａｒｅ）が達成（実現）される。最後に、Η をｓｇｎ（ｕ_ｉ（ｎ））に設定し、かつΒをμｓｇｎ（ｅ（ｎ））に設定して、漏洩符号−符号（ｌｅａｋｙ ｓｉｇｎ−ｓｉｇｎ）ＬＭＳアルゴリズムを達成（実現）することができる。また、フィルタ係数は、再帰的最小二乗（ｒｅｃｕｒｓｉｖｅ ｌｅａｓｔ ｓｑｕａｒｅ）アルゴリズムを用いて計算してもよい。
【００７９】
フィルタ１０の実施例であるワープ形ＦＩＲフィルタは、図７にさらに詳細に示されている。図６および図７の上側の遅延線（ライン）より下の回路が同一であることに注意されたい。ワーピング（ｗａｒｐｉｎｇ）パラメータγ は、０．５に等しいことが好ましい。γ＝０の場合、ワープ形ＦＩＲフィルタは、ＦＩＲフィルタに変わることに注意されたい。
【００８０】
図８に、ワープ形ＦＩＲフィルタの無限インパルス応答とＦＩＲフィルタの有限応答のグラフが示されている。このグラフから、ワープ形ＦＩＲフィルタでは、本来的に、ＦＩＲフィルタよりも、所望の伝達関数により近い値が得られることがわかる。
【００８１】
図９には、他のフィルタ１０４の所望の伝達関数Ｈに適応する適応フィルタ１０２の伝達関数 Ｈ_ａを決定するテスト回路１００のブロック図が示されている。グラフの曲線により、適応フィルタ１０２がＦＩＲフィルタである場合のエラー（誤差）信号１０６のパワー・スペクトル１１０とともに、適応フィルタ１０２がワープ形ＦＩＲフィルタである場合のエラー（誤差）信号１０６のパワー・スペクトル１０８が示されている。ＦＩＲフィルタとワープ形ＦＩＲフィルタは、同一個数のタブを有する。６〜７ｋＨｚ 未満において、ワープ形ＦＩＲフィルタは、エラー信号を１５ｄＢまで改善することがわかる。出力トランスデューサ５の出力部が典型的には、６〜８ｋＨｚ 程度の遮断周波数を有するため、８ｋＨｚ を超える部分におけるワープ形ＦＩＲフィルタの性能は重要ではない。サンプリング周波数を変化させることによって、周波数軸に沿って示されている周波数値が変化することに注意されたい。また、γを調節（調整）してエラー信号１０６のスペクトルを、特定のタイプの聴覚障害等の特定の用途に合わせて最適化しうることにも注意されたい。
【００８２】
図２に、各チャネル（チャンネル）が図１に示された単一チャネルの実施例と一般に同じように動作する、この発明にしたがった補聴器の多重チャネル（マルチチャネル）の実施例が示されている。図２の参照符号に添え字が付け加えられていることを除いて、図１および図２の対応する部分は、同じ参照符号で示されている。簡単にするために、図２には３個のチャネルのみが図示されている。しかしながら、この補聴器は、同図に示されているようなチャネルをいかなる適切な個数でも含みうることに注意されたい。
【００８３】
図２にしたがったこの発明の多重チャネルの実施例は、図１に示された単一チャネルの実施例と同じ部分に加えて、帯域フィルタ信号（帯域フィルタリングされた信号）４ａ、４ｉ、４ｎを出力するフィルタ・バンク３を備えている。結合ノード９ａ、９ｉ、９ｎにおいて、それぞれの信号４ａ、４ｉ、４ｎが組み合わされて、それぞれの信号８６ａ、８６ｉ、８６ｎが形成される。これらの信号８６ａ、８６ｉ、８６ｎは、多重チャネル形ディジタル処理装置７に供給され、使用者の聴覚障害に合致する所望の特性にしたがって処理される。これは、個別のチャネルにおける異なるゲイン設定の調節を伴いうる。さらに、この処理には、圧縮機能（ｃｏｍｐｒｅｓｓｏｒ ｆｕｎｃｔｉｏｎｓ） も含まれうる。さらにまた、雑音の減少等の他の機能が信号処理装置によって達成されうる。
【００８４】
ディジタル信号処理装置７からの出力信号はフィルタ・バンク１６に送られ、このフィルタ・バンクにおいて、１組の適応フィルタ１０ａ、１０ｉ、１０ｎにおける異なる周波数帯域またはチャネルに対応する帯域フィルタ信号（帯域フィルタリングされた信号）８３ａ、８３ｉ、８３ｎに分割される。好ましくは、フィルタ・バンク１６は、ディジタル四次フィルタからなる。
【００８５】
フィルタリングされた信号８５ａ、８５ｉ、８５ｎは、適応フィルタ１０ａ、１０ｉ、１０ｎから、それぞれの結合ノード９ａ、９ｉ、９ｎに供給されて、信号４ａ、４ｉ、４ｎから減じられ、信号８６ａ、８６ｉ、８６ｎが生成される。図１の実施例の場合と同様に、任意の遅延線Δにより出力信号８０を遅延させることができる。好ましくは、この遅延は、実質的に出力トランスデューサ５から入力トランスデューサ１までの音の最大伝搬時間に等しい。
【００８６】
処理装置７は、そのチャネルの信号を組み合わせて単一の出力信号８０にする。
【００８７】
多重チャネルの実施例においては、それぞれのチャネルの適応速度（ａｄａｐｔａｔｉｏｎ ｒａｔｅｓ）は、互いに異なりうる。このため、より高い適応速度を適用して、その結果として帰還共振が起こりやすい周波数で望ましくない歪みを発生させることができる。このことは、帰還共振が所望の信号にとって重要でない周波数で起こる場合に有利な特徴となる。
【００８８】
また、信号検出は、広範な周波数範囲において行なう方が困難である。このため、多重チャネルの装置では、単一チャネルの装置よりも、間違った信号検出による収束誤差（ｃｏｎｖｅｒｇｅｎｃｅ ｅｒｒｏｒｓ）が生じにくい。
【００８９】
ひとつの実施例において、制御装置１３は、適応フィルタ１０、１０ａ、１０ｉ、１０ｎのフィルタ係数の適応速度を、Ｇ_０からＧ_ａのゲイン間隔において、処理装置における実際の動作ゲインの関数として制御する。
【００９０】
図３に図示されている補聴器は、図１の補聴器に追加の測定装置を設けたものに対応する。対応する部分は同一の参照符号で示されており、これらの部分の動作の説明を繰り返すことはしない。図３に示された補聴器は、さらに、第１の適応フィルタ１０と並行して動作する、すなわち第１の適応フィルタ１０と同じ信号に対して作用するが、第１の適応フィルタ１０の第１の収束速度（ｃｏｎｖｅｒｇｅｎｃｅ ｒａｔｅ）より低速の第２の収束速度で動作する第２の適応フィルタ１１を含む。第２の適応フィルタ１１の出力８５は、結合ノード９に供給されて、信号４から減じられ、処理装置７に入力される信号８６が生成され、これによって音響帰還信号は、処理装置７による処理の前に、信号から実質的に除去される。第１の適応フィルタ１０の出力８９は処理装置の入力の補正には用いられないことに注意されたい。
【００９１】
この実施例において、制御装置１３は、第１の適応フィルタ１０のパラメータを決定することにより音響帰還の量を推定するようになっている。高い第１の収束速度により、第１の適応フィルタ１０は、第２の適応フィルタ１１よりも、音響帰還を経時的により厳密に追跡しうる。さらに、第１の適応フィルタ１０の出力信号８９が入力トランスデューサ信号４から減じられないため、所望の信号が第１の適応フィルタ１０によって歪まされることはない。
【００９２】
第２の適応フィルタ１１は、いかなる種類の適応フィルタであってもよいが、好ましくは、パワー正規化最小二乗平均（ｐｏｗｅｒ−ｎｏｒｍａｌｉｓｅｄ Ｌｅａｓｔ Ｍｅａｎ Ｓｑｕａｒｅ）（ｐｏｗｅｒ−ｎＬＭＳ）アルゴリズムを用いるＦＩＲフィルタまたはワープ形ＦＩＲフィルタである。
【００９３】
第２の適応フィルタ１１は、フィルタリングされた信号８９を第２の結合ノード１２に出力し、このノードにおいて前記信号は第１の結合ノード９からの信号８６と組み合わされる。結合ノード１２からの出力信号９０は、第２の適応フィルタ１１に入力されて、フィルタ係数が調節される。
【００９４】
図３に示された実施例の重要な利点は、第１の適応フィルタ１０によって生成された出力信号が入力トランスデューサ１から出力トランスデューサ５までの主要信号経路に供給されないことである。この主要信号経路は、入力トランスデューサ１と、ディジタル変換手段（図示せず）と、結合ノード９と、ディジタル処理装置７と、出力トランスデューサ５とからなる。したがって、第１の適応フィルタ１０による信号処理が主要信号経路における信号に直接影響することはない。このため、主要信号経路における信号の信号歪みが第１の適応フィルタ１０によって創出されることはなく、したがって第１の適応フィルタ１０の適応速度（ａｄａｐｔａｔｉｏｎ ｒａｔｅ）を第２の適応フィルタ１１の適応速度より実質的に高くすることができる。第１の適応フィルタ１０の適応速度を第２の適応フィルタ１１の適応速度より実質的に高くすることができるため、帰還経路は、第１の適応フィルタ１０によって、第２の適応フィルタ１１によるよりも、経時的にはるかに厳密に監視されうる。好ましくは、第１の適応速度は、固定された高適応速度であるが、この適応速度は、たとえば１個またはそれ以上のスケーリング・ファクタを補正することによって調節されうる。たとえば、第１の適応フィルタの適応速度を、処理装置における実際のゲインまたは入力パワー・レベルにしたがって調節することが好ましいかもしれない。
【００９５】
適応速度（ａｄａｐｔａｔｉｏｎ ｒａｔｅ）の調節は、動作の異なるモード毎に相違しうる。
【００９６】
音響環境の急速な変化が起こった場合に、図３の第２の適応フィルタ１１は、こうした変化に即座に適応してこの変化を補償することができない。したがって、未補償の帰還信号が発生し始めることになる。しかしながら、第１の適応フィルタ１０は、第２の適応フィルタ１１よりはるかに高速であり、かつ帰還経路における変化に適応する。
【００９７】
ひとつの実施例において、制御装置は、帰還経路における変化に対する第１の適応フィルタ１０の迅速な応答に基づいて、たとえばμの値を制御するといったように、第２の適応フィルタ１１の適応速度を制御する。このため、第１の適応フィルタ１０の特性、たとえば減衰等のフィルタリング特性により帰還経路における変化が示されると、第２の適応フィルタ１１はそれに応じて制御される。たとえばゲインが帰還限界に近い場合は、第２の適応フィルタ１１の適応速度が増加する。第２の適応フィルタ１１の適応速度を増加させることにより、第２の適応フィルタは、より迅速に、たとえば音響帰還が望ましくない音の発生を招く前に、音響帰還の変化を補償しうる。
【００９８】
音響帰還の量は、好ましくは、第１の適応フィルタ１０のパラメータを決定すること、またはこれに代えて、もしくは加えて、第２の適応フィルタ１１のパラメータを決定することによって推定されうることに注意されたい。たとえば、それぞれの適応フィルタ１０、１１の入力信号と出力信号との比は、音響帰還経路を含む帰還経路の減衰の推定の要素となるので、この比を決定してもよい。さらに、このような計算を平均化された信号に基づいて行ない、雑音およびスピーチによる影響と収束誤差を抑制することが望ましい。これに代えて、所望の特性の平均を決定してもよい。好ましくは、上記のタイプのパワー推定を各信号に関して使用する。これに代えて、フィルタ係数を適切に変換することによって、適応フィルタ１０、１１のうちの一方のパラメータを決定してもよい。
【００９９】
また他の実施例において、制御装置は、帰還における変化が第１の適応フィルタ１０により検出されると、ディジタル処理装置におけるゲインを低下させる。特に、このことは、ディジタル処理装置の異なるチャネルにおいて選択的に実行されうる。
【０１００】
第１のパラメータの決定に基づいて、制御装置は、望ましくない音声（音）信号の発生を防ぐために処理装置が超えてはならない最大ゲイン値 Ｇ_ｍａｘを計算することができる。多重チャネル形補聴器においては、各チャネル毎に別個のＧ_ｍａｘ値があってもよい。
【０１０１】
さらに他の実施例において、制御装置は、Ｇ_０からＧ_ａへのゲイン間隔を変更する。このため、第２の適応フィルタ１１により、装置が不安定に近くなっていることが検出されると、この情報を用いてゲイン下限 Ｇ_０を低下させ、これによって、 Ｇ_ａを特定レベルに維持することが望まれる場合には、ゲイン間隔全体が下方に移動するか、またはゲイン間隔が拡大される。ゲイン下限 Ｇ_０だけが変更される場合は、λおよびμの曲線が異なる間隔を包含するように変更されることが好ましい。
【０１０２】
これに関しては、ゲインとλおよびμとの関係は、図１０に示された関数とは異なることに注意されたい。
【０１０３】
図４に、各チャネルが一般に図３に示された単一チャネルの実施例と同じように動作する、この発明にしたがった補聴器の多重チャネル（マルチチャネル）の実施例が示されている。図３および図４の対応する部分は、図３の参照符号に添え字が付け加えられていることを除いて、同じ参照符号で示されている。簡単のために、３個のチャネルのみが図４に示されている。しかしながら、この補聴器は、同図に示されているようなチャネルを適切な個数で含みうることに注意されたい。簡単にするために、制御ラインは、図４においては省略されている。
【０１０４】
図４にしたがったこの発明の多重チャネルの実施例は、図３に示された単一チャネルの実施例と同じ構成部分に加えて、帯域フィルタ信号（帯域フィルタリングされた信号）８３ａ、８３ｉ、８３ｎを第２の適応フィルタ１１ａ、１１ｉ、１１ｎの組 （ｓｅｔ）に出力するフィルタ・バンク１６を備えている。それぞれの適応フィルタ１１ａ、１１ｉ、１１ｎは、それぞれの結合ノード１２ａ、１２ｉ、１２ｎに、結合ノード９ａ、９ｉ、９ｎからのそれぞれの信号８６ａ、８６ｉ、８６ｎと組み合わされるフィルタリングされた信号を供給する。
【０１０５】
図４に示された多重チャネルの実施例では、帰還経路の伝達関数のより詳細な推定が達成される。さらに、信号処理は、より低い周波数帯域においてより低いサンプリング周波数で行なわれ得、これはデシメーション（ｄｅｃｉｍａｔｉｏｎ）として知られた技術である。第１の適応フィルタの組からの出力信号は主要信号経路には供給されないため、いかなるアンチ・エリアジング（ａｎｔｉ−ａｌｉａｓｉｎｇ） フィルタもシステムには必要とされないので、デシメーションは、これらのフィルタにおいて特に簡単に用いられうる。
【０１０６】
図４に示された実施例は、図３に示された実施例と同じように制御されうる。しかしながら、図４に示された実施例では、各個別のチャネルにおいてゲインを選択的に低下させることと、第２の適応フィルタ１１ａ、１１ｉ、１１ｎの組の各個別の適応フィルタの適応速度を選択的に調節することが可能である。このため、帰還共振が起こりにくい周波数においてゲインを高い値に維持することができ、かつ歪みを低レベルに維持することができるというさらに他の利点が得られる。
【０１０７】
図５に、図４に示された実施例と同様の構成であり、かつ同様に動作する多重チャネルの実施例が示されている。しかし、図５に示された実施例は、単一の適応フィルタ１１によって構成される第２の適応フィルタの組を有しており、かつ結合ノード９が単一の結合ノードであるため、より簡素である。
【０１０８】
多くのその他の実施例が、処理装置におけるチャネルと第１および第２の適応フィルタの組における適応フィルタの個数を変化させることによって得られる。また、処理装置におけるチャネルの個数は、第１の適応フィルタの組におけるフィルタの個数と異なっていてもよく、同様に前記フィルタの個数は、第２の適応フィルタの組におけるフィルタの個数と異なっていてもよい。
【０１０９】
特に、比較的少数のチャネルを有するディジタル信号処理装置７と、より多くのフィルタを含む第２の適応フィルタの組とを用いることが可能である。これに代えて、第２の適応フィルタの組の個別の適応フィルタは、ディジタル信号処理装置７におけるチャネルとの組み合わせの上で動作するものでもよく、たとえばディジタル信号処理装置７の２個またはそれ以上のチャネルが第１の適応フィルタの組の特定の適応フィルタにより決定される同じ Ｇ_ｍａｘを用いて動作するか、またはディジタル信号処理装置７の１個のチャネルが第１の適応フィルタの組の適応フィルタにより決定される２個以上のゲインのうちの最も低いゲインであるＧ_ｍａｘを用いて動作してもよい。 しかしながら、現在のところは、単一の第２の適応フィルタ１１と多重チャネルの第１の適応フィルタ１０の組とを備えた実施例が好ましい。
【０１１０】
図１１に、周波数の関数としての動作ゲインのグラフが示されている。実線で示されている上側の曲線は、この発明にしたがった補聴器によって望ましくない音の発生を伴うことなしに得られる最大動作ゲインを示し、破線で示されている下側の曲線は、公知の補聴器の場合の対応するゲインを示す。
【図面の簡単な説明】
【図１】
この発明にしたがう補聴器のブロック図である。
【図２】
各チャネルが図１に示された補聴器に対応する多重チャネル形補聴器のブロック図である。
【図３】
この発明にしたがう測定装置を内蔵する補聴器のブロック図である。
【図４】
各チャネルが図３に示された補聴器に対応する多重チャネル形補聴器のブロック図である。
【図５】
単一の帯域適応フィルタを有する多重チャネル形補聴器のブロック図である。
【図６】
この発明にしたがう更新アルゴリズムを実現するＬＭＳ形ＦＩＲフィルタを示すブロック図である。
【図７】
この発明にしたがう更新アルゴリズムを実現するＬＭＳ形ワープＦＩＲフィルタを示すブロック図である。
【図８】
ＦＩＲフィルタのインパルス応答とワープ形ＦＩＲフィルタのインパルス応答との比較を示すグラフである。
【図９】
ＦＩＲフィルタおよびワープ形ＦＩＲフィルタの所望の伝達関数からの偏差を示すグラフである。
【図１０】
ディジタル処理装置におけるゲインに依存するフィルタ係数の変動を表す図である。
【図１１】
この発明により達成される最大可能ゲインの改善を示す図である。[0001]
【Technical field】
The present invention relates to a hearing aid including an adaptive filter that suppresses acoustic feedback in the hearing aid.
[0002]
[Prior art]
It is well known in the hearing aid art that acoustic feedback results in the generation of undesirable acoustic signals that can be heard by the hearing aid user.
[0003]
Acoustic feedback occurs when the hearing aid's input transducer receives and detects the acoustic output signal generated by the output transducer. Amplification of the detected signal may lead to the generation of a stronger sound output signal, and eventually the hearing aid may oscillate.
[0004]
It is well known to provide an adaptive filter in a hearing aid to compensate for acoustic feedback. The adaptive filter estimates a transfer function from the output to the input of the hearing aid, including the acoustic propagation path from the output transducer to the input transducer. The input of the adaptive filter is connected to the output of the hearing aid, and the output signal of the adaptive filter is subtracted from the input transducer signal to compensate for acoustic feedback. Such a hearing aid is disclosed in U.S. Pat. No. 5,402,496.
[0005]
In such a system, the adaptive filter acts to remove the correlation from the input signal, while the signals representing speech (language) and music are signals with significant autocorrelation. Thus, removing the correlation from the signals representing speech and music will distort these signals, and such distortion is, of course, undesirable, and the adaptive filter cannot be adapted too quickly. Thus, the convergence rate of the adaptive filter in known hearing aids ensures that the desired high convergence rate can cope with sudden changes in the acoustic environment, and that the signals representing speech and music remain undistorted. It is a compromise with the desired low convergence speed that can be kept.
[0006]
Lack of speed of adaptation still leads to the generation of unwanted acoustic signals due to acoustic feedback. Undesirable acoustic signal generation is most likely to occur at frequencies with high feedback loop gain. The loop gain is the product of the attenuation in the acoustic feedback path and the gain from the input to the output of the hearing aid.
[0007]
Acoustic feedback is achieved by a known CIC hearing aid with vent (CIC =ccompletein thecanal (completely in the ear canal)). The presence of the vent and the short distance between the input and output transducers of the hearing aid result in low attenuation of the acoustic feedback path from the output transducer to the input transducer, and the short delay time maintains the correlation in the signal. It is.
[0008]
Various strategies are well known in the prior art to address acoustic feedback. For example, it is well known to keep the loop gain below a certain limit to prevent feedback resonance from occurring. It is also known to perform phase adjustment (adjustment) of a feedback signal, perform frequency transposition, and compensate the feedback signal.
[0009]
Typically, the acoustic environment of a hearing aid changes over time, and often changes rapidly over time, such that the propagation of sound from the output transducer of the hearing aid to its input transducer changes dramatically. By way of example, such a change may be caused by a change in the position of the user in the room, for example from a free sound field position in the center of the room to a position near a sound-reflecting wall. Changes can also occur if the user yawns or if the user places the telephone handset in his ear. These various forms of change are known to include changes in the return path attenuation of over 20 dB, some of which may be almost instantaneous.
[0010]
It is known to keep the loop gain below safe limits by limiting the gain adjustment in the hearing aid to the maximum allowable gain based on experience. However, a great deal of safety margin is needed to address the above changes in the acoustic environment and the changes in the physical fit of the hearing aid to the wearer. It is also known to determine the maximum allowable gain when tailoring a hearing aid to a particular user. Nevertheless, great safety margins are still needed. In situations where the gain can be adjusted to a value higher than the maximum allowable gain without undesired sound generation, the provision of the above-mentioned safety margin makes full use of the capabilities of the hearing aid. Hinder.
[0011]
To compensate for severe hearing impairment, it is desirable to be able to set a high gain in the hearing aid. However, the danger of oscillation, also called feedback resonance, limits the maximum gain that can be used, even in situations where the attenuation in the acoustic feedback path is high.
[0012]
DE-A-1 802 2568 A and U.S. Pat. No. 5,016,280 disclose hearing aids including a measuring system for determining the characteristics of the acoustic return path. A test signal is transmitted through the system and the characteristics of the feedback path are determined.
[0013]
In DE 198 02 568 A, the coefficients of the digital filter are determined based on the impulse response of the feedback path, and in U.S. Pat. No. 5,016,280 the filter coefficients of the adaptive compensation filter are determined in the feedback path. Is calculated using a leaky LMS algorithm acting on the transmitted white noise signal.
[0014]
Each of these measuring devices is rather complex and the time for the decision is relatively long, during which the normal functioning of the hearing aid is interrupted. Thus, the decision is made only at certain occasions, for example, when the user switches on the hearing aid. Therefore, a relatively high safety margin for gain is still needed to cope with changes in the acoustic environment between decision points.
[0015]
U.S. Pat. No. 5,619,580 discloses a hearing aid with an adaptive filter and a continuous motion measuring device. A pseudo-random noise signal is added to the output signal. The monitoring system controls the gain of the hearing aid so that the loop gain is kept below a certain value that is frequency dependent. The filter coefficients of the adaptive filter are monitored and the frequency of updating these coefficients is adjusted according to a statistical analysis, which complicates the system. Other disadvantages of this system are that a noise generator is required and that the generated noise signal is always present. In addition, this system increases the adaptation rate. Therefore, even when the hearing aid is not operating close to resonance, signal quality is degraded when a change in the acoustic environment is detected.
[0016]
DISCLOSURE OF THE INVENTION
For this reason, hearing aids are improved so that the operating gain in certain acoustic environments can be higher than in known hearing aids, overcoming the above disadvantages and substantially eliminating gain margin requirements. Is needed.
[0017]
According to a first aspect of the present invention, the above and other objects are attained in a method for suppressing acoustic feedback in a hearing aid, comprising: converting an audio input signal into a first electrical signal; Splitting the signal into a first set of band-filtered electrical signals; processing each of the band-filtered first electrical signals individually; Adding a second electrical signal to a second electrical signal; converting the second electrical signal to an acoustic output signal; and band-filtering the second electrical signal (s). And adaptively filtering the second band-passed electrical signal and applying the filtered signal to an input of the processing device. Estimating acoustic feedback by generating a third signal by adapting each signal at a respective first convergence rate; and determining a first parameter of an acoustic feedback loop of the hearing aid. And adjusting the second parameter of the hearing aid according to the first parameter to compensate for acoustic feedback, thereby reducing the generation of undesirable sounds such as howling and signal distortion. This is to prevent substantially.
[0018]
According to a second aspect of the invention, the above and other objects are achieved by a hearing aid with an adaptive filter that compensates for acoustic feedback. The adaptive filter serves to estimate the transfer function from the output to the input of the hearing aid, including the acoustic path from the output transducer to the input transducer. The input of the adaptive filter is connected to the electrical output of the hearing aid, and the output signal of the adaptive filter can be subtracted from the input transducer signal to compensate for acoustic feedback. The hearing aid further includes an input transducer that converts the acoustic input signal into a first electrical signal, and splits the first electrical signal into a first set of band-filtered electrical signals. A) a first filter bank having a bandpass filter, and separately processing each of the band-filtered first electrical signals and adding the processed electrical signals to a second electrical signal. A second electrical signal and an output transducer for converting the second electrical signal into an acoustic output signal. The hearing aid further comprises: a second filter bank having a plurality of bandpass filters for dividing the second electrical signal into a band-filtered second set of electrical signals; Generating a third (plural) electric signal by filtering the band-passed electric signal of said first electric signal and converting each of said third electric signals into a respective signal at an input side of said processing device to a respective first signal. A first set of adaptive filters having first filter coefficients for estimating acoustic feedback by adapting at a convergence rate.
[0019]
A unique feature of the hearing aid is that it determines the first parameter of the acoustic feedback loop of the hearing aid and adjusts (adjusts) the second parameter of the hearing aid in response to the first parameter. In order to compensate for this and thereby substantially prevent the generation of undesired sounds.
[0020]
An important advantage of the present invention is that whenever the hearing aid operates at a high risk of generating unwanted sounds, the controller automatically adjusts the parameters of the electronic feedback loop to generate such sounds. Therefore, the condition of the gain safety margin is significantly relaxed.
[0021]
In the following, the frequency range of the bandpass filter is also called channel.
[0022]
In a simple embodiment of the invention, the hearing aid is a single-channel hearing aid, ie a hearing aid that processes incoming signals in only one frequency band. To this end, the first filter bank is constituted by a single bandpass filter, which can be constituted by an inherent bandpass filter in the electronic circuit, i.e. if any special circuit is It does not mean that. Correspondingly, the addition of the processed electrical signals at the processing unit (processor) is reduced to the task of providing a single processed electrical signal at the output of the processing unit. Further, the second filter bank is constituted by a single bandpass filter, and the first set of adaptive filters is constituted by a single adaptive filter.
[0023]
In general, hearing loss varies as a function of frequency in different ways for each individual user. To this end, the processing device is preferably divided into a plurality of channels so that each of the frequency bands can be subjected to different processing, for example amplification using different gains. Correspondingly, the hearing aid may comprise a first set of adaptive filters having a plurality of adaptive filters for individually filtering the signals in each frequency band, thereby providing acoustic feedback in each channel of the hearing aid. The ability to control individually is obtained. Preferably, the frequency band of the first set of adaptive filters is substantially identical to the frequency band of the first filter bank so that the bandpass filter does not degrade the effect of the adaptive filter.
[0024]
In one embodiment of the present invention, the first set of adaptive filters subtracts the electrical output of the hearing aid from the input to the processing device, and the difference signal is used to modify the filter coefficients, as described below. This difference signal is not used to correct the input signal to the processing device, thus avoiding signal distortion. Thus, in this embodiment of the invention, the first adaptive filter is used to estimate the acoustic feedback signal without distorting the processed signal. Furthermore, in this embodiment, at least one adaptive filter of the first set of adaptive filters operates on the respective decimated second bandpass filtered electrical signal, so that the adaptive filter output signal is processed. The signal processing power requirements can be minimized without the need for additional filters because they do not directly affect the resulting signal.
[0025]
In another embodiment of the invention, the first set of adaptive filters subtracts the electrical output of the hearing aid from the electrical signal from the input transducer, and the difference signal is used to modify the filter coefficients, and the input of the processing unit. , Whereby the acoustic feedback signal is substantially removed from the signal prior to processing by the processing device. In this embodiment, a third filter bank, which is substantially the same as the first filter bank, is used to convert the processed individual signals from each processing unit (processor) channel to an output signal from the processing unit. If it is added to the processing unit before summing, the decimation of the signal can be used in the set of processing unit and the first adaptive filter.
[0026]
The loop gain of the acoustic feedback loop, i.e., the gain of the acoustic feedback path from the output transducer to the input transducer, including the transfer function of the transducer, and the gain of the electronics included in the signal path from the input to the output of the hearing aid. By monitoring the loop gain of the acoustic feedback loop, which is the sum of As the loop gain approaches unity, certain actions are taken to prevent unwanted sound generation. Since the first set of adaptive filters produces a signal corresponding to the signal generated by the acoustic feedback, the acoustics are monitored by monitoring the attenuation in the first set of adaptive filters and the gain in the corresponding channel of the processor. An index of the loop gain of the feedback loop is obtained. Thus, for example, by determining the individual ratio of the amplitude of the signal at the input of the individual filter to the amplitude of the signal at the corresponding output of the individual filter, the control unit can control the attenuation in the first set of adaptive filters. Can be monitored. In addition, control may be performed, for example, by similarly determining the input and output signal levels of the individual processing unit (processor) channels, or by reading the value of a register in the processing unit that includes the current gain value of the individual processing unit channel. The device may be configured to monitor the gain of individual channels of the processing device. In general, the gain of a processor channel is different for different channels and depends on the input level.
[0027]
Based on monitoring a first parameter of the acoustic feedback loop, such as the loop gain, the gain of the processor channel, the attenuation of one adaptive filter of the first set of adaptive filters, etc. The parameters can be adjusted to prevent unwanted sound generation. For example, the gain of at least one processor channel may be modified, eg, reduced, to keep the acoustic feedback loop gain less than one.
[0028]
The second parameter is the maximum gain limit G that the processing unit gain must not exceed within a particular channel._maxIt may be. The adaptation rate of the first set of adaptive filters is kept constant, while the maximum gain limit G for a particular channel of the processing unit._maxIs reduced whenever the hearing aid approaches a high risk of generating unwanted sounds in its channel, for example caused by sudden changes in the acoustic environment. For example, the maximum gain limit for a particular channel G_max, While the first adaptive filter adapts to the changed acoustic environment and returns to its original value when the adaptive filter has finished adapting to the new situation. This does not cause any distortion of the desired signal.
[0029]
An important advantage of this embodiment of the invention is that the operating gain of the hearing aid can be very high without the risk of unwanted sound being generated, since the gain automatically decreases as the feedback loop approaches resonance. It can be done. Thus, a gain safety margin is not substantially required.
[0030]
In embodiments where the bandpass filter (s) of the second filter bank are substantially identical to the respective bandpass filters of the first filter bank, each channel is individually controlled based on a decision in that channel. Therefore, it is possible to prevent the gain from being reduced due to the influence of the frequency outside the channel.
[0031]
Further, in an embodiment of the present invention in which the difference signal from the first adaptive filter is provided to an input of the processing device, the second parameter is the first convergence speed or the adaptive speed of the first set of adaptive filters. It can be. For example, the adaptation rate of the filter is such that whenever the hearing aid approaches a state of high risk of generating unwanted sounds, for example caused by sudden changes in the acoustic environment, the first adaptive filter It may depend on the operating gain of the processing device in such a way that the adaptation speed is increased and the change is compensated quickly.
[0032]
The convergence rate of the first set of adaptive filters may be adjusted by modifying an algorithm that updates the filter coefficients of the adaptive filter. As described further below, the algorithm may include one or more scaling factors that may be adjusted (adjusted) in response to the determination of the first parameter. For example, the one or more scaling factors can be adjusted as a predetermined function of the operating gain of the processing device.
[0033]
An important advantage of this embodiment is that the operating gain of the hearing aid can be reduced without the risk of generating unwanted sounds, as the adaptive filter adapts more quickly to the situation as the acoustic feedback loop gain approaches resonance. It can be very expensive. As already explained, the quick adaptation of the adaptive filter causes distortion of the desired signal. However, as soon as the adaptive filter has finished adapting, the convergence speed is reduced and the desired signal is no longer distorted. Further, this distortion can occur in a frequency band that does not affect the understanding of the received voice (sound) signal.
[0034]
G₀To G_aA gain interval up to may be provided in the hearing aid. G₀ Is a predetermined lower limit of gain below which feedback resonance and undesirable sound generation cannot occur. G₀Can be determined in a mounting adjustment procedure. G_aIs an adjustable (adjustable) gain upper limit that is adjusted (adjusted) according to the desired audio quality. Preferably, G_aIs adjusted by the procedure of mounting adjustment.
[0035]
The convergence speed is, for example, G₀To G_a, The gain of the processing device may vary as a predetermined function, such as a linear or non-linear function. For example, one or more scaling factors of the adaptive filter update algorithm may be, for example, G G₀To G_a, The gain of the processing device may vary as a predetermined function, such as a linear or non-linear function.
[0036]
When adjusting the hearing aid for an individual user, the transfer characteristics of the return path are measured. Based on these characteristics, G has an appropriate safety margin.₀And G_aIs determined and stored in the hearing aid. G₀To determine, several factors are taken into account. The feedback path characteristic is not constant as described above. Thus, if the feedback compensation is too slow, sudden changes can cause feedback resonance. Further, it may be difficult to predict the magnitude and duration of the return path attenuation change. On the other hand, rapid adaptation can lead to unacceptable distortion of the desired signal, and the level of unacceptable distortion is also of subjective magnitude.
[0037]
However, in a situation where the characteristics of the acoustic return path have been stable for a certain period of time, it is possible to accurately estimate the characteristics of the return path. In such a situation, the relationship between the signal at the input of the first set of adaptive filters and the signal at the output of the first set of adaptive filters is such that the characteristics of the acoustic feedback path, eg, attenuation, are accurate. This is because it is a good measure. When the gain characteristics of the digital processing device and the gain characteristics of the acoustic feedback signal are known, it is possible to obtain an estimate regarding the acoustic feedback loop. With this knowledge, G changes dynamically₀The value can be introduced into the hearing aid. In one embodiment, G₀To G_aIs G₀Has a fixed magnitude, regardless of the change in₀It can move according to the change of.
[0038]
According to a preferred embodiment of the present invention, the hearing aid operates in parallel with the first set of adaptive filters, ie operates on the same signal, but the first convergence of the first set of adaptive filters. The method further comprises a second set of adaptive filters having a second convergence speed that is lower than the speed. The output of the second set of adaptive filters is provided to a corresponding input of the processing device, whereby the acoustic feedback signal is substantially removed from the signal prior to processing by the processing device. The output of the first set of adaptive filters is not used to modify the input signal of the processing device.
[0039]
In this embodiment, the controller is configured to estimate the amount of acoustic feedback by determining the parameters of the first set of adaptive filters. Due to the fast first convergence speed, the first adaptive filter may track acoustic feedback more closely over time than the second adaptive filter. Furthermore, the output signal of the first adaptive filter is not subtracted from the input transducer signal, so that the desired signal is not distorted by the first adaptive filter.
[0040]
Thus, according to the preferred embodiment of the present invention, the second filter coefficient for suppressing feedback in the hearing aid by filtering the second band-filtered electric signal into each fourth electric signal is provided. Generating a fifth electrical signal by subtracting the fourth electrical signal from the respective first band-passed electrical signal and processing the fifth electrical signal. A coupling node supplying the device, wherein the second filter coefficient is updated at a second convergence speed that is lower than the first convergence speed.
[0041]
The amount of acoustic feedback can be estimated by determining the respective ratio of the amplitude of the signal at the input of the first set of adaptive filters to the amplitude of the signal at the output of the first set of adaptive filters. . In this way, a fast response to changes in the acoustic return path is achieved, while requiring very little processor power.
[0042]
The second parameter may be a second convergence speed or an adaptation speed of the second set of adaptive filters. For example, the adaptation rate of the filtering may be such that whenever the hearing aid approaches a high risk of generating unwanted sounds, for example caused by sudden changes in the acoustic environment, the adaptation rate of the second adaptive filter increases. The change may then be made dependent on the operating gain of the processing device, or the attenuation of the first set of adaptive filters, or a combination thereof, in a manner that quickly compensates for this change. .
[0043]
As already described for the first set of adaptive filters, the convergence speed of the second set of adaptive filters can be adjusted by modifying the algorithm that updates the filter coefficients of the adaptive filters. As described further below, the algorithm may include one or more scaling factors that may be adjusted in response to the determination of the first parameter. For example, one or more scaling factors may be set as a predetermined function of the operating gain of the processing device.
[0044]
The second set of adaptive filters provides individual filtering of the signal in each frequency band. Preferably, the frequency band of the second set of adaptive filters is substantially identical to the frequency band of the first filter bank.
[0045]
The frequency bands of the second set of adaptive filters may differ in number and range from the frequency bands of the first filter bank and the first set of adaptive filters. However, in a preferred embodiment of the present invention, the first filter bank comprises a plurality of bandpass filters, while the second set of adaptive filters modifies the processing unit input signal in a single frequency band. Is performed by a single adaptive filter. Thus, a hearing aid having frequency dependent compensation capability can be provided with a simple single band acoustic feedback compensation loop.
[0046]
Thus, according to a preferred embodiment of the present invention, the second adaptive signal having the second filter coefficient for suppressing feedback in the hearing aid by filtering the second electrical signal into a fourth electrical signal. A filter, and generating a fifth electrical signal by subtracting the fourth electrical signal from the first electrical signal, and supplying the fifth electrical signal to a respective bandpass filter of the first filter bank. A hearing aid wherein the second filter coefficient is updated at a second convergence speed lower than the first convergence speed.
[0047]
To this end, in a preferred embodiment of the invention, the processing unit and the first adaptive filter are divided into a plurality of channels covering the same frequency band, while the second adaptive filter is Are not divided into a plurality of channels. In addition, the controller is responsive to the determination of the attenuation of the channel of the corresponding first adaptive filter, the individual maximum gain limit G of each processing unit (processor) channel._maxMay be controlled. The controller may further include a G for the corresponding processor channel._max It may be arranged to increase the second convergence speed of one filter of the second set of adaptive filters when limited by a limit, thereby reducing the duration of the gain limit. Further, the controller may be configured to adjust the gain limit and / or the convergence speed according to the current operating mode of the hearing aid. The term operating mode is explained below.
[0048]
Preferably, the at least one adaptive filter is a finite impulse response (FIR) filter, and more preferably, the at least one adaptive filter is a warped FIR filter or a warped infinite impulse response. ) (Warped) filters, such as (IIR) filters and the like.
[0049]
In the warped FIR filter of this example, the unit delays are replaced by first order allpass sections. However, warping can also be achieved with a second or higher order all-pass. The primary all-pass has the following z-transform.
(Equation 6)

Here, γ is a warping parameter. As a result, the fixed delay in the FIR filter is replaced with a frequency-dependent delay, so that a large delay is obtained at a low frequency and a delay is small at a high frequency. Also note that the allpass elements are internally recursive, so that warped FIR filters have an infinite impulse response. Thus, the term warped FIR, although somewhat contradictory, well describes the structural analogy with a transversal FIR filter.
[0050]
In an embodiment of the present invention, the order of a warped FIR filter may be significantly lower than the order of a comparable specification FIR filter. Thus, for a given circuit configuration, a warped FIR filter may provide better filter characteristics than an FIR filter. In addition, the warping parameter γ can be used as a transfer function, a control parameter that controls the positioning of the resonance and cut-off frequencies in the frequency spectrum, thereby producing an error which is the difference between the filter output signal and the desired signal. The spectrum of the signal e (n) can be minimized within the desired frequency range.
[0051]
In the FIR filter or the warped FIR filter, the next sample Y (t + T) is calculated according to the following equation.
(Equation 7)

It is.
[0052]
uIs an N-dimensional vector containing the latest N samples of the signal u,cIs a vector containing the N coefficients of the Nth order filter. T is a sampling period.
[0053]
In the above equation, u (t) is the actual value at the real time (current time) t, and u (t−iT) is the signal value i times before the real time t before the sampling period. In discrete-time systems, shorthand notation is often used. Here, the symbol u (i) represents the signal value at the time t-iT, that is, u (t-iT) in the above equation.
[0054]
For example, Paul S.A. published in 1997 by Crewer Academic Publishers. R. Referring to Dinitz's adaptive filtering (Adaptive Filtering by Paulo S. R. Diniz, Kluer Academic Publishers, 1997), the least mean square algorithm (least mean square) in which the following equation is used to update the filter coefficient in the adaptive filter is used. well known.
(Equation 8)

[0055]
Using the above shorthand notation (where n is the reference number of the actual sample), the above equation can be rewritten as:
(Equation 9)

[0056]
Or, further simplified and rewritten as follows:
(Equation 10)

Here, i indicates an individual vector element.
[0057]
It is preferable to use a leaky least mean square algorithm of the following equation for updating the filter coefficient.
(Equation 11)

Where u_iIs a set of signal values derived from the output signal of the digital processing device in the nth sampling period and the i-1 preceding sampling periods, and c_iIs a set of filter coefficients, e is the current value of the error signal, and λ and μ are scaling factors. The value of μ is generally 10^-6And the value of λ is generally about 0.99. λ is called leakage (leakage), and when λ <1, the filter coefficients have their respective initial values c_iDrift toward (0). μ is the convergence speed, which determines the speed at which the adaptive filter adapts to changes. This adaptation speed increases as the value of μ increases.
[0058]
Further, it is advantageous to normalize the algorithm so that the adaptive filter does not substantially respond to instantaneous dynamic changes in the input signal. Note that for estimating the acoustic feedback signal, the desired input signal is inappropriate and results in noise that degrades the convergence performance of the adaptive filter. The normalized algorithm of the following formula is called a normalized least mean square (nLMS) algorithm.
(Equation 12)

[0059]
However, in the above equation, a large processing power is required for calculating the power. Therefore, it is preferable to use the power estimation according to the following equation.
(Equation 13)

Where α is P_uA predetermined constant that determines the rate at which the estimate changes. This algorithm is called a power-normalized least mean square algorithm. Also, the power estimate can be based on the output signal from the input transducer so as to minimize the impact of the sudden change in the power of the input signal on the adaptive algorithm.
[0060]
Furthermore, the adaptive filter coefficients may be updated using a third update algorithm called a leaky sign least mean square algorithm such as the following equation.
[Equation 14]

Where μ_sIs the product of the sign of the e (n) signal and μ.
[0061]
Furthermore, a fourth update algorithm such as the following equation called a leaky code-sign least mean square algorithm may be used for the adaptive filter coefficients.
(Equation 15)

Here, sgn (u_i(N)) is u_iThis is the sign of (n).
[0062]
The filter coefficients may be updated based on, for example, the difference signal undergoing processing such as being combined with another signal, averaged, or otherwise filtered. Filtering may be performed in a focused manner as known in the art.
[0063]
It should also be noted that, in a multi-channel hearing aid according to the invention, the adaptive filters of the channels need not have the same number of taps. For example, it is desirable to include more taps in an adaptive filter operating in the low frequency channel.
[0064]
As described above, the controller can adjust λ and μ in response to the determination of the first parameter of the acoustic feedback loop of the hearing aid.
[0065]
Different sets of hearing aid parameters are set for each of the various types of sounds that the user desires to hear, for example speech or music, and for which the user is placed, for example, silence, noise, It can be provided for each of various acoustic environments such as reverberation, crowd, outdoor, indoor, and headset. For example, various gain settings may be provided as a function of the frequency, various gain settings may be provided as a function of the input signal level, and various convergence speeds may be provided as a function of the operation gain of the processing device. And so on. Each set of parameters defines a particular mode of operation of the hearing aid, and when the hearing aid operates with a particular set of parameters, it is said to operate in the corresponding mode. Thus, in certain operating modes, certain parameter values of the hearing aid are set in order to properly process the corresponding specific sound in a specific acoustic environment. Similarly, automatic adjustment of the parameters can be performed according to the current mode of operation.
[0066]
The type of sound is selected by the user or automatically by the hearing aid, for example by frequency analysis, analysis of the signal-to-noise ratio at various frequencies, acoustic dynamic analysis, language recognition, neural network recognition, etc. Can be detected.
[0067]
Similarly, the type of acoustic environment can be selected by the user or by means of, for example, frequency analysis, analysis of the signal-to-noise ratio at various frequencies, acoustic dynamic analysis, recognition by neural networks, etc. Can be detected automatically.
[0068]
For example, a user may want to listen to music. The first convergence speed of the first adaptive filter may then be set to a value that matches the automatic correlation of the music. Further, the gain adjustment or the adjustment of the first convergence speed may also be performed in accordance with the automatic correlation of music. If the first rate of convergence, eg, one or more scaling factors, is controlled as a function of the gain of the processing device, the function involves a particular sound, such as music or speech, that the user has decided to listen to. It can be selected from a set of functions adapted for use in a particular acoustic environment.
[0069]
Further, the adjustment may also be made according to the rate of change of the measured parameter, for example, the feedback gain of the acoustic return path.
[0070]
【Example】
Hereinafter, the present invention will be described in more detail with reference to the accompanying drawings.
[0071]
FIG. 1 is a schematic block diagram of an embodiment of the present invention. It will be obvious to those skilled in the art that the circuit shown in FIG. 1 can be implemented using digital or analog circuits or a combination thereof. In this embodiment, digital signal processing is employed, so that the processing unit (processor) 7 and the adaptive filter 10 are digital signal processing circuits. In this embodiment, all the digital circuits of the hearing aid are provided on a single digital signal processing chip or distributed in a suitable manner on a plurality of integrated circuit chips.
[0072]
In this hearing aid, an input transducer 1 such as a microphone is used to receive an audio signal (sound signal), which is converted into a corresponding electrical signal representing the received audio signal (received sound signal). The hearing aid may comprise a plurality of input transducers 1, which may for example provide a certain directional sensing characteristic. The input transducer 1 has a transfer function H_mHaving. The input transducer 1 converts an audio signal (sound signal) into an analog signal. This analog signal is sampled and digitized by an A / D converter (not shown), and becomes a digital signal 4 that is subjected to digital signal processing in a hearing aid. The digital signal 4 is sent to a coupling node 9 where it is combined with a feedback compensation signal 85 described below. The coupling node 9 outputs an output signal 86 which is applied to a digital signal processor (digital signal processor) 7 and amplified according to the desired frequency characteristics and the compression function to compensate for the hearing impairment of the user. An output signal 80 suitable for the operation is obtained.
[0073]
The output signal 80 is sent to the output transducer 5 and an optional (optional) delay Δ, and a delayed signal 83 is provided to the adaptive filter 10. The output transducer 5 converts the output signal 80 into an acoustic output signal 6. A portion of this acoustic signal has a transfer function H_fbTo the input transducer 1 along the return path Preferably, the time delay of the delay line Δ is substantially equal to the transit time of the signal 6 from the output transducer 5 to the input transducer 1. Other time delays may be selected. However, shorter or zero time delays complicate the filtering. For example, if the filter is a finite impulse response filter, a longer filter, ie, a filter with more taps, is required. To this end, a further delay is inserted into the circuit at the output of the processing device 7 to provide a delayed signal to the output transducer 5 and an optional delay Δ, whereby the input signal 4 and the filtered signal 85 May be reduced.
[0074]
In the adaptive filter 10, the delayed signal 83 is filtered to obtain a filtered signal 85, which is an estimate of the acoustic feedback. That is, the filtered signal 85 is an estimate of a portion of the transducer generated signal 4 that is generated by receiving sound from the output transducer 5. The filtered signal 85 is subtracted from the digital input signal 4 at the coupling node 9, resulting in a feedback compensation signal 86, which is input to the digital processing device 7. To compensate for changes in the acoustic feedback path, the filter coefficients of adaptive filter 10 are continuously updated such that filtered signal 85 remains substantially identical to feedback signal 6.
[0075]
The filter 10 is a finite impulse response (FIR) filter or a warped FIR filter having a leaky sign-sign least mean square algorithm as described above.
[0076]
The control device adjusts (adjusts) λ and μ according to the actual gain in the processing device 7. A graph of the scaling factors λ and μ as a function of gain is shown in FIG. Note that these functions depend on the mode of operation of the hearing aid. There is provided a selectable subset of functions as shown in FIG. 10 that can be selected by the controller 13 according to the current (current) mode of operation of the hearing aid. Further, these functions are selected according to the rate of change of the measured parameter, such as, for example, the attenuation in the acoustic return path.
[0077]
In the embodiment of FIG. 1, the control unit 13 receives information from the digital processing unit 7 via line 15. In accordance with the information received over line 15 regarding the current operating gain in digital processing device 7, the controller adjusts the adaptation rate of the filter coefficients of adaptive filter 10. Note that in this figure, dashed lines and arrows indicate control lines that do not form part of the signal path of the processed signal.
[0078]
An FIR filter, which is an embodiment of the filter 10, is shown in more detail in FIG. For simplicity, only the first four taps are shown, but the filter may include any suitable number of taps. When the operator Η is set to 1 and the operator Β is set to μ (e (n)), a leaky least mean square algorithm is achieved (realized). When λ is set to 1, a simple least mean square algorithm is achieved. When Η is set to 1 and Β is set to μsgn (e (n)), a leaky sign least squares algorithm is achieved (realized). Finally, Η is replaced by sgn (u_i(N)) and Β is set to μsgn (e (n)) to achieve (realize) a leaky sign-sign LMS algorithm. Also, the filter coefficients may be calculated using a recursive least squares algorithm.
[0079]
A warped FIR filter which is an embodiment of the filter 10 is shown in more detail in FIG. Note that the circuits below the upper delay line (line) in FIGS. 6 and 7 are identical. The warping parameter γ is preferably equal to 0.5. Note that if γ = 0, the warped FIR filter is replaced by an FIR filter.
[0080]
FIG. 8 shows a graph of the infinite impulse response of the warped FIR filter and the finite response of the FIR filter. From this graph, it can be seen that the warp-type FIR filter inherently obtains a value closer to the desired transfer function than the FIR filter.
[0081]
FIG. 9 shows the transfer function H of the adaptive filter 102 adapted to the desired transfer function H of the other filter 104._aIs shown in a block diagram of a test circuit 100 that determines The curve of the graph shows the power spectrum 110 of the error (error) signal 106 when the adaptive filter 102 is an FIR filter, and the power spectrum of the error (error) signal 106 when the adaptive filter 102 is a warped FIR filter. 108 is shown. The FIR filter and the warped FIR filter have the same number of tabs. It can be seen that below 6-7 kHz, the warped FIR filter improves the error signal to 15 dB. Since the output of the output transducer 5 typically has a cutoff frequency on the order of 6 to 8 kHz, the performance of the warped FIR filter above 8 kHz is not critical. Note that changing the sampling frequency changes the frequency values shown along the frequency axis. It should also be noted that gamma can be adjusted (tuned) to optimize the spectrum of error signal 106 for a particular application, such as a particular type of hearing impairment.
[0082]
FIG. 2 shows a multi-channel embodiment of a hearing aid according to the invention, in which each channel (channel) operates generally in the same way as the single-channel embodiment shown in FIG. I have. Except for the suffixes added to the reference numbers in FIG. 2, corresponding parts in FIGS. 1 and 2 are designated by the same reference numbers. For simplicity, only three channels are shown in FIG. However, it should be noted that the hearing aid may include any suitable number of channels as shown in the figure.
[0083]
The multi-channel embodiment of the invention according to FIG. 2 has the same parts as the single-channel embodiment shown in FIG. 1, but also includes band-pass filtered signals (band-filtered signals) 4a, 4i, 4n. It has a filter bank 3 for output. At coupling nodes 9a, 9i, 9n, respective signals 4a, 4i, 4n are combined to form respective signals 86a, 86i, 86n. These signals 86a, 86i, 86n are supplied to a multi-channel digital processing device 7 and processed according to the desired characteristics that match the hearing impairment of the user. This may involve adjusting different gain settings on individual channels. In addition, the processing may include compression functions. Furthermore, other functions, such as noise reduction, can be achieved by the signal processing device.
[0084]
The output signal from the digital signal processor 7 is sent to a filter bank 16 in which a band-pass filter signal (band-filtered) corresponding to different frequency bands or channels in a set of adaptive filters 10a, 10i, 10n. Signal 83a, 83i, 83n. Preferably, filter bank 16 comprises a digital fourth order filter.
[0085]
Filtered signals 85a, 85i, 85n are supplied from adaptive filters 10a, 10i, 10n to respective coupling nodes 9a, 9i, 9n and subtracted from signals 4a, 4i, 4n to obtain signals 86a, 86i, 86n. Is generated. As in the case of the embodiment of FIG. 1, the output signal 80 can be delayed by an arbitrary delay line Δ. Preferably, this delay is substantially equal to the maximum propagation time of the sound from the output transducer 5 to the input transducer 1.
[0086]
The processing device 7 combines the signals of the channels into a single output signal 80.
[0087]
In a multi-channel embodiment, the adaptation rates of each channel may be different from each other. Thus, higher adaptation speeds can be applied, resulting in undesirable distortion at frequencies where feedback resonance is likely to occur. This is an advantageous feature when the feedback resonance occurs at a frequency that is not important for the desired signal.
[0088]
Further, it is more difficult to perform signal detection in a wide frequency range. For this reason, convergence errors due to incorrect signal detection are less likely to occur in a multi-channel device than in a single-channel device.
[0089]
In one embodiment, the control device 13 determines the adaptive speed of the filter coefficients of the adaptive filters 10, 10a, 10i, and 10n as G₀To G_aIs controlled as a function of the actual operation gain in the processing device.
[0090]
The hearing aid illustrated in FIG. 3 corresponds to the hearing aid of FIG. 1 with an additional measuring device. Corresponding parts are designated by the same reference numerals and the description of the operation of these parts will not be repeated. The hearing aid shown in FIG. 3 further operates in parallel with the first adaptive filter 10, ie operates on the same signal as the first adaptive filter 10, but with the first of the first adaptive filter 10. And a second adaptive filter 11 that operates at a second convergence speed that is lower than the convergence rate. The output 85 of the second adaptive filter 11 is supplied to the coupling node 9 and is subtracted from the signal 4 to generate a signal 86 which is input to the processing device 7, whereby the acoustic feedback signal is processed by the processing device 7. Before being substantially removed from the signal. Note that the output 89 of the first adaptive filter 10 is not used to correct the input of the processing device.
[0091]
In this embodiment, the control device 13 estimates the amount of acoustic feedback by determining the parameters of the first adaptive filter 10. Due to the higher first convergence speed, the first adaptive filter 10 can track acoustic feedback more closely over time than the second adaptive filter 11. Further, the output signal 89 of the first adaptive filter 10 is not subtracted from the input transducer signal 4, so that the desired signal is not distorted by the first adaptive filter 10.
[0092]
The second adaptive filter 11 may be any type of adaptive filter, but is preferably an FIR filter using a power-normalized least mean square (power-nLMS) algorithm or a warped type. It is an FIR filter.
[0093]
The second adaptive filter 11 outputs the filtered signal 89 to a second combining node 12, where said signal is combined with the signal 86 from the first combining node 9. The output signal 90 from the coupling node 12 is input to the second adaptive filter 11, and the filter coefficient is adjusted.
[0094]
An important advantage of the embodiment shown in FIG. 3 is that the output signal generated by the first adaptive filter 10 is not provided in the main signal path from the input transducer 1 to the output transducer 5. This main signal path consists of an input transducer 1, digital conversion means (not shown), a coupling node 9, a digital processing device 7, and an output transducer 5. Therefore, the signal processing by the first adaptive filter 10 does not directly affect the signal in the main signal path. Therefore, no signal distortion of the signal in the main signal path is created by the first adaptive filter 10, and therefore the adaptation rate of the first adaptive filter 10 is reduced by the adaptive rate of the second adaptive filter 11. It can be substantially higher. Since the adaptive speed of the first adaptive filter 10 can be substantially higher than the adaptive speed of the second adaptive filter 11, the feedback path is made smaller by the first adaptive filter 10 than by the second adaptive filter 11. Can also be monitored much more closely over time. Preferably, the first adaptation speed is a fixed high adaptation speed, but this adaptation speed may be adjusted, for example, by correcting one or more scaling factors. For example, it may be preferable to adjust the adaptation speed of the first adaptive filter according to the actual gain or input power level in the processing device.
[0095]
Adjustment of the adaptation rate may be different for different modes of operation.
[0096]
If a rapid change in the acoustic environment occurs, the second adaptive filter 11 of FIG. 3 cannot immediately adapt to such a change and compensate for this change. Therefore, an uncompensated feedback signal starts to be generated. However, the first adaptive filter 10 is much faster than the second adaptive filter 11 and adapts to changes in the feedback path.
[0097]
In one embodiment, the controller adjusts the adaptation speed of the second adaptive filter 11 based on the quick response of the first adaptive filter 10 to changes in the feedback path, such as controlling the value of μ. Control. Therefore, when a change in the feedback path is indicated by the characteristic of the first adaptive filter 10, for example, a filtering characteristic such as attenuation, the second adaptive filter 11 is controlled accordingly. For example, when the gain is close to the feedback limit, the adaptation speed of the second adaptive filter 11 increases. By increasing the adaptation speed of the second adaptive filter 11, the second adaptive filter can compensate for changes in the acoustic feedback more quickly, for example, before the acoustic feedback leads to the generation of unwanted sound.
[0098]
The amount of acoustic feedback can preferably be estimated by determining the parameters of the first adaptive filter 10, or alternatively or additionally, by determining the parameters of the second adaptive filter 11. Please be careful. For example, since the ratio between the input signal and the output signal of each of the adaptive filters 10 and 11 is an element for estimating the attenuation of the feedback path including the acoustic feedback path, this ratio may be determined. Further, it is desirable to perform such calculation based on the averaged signal to suppress the influence of noise and speech and the convergence error. Alternatively, an average of the desired characteristics may be determined. Preferably, a power estimation of the type described above is used for each signal. Alternatively, one of the parameters of the adaptive filters 10 and 11 may be determined by appropriately converting the filter coefficients.
[0099]
In yet another embodiment, the control device reduces the gain in the digital processing device when a change in feedback is detected by the first adaptive filter 10. In particular, this may be selectively performed on different channels of the digital processing device.
[0100]
Based on the determination of the first parameter, the control unit determines that the maximum gain value G that the processing unit must not exceed in order to prevent the generation of unwanted audio (sound) signals._maxCan be calculated. In a multi-channel hearing aid, a separate G_maxThere may be values.
[0101]
In yet another embodiment, the control device is G₀To G_aChange the gain interval to. For this reason, when the second adaptive filter 11 detects that the apparatus is approaching unstable, the gain lower limit G is used by using this information.₀, Thereby reducing G_aIf it is desired to maintain at a particular level, the entire gain interval moves down or the gain interval is increased. Gain lower limit G₀Is preferably changed so that the λ and μ curves cover different intervals.
[0102]
In this regard, note that the relationship between gain and λ and μ differs from the function shown in FIG.
[0103]
FIG. 4 shows a multi-channel embodiment of a hearing aid according to the invention, in which each channel operates generally in the same manner as the single-channel embodiment shown in FIG. Corresponding parts in FIGS. 3 and 4 are designated by the same reference numerals, except that a suffix is added to the reference numerals in FIG. For simplicity, only three channels are shown in FIG. However, it should be noted that the hearing aid may include any suitable number of channels as shown in the figure. For simplicity, the control lines have been omitted from FIG.
[0104]
The multi-channel embodiment of the present invention according to FIG. 4 has the same components as the single-channel embodiment shown in FIG. 3, but also includes band-pass filter signals (band-filtered signals) 83a, 83i, 83n. To a second set of adaptive filters 11a, 11i, 11n. Each adaptive filter 11a, 11i, 11n provides a respective coupling node 12a, 12i, 12n with a filtered signal combined with a respective signal 86a, 86i, 86n from coupling node 9a, 9i, 9n.
[0105]
In the multi-channel embodiment shown in FIG. 4, a more detailed estimation of the transfer function of the feedback path is achieved. Further, signal processing may be performed at lower sampling frequencies in lower frequency bands, a technique known as decimation. Decimation is particularly simple in these filters because the output signal from the first set of adaptive filters is not fed into the main signal path, so no anti-aliasing filters are required in the system. Can be used for
[0106]
The embodiment shown in FIG. 4 can be controlled in the same way as the embodiment shown in FIG. However, in the embodiment shown in FIG. 4, the gain is selectively reduced in each individual channel and the adaptation speed of each individual adaptive filter of the set of second adaptive filters 11a, 11i, 11n is selected. It can be adjusted manually. For this reason, there is another advantage that the gain can be maintained at a high value at a frequency at which feedback resonance does not easily occur, and the distortion can be maintained at a low level.
[0107]
FIG. 5 shows an embodiment of a multiplex channel having a configuration similar to that of the embodiment shown in FIG. 4 and operating similarly. However, the embodiment shown in FIG. 5 has a second set of adaptive filters constituted by a single adaptive filter 11, and moreover because the joining node 9 is a single joining node. It is simple.
[0108]
Many other embodiments are obtained by varying the number of adaptive filters in the set of channels and first and second adaptive filters in the processing device. Also, the number of channels in the processing device may be different from the number of filters in the first set of adaptive filters, and similarly the number of filters is different from the number of filters in the second set of adaptive filters. You may.
[0109]
In particular, it is possible to use a digital signal processing device 7 having a relatively small number of channels and a second set of adaptive filters including more filters. Alternatively, the individual adaptive filters of the second set of adaptive filters may operate on combinations with channels in the digital signal processor 7, for example two or more of the digital signal processors 7. Are determined by a particular adaptive filter of the first set of adaptive filters._maxOr one channel of the digital signal processor 7 is the lowest gain of two or more gains determined by the adaptive filters of the first set of adaptive filters._maxMay be used. However, it is presently preferred that the embodiment comprise a single second adaptive filter 11 and a set of multi-channel first adaptive filters 10.
[0110]
FIG. 11 shows a graph of the operating gain as a function of frequency. The upper curve, shown as a solid line, shows the maximum operating gain obtained by the hearing aid according to the invention without undesirable sound generation, and the lower curve, shown as a dashed line, is a known curve. Shows the corresponding gain for a hearing aid.
[Brief description of the drawings]
FIG.
1 is a block diagram of a hearing aid according to the present invention.
FIG. 2
FIG. 2 is a block diagram of a multi-channel hearing aid in which each channel corresponds to the hearing aid shown in FIG. 1.
FIG. 3
1 is a block diagram of a hearing aid incorporating a measuring device according to the present invention.
FIG. 4
FIG. 4 is a block diagram of a multi-channel hearing aid in which each channel corresponds to the hearing aid shown in FIG. 3.
FIG. 5
FIG. 2 is a block diagram of a multi-channel hearing aid having a single band adaptive filter.
FIG. 6
FIG. 3 is a block diagram illustrating an LMS FIR filter that implements an update algorithm according to the present invention.
FIG. 7
FIG. 3 is a block diagram illustrating an LMS-type warp FIR filter that implements an update algorithm according to the present invention.
FIG. 8
4 is a graph showing a comparison between an impulse response of an FIR filter and an impulse response of a warped FIR filter.
FIG. 9
5 is a graph showing deviations of a FIR filter and a warped FIR filter from a desired transfer function.
FIG. 10
FIG. 4 is a diagram illustrating a variation of a filter coefficient depending on a gain in the digital processing device.
FIG. 11
FIG. 4 illustrates the improvement in maximum possible gain achieved by the present invention.

Claims (28)

An input transducer (1) for converting an acoustic input signal into a first electrical signal (4);
A first filter bank (3) having a bandpass filter for dividing said first electric signal (4) into a set of bandpass-filtered first electric signals (4 _i );
By individually processing each of the band-filtered first electrical signals (4 _i , 86) and adding the processed electrical signals to a second electrical signal (80), A processing device (7) for generating an electric signal (80) of
An output transducer (5) for converting the second electrical signal (80) into an acoustic output signal (6);
A second filter bank (16) having a plurality of bandpass filters for dividing said second electrical signal (80) into a set of bandpassed second electrical signals ( _80i );
A third electric signal (85) is generated by filtering the band-filtered second electric signal ( _80i ), and each of the third signals (85) is processed by the processing device (7). A first set of adaptive filters (10) having first filter coefficients for estimating acoustic feedback by adapting respective signals on the input side at respective first convergence rates;
Determining a first parameter of an acoustic feedback loop of the hearing aid and adjusting a second parameter of the hearing aid in response to the first parameter to compensate for acoustic feedback and substantially avoid generation of undesirable sound A control device to
Hearing aid with. The first of the at least one adaptive filter set (10) of the adaptive filter acts on the second electrical signal band-filtered are respective decimated (80 _i), the hearing aid of claim 1. Hearing aid according to claim 1 or 2, wherein the first filter bank (3) is constituted by a single bandpass filter. Hearing aid according to claim 1 or 3, wherein the second filter bank (16) is constituted by a single bandpass filter and the first set of adaptive filters is constituted by a single adaptive filter. . Hearing aid according to claim 1 or 2, wherein the plurality of bandpass filters of the second filter bank (16) are substantially identical to the respective bandpass filters of the first filter bank (3). Hearing aid according to claim 4, wherein the first set of adaptive filters filters the second electrical signal (80) and adapts to the first electrical signal (4). Further comprising a coupling node (9) for subtracting the third signal (85) from the first electrical signal (4), wherein the reduced signal is provided to the processing device (7). Item 7. A hearing aid according to item 6. Wherein the first set of adaptive filters with filtering each of the band filtered second electrical signals (80 _i), to adapt to each of the band filtered first electrical signals (4 _i) A hearing aid according to claim 5. A coupling node (9) for subtracting the third signal (85) from each of the band-filtered first electrical signals (4 _i ), wherein the reduced signal is coupled to the processing device ( Hearing aid according to claim 8, supplied to 7). A second adaptive filter (11) having a second filter coefficient for suppressing feedback in the hearing aid by filtering the second electrical signal (80) into a fourth electrical signal (85);
A fifth electric signal (86) is generated by subtracting the fourth electric signal (85) from the first electric signal (4), and the fifth electric signal (86) is converted to the first electric signal (86). A coupling node (9) that supplies each of said bandpass filters of a filter bank;
The hearing aid according to claim 6, wherein the second filter coefficient is updated at a second convergence speed lower than the first convergence speed. A second filter coefficient having a second filter coefficient for suppressing feedback in a hearing aid by filtering the band-passed second electrical signal (80 _i ) into the respective fourth electrical signal (85 _i ). A set of adaptive filters (11);
To generate a fifth electrical signal (86 _i) by subtracting from each of said first electrical signal fourth electrical signals (85 _i) is the band filter (4 _i), the fifth electric A coupling node (9) for supplying a signal (86 _i ) to the processing unit (7), wherein the second filter coefficient is updated at a second convergence speed lower than the first convergence speed. The hearing aid according to claim 8, wherein Hearing aid according to any of the preceding claims, wherein the first parameter is an operating gain of the processing device (7). A hearing aid according to any of the preceding claims, wherein the first parameter is a parameter of the first set of adaptive filters. The first parameter is the ratio of the magnitude of the signal (88) at the input of the first adaptive filter of the first set of adaptive filters (11) to the magnitude of the signal (89) at the corresponding output. The hearing aid according to claim 13, wherein A hearing aid according to any one of the preceding claims, wherein the second parameter is a gain of the processing device (7). The hearing aid according to claim 1, wherein the second parameter is the first convergence speed of the first filter coefficient. 17. A hearing aid according to any one of claims 12 to 16, dependent on claim 9 or 10, wherein the second parameter is the second convergence speed of the second filter coefficient. Leaky least mean square algorithm

Further comprising means for updating the filter coefficients according to
Here, c _i (n + 1) is the updated value of the i-th filter coefficient, c _i (n) is the current value of the i-th filter coefficient, and c _i (0) is the initial value of the i-th filter coefficient. U _i (n) is the (ni) th sample of the output signal of the processing device, e (n) is the current sample of the second electrical signal (86), and λ is 18. A hearing aid according to any of the preceding claims, wherein leakage is, μ is convergence, and λ and μ determine the first convergence speed. Normalized least mean square

Further comprising means for updating the filter coefficients according to
Where u (n) is an N-order vector containing the latest N samples of signal u, c (n) is a vector containing the current values of the N filter coefficients, and c (0) is N A vector containing initial values of the filter coefficients, c (n + 1) is an updated value of the N filter coefficients, and e (n) is a current sample of the second electrical signal (86). Item 18. A hearing aid according to any one of Items 1 to 17. Power-normalized least mean square algorithm

Further comprising means for updating the filter coefficients according to
18. The hearing aid according to claim 1, wherein α is a predetermined constant that determines the speed at which the _Pu estimation changes. Leaky code least mean square algorithm

Further comprising means for updating the filter coefficients according to
Here, c _i (n + 1) is the updated value of the i-th filter coefficient, c _i (n) is the current value of the i-th filter coefficient, and c _i (0) is the initial value of the i-th filter coefficient. U _i (n) is the (ni) th sample of the output signal of the processing device, e (n) is the current sample of the second electrical signal (86), and λ is 18. Leakage, μ is convergence, μ _s is the product of the sign of the e (n) signal and μ, and λ and μ determine the first convergence speed. A hearing aid according to one of the preceding claims. Leaky code-code least mean square algorithm

Further comprising means for updating the filter coefficients according to
Here, c _i (n + 1) is the updated value of the i-th filter coefficient, c _i (n) is the current value of the i-th filter coefficient, and c _i (0) is the initial value of the i-th filter coefficient. U _i (n) is the (ni) th sample of the output signal of the processing device, e (n) is the current sample of the second electrical signal (86), and λ is a leakage, mu is the convergence _{coefficient, sgn (u i (n)} ) _is the sign of _u i (n), _λ and mu determines the first convergence rate, claims 1-17 Hearing aid according to any one of the preceding claims. Hearing aid according to one of the preceding claims, wherein at least one of the first and second sets of adaptive filters (10, 11) comprises a finite impulse response filter. Hearing aid according to any of the preceding claims, wherein at least one of the first and second sets of adaptive filters (10, 11) comprises a warped finite impulse response filter. The hearing aid according to any one of claims 1 to 24, wherein the control device adjusts a second parameter of the hearing aid according to the first parameter and an actual acoustic environment. An input transducer (1) for converting an acoustic input signal into a first electrical signal (4);
A processing unit (7) for processing the first electric signal (4 _i , 86) to produce a second electric signal (80), thereby generating a second electric signal (80);
An output transducer (5) for converting the second electrical signal (80) into an acoustic output signal (6);
The second electrical signal (80) is filtered to generate a third electrical signal (85), and each of the third electrical signals (85) is converted to a respective one of an input side of the processing device (7). An adaptive filter (10) having filter coefficients for estimating acoustic feedback by adapting to the signal.
Hearing aid characterized in that the adaptive filter (10) is a warped adaptive filter. The hearing aid according to claim 26, wherein the warped filter is a warped FIR filter. In the method of suppressing acoustic feedback in hearing aids,
Converting the acoustic input signal into a first electrical signal (4);
Splitting the first electrical signal (4) into a set of band-filtered first electrical signals (4 _i );
Individually processing each of said band-filtered first electrical signals (4 _i , 86);
Adding the processed electrical signal to a second electrical signal (80);
Converting the second electrical signal (80) into an audio output signal (6);
Splitting the second electrical signal (80) into a set of band-passed second electrical signals ( _80i );
A third electric signal (85) is generated by adaptively filtering the band-filtered second electric signal (80 _i ), and the filtered signal (85) is input to a processing apparatus (7). Estimating the acoustic feedback by adapting the respective signals of at a respective first convergence rate;
Determining a first parameter of the acoustic feedback loop of the hearing aid;
Compensating for acoustic feedback by adjusting a second parameter of the hearing aid in response to said first parameter;
A method for suppressing acoustic feedback that substantially prevents the generation of undesirable sounds.

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