JP3701441B2 - Audio equipment - Google Patents

Description

により、適応フィルタの出力信号ｖ_w(n)の平均値Ｅ[ｖ_w ²(n)]を演算する。監視部７１ｃは該平均値Ｅ[ｖ_w ²(n)]が設定値以上になったか監視し、適応フィルタ係数が一定値に収束する前に、設定値以上になったとき、適応フィルタ係数の同定動作を停止するための動作停止信号ＡＬＭを出力する。すなわち、第１実施例は適応フィルタ出力が増大した時、異常状態発生と判断するものである。
動作停止信号ＡＬＭの発生により、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度適応フィルタ係数を同定するための調整を行う。
【００２９】
（Ｂ）本発明の第２実施例
図３は本発明の第２実施例の構成図であり、図１の第１実施例と同一部分には同一符号を付している。７０は異常検出部であり、７１ａは適応フィルタの出力信号ｖ_w(n)を取り込む適応フィルタ出力取得部、７１ｂは適応フィルタの出力信号ｖ_w(n)の平均値を演算する平均値演算部、７２ａはマイク出力ｖ_m(n)を取り込むマイク出力取得部、７２ｂはマイク出力ｖ_m(n)の平均値を演算する平均値演算部、７２ｃは上記２つの平均値の比βを演算する比演算部、７２ｄは前記比βが一定値以上になったか監視し、一定値以上になったとき適応フィルタ係数の同定動作を停止する信号ＡＬＭを出力する監視部、８０はオーディオ装置全体を制御するシステム制御部である。
【００３０】
第２実施例は、異常状態において、適応フィルタ出力が増大し、マイク出力が小さくなる点を考慮して異常状態を検出するものである。
適応フィルタ５６ｂの係数値を同定する調整モードにおいて、システムコントローラ８０は図示しないホワイトノイズ発生部から発生する定常ホワイトノイズｕ_n(n)を入力端子５１に入力する。適応信号処理装置５６は、ホワイトノイズが入力されるとエラー信号のパワーが最小となるように適応信号処理を行い、適応フィルタ５６ｂの係数を一定値に収束させ、該一定値をオーディオ信号入力時の適応フィルタの係数値として同定する。
かかる適応フィルタ係数の同定制御時、適応フィルタ出力取得部７１ａは適応フィルタの出力信号ｖ_w(n)を取り込み、平均値演算部７１ｂは(1)式により、適応フィルタの出力信号ｖ_w(n)の平均値Ｅ[ｖ_w ²(n)]を演算する。
【００３１】
又、マイク出力取得部７２ａはマイク出力ｖ_m(n)を取り込み、平均値演算部７２ｂは次式
【数２】

により、マイク出力信号ｖ_m(n)の平均値Ｅ[ｖ_m ²(n)]を演算する。
【００３２】
適応フィルタの出力信号の平均値Ｅ[ｖ_w ²(n)]及びマイク出力信号の平均値Ｅ[ｖ_m ²(n)]が求まれば、比演算部７２ｃはこれら２つの平均値の比βを次式
β＝Ｅ[ｖ_w ²(n)]／Ｅ[ｖ_m ²(n)] (3)
により演算する。監視部７２ｄは前記比βが設定値以上になったか監視し、適応フィルタ係数が一定値に収束する前に、設定値以上になったとき、適応フィルタ係数の同定動作を停止する動作停止信号ＡＬＭを出力する。
これにより、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度適応フィルタ係数を同定するための調整を行う。
【００３３】
（Ｃ）第３実施例
図４は本発明の第３実施例の構成図であり、図１の第１実施例と同一部分には同一符号を付している。７０は異常検出部であり、７１ａは適応フィルタの出力信号ｖ_w(n)を取り込む適応フィルタ出力取得部、７１ｂは適応フィルタの出力信号ｖ_w(n)の平均値を演算する平均値演算部、７３ａは定常ホワイトノイズｕ(n)を取り込むノイズ取得部、７３ｂは定常ホワイトノイズ信号ｕ(n)の平均値を演算する平均値演算部、７３ｃは上記２つの平均値の比αを演算する比演算部、７３ｄは前記比αが一定値以上になったか監視し、一定値以上になったとき適応フィルタ係数の同定動作を停止する信号ＡＬＭを出力する監視部、８０はオーディオ装置全体を制御するシステム制御部である。
【００３４】
第３実施例は、異常状態において、適応フィルタ出力が増大する点を考慮して異常状態を検出するものである。
適応フィルタ５６ｂの係数値を同定する調整モードにおいて、システムコントローラ８０は図示しないホワイトノイズ発生部から発生する定常ホワイトノイズｕ_n(n)を入力端子５１に入力する。適応信号処理装置５６は、ホワイトノイズが入力されるとエラー信号のパワーが最小となるように適応信号処理を行い、適応フィルタ５６ｂの係数を一定値に収束させ、該一定値をオーディオ信号入力時の適応フィルタの係数値として同定する。かかる適応フィルタ係数の同定制御時、適応フィルタ出力取得部７１ａは適応フィルタの出力信号ｖ_w(n)を取り込み、平均値演算部７１ｂは(1)式により、適応フィルタの出力信号ｖ_w(n)の平均値Ｅ[ｖ_w ²(n)]を演算する。
【００３５】
又、ノイズ取得部７３ａは定常ホワイトノイズｕ(n)を取り込み、平均値演算部７３ｂは次式
【数３】

により、ホワイトノイズｕ(n)の平均値Ｅ[ｕ²(n)]を演算する。
【００３６】
適応フィルタの出力信号の平均値Ｅ[ｖ_w ²(n)]及びホワイトノイズの平均値Ｅ[ｕ²(n)]が求まれば、比演算部７３ｃはこれら２つの平均値の比αを次式
α＝Ｅ[ｖ_w ²(n)]／Ｅ[ｕ²(n)] (5)
により演算する。監視部７３ｄは比αが設定値以上になったか監視し、適応フィルタ係数が一定値に収束する前に、設定値以上になったとき、適応フィルタ係数の同定動作を停止する動作停止信号ＡＬＭを出力する。
これにより、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度適応フィルタ係数を同定するための調整を行う。
【００３７】
（Ｄ）第４実施例
図５は本発明の第４実施例の構成図であり、図１の第１実施例と同一部分には同一符号を付している。７４ａは、フィルタ係数監視部であり、適応フィルタの各係数値ａ₀〜ａ_nを監視し、少なくとも２つの係数値が最大(＝１)になった時、適応フィルタ係数の同定動作を停止する信号ＡＬＭを出力する監視部、８０はオーディオ装置全体を制御するシステム制御部である。第４実施例は、異常状態において、適応フィルタ係数が増大する点を考慮して異常状態を検出するものである。
【００３８】
適応フィルタ５６ｂの係数値を同定する調整モードにおいて、システムコントローラ８０は図示しないホワイトノイズ発生部から発生する定常ホワイトノイズｕ_n(n)を入力端子５１に入力する。適応信号処理装置５６は、ホワイトノイズが入力されるとエラー信号のパワーが最小となるように適応信号処理を行い、適応フィルタ５６ｂの係数を一定値に収束させ、該一定値をオーディオ信号入力時の適応フィルタの係数値として同定する。
【００３９】
かかる適応フィルタ係数の同定制御時、フィルタ係数監視部７４ａは適応フィルタの各係数値ａ₀〜ａ_nを監視し、少なくとも２つの係数値の絶対値が１になったか監視する。すなわち、フィルタ係数監視部７４ａは
｜ａ_m｜＝１ （０≦ｍ≦ｎ）
｜ａ_i｜＝１ （０≦i≦ｎ, ｉ≠ｍ）
が満たされたかチェックし、適応フィルタ係数が一定値に収束する前に、上式が満たされれば、適応フィルタ係数の同定動作を停止する動作停止信号ＡＬＭを出力する。これにより、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度適応フィルタ係数を同定するための調整を行う。
【００４０】
（Ｅ）第５実施例
（ａ）構成
適応イコライザ機能を備えたオーディオ装置（図１）において、適応信号処理部５６ａはエラー信号ｅ(n)と信号処理フィルタ５６ｃを介して入力される適応信号処理用参照信号ｒ(n)を用いて観測点におけるオーディオ信号が目標信号と等しくなるように適応信号処理（ＬＭＳ適応アルゴリズムに従った処理）を行って適応フィルタ５６ｂの係数を同定する。信号処理フィルタ５６ｃは制御音源 ５８から観測点までの伝搬特性を入力信号に畳み込んで適応信号処理用参照信号ｒ(n)を発生するものである。従って、適応イコライザ制御機能を備えたオーディオ装置では、適応フィルタの係数を同定する前に、信号処理用フィルタ５６ｂの係数、すなわち、制御音源から観測点までの伝搬特性を同定する必要がある。
【００４１】
図６はオーディオ装置の接続を変えて、制御音源５８から観測点までの伝搬特性を同定できるように、かつ、該伝搬特性の同定制御時にオーディオ装置の異常を検出できるようにした第５実施例の構成図である。
図中、５１は信号入力端子であり、伝搬特性の同定制御時に定常ホワイトノイズｕ(n)が入力される。５３は車室内音響空間の聴取位置(観測点)における音を検出するマイク、５４は検出されたオーディオ信号を増幅するマイクアンプ、５５はマイク出力ｖ_m(n)と適応フィルタの出力信号ｖ_w(n)との誤差e(n)を演算する演算部、５６は前記誤差ｅ(n)のパワーが最小となるように信号ｖ_w(n)を発生する適応信号処理装置であり、適応信号処理部（ＬＭＳ）５６ａと適応フィルタ５６ｂで構成されている。５７は入力信号を増幅するパワーアンプ、５８は入力信号に応じた音を車室内音響空間に放射する制御音源であるスピーカ（１つしか示してないが図１４で説明したように２つ存在する場合もある）、５９はローパスフィルタ（ＬＰＦ）で、特定帯域例えば２００Ｈｚ以下の低音域（制御帯域）の信号成分を通過して適応信号処理装置５６及びパワーアンプ５７に入力するものである。
【００４２】
９０は異常検出部であり、９１ａはマイク出力信号ｖ_m(n)を取り込むマイク出力取得部、９１ｂは(2)式によりマイク出力信号ｖ_m(n)の平均値を演算する平均値演算部、９１ｃは該平均値が一定値以下になったか監視し、一定値以下になったとき伝搬特性の同定動作を停止する信号ＡＬＭを出力する監視部、８０はオーディオ装置全体を制御するシステム制御部である。
【００４３】
（ｂ）伝搬特性の同定制御
制御音源から観測点までの伝搬特性を同定するために、システムコントローラ８０は定常ホワイトノイズｕ(n)をローパスフィルタ５９に入力する。このホワイトノイズはパワーアンプ５７を介して制御音源であるスピーカ５８に入力し、音響空間に放射される。観測点のマイク５３はホワイトノイズを検出し、演算部５５は検出信号ｖ_m(n)と適応フィルタ出力信号ｖ_w(n)との差分をエラー信号ｅ(n)として出力する。適応信号処理部（ＬＭＳ）５６ａはエラー信号ｅ(n)と定常ホワイトノイズｕ(n)を用いて適応信号処理を行って適応フィルタ５６ｂの係数を決定する。
【００４４】
適応フィルタ５６ｂは決定された係数に従ってホワイトノイズｕ(n)にデジタルフィルタ処理を施して信号ｖ_w(n)を出力する。かかるＬＭＳ適応信号処理を継続して実行するとエラー信号ｅ(n)のパワーが最小となるように適応フィルタ５６ｂの係数が所定値に収束する。これにより、制御音源であるスピーカ５８からマイク５３位置までの制御帯域（200Hz以下）の信号伝搬系の伝搬特性が適応フィルタ５６ｂに設定されたことになる（伝搬特性の同定）。従って、この適応フィルタ５６ｂの係数値を保存し、第１〜第４実施例の適応フィルタ係数の同定制御時に信号処理用フィルタ５６ｃに設定する。
【００４５】
（ｃ）異常検出原理
図７は異常検出の原理説明図である。図１４で説明したように、２つの制御音源用スピーカ５８ａ，５８ｂの接続が逆相になっていると、▲１▼マイク５３に到達する時点で互いに打ち消しあってしまい、マイク出力は小さい値となる。▲２▼このため、適応信号処理装置５６は誤差ｅが小さくなるように適応フィルタの係数値を絶対値が減小する方向に更新する。
以上は２つの制御音源を逆相接続した場合であるが、１つの制御音源のみが存在し、該制御音源とパワーアンプ間の接続線が断線した場合も同様である。すなわち、制御音源とパワーアンプ間が接続されないと、制御音源から音がでず、マイク出力が零になる、このため、適応信号処理装置５６は誤差ｅが小さくなるように適応フィルタの係数値を絶対値が減小する方向に更新する。以上より、逆相接続、信号線の断線等の異常が発生すると、
Ａ．適応フィルタ係数が小さくなり、更に、
Ｂ．適応フィルタ出力が減小する。または、
Ｃ．マイク出力が小い。
従って、これらを考慮してオーディオ装置の異常を検出することができる。
【００４６】
（ｄ）異常検出制御
伝搬特性の同定制御時、マイク出力取得部９１ａはマイク出力ｖ_m(n)を取り込み、平均値演算部９１ｂは(2)式によりマイク出力信号ｖ_m(n)の平均値Ｅ[ｖ_m ²(n)]を演算する。監視部９１ｃは該平均値が一定値以下になったか監視し、一定値以下になったとき伝搬特性の同定動作を停止する信号ＡＬＭを出力する。すなわち、上記Ｃ．の基準に従って異常状態を検出する。
動作停止信号の発生により、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度伝搬特性を同定するための調整を行う。
【００４７】
（Ｆ）第６実施例
図８は本発明の第６実施例の構成図であり、図６の第５実施例と同一部分には同一符号を付している。９０は異常検出部であり、９２ａは適応フィルタの出力信号ｖ_w(n)を取り込む適応フィルタ出力取得部、９２ｂは適応フィルタの出力信号ｖ_w(n)の平均値を演算する平均値演算部、９２ｃは該平均値が一定値以下になったか監視し、一定値以下になったとき伝搬特性の同定動作を停止する信号ＡＬＭを出力する監視部，８０はオーディオ装置全体を制御するシステム制御部である。
【００４８】
伝搬特性の同定制御時、適応フィルタ出力取得部９２ａは適応フィルタの出力信号ｖ_w(n)を取り込み、平均値演算部９２ｂは(1)式により適応フィルタ出力信号ｖ_w(n)の平均値Ｅ[ｖ_w ²(n)]を演算する。監視部９２ｃは該平均値が一定値以下になったか監視し、一定値以下になったとき伝搬特性の同定動作を停止する信号ＡＬＭを出力する。すなわち、上記Ｂ．の基準に従って異常状態を検出する。
動作停止信号の発生により、システム制御部８０は同定動作を停止し、図示しない異常検出ランプを点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度伝搬特性を同定するための調整を行う。
【００４９】
（Ｇ）第７実施例
図９は本発明の第７実施例の構成図であり、図６の第５実施例と同一部分には同一符号を付している。９０は異常検出部であり、９３ａは適応フィルタ５６ｂの各係数値の二乗和（ノルム）を演算するノルム演算部、９３ｄは二乗和が一定値より大きくなったか監視し、一定値より大きくならない場合には音響伝搬特性の同定動作を停止する信号ＡＬＭを出力する監視部、８０はオーディオ装置全体を制御するシステム制御部である。
【００５０】
伝搬特性の同定制御時、ノルム演算部９３ａは次式
Ｎω(n)＝ａ₀ ²＋ａ₁ ²＋ａ₂ ²＋・・・ａ_n ² (6)
により、各係数値の二乗和（ノルム）を演算し、監視部９３ｂはノルムが一定値以上になったか監視し、所定時間経過しても一定値以上にならないとき伝搬特性の同定動作を停止する信号ＡＬＭを出力する。すなわち、上記Ａ．の基準に従って異常状態を検出する。
動作停止信号の発生により、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度適応フィルタ係数を同定するための調整を行う。
【００５１】
（Ｈ）第８実施例
図１０は本発明の第８実施例の構成図であり、図６の第５実施例と同一部分には同一符号を付している。９０は異常検出部であり、９４ａはホワイトノイズ ｕ(n)と観測点におけるオーディオ信号ｖ_m(n)との相互相関値を演算する相互相関演算部、９４ｂは相互相関値のピーク値を算出するピーク値算出部、９４ｃは相互相関のピーク値が一定値に以上になったか監視し、一定値以上にならなければ音響伝搬特性の同定動作を停止する信号を出力する監視部、８０はオーディオ装置全体を制御するシステム制御部である。
オーディオ装置部分に異常がなければ、同定制御により適応フィルタ５６ｂの係数値は制御音源からマイクまでの伝搬特性を次第に表現するようになる。かかる場合、定常ホワイトノイズｕ(n)とマイク検出信号ｖ_m(n)の相関度が強くなり、相互相関のピーク値は設定値以上になる。
【００５２】
そこで、 伝搬特性の同定制御時、相互相関演算部９４ａは次式
【数４】

により、ホワイトノイズｕ(n)とマイク検出信号ｖ_m(n)の相互相関値を演算する。(7)式において、ｍはシフトしたサンプリング数である。この(7)式は信号ｕ(n)と信号ｖ_m(n+m)の対応する値の積を演算してその総和を相関関数値とするものであり、両信号が時間的に一致するとき相関関数値が最大になる。相互相関演算部９４ａは信号 ｖ_m(n)を１サンプリングづつ順にずらしてゆき、その都度、(7)式により両信号波形の相関関数値を演算する。ピーク値算出部９４ｂは相互相関値のピーク値を算出し、監視部９４ｃは相互相関のピーク値が設定値以上になったか監視し、所定時間経過しても設定値以上にならなければ、伝搬特性の同定動作を停止する信号ＡＬＭを出力する。
動作停止信号の発生により、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度適応フィルタ係数を同定するための調整を行う。
【００５３】
（Ｉ）第９実施例
図１１は本発明の第９実施例の構成図であり、図１０の第８実施例と同一部分には同一符号を付している。９０は異常検出部であり、９４ａ′はホワイトノイズｕ(n)と適応フィルタ出力信号ｖ_w(n)との相互相関値を演算する相互相関演算部、９４ｂ′は相互相関値のピーク値を算出するピーク値算出部、９４ｃ′は相互相関のピーク値が一定値に以上になったか監視し、一定値以上にならなければ音響伝搬特性の同定動作を停止する信号を出力する監視部、８０はオーディオ装置全体を制御するシステム制御部である。
オーディオ装置部分に異常がなければ、同定制御により適応フィルタ５６ｂの係数値は制御音源からマイクまでの伝搬特性を次第に表現するようになる。かかる場合、定常ホワイトノイズｕ(n)と適応フィルタ出力信号ｖ_w(n)の相関度が強くなり、相互相関のピーク値は設定値以上になる。
【００５４】
そこで、 伝搬特性の同定制御時、相互相関演算部９４ａ′は次式
【数５】

により、ホワイトノイズｕ(n)と適応フィルタ出力信号ｖ_w(n)の相互相関値を演算する。(8)式において、ｍはシフトしたサンプリング数である。この(8)式は信号ｕ(n)と信号ｖ_w(n+m)の対応する値の積を演算してその総和を相関関数値とするものであり、両信号が時間的に一致するとき相関関数値が最大になる。相互相関演算部９４ａ′は信号 ｖ_w(n)を１サンプリングづつ順にずらしてゆき、その都度、(8)式により両信号波形の相関関数値を演算する。ピーク値算出部９４ｂ′は相互相関値のピーク値を算出し、監視部９４ｃ′は相互相関のピーク値が設定値以上になったか監視し、所定時間経過しても設定値以上にならなければ、伝搬特性の同定動作を停止する信号ＡＬＭを出力する。
【００５５】
動作停止信号の発生により、システム制御部８０は同定動作を停止し、図示しない異常検出ランプ等を点灯する。作業者は誤接続、断線、接続忘れ等のオーディオ装置の異常を検出し、異常除去後、再度適応フィルタ係数を同定するための調整を行う。
以上、本発明を実施例により説明したが、本発明は請求の範囲に記載した本発明の主旨に従い種々の変形が可能であり、本発明はこれらを排除するものではない。
【００５６】
【発明の効果】
以上本発明によれば、適応イコライザ制御機能を備えたオーディオ装置において、誤接続、断線、接続忘れ等の異常状態を調整時に事前に検出でき、オーディオ装置が故障する事態を防止することができる。
【図面の簡単な説明】
【図１】第１実施例の構成図である。
【図２】異常時の状態説明図である。
【図３】第２実施例の構成図である。
【図４】第３実施例の構成図である。
【図５】第４実施例の構成図である。
【図６】第５実施例の構成図である。
【図７】異常時の状態説明図である。
【図８】第６実施例の構成図である。
【図９】第７実施例の構成図である。
【図１０】第８実施例の構成図である。
【図１１】第９実施例の構成図である。
【図１２】適応等化システムの基本構成図である。
【図１３】特定帯域のみターゲットにした適応イコライザの構成図である。
【図１４】正相接続と逆相接続の制御音源説明図である。
【符号の説明】
５２・・目標応答設定部
５３・・マイク
５４・・マイクアンプ
５５・・演算部
５６・・適応信号処理装置
５６ａ・・適応信号処理部（ＬＭＳ）
５６ｂ・・適応フィルタ
５６ｃ・・信号処理用フィルタ
５７・・パワーアンプ
５８・・制御音源である第１のスピーカ
５９・・ローパスフィルタ（ＬＰＦ）
６０・・非制御音源である第２のスピーカ
７０・・異常検出部
７１ａ・・適応フィルタ出力取得部
７１ｂ・・平均値演算部
７１ｃ・・監視部
８０・・システム制御部[0001]
[Industrial application fields]
The present invention relates to an audio apparatus, and more particularly, to an audio apparatus having an equalizer control function for controlling an audio signal in a specific band to be a target response.
[0002]
[Prior art]
The vehicle interior is a sealed narrow space. Therefore, since reflection occurs in a short time and sound waves interfere with each other, the propagation characteristics up to the listening point are very complicated. In addition, since music or the like is being listened to in an asymmetrical place, the propagation characteristics from the left and right speakers are also greatly different. There is a demand for an audio device that eliminates the adverse effects of the vehicle interior and improves the acoustic characteristics of the vehicle interior. For this reason, control has been proposed that uses an adaptive equalizer to achieve desired characteristics including amplitude and phase characteristics at a plurality of points (control points) in the reproduction space.
[0003]
FIG. 12 is a basic configuration diagram of an adaptive equalization system, where 1 is an audio source (tuner, tape deck, CD player, etc.) that outputs an audio signal x (n), and 2 is a target response characteristic (impulse response characteristic) H. A target response setting unit that is set and receives the audio signal x (n) and outputs the target signal d (n), 4 is a microphone that detects sound at the listening position (observation point) in the vehicle interior acoustic space, and 5 is detection A calculation unit for calculating an error e (n) between the audio signal d ^ (n) and the target signal d (n) output from the target response setting unit 2, and a power of the error e (n) is minimum An adaptive signal processing device 7 that generates a signal y (n) so as to become a speaker (control sound source) that radiates sound corresponding to the signal y (n) to the vehicle interior acoustic space 8.
[0004]
The target response characteristic H of the target response setting unit 2 is set to a characteristic corresponding to the sound field space to be reproduced. For example, when a delay time that is about half of the number of taps of the adaptive filter is t, a flat characteristic (gain 1 characteristic) having the delay time t is set in the entire audio frequency band. The delay time t is for the adaptive filter to accurately approximate the inverse characteristic of the acoustic system. The target response setting unit 2 having such a target response characteristic is an FIR (Finite Impulse Response) type digital This can be realized by setting the tap coefficient corresponding to the delay time t of the filter to 1 and setting the other tap coefficients to 0.
[0005]
The adaptive signal processing device 6 is input with the audio signal x (n) as a reference signal and the error signal e (n) output from the arithmetic unit 5, and the power of the error signal is minimized. The adaptive signal processing is performed so that the signal y (n) is output. The adaptive signal processing device 6 includes an adaptive signal processing unit (LMS) 6a, an adaptive filter 6b having a digital filter configuration of FIR type, and an audio signal x (n) propagation characteristics of an acoustic propagation system from a speaker 7 to a listening position ( It has a filter 6c that generates a reference signal (filtered reference signal) r (n) to be used for adaptive signal processing by convolving the transfer function C ^.
[0006]
The adaptive signal processing unit 6a receives the error signal e (n) at the listening position and the reference signal r (n) for adaptive signal processing input through the signal processing filter 6c, and uses these signals for audio at the listening position. Adaptive signal processing is performed so that the signal d ^ (n) is equal to the target signal d (n), and the coefficient of the adaptive filter 6b is determined. For example, the adaptive signal processing unit 6a determines the coefficient of the adaptive filter 6b according to a known LMS (Least Mean Square) adaptive algorithm so that the power of the error signal e (n) is minimized. The adaptive filter 6b performs digital filter processing on the audio signal x (n) according to the coefficient determined by the adaptive signal processing unit 6a and outputs a signal y (n). Therefore, if the coefficient of the adaptive filter 6b converges so that the power of the error signal e (n) is minimized by the adaptive signal processing, the audio signal d ^ (n) is equal to the target signal d (n) at the listening position. Thus, it is possible to obtain the same effect as listening to a sound in an ideal space having the propagation characteristic H (frequency characteristic is flat) set in the target response setting unit 2.
[0007]
In the adaptive equalization system, if the entire band is controlled, a huge amount of calculation is required, and there is a problem that several tens of DSPs are required if the processing is performed in real time. Therefore, an adaptive equalizer that targets only the low frequency range has been proposed in order to improve the reproduction quality of a specific frequency band, for example, 200 Hz or lower.
[0008]
FIG. 13 is a configuration diagram of such an adaptive equalizer, and the same components as those in FIG. 12 are denoted by the same reference numerals. The difference from FIG. 12 is that (1) a band-pass filter (LPF) 9 that passes the low frequency band is provided after the audio source 1 and the output is input to the target response setting unit 2 and the adaptive signal processing device 6. (2) Provided with a second speaker (uncontrolled sound source) 10 that receives audio signals in the entire band and radiates audio sound into the vehicle interior acoustic space 8; (3) Preceding the second speaker 10 Is provided with a delay device 11 for delaying the audio signal, the signal delay time from the output end of the audio source 1 to the microphone 4 via the first speaker 7 is t, and the signal delay from the second speaker 10 to the microphone 4 is When the time is td, the time Δt (= t−td) is set in the delay device 11 and the audio signal of the entire band is delayed by Δt and input to the second speaker 10.
According to this adaptive equalizer, the adaptive filter 6b becomes an inverse filter of the reproduction system (propagation characteristics from the speaker 7 to the microphone position) only in the low sound range (specific band) limited by the low-pass filter 9, and the target signal d ( n) to work. Then, the amount of calculation can be reduced by controlling only the specific band as compared with the adaptive equalization system of FIG.
[0009]
[Problems to be solved by the invention]
By the way, in the conventional audio system provided with the adaptive equalizer control function, even if there is an erroneous connection such as reverse phase, forgetting connection, disconnection, etc., these abnormalities cannot be recognized and abnormal operations may occur. When such an abnormal operation occurs, it may lead to a rapid increase in the speaker output signal, and consequently to a system failure.
For example, if the first speaker 7 that is the control sound source and the adaptive filter 6b are not connected, the error signal e becomes large. Therefore, in order to reduce the error signal, adaptive signal processing is performed so that the output of the adaptive filter is increased, and the coefficient value of the adaptive filter is updated in the direction in which the absolute value increases. However, since there is no sound from the first speaker 7, the microphone output does not increase. Therefore, the error signal does not change, and in response to this, the adaptive filter updates the filter coefficient in a direction in which the output further increases. If such an operation is repeated, the output of the adaptive filter continues to increase, and the system does not operate normally. In the worst case, the power amplifier (not shown, but provided between the adaptive filter and the speaker) is destroyed.
If there is not enough power, there may be a total of two speakers for the control sound source, one on each side. In this case, as shown in FIG. 14, if the control sound sources (speakers) 7a and 7b are mistakenly connected in the normal phase and the reverse phase, the control sound sources 7a and 7b are arranged at substantially the same distance from the observation point (microphone 4). Therefore, when the outputs from the respective control sound sources reach the microphone 4, they cancel each other.
[0010]
For this reason, the microphone output becomes a small value, and the error signal e with the target response becomes large. Therefore, in order to reduce the error signal, adaptive signal processing is performed so that the output of the adaptive filter is increased, and the coefficient value of the adaptive filter is updated in the direction in which the absolute value increases. However, since the control sound source output is canceled near the microphone, the microphone output does not increase. Therefore, the error signal does not change so much, and the adaptive filter updates the filter coefficient in the direction in which the output further increases in response to this. If such an operation is repeated, the output of the control sound source continues to increase, and the system does not operate normally. In the worst case, the control sound source and the power amplifier are destroyed.
As described above, an object of the present invention is to detect an abnormal state in an audio apparatus provided with an adaptive equalizer and prevent a situation in which the audio apparatus breaks down.
[0011]
[Means for Solving the Problems]
  According to the present invention, there is provided a filter that passes a specific band component of an audio signal, target signal output means that outputs a target signal, and an error signal that is a difference between the target signal and an audio signal detected at an observation point. An output means, an audio signal and an error signal are input, adaptive signal processing is performed so that the power of the error signal is minimized, an adaptive filter coefficient is determined, and the adaptive filter is used to filter an audio signal in a specific band. An adaptive signal processing device that outputs the audio signal output from the adaptive filter, a control speaker that radiates audio sound to an acoustic space, and an audio signal in a band other than the specific band is input It is equipped with an uncontrolled speaker that radiates audio sound into the acoustic space, and audio signals at observation points in a specific band. A There audio device for controlling so as to match the target signal,When the stationary white noise is input to the filter and the coefficient value of the adaptive filter is identified, the average value of the output signal of the adaptive filter Ew Average value calculation unit that calculates the average value of white noise at the observation point Em An average value calculation unit for calculating the ratio of the two average values Ew / Em A ratio calculating unit that calculates whether the ratio exceeds a certain value, and a monitoring unit that outputs a signal for stopping the identification operation of the adaptive filter coefficient when the ratio exceeds a certain value, an identification operation based on the stop signal And a control unit for notifying the abnormality of the audio device,This is achieved by an audio device comprising
[0020]
  According to the present invention, the above problem isSteady white noise is input to the filter, a specific band component output from the filter is input to a control sound source and an adaptive filter, an output of the adaptive filter is set as a target signal, and white noise detected at the target signal and an observation point When adaptive processing is performed so that the power of the error signal, which is the difference between the two, is minimized and the coefficient value of the adaptive filter is converged to identify the acoustic propagation characteristics from the control sound source to the observation point, white noise and the adaptive filter output An arithmetic unit that calculates the cross-correlation value with the signal, a peak value calculation unit that calculates the peak value of the cross-correlation value, and monitors whether the peak value exceeds a certain value. A monitoring unit for outputting a signal for stopping the identification operation, a control unit for subsequently stopping the identification operation based on the stop signal, and for notifying the abnormality of the audio device,This is achieved by an audio device comprising
[0021]
DETAILED DESCRIPTION OF THE INVENTION
(A) First embodiment of the present invention
(A) Configuration
FIG. 1 is a block diagram of a first embodiment of the present invention. In the figure, 51 is a signal input terminal, and stationary white noise u (n) is inputted at the time of adjustment for identifying the coefficient value of the adaptive filter, and from the audio source (tuner, tape deck, CD player, etc.) other than at the time of adjustment. An audio signal is input. 52 is a target response setting unit that sets a target response characteristic (impulse response) h and outputs a target signal d (n), 53 is a microphone that detects sound at the listening position (observation point) in the vehicle interior acoustic space, and 54 is A microphone amplifier for amplifying the detected audio signal, 55 is a microphone output v_mAn arithmetic unit 56 that calculates an error e (n) between (n) and the target signal d (n) output from the target response setting unit 52, and a signal v so that the power of the error e (n) is minimized._wis an adaptive signal processing device that generates (n), and includes an adaptive signal processing unit (LMS) 56a, an adaptive filter 56b, and a signal processing filter 56c that generates a reference signal r (n) used for adaptive signal processing. .
[0022]
57 is an output signal v of the adaptive filter 56b._wA power amplifier for amplifying (n), 58 is an output signal v of the adaptive filter_wA first speaker which is a control sound source that radiates sound according to (n) to the vehicle interior acoustic space (only one is shown, but there may be two as described in FIG. 14), 59 is a low-pass A filter (LPF) that passes a signal component in a low frequency range (control band) of a specific band, for example, 200 Hz or less, and inputs it to the adaptive signal processing device 56. A second speaker 61, which is a non-control sound source that radiates into the indoor space acoustic space, is a power amplifier that amplifies an audio signal in the entire band.
Note that an audio signal having a band of 200 Hz or more may be input to the second speaker 60. The output of the low-pass filter 59 can also be input to the signal processing filter 56c.
[0023]
Reference numeral 70 denotes an abnormality detection unit, and reference numeral 71a denotes an output signal v of the adaptive filter._wAn adaptive filter output acquisition unit for capturing (n), 71b is an output signal v of the adaptive filter_wAn average value calculation unit 71c that calculates the average value of (n) monitors whether the average value exceeds a certain value, and outputs a signal ALM that stops the adaptive filter coefficient identification operation when the average value exceeds the certain value. A monitoring unit 80 is a system control unit that controls the entire audio apparatus.
[0024]
(B) Principle of abnormality detection
FIG. 2 is a diagram for explaining the principle of abnormality detection.
As described with reference to FIG. 14, when the connection between the two control sound source speakers 57a and 57b is reversed, (1) when they reach the microphone 4, they cancel each other, and the microphone output becomes a small value. The error e from the target response becomes large. Therefore, (2) in order to reduce the error e, the adaptive signal processing unit updates the absolute value of the coefficient value of the adaptive filter in the direction of increasing. However, since the control sound source output is canceled near the microphone, the microphone output does not increase. Therefore, (3) The error e does not change so much, and the adaptive filter updates the filter coefficient in the direction in which the output further increases in response to this. If such an operation is repeated, the control sound source output continues to increase and does not operate normally. In the worst case, the control sound source and the power amplifier are destroyed.
[0025]
The above is the case where two control sound sources are connected in reverse phase, but the same applies when only one control sound source exists and the connection between the control sound source and the power amplifier is broken. That is, if the control sound source and the power amplifier are not connected, no sound is produced from the control sound source, and the microphone output becomes zero. For this reason, the adaptive filter updates the filter coefficient in the direction in which the output increases. Such an operation is repeated, the filter output continues to increase, does not operate normally, and in the worst case, the power amplifier is destroyed.
[0026]
From the above, when abnormalities such as reverse phase connection, disconnection of signal line between power amplifier and control sound source, forget connection, etc. occur,
A. The adaptive filter coefficient increases,
B. The adaptive filter output increases, or
C. Microphone output is small.
Therefore, it is possible to detect an abnormality of the audio device in consideration of these.
[0027]
(C) Control
In an audio apparatus having an adaptive equalizer control function, the coefficient value of the adaptive filter 56b is identified (adjustment mode) before inputting an audio signal. When the audio signal is input (audio mode), the identified coefficient value is fixedly set in the adaptive filter, and the adaptive filter coefficient is not updated by the adaptive signal processing.
Now, in the adjustment mode, the system controller 80 is a stationary white noise (noise having a flat characteristic over the entire audio signal) u generated from a white noise generator (not shown) u._n(n) is input to the input terminal 51.
The adaptive signal processing device 56 performs adaptive signal processing so that the power of the error signal is minimized, converges the coefficient of the adaptive filter 56b to a constant value, and identifies the constant value as the coefficient value of the adaptive filter when the audio signal is input. To do.
[0028]
During the filter coefficient identification control, the adaptive filter output acquisition unit 71a outputs the output signal v of the adaptive filter._w(n) is taken in, and the average value calculation unit 71b
[Expression 1]

Thus, the output signal v of the adaptive filter_wAverage value E [v of (n)_w ²(n)] is calculated. The monitoring unit 71c calculates the average value E [v_w ²(n)] is monitored to see if it exceeds the set value. When the adaptive filter coefficient exceeds the set value before it converges to a certain value, an operation stop signal ALM for stopping the adaptive filter coefficient identification operation is displayed. Output. That is, the first embodiment determines that an abnormal state has occurred when the adaptive filter output increases.
When the operation stop signal ALM is generated, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the adaptive filter coefficient again.
[0029]
(B) Second embodiment of the present invention
FIG. 3 is a block diagram of the second embodiment of the present invention. The same reference numerals are given to the same parts as those of the first embodiment of FIG. Reference numeral 70 denotes an abnormality detection unit, and reference numeral 71a denotes an output signal v of the adaptive filter._wAn adaptive filter output acquisition unit for capturing (n), 71b is an output signal v of the adaptive filter_wAn average value calculation unit 72a for calculating the average value of (n), 72a is a microphone output v_mA microphone output acquisition unit for capturing (n), 72b is a microphone output v_m(n) an average value calculating unit for calculating the average value, 72c is a ratio calculating unit for calculating the ratio β of the two average values, and 72d monitors whether the ratio β is equal to or greater than a certain value. A monitoring unit 80 for outputting a signal ALM for stopping the adaptive filter coefficient identification operation when it becomes, and a system control unit 80 for controlling the entire audio apparatus.
[0030]
In the second embodiment, the abnormal state is detected in consideration of the point that the adaptive filter output increases and the microphone output decreases in the abnormal state.
In the adjustment mode for identifying the coefficient value of the adaptive filter 56b, the system controller 80 causes the steady white noise u generated from a white noise generator (not shown)._n(n) is input to the input terminal 51. The adaptive signal processing device 56 performs adaptive signal processing so that the power of the error signal is minimized when white noise is input, converges the coefficient of the adaptive filter 56b to a constant value, and the constant value is input to the audio signal. Are identified as the coefficient values of the adaptive filter.
During the adaptive filter coefficient identification control, the adaptive filter output acquisition unit 71a outputs the output signal v of the adaptive filter._w(n) is taken in, and the average value calculation unit 71b calculates the output signal v of the adaptive filter by the equation (1)_wAverage value E [v of (n)_w ²(n)] is calculated.
[0031]
In addition, the microphone output acquisition unit 72 a_m(n) is taken in, and the average value calculation unit 72b
[Expression 2]

By the microphone output signal v_mAverage value E [v of (n)_m ²(n)] is calculated.
[0032]
Average value E [v of output signal of adaptive filter_w ²(n)] and the average value E [v of the microphone output signal_m ²(n)] is obtained, the ratio calculation unit 72c calculates the ratio β of these two average values as
β = E [v_w ²(n)] / E [v_m ²(n)] (3)
Calculate by The monitoring unit 72d monitors whether the ratio β is equal to or higher than a set value, and when the adaptive filter coefficient becomes equal to or higher than the set value before the adaptive filter coefficient converges to a certain value, the operation stop signal ALM that stops the identification operation of the adaptive filter coefficient. Is output.
As a result, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the adaptive filter coefficient again.
[0033]
(C) Third embodiment
FIG. 4 is a block diagram of the third embodiment of the present invention. The same reference numerals are given to the same parts as those of the first embodiment of FIG. Reference numeral 70 denotes an abnormality detection unit, and reference numeral 71a denotes an output signal v of the adaptive filter._wAn adaptive filter output acquisition unit for capturing (n), 71b is an output signal v of the adaptive filter_wan average value calculation unit for calculating the average value of (n), 73a is a noise acquisition unit for capturing the steady white noise u (n), 73b is an average value calculation unit for calculating the average value of the steady white noise signal u (n), 73c is a ratio calculation unit for calculating the ratio α of the two average values, 73d is a signal ALM that monitors whether the ratio α is equal to or greater than a certain value and stops the identification operation of the adaptive filter coefficient when the ratio α is equal to or greater than the certain value. , 80 is a system control unit for controlling the entire audio apparatus.
[0034]
In the third embodiment, the abnormal state is detected in consideration of the point that the adaptive filter output increases in the abnormal state.
In the adjustment mode for identifying the coefficient value of the adaptive filter 56b, the system controller 80 causes the steady white noise u generated from a white noise generator (not shown)._n(n) is input to the input terminal 51. The adaptive signal processing device 56 performs adaptive signal processing so that the power of the error signal is minimized when white noise is input, converges the coefficient of the adaptive filter 56b to a constant value, and the constant value is input to the audio signal. Are identified as the coefficient values of the adaptive filter. During the adaptive filter coefficient identification control, the adaptive filter output acquisition unit 71a outputs the output signal v of the adaptive filter._w(n) is taken in, and the average value calculation unit 71b calculates the output signal v of the adaptive filter by the equation (1)_wAverage value E [v of (n)_w ²(n)] is calculated.
[0035]
The noise acquisition unit 73a takes in the stationary white noise u (n), and the average value calculation unit 73b
[Equation 3]

The average value E [u of white noise u (n)²(n)] is calculated.
[0036]
Average value E [v of output signal of adaptive filter_w ²(n)] and the average value of white noise E [u²If (n)] is obtained, the ratio calculation unit 73c calculates the ratio α of these two average values as follows:
α = E [v_w ²(n)] / E [u²(n)] (5)
Calculate by The monitoring unit 73d monitors whether the ratio α is equal to or greater than a set value, and outputs an operation stop signal ALM that stops the identification operation of the adaptive filter coefficient when the ratio becomes equal to or greater than the set value before the adaptive filter coefficient converges to a certain value. Output.
As a result, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the adaptive filter coefficient again.
[0037]
(D) Fourth embodiment
FIG. 5 is a block diagram of the fourth embodiment of the present invention. The same reference numerals are given to the same parts as those of the first embodiment of FIG. 74a is a filter coefficient monitoring unit, and each coefficient value a of the adaptive filter₀~ A_nIs a monitoring unit that outputs a signal ALM that stops the identification operation of the adaptive filter coefficient when at least two coefficient values become maximum (= 1), and 80 is a system control unit that controls the entire audio apparatus. . In the fourth embodiment, the abnormal state is detected in consideration of the point that the adaptive filter coefficient increases in the abnormal state.
[0038]
In the adjustment mode for identifying the coefficient value of the adaptive filter 56b, the system controller 80 causes the steady white noise u generated from a white noise generator (not shown)._n(n) is input to the input terminal 51. The adaptive signal processing device 56 performs adaptive signal processing so that the power of the error signal is minimized when white noise is input, converges the coefficient of the adaptive filter 56b to a constant value, and the constant value is input to the audio signal. Are identified as the coefficient values of the adaptive filter.
[0039]
During such adaptive filter coefficient identification control, the filter coefficient monitoring unit 74a uses each coefficient value a of the adaptive filter.₀~ A_nAnd monitor whether the absolute value of at least two coefficient values has become 1. That is, the filter coefficient monitoring unit 74a
| A_m| = 1 (0 ≦ m ≦ n)
| A_i| = 1 (0 ≦ i ≦ n, i ≠ m)
Is satisfied, and if the above equation is satisfied before the adaptive filter coefficient converges to a constant value, an operation stop signal ALM for stopping the adaptive filter coefficient identification operation is output. As a result, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the adaptive filter coefficient again.
[0040]
(E) Fifth embodiment
(A) Configuration
In the audio apparatus (FIG. 1) having the adaptive equalizer function, the adaptive signal processing unit 56a uses the error signal e (n) and the adaptive signal processing reference signal r (n) input through the signal processing filter 56c. Adaptive signal processing (processing according to the LMS adaptive algorithm) is performed so that the audio signal at the observation point becomes equal to the target signal, and the coefficient of the adaptive filter 56b is identified. The signal processing filter 56c convolves the input signal with the propagation characteristic from the control sound source 58 to the observation point to generate an adaptive signal processing reference signal r (n). Therefore, in an audio apparatus having an adaptive equalizer control function, it is necessary to identify the coefficient of the signal processing filter 56b, that is, the propagation characteristic from the control sound source to the observation point, before identifying the coefficient of the adaptive filter.
[0041]
FIG. 6 shows a fifth embodiment in which the connection of the audio device is changed so that the propagation characteristic from the control sound source 58 to the observation point can be identified, and the abnormality of the audio device can be detected during the identification control of the propagation characteristic. FIG.
In the figure, 51 is a signal input terminal to which steady white noise u (n) is input during propagation characteristic identification control. 53 is a microphone that detects sound at the listening position (observation point) in the vehicle interior acoustic space, 54 is a microphone amplifier that amplifies the detected audio signal, and 55 is a microphone output v._m(n) and output signal v of the adaptive filter_wa calculation unit 56 for calculating an error e (n) with respect to (n), a signal v so that the power of the error e (n) is minimized;_wThis is an adaptive signal processing device that generates (n), and includes an adaptive signal processing unit (LMS) 56a and an adaptive filter 56b. 57 is a power amplifier that amplifies the input signal, and 58 is a speaker that is a control sound source that radiates sound according to the input signal to the vehicle interior acoustic space (only one speaker is shown, but there are two as described in FIG. 14). 59 is a low-pass filter (LPF) that passes a signal component in a low frequency range (control band) of a specific band, for example, 200 Hz or less, and inputs the signal component to the adaptive signal processing device 56 and the power amplifier 57.
[0042]
90 is an abnormality detection unit, 91a is a microphone output signal v_mA microphone output acquisition unit for capturing (n), 91b is a microphone output signal v according to equation (2)._mAn average value calculation unit 91c that calculates the average value of (n) monitors whether the average value is below a certain value, and outputs a signal ALM that stops the identification operation of the propagation characteristics when it falls below a certain value A system control unit 80 controls the entire audio apparatus.
[0043]
(B) Propagation characteristic identification control
In order to identify the propagation characteristic from the control sound source to the observation point, the system controller 80 inputs the stationary white noise u (n) to the low-pass filter 59. The white noise is input to the speaker 58, which is a control sound source, via the power amplifier 57, and is radiated to the acoustic space. The microphone 53 at the observation point detects white noise, and the calculation unit 55 detects the detection signal v._m(n) and adaptive filter output signal v_wThe difference from (n) is output as an error signal e (n). The adaptive signal processing unit (LMS) 56a performs adaptive signal processing using the error signal e (n) and the stationary white noise u (n) to determine the coefficient of the adaptive filter 56b.
[0044]
The adaptive filter 56b performs digital filter processing on the white noise u (n) according to the determined coefficient, and outputs the signal v_wOutput (n). When such LMS adaptive signal processing is continuously executed, the coefficient of the adaptive filter 56b converges to a predetermined value so that the power of the error signal e (n) is minimized. As a result, the propagation characteristic of the signal propagation system in the control band (200 Hz or less) from the speaker 58 as the control sound source to the position of the microphone 53 is set in the adaptive filter 56b (identification of propagation characteristic). Therefore, the coefficient value of the adaptive filter 56b is stored and set in the signal processing filter 56c during the adaptive filter coefficient identification control of the first to fourth embodiments.
[0045]
(C) Abnormality detection principle
FIG. 7 is an explanatory diagram of the principle of abnormality detection. As described with reference to FIG. 14, if the connection between the two control sound source speakers 58a and 58b is reversed, (1) when they reach the microphone 53, they cancel each other, and the microphone output is a small value. Become. (2) For this reason, the adaptive signal processing device 56 updates the coefficient value of the adaptive filter so that the absolute value decreases so that the error e becomes small.
The above is the case where two control sound sources are connected in reverse phase, but the same applies when only one control sound source exists and the connection line between the control sound source and the power amplifier is disconnected. That is, if the control sound source and the power amplifier are not connected, no sound is output from the control sound source, and the microphone output becomes zero. Therefore, the adaptive signal processing device 56 sets the coefficient value of the adaptive filter so that the error e becomes small. Update in the direction of decreasing absolute value. From the above, when an abnormality such as reverse phase connection or signal line disconnection occurs,
A. The adaptive filter coefficients are reduced, and
B. Adaptive filter output decreases. Or
C. Microphone output is small.
Therefore, it is possible to detect an abnormality of the audio device in consideration of these.
[0046]
(D) Abnormality detection control
During the propagation characteristic identification control, the microphone output acquisition unit 91a performs the microphone output v._m(n) is taken in, and the average value calculation unit 91b calculates the microphone output signal v according to the equation (2)._mAverage value E [v of (n)_m ²(n)] is calculated. The monitoring unit 91c monitors whether the average value is equal to or less than a certain value, and outputs a signal ALM that stops the propagation characteristic identification operation when the average value is equal to or less than the certain value. That is, the C.I. An abnormal condition is detected according to the criteria.
When the operation stop signal is generated, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the propagation characteristic again.
[0047]
(F) Sixth embodiment
FIG. 8 is a block diagram of the sixth embodiment of the present invention. The same reference numerals are assigned to the same parts as those of the fifth embodiment of FIG. Reference numeral 90 denotes an abnormality detection unit, and 92a denotes an output signal v of the adaptive filter_wAn adaptive filter output acquisition unit for capturing (n), 92b is an output signal v of the adaptive filter_wAn average value calculation unit 92c that calculates the average value of (n) monitors whether the average value is below a certain value, and outputs a signal ALM that stops the identification operation of the propagation characteristics when it falls below a certain value Reference numeral 80 denotes a system control unit that controls the entire audio apparatus.
[0048]
During the propagation characteristic identification control, the adaptive filter output acquisition unit 92a outputs the output signal v of the adaptive filter._w(n) is taken in, and the average value calculation unit 92b calculates the adaptive filter output signal v by the equation (1)._wAverage value E [v of (n)_w ²(n)] is calculated. The monitoring unit 92c monitors whether the average value has become a certain value or less, and outputs a signal ALM for stopping the propagation characteristic identification operation when the average value has become the certain value or less. That is, the B. An abnormal condition is detected according to the criteria.
When the operation stop signal is generated, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the propagation characteristic again.
[0049]
(G) Seventh embodiment
FIG. 9 is a block diagram of the seventh embodiment of the present invention. The same reference numerals are assigned to the same parts as those of the fifth embodiment of FIG. 90 is an abnormality detection unit, 93a is a norm calculation unit that calculates the square sum (norm) of each coefficient value of the adaptive filter 56b, and 93d monitors whether the sum of squares is larger than a certain value. Is a monitoring unit that outputs a signal ALM for stopping the identification operation of acoustic propagation characteristics, and 80 is a system control unit that controls the entire audio apparatus.
[0050]
During identification control of propagation characteristics, the norm calculation unit 93a is
Nω (n) = a₀ ²+ A₁ ²+ A₂ ²+ ... a_n ²(6)
Thus, the square sum (norm) of each coefficient value is calculated, and the monitoring unit 93b monitors whether the norm is equal to or greater than a certain value, and stops the identification operation of the propagation characteristic when the norm does not become equal to or greater than a certain value even after a predetermined time. The signal ALM is output. That is, the A. above. An abnormal condition is detected according to the criteria.
When the operation stop signal is generated, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the adaptive filter coefficient again.
[0051]
(H) Eighth embodiment
FIG. 10 is a block diagram of an eighth embodiment of the present invention. The same reference numerals are given to the same parts as those of the fifth embodiment of FIG. Reference numeral 90 denotes an anomaly detection unit, and 94a denotes white noise u (n) and an audio signal v at the observation point._ma cross-correlation calculating unit that calculates a cross-correlation value with (n), 94b is a peak value calculating unit that calculates the peak value of the cross-correlation value, and 94c monitors whether the peak value of the cross-correlation has reached a certain value or more, A monitoring unit 80 for outputting a signal for stopping the identification operation of the acoustic propagation characteristics unless it exceeds a certain value, and a system control unit 80 for controlling the entire audio apparatus.
If there is no abnormality in the audio apparatus part, the coefficient value of the adaptive filter 56b gradually expresses the propagation characteristic from the control sound source to the microphone by the identification control. In such a case, stationary white noise u (n) and microphone detection signal v_mThe degree of correlation (n) becomes stronger, and the peak value of the cross-correlation is greater than or equal to the set value.
[0052]
Therefore, during the propagation characteristic identification control, the cross-correlation calculating unit 94a
[Expression 4]

White noise u (n) and microphone detection signal v_mThe cross-correlation value of (n) is calculated. In equation (7), m is the number of shifted samples. This equation (7) is for signal u (n) and signal v_mThe product of the corresponding values of (n + m) is calculated and the sum is used as the correlation function value, and the correlation function value is maximized when both signals coincide in time. The cross-correlation calculation unit 94a receives the signal v_m(n) is shifted in order of one sample at a time, and each time the correlation function value of both signal waveforms is calculated by equation (7). The peak value calculation unit 94b calculates the peak value of the cross-correlation value, and the monitoring unit 94c monitors whether or not the cross-correlation peak value exceeds the set value. A signal ALM for stopping the characteristic identification operation is output.
When the operation stop signal is generated, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the adaptive filter coefficient again.
[0053]
(I) Ninth embodiment
FIG. 11 is a block diagram of the ninth embodiment of the present invention. Components identical with those of the eighth embodiment shown in FIG. 10 are designated by like reference characters. Reference numeral 90 denotes an abnormality detection unit, and 94a ′ denotes white noise u (n) and an adaptive filter output signal v._wa cross-correlation calculation unit for calculating a cross-correlation value with (n), 94b 'is a peak value calculation unit for calculating the peak value of the cross-correlation value, and 94c' is used to monitor whether the cross-correlation peak value has reached a certain value If the value does not exceed a certain value, the monitoring unit 80 outputs a signal for stopping the acoustic propagation characteristic identification operation, and 80 is a system control unit for controlling the entire audio apparatus.
If there is no abnormality in the audio apparatus part, the coefficient value of the adaptive filter 56b gradually expresses the propagation characteristic from the control sound source to the microphone by the identification control. In such a case, stationary white noise u (n) and adaptive filter output signal v_wThe degree of correlation (n) becomes stronger, and the peak value of the cross-correlation is greater than or equal to the set value.
[0054]
Therefore, during propagation characteristic identification control, the cross-correlation calculation unit 94a '
[Equation 5]

White noise u (n) and adaptive filter output signal v_wThe cross-correlation value of (n) is calculated. In equation (8), m is the shifted sampling number. This equation (8) is for signal u (n) and signal v_wThe product of the corresponding values of (n + m) is calculated and the sum is used as the correlation function value, and the correlation function value is maximized when both signals coincide in time. The cross-correlation calculation unit 94a '_w(n) is shifted in order of one sample at a time, and each time, the correlation function value of both signal waveforms is calculated by equation (8). The peak value calculation unit 94b ′ calculates the peak value of the cross-correlation value, and the monitoring unit 94c ′ monitors whether the cross-correlation peak value is equal to or higher than the set value. The signal ALM for stopping the identification operation of the propagation characteristics is output.
[0055]
When the operation stop signal is generated, the system control unit 80 stops the identification operation and turns on an abnormality detection lamp (not shown). The operator detects an abnormality of the audio device such as erroneous connection, disconnection, or forgetting connection, and after removing the abnormality, performs adjustment for identifying the adaptive filter coefficient again.
The present invention has been described with reference to the embodiments. However, the present invention can be variously modified in accordance with the gist of the present invention described in the claims, and the present invention does not exclude these.
[0056]
【The invention's effect】
As described above, according to the present invention, in an audio apparatus having an adaptive equalizer control function, an abnormal state such as erroneous connection, disconnection, or forgetting connection can be detected in advance during adjustment, and a situation in which the audio apparatus fails can be prevented.
[Brief description of the drawings]
FIG. 1 is a configuration diagram of a first embodiment.
FIG. 2 is a diagram illustrating a state at the time of abnormality.
FIG. 3 is a configuration diagram of a second embodiment.
FIG. 4 is a configuration diagram of a third embodiment.
FIG. 5 is a configuration diagram of a fourth embodiment.
FIG. 6 is a configuration diagram of a fifth embodiment.
FIG. 7 is an explanatory diagram of a state at the time of abnormality.
FIG. 8 is a configuration diagram of a sixth embodiment.
FIG. 9 is a block diagram of a seventh embodiment.
FIG. 10 is a configuration diagram of an eighth embodiment.
FIG. 11 is a configuration diagram of the ninth embodiment.
FIG. 12 is a basic configuration diagram of an adaptive equalization system.
FIG. 13 is a configuration diagram of an adaptive equalizer that targets only a specific band.
FIG. 14 is an explanatory diagram of control sound sources for normal phase connection and reverse phase connection;
[Explanation of symbols]
52 .. Target response setting part
53. Mike
54 .. Microphone amplifier
55 .. Operation part
56 .. Adaptive signal processing device
56a ・ ・ Adaptive signal processor (LMS)
56b ・ ・ Adaptive filter
56c .. Filter for signal processing
57. Power amplifier
58..First speaker as control sound source
59 .. Low pass filter (LPF)
60 .. Second speaker as non-control sound source
70 .. Abnormality detection part
71a..Adaptive filter output acquisition unit
71b .. Average value calculator
71c ・ ・ Monitoring part
80 .. System control part

Claims (2)

A filter that passes a specific band component of the audio signal, target signal output means for outputting the target signal, means for outputting an error signal that is a difference between the target signal and the audio signal detected at the observation point, and an audio signal in the specific band and is input to said error signal, the power of the error signal is subjected to adaptive signal processing to minimize to determine the coefficients of the adaptive filter, outputs performs a filtering process on the audio signal of the specific band by the adaptive filter An adaptive signal processing unit that receives the audio signal output from the adaptive filter and radiates the audio sound to an acoustic space; receives an audio signal in the entire band or a band other than the specific band; comprising a non-control speaker for radiating to a space, an audio signal at the observation point of the specific bands An audio device for controlling so as to match the target signal,
Stationary white noise is input to the filter which passes through the specific band component in identifying coefficient value of said adaptive filter, the average value calculator for calculating an average value Ew of the output signal of the adaptive filter,
An average value calculator that calculates the average value Em of white noise at the observation point,
A ratio calculation unit for calculating a ratio Ew / Em of the two average values;
A monitoring unit that monitors whether the ratio has become a certain value or more, and outputs a signal for stopping the identification operation of the adaptive filter coefficient when the ratio has become a certain value or more,
A control unit that subsequently stops the identification operation based on the stop signal, and notifies the abnormality of the audio device,
An audio apparatus comprising: The difference of the filter, the control speaker to by an audio sound input audio signal of a specific band which is output from the filter to emit the acoustic space, the target signal and the detected audio signal at the observation point that pass through a particular band component of the audio signal means for outputting an error signal is, the inputted audio signal and the error signal in a specific band, adaptive signal power of said error signal to determine the coefficients of the adaptive filter by performing adaptive signal processing so as to minimize processing In an audio device comprising a unit,
Steady white noise is input to the filter, a specific band component output from the filter is input to the control speaker and an adaptive filter, and the adaptive filter output is set as a target signal, and the white noise detected at the target signal and the observation point When the acoustic propagation characteristics from the control speaker to the observation point are identified by performing the adaptive processing so that the power of the error signal that is the difference between the two and the adaptive filter is converged to converge the coefficient value of the adaptive filter,
Calculator for calculating a cross-correlation value between the white noise and the adaptive filter output signal,
A peak value calculation unit for calculating the peak value of the cross-correlation value,
A monitoring unit that monitors whether the peak value becomes a certain value or more, and outputs a signal that stops the identification operation of the acoustic propagation characteristics if not exceeding the certain value,
A control unit that subsequently stops the identification operation based on the stop signal, and notifies the abnormality of the audio device,
An audio apparatus comprising:

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